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From: Colin Perkins <csp@csperkins.org>
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] DECISION: Default mechanism to map RTP data to m- line is based on PT?
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On 29 Jun 2013, at 16:44, Paul Kyzivat wrote:
> On 6/29/13 8:44 AM, Colin Perkins wrote:
>> On 26 Jun 2013, at 13:20, Christer Holmberg wrote:
>>> Hi Colin,
>>>=20
>>>>> Emil suggested that the default, "MTI", mechanism for mapping RTP =
data to m- lines should be based on PT. Applications are
>>>>> allowed to use whatever other mechanisms, but usage of such =
mechanisms must be negotiated (or, applications need to have
>>>>> some other means knowing that the other endpoint support such =
mechanisms).
>>>>>=20
>>>>> Q3: Do we need to specify a default, MTI, mechanism for mapping =
RTP data to m- lines?
>>>>=20
>>>> I think the BUNDLE specification needs to define how RTP flows can =
be associated with m=3D lines, but ought not mandate that applications
>>>> do so. The document that describes how WebRTC uses BUNDLE might =
want to mandate that WebRTC applications use the mechanism.
>>>=20
>>> Note that there is a separate discussion, BUNDLE DISCUSION: Always =
mandate mechanism to map received data to m- line?, on whether BUNDLE =
shall mandate that the receiver must always be able to map received data =
to an m- line.
>>>=20
>>> So far only Paul has commented on that discussion, but I assume it's =
just a question of hours before your input will be posted? :)
>>>=20
>>>>> Q4: If Q3, do we mandate applications to support, and use (unless =
applications are made aware of other mechanisms supported by all =
endpoints) PT for mapping received RTP media?
>>>>=20
>>>> I think we should specify how the combination of PT values and =
a=3Dssrc: lines can be used to associate RTP data with m=3D lines, for =
applications that want to do so.
>>>>=20
>>>> I don't think just using PT values is sufficient, since the PT =
space is small (I can imagine applications that want to send lots of =
flows using
>>>> the same codec configuration, and it's more natural to give them =
all the same PT and use a=3Dssrc: lines to distinguish).
>>>=20
>>> If you run out of PTs, and have to re-use the same PT in multiple m- =
lines (for the same codec configuration, though), you will obviously =
need something else for mapping RTP to m- lines, and we should probably =
state that.
>>>=20
>>> But, my take from your reply is that you don't think we should =
specify a default mechanism. We CAN describe different mechanisms (PT, =
PT+SSRC etc), but it would be up to applications to decide what to use. =
Correct?
>>=20
>>=20
>> No.
>>=20
>> Specifying a "default" suggests that there might be alternative ways =
of doing this. If we specify an algorithm for mapping RTP flows to m=3D =
lines, we should specify a single algorithm. We don't need multiple ways =
of doing this. I think that single algorithm should look at the PT and =
the a=3Dssrc: lines. Just looking at the PT isn't sufficient.
>=20
> But we *will* have alternative ways of doing this:
>=20
> - one way is by SSRC, specifying which SSRC goes to which m-line
>=20
> - another way could be by appid (from Roni's new draft)
>=20
> The only thing that is special about PT is that *every* RTP media =
section contains PTs, so it is always available to be used.

Every RTP packet contains an SSRC, so that's always available to use =
too. You may not chose to use it, but it seems strange to ignore it.

Colin

> If I want to use appid, I may very well *not* want to use ssrc as =
well. I may want the same ssrc to associate with *different* m-lines in =
the bundle depending on which appid it has.
>=20
> And in that case I may well not want to use PT either.
>=20
> 	Thanks,
> 	Paul
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic



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




From ron.even.tlv@gmail.com  Mon Jul  1 04:40:36 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Paul Kyzivat'" <pkyzivat@alum.mit.edu>
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Date: Mon, 1 Jul 2013 14:38:59 +0300
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Subject: Re: [MMUSIC] FW: New Version Notification	for	draft-even-mmusic-application-token-00.txt
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: 30 June, 2013 7:03 PM
> To: Roni Even
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] FW: New Version Notification for draft-even-
> mmusic-application-token-00.txt
> 
> On 6/30/13 1:29 AM, Roni Even wrote:
> > Inline
> >
> >> -----Original Message-----
> >> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >> Behalf Of Paul Kyzivat
> >> Sent: 29 June, 2013 6:59 PM
> >> To: mmusic@ietf.org
> >> Subject: Re: [MMUSIC] FW: New Version Notification for draft-even-
> >> mmusic-application-token-00.txt
> >>
> >> I looked this over, and it seems to solve many of the issues at hand.
> >> There are no doubt some nits to iron out, but we should get agreement
> >> over the concepts before worrying too much about the details.
> >>
> >> A couple of things I wanted to comment on:
> >>
> >> You give examples of form:
> >>         a=ssrc:11111 AppID:1
> >> But you don't say too much about the meaning of this. The reason I
> >> ask is that you say elsewhere that the appid can move from one ssrc to
> another.
> >> My question is: If this form is used, can that appid later be bound
> >> to a different ssrc via the header extension? Also, is the following
legal?
> >>         a=ssrc:11111,22222 AppID:1
> > [Roni Even] The AppID can be mapped later using SDES or header
> > extension to a new SSRC and our view is that this is a replacement. An
> > AppID can be mapped to a single SSRC at a time.
> 
> OK. That's fine, but then I think the draft should say more about it.


[Roni Even] In section 3 there is the following text
" Each value of the AppID maps to one SSRC at a time.  When a new SSRC
   is mapped to an existing AppID using an RTP header extension or SDES
   message, it replaces the previous RTP stream for this application
   usage."

> 
> Also, a followup question:
> 
> Suppose I start with an offer with
> 
>         a=ssrc:11111 AppID:1
> 
> and then use SDES or header extension to change move AppID 1 to ssrc
> 22222.
> 
> If there is then another O/A, with the same SDP as before:
> 
>         a=ssrc:11111 AppID:1
> 
> Does that change the mapping back to 11111, or does it stay as 22222?
> 
> This impacts how closely the SDP signaling must be coupled to the media
> processing.
[Roni Even]This needs clarification in the previous text. If the mapping is
changed using SDES or RTP header it is the current mapping so if a new offer
is sent it should use the current SSRC mapping.  

> 
> > We looked at having more than one SSRC mapped to a single AppID and
> > decided to leave it to discussion. It will require to distinguish
> > between adding and SSRC and replacing an SSRC when a new mapping is
> specified.
> > My view is that the unique mapping is clearer and I assume we can look
> > at using SDP grouping if want to associate two RTP media streams to an
> > application usage.
> 
> I asked about
> 
>         a=ssrc:11111,22222 AppID:1
> 
> just because it seemed to be allowed by the syntax. My next question would
> be what it means. E.g, it *could* mean that it is expected that
> 11111 and 22222 may both be used, but won't be used simultaneously. If it
> turned out that they *were* used simultaneously, then the effect would be
> to switch rapidly between them.
[Roni Even] We have a problem in the syntax. We did not mean to allow
mapping two SSRCs to the same AppID. The usage for the syntax was to allow a
preannounce of AppIDs for example "a=appID:2,3" to allow them to be used
later in SDES or RTP header and know to which m-line they correspond. So we
will need to change the syntax.

> 
> 	Thanks,
> 	Paul
> 
> > Roni
> >
> >>
> >> My other issue is with:
> >>      In the CLUE WG case the mapping is from an RTP stream to a CLUE
> media
> >>      capture specified in the CLUE framework [I-D.ietf-clue-framework].
> >> This should be to a CLUE capture-encoding.
> > [Roni Even] OK, will fix it
> >>
> >> 	Thanks,
> >> 	Paul
> >>
> >> On 6/28/13 11:54 AM, Roni Even wrote:
> >>> Hi,
> >>> We have submitted this document that  defines a mechanism to provide
> >> the mapping between the
> >>>      SSRCs of RTP streams and the application semantics by defining
> >> extensions to RTP and RTCP messages.
> >>>
> >>> It defines a new SDP attribute, RTCP SDES message and RTP header
> >> extension for this puropose. The document explains how it can be used
> >> with the different multiplexing proposal (Plan A, Plan B and no plan).
> >>>
> >>> It can be used also in CLUE to map SSRCs to Clue media capture.
> >>>
> >>> Please review and send comments
> >>> Thanks
> >>> Roni Even
> >>>
> >>> ________________________________________
> >>> From: internet-drafts@ietf.org [internet-drafts@ietf.org]
> >>> Sent: Friday, June 28, 2013 6:27 PM
> >>> To: Jonathan Lennox; Qin Wu; Roni Even
> >>> Subject: New Version Notification for   draft-even-mmusic-application-
> >> token-00.txt
> >>>
> >>> A new version of I-D, draft-even-mmusic-application-token-00.txt
> >>> has been successfully submitted by Roni Even and posted to the IETF
> >>> repository.
> >>>
> >>> Filename:        draft-even-mmusic-application-token
> >>> Revision:        00
> >>> Title:           The Session Description Protocol (SDP) Application
> > Token
> >> Attribute
> >>> Creation date:   2013-06-28
> >>> Group:           Individual Submission
> >>> Number of pages: 11
> >>> URL:
http://www.ietf.org/internet-drafts/draft-even-mmusic-
> >> application-token-00.txt
> >>> Status:          http://datatracker.ietf.org/doc/draft-even-mmusic-
> >> application-token
> >>> Htmlized:
> > http://tools.ietf.org/html/draft-even-mmusic-application-
> >> token-00
> >>>
> >>>
> >>> Abstract:
> >>>      The RTP fixed header includes the payload type number and the
SSRC
> >>>      values of the RTP stream.  RTP defines how you de-multiplex
streams
> >>>      within an RTP session, but in some use cases applications need
> >>>      further identifiers in order to identify the application
semantics
> >>>      associated with particular streams within the session.
> >>>
> >>>      This document defines a mechanism to provide the mapping
> >>> between
> >> the
> >>>      SSRCs of RTP streams and the application semantics by defining
> >>>      extensions to RTP and RTCP messages.
> >>>
> >>>
> >>>
> >>>
> >>> The IETF Secretariat
> >>> _______________________________________________
> >>> 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: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: Application to Network Communication
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On Jun 29, 2013, at 24:27 AM, Christer Holmberg <christer.holmberg@ericsson=
.com> wrote:

> Hi,
>=20
>>> I guess my questions will be answered once I read the draft, but I'll a=
sk a couple of high-level questions at this point anyway :)
>>=20
>> I am afraid that the draft will open up for more questions than answers.=
;-)
>=20
> Well, at least it's good if the questions come before the draft is publis=
hed as RFC :)
>=20
>>> Q1:	How will you associate the STUN requests with a given media flow? I=
f your answer is based on IP address:port, do you=20
>>>                       then assume that there will be only a single flow=
 per IP address:port?
>>>=20
>>=20
>> The STUN messages are sent on the same 5-tuple as media, so the metadata=
 in the STUN packets needs to describe whatever is being sent on that 5-tup=
le.
>>=20
>> Introducing BUNDLE into this is interesting. But again I think that is u=
p to what the metadata want to describe. If audio and video is flowing on t=
he same 5-tuple, that needs to be described.=20
>=20
> BUNDLE is one case, but there are also discussions about sending multiple=
 media flows of the same media type, separated by SSRC, on the same 5-tuple=
. The SDP a=3Dssrc attribute might be enough to signal that, so you wouldn'=
t need BUNDLE.

That is true. Thanks for the pointer. We some text around multiplexing scen=
arios and the problems associated with it.

>=20
>>> Q2:	You say that the network will provide information using STUN reques=
ts. Will the network piggyback information in the STUN requests sent by the=
 ICE endpoints, or will the network generate their own STUN requests?
>>>=20
>> Piggyback. The plan is to add attributes the network elements can write =
information into. That might save some mallocs on the network side.=20
>=20
> Keep in mind that if one endpoint is ICE lite, it won't send any STUN con=
nectivity check requests (only responses) or keep-alives. That might not be=
 an issue in UA-to-UA scenarios, where both ICE endpoints are typically ful=
l ICE.=20
>=20
> But, if one ICE endpoint is located within the network it will often be I=
CE lite. Now, I don't think you would have to mandate those to become full =
ICE, but I guess they at least would need to send keep-alives, in order to =
allow piggybacking of data in both directions.
>=20
Yes. For now we are focusing on solving two problems, how to get better tre=
atment of flows in the network and how to improve ICE path selection. To so=
lve both of these in the ICE-lite scenario I think we need to introduce a M=
ALICE-light version that does what you describe above (Send some conn check=
s and answer checks with reflected metadata attributes.)

Will post the draft later today.
.-.
P=E5l-Erik


> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
>> -----Original Message-----
>> From: Pal Martinsen (palmarti) [mailto:palmarti@cisco.com]=20
>> Sent: 28. kes=E4kuuta 2013 10:13
>> To: Christer Holmberg
>> Cc: mmusic
>> Subject: Re: Application to Network Communication
>>=20
>>=20
>> On Jun 27, 2013, at 11:42 AM, Christer Holmberg <christer.holmberg@erics=
son.com> wrote:
>>=20
>>> Hi,
>>>=20
>>> I am not really sure what you mean by "how network should treat differe=
nt streams".
>>>=20
>>> What kind of network entities are you referring to, and what kind of tr=
eatment?
>>>=20
>>=20
>> Routers, switches, firewalls, NATs, ALGs, SBCs, B2Bs. And a few acronyms=
 I probably don=B4t know exist. In short any device in the network that rig=
htfully or not is or inserts itself in the media path.=20
>>=20
>> The name of the draft that currently is being drafted is MALICE (Metadat=
a Attribute signaLing with ICE). The basic idea is to signal a set of descr=
iptive metadata attributes in some of the STUN messages that are sent when =
an agent does ICE. The way STUN is crafted makes it easy for network entiti=
es to distinguish the STUN metadata packets and act on the information.=20
>>=20
>> One of the short term use cases this can solve is the problem where DSCP=
 markings can be lost when the packet crosses administrative domains. If me=
tadata is sent in-band with STUN it can for instance describe this stream a=
s video. If SRTP is in use it would be hard/impossible for any network enti=
ty to know that the stream is video without this in-band metadata. So if ne=
twork A and B want to treat video differently by having their own DSCP mark=
ings they can now do this.=20
>>=20
>> MALICE tries to just be a framework for sending the metadata, thus enabl=
ing application to network communication. It also tries to describe how rou=
ters can add useful information to the metadata packets, and how that infor=
mation can be sent back to the application without relying on address spoof=
ing or opening up for multiplying attacks. This opens up for network to app=
lication communication. Like ECN, but more descriptive and it works on more=
 platforms. (I am no expert n the area, but I am under the impression that =
it is difficult to get the complete IP header up to the application on some=
 platforms)
>>=20
>> This can be used in a multitude of ways; Optimising the ICE path selecti=
on based on actual network feedback, making it easier for video application=
 to choose the correct starting bandwidth, can enable better down/and up sp=
eeding of video streams. Hopefully there are a lot of other use cases as we=
ll.=20
>>=20
>> I don=B4t expect this to be an easy problem to solve. So for now I think=
 the big question is:
>> "Is it worthwile trying to come up with machanisms that enable better ap=
plication to/from network communication?"
>>=20
>> .-.
>> P=E5l-Erik
>>=20
>>=20
>>> Regards,
>>>=20
>>> Christer
>>>=20
>>>=20
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behal=
f Of Pal Martinsen (palmarti)
>>> Sent: 27. kes=E4kuuta 2013 11:36
>>> To: mmusic
>>> Subject: [MMUSIC] Application to Network Communication
>>>=20
>>> Hi All,
>>>=20
>>> We see a need for better application to network communication to convey=
 how the network should/could treat the different streams described in the =
medalines of the SDP. This is especially important as video codecs have bec=
ome very elastic regarding network bandwidth. It would be useful for the ne=
twork to know the nature regarding the usage of the video stream. The netwo=
rk could treat the stream very differently whatever it is realtime video re=
quiring low latency and high bandwidth or is it a more static presentation =
flow carrying a slideshow. Getting information from the network to the appl=
ication can also prove to be useful.
>>>=20
>>> Todays existing solutions can be difficult to implement on some devices=
 as the information is carried in the IP header or the markings in the pack=
ets is ignored across administrative domains. Other solution treats the net=
work as a black box and tries to fix problems when they are discovered. Bot=
h the network and the application would benefit if information regarding th=
ings that is about to happen as opposed to just happened could be conveyed.
>>>=20
>>> We are considering extending ICE connectivity checks and keepalives (ST=
UN) to provide this communication from the endpoint to the network and from=
 the network to the endpoint.
>>>=20
>>> Are other folks seeing this problem as well, or are existing techniques=
 and throwing bandwidth at the problem sufficient today and sufficient tomo=
rrow?
>>>=20
>>> Any comments, feedback and ideas are very welcome.
>>>=20
>>> Regards
>>> P=E5l-Erik Martinsen
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>=20
>=20


From pkyzivat@alum.mit.edu  Mon Jul  1 10:03:06 2013
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Date: Mon, 01 Jul 2013 13:02:58 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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To: Roni Even <ron.even.tlv@gmail.com>
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] FW: New Version Notification	for	draft-even-mmusic-application-token-00.txt
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Hi Roni. See inline.

On 7/1/13 7:38 AM, Roni Even wrote:
>
>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>> Sent: 30 June, 2013 7:03 PM
>> To: Roni Even
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] FW: New Version Notification for draft-even-
>> mmusic-application-token-00.txt
>>
>> On 6/30/13 1:29 AM, Roni Even wrote:
>>> Inline
>>>
>>>> -----Original Message-----
>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>>> Behalf Of Paul Kyzivat
>>>> Sent: 29 June, 2013 6:59 PM
>>>> To: mmusic@ietf.org
>>>> Subject: Re: [MMUSIC] FW: New Version Notification for draft-even-
>>>> mmusic-application-token-00.txt
>>>>
>>>> I looked this over, and it seems to solve many of the issues at hand.
>>>> There are no doubt some nits to iron out, but we should get agreement
>>>> over the concepts before worrying too much about the details.
>>>>
>>>> A couple of things I wanted to comment on:
>>>>
>>>> You give examples of form:
>>>>          a=ssrc:11111 AppID:1
>>>> But you don't say too much about the meaning of this. The reason I
>>>> ask is that you say elsewhere that the appid can move from one ssrc to
>> another.
>>>> My question is: If this form is used, can that appid later be bound
>>>> to a different ssrc via the header extension? Also, is the following
> legal?
>>>>          a=ssrc:11111,22222 AppID:1
>>> [Roni Even] The AppID can be mapped later using SDES or header
>>> extension to a new SSRC and our view is that this is a replacement. An
>>> AppID can be mapped to a single SSRC at a time.
>>
>> OK. That's fine, but then I think the draft should say more about it.
>
>
> [Roni Even] In section 3 there is the following text
> " Each value of the AppID maps to one SSRC at a time.  When a new SSRC
>     is mapped to an existing AppID using an RTP header extension or SDES
>     message, it replaces the previous RTP stream for this application
>     usage."

It still isn't clear to me. You have three ways to communicate the mapping:

- SDP
- SDES
- header extension

I think the relationship and precedence between SDES and header 
extension is well defined in general, so I guess that doesn't need 
further elaboration.

But the relationship between the SDP and the RTP/RTCP mechanisms isn't 
so clear. This is especially problematic because it is impossible to 
synchronize the arrival of SDP with the arrival of RTP/RTCP.

The simplest answer would be to use the SDP as a *default* if the 
mapping hasn't been received in the media, with changes in the SDP 
having no effect on the mapping of a particular AppId once it has been 
mentioned in the media. But I don't know if that is a sufficient mechanism.

Alternatively, the mapping can be defined by whichever has been received 
most recently. But that may cause artifacts.

I just think this needs to be made very clear.

>> Also, a followup question:
>>
>> Suppose I start with an offer with
>>
>>          a=ssrc:11111 AppID:1
>>
>> and then use SDES or header extension to change move AppID 1 to ssrc
>> 22222.
>>
>> If there is then another O/A, with the same SDP as before:
>>
>>          a=ssrc:11111 AppID:1
>>
>> Does that change the mapping back to 11111, or does it stay as 22222?
>>
>> This impacts how closely the SDP signaling must be coupled to the media
>> processing.
> [Roni Even]This needs clarification in the previous text. If the mapping is
> changed using SDES or RTP header it is the current mapping so if a new offer
> is sent it should use the current SSRC mapping.

See comments above.

>>> We looked at having more than one SSRC mapped to a single AppID and
>>> decided to leave it to discussion. It will require to distinguish
>>> between adding and SSRC and replacing an SSRC when a new mapping is
>> specified.
>>> My view is that the unique mapping is clearer and I assume we can look
>>> at using SDP grouping if want to associate two RTP media streams to an
>>> application usage.
>>
>> I asked about
>>
>>          a=ssrc:11111,22222 AppID:1
>>
>> just because it seemed to be allowed by the syntax. My next question would
>> be what it means. E.g, it *could* mean that it is expected that
>> 11111 and 22222 may both be used, but won't be used simultaneously. If it
>> turned out that they *were* used simultaneously, then the effect would be
>> to switch rapidly between them.
> [Roni Even] We have a problem in the syntax. We did not mean to allow
> mapping two SSRCs to the same AppID. The usage for the syntax was to allow a
> preannounce of AppIDs for example "a=appID:2,3" to allow them to be used
> later in SDES or RTP header and know to which m-line they correspond. So we
> will need to change the syntax.

OK. So if the intent is not to allow it, then sure, the syntax can be 
tweaked so it isn't possible to say it.

	Thanks,
	Paul


From iesg-secretary@ietf.org  Mon Jul  1 11:56:24 2013
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Subject: [MMUSIC] Protocol Action: 'Session Description Protocol (SDP) Extension For	Setting Audio and Video Media Streams Over Circuit-Switched	Bearers In The Public Switched Telephone Network (PSTN)' to	Proposed Standard (draft-ietf-mmusic-sdp-cs-21.txt)
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The IESG has approved the following document:
- 'Session Description Protocol (SDP) Extension For Setting Audio and
   Video Media Streams Over Circuit-Switched Bearers In The Public
   Switched Telephone Network (PSTN)'
  (draft-ietf-mmusic-sdp-cs-21.txt) as Proposed Standard

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

The IESG contact persons are Gonzalo Camarillo and Richard Barnes.

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




*Technical Summary
*

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

*Working Group Summary
*

The WG had some discussion around the format to use for E.164 numbers
and whether to align this with the existing definition in RFC 3108. The
RFC 3108 definition was seen as deficient and the WG agreed it was
better to align with relevant parts of the tel URI format defined in RFC
3966, not least since SDP address types are defined in the context of a
particular network type, and hence RFC 3108 compatibility is not a
concern (the implication is that the "E164" address type may differ
between network types in SDP).

*Document Quality
*

There are currently no known implementations of the draft, however the
draft is a dependency for 3GPP, so future implementations are expected.

The document has received good overall review in the WG, some of which
resulted in changes to the detailed specification. The document has been
reviewed in detail several times, including of the last few versions.
The major contributors to these as well as earlier discussions are
listed in the Acknowledgements section of the document.


*Personnel
*

Document Shepherd: Flemming Andreasen
Responsible AD: Gonzalo Camarillo 

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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] DECISION: Default mechanism to map RTP data to m- line is based on PT?
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On 7/1/13 5:55 AM, Colin Perkins wrote:
> On 29 Jun 2013, at 16:44, Paul Kyzivat wrote:
>> On 6/29/13 8:44 AM, Colin Perkins wrote:
>>> On 26 Jun 2013, at 13:20, Christer Holmberg wrote:
>>>> Hi Colin,
>>>>
>>>>>> Emil suggested that the default, "MTI", mechanism for mapping RTP data to m- lines should be based on PT. Applications are
>>>>>> allowed to use whatever other mechanisms, but usage of such mechanisms must be negotiated (or, applications need to have
>>>>>> some other means knowing that the other endpoint support such mechanisms).
>>>>>>
>>>>>> Q3: Do we need to specify a default, MTI, mechanism for mapping RTP data to m- lines?
>>>>>
>>>>> I think the BUNDLE specification needs to define how RTP flows can be associated with m= lines, but ought not mandate that applications
>>>>> do so. The document that describes how WebRTC uses BUNDLE might want to mandate that WebRTC applications use the mechanism.
>>>>
>>>> Note that there is a separate discussion, BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?, on whether BUNDLE shall mandate that the receiver must always be able to map received data to an m- line.
>>>>
>>>> So far only Paul has commented on that discussion, but I assume it's just a question of hours before your input will be posted? :)
>>>>
>>>>>> Q4: If Q3, do we mandate applications to support, and use (unless applications are made aware of other mechanisms supported by all endpoints) PT for mapping received RTP media?
>>>>>
>>>>> I think we should specify how the combination of PT values and a=ssrc: lines can be used to associate RTP data with m= lines, for applications that want to do so.
>>>>>
>>>>> I don't think just using PT values is sufficient, since the PT space is small (I can imagine applications that want to send lots of flows using
>>>>> the same codec configuration, and it's more natural to give them all the same PT and use a=ssrc: lines to distinguish).
>>>>
>>>> If you run out of PTs, and have to re-use the same PT in multiple m- lines (for the same codec configuration, though), you will obviously need something else for mapping RTP to m- lines, and we should probably state that.
>>>>
>>>> But, my take from your reply is that you don't think we should specify a default mechanism. We CAN describe different mechanisms (PT, PT+SSRC etc), but it would be up to applications to decide what to use. Correct?
>>>
>>>
>>> No.
>>>
>>> Specifying a "default" suggests that there might be alternative ways of doing this. If we specify an algorithm for mapping RTP flows to m= lines, we should specify a single algorithm. We don't need multiple ways of doing this. I think that single algorithm should look at the PT and the a=ssrc: lines. Just looking at the PT isn't sufficient.
>>
>> But we *will* have alternative ways of doing this:
>>
>> - one way is by SSRC, specifying which SSRC goes to which m-line
>>
>> - another way could be by appid (from Roni's new draft)
>>
>> The only thing that is special about PT is that *every* RTP media section contains PTs, so it is always available to be used.
>
> Every RTP packet contains an SSRC, so that's always available to use too. You may not chose to use it, but it seems strange to ignore it.

The think about PT is that it is guaranteed to be in the SDP.

There are lots of things in the RTP, but most of them are sometimes, or 
usually, not in the SDP.

While we have a way of putting SSRC into the SDP, we have heard much 
discussion of cases where the SSRC values aren't known at the time the 
SDP is sent.

The AppId draft, if it progresses, will provide another way. That makes 
three mechanisms. We *could* define an *algorithm* for mapping that is 
based on the sometimes presence of some or all of these in the SDP. But 
then I think it may need to be extensible for new attributes that might 
be used for this.

I did specify an abstract algorithm of this sort in an earlier reply.

(I suspect we aren't in fundamental disagreement. We are probably just 
talking about this is very different ways. For me the fundamental point 
is that a valid O/A must always define (perhaps implicitly) the 
algorithm to do the association.)

	Thanks,
	Paul

> Colin
>
>> If I want to use appid, I may very well *not* want to use ssrc as well. I may want the same ssrc to associate with *different* m-lines in the bundle depending on which appid it has.
>>
>> And in that case I may well not want to use PT either.
>>
>> 	Thanks,
>> 	Paul
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
>
>


From ron.even.tlv@gmail.com  Tue Jul  2 05:33:49 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Paul Kyzivat'" <pkyzivat@alum.mit.edu>
References: <20130628152721.7537.5884.idtracker@ietfa.amsl.com>	<760B7D45D1EFF74988DBF5C2122830C2299A10BE@szxpml504-mbs.exmail.huawei.com> <51CF0435.3050305@alum.mit.edu> <047901ce7552$de032840$9a0978c0$@gmail.com> <51D0569C.3020508@alum.mit.edu> <051301ce764f$95a89370$c0f9ba50$@gmail.com> <51D1B642.8080608@alum.mit.edu>
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Date: Tue, 2 Jul 2013 15:32:01 +0300
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] FW: New Version Notification	for	draft-even-mmusic-application-token-00.txt
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: 01 July, 2013 8:03 PM
> To: Roni Even
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] FW: New Version Notification for draft-even-
> mmusic-application-token-00.txt
> 
> Hi Roni. See inline.
> 
> On 7/1/13 7:38 AM, Roni Even wrote:
> >
> >
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> >> Sent: 30 June, 2013 7:03 PM
> >> To: Roni Even
> >> Cc: mmusic@ietf.org
> >> Subject: Re: [MMUSIC] FW: New Version Notification for draft-even-
> >> mmusic-application-token-00.txt
> >>
> >> On 6/30/13 1:29 AM, Roni Even wrote:
> >>> Inline
> >>>
> >>>> -----Original Message-----
> >>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org]
> On
> >>>> Behalf Of Paul Kyzivat
> >>>> Sent: 29 June, 2013 6:59 PM
> >>>> To: mmusic@ietf.org
> >>>> Subject: Re: [MMUSIC] FW: New Version Notification for draft-even-
> >>>> mmusic-application-token-00.txt
> >>>>
> >>>> I looked this over, and it seems to solve many of the issues at hand.
> >>>> There are no doubt some nits to iron out, but we should get
> >>>> agreement over the concepts before worrying too much about the
> details.
> >>>>
> >>>> A couple of things I wanted to comment on:
> >>>>
> >>>> You give examples of form:
> >>>>          a=ssrc:11111 AppID:1
> >>>> But you don't say too much about the meaning of this. The reason I
> >>>> ask is that you say elsewhere that the appid can move from one ssrc
> >>>> to
> >> another.
> >>>> My question is: If this form is used, can that appid later be bound
> >>>> to a different ssrc via the header extension? Also, is the
> >>>> following
> > legal?
> >>>>          a=ssrc:11111,22222 AppID:1
> >>> [Roni Even] The AppID can be mapped later using SDES or header
> >>> extension to a new SSRC and our view is that this is a replacement.
> >>> An AppID can be mapped to a single SSRC at a time.
> >>
> >> OK. That's fine, but then I think the draft should say more about it.
> >
> >
> > [Roni Even] In section 3 there is the following text " Each value of
> > the AppID maps to one SSRC at a time.  When a new SSRC
> >     is mapped to an existing AppID using an RTP header extension or SDES
> >     message, it replaces the previous RTP stream for this application
> >     usage."
> 
> It still isn't clear to me. You have three ways to communicate the
mapping:
> 
> - SDP
> - SDES
> - header extension
> 
> I think the relationship and precedence between SDES and header extension
> is well defined in general, so I guess that doesn't need further
elaboration.
> 
> But the relationship between the SDP and the RTP/RTCP mechanisms isn't so
> clear. This is especially problematic because it is impossible to
synchronize
> the arrival of SDP with the arrival of RTP/RTCP.
> 
> The simplest answer would be to use the SDP as a *default* if the mapping
> hasn't been received in the media, with changes in the SDP having no
effect
> on the mapping of a particular AppId once it has been mentioned in the
> media. But I don't know if that is a sufficient mechanism.


[Roni Even] I think this make sense because the usage of the SDES and RTP
header is to allow changes without a new O/A what we need to specify is what
should be the SDP in a case when we used mapping to SSRC in the SDP and
after using SDES to remap we will have a new O/A. I think that we should not
try to synchronize the SDP but maybe add a flag that will say the SEDS / RTP
header should be used for the mapping.

> 
> Alternatively, the mapping can be defined by whichever has been received
> most recently. But that may cause artifacts.
> 
> I just think this needs to be made very clear.
> 
> >> Also, a followup question:
> >>
> >> Suppose I start with an offer with
> >>
> >>          a=ssrc:11111 AppID:1
> >>
> >> and then use SDES or header extension to change move AppID 1 to ssrc
> >> 22222.
> >>
> >> If there is then another O/A, with the same SDP as before:
> >>
> >>          a=ssrc:11111 AppID:1
> >>
> >> Does that change the mapping back to 11111, or does it stay as 22222?
> >>
> >> This impacts how closely the SDP signaling must be coupled to the
> >> media processing.
> > [Roni Even]This needs clarification in the previous text. If the
> > mapping is changed using SDES or RTP header it is the current mapping
> > so if a new offer is sent it should use the current SSRC mapping.
> 
> See comments above.
> 
> >>> We looked at having more than one SSRC mapped to a single AppID and
> >>> decided to leave it to discussion. It will require to distinguish
> >>> between adding and SSRC and replacing an SSRC when a new mapping is
> >> specified.
> >>> My view is that the unique mapping is clearer and I assume we can
> >>> look at using SDP grouping if want to associate two RTP media
> >>> streams to an application usage.
> >>
> >> I asked about
> >>
> >>          a=ssrc:11111,22222 AppID:1
> >>
> >> just because it seemed to be allowed by the syntax. My next question
> >> would be what it means. E.g, it *could* mean that it is expected that
> >> 11111 and 22222 may both be used, but won't be used simultaneously.
> >> If it turned out that they *were* used simultaneously, then the
> >> effect would be to switch rapidly between them.
> > [Roni Even] We have a problem in the syntax. We did not mean to allow
> > mapping two SSRCs to the same AppID. The usage for the syntax was to
> > allow a preannounce of AppIDs for example "a=appID:2,3" to allow them
> > to be used later in SDES or RTP header and know to which m-line they
> > correspond. So we will need to change the syntax.
> 
> OK. So if the intent is not to allow it, then sure, the syntax can be
tweaked so
> it isn't possible to say it.
> 
> 	Thanks,
> 	Paul


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From: "Pal Martinsen (palmarti)" <palmarti@cisco.com>
To: mmusic <mmusic@ietf.org>
Thread-Topic: New Version Notification for draft-martinsen-mmusic-malice-00.txt
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Subject: [MMUSIC] New Version Notification for draft-martinsen-mmusic-malice-00.txt
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Hi,

We submitted a new draft called MALICE today. It is all about smarter appli=
cation/network communication. Se abstract for some details. Read the whole =
draft to get a lot more details.

>=20
> A new version of I-D, draft-martinsen-mmusic-malice-00.txt
> has been successfully submitted by Reinaldo Penno and posted to the
> IETF repository.
>=20
> Filename:	 draft-martinsen-mmusic-malice
> Revision:	 00
> Title:		 Meta-data Attribute signaLling with ICE
> Creation date:	 2013-07-02
> Group:		 Individual Submission
> Number of pages: 33
> URL:             http://www.ietf.org/internet-drafts/draft-martinsen-mmus=
ic-malice-00.txt
> Status:          http://datatracker.ietf.org/doc/draft-martinsen-mmusic-m=
alice
> Htmlized:        http://tools.ietf.org/html/draft-martinsen-mmusic-malice=
-00
>=20
>=20
> Abstract:
>   It can be useful for applications to provide flow metadata
>   information to on-path devices to influence flow treatment in the
>   network.  Provided that the network is able to provide useful
>   feedback, this can also influence path selection if an application
>   have multiple flow paths to choose from.
>=20
>   This draft describes how this can be achieved by adding metadata to
>   the STUN packets sent during the ICE connectivity checks or a
>   slightly modified version of the keep-alive mechanism.  Devices on
>   the media path can use the metadata information to prioritize the
>   flow, perform traffic engineering, or provide network analytics and
>   notifications as requested by the endpoints.  On-path devices can
>   append or modify the existing metadata information in the STUN/ICE
>   messages to enable feedback to other on-path devices or the
>   applications in both ends of the media session.
>=20
>   This document describes a framework mechanism for how such metadata
>   can be transported by STUN when ICE is in use and it covers the
>   endpoint and on path device processing.  The functionality described
>   here is referred to as MALICE.

.-.
P=E5l-Erik=

From pkyzivat@alum.mit.edu  Tue Jul  2 07:05:37 2013
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Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
> Informative descriptions of possible mappings are fine, but nothing normative, please. Some applications won't care about the mapping, and others may care but use a different mapping from those described. The only normative statement you could make is something like: "If you care about the mapping, you MUST have a mapping mechanism. You MAY use the mechanism(s) described here, or some other mechanism." But why mandate such a tautology?
>
> The assumption below that multiple m-lines always require different m-line-specific processing of the packets also assumes Plan B, i.e. that multiple streams with similar processing should always be signaled with a single m-line (for example, using a=ssrc or max-ssrc). Plan A purists would use multiple m-lines even for multiple streams with the same processing. I don't think we want to force bundle to pick a plan.

I'm going to keep asking this question until I get an answer that makes 
sense:

Please explain to me a meaningful situation when there are multiple 
m-lines in a bundle and there is insufficient information to associate 
each packet with an m-line.

The *point* of plan A is that there be an m-line per "flow", and that 
the offerer is enumerating the flows. So clearly it knows the mapping. 
If the answerer isn't capable of doing the same mapping, then the use of 
plan A has failed.

The part that I want to be normative is that an O/A is not valid unless 
the mapping is possible with the information in the O/A, and that the 
offerer and the answerer would agree about the mapping of each packet. I 
don't care if they actually do the mapping, though if they are doing any 
sensible processing then they will be doing something *equivalent* to 
the mapping.

	Thanks,
	Paul

> Mo
>
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: Saturday, June 29, 2013 12:12 PM
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>
> On 6/29/13 8:50 AM, Colin Perkins wrote:
>> Christer,
>>
>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
>>> There has been some discussions about whether BUNDLE should mandate that users are mandated to always (no matter what transport protocols are used in the BUNDLE group) have a mechanism to map received data to an m- line, or whether it from a generic BUNDLE perspective should be optional - IF there would be cases where it's not needed.
>>>
>>> We haven't had that much discussion about it yet, so I will not ask a DECISION question at this point, but I would really like to get some input from people who have opinions about this :)
>>
>> My opinion: BUNDLE needs to mandate a single algorithm for mapping from RTP flows to m= lines, for those applications that care about such a mapping. I don't think it should mandate that all applications need to care.
>
> I just replied elsewhere on this.
> My opinion turns out to be the exact opposite of yours: we need to
> support multiple algorithms, and that all applications should care.
>
> I won't repeat the stuff about multiple algorithms.
>
> I will repeat what I have said about the need for being able to associate:
>
> There must have been *some* reason that two m-lines were used, rather
> than just one. Each m-line heads a media section of the SDP that
> contains a bunch of declarations. Something must be different in those
> two media sections, or else a single media section would have been
> enough. Presumably whatever it is that is different is intended to
> affect how received packets are processed or interpreted. (If not, again
> there is no need for it to be there.) If you can't associate the packet
> with the m-line, then you don't know which of the multiple
> interpretations to apply to the packet.
>
> Maybe there is some exception to this, though I haven't come up with
> anything. If there is, then I hope someone will call it out. If so, then
> the constraint can be tightened.
>
> 	Thanks,
> 	Paul
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From pkyzivat@alum.mit.edu  Tue Jul  2 08:01:00 2013
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sctp-sdp-04.txt
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On 6/30/13 12:34 PM, Salvatore Loreto wrote:
> Hi there,
>
> I have produced a new version to address all the comments received on
> the mailing list
>
> Comments, feedback and questions are very well appreciated and requested
>
> cheers
> Salvatore

Here are some things that I found. I'm pretty sure some of them have 
been discussed on the list before.

In section 5.1:

    The sctpmap attribute maps from a port number (as used in an "m="
    line) to an encoding name denoting the payload format to be used on
    top of the SCTP association or the actual protocol running on top of
    it.

While the earlier examples may clarify, this text isn't clear about what 
"port" means. The m-line contains a <port> field, and a <fmt> field that 
has been said to contain an sctp-port. I think this text should 
reference the <fmt> field:

    The sctpmap attribute maps from a <fmt> number (as used in an "m="
    line) to an encoding name denoting the payload format to be used on
    top of the SCTP association or the actual protocol running on top of
    it.

Also in this section you define <protocol> as <labelstring>. There is no 
specification of how/where such things are defined - e.g. a registry. 
That can't be.

In the case of rtpmap, the <media> field from the m-line, plus the 
<encoding-name> from the rtpmap-line form a mime-type that is registered 
in IANA. Some of the examples suggest that the same approach is intended 
for sctpmap. But this needs to be clearly specified. If you don't intend 
to follow this approach, then for the examples (such as T38) to be valid 
there will need to be definitions of them is whatever registration 
scheme you propose.

Section 9.3:

The example shows a=webrtc-DataChannel lines.

Are these defined in any rfc or draft? I can't find one. I think you 
should either use real examples, that are defined somewhere, or else 
make clear that you are using a hypothetical example.

	Thanks,
	Paul


From mzanaty@cisco.com  Tue Jul  2 09:09:11 2013
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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An attempt at "an answer that makes sense". (Just snippets, not full SDP.)

m=3Dvideo 10000 RTP/SAVPF 96
a=3Drtpmap:96 H264/90000
a=3Dfmtp:96 profile-level-id=3D42001F <--- HD 720p30 main speaker

m=3Dvideo 10000 RTP/SAVPF 97
a=3Drtpmap:97 H264/90000
a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail
a=3Drecvonly

m=3Dvideo 10000 RTP/SAVPF 97
a=3Drtpmap:97 H264/90000
a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail
a=3Drecvonly

m=3Dvideo 10000 RTP/SAVPF 97
a=3Drtpmap:97 H264/90000
a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail
a=3Drecvonly

m=3Dvideo 10000 RTP/SAVPF 97
a=3Drtpmap:97 H264/90000
a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail
a=3Drecvonly

No need to map thumbnail packets to specific m-lines, so reuse of PT=3D97 i=
s fine. Just render the last 4 thumbnail sources received (in addition to t=
he main speaker).

You could mandate that PT must be unique to allow unique mapping.
=20
You could mandate that multiple m-lines of thumbnails must be combined into=
 a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.

Which do you want to mandate and why?

Cheers,
Mo


-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]=20
Sent: Tuesday, July 02, 2013 10:06 AM
To: Mo Zanaty (mzanaty)
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map rec=
eived data to m- line?

On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
> Informative descriptions of possible mappings are fine, but nothing norma=
tive, please. Some applications won't care about the mapping, and others ma=
y care but use a different mapping from those described. The only normative=
 statement you could make is something like: "If you care about the mapping=
, you MUST have a mapping mechanism. You MAY use the mechanism(s) described=
 here, or some other mechanism." But why mandate such a tautology?
>
> The assumption below that multiple m-lines always require different m-lin=
e-specific processing of the packets also assumes Plan B, i.e. that multipl=
e streams with similar processing should always be signaled with a single m=
-line (for example, using a=3Dssrc or max-ssrc). Plan A purists would use m=
ultiple m-lines even for multiple streams with the same processing. I don't=
 think we want to force bundle to pick a plan.

I'm going to keep asking this question until I get an answer that makes=20
sense:

Please explain to me a meaningful situation when there are multiple=20
m-lines in a bundle and there is insufficient information to associate=20
each packet with an m-line.

The *point* of plan A is that there be an m-line per "flow", and that=20
the offerer is enumerating the flows. So clearly it knows the mapping.=20
If the answerer isn't capable of doing the same mapping, then the use of=20
plan A has failed.

The part that I want to be normative is that an O/A is not valid unless=20
the mapping is possible with the information in the O/A, and that the=20
offerer and the answerer would agree about the mapping of each packet. I=20
don't care if they actually do the mapping, though if they are doing any=20
sensible processing then they will be doing something *equivalent* to=20
the mapping.

	Thanks,
	Paul

> Mo
>
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf =
Of Paul Kyzivat
> Sent: Saturday, June 29, 2013 12:12 PM
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map r=
eceived data to m- line?
>
> On 6/29/13 8:50 AM, Colin Perkins wrote:
>> Christer,
>>
>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
>>> There has been some discussions about whether BUNDLE should mandate tha=
t users are mandated to always (no matter what transport protocols are used=
 in the BUNDLE group) have a mechanism to map received data to an m- line, =
or whether it from a generic BUNDLE perspective should be optional - IF the=
re would be cases where it's not needed.
>>>
>>> We haven't had that much discussion about it yet, so I will not ask a D=
ECISION question at this point, but I would really like to get some input f=
rom people who have opinions about this :)
>>
>> My opinion: BUNDLE needs to mandate a single algorithm for mapping from =
RTP flows to m=3D lines, for those applications that care about such a mapp=
ing. I don't think it should mandate that all applications need to care.
>
> I just replied elsewhere on this.
> My opinion turns out to be the exact opposite of yours: we need to
> support multiple algorithms, and that all applications should care.
>
> I won't repeat the stuff about multiple algorithms.
>
> I will repeat what I have said about the need for being able to associate=
:
>
> There must have been *some* reason that two m-lines were used, rather
> than just one. Each m-line heads a media section of the SDP that
> contains a bunch of declarations. Something must be different in those
> two media sections, or else a single media section would have been
> enough. Presumably whatever it is that is different is intended to
> affect how received packets are processed or interpreted. (If not, again
> there is no need for it to be there.) If you can't associate the packet
> with the m-line, then you don't know which of the multiple
> interpretations to apply to the packet.
>
> Maybe there is some exception to this, though I haven't come up with
> anything. If there is, then I hope someone will call it out. If so, then
> the constraint can be tightened.
>
> 	Thanks,
> 	Paul
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From christer.holmberg@ericsson.com  Tue Jul  2 11:30:43 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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Hi,

I am not Paul, but just to verify that I understand his issue, the question=
 is:

Why do you need 4 m- lines for the thumbnails, when you could use 1 m- line=
?

Regards,

Christer

-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] P=
uolesta Mo Zanaty (mzanaty)
L=E4hetetty: 2. hein=E4kuuta 2013 19:09
Vastaanottaja: Paul Kyzivat
Kopio: mmusic@ietf.org
Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map receiv=
ed data to m- line?

An attempt at "an answer that makes sense". (Just snippets, not full SDP.)

m=3Dvideo 10000 RTP/SAVPF 96
a=3Drtpmap:96 H264/90000
a=3Dfmtp:96 profile-level-id=3D42001F <--- HD 720p30 main speaker

m=3Dvideo 10000 RTP/SAVPF 97
a=3Drtpmap:97 H264/90000
a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvonly

m=3Dvideo 10000 RTP/SAVPF 97
a=3Drtpmap:97 H264/90000
a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvonly

m=3Dvideo 10000 RTP/SAVPF 97
a=3Drtpmap:97 H264/90000
a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvonly

m=3Dvideo 10000 RTP/SAVPF 97
a=3Drtpmap:97 H264/90000
a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvonly

No need to map thumbnail packets to specific m-lines, so reuse of PT=3D97 i=
s fine. Just render the last 4 thumbnail sources received (in addition to t=
he main speaker).

You could mandate that PT must be unique to allow unique mapping.
=20
You could mandate that multiple m-lines of thumbnails must be combined into=
 a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.

Which do you want to mandate and why?

Cheers,
Mo


-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
Sent: Tuesday, July 02, 2013 10:06 AM
To: Mo Zanaty (mzanaty)
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map rec=
eived data to m- line?

On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
> Informative descriptions of possible mappings are fine, but nothing norma=
tive, please. Some applications won't care about the mapping, and others ma=
y care but use a different mapping from those described. The only normative=
 statement you could make is something like: "If you care about the mapping=
, you MUST have a mapping mechanism. You MAY use the mechanism(s) described=
 here, or some other mechanism." But why mandate such a tautology?
>
> The assumption below that multiple m-lines always require different m-lin=
e-specific processing of the packets also assumes Plan B, i.e. that multipl=
e streams with similar processing should always be signaled with a single m=
-line (for example, using a=3Dssrc or max-ssrc). Plan A purists would use m=
ultiple m-lines even for multiple streams with the same processing. I don't=
 think we want to force bundle to pick a plan.

I'm going to keep asking this question until I get an answer that makes
sense:

Please explain to me a meaningful situation when there are multiple m-lines=
 in a bundle and there is insufficient information to associate each packet=
 with an m-line.

The *point* of plan A is that there be an m-line per "flow", and that the o=
fferer is enumerating the flows. So clearly it knows the mapping.=20
If the answerer isn't capable of doing the same mapping, then the use of pl=
an A has failed.

The part that I want to be normative is that an O/A is not valid unless the=
 mapping is possible with the information in the O/A, and that the offerer =
and the answerer would agree about the mapping of each packet. I don't care=
 if they actually do the mapping, though if they are doing any sensible pro=
cessing then they will be doing something *equivalent* to the mapping.

	Thanks,
	Paul

> Mo
>
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On=20
> Behalf Of Paul Kyzivat
> Sent: Saturday, June 29, 2013 12:12 PM
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map r=
eceived data to m- line?
>
> On 6/29/13 8:50 AM, Colin Perkins wrote:
>> Christer,
>>
>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
>>> There has been some discussions about whether BUNDLE should mandate tha=
t users are mandated to always (no matter what transport protocols are used=
 in the BUNDLE group) have a mechanism to map received data to an m- line, =
or whether it from a generic BUNDLE perspective should be optional - IF the=
re would be cases where it's not needed.
>>>
>>> We haven't had that much discussion about it yet, so I will not ask=20
>>> a DECISION question at this point, but I would really like to get=20
>>> some input from people who have opinions about this :)
>>
>> My opinion: BUNDLE needs to mandate a single algorithm for mapping from =
RTP flows to m=3D lines, for those applications that care about such a mapp=
ing. I don't think it should mandate that all applications need to care.
>
> I just replied elsewhere on this.
> My opinion turns out to be the exact opposite of yours: we need to=20
> support multiple algorithms, and that all applications should care.
>
> I won't repeat the stuff about multiple algorithms.
>
> I will repeat what I have said about the need for being able to associate=
:
>
> There must have been *some* reason that two m-lines were used, rather=20
> than just one. Each m-line heads a media section of the SDP that=20
> contains a bunch of declarations. Something must be different in those=20
> two media sections, or else a single media section would have been=20
> enough. Presumably whatever it is that is different is intended to=20
> affect how received packets are processed or interpreted. (If not,=20
> again there is no need for it to be there.) If you can't associate the=20
> packet with the m-line, then you don't know which of the multiple=20
> interpretations to apply to the packet.
>
> Maybe there is some exception to this, though I haven't come up with=20
> anything. If there is, then I hope someone will call it out. If so,=20
> then the constraint can be tightened.
>
> 	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 jonathan@vidyo.com  Tue Jul  2 11:56:57 2013
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Date: Tue, 2 Jul 2013 14:56:48 -0400
Thread-Topic: [MMUSIC] FW: New Version	Notification	for draft-even-mmusic-application-token-00.txt
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On Jul 1, 2013, at 1:02 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> It still isn't clear to me. You have three ways to communicate the mappin=
g:
>=20
> - SDP
> - SDES
> - header extension
>=20
> I think the relationship and precedence between SDES and header=20
> extension is well defined in general, so I guess that doesn't need=20
> further elaboration.

Actually, it probably needs some further elaboration in cases where RTP pac=
kets arrive out of order (either with respect to themselves or with their R=
TCP).  But straightforward rules for this can probably be defined fairly ea=
sily.=20

> But the relationship between the SDP and the RTP/RTCP mechanisms isn't=20
> so clear. This is especially problematic because it is impossible to=20
> synchronize the arrival of SDP with the arrival of RTP/RTCP.
>=20
> The simplest answer would be to use the SDP as a *default* if the=20
> mapping hasn't been received in the media, with changes in the SDP=20
> having no effect on the mapping of a particular AppId once it has been=20
> mentioned in the media. But I don't know if that is a sufficient mechanis=
m.
>=20
> Alternatively, the mapping can be defined by whichever has been received=
=20
> most recently. But that may cause artifacts.
>=20
> I just think this needs to be made very clear.

If we have AppIDs in SDP, I definitely think you need an "RTP associations =
always win" semantic.

On further reflection, though, I'm now wondering whether to what extent it'=
s actually necessary to have per-SSRC AppIDs defined in SDP at all, since t=
he point of AppID is to define an *alternative* to signaling SSRCs in SDP, =
by instead defining application semantics with a layer of indirection.

I think (and this was a bit of a point of disagreement between Roni and me =
in writing the draft) that "Plan B"-style signaling with AppID should use a=
=3DAppID:token attribute:value signaling in place of RFC 5576 a=3Dssrc:numb=
er attribute:value signaling.  I.e., it's similar semantics to RFC 5576 sig=
naling, but with the extra indirection layer.

--
Jonathan Lennox
jonathan@vidyo.com



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Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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+1
(Except that I *am* Paul)

On 7/2/13 2:30 PM, Christer Holmberg wrote:
> Hi,
>
> I am not Paul, but just to verify that I understand his issue, the question is:
>
> Why do you need 4 m- lines for the thumbnails, when you could use 1 m- line?
>
> Regards,
>
> Christer
>
> -----Alkuperäinen viesti-----
> Lähettäjä: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] Puolesta Mo Zanaty (mzanaty)
> Lähetetty: 2. heinäkuuta 2013 19:09
> Vastaanottaja: Paul Kyzivat
> Kopio: mmusic@ietf.org
> Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>
> An attempt at "an answer that makes sense". (Just snippets, not full SDP.)
>
> m=video 10000 RTP/SAVPF 96
> a=rtpmap:96 H264/90000
> a=fmtp:96 profile-level-id=42001F <--- HD 720p30 main speaker
>
> m=video 10000 RTP/SAVPF 97
> a=rtpmap:97 H264/90000
> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>
> m=video 10000 RTP/SAVPF 97
> a=rtpmap:97 H264/90000
> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>
> m=video 10000 RTP/SAVPF 97
> a=rtpmap:97 H264/90000
> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>
> m=video 10000 RTP/SAVPF 97
> a=rtpmap:97 H264/90000
> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>
> No need to map thumbnail packets to specific m-lines, so reuse of PT=97 is fine. Just render the last 4 thumbnail sources received (in addition to the main speaker).
>
> You could mandate that PT must be unique to allow unique mapping.
>
> You could mandate that multiple m-lines of thumbnails must be combined into a single m-line with a=ssrc/max-ssrc/multiple-render/whatever.
>
> Which do you want to mandate and why?
>
> Cheers,
> Mo
>
>
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Tuesday, July 02, 2013 10:06 AM
> To: Mo Zanaty (mzanaty)
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>
> On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
>> Informative descriptions of possible mappings are fine, but nothing normative, please. Some applications won't care about the mapping, and others may care but use a different mapping from those described. The only normative statement you could make is something like: "If you care about the mapping, you MUST have a mapping mechanism. You MAY use the mechanism(s) described here, or some other mechanism." But why mandate such a tautology?
>>
>> The assumption below that multiple m-lines always require different m-line-specific processing of the packets also assumes Plan B, i.e. that multiple streams with similar processing should always be signaled with a single m-line (for example, using a=ssrc or max-ssrc). Plan A purists would use multiple m-lines even for multiple streams with the same processing. I don't think we want to force bundle to pick a plan.
>
> I'm going to keep asking this question until I get an answer that makes
> sense:
>
> Please explain to me a meaningful situation when there are multiple m-lines in a bundle and there is insufficient information to associate each packet with an m-line.
>
> The *point* of plan A is that there be an m-line per "flow", and that the offerer is enumerating the flows. So clearly it knows the mapping.
> If the answerer isn't capable of doing the same mapping, then the use of plan A has failed.
>
> The part that I want to be normative is that an O/A is not valid unless the mapping is possible with the information in the O/A, and that the offerer and the answerer would agree about the mapping of each packet. I don't care if they actually do the mapping, though if they are doing any sensible processing then they will be doing something *equivalent* to the mapping.
>
> 	Thanks,
> 	Paul
>
>> Mo
>>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>> Behalf Of Paul Kyzivat
>> Sent: Saturday, June 29, 2013 12:12 PM
>> To: mmusic@ietf.org
>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>>
>> On 6/29/13 8:50 AM, Colin Perkins wrote:
>>> Christer,
>>>
>>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
>>>> There has been some discussions about whether BUNDLE should mandate that users are mandated to always (no matter what transport protocols are used in the BUNDLE group) have a mechanism to map received data to an m- line, or whether it from a generic BUNDLE perspective should be optional - IF there would be cases where it's not needed.
>>>>
>>>> We haven't had that much discussion about it yet, so I will not ask
>>>> a DECISION question at this point, but I would really like to get
>>>> some input from people who have opinions about this :)
>>>
>>> My opinion: BUNDLE needs to mandate a single algorithm for mapping from RTP flows to m= lines, for those applications that care about such a mapping. I don't think it should mandate that all applications need to care.
>>
>> I just replied elsewhere on this.
>> My opinion turns out to be the exact opposite of yours: we need to
>> support multiple algorithms, and that all applications should care.
>>
>> I won't repeat the stuff about multiple algorithms.
>>
>> I will repeat what I have said about the need for being able to associate:
>>
>> There must have been *some* reason that two m-lines were used, rather
>> than just one. Each m-line heads a media section of the SDP that
>> contains a bunch of declarations. Something must be different in those
>> two media sections, or else a single media section would have been
>> enough. Presumably whatever it is that is different is intended to
>> affect how received packets are processed or interpreted. (If not,
>> again there is no need for it to be there.) If you can't associate the
>> packet with the m-line, then you don't know which of the multiple
>> interpretations to apply to the packet.
>>
>> Maybe there is some exception to this, though I haven't come up with
>> anything. If there is, then I hope someone will call it out. If so,
>> then the constraint can be tightened.
>>
>> 	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 mzanaty@cisco.com  Tue Jul  2 13:33:33 2013
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
Thread-Index: AQHOdON5iUTc8DI+/Em0svjaZWQNnJlPEy+AgAKxRwD//7wNkIAAjfgAgAAZDgD//7QIEA==
Date: Tue, 2 Jul 2013 20:33:02 +0000
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References: <7594FB04B1934943A5C02806D1A2204B1C3BA9EF@ESESSMB209.ericsson.se> <511DF9CE-FEDD-4175-BF36-D44ACBE285DE@csperkins.org> <51CF0768.7030504@alum.mit.edu> <3879D71E758A7E4AA99A35DD8D41D3D91D48CF17@xmb-rcd-x14.cisco.com> <51D2DE2A.20300@alum.mit.edu> <3879D71E758A7E4AA99A35DD8D41D3D91D48DBCF@xmb-rcd-x14.cisco.com> <7594FB04B1934943A5C02806D1A2204B1C3C19FE@ESESSMB209.ericsson.se> <51D33146.9010402@alum.mit.edu>
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Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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> You could mandate that multiple m-lines of thumbnails must be combined in=
to a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.

It seems you both want to mandate this (Plan B). Why? I see no reason for b=
undle to pick a plan. Let the plan wars rage outside bundle.

Mo

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]=20
Sent: Tuesday, July 02, 2013 4:00 PM
To: Christer Holmberg
Cc: Mo Zanaty (mzanaty); mmusic@ietf.org
Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?

+1
(Except that I *am* Paul)

On 7/2/13 2:30 PM, Christer Holmberg wrote:
> Hi,
>
> I am not Paul, but just to verify that I understand his issue, the questi=
on is:
>
> Why do you need 4 m- lines for the thumbnails, when you could use 1 m- li=
ne?
>
> Regards,
>
> Christer
>
> -----Alkuper=E4inen viesti-----
> L=E4hett=E4j=E4: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org]=
 Puolesta Mo Zanaty (mzanaty)
> L=E4hetetty: 2. hein=E4kuuta 2013 19:09
> Vastaanottaja: Paul Kyzivat
> Kopio: mmusic@ietf.org
> Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map rece=
ived data to m- line?
>
> An attempt at "an answer that makes sense". (Just snippets, not full SDP.=
)
>
> m=3Dvideo 10000 RTP/SAVPF 96
> a=3Drtpmap:96 H264/90000
> a=3Dfmtp:96 profile-level-id=3D42001F <--- HD 720p30 main speaker
>
> m=3Dvideo 10000 RTP/SAVPF 97
> a=3Drtpmap:97 H264/90000
> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvon=
ly
>
> m=3Dvideo 10000 RTP/SAVPF 97
> a=3Drtpmap:97 H264/90000
> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvon=
ly
>
> m=3Dvideo 10000 RTP/SAVPF 97
> a=3Drtpmap:97 H264/90000
> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvon=
ly
>
> m=3Dvideo 10000 RTP/SAVPF 97
> a=3Drtpmap:97 H264/90000
> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvon=
ly
>
> No need to map thumbnail packets to specific m-lines, so reuse of PT=3D97=
 is fine. Just render the last 4 thumbnail sources received (in addition to=
 the main speaker).
>
> You could mandate that PT must be unique to allow unique mapping.
>
> You could mandate that multiple m-lines of thumbnails must be combined in=
to a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.
>
> Which do you want to mandate and why?
>
> Cheers,
> Mo
>
>
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Tuesday, July 02, 2013 10:06 AM
> To: Mo Zanaty (mzanaty)
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map r=
eceived data to m- line?
>
> On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
>> Informative descriptions of possible mappings are fine, but nothing norm=
ative, please. Some applications won't care about the mapping, and others m=
ay care but use a different mapping from those described. The only normativ=
e statement you could make is something like: "If you care about the mappin=
g, you MUST have a mapping mechanism. You MAY use the mechanism(s) describe=
d here, or some other mechanism." But why mandate such a tautology?
>>
>> The assumption below that multiple m-lines always require different m-li=
ne-specific processing of the packets also assumes Plan B, i.e. that multip=
le streams with similar processing should always be signaled with a single =
m-line (for example, using a=3Dssrc or max-ssrc). Plan A purists would use =
multiple m-lines even for multiple streams with the same processing. I don'=
t think we want to force bundle to pick a plan.
>
> I'm going to keep asking this question until I get an answer that makes
> sense:
>
> Please explain to me a meaningful situation when there are multiple m-lin=
es in a bundle and there is insufficient information to associate each pack=
et with an m-line.
>
> The *point* of plan A is that there be an m-line per "flow", and that the=
 offerer is enumerating the flows. So clearly it knows the mapping.
> If the answerer isn't capable of doing the same mapping, then the use of =
plan A has failed.
>
> The part that I want to be normative is that an O/A is not valid unless t=
he mapping is possible with the information in the O/A, and that the offere=
r and the answerer would agree about the mapping of each packet. I don't ca=
re if they actually do the mapping, though if they are doing any sensible p=
rocessing then they will be doing something *equivalent* to the mapping.
>
> 	Thanks,
> 	Paul
>
>> Mo
>>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>> Behalf Of Paul Kyzivat
>> Sent: Saturday, June 29, 2013 12:12 PM
>> To: mmusic@ietf.org
>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map =
received data to m- line?
>>
>> On 6/29/13 8:50 AM, Colin Perkins wrote:
>>> Christer,
>>>
>>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
>>>> There has been some discussions about whether BUNDLE should mandate th=
at users are mandated to always (no matter what transport protocols are use=
d in the BUNDLE group) have a mechanism to map received data to an m- line,=
 or whether it from a generic BUNDLE perspective should be optional - IF th=
ere would be cases where it's not needed.
>>>>
>>>> We haven't had that much discussion about it yet, so I will not ask
>>>> a DECISION question at this point, but I would really like to get
>>>> some input from people who have opinions about this :)
>>>
>>> My opinion: BUNDLE needs to mandate a single algorithm for mapping from=
 RTP flows to m=3D lines, for those applications that care about such a map=
ping. I don't think it should mandate that all applications need to care.
>>
>> I just replied elsewhere on this.
>> My opinion turns out to be the exact opposite of yours: we need to
>> support multiple algorithms, and that all applications should care.
>>
>> I won't repeat the stuff about multiple algorithms.
>>
>> I will repeat what I have said about the need for being able to associat=
e:
>>
>> There must have been *some* reason that two m-lines were used, rather
>> than just one. Each m-line heads a media section of the SDP that
>> contains a bunch of declarations. Something must be different in those
>> two media sections, or else a single media section would have been
>> enough. Presumably whatever it is that is different is intended to
>> affect how received packets are processed or interpreted. (If not,
>> again there is no need for it to be there.) If you can't associate the
>> packet with the m-line, then you don't know which of the multiple
>> interpretations to apply to the packet.
>>
>> Maybe there is some exception to this, though I haven't come up with
>> anything. If there is, then I hope someone will call it out. If so,
>> then the constraint can be tightened.
>>
>> 	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 pkyzivat@alum.mit.edu  Tue Jul  2 15:40:14 2013
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References: <7594FB04B1934943A5C02806D1A2204B1C3BA9EF@ESESSMB209.ericsson.se> <511DF9CE-FEDD-4175-BF36-D44ACBE285DE@csperkins.org> <51CF0768.7030504@alum.mit.edu> <3879D71E758A7E4AA99A35DD8D41D3D91D48CF17@xmb-rcd-x14.cisco.com> <51D2DE2A.20300@alum.mit.edu> <3879D71E758A7E4AA99A35DD8D41D3D91D48DBCF@xmb-rcd-x14.cisco.com> <7594FB04B1934943A5C02806D1A2204B1C3C19FE@ESESSMB209.ericsson.se> <51D33146.9010402@alum.mit.edu> <3879D71E758A7E4AA99A35DD8D41D3D91D48DE77@xmb-rcd-x14.cisco.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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On 7/2/13 4:33 PM, Mo Zanaty (mzanaty) wrote:
>> You could mandate that multiple m-lines of thumbnails must be combined into a single m-line with a=ssrc/max-ssrc/multiple-render/whatever.
>
> It seems you both want to mandate this (Plan B). Why? I see no reason for bundle to pick a plan. Let the plan wars rage outside bundle.

So you want to use plan A *without* the association, so that the *only* 
purpose of the extra m-lines is to denote the number of flows?

How would the receiver know if an incoming packet was associated with 
one of the flows defined by the m-lines?

How would the receiver know if each ssrc should be treated as a 
different flow, or if appid should be used to define flows?

I think you abuse plan A. Its different with plan B or No Plan. There 
you still need the ability to associate packets to m-lines, but that 
isn't sufficient to map them to flows. It is assumed that "extra" 
information, learned by different means, is used to sort out the flows.

	Thanks,
	Paul

> Mo
>
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Tuesday, July 02, 2013 4:00 PM
> To: Christer Holmberg
> Cc: Mo Zanaty (mzanaty); mmusic@ietf.org
> Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>
> +1
> (Except that I *am* Paul)
>
> On 7/2/13 2:30 PM, Christer Holmberg wrote:
>> Hi,
>>
>> I am not Paul, but just to verify that I understand his issue, the question is:
>>
>> Why do you need 4 m- lines for the thumbnails, when you could use 1 m- line?
>>
>> Regards,
>>
>> Christer
>>
>> -----Alkuperäinen viesti-----
>> Lähettäjä: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] Puolesta Mo Zanaty (mzanaty)
>> Lähetetty: 2. heinäkuuta 2013 19:09
>> Vastaanottaja: Paul Kyzivat
>> Kopio: mmusic@ietf.org
>> Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>>
>> An attempt at "an answer that makes sense". (Just snippets, not full SDP.)
>>
>> m=video 10000 RTP/SAVPF 96
>> a=rtpmap:96 H264/90000
>> a=fmtp:96 profile-level-id=42001F <--- HD 720p30 main speaker
>>
>> m=video 10000 RTP/SAVPF 97
>> a=rtpmap:97 H264/90000
>> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>>
>> m=video 10000 RTP/SAVPF 97
>> a=rtpmap:97 H264/90000
>> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>>
>> m=video 10000 RTP/SAVPF 97
>> a=rtpmap:97 H264/90000
>> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>>
>> m=video 10000 RTP/SAVPF 97
>> a=rtpmap:97 H264/90000
>> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>>
>> No need to map thumbnail packets to specific m-lines, so reuse of PT=97 is fine. Just render the last 4 thumbnail sources received (in addition to the main speaker).
>>
>> You could mandate that PT must be unique to allow unique mapping.
>>
>> You could mandate that multiple m-lines of thumbnails must be combined into a single m-line with a=ssrc/max-ssrc/multiple-render/whatever.
>>
>> Which do you want to mandate and why?
>>
>> Cheers,
>> Mo
>>
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>> Sent: Tuesday, July 02, 2013 10:06 AM
>> To: Mo Zanaty (mzanaty)
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>>
>> On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
>>> Informative descriptions of possible mappings are fine, but nothing normative, please. Some applications won't care about the mapping, and others may care but use a different mapping from those described. The only normative statement you could make is something like: "If you care about the mapping, you MUST have a mapping mechanism. You MAY use the mechanism(s) described here, or some other mechanism." But why mandate such a tautology?
>>>
>>> The assumption below that multiple m-lines always require different m-line-specific processing of the packets also assumes Plan B, i.e. that multiple streams with similar processing should always be signaled with a single m-line (for example, using a=ssrc or max-ssrc). Plan A purists would use multiple m-lines even for multiple streams with the same processing. I don't think we want to force bundle to pick a plan.
>>
>> I'm going to keep asking this question until I get an answer that makes
>> sense:
>>
>> Please explain to me a meaningful situation when there are multiple m-lines in a bundle and there is insufficient information to associate each packet with an m-line.
>>
>> The *point* of plan A is that there be an m-line per "flow", and that the offerer is enumerating the flows. So clearly it knows the mapping.
>> If the answerer isn't capable of doing the same mapping, then the use of plan A has failed.
>>
>> The part that I want to be normative is that an O/A is not valid unless the mapping is possible with the information in the O/A, and that the offerer and the answerer would agree about the mapping of each packet. I don't care if they actually do the mapping, though if they are doing any sensible processing then they will be doing something *equivalent* to the mapping.
>>
>> 	Thanks,
>> 	Paul
>>
>>> Mo
>>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>> Behalf Of Paul Kyzivat
>>> Sent: Saturday, June 29, 2013 12:12 PM
>>> To: mmusic@ietf.org
>>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>>>
>>> On 6/29/13 8:50 AM, Colin Perkins wrote:
>>>> Christer,
>>>>
>>>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
>>>>> There has been some discussions about whether BUNDLE should mandate that users are mandated to always (no matter what transport protocols are used in the BUNDLE group) have a mechanism to map received data to an m- line, or whether it from a generic BUNDLE perspective should be optional - IF there would be cases where it's not needed.
>>>>>
>>>>> We haven't had that much discussion about it yet, so I will not ask
>>>>> a DECISION question at this point, but I would really like to get
>>>>> some input from people who have opinions about this :)
>>>>
>>>> My opinion: BUNDLE needs to mandate a single algorithm for mapping from RTP flows to m= lines, for those applications that care about such a mapping. I don't think it should mandate that all applications need to care.
>>>
>>> I just replied elsewhere on this.
>>> My opinion turns out to be the exact opposite of yours: we need to
>>> support multiple algorithms, and that all applications should care.
>>>
>>> I won't repeat the stuff about multiple algorithms.
>>>
>>> I will repeat what I have said about the need for being able to associate:
>>>
>>> There must have been *some* reason that two m-lines were used, rather
>>> than just one. Each m-line heads a media section of the SDP that
>>> contains a bunch of declarations. Something must be different in those
>>> two media sections, or else a single media section would have been
>>> enough. Presumably whatever it is that is different is intended to
>>> affect how received packets are processed or interpreted. (If not,
>>> again there is no need for it to be there.) If you can't associate the
>>> packet with the m-line, then you don't know which of the multiple
>>> interpretations to apply to the packet.
>>>
>>> Maybe there is some exception to this, though I haven't come up with
>>> anything. If there is, then I hope someone will call it out. If so,
>>> then the constraint can be tightened.
>>>
>>> 	Thanks,
>>> 	Paul
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
>


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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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Date: Wed, 3 Jul 2013 05:06:32 +0000
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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To clarify, I have no plan to use or abuse Plan A. The implementations I'm =
most interested in are all closer to Plan B. However, I'm aware of widely d=
eployed implementations (from other vendors) which are closer to Plan A in =
general, and specifically for the thumbnail example below (one m-line per t=
humbnail, all with the same port/PT). While I personally would not chose th=
is, I see no reason to forbid it, because I recognize it is just personal p=
reference.

Mo

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]=20
Sent: Tuesday, July 02, 2013 6:40 PM
To: Mo Zanaty (mzanaty)
Cc: Christer Holmberg; mmusic@ietf.org
Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?

On 7/2/13 4:33 PM, Mo Zanaty (mzanaty) wrote:
>> You could mandate that multiple m-lines of thumbnails must be combined i=
nto a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.
>
> It seems you both want to mandate this (Plan B). Why? I see no reason for=
 bundle to pick a plan. Let the plan wars rage outside bundle.

So you want to use plan A *without* the association, so that the *only*=20
purpose of the extra m-lines is to denote the number of flows?

How would the receiver know if an incoming packet was associated with=20
one of the flows defined by the m-lines?

How would the receiver know if each ssrc should be treated as a=20
different flow, or if appid should be used to define flows?

I think you abuse plan A. Its different with plan B or No Plan. There=20
you still need the ability to associate packets to m-lines, but that=20
isn't sufficient to map them to flows. It is assumed that "extra"=20
information, learned by different means, is used to sort out the flows.

	Thanks,
	Paul

> Mo
>
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Tuesday, July 02, 2013 4:00 PM
> To: Christer Holmberg
> Cc: Mo Zanaty (mzanaty); mmusic@ietf.org
> Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to m=
ap received data to m- line?
>
> +1
> (Except that I *am* Paul)
>
> On 7/2/13 2:30 PM, Christer Holmberg wrote:
>> Hi,
>>
>> I am not Paul, but just to verify that I understand his issue, the quest=
ion is:
>>
>> Why do you need 4 m- lines for the thumbnails, when you could use 1 m- l=
ine?
>>
>> Regards,
>>
>> Christer
>>
>> -----Alkuper=E4inen viesti-----
>> L=E4hett=E4j=E4: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org=
] Puolesta Mo Zanaty (mzanaty)
>> L=E4hetetty: 2. hein=E4kuuta 2013 19:09
>> Vastaanottaja: Paul Kyzivat
>> Kopio: mmusic@ietf.org
>> Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map rec=
eived data to m- line?
>>
>> An attempt at "an answer that makes sense". (Just snippets, not full SDP=
.)
>>
>> m=3Dvideo 10000 RTP/SAVPF 96
>> a=3Drtpmap:96 H264/90000
>> a=3Dfmtp:96 profile-level-id=3D42001F <--- HD 720p30 main speaker
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> No need to map thumbnail packets to specific m-lines, so reuse of PT=3D9=
7 is fine. Just render the last 4 thumbnail sources received (in addition t=
o the main speaker).
>>
>> You could mandate that PT must be unique to allow unique mapping.
>>
>> You could mandate that multiple m-lines of thumbnails must be combined i=
nto a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.
>>
>> Which do you want to mandate and why?
>>
>> Cheers,
>> Mo
>>
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>> Sent: Tuesday, July 02, 2013 10:06 AM
>> To: Mo Zanaty (mzanaty)
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map =
received data to m- line?
>>
>> On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
>>> Informative descriptions of possible mappings are fine, but nothing nor=
mative, please. Some applications won't care about the mapping, and others =
may care but use a different mapping from those described. The only normati=
ve statement you could make is something like: "If you care about the mappi=
ng, you MUST have a mapping mechanism. You MAY use the mechanism(s) describ=
ed here, or some other mechanism." But why mandate such a tautology?
>>>
>>> The assumption below that multiple m-lines always require different m-l=
ine-specific processing of the packets also assumes Plan B, i.e. that multi=
ple streams with similar processing should always be signaled with a single=
 m-line (for example, using a=3Dssrc or max-ssrc). Plan A purists would use=
 multiple m-lines even for multiple streams with the same processing. I don=
't think we want to force bundle to pick a plan.
>>
>> I'm going to keep asking this question until I get an answer that makes
>> sense:
>>
>> Please explain to me a meaningful situation when there are multiple m-li=
nes in a bundle and there is insufficient information to associate each pac=
ket with an m-line.
>>
>> The *point* of plan A is that there be an m-line per "flow", and that th=
e offerer is enumerating the flows. So clearly it knows the mapping.
>> If the answerer isn't capable of doing the same mapping, then the use of=
 plan A has failed.
>>
>> The part that I want to be normative is that an O/A is not valid unless =
the mapping is possible with the information in the O/A, and that the offer=
er and the answerer would agree about the mapping of each packet. I don't c=
are if they actually do the mapping, though if they are doing any sensible =
processing then they will be doing something *equivalent* to the mapping.
>>
>> 	Thanks,
>> 	Paul
>>
>>> Mo
>>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>> Behalf Of Paul Kyzivat
>>> Sent: Saturday, June 29, 2013 12:12 PM
>>> To: mmusic@ietf.org
>>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?
>>>
>>> On 6/29/13 8:50 AM, Colin Perkins wrote:
>>>> Christer,
>>>>
>>>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
>>>>> There has been some discussions about whether BUNDLE should mandate t=
hat users are mandated to always (no matter what transport protocols are us=
ed in the BUNDLE group) have a mechanism to map received data to an m- line=
, or whether it from a generic BUNDLE perspective should be optional - IF t=
here would be cases where it's not needed.
>>>>>
>>>>> We haven't had that much discussion about it yet, so I will not ask
>>>>> a DECISION question at this point, but I would really like to get
>>>>> some input from people who have opinions about this :)
>>>>
>>>> My opinion: BUNDLE needs to mandate a single algorithm for mapping fro=
m RTP flows to m=3D lines, for those applications that care about such a ma=
pping. I don't think it should mandate that all applications need to care.
>>>
>>> I just replied elsewhere on this.
>>> My opinion turns out to be the exact opposite of yours: we need to
>>> support multiple algorithms, and that all applications should care.
>>>
>>> I won't repeat the stuff about multiple algorithms.
>>>
>>> I will repeat what I have said about the need for being able to associa=
te:
>>>
>>> There must have been *some* reason that two m-lines were used, rather
>>> than just one. Each m-line heads a media section of the SDP that
>>> contains a bunch of declarations. Something must be different in those
>>> two media sections, or else a single media section would have been
>>> enough. Presumably whatever it is that is different is intended to
>>> affect how received packets are processed or interpreted. (If not,
>>> again there is no need for it to be there.) If you can't associate the
>>> packet with the m-line, then you don't know which of the multiple
>>> interpretations to apply to the packet.
>>>
>>> Maybe there is some exception to this, though I haven't come up with
>>> anything. If there is, then I hope someone will call it out. If so,
>>> then the constraint can be tightened.
>>>
>>> 	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 christer.holmberg@ericsson.com  Wed Jul  3 00:26:49 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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Date: Wed, 3 Jul 2013 07:26:34 +0000
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References: <7594FB04B1934943A5C02806D1A2204B1C3BA9EF@ESESSMB209.ericsson.se> <511DF9CE-FEDD-4175-BF36-D44ACBE285DE@csperkins.org> <51CF0768.7030504@alum.mit.edu> <3879D71E758A7E4AA99A35DD8D41D3D91D48CF17@xmb-rcd-x14.cisco.com> <51D2DE2A.20300@alum.mit.edu> <3879D71E758A7E4AA99A35DD8D41D3D91D48DBCF@xmb-rcd-x14.cisco.com> <7594FB04B1934943A5C02806D1A2204B1C3C19FE@ESESSMB209.ericsson.se> <51D33146.9010402@alum.mit.edu> <3879D71E758A7E4AA99A35DD8D41D3D91D48DE77@xmb-rcd-x14.cisco.com> <51D356C7.2030909@alum.mit.edu> <3879D71E758A7E4AA99A35DD8D41D3D91D48E258@xmb-rcd-x14.cisco.com>
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Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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Hi Mo,

I think what you are saying is that you want to have one thumbnail flow per=
 m- line. So far so good.

And, WITHOUT BUNDLE, you would be able to demux the flows based on the 5-tu=
ple (as each m- line would have a unique address:port). So far go good.

But, in your example, WITH BUNDLE, all 4 m- lines share the same 5-tuple, s=
o if you want to demux the streams you need to have SOME other mechanism th=
an the 5-tuple. Will you demux based on SSRC?

Regards,

Christer






-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: Mo Zanaty (mzanaty) [mailto:mzanaty@cisco.com]=20
L=E4hetetty: 3. hein=E4kuuta 2013 8:07
Vastaanottaja: Paul Kyzivat
Kopio: Christer Holmberg; mmusic@ietf.org
Aihe: RE: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map re=
ceived data to m- line?

To clarify, I have no plan to use or abuse Plan A. The implementations I'm =
most interested in are all closer to Plan B. However, I'm aware of widely d=
eployed implementations (from other vendors) which are closer to Plan A in =
general, and specifically for the thumbnail example below (one m-line per t=
humbnail, all with the same port/PT). While I personally would not chose th=
is, I see no reason to forbid it, because I recognize it is just personal p=
reference.

Mo

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
Sent: Tuesday, July 02, 2013 6:40 PM
To: Mo Zanaty (mzanaty)
Cc: Christer Holmberg; mmusic@ietf.org
Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?

On 7/2/13 4:33 PM, Mo Zanaty (mzanaty) wrote:
>> You could mandate that multiple m-lines of thumbnails must be combined i=
nto a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.
>
> It seems you both want to mandate this (Plan B). Why? I see no reason for=
 bundle to pick a plan. Let the plan wars rage outside bundle.

So you want to use plan A *without* the association, so that the *only* pur=
pose of the extra m-lines is to denote the number of flows?

How would the receiver know if an incoming packet was associated with one o=
f the flows defined by the m-lines?

How would the receiver know if each ssrc should be treated as a different f=
low, or if appid should be used to define flows?

I think you abuse plan A. Its different with plan B or No Plan. There you s=
till need the ability to associate packets to m-lines, but that isn't suffi=
cient to map them to flows. It is assumed that "extra"=20
information, learned by different means, is used to sort out the flows.

	Thanks,
	Paul

> Mo
>
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Tuesday, July 02, 2013 4:00 PM
> To: Christer Holmberg
> Cc: Mo Zanaty (mzanaty); mmusic@ietf.org
> Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to m=
ap received data to m- line?
>
> +1
> (Except that I *am* Paul)
>
> On 7/2/13 2:30 PM, Christer Holmberg wrote:
>> Hi,
>>
>> I am not Paul, but just to verify that I understand his issue, the quest=
ion is:
>>
>> Why do you need 4 m- lines for the thumbnails, when you could use 1 m- l=
ine?
>>
>> Regards,
>>
>> Christer
>>
>> -----Alkuper=E4inen viesti-----
>> L=E4hett=E4j=E4: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org=
]=20
>> Puolesta Mo Zanaty (mzanaty)
>> L=E4hetetty: 2. hein=E4kuuta 2013 19:09
>> Vastaanottaja: Paul Kyzivat
>> Kopio: mmusic@ietf.org
>> Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map rec=
eived data to m- line?
>>
>> An attempt at "an answer that makes sense". (Just snippets, not full=20
>> SDP.)
>>
>> m=3Dvideo 10000 RTP/SAVPF 96
>> a=3Drtpmap:96 H264/90000
>> a=3Dfmtp:96 profile-level-id=3D42001F <--- HD 720p30 main speaker
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> No need to map thumbnail packets to specific m-lines, so reuse of PT=3D9=
7 is fine. Just render the last 4 thumbnail sources received (in addition t=
o the main speaker).
>>
>> You could mandate that PT must be unique to allow unique mapping.
>>
>> You could mandate that multiple m-lines of thumbnails must be combined i=
nto a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.
>>
>> Which do you want to mandate and why?
>>
>> Cheers,
>> Mo
>>
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>> Sent: Tuesday, July 02, 2013 10:06 AM
>> To: Mo Zanaty (mzanaty)
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map =
received data to m- line?
>>
>> On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
>>> Informative descriptions of possible mappings are fine, but nothing nor=
mative, please. Some applications won't care about the mapping, and others =
may care but use a different mapping from those described. The only normati=
ve statement you could make is something like: "If you care about the mappi=
ng, you MUST have a mapping mechanism. You MAY use the mechanism(s) describ=
ed here, or some other mechanism." But why mandate such a tautology?
>>>
>>> The assumption below that multiple m-lines always require different m-l=
ine-specific processing of the packets also assumes Plan B, i.e. that multi=
ple streams with similar processing should always be signaled with a single=
 m-line (for example, using a=3Dssrc or max-ssrc). Plan A purists would use=
 multiple m-lines even for multiple streams with the same processing. I don=
't think we want to force bundle to pick a plan.
>>
>> I'm going to keep asking this question until I get an answer that=20
>> makes
>> sense:
>>
>> Please explain to me a meaningful situation when there are multiple m-li=
nes in a bundle and there is insufficient information to associate each pac=
ket with an m-line.
>>
>> The *point* of plan A is that there be an m-line per "flow", and that th=
e offerer is enumerating the flows. So clearly it knows the mapping.
>> If the answerer isn't capable of doing the same mapping, then the use of=
 plan A has failed.
>>
>> The part that I want to be normative is that an O/A is not valid unless =
the mapping is possible with the information in the O/A, and that the offer=
er and the answerer would agree about the mapping of each packet. I don't c=
are if they actually do the mapping, though if they are doing any sensible =
processing then they will be doing something *equivalent* to the mapping.
>>
>> 	Thanks,
>> 	Paul
>>
>>> Mo
>>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On=20
>>> Behalf Of Paul Kyzivat
>>> Sent: Saturday, June 29, 2013 12:12 PM
>>> To: mmusic@ietf.org
>>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?
>>>
>>> On 6/29/13 8:50 AM, Colin Perkins wrote:
>>>> Christer,
>>>>
>>>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
>>>>> There has been some discussions about whether BUNDLE should mandate t=
hat users are mandated to always (no matter what transport protocols are us=
ed in the BUNDLE group) have a mechanism to map received data to an m- line=
, or whether it from a generic BUNDLE perspective should be optional - IF t=
here would be cases where it's not needed.
>>>>>
>>>>> We haven't had that much discussion about it yet, so I will not=20
>>>>> ask a DECISION question at this point, but I would really like to=20
>>>>> get some input from people who have opinions about this :)
>>>>
>>>> My opinion: BUNDLE needs to mandate a single algorithm for mapping fro=
m RTP flows to m=3D lines, for those applications that care about such a ma=
pping. I don't think it should mandate that all applications need to care.
>>>
>>> I just replied elsewhere on this.
>>> My opinion turns out to be the exact opposite of yours: we need to=20
>>> support multiple algorithms, and that all applications should care.
>>>
>>> I won't repeat the stuff about multiple algorithms.
>>>
>>> I will repeat what I have said about the need for being able to associa=
te:
>>>
>>> There must have been *some* reason that two m-lines were used,=20
>>> rather than just one. Each m-line heads a media section of the SDP=20
>>> that contains a bunch of declarations. Something must be different=20
>>> in those two media sections, or else a single media section would=20
>>> have been enough. Presumably whatever it is that is different is=20
>>> intended to affect how received packets are processed or=20
>>> interpreted. (If not, again there is no need for it to be there.) If=20
>>> you can't associate the packet with the m-line, then you don't know=20
>>> which of the multiple interpretations to apply to the packet.
>>>
>>> Maybe there is some exception to this, though I haven't come up with=20
>>> anything. If there is, then I hope someone will call it out. If so,=20
>>> then the constraint can be tightened.
>>>
>>> 	Thanks,
>>> 	Paul
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
>


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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Date: Wed, 3 Jul 2013 11:38:00 +0000
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References: <7594FB04B1934943A5C02806D1A2204B1C3BA9EF@ESESSMB209.ericsson.se> <511DF9CE-FEDD-4175-BF36-D44ACBE285DE@csperkins.org> <51CF0768.7030504@alum.mit.edu> <3879D71E758A7E4AA99A35DD8D41D3D91D48CF17@xmb-rcd-x14.cisco.com> <51D2DE2A.20300@alum.mit.edu> <3879D71E758A7E4AA99A35DD8D41D3D91D48DBCF@xmb-rcd-x14.cisco.com> <7594FB04B1934943A5C02806D1A2204B1C3C19FE@ESESSMB209.ericsson.se> <51D33146.9010402@alum.mit.edu> <3879D71E758A7E4AA99A35DD8D41D3D91D48DE77@xmb-rcd-x14.cisco.com> <51D356C7.2030909@alum.mit.edu>, <3879D71E758A7E4AA99A35DD8D41D3D91D48E258@xmb-rcd-x14.cisco.com>
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Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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Hi,

I guess one advantage with 4 m- lines is that, if the remote endpoint does =
not support BUNDLE, it will allow to have a separate 5-tuple for each thumb=
nail stream, without having to send a "fallack" offer.

When both endpoints do support BUNDLE it makes little sense, but when think=
ing about it I really wonder if there is a reason to forbid it.


Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Mo Zanaty (mzanaty) [mzanaty@cisco.com]
To: Paul Kyzivat [pkyzivat@alum.mit.edu]
CC: Christer Holmberg [christer.holmberg@ericsson.com]; mmusic@ietf.org [mm=
usic@ietf.org]
Subject: RE: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?
To clarify, I have no plan to use or abuse Plan A. The implementations I'm =
most interested in are all closer to Plan B. However, I'm aware of widely d=
eployed implementations (from other vendors) which are closer to Plan A in =
general, and specifically for the thumbnail example below (one m-line per t=
humbnail, all with the same port/PT). While I personally would not chose th=
is, I see no reason to forbid it, because I recognize it is just personal p=
reference.

Mo

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
Sent: Tuesday, July 02, 2013 6:40 PM
To: Mo Zanaty (mzanaty)
Cc: Christer Holmberg; mmusic@ietf.org
Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?

On 7/2/13 4:33 PM, Mo Zanaty (mzanaty) wrote:
>> You could mandate that multiple m-lines of thumbnails must be combined i=
nto a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.
>
> It seems you both want to mandate this (Plan B). Why? I see no reason for=
 bundle to pick a plan. Let the plan wars rage outside bundle.

So you want to use plan A *without* the association, so that the *only*
purpose of the extra m-lines is to denote the number of flows?

How would the receiver know if an incoming packet was associated with
one of the flows defined by the m-lines?

How would the receiver know if each ssrc should be treated as a
different flow, or if appid should be used to define flows?

I think you abuse plan A. Its different with plan B or No Plan. There
you still need the ability to associate packets to m-lines, but that
isn't sufficient to map them to flows. It is assumed that "extra"
information, learned by different means, is used to sort out the flows.

        Thanks,
        Paul

> Mo
>
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Tuesday, July 02, 2013 4:00 PM
> To: Christer Holmberg
> Cc: Mo Zanaty (mzanaty); mmusic@ietf.org
> Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to m=
ap received data to m- line?
>
> +1
> (Except that I *am* Paul)
>
> On 7/2/13 2:30 PM, Christer Holmberg wrote:
>> Hi,
>>
>> I am not Paul, but just to verify that I understand his issue, the quest=
ion is:
>>
>> Why do you need 4 m- lines for the thumbnails, when you could use 1 m- l=
ine?
>>
>> Regards,
>>
>> Christer
>>
>> -----Alkuper=E4inen viesti-----
>> L=E4hett=E4j=E4: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org=
] Puolesta Mo Zanaty (mzanaty)
>> L=E4hetetty: 2. hein=E4kuuta 2013 19:09
>> Vastaanottaja: Paul Kyzivat
>> Kopio: mmusic@ietf.org
>> Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map rec=
eived data to m- line?
>>
>> An attempt at "an answer that makes sense". (Just snippets, not full SDP=
.)
>>
>> m=3Dvideo 10000 RTP/SAVPF 96
>> a=3Drtpmap:96 H264/90000
>> a=3Dfmtp:96 profile-level-id=3D42001F <--- HD 720p30 main speaker
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> No need to map thumbnail packets to specific m-lines, so reuse of PT=3D9=
7 is fine. Just render the last 4 thumbnail sources received (in addition t=
o the main speaker).
>>
>> You could mandate that PT must be unique to allow unique mapping.
>>
>> You could mandate that multiple m-lines of thumbnails must be combined i=
nto a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.
>>
>> Which do you want to mandate and why?
>>
>> Cheers,
>> Mo
>>
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>> Sent: Tuesday, July 02, 2013 10:06 AM
>> To: Mo Zanaty (mzanaty)
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map =
received data to m- line?
>>
>> On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
>>> Informative descriptions of possible mappings are fine, but nothing nor=
mative, please. Some applications won't care about the mapping, and others =
may care but use a different mapping from those described. The only normati=
ve statement you could make is something like: "If you care about the mappi=
ng, you MUST have a mapping mechanism. You MAY use the mechanism(s) describ=
ed here, or some other mechanism." But why mandate such a tautology?
>>>
>>> The assumption below that multiple m-lines always require different m-l=
ine-specific processing of the packets also assumes Plan B, i.e. that multi=
ple streams with similar processing should always be signaled with a single=
 m-line (for example, using a=3Dssrc or max-ssrc). Plan A purists would use=
 multiple m-lines even for multiple streams with the same processing. I don=
't think we want to force bundle to pick a plan.
>>
>> I'm going to keep asking this question until I get an answer that makes
>> sense:
>>
>> Please explain to me a meaningful situation when there are multiple m-li=
nes in a bundle and there is insufficient information to associate each pac=
ket with an m-line.
>>
>> The *point* of plan A is that there be an m-line per "flow", and that th=
e offerer is enumerating the flows. So clearly it knows the mapping.
>> If the answerer isn't capable of doing the same mapping, then the use of=
 plan A has failed.
>>
>> The part that I want to be normative is that an O/A is not valid unless =
the mapping is possible with the information in the O/A, and that the offer=
er and the answerer would agree about the mapping of each packet. I don't c=
are if they actually do the mapping, though if they are doing any sensible =
processing then they will be doing something *equivalent* to the mapping.
>>
>>       Thanks,
>>       Paul
>>
>>> Mo
>>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>> Behalf Of Paul Kyzivat
>>> Sent: Saturday, June 29, 2013 12:12 PM
>>> To: mmusic@ietf.org
>>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?
>>>
>>> On 6/29/13 8:50 AM, Colin Perkins wrote:
>>>> Christer,
>>>>
>>>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
>>>>> There has been some discussions about whether BUNDLE should mandate t=
hat users are mandated to always (no matter what transport protocols are us=
ed in the BUNDLE group) have a mechanism to map received data to an m- line=
, or whether it from a generic BUNDLE perspective should be optional - IF t=
here would be cases where it's not needed.
>>>>>
>>>>> We haven't had that much discussion about it yet, so I will not ask
>>>>> a DECISION question at this point, but I would really like to get
>>>>> some input from people who have opinions about this :)
>>>>
>>>> My opinion: BUNDLE needs to mandate a single algorithm for mapping fro=
m RTP flows to m=3D lines, for those applications that care about such a ma=
pping. I don't think it should mandate that all applications need to care.
>>>
>>> I just replied elsewhere on this.
>>> My opinion turns out to be the exact opposite of yours: we need to
>>> support multiple algorithms, and that all applications should care.
>>>
>>> I won't repeat the stuff about multiple algorithms.
>>>
>>> I will repeat what I have said about the need for being able to associa=
te:
>>>
>>> There must have been *some* reason that two m-lines were used, rather
>>> than just one. Each m-line heads a media section of the SDP that
>>> contains a bunch of declarations. Something must be different in those
>>> two media sections, or else a single media section would have been
>>> enough. Presumably whatever it is that is different is intended to
>>> affect how received packets are processed or interpreted. (If not,
>>> again there is no need for it to be there.) If you can't associate the
>>> packet with the m-line, then you don't know which of the multiple
>>> interpretations to apply to the packet.
>>>
>>> Maybe there is some exception to this, though I haven't come up with
>>> anything. If there is, then I hope someone will call it out. If so,
>>> then the constraint can be tightened.
>>>
>>>      Thanks,
>>>      Paul
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
>


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t; color:black">
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div><br>
Hi,</div>
<div>&nbsp;</div>
<div>I guess one advantage with 4 m- lines is that, if the remote endpoint =
does not support BUNDLE, it will allow to have a separate 5-tuple for each =
thumbnail stream, without having to send a &quot;fallack&quot; offer.</div>
<div>&nbsp;</div>
<div>When both endpoints do support BUNDLE it makes little sense, but when =
thinking about it I really wonder if there is a reason to forbid it.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Mo Zanaty (mzanaty) [mzanaty@cisco.com]<br>
<b>To:</b> Paul Kyzivat [pkyzivat@alum.mit.edu]<br>
<b>CC:</b> Christer Holmberg [christer.holmberg@ericsson.com]; mmusic@ietf.=
org [mmusic@ietf.org]<br>
<b>Subject:</b> RE: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism=
 to map received data to m- line?</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">To clarify, I have no plan to use or abuse Plan A.=
 The implementations I'm most interested in are all closer to Plan B. Howev=
er, I'm aware of widely deployed implementations (from other vendors) which=
 are closer to Plan A in general,
 and specifically for the thumbnail example below (one m-line per thumbnail=
, all with the same port/PT). While I personally would not chose this, I se=
e no reason to forbid it, because I recognize it is just personal preferenc=
e.<br>
<br>
Mo<br>
<br>
-----Original Message-----<br>
From: Paul Kyzivat [<a href=3D"mailto:pkyzivat@alum.mit.edu">mailto:pkyziva=
t@alum.mit.edu</a>]
<br>
Sent: Tuesday, July 02, 2013 6:40 PM<br>
To: Mo Zanaty (mzanaty)<br>
Cc: Christer Holmberg; mmusic@ietf.org<br>
Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?<br>
<br>
On 7/2/13 4:33 PM, Mo Zanaty (mzanaty) wrote:<br>
&gt;&gt; You could mandate that multiple m-lines of thumbnails must be comb=
ined into a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.<=
br>
&gt;<br>
&gt; It seems you both want to mandate this (Plan B). Why? I see no reason =
for bundle to pick a plan. Let the plan wars rage outside bundle.<br>
<br>
So you want to use plan A *without* the association, so that the *only* <br=
>
purpose of the extra m-lines is to denote the number of flows?<br>
<br>
How would the receiver know if an incoming packet was associated with <br>
one of the flows defined by the m-lines?<br>
<br>
How would the receiver know if each ssrc should be treated as a <br>
different flow, or if appid should be used to define flows?<br>
<br>
I think you abuse plan A. Its different with plan B or No Plan. There <br>
you still need the ability to associate packets to m-lines, but that <br>
isn't sufficient to map them to flows. It is assumed that &quot;extra&quot;=
 <br>
information, learned by different means, is used to sort out the flows.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
<br>
&gt; Mo<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: Paul Kyzivat [<a href=3D"mailto:pkyzivat@alum.mit.edu">mailto:pk=
yzivat@alum.mit.edu</a>]<br>
&gt; Sent: Tuesday, July 02, 2013 4:00 PM<br>
&gt; To: Christer Holmberg<br>
&gt; Cc: Mo Zanaty (mzanaty); mmusic@ietf.org<br>
&gt; Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism t=
o map received data to m- line?<br>
&gt;<br>
&gt; &#43;1<br>
&gt; (Except that I *am* Paul)<br>
&gt;<br>
&gt; On 7/2/13 2:30 PM, Christer Holmberg wrote:<br>
&gt;&gt; Hi,<br>
&gt;&gt;<br>
&gt;&gt; I am not Paul, but just to verify that I understand his issue, the=
 question is:<br>
&gt;&gt;<br>
&gt;&gt; Why do you need 4 m- lines for the thumbnails, when you could use =
1 m- line?<br>
&gt;&gt;<br>
&gt;&gt; Regards,<br>
&gt;&gt;<br>
&gt;&gt; Christer<br>
&gt;&gt;<br>
&gt;&gt; -----Alkuper=E4inen viesti-----<br>
&gt;&gt; L=E4hett=E4j=E4: mmusic-bounces@ietf.org [<a href=3D"mailto:mmusic=
-bounces@ietf.org">mailto:mmusic-bounces@ietf.org</a>] Puolesta Mo Zanaty (=
mzanaty)<br>
&gt;&gt; L=E4hetetty: 2. hein=E4kuuta 2013 19:09<br>
&gt;&gt; Vastaanottaja: Paul Kyzivat<br>
&gt;&gt; Kopio: mmusic@ietf.org<br>
&gt;&gt; Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to m=
ap received data to m- line?<br>
&gt;&gt;<br>
&gt;&gt; An attempt at &quot;an answer that makes sense&quot;. (Just snippe=
ts, not full SDP.)<br>
&gt;&gt;<br>
&gt;&gt; m=3Dvideo 10000 RTP/SAVPF 96<br>
&gt;&gt; a=3Drtpmap:96 H264/90000<br>
&gt;&gt; a=3Dfmtp:96 profile-level-id=3D42001F &lt;--- HD 720p30 main speak=
er<br>
&gt;&gt;<br>
&gt;&gt; m=3Dvideo 10000 RTP/SAVPF 97<br>
&gt;&gt; a=3Drtpmap:97 H264/90000<br>
&gt;&gt; a=3Dfmtp:97 profile-level-id=3D42000C &lt;--- SD 240p15 thumbnail =
a=3Drecvonly<br>
&gt;&gt;<br>
&gt;&gt; m=3Dvideo 10000 RTP/SAVPF 97<br>
&gt;&gt; a=3Drtpmap:97 H264/90000<br>
&gt;&gt; a=3Dfmtp:97 profile-level-id=3D42000C &lt;--- SD 240p15 thumbnail =
a=3Drecvonly<br>
&gt;&gt;<br>
&gt;&gt; m=3Dvideo 10000 RTP/SAVPF 97<br>
&gt;&gt; a=3Drtpmap:97 H264/90000<br>
&gt;&gt; a=3Dfmtp:97 profile-level-id=3D42000C &lt;--- SD 240p15 thumbnail =
a=3Drecvonly<br>
&gt;&gt;<br>
&gt;&gt; m=3Dvideo 10000 RTP/SAVPF 97<br>
&gt;&gt; a=3Drtpmap:97 H264/90000<br>
&gt;&gt; a=3Dfmtp:97 profile-level-id=3D42000C &lt;--- SD 240p15 thumbnail =
a=3Drecvonly<br>
&gt;&gt;<br>
&gt;&gt; No need to map thumbnail packets to specific m-lines, so reuse of =
PT=3D97 is fine. Just render the last 4 thumbnail sources received (in addi=
tion to the main speaker).<br>
&gt;&gt;<br>
&gt;&gt; You could mandate that PT must be unique to allow unique mapping.<=
br>
&gt;&gt;<br>
&gt;&gt; You could mandate that multiple m-lines of thumbnails must be comb=
ined into a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.<=
br>
&gt;&gt;<br>
&gt;&gt; Which do you want to mandate and why?<br>
&gt;&gt;<br>
&gt;&gt; Cheers,<br>
&gt;&gt; Mo<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; -----Original Message-----<br>
&gt;&gt; From: Paul Kyzivat [<a href=3D"mailto:pkyzivat@alum.mit.edu">mailt=
o:pkyzivat@alum.mit.edu</a>]<br>
&gt;&gt; Sent: Tuesday, July 02, 2013 10:06 AM<br>
&gt;&gt; To: Mo Zanaty (mzanaty)<br>
&gt;&gt; Cc: mmusic@ietf.org<br>
&gt;&gt; Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism t=
o map received data to m- line?<br>
&gt;&gt;<br>
&gt;&gt; On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:<br>
&gt;&gt;&gt; Informative descriptions of possible mappings are fine, but no=
thing normative, please. Some applications won't care about the mapping, an=
d others may care but use a different mapping from those described. The onl=
y normative statement you could make is something
 like: &quot;If you care about the mapping, you MUST have a mapping mechani=
sm. You MAY use the mechanism(s) described here, or some other mechanism.&q=
uot; But why mandate such a tautology?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; The assumption below that multiple m-lines always require diff=
erent m-line-specific processing of the packets also assumes Plan B, i.e. t=
hat multiple streams with similar processing should always be signaled with=
 a single m-line (for example, using a=3Dssrc
 or max-ssrc). Plan A purists would use multiple m-lines even for multiple =
streams with the same processing. I don't think we want to force bundle to =
pick a plan.<br>
&gt;&gt;<br>
&gt;&gt; I'm going to keep asking this question until I get an answer that =
makes<br>
&gt;&gt; sense:<br>
&gt;&gt;<br>
&gt;&gt; Please explain to me a meaningful situation when there are multipl=
e m-lines in a bundle and there is insufficient information to associate ea=
ch packet with an m-line.<br>
&gt;&gt;<br>
&gt;&gt; The *point* of plan A is that there be an m-line per &quot;flow&qu=
ot;, and that the offerer is enumerating the flows. So clearly it knows the=
 mapping.<br>
&gt;&gt; If the answerer isn't capable of doing the same mapping, then the =
use of plan A has failed.<br>
&gt;&gt;<br>
&gt;&gt; The part that I want to be normative is that an O/A is not valid u=
nless the mapping is possible with the information in the O/A, and that the=
 offerer and the answerer would agree about the mapping of each packet. I d=
on't care if they actually do the mapping,
 though if they are doing any sensible processing then they will be doing s=
omething *equivalent* to the mapping.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
&gt;&gt;<br>
&gt;&gt;&gt; Mo<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; -----Original Message-----<br>
&gt;&gt;&gt; From: mmusic-bounces@ietf.org [<a href=3D"mailto:mmusic-bounce=
s@ietf.org">mailto:mmusic-bounces@ietf.org</a>] On<br>
&gt;&gt;&gt; Behalf Of Paul Kyzivat<br>
&gt;&gt;&gt; Sent: Saturday, June 29, 2013 12:12 PM<br>
&gt;&gt;&gt; To: mmusic@ietf.org<br>
&gt;&gt;&gt; Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechani=
sm to map received data to m- line?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; On 6/29/13 8:50 AM, Colin Perkins wrote:<br>
&gt;&gt;&gt;&gt; Christer,<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; On 25 Jun 2013, at 12:16, Christer Holmberg wrote:<br>
&gt;&gt;&gt;&gt;&gt; There has been some discussions about whether BUNDLE s=
hould mandate that users are mandated to always (no matter what transport p=
rotocols are used in the BUNDLE group) have a mechanism to map received dat=
a to an m- line, or whether it from a generic BUNDLE
 perspective should be optional - IF there would be cases where it's not ne=
eded.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; We haven't had that much discussion about it yet, so I=
 will not ask<br>
&gt;&gt;&gt;&gt;&gt; a DECISION question at this point, but I would really =
like to get<br>
&gt;&gt;&gt;&gt;&gt; some input from people who have opinions about this :)=
<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; My opinion: BUNDLE needs to mandate a single algorithm for=
 mapping from RTP flows to m=3D lines, for those applications that care abo=
ut such a mapping. I don't think it should mandate that all applications ne=
ed to care.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I just replied elsewhere on this.<br>
&gt;&gt;&gt; My opinion turns out to be the exact opposite of yours: we nee=
d to<br>
&gt;&gt;&gt; support multiple algorithms, and that all applications should =
care.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I won't repeat the stuff about multiple algorithms.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I will repeat what I have said about the need for being able t=
o associate:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; There must have been *some* reason that two m-lines were used,=
 rather<br>
&gt;&gt;&gt; than just one. Each m-line heads a media section of the SDP th=
at<br>
&gt;&gt;&gt; contains a bunch of declarations. Something must be different =
in those<br>
&gt;&gt;&gt; two media sections, or else a single media section would have =
been<br>
&gt;&gt;&gt; enough. Presumably whatever it is that is different is intende=
d to<br>
&gt;&gt;&gt; affect how received packets are processed or interpreted. (If =
not,<br>
&gt;&gt;&gt; again there is no need for it to be there.) If you can't assoc=
iate the<br>
&gt;&gt;&gt; packet with the m-line, then you don't know which of the multi=
ple<br>
&gt;&gt;&gt; interpretations to apply to the packet.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Maybe there is some exception to this, though I haven't come u=
p with<br>
&gt;&gt;&gt; anything. If there is, then I hope someone will call it out. I=
f so,<br>
&gt;&gt;&gt; then the constraint can be tightened.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
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&gt;&gt;&gt;<br>
&gt;&gt;&gt; _______________________________________________<br>
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From emil@sip-communicator.org  Wed Jul  3 07:13:59 2013
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Date: Wed, 03 Jul 2013 16:13:26 +0200
From: Emil Ivov <emcho@jitsi.org>
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Subject: [MMUSIC] Trickle ICE for SIP Questions
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Hey all,

Christer, Enrico and I are preparing the next version of Trickle ICE for 
SIP. Now that discussions on BUNDLE and the plans seem to be winding 
down, we wanted to run a few questions by the working group.

Q1: Making reliable provisional responses and PRACK mandatory. Obviously 
this would be nice to avoid, so the question is: is there a reasonable 
mechanism to achieve this (and by reasonable, we mean something that 
wouldn't be harder than implementing support for PRACK).

There was some discussion about this back in April and there was a 
suggestion for implementing a 5245-style hack where the answerer 
basically resends the 180 until it knows that it has been received.

5245 uses connectivity checks for this (i.e. it stops 180 
retransmissions when the first connectivity check is received) but we 
don't have that option here since the 180 may contain either none or 
only host candidates so there are strong chances that no binding request 
would be received on them.

Thomas also suggested a second option which would be to also use INFO 
requests with trickled candidates as an indication that 180 was 
received. This however wouldn't work with half trickle so we are 
basically back to vanilla ICE for all (non-re) INVITEs.

Another option would be to mandate an INFO request with 
"end-of-candidates" in response to the 180, but that would be just the 
same as mandating PRACK support.

Thomas also suggested that the answerer can start sending INFOs right 
after it sends its answer in the 180 and then it can just resend the 180 
if the INFOs result in a 481 response.

Personally I think this could potentially be made to work, but it would 
imply a level of complexity that considerably exceeds PRACK support.

Opinions?

Q2: How do we send INFOs? Are they blocking or do we just send them in 
parallel? If the latter, then what happens when an INFO fails because it 
is received out of order? Do we just tell the application to resend the 
candidates asap?

This also leads to the following question:

Q3: What exactly do we send in INFOs? Just the latest batch of freshly 
learned candidates or all candidates we've learned so far? Dale 
suggested that if we do this cumulatively we wouldn't need to worry 
about the case with the out-of-order INFOs from Q2 since the information 
gets resent anyway. A drawback here would obviously be that this adds 
more complexity for trickle ICE users (WebRTC applications specifically)

A third option would be to allow both and leave it to the application.

Comments are most welcome!

Emil

-- 
https://jitsi.org

From partha@parthasarathi.co.in  Wed Jul  3 09:32:03 2013
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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Hi all,

In case BUNDLE, the same type of =93m=94 lines shall be diffentiated by =
SDP
label attribute? I'm asking this question as I heard that SSRC in RTP is =
not
created to differentiate RTP session.

In fact in SIPREC, the label attribute is used to differentiate the two
different "m" line of the same type.

Thanks
Partha

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Christer Holmberg
> Sent: Wednesday, July 03, 2013 12:57 PM
> To: Mo Zanaty (mzanaty); Paul Kyzivat
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to =
map
> received data to m- line?
>=20
> Hi Mo,
>=20
> I think what you are saying is that you want to have one thumbnail =
flow
> per m- line. So far so good.
>=20
> And, WITHOUT BUNDLE, you would be able to demux the flows based on the
> 5-tuple (as each m- line would have a unique address:port). So far go
> good.
>=20
> But, in your example, WITH BUNDLE, all 4 m- lines share the same 5-
> tuple, so if you want to demux the streams you need to have SOME other
> mechanism than the 5-tuple. Will you demux based on SSRC?
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
> -----Alkuper=E4inen viesti-----
> L=E4hett=E4j=E4: Mo Zanaty (mzanaty) [mailto:mzanaty@cisco.com]
> L=E4hetetty: 3. hein=E4kuuta 2013 8:07
> Vastaanottaja: Paul Kyzivat
> Kopio: Christer Holmberg; mmusic@ietf.org
> Aihe: RE: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to
> map received data to m- line?
>=20
> To clarify, I have no plan to use or abuse Plan A. The implementations
> I'm most interested in are all closer to Plan B. However, I'm aware of
> widely deployed implementations (from other vendors) which are closer
> to Plan A in general, and specifically for the thumbnail example below
> (one m-line per thumbnail, all with the same port/PT). While I
> personally would not chose this, I see no reason to forbid it, because
> I recognize it is just personal preference.
>=20
> Mo
>=20
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Tuesday, July 02, 2013 6:40 PM
> To: Mo Zanaty (mzanaty)
> Cc: Christer Holmberg; mmusic@ietf.org
> Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism =
to
> map received data to m- line?
>=20
> On 7/2/13 4:33 PM, Mo Zanaty (mzanaty) wrote:
> >> You could mandate that multiple m-lines of thumbnails must be
> combined into a single m-line with a=3Dssrc/max-ssrc/multiple-
> render/whatever.
> >
> > It seems you both want to mandate this (Plan B). Why? I see no =
reason
> for bundle to pick a plan. Let the plan wars rage outside bundle.
>=20
> So you want to use plan A *without* the association, so that the =
*only*
> purpose of the extra m-lines is to denote the number of flows?
>=20
> How would the receiver know if an incoming packet was associated with
> one of the flows defined by the m-lines?
>=20
> How would the receiver know if each ssrc should be treated as a
> different flow, or if appid should be used to define flows?
>=20
> I think you abuse plan A. Its different with plan B or No Plan. There
> you still need the ability to associate packets to m-lines, but that
> isn't sufficient to map them to flows. It is assumed that "extra"
> information, learned by different means, is used to sort out the =
flows.
>=20
> 	Thanks,
> 	Paul
>=20
> > Mo
> >
> > -----Original Message-----
> > From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> > Sent: Tuesday, July 02, 2013 4:00 PM
> > To: Christer Holmberg
> > Cc: Mo Zanaty (mzanaty); mmusic@ietf.org
> > Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism
> to map received data to m- line?
> >
> > +1
> > (Except that I *am* Paul)
> >
> > On 7/2/13 2:30 PM, Christer Holmberg wrote:
> >> Hi,
> >>
> >> I am not Paul, but just to verify that I understand his issue, the
> question is:
> >>
> >> Why do you need 4 m- lines for the thumbnails, when you could use 1
> m- line?
> >>
> >> Regards,
> >>
> >> Christer
> >>
> >> -----Alkuper=E4inen viesti-----
> >> L=E4hett=E4j=E4: mmusic-bounces@ietf.org =
[mailto:mmusic-bounces@ietf.org]
> >> Puolesta Mo Zanaty (mzanaty)
> >> L=E4hetetty: 2. hein=E4kuuta 2013 19:09
> >> Vastaanottaja: Paul Kyzivat
> >> Kopio: mmusic@ietf.org
> >> Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to =
map
> received data to m- line?
> >>
> >> An attempt at "an answer that makes sense". (Just snippets, not =
full
> >> SDP.)
> >>
> >> m=3Dvideo 10000 RTP/SAVPF 96
> >> a=3Drtpmap:96 H264/90000
> >> a=3Dfmtp:96 profile-level-id=3D42001F <--- HD 720p30 main speaker
> >>
> >> m=3Dvideo 10000 RTP/SAVPF 97
> >> a=3Drtpmap:97 H264/90000
> >> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail
> a=3Drecvonly
> >>
> >> m=3Dvideo 10000 RTP/SAVPF 97
> >> a=3Drtpmap:97 H264/90000
> >> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail
> a=3Drecvonly
> >>
> >> m=3Dvideo 10000 RTP/SAVPF 97
> >> a=3Drtpmap:97 H264/90000
> >> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail
> a=3Drecvonly
> >>
> >> m=3Dvideo 10000 RTP/SAVPF 97
> >> a=3Drtpmap:97 H264/90000
> >> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail
> a=3Drecvonly
> >>
> >> No need to map thumbnail packets to specific m-lines, so reuse of
> PT=3D97 is fine. Just render the last 4 thumbnail sources received (in
> addition to the main speaker).
> >>
> >> You could mandate that PT must be unique to allow unique mapping.
> >>
> >> You could mandate that multiple m-lines of thumbnails must be
> combined into a single m-line with a=3Dssrc/max-ssrc/multiple-
> render/whatever.
> >>
> >> Which do you want to mandate and why?
> >>
> >> Cheers,
> >> Mo
> >>
> >>
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> >> Sent: Tuesday, July 02, 2013 10:06 AM
> >> To: Mo Zanaty (mzanaty)
> >> Cc: mmusic@ietf.org
> >> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to
> map received data to m- line?
> >>
> >> On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
> >>> Informative descriptions of possible mappings are fine, but =
nothing
> normative, please. Some applications won't care about the mapping, and
> others may care but use a different mapping from those described. The
> only normative statement you could make is something like: "If you =
care
> about the mapping, you MUST have a mapping mechanism. You MAY use the
> mechanism(s) described here, or some other mechanism." But why mandate
> such a tautology?
> >>>
> >>> The assumption below that multiple m-lines always require =
different
> m-line-specific processing of the packets also assumes Plan B, i.e.
> that multiple streams with similar processing should always be =
signaled
> with a single m-line (for example, using a=3Dssrc or max-ssrc). Plan A
> purists would use multiple m-lines even for multiple streams with the
> same processing. I don't think we want to force bundle to pick a plan.
> >>
> >> I'm going to keep asking this question until I get an answer that
> >> makes
> >> sense:
> >>
> >> Please explain to me a meaningful situation when there are multiple
> m-lines in a bundle and there is insufficient information to associate
> each packet with an m-line.
> >>
> >> The *point* of plan A is that there be an m-line per "flow", and
> that the offerer is enumerating the flows. So clearly it knows the
> mapping.
> >> If the answerer isn't capable of doing the same mapping, then the
> use of plan A has failed.
> >>
> >> The part that I want to be normative is that an O/A is not valid
> unless the mapping is possible with the information in the O/A, and
> that the offerer and the answerer would agree about the mapping of =
each
> packet. I don't care if they actually do the mapping, though if they
> are doing any sensible processing then they will be doing something
> *equivalent* to the mapping.
> >>
> >> 	Thanks,
> >> 	Paul
> >>
> >>> Mo
> >>>
> >>> -----Original Message-----
> >>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >>> Behalf Of Paul Kyzivat
> >>> Sent: Saturday, June 29, 2013 12:12 PM
> >>> To: mmusic@ietf.org
> >>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism =
to
> map received data to m- line?
> >>>
> >>> On 6/29/13 8:50 AM, Colin Perkins wrote:
> >>>> Christer,
> >>>>
> >>>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
> >>>>> There has been some discussions about whether BUNDLE should
> mandate that users are mandated to always (no matter what transport
> protocols are used in the BUNDLE group) have a mechanism to map
> received data to an m- line, or whether it from a generic BUNDLE
> perspective should be optional - IF there would be cases where it's =
not
> needed.
> >>>>>
> >>>>> We haven't had that much discussion about it yet, so I will not
> >>>>> ask a DECISION question at this point, but I would really like =
to
> >>>>> get some input from people who have opinions about this :)
> >>>>
> >>>> My opinion: BUNDLE needs to mandate a single algorithm for =
mapping
> from RTP flows to m=3D lines, for those applications that care about =
such
> a mapping. I don't think it should mandate that all applications need
> to care.
> >>>
> >>> I just replied elsewhere on this.
> >>> My opinion turns out to be the exact opposite of yours: we need to
> >>> support multiple algorithms, and that all applications should =
care.
> >>>
> >>> I won't repeat the stuff about multiple algorithms.
> >>>
> >>> I will repeat what I have said about the need for being able to
> associate:
> >>>
> >>> There must have been *some* reason that two m-lines were used,
> >>> rather than just one. Each m-line heads a media section of the SDP
> >>> that contains a bunch of declarations. Something must be different
> >>> in those two media sections, or else a single media section would
> >>> have been enough. Presumably whatever it is that is different is
> >>> intended to affect how received packets are processed or
> >>> interpreted. (If not, again there is no need for it to be there.)
> If
> >>> you can't associate the packet with the m-line, then you don't =
know
> >>> which of the multiple interpretations to apply to the packet.
> >>>
> >>> Maybe there is some exception to this, though I haven't come up
> with
> >>> anything. If there is, then I hope someone will call it out. If =
so,
> >>> then the constraint can be tightened.
> >>>
> >>> 	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
> >>
> >
> >
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Parthasarathi R'" <partha@parthasarathi.co.in>, "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Mo Zanaty \(mzanaty\)'" <mzanaty@cisco.com>, "'Paul Kyzivat'" <pkyzivat@alum.mit.edu>
References: <7594FB04B1934943A5C02806D1A2204B1C3BA9EF@ESESSMB209.ericsson.se>	<511DF9CE-FEDD-4175-BF36-D44ACBE285DE@csperkins.org>	<51CF0768.7030504@alum.mit.edu>	<3879D71E758A7E4AA99A35DD8D41D3D91D48CF17@xmb-rcd-x14.cisco.com>	<51D2DE2A.20300@alum.mit.edu>	<3879D71E758A7E4AA99A35DD8D41D3D91D48DBCF@xmb-rcd-x14.cisco.com>	<7594FB04B1934943A5C02806D1A2204B1C3C19FE@ESESSMB209.ericsson.se>	<51D33146.9010402@alum.mit.edu>	<3879D71E758A7E4AA99A35DD8D41D3D91D48DE77@xmb-rcd-x14.cisco.com>	<51D356C7.2030909@alum.mit.edu>	<3879D71E758A7E4AA99A35DD8D41D3D91D48E258@xmb-rcd-x14.cisco.com>	<7594FB04B1934943A5C02806D1A2204B1C3C1CD1@ESESSMB209.ericsson.se> <00e101ce780a$d189acf0$749d06d0$@co.in>
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map	received data to m- line?
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Hi Christer,

Sorry, label attribute shall be used to distinguish in application layer
only and not on RTP layer demux as label is not possible to be pushed in =
the
RTP session.

AFAIK, the proposal to use SSRC or CNAME for distinguishing the RTP =
streams
are discussed in SIPREC but both are not selected.

Thanks
Partha

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Parthasarathi R
> Sent: Wednesday, July 03, 2013 10:02 PM
> To: 'Christer Holmberg'; 'Mo Zanaty (mzanaty)'; 'Paul Kyzivat'
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to =
map
> received data to m- line?
>=20
> Hi all,
>=20
> In case BUNDLE, the same type of =93m=94 lines shall be diffentiated =
by SDP
> label attribute? I'm asking this question as I heard that SSRC in RTP
> is not
> created to differentiate RTP session.
>=20
> In fact in SIPREC, the label attribute is used to differentiate the =
two
> different "m" line of the same type.
>=20
> Thanks
> Partha
>=20
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> > Behalf Of Christer Holmberg
> > Sent: Wednesday, July 03, 2013 12:57 PM
> > To: Mo Zanaty (mzanaty); Paul Kyzivat
> > Cc: mmusic@ietf.org
> > Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to
> map
> > received data to m- line?
> >
> > Hi Mo,
> >
> > I think what you are saying is that you want to have one thumbnail
> flow
> > per m- line. So far so good.
> >
> > And, WITHOUT BUNDLE, you would be able to demux the flows based on
> the
> > 5-tuple (as each m- line would have a unique address:port). So far =
go
> > good.
> >
> > But, in your example, WITH BUNDLE, all 4 m- lines share the same 5-
> > tuple, so if you want to demux the streams you need to have SOME
> other
> > mechanism than the 5-tuple. Will you demux based on SSRC?
> >
> > Regards,
> >
> > Christer
> >
> >
> >
> >
> >
> >
> > -----Alkuper=E4inen viesti-----
> > L=E4hett=E4j=E4: Mo Zanaty (mzanaty) [mailto:mzanaty@cisco.com]
> > L=E4hetetty: 3. hein=E4kuuta 2013 8:07
> > Vastaanottaja: Paul Kyzivat
> > Kopio: Christer Holmberg; mmusic@ietf.org
> > Aihe: RE: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to
> > map received data to m- line?
> >
> > To clarify, I have no plan to use or abuse Plan A. The
> implementations
> > I'm most interested in are all closer to Plan B. However, I'm aware
> of
> > widely deployed implementations (from other vendors) which are =
closer
> > to Plan A in general, and specifically for the thumbnail example
> below
> > (one m-line per thumbnail, all with the same port/PT). While I
> > personally would not chose this, I see no reason to forbid it,
> because
> > I recognize it is just personal preference.
> >
> > Mo
> >
> > -----Original Message-----
> > From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> > Sent: Tuesday, July 02, 2013 6:40 PM
> > To: Mo Zanaty (mzanaty)
> > Cc: Christer Holmberg; mmusic@ietf.org
> > Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism
> to
> > map received data to m- line?
> >
> > On 7/2/13 4:33 PM, Mo Zanaty (mzanaty) wrote:
> > >> You could mandate that multiple m-lines of thumbnails must be
> > combined into a single m-line with a=3Dssrc/max-ssrc/multiple-
> > render/whatever.
> > >
> > > It seems you both want to mandate this (Plan B). Why? I see no
> reason
> > for bundle to pick a plan. Let the plan wars rage outside bundle.
> >
> > So you want to use plan A *without* the association, so that the
> *only*
> > purpose of the extra m-lines is to denote the number of flows?
> >
> > How would the receiver know if an incoming packet was associated =
with
> > one of the flows defined by the m-lines?
> >
> > How would the receiver know if each ssrc should be treated as a
> > different flow, or if appid should be used to define flows?
> >
> > I think you abuse plan A. Its different with plan B or No Plan. =
There
> > you still need the ability to associate packets to m-lines, but that
> > isn't sufficient to map them to flows. It is assumed that "extra"
> > information, learned by different means, is used to sort out the
> flows.
> >
> > 	Thanks,
> > 	Paul
> >
> > > Mo
> > >
> > > -----Original Message-----
> > > From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> > > Sent: Tuesday, July 02, 2013 4:00 PM
> > > To: Christer Holmberg
> > > Cc: Mo Zanaty (mzanaty); mmusic@ietf.org
> > > Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate
> mechanism
> > to map received data to m- line?
> > >
> > > +1
> > > (Except that I *am* Paul)
> > >
> > > On 7/2/13 2:30 PM, Christer Holmberg wrote:
> > >> Hi,
> > >>
> > >> I am not Paul, but just to verify that I understand his issue, =
the
> > question is:
> > >>
> > >> Why do you need 4 m- lines for the thumbnails, when you could use
> 1
> > m- line?
> > >>
> > >> Regards,
> > >>
> > >> Christer
> > >>
> > >> -----Alkuper=E4inen viesti-----
> > >> L=E4hett=E4j=E4: mmusic-bounces@ietf.org [mailto:mmusic-
> bounces@ietf.org]
> > >> Puolesta Mo Zanaty (mzanaty)
> > >> L=E4hetetty: 2. hein=E4kuuta 2013 19:09
> > >> Vastaanottaja: Paul Kyzivat
> > >> Kopio: mmusic@ietf.org
> > >> Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to
> map
> > received data to m- line?
> > >>
> > >> An attempt at "an answer that makes sense". (Just snippets, not
> full
> > >> SDP.)
> > >>
> > >> m=3Dvideo 10000 RTP/SAVPF 96
> > >> a=3Drtpmap:96 H264/90000
> > >> a=3Dfmtp:96 profile-level-id=3D42001F <--- HD 720p30 main speaker
> > >>
> > >> m=3Dvideo 10000 RTP/SAVPF 97
> > >> a=3Drtpmap:97 H264/90000
> > >> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail
> > a=3Drecvonly
> > >>
> > >> m=3Dvideo 10000 RTP/SAVPF 97
> > >> a=3Drtpmap:97 H264/90000
> > >> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail
> > a=3Drecvonly
> > >>
> > >> m=3Dvideo 10000 RTP/SAVPF 97
> > >> a=3Drtpmap:97 H264/90000
> > >> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail
> > a=3Drecvonly
> > >>
> > >> m=3Dvideo 10000 RTP/SAVPF 97
> > >> a=3Drtpmap:97 H264/90000
> > >> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail
> > a=3Drecvonly
> > >>
> > >> No need to map thumbnail packets to specific m-lines, so reuse of
> > PT=3D97 is fine. Just render the last 4 thumbnail sources received =
(in
> > addition to the main speaker).
> > >>
> > >> You could mandate that PT must be unique to allow unique mapping.
> > >>
> > >> You could mandate that multiple m-lines of thumbnails must be
> > combined into a single m-line with a=3Dssrc/max-ssrc/multiple-
> > render/whatever.
> > >>
> > >> Which do you want to mandate and why?
> > >>
> > >> Cheers,
> > >> Mo
> > >>
> > >>
> > >> -----Original Message-----
> > >> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> > >> Sent: Tuesday, July 02, 2013 10:06 AM
> > >> To: Mo Zanaty (mzanaty)
> > >> Cc: mmusic@ietf.org
> > >> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism
> to
> > map received data to m- line?
> > >>
> > >> On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
> > >>> Informative descriptions of possible mappings are fine, but
> nothing
> > normative, please. Some applications won't care about the mapping,
> and
> > others may care but use a different mapping from those described. =
The
> > only normative statement you could make is something like: "If you
> care
> > about the mapping, you MUST have a mapping mechanism. You MAY use =
the
> > mechanism(s) described here, or some other mechanism." But why
> mandate
> > such a tautology?
> > >>>
> > >>> The assumption below that multiple m-lines always require
> different
> > m-line-specific processing of the packets also assumes Plan B, i.e.
> > that multiple streams with similar processing should always be
> signaled
> > with a single m-line (for example, using a=3Dssrc or max-ssrc). Plan =
A
> > purists would use multiple m-lines even for multiple streams with =
the
> > same processing. I don't think we want to force bundle to pick a
> plan.
> > >>
> > >> I'm going to keep asking this question until I get an answer that
> > >> makes
> > >> sense:
> > >>
> > >> Please explain to me a meaningful situation when there are
> multiple
> > m-lines in a bundle and there is insufficient information to
> associate
> > each packet with an m-line.
> > >>
> > >> The *point* of plan A is that there be an m-line per "flow", and
> > that the offerer is enumerating the flows. So clearly it knows the
> > mapping.
> > >> If the answerer isn't capable of doing the same mapping, then the
> > use of plan A has failed.
> > >>
> > >> The part that I want to be normative is that an O/A is not valid
> > unless the mapping is possible with the information in the O/A, and
> > that the offerer and the answerer would agree about the mapping of
> each
> > packet. I don't care if they actually do the mapping, though if they
> > are doing any sensible processing then they will be doing something
> > *equivalent* to the mapping.
> > >>
> > >> 	Thanks,
> > >> 	Paul
> > >>
> > >>> Mo
> > >>>
> > >>> -----Original Message-----
> > >>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] =
On
> > >>> Behalf Of Paul Kyzivat
> > >>> Sent: Saturday, June 29, 2013 12:12 PM
> > >>> To: mmusic@ietf.org
> > >>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism
> to
> > map received data to m- line?
> > >>>
> > >>> On 6/29/13 8:50 AM, Colin Perkins wrote:
> > >>>> Christer,
> > >>>>
> > >>>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
> > >>>>> There has been some discussions about whether BUNDLE should
> > mandate that users are mandated to always (no matter what transport
> > protocols are used in the BUNDLE group) have a mechanism to map
> > received data to an m- line, or whether it from a generic BUNDLE
> > perspective should be optional - IF there would be cases where it's
> not
> > needed.
> > >>>>>
> > >>>>> We haven't had that much discussion about it yet, so I will =
not
> > >>>>> ask a DECISION question at this point, but I would really like
> to
> > >>>>> get some input from people who have opinions about this :)
> > >>>>
> > >>>> My opinion: BUNDLE needs to mandate a single algorithm for
> mapping
> > from RTP flows to m=3D lines, for those applications that care about
> such
> > a mapping. I don't think it should mandate that all applications =
need
> > to care.
> > >>>
> > >>> I just replied elsewhere on this.
> > >>> My opinion turns out to be the exact opposite of yours: we need
> to
> > >>> support multiple algorithms, and that all applications should
> care.
> > >>>
> > >>> I won't repeat the stuff about multiple algorithms.
> > >>>
> > >>> I will repeat what I have said about the need for being able to
> > associate:
> > >>>
> > >>> There must have been *some* reason that two m-lines were used,
> > >>> rather than just one. Each m-line heads a media section of the
> SDP
> > >>> that contains a bunch of declarations. Something must be
> different
> > >>> in those two media sections, or else a single media section =
would
> > >>> have been enough. Presumably whatever it is that is different is
> > >>> intended to affect how received packets are processed or
> > >>> interpreted. (If not, again there is no need for it to be =
there.)
> > If
> > >>> you can't associate the packet with the m-line, then you don't
> know
> > >>> which of the multiple interpretations to apply to the packet.
> > >>>
> > >>> Maybe there is some exception to this, though I haven't come up
> > with
> > >>> anything. If there is, then I hope someone will call it out. If
> so,
> > >>> then the constraint can be tightened.
> > >>>
> > >>> 	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
> > >>
> > >
> > >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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Subject: Re: [MMUSIC] comments on draft-ietf-mmusic-sdp-g723-g729-03
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Hi Aurelien,

Sorry for the delay response. Thanks for the review comments. 

Please read inline.

Thanks
Partha

> -----Original Message-----
> From: aurelien.sollaud@orange.com [mailto:aurelien.sollaud@orange.com]
> Sent: Thursday, June 27, 2013 1:52 PM
> To: mmusic@ietf.org; mperumal@cisco.com; partha@parthasarathi.co.in
> Subject: comments on draft-ietf-mmusic-sdp-g723-g729-03
> 
> Hi
> 
> I reviewed draft-ietf-mmusic-sdp-g723-g729-03 and I have some comments.
> They are quite symmetrical between G723 and G729.
> 
> 
> ** Section 3
> 
> 1)
> -----
>    Based on the above, it is best not to use comfort noise frames if
> the
>    SDP offer or answer indicates that comfort noise is not supported.
> -----
> The words "it is best not to use" are not strong enough, because it can
> not work if one of the party do not support comfort noise frames (SID
> frames in the ITU-T Annexes specs). It is rather a "MUST NOT use".
> 
> The word "restrict the use" in the RFC3551 quote is to be interpreted
> as "forbid the use".
> 
<Partha> It is fine. We will add this as a normative statement in case no
objection </Partha>

> 
> 
> ** Section 3.1
> 
> 2)
> After the first paragraph, I suggest to insert the new paragraph below:
> 
> ---NEW
>  The Annex A of the codec concerns both sending and receiving, so both
>  sides of a bi-directional session MUST have the same setting.  If
>  one party indicates that it does not support Annex A, Annex A  must be
>  deactivated both ways.
> -----
> That defines the general rationale of the following paragraphs.
>
<Partha> I agree with that it is the objective of this O/A requirement but
how this statement qualifies for normative statement is not clear. From the
offerer perspective, there is no control on same setting in both sides. In
case you look at the combine requirements of this draft, this object is met.
</Partha>
 
> 
> 3)
> -----
>    When the offer has G723 with annexa=no then the answer MUST NOT have
>    annexa=yes for G723.  Thus the annexa parameter can be turned off by
>    the answerer, but cannot be turned on.
> -----
> I don't think this new requirement is needed.
> Anyway implementation that will not implement this RFC-to-be will not
> comply with it.
> 
<Partha> As per based G723 RFC, it is expected for all the implementation to
support annexa. This issue is discussed in
http://www.ietf.org/mail-archive/web/mmusic/current/msg11165.html </Partha> 

> Plus, with the following paragraph, it breaks the equivalence between
> annexa=yes and annexa omitted.
> 
> Note that RFC3264 allows the SDP answer to include media format that
> are not present in the SDP offer, so it could be the same for the Annex
> A support.
> 
<Partha> RFC 3264 recommends to define offer/answer behavior per media
format whereas it is not defined for G723 & G729 and it is addressed as part
of this draft </Partha>

> 
> 4)
> -----
>    When the offer has G723 with annexa=no, the reason for not mandating
>    that the answer MUST have annexa=no for G723 is that there are there
>    implementations that omit the annexa parameter in answer and expect
>    the least common denominator to be used.
> -----
> 
> Add the following sentence at the end of the paragraph, to be explicit
> about the "least common denominator":
>    In such cases the offerer and answerer MUST proceed as if G723 is
>    negotiated with annexa=no.
> 
<Partha> Agreed. We will update the text </Partha>
> 
> 
> ** Section 3.2
> 
> 5)
> After the first two paragraphs, I suggest to insert the new paragraph
> below:
> 
> ---NEW
>  The Annex B of the codec concerns both sending and receiving, so both
>  sides of a bi-directional session MUST have the same setting.  If
>  one party indicates that it does not support Annex B, Annex B must be
>  deactivated both ways.
> -----
> That defines the general rationale of the following paragraphs.
> 

<Partha> Same as comment 2 reply. I agree with that it is the objective of
this O/A requirement but how this statement qualifies for normative
statement is not clear. From the offerer perspective, there is no control on
same setting in both sides. In case you look at the combine requirements of
this draft, this object is met. </Partha>

> 
> 6)
> -----
>    When the offer or answer has G729 and the annexb parameter is
> absent,
>    it MUST be considered as if the offer or answer has G729 with
>    annexb=yes.
> -----
> This paragraph is redundant with what is said before.
> Remove it.
> 
> 

<Partha> As per based G729 RFC, it is expected for all the implementation to
support annexb. This issue is discussed in
http://www.ietf.org/mail-archive/web/mmusic/current/msg11165.html </Partha>

> 7)
> -----
>    When the offer has G729 with annexb=no then the answer MUST NOT have
>    annexb=yes for G729.  Thus the annexb parameter can be turned off by
>    the answerer, but cannot be turned on.
> -----
> I don't think this new requirement is needed.
> For the same reason as for G723
> 
<Partha> Same reply as comment 3. As per based G729 RFC, it is expected for
all the implementation to support annexb. This issue is discussed in
http://www.ietf.org/mail-archive/web/mmusic/current/msg11165.html </Partha>

> 
> 8)
> -----
>    When the offer has G729 with annexa=no, the reason for not mandating
>    that the answer MUST have annexa=no for G729 is that there are there
>    implementations that omit the annexa parameter in answer and expect
>    the least common denominator to be used.
> -----
> 
> Replace annexa by annexb  (copy/paste error I guess)
> 
<Partha> Agreed </Partha>
> And add the following sentence at the end of the paragraph, to be
> explicit about the "least common denominator":
>    In such cases the offerer and answerer MUST proceed as if G729 is
>    negotiated with annexb=no.
> 
<Parta> Agreed </Partha>
> 
> 
> Best regards,
> Aurelien
> 
> 
> _______________________________________________________________________
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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Hi Christer,

If by "demux" you mean classify as separate RTP streams, then yes, that is =
always by SSRC (regardless of SDP).

If by "demux" you mean map to unique SDP m-lines for unique context of othe=
r attributes, that unique context is not always necessary for all apps, hen=
ce why I prefer not mandating this.

I think you worded it well: "if you want to demux the streams you need to h=
ave SOME other mechanism". Emphasis on "if" (when demux refers to mapping t=
o unique SDP m-lines). This would be fine to mandate if you include the "if=
".

Cheers,
Mo

-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
Sent: Wednesday, July 03, 2013 3:27 AM
To: Mo Zanaty (mzanaty); Paul Kyzivat
Cc: mmusic@ietf.org
Subject: VS: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?

Hi Mo,

I think what you are saying is that you want to have one thumbnail flow per=
 m- line. So far so good.

And, WITHOUT BUNDLE, you would be able to demux the flows based on the 5-tu=
ple (as each m- line would have a unique address:port). So far go good.

But, in your example, WITH BUNDLE, all 4 m- lines share the same 5-tuple, s=
o if you want to demux the streams you need to have SOME other mechanism th=
an the 5-tuple. Will you demux based on SSRC?

Regards,

Christer






-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: Mo Zanaty (mzanaty) [mailto:mzanaty@cisco.com]=20
L=E4hetetty: 3. hein=E4kuuta 2013 8:07
Vastaanottaja: Paul Kyzivat
Kopio: Christer Holmberg; mmusic@ietf.org
Aihe: RE: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map re=
ceived data to m- line?

To clarify, I have no plan to use or abuse Plan A. The implementations I'm =
most interested in are all closer to Plan B. However, I'm aware of widely d=
eployed implementations (from other vendors) which are closer to Plan A in =
general, and specifically for the thumbnail example below (one m-line per t=
humbnail, all with the same port/PT). While I personally would not chose th=
is, I see no reason to forbid it, because I recognize it is just personal p=
reference.

Mo

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
Sent: Tuesday, July 02, 2013 6:40 PM
To: Mo Zanaty (mzanaty)
Cc: Christer Holmberg; mmusic@ietf.org
Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?

On 7/2/13 4:33 PM, Mo Zanaty (mzanaty) wrote:
>> You could mandate that multiple m-lines of thumbnails must be combined i=
nto a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.
>
> It seems you both want to mandate this (Plan B). Why? I see no reason for=
 bundle to pick a plan. Let the plan wars rage outside bundle.

So you want to use plan A *without* the association, so that the *only* pur=
pose of the extra m-lines is to denote the number of flows?

How would the receiver know if an incoming packet was associated with one o=
f the flows defined by the m-lines?

How would the receiver know if each ssrc should be treated as a different f=
low, or if appid should be used to define flows?

I think you abuse plan A. Its different with plan B or No Plan. There you s=
till need the ability to associate packets to m-lines, but that isn't suffi=
cient to map them to flows. It is assumed that "extra"=20
information, learned by different means, is used to sort out the flows.

	Thanks,
	Paul

> Mo
>
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Tuesday, July 02, 2013 4:00 PM
> To: Christer Holmberg
> Cc: Mo Zanaty (mzanaty); mmusic@ietf.org
> Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to m=
ap received data to m- line?
>
> +1
> (Except that I *am* Paul)
>
> On 7/2/13 2:30 PM, Christer Holmberg wrote:
>> Hi,
>>
>> I am not Paul, but just to verify that I understand his issue, the quest=
ion is:
>>
>> Why do you need 4 m- lines for the thumbnails, when you could use 1 m- l=
ine?
>>
>> Regards,
>>
>> Christer
>>
>> -----Alkuper=E4inen viesti-----
>> L=E4hett=E4j=E4: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org=
]=20
>> Puolesta Mo Zanaty (mzanaty)
>> L=E4hetetty: 2. hein=E4kuuta 2013 19:09
>> Vastaanottaja: Paul Kyzivat
>> Kopio: mmusic@ietf.org
>> Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map rec=
eived data to m- line?
>>
>> An attempt at "an answer that makes sense". (Just snippets, not full=20
>> SDP.)
>>
>> m=3Dvideo 10000 RTP/SAVPF 96
>> a=3Drtpmap:96 H264/90000
>> a=3Dfmtp:96 profile-level-id=3D42001F <--- HD 720p30 main speaker
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> No need to map thumbnail packets to specific m-lines, so reuse of PT=3D9=
7 is fine. Just render the last 4 thumbnail sources received (in addition t=
o the main speaker).
>>
>> You could mandate that PT must be unique to allow unique mapping.
>>
>> You could mandate that multiple m-lines of thumbnails must be combined i=
nto a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.
>>
>> Which do you want to mandate and why?
>>
>> Cheers,
>> Mo
>>
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>> Sent: Tuesday, July 02, 2013 10:06 AM
>> To: Mo Zanaty (mzanaty)
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map =
received data to m- line?
>>
>> On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
>>> Informative descriptions of possible mappings are fine, but nothing nor=
mative, please. Some applications won't care about the mapping, and others =
may care but use a different mapping from those described. The only normati=
ve statement you could make is something like: "If you care about the mappi=
ng, you MUST have a mapping mechanism. You MAY use the mechanism(s) describ=
ed here, or some other mechanism." But why mandate such a tautology?
>>>
>>> The assumption below that multiple m-lines always require different m-l=
ine-specific processing of the packets also assumes Plan B, i.e. that multi=
ple streams with similar processing should always be signaled with a single=
 m-line (for example, using a=3Dssrc or max-ssrc). Plan A purists would use=
 multiple m-lines even for multiple streams with the same processing. I don=
't think we want to force bundle to pick a plan.
>>
>> I'm going to keep asking this question until I get an answer that=20
>> makes
>> sense:
>>
>> Please explain to me a meaningful situation when there are multiple m-li=
nes in a bundle and there is insufficient information to associate each pac=
ket with an m-line.
>>
>> The *point* of plan A is that there be an m-line per "flow", and that th=
e offerer is enumerating the flows. So clearly it knows the mapping.
>> If the answerer isn't capable of doing the same mapping, then the use of=
 plan A has failed.
>>
>> The part that I want to be normative is that an O/A is not valid unless =
the mapping is possible with the information in the O/A, and that the offer=
er and the answerer would agree about the mapping of each packet. I don't c=
are if they actually do the mapping, though if they are doing any sensible =
processing then they will be doing something *equivalent* to the mapping.
>>
>> 	Thanks,
>> 	Paul
>>
>>> Mo
>>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On=20
>>> Behalf Of Paul Kyzivat
>>> Sent: Saturday, June 29, 2013 12:12 PM
>>> To: mmusic@ietf.org
>>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?
>>>
>>> On 6/29/13 8:50 AM, Colin Perkins wrote:
>>>> Christer,
>>>>
>>>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
>>>>> There has been some discussions about whether BUNDLE should mandate t=
hat users are mandated to always (no matter what transport protocols are us=
ed in the BUNDLE group) have a mechanism to map received data to an m- line=
, or whether it from a generic BUNDLE perspective should be optional - IF t=
here would be cases where it's not needed.
>>>>>
>>>>> We haven't had that much discussion about it yet, so I will not=20
>>>>> ask a DECISION question at this point, but I would really like to=20
>>>>> get some input from people who have opinions about this :)
>>>>
>>>> My opinion: BUNDLE needs to mandate a single algorithm for mapping fro=
m RTP flows to m=3D lines, for those applications that care about such a ma=
pping. I don't think it should mandate that all applications need to care.
>>>
>>> I just replied elsewhere on this.
>>> My opinion turns out to be the exact opposite of yours: we need to=20
>>> support multiple algorithms, and that all applications should care.
>>>
>>> I won't repeat the stuff about multiple algorithms.
>>>
>>> I will repeat what I have said about the need for being able to associa=
te:
>>>
>>> There must have been *some* reason that two m-lines were used,=20
>>> rather than just one. Each m-line heads a media section of the SDP=20
>>> that contains a bunch of declarations. Something must be different=20
>>> in those two media sections, or else a single media section would=20
>>> have been enough. Presumably whatever it is that is different is=20
>>> intended to affect how received packets are processed or=20
>>> interpreted. (If not, again there is no need for it to be there.) If=20
>>> you can't associate the packet with the m-line, then you don't know=20
>>> which of the multiple interpretations to apply to the packet.
>>>
>>> Maybe there is some exception to this, though I haven't come up with=20
>>> anything. If there is, then I hope someone will call it out. If so,=20
>>> then the constraint can be tightened.
>>>
>>> 	Thanks,
>>> 	Paul
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
>


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From: Dan Wing <dwing@cisco.com>
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To: Emil Ivov <emcho@jitsi.org>
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Cc: MMUSIC IETF WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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On Jul 3, 2013, at 7:13 AM, Emil Ivov <emcho@jitsi.org> wrote:

> Hey all,
>=20
> Christer, Enrico and I are preparing the next version of Trickle ICE =
for SIP. Now that discussions on BUNDLE and the plans seem to be winding =
down, we wanted to run a few questions by the working group.
>=20
> Q1: Making reliable provisional responses and PRACK mandatory. =
Obviously this would be nice to avoid, so the question is: is there a =
reasonable mechanism to achieve this (and by reasonable, we mean =
something that wouldn't be harder than implementing support for PRACK).
>=20
> There was some discussion about this back in April and there was a =
suggestion for implementing a 5245-style hack where the answerer =
basically resends the 180 until it knows that it has been received.
>=20
> 5245 uses connectivity checks for this (i.e. it stops 180 =
retransmissions when the first connectivity check is received) but we =
don't have that option here since the 180 may contain either none or =
only host candidates so there are strong chances that no binding request =
would be received on them.

I might work, because each re-transmission would probably have =
additional candidates (because the STUN/TURN transactions will have =
completed), and thus more likely that those re-transmissions (containing =
those additional candidates) would result in the remote endpoint sending =
a STUN binding request.

-d



> Thomas also suggested a second option which would be to also use INFO =
requests with trickled candidates as an indication that 180 was =
received. This however wouldn't work with half trickle so we are =
basically back to vanilla ICE for all (non-re) INVITEs.
>=20
> Another option would be to mandate an INFO request with =
"end-of-candidates" in response to the 180, but that would be just the =
same as mandating PRACK support.
>=20
> Thomas also suggested that the answerer can start sending INFOs right =
after it sends its answer in the 180 and then it can just resend the 180 =
if the INFOs result in a 481 response.
>=20
> Personally I think this could potentially be made to work, but it =
would imply a level of complexity that considerably exceeds PRACK =
support.
>=20
> Opinions?
>=20
> Q2: How do we send INFOs? Are they blocking or do we just send them in =
parallel? If the latter, then what happens when an INFO fails because it =
is received out of order? Do we just tell the application to resend the =
candidates asap?
>=20
> This also leads to the following question:
>=20
> Q3: What exactly do we send in INFOs? Just the latest batch of freshly =
learned candidates or all candidates we've learned so far? Dale =
suggested that if we do this cumulatively we wouldn't need to worry =
about the case with the out-of-order INFOs from Q2 since the information =
gets resent anyway. A drawback here would obviously be that this adds =
more complexity for trickle ICE users (WebRTC applications specifically)
>=20
> A third option would be to allow both and leave it to the application.
>=20
> Comments are most welcome!
>=20
> Emil
>=20
> --=20
> https://jitsi.org
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From emil@sip-communicator.org  Wed Jul  3 12:29:26 2013
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Date: Wed, 03 Jul 2013 21:29:18 +0200
From: Emil Ivov <emcho@jitsi.org>
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hey Dan,

On 03.07.13, 21:01, Dan Wing wrote:
>
> On Jul 3, 2013, at 7:13 AM, Emil Ivov <emcho@jitsi.org> wrote:
>
>> Hey all,
>>
>> Christer, Enrico and I are preparing the next version of Trickle
>> ICE for SIP. Now that discussions on BUNDLE and the plans seem to
>> be winding down, we wanted to run a few questions by the working
>> group.
>>
>> Q1: Making reliable provisional responses and PRACK mandatory.
>> Obviously this would be nice to avoid, so the question is: is there
>> a reasonable mechanism to achieve this (and by reasonable, we mean
>> something that wouldn't be harder than implementing support for
>> PRACK).
>>
>> There was some discussion about this back in April and there was a
>> suggestion for implementing a 5245-style hack where the answerer
>> basically resends the 180 until it knows that it has been
>> received.
>>
>> 5245 uses connectivity checks for this (i.e. it stops 180
>> retransmissions when the first connectivity check is received) but
>> we don't have that option here since the 180 may contain either
>> none or only host candidates so there are strong chances that no
>> binding request would be received on them.
>
> I might work, because each re-transmission would probably have
> additional candidates (because the STUN/TURN transactions will have
> completed), and thus more likely that those re-transmissions
> (containing those additional candidates) would result in the remote
> endpoint sending a STUN binding request.

If we do it cumulatively, then yes. We can then move to INFOs the moment 
we get either a first check (which would more often be the case with 
half trickle) or a first INFO request (which would be the case for full 
trickle).

This could indeed work. However it would also imply that we keep sending 
candidates cumulatively even in the INFO requests since we wouldn't know 
how many of the 180s got through.

This makes me slightly uncomfortable as it would imply that someone 
would have to maintain state on the sending side, and do a number of 
comparisons on the receiving side ...

Emil

--
https://jitsi.org



>> Thomas also suggested a second option which would be to also use
>> INFO requests with trickled candidates as an indication that 180
>> was received. This however wouldn't work with half trickle so we
>> are basically back to vanilla ICE for all (non-re) INVITEs.
>>
>> Another option would be to mandate an INFO request with
>> "end-of-candidates" in response to the 180, but that would be just
>> the same as mandating PRACK support.
>>
>> Thomas also suggested that the answerer can start sending INFOs
>> right after it sends its answer in the 180 and then it can just
>> resend the 180 if the INFOs result in a 481 response.
>>
>> Personally I think this could potentially be made to work, but it
>> would imply a level of complexity that considerably exceeds PRACK
>> support.
>>
>> Opinions?
>>
>> Q2: How do we send INFOs? Are they blocking or do we just send them
>> in parallel? If the latter, then what happens when an INFO fails
>> because it is received out of order? Do we just tell the
>> application to resend the candidates asap?
>>
>> This also leads to the following question:
>>
>> Q3: What exactly do we send in INFOs? Just the latest batch of
>> freshly learned candidates or all candidates we've learned so far?
>> Dale suggested that if we do this cumulatively we wouldn't need to
>> worry about the case with the out-of-order INFOs from Q2 since the
>> information gets resent anyway. A drawback here would obviously be
>> that this adds more complexity for trickle ICE users (WebRTC
>> applications specifically)
>>
>> A third option would be to allow both and leave it to the
>> application.
>>
>> Comments are most welcome!
>>
>> Emil

From christer.holmberg@ericsson.com  Thu Jul  4 00:10:49 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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Hi,

I meant mapping to unique m- line :)

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Mo Zanaty (mzanaty) [mzanaty@cisco.com]
To: Christer Holmberg [christer.holmberg@ericsson.com]; Paul Kyzivat [pkyzi=
vat@alum.mit.edu]
CC: mmusic@ietf.org [mmusic@ietf.org]
Subject: RE: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?
Hi Christer,

If by "demux" you mean classify as separate RTP streams, then yes, that is =
always by SSRC (regardless of SDP).

If by "demux" you mean map to unique SDP m-lines for unique context of othe=
r attributes, that unique context is not always necessary for all apps, hen=
ce why I prefer not mandating this.

I think you worded it well: "if you want to demux the streams you need to h=
ave SOME other mechanism". Emphasis on "if" (when demux refers to mapping t=
o unique SDP m-lines). This would be fine to mandate if you include the "if=
".

Cheers,
Mo

-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
Sent: Wednesday, July 03, 2013 3:27 AM
To: Mo Zanaty (mzanaty); Paul Kyzivat
Cc: mmusic@ietf.org
Subject: VS: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?

Hi Mo,

I think what you are saying is that you want to have one thumbnail flow per=
 m- line. So far so good.

And, WITHOUT BUNDLE, you would be able to demux the flows based on the 5-tu=
ple (as each m- line would have a unique address:port). So far go good.

But, in your example, WITH BUNDLE, all 4 m- lines share the same 5-tuple, s=
o if you want to demux the streams you need to have SOME other mechanism th=
an the 5-tuple. Will you demux based on SSRC?

Regards,

Christer






-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: Mo Zanaty (mzanaty) [mailto:mzanaty@cisco.com]
L=E4hetetty: 3. hein=E4kuuta 2013 8:07
Vastaanottaja: Paul Kyzivat
Kopio: Christer Holmberg; mmusic@ietf.org
Aihe: RE: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map re=
ceived data to m- line?

To clarify, I have no plan to use or abuse Plan A. The implementations I'm =
most interested in are all closer to Plan B. However, I'm aware of widely d=
eployed implementations (from other vendors) which are closer to Plan A in =
general, and specifically for the thumbnail example below (one m-line per t=
humbnail, all with the same port/PT). While I personally would not chose th=
is, I see no reason to forbid it, because I recognize it is just personal p=
reference.

Mo

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
Sent: Tuesday, July 02, 2013 6:40 PM
To: Mo Zanaty (mzanaty)
Cc: Christer Holmberg; mmusic@ietf.org
Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?

On 7/2/13 4:33 PM, Mo Zanaty (mzanaty) wrote:
>> You could mandate that multiple m-lines of thumbnails must be combined i=
nto a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.
>
> It seems you both want to mandate this (Plan B). Why? I see no reason for=
 bundle to pick a plan. Let the plan wars rage outside bundle.

So you want to use plan A *without* the association, so that the *only* pur=
pose of the extra m-lines is to denote the number of flows?

How would the receiver know if an incoming packet was associated with one o=
f the flows defined by the m-lines?

How would the receiver know if each ssrc should be treated as a different f=
low, or if appid should be used to define flows?

I think you abuse plan A. Its different with plan B or No Plan. There you s=
till need the ability to associate packets to m-lines, but that isn't suffi=
cient to map them to flows. It is assumed that "extra"
information, learned by different means, is used to sort out the flows.

        Thanks,
        Paul

> Mo
>
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Tuesday, July 02, 2013 4:00 PM
> To: Christer Holmberg
> Cc: Mo Zanaty (mzanaty); mmusic@ietf.org
> Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to m=
ap received data to m- line?
>
> +1
> (Except that I *am* Paul)
>
> On 7/2/13 2:30 PM, Christer Holmberg wrote:
>> Hi,
>>
>> I am not Paul, but just to verify that I understand his issue, the quest=
ion is:
>>
>> Why do you need 4 m- lines for the thumbnails, when you could use 1 m- l=
ine?
>>
>> Regards,
>>
>> Christer
>>
>> -----Alkuper=E4inen viesti-----
>> L=E4hett=E4j=E4: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org=
]
>> Puolesta Mo Zanaty (mzanaty)
>> L=E4hetetty: 2. hein=E4kuuta 2013 19:09
>> Vastaanottaja: Paul Kyzivat
>> Kopio: mmusic@ietf.org
>> Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map rec=
eived data to m- line?
>>
>> An attempt at "an answer that makes sense". (Just snippets, not full
>> SDP.)
>>
>> m=3Dvideo 10000 RTP/SAVPF 96
>> a=3Drtpmap:96 H264/90000
>> a=3Dfmtp:96 profile-level-id=3D42001F <--- HD 720p30 main speaker
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> m=3Dvideo 10000 RTP/SAVPF 97
>> a=3Drtpmap:97 H264/90000
>> a=3Dfmtp:97 profile-level-id=3D42000C <--- SD 240p15 thumbnail a=3Drecvo=
nly
>>
>> No need to map thumbnail packets to specific m-lines, so reuse of PT=3D9=
7 is fine. Just render the last 4 thumbnail sources received (in addition t=
o the main speaker).
>>
>> You could mandate that PT must be unique to allow unique mapping.
>>
>> You could mandate that multiple m-lines of thumbnails must be combined i=
nto a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.
>>
>> Which do you want to mandate and why?
>>
>> Cheers,
>> Mo
>>
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>> Sent: Tuesday, July 02, 2013 10:06 AM
>> To: Mo Zanaty (mzanaty)
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map =
received data to m- line?
>>
>> On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
>>> Informative descriptions of possible mappings are fine, but nothing nor=
mative, please. Some applications won't care about the mapping, and others =
may care but use a different mapping from those described. The only normati=
ve statement you could make is something like: "If you care about the mappi=
ng, you MUST have a mapping mechanism. You MAY use the mechanism(s) describ=
ed here, or some other mechanism." But why mandate such a tautology?
>>>
>>> The assumption below that multiple m-lines always require different m-l=
ine-specific processing of the packets also assumes Plan B, i.e. that multi=
ple streams with similar processing should always be signaled with a single=
 m-line (for example, using a=3Dssrc or max-ssrc). Plan A purists would use=
 multiple m-lines even for multiple streams with the same processing. I don=
't think we want to force bundle to pick a plan.
>>
>> I'm going to keep asking this question until I get an answer that
>> makes
>> sense:
>>
>> Please explain to me a meaningful situation when there are multiple m-li=
nes in a bundle and there is insufficient information to associate each pac=
ket with an m-line.
>>
>> The *point* of plan A is that there be an m-line per "flow", and that th=
e offerer is enumerating the flows. So clearly it knows the mapping.
>> If the answerer isn't capable of doing the same mapping, then the use of=
 plan A has failed.
>>
>> The part that I want to be normative is that an O/A is not valid unless =
the mapping is possible with the information in the O/A, and that the offer=
er and the answerer would agree about the mapping of each packet. I don't c=
are if they actually do the mapping, though if they are doing any sensible =
processing then they will be doing something *equivalent* to the mapping.
>>
>>       Thanks,
>>       Paul
>>
>>> Mo
>>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>> Behalf Of Paul Kyzivat
>>> Sent: Saturday, June 29, 2013 12:12 PM
>>> To: mmusic@ietf.org
>>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?
>>>
>>> On 6/29/13 8:50 AM, Colin Perkins wrote:
>>>> Christer,
>>>>
>>>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
>>>>> There has been some discussions about whether BUNDLE should mandate t=
hat users are mandated to always (no matter what transport protocols are us=
ed in the BUNDLE group) have a mechanism to map received data to an m- line=
, or whether it from a generic BUNDLE perspective should be optional - IF t=
here would be cases where it's not needed.
>>>>>
>>>>> We haven't had that much discussion about it yet, so I will not
>>>>> ask a DECISION question at this point, but I would really like to
>>>>> get some input from people who have opinions about this :)
>>>>
>>>> My opinion: BUNDLE needs to mandate a single algorithm for mapping fro=
m RTP flows to m=3D lines, for those applications that care about such a ma=
pping. I don't think it should mandate that all applications need to care.
>>>
>>> I just replied elsewhere on this.
>>> My opinion turns out to be the exact opposite of yours: we need to
>>> support multiple algorithms, and that all applications should care.
>>>
>>> I won't repeat the stuff about multiple algorithms.
>>>
>>> I will repeat what I have said about the need for being able to associa=
te:
>>>
>>> There must have been *some* reason that two m-lines were used,
>>> rather than just one. Each m-line heads a media section of the SDP
>>> that contains a bunch of declarations. Something must be different
>>> in those two media sections, or else a single media section would
>>> have been enough. Presumably whatever it is that is different is
>>> intended to affect how received packets are processed or
>>> interpreted. (If not, again there is no need for it to be there.) If
>>> you can't associate the packet with the m-line, then you don't know
>>> which of the multiple interpretations to apply to the packet.
>>>
>>> Maybe there is some exception to this, though I haven't come up with
>>> anything. If there is, then I hope someone will call it out. If so,
>>> then the constraint can be tightened.
>>>
>>>      Thanks,
>>>      Paul
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
>


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<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">I meant mapping to unique m- line :)</font></di=
v>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Mo Zanaty (mzanaty) [mzanaty@cisco.com]<br>
<b>To:</b> Christer Holmberg [christer.holmberg@ericsson.com]; Paul Kyzivat=
 [pkyzivat@alum.mit.edu]<br>
<b>CC:</b> mmusic@ietf.org [mmusic@ietf.org]<br>
<b>Subject:</b> RE: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism=
 to map received data to m- line?</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">Hi Christer,<br>
<br>
If by &quot;demux&quot; you mean classify as separate RTP streams, then yes=
, that is always by SSRC (regardless of SDP).<br>
<br>
If by &quot;demux&quot; you mean map to unique SDP m-lines for unique conte=
xt of other attributes, that unique context is not always necessary for all=
 apps, hence why I prefer not mandating this.<br>
<br>
I think you worded it well: &quot;if you want to demux the streams you need=
 to have SOME other mechanism&quot;. Emphasis on &quot;if&quot; (when demux=
 refers to mapping to unique SDP m-lines). This would be fine to mandate if=
 you include the &quot;if&quot;.<br>
<br>
Cheers,<br>
Mo<br>
<br>
-----Original Message-----<br>
From: Christer Holmberg [<a href=3D"mailto:christer.holmberg@ericsson.com">=
mailto:christer.holmberg@ericsson.com</a>]
<br>
Sent: Wednesday, July 03, 2013 3:27 AM<br>
To: Mo Zanaty (mzanaty); Paul Kyzivat<br>
Cc: mmusic@ietf.org<br>
Subject: VS: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?<br>
<br>
Hi Mo,<br>
<br>
I think what you are saying is that you want to have one thumbnail flow per=
 m- line. So far so good.<br>
<br>
And, WITHOUT BUNDLE, you would be able to demux the flows based on the 5-tu=
ple (as each m- line would have a unique address:port). So far go good.<br>
<br>
But, in your example, WITH BUNDLE, all 4 m- lines share the same 5-tuple, s=
o if you want to demux the streams you need to have SOME other mechanism th=
an the 5-tuple. Will you demux based on SSRC?<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
<br>
<br>
<br>
<br>
<br>
-----Alkuper=E4inen viesti-----<br>
L=E4hett=E4j=E4: Mo Zanaty (mzanaty) [<a href=3D"mailto:mzanaty@cisco.com">=
mailto:mzanaty@cisco.com</a>]
<br>
L=E4hetetty: 3. hein=E4kuuta 2013 8:07<br>
Vastaanottaja: Paul Kyzivat<br>
Kopio: Christer Holmberg; mmusic@ietf.org<br>
Aihe: RE: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map re=
ceived data to m- line?<br>
<br>
To clarify, I have no plan to use or abuse Plan A. The implementations I'm =
most interested in are all closer to Plan B. However, I'm aware of widely d=
eployed implementations (from other vendors) which are closer to Plan A in =
general, and specifically for the
 thumbnail example below (one m-line per thumbnail, all with the same port/=
PT). While I personally would not chose this, I see no reason to forbid it,=
 because I recognize it is just personal preference.<br>
<br>
Mo<br>
<br>
-----Original Message-----<br>
From: Paul Kyzivat [<a href=3D"mailto:pkyzivat@alum.mit.edu">mailto:pkyziva=
t@alum.mit.edu</a>]<br>
Sent: Tuesday, July 02, 2013 6:40 PM<br>
To: Mo Zanaty (mzanaty)<br>
Cc: Christer Holmberg; mmusic@ietf.org<br>
Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map=
 received data to m- line?<br>
<br>
On 7/2/13 4:33 PM, Mo Zanaty (mzanaty) wrote:<br>
&gt;&gt; You could mandate that multiple m-lines of thumbnails must be comb=
ined into a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.<=
br>
&gt;<br>
&gt; It seems you both want to mandate this (Plan B). Why? I see no reason =
for bundle to pick a plan. Let the plan wars rage outside bundle.<br>
<br>
So you want to use plan A *without* the association, so that the *only* pur=
pose of the extra m-lines is to denote the number of flows?<br>
<br>
How would the receiver know if an incoming packet was associated with one o=
f the flows defined by the m-lines?<br>
<br>
How would the receiver know if each ssrc should be treated as a different f=
low, or if appid should be used to define flows?<br>
<br>
I think you abuse plan A. Its different with plan B or No Plan. There you s=
till need the ability to associate packets to m-lines, but that isn't suffi=
cient to map them to flows. It is assumed that &quot;extra&quot;
<br>
information, learned by different means, is used to sort out the flows.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
<br>
&gt; Mo<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: Paul Kyzivat [<a href=3D"mailto:pkyzivat@alum.mit.edu">mailto:pk=
yzivat@alum.mit.edu</a>]<br>
&gt; Sent: Tuesday, July 02, 2013 4:00 PM<br>
&gt; To: Christer Holmberg<br>
&gt; Cc: Mo Zanaty (mzanaty); mmusic@ietf.org<br>
&gt; Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism t=
o map received data to m- line?<br>
&gt;<br>
&gt; &#43;1<br>
&gt; (Except that I *am* Paul)<br>
&gt;<br>
&gt; On 7/2/13 2:30 PM, Christer Holmberg wrote:<br>
&gt;&gt; Hi,<br>
&gt;&gt;<br>
&gt;&gt; I am not Paul, but just to verify that I understand his issue, the=
 question is:<br>
&gt;&gt;<br>
&gt;&gt; Why do you need 4 m- lines for the thumbnails, when you could use =
1 m- line?<br>
&gt;&gt;<br>
&gt;&gt; Regards,<br>
&gt;&gt;<br>
&gt;&gt; Christer<br>
&gt;&gt;<br>
&gt;&gt; -----Alkuper=E4inen viesti-----<br>
&gt;&gt; L=E4hett=E4j=E4: mmusic-bounces@ietf.org [<a href=3D"mailto:mmusic=
-bounces@ietf.org">mailto:mmusic-bounces@ietf.org</a>]
<br>
&gt;&gt; Puolesta Mo Zanaty (mzanaty)<br>
&gt;&gt; L=E4hetetty: 2. hein=E4kuuta 2013 19:09<br>
&gt;&gt; Vastaanottaja: Paul Kyzivat<br>
&gt;&gt; Kopio: mmusic@ietf.org<br>
&gt;&gt; Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to m=
ap received data to m- line?<br>
&gt;&gt;<br>
&gt;&gt; An attempt at &quot;an answer that makes sense&quot;. (Just snippe=
ts, not full <br>
&gt;&gt; SDP.)<br>
&gt;&gt;<br>
&gt;&gt; m=3Dvideo 10000 RTP/SAVPF 96<br>
&gt;&gt; a=3Drtpmap:96 H264/90000<br>
&gt;&gt; a=3Dfmtp:96 profile-level-id=3D42001F &lt;--- HD 720p30 main speak=
er<br>
&gt;&gt;<br>
&gt;&gt; m=3Dvideo 10000 RTP/SAVPF 97<br>
&gt;&gt; a=3Drtpmap:97 H264/90000<br>
&gt;&gt; a=3Dfmtp:97 profile-level-id=3D42000C &lt;--- SD 240p15 thumbnail =
a=3Drecvonly<br>
&gt;&gt;<br>
&gt;&gt; m=3Dvideo 10000 RTP/SAVPF 97<br>
&gt;&gt; a=3Drtpmap:97 H264/90000<br>
&gt;&gt; a=3Dfmtp:97 profile-level-id=3D42000C &lt;--- SD 240p15 thumbnail =
a=3Drecvonly<br>
&gt;&gt;<br>
&gt;&gt; m=3Dvideo 10000 RTP/SAVPF 97<br>
&gt;&gt; a=3Drtpmap:97 H264/90000<br>
&gt;&gt; a=3Dfmtp:97 profile-level-id=3D42000C &lt;--- SD 240p15 thumbnail =
a=3Drecvonly<br>
&gt;&gt;<br>
&gt;&gt; m=3Dvideo 10000 RTP/SAVPF 97<br>
&gt;&gt; a=3Drtpmap:97 H264/90000<br>
&gt;&gt; a=3Dfmtp:97 profile-level-id=3D42000C &lt;--- SD 240p15 thumbnail =
a=3Drecvonly<br>
&gt;&gt;<br>
&gt;&gt; No need to map thumbnail packets to specific m-lines, so reuse of =
PT=3D97 is fine. Just render the last 4 thumbnail sources received (in addi=
tion to the main speaker).<br>
&gt;&gt;<br>
&gt;&gt; You could mandate that PT must be unique to allow unique mapping.<=
br>
&gt;&gt;<br>
&gt;&gt; You could mandate that multiple m-lines of thumbnails must be comb=
ined into a single m-line with a=3Dssrc/max-ssrc/multiple-render/whatever.<=
br>
&gt;&gt;<br>
&gt;&gt; Which do you want to mandate and why?<br>
&gt;&gt;<br>
&gt;&gt; Cheers,<br>
&gt;&gt; Mo<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; -----Original Message-----<br>
&gt;&gt; From: Paul Kyzivat [<a href=3D"mailto:pkyzivat@alum.mit.edu">mailt=
o:pkyzivat@alum.mit.edu</a>]<br>
&gt;&gt; Sent: Tuesday, July 02, 2013 10:06 AM<br>
&gt;&gt; To: Mo Zanaty (mzanaty)<br>
&gt;&gt; Cc: mmusic@ietf.org<br>
&gt;&gt; Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism t=
o map received data to m- line?<br>
&gt;&gt;<br>
&gt;&gt; On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:<br>
&gt;&gt;&gt; Informative descriptions of possible mappings are fine, but no=
thing normative, please. Some applications won't care about the mapping, an=
d others may care but use a different mapping from those described. The onl=
y normative statement you could make is something
 like: &quot;If you care about the mapping, you MUST have a mapping mechani=
sm. You MAY use the mechanism(s) described here, or some other mechanism.&q=
uot; But why mandate such a tautology?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; The assumption below that multiple m-lines always require diff=
erent m-line-specific processing of the packets also assumes Plan B, i.e. t=
hat multiple streams with similar processing should always be signaled with=
 a single m-line (for example, using a=3Dssrc
 or max-ssrc). Plan A purists would use multiple m-lines even for multiple =
streams with the same processing. I don't think we want to force bundle to =
pick a plan.<br>
&gt;&gt;<br>
&gt;&gt; I'm going to keep asking this question until I get an answer that =
<br>
&gt;&gt; makes<br>
&gt;&gt; sense:<br>
&gt;&gt;<br>
&gt;&gt; Please explain to me a meaningful situation when there are multipl=
e m-lines in a bundle and there is insufficient information to associate ea=
ch packet with an m-line.<br>
&gt;&gt;<br>
&gt;&gt; The *point* of plan A is that there be an m-line per &quot;flow&qu=
ot;, and that the offerer is enumerating the flows. So clearly it knows the=
 mapping.<br>
&gt;&gt; If the answerer isn't capable of doing the same mapping, then the =
use of plan A has failed.<br>
&gt;&gt;<br>
&gt;&gt; The part that I want to be normative is that an O/A is not valid u=
nless the mapping is possible with the information in the O/A, and that the=
 offerer and the answerer would agree about the mapping of each packet. I d=
on't care if they actually do the mapping,
 though if they are doing any sensible processing then they will be doing s=
omething *equivalent* to the mapping.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
&gt;&gt;<br>
&gt;&gt;&gt; Mo<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; -----Original Message-----<br>
&gt;&gt;&gt; From: mmusic-bounces@ietf.org [<a href=3D"mailto:mmusic-bounce=
s@ietf.org">mailto:mmusic-bounces@ietf.org</a>] On
<br>
&gt;&gt;&gt; Behalf Of Paul Kyzivat<br>
&gt;&gt;&gt; Sent: Saturday, June 29, 2013 12:12 PM<br>
&gt;&gt;&gt; To: mmusic@ietf.org<br>
&gt;&gt;&gt; Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechani=
sm to map received data to m- line?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; On 6/29/13 8:50 AM, Colin Perkins wrote:<br>
&gt;&gt;&gt;&gt; Christer,<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; On 25 Jun 2013, at 12:16, Christer Holmberg wrote:<br>
&gt;&gt;&gt;&gt;&gt; There has been some discussions about whether BUNDLE s=
hould mandate that users are mandated to always (no matter what transport p=
rotocols are used in the BUNDLE group) have a mechanism to map received dat=
a to an m- line, or whether it from a generic BUNDLE
 perspective should be optional - IF there would be cases where it's not ne=
eded.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; We haven't had that much discussion about it yet, so I=
 will not <br>
&gt;&gt;&gt;&gt;&gt; ask a DECISION question at this point, but I would rea=
lly like to <br>
&gt;&gt;&gt;&gt;&gt; get some input from people who have opinions about thi=
s :)<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; My opinion: BUNDLE needs to mandate a single algorithm for=
 mapping from RTP flows to m=3D lines, for those applications that care abo=
ut such a mapping. I don't think it should mandate that all applications ne=
ed to care.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I just replied elsewhere on this.<br>
&gt;&gt;&gt; My opinion turns out to be the exact opposite of yours: we nee=
d to <br>
&gt;&gt;&gt; support multiple algorithms, and that all applications should =
care.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I won't repeat the stuff about multiple algorithms.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I will repeat what I have said about the need for being able t=
o associate:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; There must have been *some* reason that two m-lines were used,=
 <br>
&gt;&gt;&gt; rather than just one. Each m-line heads a media section of the=
 SDP <br>
&gt;&gt;&gt; that contains a bunch of declarations. Something must be diffe=
rent <br>
&gt;&gt;&gt; in those two media sections, or else a single media section wo=
uld <br>
&gt;&gt;&gt; have been enough. Presumably whatever it is that is different =
is <br>
&gt;&gt;&gt; intended to affect how received packets are processed or <br>
&gt;&gt;&gt; interpreted. (If not, again there is no need for it to be ther=
e.) If <br>
&gt;&gt;&gt; you can't associate the packet with the m-line, then you don't=
 know <br>
&gt;&gt;&gt; which of the multiple interpretations to apply to the packet.<=
br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Maybe there is some exception to this, though I haven't come u=
p with <br>
&gt;&gt;&gt; anything. If there is, then I hope someone will call it out. I=
f so, <br>
&gt;&gt;&gt; then the constraint can be tightened.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&gt;&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; _______________________________________________<br>
&gt;&gt;&gt; mmusic mailing list<br>
&gt;&gt;&gt; mmusic@ietf.org<br>
&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https=
://www.ietf.org/mailman/listinfo/mmusic</a><br>
&gt;&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; mmusic mailing list<br>
&gt;&gt; mmusic@ietf.org<br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://w=
ww.ietf.org/mailman/listinfo/mmusic</a><br>
&gt;&gt;<br>
&gt;<br>
&gt;<br>
<br>
</div>
</span></font>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C3C2DE2ESESSMB209erics_--

From aurelien.sollaud@orange.com  Thu Jul  4 03:14:15 2013
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From: <aurelien.sollaud@orange.com>
To: Parthasarathi R <partha@parthasarathi.co.in>, "mmusic@ietf.org" <mmusic@ietf.org>, "mperumal@cisco.com" <mperumal@cisco.com>
Thread-Topic: comments on draft-ietf-mmusic-sdp-g723-g729-03
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Hello

My answers inline
(I removed the agreed parts)

Aurelien

-----Message d'origine-----
De=A0: Parthasarathi R [mailto:partha@parthasarathi.co.in]
Envoy=E9=A0: mercredi 3 juillet 2013 19:29
=C0=A0: SOLLAUD Aurelien OLNC/OLN; mmusic@ietf.org; mperumal@cisco.com Obje=
t=A0: RE: comments on draft-ietf-mmusic-sdp-g723-g729-03

Hi Aurelien,

Sorry for the delay response. Thanks for the review comments.=20

Please read inline.

Thanks
Partha

> -----Original Message-----
> From: aurelien.sollaud@orange.com [mailto:aurelien.sollaud@orange.com]
> Sent: Thursday, June 27, 2013 1:52 PM
> To: mmusic@ietf.org; mperumal@cisco.com; partha@parthasarathi.co.in
> Subject: comments on draft-ietf-mmusic-sdp-g723-g729-03
>=20

  (snip)
>=20
>=20
> ** Section 3.1
>=20
> 2)
> After the first paragraph, I suggest to insert the new paragraph below:
>=20
> ---NEW
>  The Annex A of the codec concerns both sending and receiving, so both=20
> sides of a bi-directional session MUST have the same setting.  If  one=20
> party indicates that it does not support Annex A, Annex A  must be=20
> deactivated both ways.
> -----
> That defines the general rationale of the following paragraphs.
>
<Partha> I agree with that it is the objective of this O/A requirement but =
how this statement qualifies for normative statement is not clear. From the=
 offerer perspective, there is no control on same setting in both sides. In=
 case you look at the combine requirements of this draft, this object is me=
t.
</Partha>

[Aurelien] OK. I agree you can turn "MUST" into "must". I think this new pa=
ragraph helps anyway.

=20
>=20
> 3)
> -----
>    When the offer has G723 with annexa=3Dno then the answer MUST NOT have
>    annexa=3Dyes for G723.  Thus the annexa parameter can be turned off by
>    the answerer, but cannot be turned on.
> -----
> I don't think this new requirement is needed.
> Anyway implementation that will not implement this RFC-to-be will not=20
> comply with it.
>=20
<Partha> As per based G723 RFC, it is expected for all the implementation t=
o support annexa. This issue is discussed in http://www.ietf.org/mail-archi=
ve/web/mmusic/current/msg11165.html </Partha>=20

[Aurelien] Yes I agree all implementations support annexa. What I called th=
e RFC-to-be is the document draft-ietf-mmusic-sdp-g723-g729. My comment was=
 that previous implementations will not comply with this "MUST NOT". A warn=
ing should be written on this. The receiver of the SDP should be tolerant a=
nd ready to receive SDP not compliant with this new requirement, without br=
eaking.
I the offer has annexa=3Dno and the answer (of an old implementation) has a=
nnexa=3Dyes, Annex A must not be used.
It is the same kind of thing you describe later about the case where the of=
fer has annexa=3Dno and the answer has no annexa at all. Annex A must not b=
e used.


  (snip)
>=20
> ** Section 3.2
>=20
> 5)
> After the first two paragraphs, I suggest to insert the new paragraph
> below:
>=20
> ---NEW
>  The Annex B of the codec concerns both sending and receiving, so both=20
> sides of a bi-directional session MUST have the same setting.  If  one=20
> party indicates that it does not support Annex B, Annex B must be=20
> deactivated both ways.
> -----
> That defines the general rationale of the following paragraphs.
>=20

<Partha> Same as comment 2 reply. I agree with that it is the objective of =
this O/A requirement but how this statement qualifies for normative stateme=
nt is not clear. From the offerer perspective, there is no control on same =
setting in both sides. In case you look at the combine requirements of this=
 draft, this object is met. </Partha>

[Aurelien] same as comment 2, you can keep it informative and turn the "MUS=
T" into "must".


>=20
> 6)
> -----
>    When the offer or answer has G729 and the annexb parameter is=20
> absent,
>    it MUST be considered as if the offer or answer has G729 with
>    annexb=3Dyes.
> -----
> This paragraph is redundant with what is said before.
> Remove it.
>=20
>=20

<Partha> As per based G729 RFC, it is expected for all the implementation t=
o support annexb. This issue is discussed in http://www.ietf.org/mail-archi=
ve/web/mmusic/current/msg11165.html </Partha>

[Aurelien] You did not understand me. I said to remove the paragraph becaus=
e it is an exact repetition of text. I quote the whole text:

----
   When the offer or answer has G729 and the annexb parameter is absent,
   the offerer or answerer knows that it has implied the default
   "annexb=3Dyes".  This is because the annexb attribute is part of the
   original registration of audio/G729 [RFC4856].  All implementations
   that support G729 understand the annexb attribute.  Hence, this case
   MUST be considered as if the offer or answer has G729 with
   annexb=3Dyes.

   When the offer or answer has G729 and the annexb parameter is absent,
   it MUST be considered as if the offer or answer has G729 with
   annexb=3Dyes.
----

As you can see, the second paragraph says the exact same thing as the first=
 one.


> 7)
> -----
>    When the offer has G729 with annexb=3Dno then the answer MUST NOT have
>    annexb=3Dyes for G729.  Thus the annexb parameter can be turned off by
>    the answerer, but cannot be turned on.
> -----
> I don't think this new requirement is needed.
> For the same reason as for G723
>=20
<Partha> Same reply as comment 3. As per based G729 RFC, it is expected for=
 all the implementation to support annexb. This issue is discussed in http:=
//www.ietf.org/mail-archive/web/mmusic/current/msg11165.html </Partha>

[Aurelien] Same as comment 3, I suggest adding a warning on previous implem=
entation that may not comply with this new requirement.

  (snip)

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From thomas.stach@siemens-enterprise.com  Thu Jul  4 06:15:12 2013
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From: "Stach, Thomas" <thomas.stach@siemens-enterprise.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
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Emil,
inline

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Emil Ivov
> Sent: Wednesday, 03 July, 2013 16:13
> To: MMUSIC IETF WG
> Subject: [MMUSIC] Trickle ICE for SIP Questions
>=20
> Hey all,
>=20
> Christer, Enrico and I are preparing the next version of Trickle ICE
> for
> SIP. Now that discussions on BUNDLE and the plans seem to be winding
> down, we wanted to run a few questions by the working group.
>=20
> Q1: Making reliable provisional responses and PRACK mandatory.
> Obviously
> this would be nice to avoid, so the question is: is there a reasonable
> mechanism to achieve this (and by reasonable, we mean something that
> wouldn't be harder than implementing support for PRACK).
>=20
> There was some discussion about this back in April and there was a
> suggestion for implementing a 5245-style hack where the answerer
> basically resends the 180 until it knows that it has been received.
>=20
> 5245 uses connectivity checks for this (i.e. it stops 180
> retransmissions when the first connectivity check is received) but we
> don't have that option here since the 180 may contain either none or
> only host candidates so there are strong chances that no binding
> request
> would be received on them.
>=20
> Thomas also suggested a second option which would be to also use INFO
> requests with trickled candidates as an indication that 180 was
> received. This however wouldn't work with half trickle so we are
> basically back to vanilla ICE for all (non-re) INVITEs.
>=20
> Another option would be to mandate an INFO request with
> "end-of-candidates" in response to the 180, but that would be just the
> same as mandating PRACK support.
>=20
> Thomas also suggested that the answerer can start sending INFOs right
> after it sends its answer in the 180 and then it can just resend the
> 180
> if the INFOs result in a 481 response.
>=20
> Personally I think this could potentially be made to work, but it would
> imply a level of complexity that considerably exceeds PRACK support.
[TS] I'm not convinced about the different level of complexity.=20
If you use PRACK you have SIP timer T1 running and would re-transmit the 18=
0 in case that the PRACK is not received.=20
Assuming that you want to wait with sending of the INFO until you saw the P=
RACK, would lead to delay in that case.
If you don't wait you would also have to treat a 481 response to the INFO.
... and you would detect quicker that the 180 was lost than by waiting for =
the PRACK.

BTW: 5245 recommends to use PRACK. However it does not mandate it, but give=
s the alternative procedures as an option.
You could do the same here.

>=20
> Opinions?
>=20
> Q2: How do we send INFOs? Are they blocking or do we just send them in
> parallel? If the latter, then what happens when an INFO fails because
> it
> is received out of order? Do we just tell the application to resend the
> candidates asap?
[TS] I would go with the proposal below and send cumulatively. You also hav=
e the CSeq: header field to detect late arrivals.

>=20
> This also leads to the following question:
>=20
> Q3: What exactly do we send in INFOs? Just the latest batch of freshly
> learned candidates or all candidates we've learned so far? Dale
> suggested that if we do this cumulatively we wouldn't need to worry
> about the case with the out-of-order INFOs from Q2 since the
> information
> gets resent anyway. A drawback here would obviously be that this adds
> more complexity for trickle ICE users (WebRTC applications
> specifically)
[TS] Again I'm not convinced about the complexity. Both peer know their can=
didates, what is the issue in putting all of them into the INFO?
>=20
> A third option would be to allow both and leave it to the application.
[TS] This now adds complexity. ;-) I would just always send all candidates.
Regards
Thomas
>=20
> Comments are most welcome!
>=20
> Emil
>=20
> --
> https://jitsi.org
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "aurelien.sollaud@orange.com" <aurelien.sollaud@orange.com>, "Parthasarathi R" <partha@parthasarathi.co.in>, "mmusic@ietf.org" <mmusic@ietf.org>
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Date: Fri, 5 Jul 2013 05:11:32 +0000
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Hi Aurelien,

Partha and I had a discussion. Please see inline for [Muthu]:

|-----Original Message-----
|From: aurelien.sollaud@orange.com [mailto:aurelien.sollaud@orange.com]
|Sent: Thursday, July 04, 2013 3:44 PM
|To: Parthasarathi R; mmusic@ietf.org; Muthu Arul Mozhi Perumal (mperumal)
|Subject: RE:`
|
|Hello
|
|My answers inline
|(I removed the agreed parts)
|
|Aurelien
|
|-----Message d'origine-----
|De=A0: Parthasarathi R [mailto:partha@parthasarathi.co.in]
|Envoy=E9=A0: mercredi 3 juillet 2013 19:29
|=C0=A0: SOLLAUD Aurelien OLNC/OLN; mmusic@ietf.org; mperumal@cisco.com Obj=
et=A0: RE: comments on draft-ietf-
|mmusic-sdp-g723-g729-03
|
|Hi Aurelien,
|
|Sorry for the delay response. Thanks for the review comments.
|
|Please read inline.
|
|Thanks
|Partha
|
|> -----Original Message-----
|> From: aurelien.sollaud@orange.com [mailto:aurelien.sollaud@orange.com]
|> Sent: Thursday, June 27, 2013 1:52 PM
|> To: mmusic@ietf.org; mperumal@cisco.com; partha@parthasarathi.co.in
|> Subject: comments on draft-ietf-mmusic-sdp-g723-g729-03
|>
|
|  (snip)
|>
|>
|> ** Section 3.1
|>
|> 2)
|> After the first paragraph, I suggest to insert the new paragraph below:
|>
|> ---NEW
|>  The Annex A of the codec concerns both sending and receiving, so both
|> sides of a bi-directional session MUST have the same setting.  If  one
|> party indicates that it does not support Annex A, Annex A  must be
|> deactivated both ways.
|> -----
|> That defines the general rationale of the following paragraphs.
|>
|<Partha> I agree with that it is the objective of this O/A requirement but=
 how this statement
|qualifies for normative statement is not clear. From the offerer perspecti=
ve, there is no control on
|same setting in both sides. In case you look at the combine requirements o=
f this draft, this object is
|met.
|</Partha>
|
|[Aurelien] OK. I agree you can turn "MUST" into "must". I think this new p=
aragraph helps anyway.

[Muthu] Agree it helps. However, the text could be reworded to make it cris=
per:
  =20
   The objective of the below statements is to make sure that if the offere=
r=20
   or answerer indicates it does not support Annex A, Annex A is not used.

Does it sound good?

|
|
|>
|> 3)
|> -----
|>    When the offer has G723 with annexa=3Dno then the answer MUST NOT hav=
e
|>    annexa=3Dyes for G723.  Thus the annexa parameter can be turned off b=
y
|>    the answerer, but cannot be turned on.
|> -----
|> I don't think this new requirement is needed.
|> Anyway implementation that will not implement this RFC-to-be will not
|> comply with it.
|>
|<Partha> As per based G723 RFC, it is expected for all the implementation =
to support annexa. This
|issue is discussed in http://www.ietf.org/mail-archive/web/mmusic/current/=
msg11165.html </Partha>
|
|[Aurelien] Yes I agree all implementations support annexa. What I called t=
he RFC-to-be is the document
|draft-ietf-mmusic-sdp-g723-g729. My comment was that previous implementati=
ons will not comply with
|this "MUST NOT". A warning should be written on this. The receiver of the =
SDP should be tolerant and
|ready to receive SDP not compliant with this new requirement, without brea=
king.
|I the offer has annexa=3Dno and the answer (of an old implementation) has =
annexa=3Dyes, Annex A must not
|be used.
|It is the same kind of thing you describe later about the case where the o=
ffer has annexa=3Dno and the
|answer has no annexa at all. Annex A must not be used.

[Muthu] Agree. I think both this paragraph and the later paragraph (that de=
scribes the case where the offer has annexa=3Dno and the answer has no anne=
xa at all) can be combined together:

   When the offer has G723 with annexa=3Dno, after the offer/answer=20
   completion the offerer and answerer MUST proceed as if G723 is=20
   negotiated with annexa=3Dno.

   When the offer has G723 with annexa=3Dno, the reason for not mandating
   that the answer MUST have annexa=3Dno for G723 is that there are there
   implementations that omit the annexa parameter in answer and expect
   the least common denominator to be used.

Does it sound good?=20

|
|
|  (snip)
|>
|> ** Section 3.2
|>
|> 5)
|> After the first two paragraphs, I suggest to insert the new paragraph
|> below:
|>
|> ---NEW
|>  The Annex B of the codec concerns both sending and receiving, so both
|> sides of a bi-directional session MUST have the same setting.  If  one
|> party indicates that it does not support Annex B, Annex B must be
|> deactivated both ways.
|> -----
|> That defines the general rationale of the following paragraphs.
|>
|
|<Partha> Same as comment 2 reply. I agree with that it is the objective of=
 this O/A requirement but
|how this statement qualifies for normative statement is not clear. From th=
e offerer perspective, there
|is no control on same setting in both sides. In case you look at the combi=
ne requirements of this
|draft, this object is met. </Partha>
|
|[Aurelien] same as comment 2, you can keep it informative and turn the "MU=
ST" into "must".

[Muthu] Same as above:=20

   The objective of the below statements is to make sure that if the offere=
r=20
   or answerer indicates it does not support Annex B, Annex B is not used.

Does it sound good?

|
|
|>
|> 6)
|> -----
|>    When the offer or answer has G729 and the annexb parameter is
|> absent,
|>    it MUST be considered as if the offer or answer has G729 with
|>    annexb=3Dyes.
|> -----
|> This paragraph is redundant with what is said before.
|> Remove it.
|>
|>
|
|<Partha> As per based G729 RFC, it is expected for all the implementation =
to support annexb. This
|issue is discussed in http://www.ietf.org/mail-archive/web/mmusic/current/=
msg11165.html </Partha>
|
|[Aurelien] You did not understand me. I said to remove the paragraph becau=
se it is an exact repetition
|of text. I quote the whole text:
|
|----
|   When the offer or answer has G729 and the annexb parameter is absent,
|   the offerer or answerer knows that it has implied the default
|   "annexb=3Dyes".  This is because the annexb attribute is part of the
|   original registration of audio/G729 [RFC4856].  All implementations
|   that support G729 understand the annexb attribute.  Hence, this case
|   MUST be considered as if the offer or answer has G729 with
|   annexb=3Dyes.
|
|   When the offer or answer has G729 and the annexb parameter is absent,
|   it MUST be considered as if the offer or answer has G729 with
|   annexb=3Dyes.
|----
|
|As you can see, the second paragraph says the exact same thing as the firs=
t one.

[Muthu] Agree, the second paragraph is redundant.

|
|
|> 7)
|> -----
|>    When the offer has G729 with annexb=3Dno then the answer MUST NOT hav=
e
|>    annexb=3Dyes for G729.  Thus the annexb parameter can be turned off b=
y
|>    the answerer, but cannot be turned on.
|> -----
|> I don't think this new requirement is needed.
|> For the same reason as for G723
|>
|<Partha> Same reply as comment 3. As per based G729 RFC, it is expected fo=
r all the implementation to
|support annexb. This issue is discussed in http://www.ietf.org/mail-
|archive/web/mmusic/current/msg11165.html </Partha>
|
|[Aurelien] Same as comment 3, I suggest adding a warning on previous imple=
mentation that may not
|comply with this new requirement.

[Muthu] Same as above:

   When the offer has G729 with annexb=3Dno, after the offer/answer=20
   completion the offerer and answerer MUST proceed as if G729 is=20
   negotiated with annexb=3Dno.

   When the offer has G729 with annexb=3Dno, the reason for not mandating
   that the answer MUST have annexb=3Dno for G729 is that there are there
   implementations that omit the annexb parameter in answer and expect
   the least common denominator to be used.

Does it sound good?

Thanks for your inputs,

Muthu

|
|  (snip)
|
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From mperumal@cisco.com  Thu Jul  4 22:24:48 2013
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "aurelien.sollaud@orange.com" <aurelien.sollaud@orange.com>, "Parthasarathi R" <partha@parthasarathi.co.in>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: comments on draft-ietf-mmusic-sdp-g723-g729-03
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Date: Fri, 5 Jul 2013 05:24:39 +0000
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Subject: Re: [MMUSIC] comments on draft-ietf-mmusic-sdp-g723-g729-03
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Fixed the subject to make it easier to search..

Muthu

|-----Original Message-----
|From: Muthu Arul Mozhi Perumal (mperumal)
|Sent: Friday, July 05, 2013 10:42 AM
|To: 'aurelien.sollaud@orange.com'; Parthasarathi R; mmusic@ietf.org
|Subject: RE: RE:`
|
|Hi Aurelien,
|
|Partha and I had a discussion. Please see inline for [Muthu]:
|
||-----Original Message-----
||From: aurelien.sollaud@orange.com [mailto:aurelien.sollaud@orange.com]
||Sent: Thursday, July 04, 2013 3:44 PM
||To: Parthasarathi R; mmusic@ietf.org; Muthu Arul Mozhi Perumal (mperumal)
||Subject: RE:`
||
||Hello
||
||My answers inline
||(I removed the agreed parts)
||
||Aurelien
||
||-----Message d'origine-----
||De=A0: Parthasarathi R [mailto:partha@parthasarathi.co.in]
||Envoy=E9=A0: mercredi 3 juillet 2013 19:29
||=C0=A0: SOLLAUD Aurelien OLNC/OLN; mmusic@ietf.org; mperumal@cisco.com Ob=
jet=A0: RE: comments on draft-
|ietf-
||mmusic-sdp-g723-g729-03
||
||Hi Aurelien,
||
||Sorry for the delay response. Thanks for the review comments.
||
||Please read inline.
||
||Thanks
||Partha
||
||> -----Original Message-----
||> From: aurelien.sollaud@orange.com [mailto:aurelien.sollaud@orange.com]
||> Sent: Thursday, June 27, 2013 1:52 PM
||> To: mmusic@ietf.org; mperumal@cisco.com; partha@parthasarathi.co.in
||> Subject: comments on draft-ietf-mmusic-sdp-g723-g729-03
||>
||
||  (snip)
||>
||>
||> ** Section 3.1
||>
||> 2)
||> After the first paragraph, I suggest to insert the new paragraph below:
||>
||> ---NEW
||>  The Annex A of the codec concerns both sending and receiving, so both
||> sides of a bi-directional session MUST have the same setting.  If  one
||> party indicates that it does not support Annex A, Annex A  must be
||> deactivated both ways.
||> -----
||> That defines the general rationale of the following paragraphs.
||>
||<Partha> I agree with that it is the objective of this O/A requirement bu=
t how this statement
||qualifies for normative statement is not clear. From the offerer perspect=
ive, there is no control on
||same setting in both sides. In case you look at the combine requirements =
of this draft, this object
|is
||met.
||</Partha>
||
||[Aurelien] OK. I agree you can turn "MUST" into "must". I think this new =
paragraph helps anyway.
|
|[Muthu] Agree it helps. However, the text could be reworded to make it cri=
sper:
|
|   The objective of the below statements is to make sure that if the offer=
er
|   or answerer indicates it does not support Annex A, Annex A is not used.
|
|Does it sound good?
|
||
||
||>
||> 3)
||> -----
||>    When the offer has G723 with annexa=3Dno then the answer MUST NOT ha=
ve
||>    annexa=3Dyes for G723.  Thus the annexa parameter can be turned off =
by
||>    the answerer, but cannot be turned on.
||> -----
||> I don't think this new requirement is needed.
||> Anyway implementation that will not implement this RFC-to-be will not
||> comply with it.
||>
||<Partha> As per based G723 RFC, it is expected for all the implementation=
 to support annexa. This
||issue is discussed in http://www.ietf.org/mail-archive/web/mmusic/current=
/msg11165.html </Partha>
||
||[Aurelien] Yes I agree all implementations support annexa. What I called =
the RFC-to-be is the
|document
||draft-ietf-mmusic-sdp-g723-g729. My comment was that previous implementat=
ions will not comply with
||this "MUST NOT". A warning should be written on this. The receiver of the=
 SDP should be tolerant and
||ready to receive SDP not compliant with this new requirement, without bre=
aking.
||I the offer has annexa=3Dno and the answer (of an old implementation) has=
 annexa=3Dyes, Annex A must not
||be used.
||It is the same kind of thing you describe later about the case where the =
offer has annexa=3Dno and the
||answer has no annexa at all. Annex A must not be used.
|
|[Muthu] Agree. I think both this paragraph and the later paragraph (that d=
escribes the case where the
|offer has annexa=3Dno and the answer has no annexa at all) can be combined=
 together:
|
|   When the offer has G723 with annexa=3Dno, after the offer/answer
|   completion the offerer and answerer MUST proceed as if G723 is
|   negotiated with annexa=3Dno.
|
|   When the offer has G723 with annexa=3Dno, the reason for not mandating
|   that the answer MUST have annexa=3Dno for G723 is that there are there
|   implementations that omit the annexa parameter in answer and expect
|   the least common denominator to be used.
|
|Does it sound good?
|
||
||
||  (snip)
||>
||> ** Section 3.2
||>
||> 5)
||> After the first two paragraphs, I suggest to insert the new paragraph
||> below:
||>
||> ---NEW
||>  The Annex B of the codec concerns both sending and receiving, so both
||> sides of a bi-directional session MUST have the same setting.  If  one
||> party indicates that it does not support Annex B, Annex B must be
||> deactivated both ways.
||> -----
||> That defines the general rationale of the following paragraphs.
||>
||
||<Partha> Same as comment 2 reply. I agree with that it is the objective o=
f this O/A requirement but
||how this statement qualifies for normative statement is not clear. From t=
he offerer perspective,
|there
||is no control on same setting in both sides. In case you look at the comb=
ine requirements of this
||draft, this object is met. </Partha>
||
||[Aurelien] same as comment 2, you can keep it informative and turn the "M=
UST" into "must".
|
|[Muthu] Same as above:
|
|   The objective of the below statements is to make sure that if the offer=
er
|   or answerer indicates it does not support Annex B, Annex B is not used.
|
|Does it sound good?
|
||
||
||>
||> 6)
||> -----
||>    When the offer or answer has G729 and the annexb parameter is
||> absent,
||>    it MUST be considered as if the offer or answer has G729 with
||>    annexb=3Dyes.
||> -----
||> This paragraph is redundant with what is said before.
||> Remove it.
||>
||>
||
||<Partha> As per based G729 RFC, it is expected for all the implementation=
 to support annexb. This
||issue is discussed in http://www.ietf.org/mail-archive/web/mmusic/current=
/msg11165.html </Partha>
||
||[Aurelien] You did not understand me. I said to remove the paragraph beca=
use it is an exact
|repetition
||of text. I quote the whole text:
||
||----
||   When the offer or answer has G729 and the annexb parameter is absent,
||   the offerer or answerer knows that it has implied the default
||   "annexb=3Dyes".  This is because the annexb attribute is part of the
||   original registration of audio/G729 [RFC4856].  All implementations
||   that support G729 understand the annexb attribute.  Hence, this case
||   MUST be considered as if the offer or answer has G729 with
||   annexb=3Dyes.
||
||   When the offer or answer has G729 and the annexb parameter is absent,
||   it MUST be considered as if the offer or answer has G729 with
||   annexb=3Dyes.
||----
||
||As you can see, the second paragraph says the exact same thing as the fir=
st one.
|
|[Muthu] Agree, the second paragraph is redundant.
|
||
||
||> 7)
||> -----
||>    When the offer has G729 with annexb=3Dno then the answer MUST NOT ha=
ve
||>    annexb=3Dyes for G729.  Thus the annexb parameter can be turned off =
by
||>    the answerer, but cannot be turned on.
||> -----
||> I don't think this new requirement is needed.
||> For the same reason as for G723
||>
||<Partha> Same reply as comment 3. As per based G729 RFC, it is expected f=
or all the implementation to
||support annexb. This issue is discussed in http://www.ietf.org/mail-
||archive/web/mmusic/current/msg11165.html </Partha>
||
||[Aurelien] Same as comment 3, I suggest adding a warning on previous impl=
ementation that may not
||comply with this new requirement.
|
|[Muthu] Same as above:
|
|   When the offer has G729 with annexb=3Dno, after the offer/answer
|   completion the offerer and answerer MUST proceed as if G729 is
|   negotiated with annexb=3Dno.
|
|   When the offer has G729 with annexb=3Dno, the reason for not mandating
|   that the answer MUST have annexb=3Dno for G729 is that there are there
|   implementations that omit the annexb parameter in answer and expect
|   the least common denominator to be used.
|
|Does it sound good?
|
|Thanks for your inputs,
|
|Muthu
|
||
||  (snip)
||
||_________________________________________________________________________=
____________________________
|_
||___________________
||
||Ce message et ses pieces jointes peuvent contenir des informations confid=
entielles ou privilegiees et
||ne doivent donc
||pas etre diffuses, exploites ou copies sans autorisation. Si vous avez re=
cu ce message par erreur,
||veuillez le signaler
||a l'expediteur et le detruire ainsi que les pieces jointes. Les messages =
electroniques etant
||susceptibles d'alteration,
||Orange decline toute responsabilite si ce message a ete altere, deforme o=
u falsifie. Merci.
||
||This message and its attachments may contain confidential or privileged i=
nformation that may be
||protected by law;
||they should not be distributed, used or copied without authorisation.
||If you have received this email in error, please notify the sender and de=
lete this message and its
||attachments.
||As emails may be altered, Orange is not liable for messages that have bee=
n modified, changed or
||falsified.
||Thank you.


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From: <aurelien.sollaud@orange.com>
To: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>, "Parthasarathi R" <partha@parthasarathi.co.in>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: comments on draft-ietf-mmusic-sdp-g723-g729-03
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Subject: Re: [MMUSIC] comments on draft-ietf-mmusic-sdp-g723-g729-03
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Hello

I fully agree with these modifications.
I have no other comment.

Thanks
Aurelien

-----Message d'origine-----
De=A0: Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]=20
Envoy=E9=A0: vendredi 5 juillet 2013 07:25
=C0=A0: SOLLAUD Aurelien OLNC/OLN; Parthasarathi R; mmusic@ietf.org
Objet=A0: RE: comments on draft-ietf-mmusic-sdp-g723-g729-03

Fixed the subject to make it easier to search..

Muthu

|-----Original Message-----
|From: Muthu Arul Mozhi Perumal (mperumal)
|Sent: Friday, July 05, 2013 10:42 AM
|To: 'aurelien.sollaud@orange.com'; Parthasarathi R; mmusic@ietf.org
|Subject: RE: RE:`
|
|Hi Aurelien,
|
|Partha and I had a discussion. Please see inline for [Muthu]:
|
||-----Original Message-----
||From: aurelien.sollaud@orange.com [mailto:aurelien.sollaud@orange.com]
||Sent: Thursday, July 04, 2013 3:44 PM
||To: Parthasarathi R; mmusic@ietf.org; Muthu Arul Mozhi Perumal=20
||(mperumal)
||Subject: RE:`
||
||Hello
||
||My answers inline
||(I removed the agreed parts)
||
||Aurelien
||
||-----Message d'origine-----
||De=A0: Parthasarathi R [mailto:partha@parthasarathi.co.in]
||Envoy=E9=A0: mercredi 3 juillet 2013 19:29 =C0=A0: SOLLAUD Aurelien OLNC/=
OLN;=20
||mmusic@ietf.org; mperumal@cisco.com Objet=A0: RE: comments on draft-
|ietf-
||mmusic-sdp-g723-g729-03
||
||Hi Aurelien,
||
||Sorry for the delay response. Thanks for the review comments.
||
||Please read inline.
||
||Thanks
||Partha
||
||> -----Original Message-----
||> From: aurelien.sollaud@orange.com=20
||> [mailto:aurelien.sollaud@orange.com]
||> Sent: Thursday, June 27, 2013 1:52 PM
||> To: mmusic@ietf.org; mperumal@cisco.com; partha@parthasarathi.co.in
||> Subject: comments on draft-ietf-mmusic-sdp-g723-g729-03
||>
||
||  (snip)
||>
||>
||> ** Section 3.1
||>
||> 2)
||> After the first paragraph, I suggest to insert the new paragraph below:
||>
||> ---NEW
||>  The Annex A of the codec concerns both sending and receiving, so=20
||> both sides of a bi-directional session MUST have the same setting.=20=
=20
||> If  one party indicates that it does not support Annex A, Annex A=20=20
||> must be deactivated both ways.
||> -----
||> That defines the general rationale of the following paragraphs.
||>
||<Partha> I agree with that it is the objective of this O/A requirement=20
||but how this statement qualifies for normative statement is not clear.=20
||From the offerer perspective, there is no control on same setting in=20
||both sides. In case you look at the combine requirements of this=20
||draft, this object
|is
||met.
||</Partha>
||
||[Aurelien] OK. I agree you can turn "MUST" into "must". I think this new =
paragraph helps anyway.
|
|[Muthu] Agree it helps. However, the text could be reworded to make it cri=
sper:
|
|   The objective of the below statements is to make sure that if the offer=
er
|   or answerer indicates it does not support Annex A, Annex A is not used.
|
|Does it sound good?
|
||
||
||>
||> 3)
||> -----
||>    When the offer has G723 with annexa=3Dno then the answer MUST NOT ha=
ve
||>    annexa=3Dyes for G723.  Thus the annexa parameter can be turned off =
by
||>    the answerer, but cannot be turned on.
||> -----
||> I don't think this new requirement is needed.
||> Anyway implementation that will not implement this RFC-to-be will=20
||> not comply with it.
||>
||<Partha> As per based G723 RFC, it is expected for all the=20
||implementation to support annexa. This issue is discussed in=20
||http://www.ietf.org/mail-archive/web/mmusic/current/msg11165.html=20
||</Partha>
||
||[Aurelien] Yes I agree all implementations support annexa. What I=20
||called the RFC-to-be is the
|document
||draft-ietf-mmusic-sdp-g723-g729. My comment was that previous=20
||implementations will not comply with this "MUST NOT". A warning should=20
||be written on this. The receiver of the SDP should be tolerant and ready =
to receive SDP not compliant with this new requirement, without breaking.
||I the offer has annexa=3Dno and the answer (of an old implementation)=20
||has annexa=3Dyes, Annex A must not be used.
||It is the same kind of thing you describe later about the case where=20
||the offer has annexa=3Dno and the answer has no annexa at all. Annex A mu=
st not be used.
|
|[Muthu] Agree. I think both this paragraph and the later paragraph=20
|(that describes the case where the offer has annexa=3Dno and the answer ha=
s no annexa at all) can be combined together:
|
|   When the offer has G723 with annexa=3Dno, after the offer/answer
|   completion the offerer and answerer MUST proceed as if G723 is
|   negotiated with annexa=3Dno.
|
|   When the offer has G723 with annexa=3Dno, the reason for not mandating
|   that the answer MUST have annexa=3Dno for G723 is that there are there
|   implementations that omit the annexa parameter in answer and expect
|   the least common denominator to be used.
|
|Does it sound good?
|
||
||
||  (snip)
||>
||> ** Section 3.2
||>
||> 5)
||> After the first two paragraphs, I suggest to insert the new=20
||> paragraph
||> below:
||>
||> ---NEW
||>  The Annex B of the codec concerns both sending and receiving, so=20
||> both sides of a bi-directional session MUST have the same setting.=20=
=20
||> If  one party indicates that it does not support Annex B, Annex B=20
||> must be deactivated both ways.
||> -----
||> That defines the general rationale of the following paragraphs.
||>
||
||<Partha> Same as comment 2 reply. I agree with that it is the=20
||objective of this O/A requirement but how this statement qualifies for=20
||normative statement is not clear. From the offerer perspective,
|there
||is no control on same setting in both sides. In case you look at the=20
||combine requirements of this draft, this object is met. </Partha>
||
||[Aurelien] same as comment 2, you can keep it informative and turn the "M=
UST" into "must".
|
|[Muthu] Same as above:
|
|   The objective of the below statements is to make sure that if the offer=
er
|   or answerer indicates it does not support Annex B, Annex B is not used.
|
|Does it sound good?
|
||
||
||>
||> 6)
||> -----
||>    When the offer or answer has G729 and the annexb parameter is=20
||> absent,
||>    it MUST be considered as if the offer or answer has G729 with
||>    annexb=3Dyes.
||> -----
||> This paragraph is redundant with what is said before.
||> Remove it.
||>
||>
||
||<Partha> As per based G729 RFC, it is expected for all the=20
||implementation to support annexb. This issue is discussed in=20
||http://www.ietf.org/mail-archive/web/mmusic/current/msg11165.html=20
||</Partha>
||
||[Aurelien] You did not understand me. I said to remove the paragraph=20
||because it is an exact
|repetition
||of text. I quote the whole text:
||
||----
||   When the offer or answer has G729 and the annexb parameter is absent,
||   the offerer or answerer knows that it has implied the default
||   "annexb=3Dyes".  This is because the annexb attribute is part of the
||   original registration of audio/G729 [RFC4856].  All implementations
||   that support G729 understand the annexb attribute.  Hence, this case
||   MUST be considered as if the offer or answer has G729 with
||   annexb=3Dyes.
||
||   When the offer or answer has G729 and the annexb parameter is absent,
||   it MUST be considered as if the offer or answer has G729 with
||   annexb=3Dyes.
||----
||
||As you can see, the second paragraph says the exact same thing as the fir=
st one.
|
|[Muthu] Agree, the second paragraph is redundant.
|
||
||
||> 7)
||> -----
||>    When the offer has G729 with annexb=3Dno then the answer MUST NOT ha=
ve
||>    annexb=3Dyes for G729.  Thus the annexb parameter can be turned off =
by
||>    the answerer, but cannot be turned on.
||> -----
||> I don't think this new requirement is needed.
||> For the same reason as for G723
||>
||<Partha> Same reply as comment 3. As per based G729 RFC, it is=20
||expected for all the implementation to support annexb. This issue is=20
||discussed in http://www.ietf.org/mail-=20
||archive/web/mmusic/current/msg11165.html </Partha>
||
||[Aurelien] Same as comment 3, I suggest adding a warning on previous=20
||implementation that may not comply with this new requirement.
|
|[Muthu] Same as above:
|
|   When the offer has G729 with annexb=3Dno, after the offer/answer
|   completion the offerer and answerer MUST proceed as if G729 is
|   negotiated with annexb=3Dno.
|
|   When the offer has G729 with annexb=3Dno, the reason for not mandating
|   that the answer MUST have annexb=3Dno for G729 is that there are there
|   implementations that omit the annexb parameter in answer and expect
|   the least common denominator to be used.
|
|Does it sound good?
|
|Thanks for your inputs,
|
|Muthu
|
||
||  (snip)
||


___________________________________________________________________________=
______________________________________________

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From emil@sip-communicator.org  Fri Jul  5 07:12:46 2013
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hey Thomas

On 04.07.13, 15:15, Stach, Thomas wrote:
> Emil,
> inline
>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>> Behalf Of Emil Ivov
>> Sent: Wednesday, 03 July, 2013 16:13
>> To: MMUSIC IETF WG
>> Subject: [MMUSIC] Trickle ICE for SIP Questions
>>
>> Hey all,
>>
>> Christer, Enrico and I are preparing the next version of Trickle ICE
>> for
>> SIP. Now that discussions on BUNDLE and the plans seem to be winding
>> down, we wanted to run a few questions by the working group.
>>
>> Q1: Making reliable provisional responses and PRACK mandatory.
>> Obviously
>> this would be nice to avoid, so the question is: is there a reasonable
>> mechanism to achieve this (and by reasonable, we mean something that
>> wouldn't be harder than implementing support for PRACK).
>>
>> There was some discussion about this back in April and there was a
>> suggestion for implementing a 5245-style hack where the answerer
>> basically resends the 180 until it knows that it has been received.
>>
>> 5245 uses connectivity checks for this (i.e. it stops 180
>> retransmissions when the first connectivity check is received) but we
>> don't have that option here since the 180 may contain either none or
>> only host candidates so there are strong chances that no binding
>> request
>> would be received on them.
>>
>> Thomas also suggested a second option which would be to also use INFO
>> requests with trickled candidates as an indication that 180 was
>> received. This however wouldn't work with half trickle so we are
>> basically back to vanilla ICE for all (non-re) INVITEs.
>>
>> Another option would be to mandate an INFO request with
>> "end-of-candidates" in response to the 180, but that would be just the
>> same as mandating PRACK support.
>>
>> Thomas also suggested that the answerer can start sending INFOs right
>> after it sends its answer in the 180 and then it can just resend the
>> 180
>> if the INFOs result in a 481 response.
>>
>> Personally I think this could potentially be made to work, but it would
>> imply a level of complexity that considerably exceeds PRACK support.
> [TS] I'm not convinced about the different level of complexity.
> If you use PRACK you have SIP timer T1 running and would re-transmit the 180 in case that the PRACK is not received.

Right.

> Assuming that you want to wait with sending of the INFO until you saw the PRACK, would lead to delay in that case.

True. Is this the only problem that we are trying to optimize? Save half 
an RTT? (This is a really a question. I am not trying to make a point. I 
am simply trying to understand why PRACK is an issue.)

> If you don't wait you would also have to treat a 481 response to the INFO.

I am always a bit nervous when it comes to interpreting error responses. 
We are indeed likely to get a 481 when sending INFOs and the 180 hasn't 
yet made it to the remote side. However 481 might also imply that our 
dialog is simply dead for some reason and in this case 3261 says:


   If the response for a request within a dialog is a 481
   (Call/Transaction Does Not Exist) or a 408 (Request Timeout), the UAC
   SHOULD terminate the dialog.

> ... and you would detect quicker that the 180 was lost than by waiting for the PRACK.

As said above: a lost 180 is not the only possible cause for a 408 in 
this case.

> BTW: 5245 recommends to use PRACK. However it does not mandate it, but gives the alternative procedures as an option.
> You could do the same here.

Of course. I should have made that clearer: all alternative approaches 
were meant as something we could have in addition to the PRACK one.

Also, keep in mind that the 180 carries an entire SDP answer. If INFOs 
are competing with 180s then they may potentially need to carry that too 
(or at least the ICE parts) and it would be good if we don't have to 
open that can of worms.

>> Opinions?
>>
>> Q2: How do we send INFOs? Are they blocking or do we just send them in
>> parallel? If the latter, then what happens when an INFO fails because
>> it
>> is received out of order? Do we just tell the application to resend the
>> candidates asap?
> [TS] I would go with the proposal below and send cumulatively. You also have the CSeq: header field to detect late arrivals.
>
>>
>> This also leads to the following question:
>>
>> Q3: What exactly do we send in INFOs? Just the latest batch of freshly
>> learned candidates or all candidates we've learned so far? Dale
>> suggested that if we do this cumulatively we wouldn't need to worry
>> about the case with the out-of-order INFOs from Q2 since the
>> information
>> gets resent anyway. A drawback here would obviously be that this adds
>> more complexity for trickle ICE users (WebRTC applications
>> specifically)
> [TS] Again I'm not convinced about the complexity. Both peer know their candidates, what is the issue in putting all of them into the INFO?

I am not sure how current WebRTC stacks handle provisioning of repeating 
candidates. If they don't then it would be up to the receiving JS app, 
to sort our which are the new candidates and which were already there.

The sending side would be similarly tricky for a JS app: it would need 
to construct SDP so that the a=candidate attributes appear under the 
right "a=mid" line.

>> A third option would be to allow both and leave it to the application.
> [TS] This now adds complexity. ;-)

Likely yes.

Emil

> I would just always send all candidates.




> Regards
> Thomas
>>
>> Comments are most welcome!
>>
>> Emil
>>
>> --
>> https://jitsi.org
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>

-- 
https://jitsi.org

From pkyzivat@alum.mit.edu  Fri Jul  5 09:47:15 2013
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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On 7/5/13 10:12 AM, Emil Ivov wrote:
> Hey Thomas
>
> On 04.07.13, 15:15, Stach, Thomas wrote:
>> Emil,
>> inline
>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>> Behalf Of Emil Ivov
>>> Sent: Wednesday, 03 July, 2013 16:13
>>> To: MMUSIC IETF WG
>>> Subject: [MMUSIC] Trickle ICE for SIP Questions
>>>
>>> Hey all,
>>>
>>> Christer, Enrico and I are preparing the next version of Trickle ICE
>>> for
>>> SIP. Now that discussions on BUNDLE and the plans seem to be winding
>>> down, we wanted to run a few questions by the working group.
>>>
>>> Q1: Making reliable provisional responses and PRACK mandatory.

Yes.

>>> Obviously
>>> this would be nice to avoid, so the question is: is there a reasonable
>>> mechanism to achieve this (and by reasonable, we mean something that
>>> wouldn't be harder than implementing support for PRACK).
>>>
>>> There was some discussion about this back in April and there was a
>>> suggestion for implementing a 5245-style hack where the answerer
>>> basically resends the 180 until it knows that it has been received.
>>>
>>> 5245 uses connectivity checks for this (i.e. it stops 180
>>> retransmissions when the first connectivity check is received) but we
>>> don't have that option here since the 180 may contain either none or
>>> only host candidates so there are strong chances that no binding
>>> request
>>> would be received on them.
>>>
>>> Thomas also suggested a second option which would be to also use INFO
>>> requests with trickled candidates as an indication that 180 was
>>> received. This however wouldn't work with half trickle so we are
>>> basically back to vanilla ICE for all (non-re) INVITEs.
>>>
>>> Another option would be to mandate an INFO request with
>>> "end-of-candidates" in response to the 180, but that would be just the
>>> same as mandating PRACK support.
>>>
>>> Thomas also suggested that the answerer can start sending INFOs right
>>> after it sends its answer in the 180 and then it can just resend the
>>> 180
>>> if the INFOs result in a 481 response.
>>>
>>> Personally I think this could potentially be made to work, but it would
>>> imply a level of complexity that considerably exceeds PRACK support.
>> [TS] I'm not convinced about the different level of complexity.
>> If you use PRACK you have SIP timer T1 running and would re-transmit
>> the 180 in case that the PRACK is not received.
>
> Right.
>
>> Assuming that you want to wait with sending of the INFO until you saw
>> the PRACK, would lead to delay in that case.
>
> True. Is this the only problem that we are trying to optimize? Save half
> an RTT? (This is a really a question. I am not trying to make a point. I
> am simply trying to understand why PRACK is an issue.)

Yes, why is it such an issue to require PRACK?
It is a good thing to implement for many reasons. So in conjunction with 
defining other new functionality, why not require it?

>> If you don't wait you would also have to treat a 481 response to the
>> INFO.
>
> I am always a bit nervous when it comes to interpreting error responses.
> We are indeed likely to get a 481 when sending INFOs and the 180 hasn't
> yet made it to the remote side. However 481 might also imply that our
> dialog is simply dead for some reason and in this case 3261 says:
>
>
>    If the response for a request within a dialog is a 481
>    (Call/Transaction Does Not Exist) or a 408 (Request Timeout), the UAC
>    SHOULD terminate the dialog.
>
>> ... and you would detect quicker that the 180 was lost than by waiting
>> for the PRACK.

This just highlights how "delicate" things get if you start trying to 
work around the lack of PRACK. It might be possible to get something 
that seems to work, but its a house of cards.

> As said above: a lost 180 is not the only possible cause for a 408 in
> this case.
>
>> BTW: 5245 recommends to use PRACK. However it does not mandate it, but
>> gives the alternative procedures as an option.
>> You could do the same here.
>
> Of course. I should have made that clearer: all alternative approaches
> were meant as something we could have in addition to the PRACK one.
>
> Also, keep in mind that the 180 carries an entire SDP answer. If INFOs
> are competing with 180s then they may potentially need to carry that too
> (or at least the ICE parts) and it would be good if we don't have to
> open that can of worms.

Yes. The trickle ICE INFOs are *relative* to what was set up in the O/A. 
In the absence of the answer the trickle ICE is problematic.

>>> Opinions?
>>>
>>> Q2: How do we send INFOs? Are they blocking or do we just send them in
>>> parallel? If the latter, then what happens when an INFO fails because
>>> it
>>> is received out of order? Do we just tell the application to resend the
>>> candidates asap?

*If* you are going to send in a non-blocking way then they should be 
cumulative.

If you treat them as blocking, then they don't have to be, but it might 
still be better.

>> [TS] I would go with the proposal below and send cumulatively. You
>> also have the CSeq: header field to detect late arrivals.

Yes, you can detect the late arrival. But you have at that point already 
acted on the one that should have been preceded by it.

>>> This also leads to the following question:
>>>
>>> Q3: What exactly do we send in INFOs? Just the latest batch of freshly
>>> learned candidates or all candidates we've learned so far? Dale
>>> suggested that if we do this cumulatively we wouldn't need to worry
>>> about the case with the out-of-order INFOs from Q2 since the
>>> information
>>> gets resent anyway.

Exactly.

>>> A drawback here would obviously be that this adds
>>> more complexity for trickle ICE users (WebRTC applications
>>> specifically)
>> [TS] Again I'm not convinced about the complexity. Both peer know
>> their candidates, what is the issue in putting all of them into the INFO?
>
> I am not sure how current WebRTC stacks handle provisioning of repeating
> candidates. If they don't then it would be up to the receiving JS app,
> to sort our which are the new candidates and which were already there.
>
> The sending side would be similarly tricky for a JS app: it would need
> to construct SDP so that the a=candidate attributes appear under the
> right "a=mid" line.

I don't know enough about ICE to comment specifically on this.

But ISTM that this is analogous to what a SIP device must do to manage 
the target set of destinations for a request - it is required to keep 
track and not add the same address to the target set more than once.

	Thanks,
	Paul

>>> A third option would be to allow both and leave it to the application.
>> [TS] This now adds complexity. ;-)
>
> Likely yes.
>
> Emil
>
>> I would just always send all candidates.
>
>
>
>
>> Regards
>> Thomas
>>>
>>> Comments are most welcome!
>>>
>>> Emil
>>>
>>> --
>>> https://jitsi.org
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>


From petithug@acm.org  Fri Jul  5 10:28:29 2013
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

Hi,

I did not receive much comments on my proposal to use STUN packets to carry
the new ICE candidates.  To summarize, instead of using the SDP in an INFO and
or UPDATE transaction, the new candidates can be sent on the same path that
the media will use by default (i.e. using the candidate that is copied in the
m= and c= line).  Because this candidate is chosen to always succeed (using a
TURN server), the candidate will always reach their destination.  This will
remove any problem in the signaling path, because it is no longer used to
update the candidates.

Thanks.

On 07/03/2013 07:13 AM, Emil Ivov wrote:
> Hey all,
> 
> Christer, Enrico and I are preparing the next version of Trickle ICE for
> SIP. Now that discussions on BUNDLE and the plans seem to be winding down,
> we wanted to run a few questions by the working group.
> 
> Q1: Making reliable provisional responses and PRACK mandatory. Obviously
> this would be nice to avoid, so the question is: is there a reasonable
> mechanism to achieve this (and by reasonable, we mean something that
> wouldn't be harder than implementing support for PRACK).
> 
> There was some discussion about this back in April and there was a
> suggestion for implementing a 5245-style hack where the answerer basically
> resends the 180 until it knows that it has been received.
> 
> 5245 uses connectivity checks for this (i.e. it stops 180 retransmissions
> when the first connectivity check is received) but we don't have that
> option here since the 180 may contain either none or only host candidates
> so there are strong chances that no binding request would be received on
> them.
> 
> Thomas also suggested a second option which would be to also use INFO
> requests with trickled candidates as an indication that 180 was received.
> This however wouldn't work with half trickle so we are basically back to
> vanilla ICE for all (non-re) INVITEs.
> 
> Another option would be to mandate an INFO request with "end-of-candidates"
> in response to the 180, but that would be just the same as mandating PRACK
> support.
> 
> Thomas also suggested that the answerer can start sending INFOs right after
> it sends its answer in the 180 and then it can just resend the 180 if the
> INFOs result in a 481 response.
> 
> Personally I think this could potentially be made to work, but it would
> imply a level of complexity that considerably exceeds PRACK support.
> 
> Opinions?
> 
> Q2: How do we send INFOs? Are they blocking or do we just send them in
> parallel? If the latter, then what happens when an INFO fails because it is
> received out of order? Do we just tell the application to resend the
> candidates asap?
> 
> This also leads to the following question:
> 
> Q3: What exactly do we send in INFOs? Just the latest batch of freshly
> learned candidates or all candidates we've learned so far? Dale suggested
> that if we do this cumulatively we wouldn't need to worry about the case
> with the out-of-order INFOs from Q2 since the information gets resent
> anyway. A drawback here would obviously be that this adds more complexity
> for trickle ICE users (WebRTC applications specifically)
> 
> A third option would be to allow both and leave it to the application.
> 
> Comments are most welcome!
> 
> Emil
> 


- -- 
Marc Petit-Huguenin
Email: marc@petit-huguenin.org
Blog: http://blog.marc.petit-huguenin.org
Profile: http://www.linkedin.com/in/petithug
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From: Emil Ivov <emcho@jitsi.org>
Date: Fri, 5 Jul 2013 23:59:40 +0200
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Cc: MMUSIC IETF WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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On Fri, Jul 5, 2013 at 7:28 PM, Marc Petit-Huguenin <petithug@acm.org> wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA256
>
> Hi,
>
> I did not receive much comments on my proposal to use STUN packets to carry
> the new ICE candidates.  To summarize, instead of using the SDP in an INFO and
> or UPDATE transaction, the new candidates can be sent on the same path that
> the media will use by default (i.e. using the candidate that is copied in the
> m= and c= line).  Because this candidate is chosen to always succeed (using a
> TURN server), the candidate will always reach their destination.  This will
> remove any problem in the signaling path, because it is no longer used to
> update the candidates.

Here's my take:
Default candidates are not guaranteed to work. Also, you only get to
place one candidate as default. What if your peer is IPv6 only and you
chose an IPv4 candidate? Or, what if you are both on the same LAN with
limited Internet connectivity? Or what if the service you use doesn't
even have a TURN server? I am sure we can come up with many other
similar situations.

The only transport that's guaranteed to work prior to ICE is
signalling. There's nothing else that's as reliable as this. (If there
was then we probably wouldn't even need ICE in the first place).

Emil

--
https://jitsi.org

>
> Thanks.
>
> On 07/03/2013 07:13 AM, Emil Ivov wrote:
>> Hey all,
>>
>> Christer, Enrico and I are preparing the next version of Trickle ICE for
>> SIP. Now that discussions on BUNDLE and the plans seem to be winding down,
>> we wanted to run a few questions by the working group.
>>
>> Q1: Making reliable provisional responses and PRACK mandatory. Obviously
>> this would be nice to avoid, so the question is: is there a reasonable
>> mechanism to achieve this (and by reasonable, we mean something that
>> wouldn't be harder than implementing support for PRACK).
>>
>> There was some discussion about this back in April and there was a
>> suggestion for implementing a 5245-style hack where the answerer basically
>> resends the 180 until it knows that it has been received.
>>
>> 5245 uses connectivity checks for this (i.e. it stops 180 retransmissions
>> when the first connectivity check is received) but we don't have that
>> option here since the 180 may contain either none or only host candidates
>> so there are strong chances that no binding request would be received on
>> them.
>>
>> Thomas also suggested a second option which would be to also use INFO
>> requests with trickled candidates as an indication that 180 was received.
>> This however wouldn't work with half trickle so we are basically back to
>> vanilla ICE for all (non-re) INVITEs.
>>
>> Another option would be to mandate an INFO request with "end-of-candidates"
>> in response to the 180, but that would be just the same as mandating PRACK
>> support.
>>
>> Thomas also suggested that the answerer can start sending INFOs right after
>> it sends its answer in the 180 and then it can just resend the 180 if the
>> INFOs result in a 481 response.
>>
>> Personally I think this could potentially be made to work, but it would
>> imply a level of complexity that considerably exceeds PRACK support.
>>
>> Opinions?
>>
>> Q2: How do we send INFOs? Are they blocking or do we just send them in
>> parallel? If the latter, then what happens when an INFO fails because it is
>> received out of order? Do we just tell the application to resend the
>> candidates asap?
>>
>> This also leads to the following question:
>>
>> Q3: What exactly do we send in INFOs? Just the latest batch of freshly
>> learned candidates or all candidates we've learned so far? Dale suggested
>> that if we do this cumulatively we wouldn't need to worry about the case
>> with the out-of-order INFOs from Q2 since the information gets resent
>> anyway. A drawback here would obviously be that this adds more complexity
>> for trickle ICE users (WebRTC applications specifically)
>>
>> A third option would be to allow both and leave it to the application.
>>
>> Comments are most welcome!
>>
>> Emil
>>
>
>
> - --
> Marc Petit-Huguenin
> Email: marc@petit-huguenin.org
> Blog: http://blog.marc.petit-huguenin.org
> Profile: http://www.linkedin.com/in/petithug
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Cc: MMUSIC IETF WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

On 07/05/2013 02:59 PM, Emil Ivov wrote:
> On Fri, Jul 5, 2013 at 7:28 PM, Marc Petit-Huguenin <petithug@acm.org>
> wrote: Hi,
> 
> I did not receive much comments on my proposal to use STUN packets to
> carry the new ICE candidates.  To summarize, instead of using the SDP in an
> INFO and or UPDATE transaction, the new candidates can be sent on the same
> path that the media will use by default (i.e. using the candidate that is
> copied in the m= and c= line).  Because this candidate is chosen to always
> succeed (using a TURN server), the candidate will always reach their
> destination.  This will remove any problem in the signaling path, because
> it is no longer used to update the candidates.
> 
>> Here's my take: Default candidates are not guaranteed to work.

Sorry, my summary was incorrect.  It is assumed that half-trickle is used (for
the same reason than in draft-ivov-mmusic-trickle-ice-sip section 3), so the
offerer sends the full set of candidates.

>> Also, you only get to place one candidate as default. What if your peer
>> is IPv6 only and you chose an IPv4 candidate?

With the offerer full set of candidates, it is easy to find the IPv6 TURN
candidate.

> Or, what if you are both on the same LAN with
>> limited Internet connectivity? Or what if the service you use doesn't 
>> even have a TURN server?

Then what about sending the answerer candidates using all the offerer
candidates available, but in reversed priority order?

>> I am sure we can come up with many other similar situations.
> 
>> The only transport that's guaranteed to work prior to ICE is signalling.
>> There's nothing else that's as reliable as this. (If there was then we
>> probably wouldn't even need ICE in the first place).

ICE is less about reliability than about latency, i.e. so far nobody proposed
to tunnel the media over SIP :-)

> 
>> Emil
> 
>> -- https://jitsi.org
> 
> 
> Thanks.
> 
> On 07/03/2013 07:13 AM, Emil Ivov wrote:
>>>> Hey all,
>>>> 
>>>> Christer, Enrico and I are preparing the next version of Trickle ICE
>>>> for SIP. Now that discussions on BUNDLE and the plans seem to be
>>>> winding down, we wanted to run a few questions by the working group.
>>>> 
>>>> Q1: Making reliable provisional responses and PRACK mandatory.
>>>> Obviously this would be nice to avoid, so the question is: is there a
>>>> reasonable mechanism to achieve this (and by reasonable, we mean
>>>> something that wouldn't be harder than implementing support for
>>>> PRACK).
>>>> 
>>>> There was some discussion about this back in April and there was a 
>>>> suggestion for implementing a 5245-style hack where the answerer
>>>> basically resends the 180 until it knows that it has been received.
>>>> 
>>>> 5245 uses connectivity checks for this (i.e. it stops 180
>>>> retransmissions when the first connectivity check is received) but we
>>>> don't have that option here since the 180 may contain either none or
>>>> only host candidates so there are strong chances that no binding
>>>> request would be received on them.
>>>> 
>>>> Thomas also suggested a second option which would be to also use
>>>> INFO requests with trickled candidates as an indication that 180 was
>>>> received. This however wouldn't work with half trickle so we are
>>>> basically back to vanilla ICE for all (non-re) INVITEs.
>>>> 
>>>> Another option would be to mandate an INFO request with
>>>> "end-of-candidates" in response to the 180, but that would be just
>>>> the same as mandating PRACK support.
>>>> 
>>>> Thomas also suggested that the answerer can start sending INFOs right
>>>> after it sends its answer in the 180 and then it can just resend the
>>>> 180 if the INFOs result in a 481 response.
>>>> 
>>>> Personally I think this could potentially be made to work, but it
>>>> would imply a level of complexity that considerably exceeds PRACK
>>>> support.
>>>> 
>>>> Opinions?
>>>> 
>>>> Q2: How do we send INFOs? Are they blocking or do we just send them
>>>> in parallel? If the latter, then what happens when an INFO fails
>>>> because it is received out of order? Do we just tell the application
>>>> to resend the candidates asap?
>>>> 
>>>> This also leads to the following question:
>>>> 
>>>> Q3: What exactly do we send in INFOs? Just the latest batch of
>>>> freshly learned candidates or all candidates we've learned so far?
>>>> Dale suggested that if we do this cumulatively we wouldn't need to
>>>> worry about the case with the out-of-order INFOs from Q2 since the
>>>> information gets resent anyway. A drawback here would obviously be
>>>> that this adds more complexity for trickle ICE users (WebRTC
>>>> applications specifically)
>>>> 
>>>> A third option would be to allow both and leave it to the
>>>> application.
>>>> 
>>>> Comments are most welcome!
>>>> 
>>>> Emil
>>>> 
> 
> 
> _______________________________________________ mmusic mailing list 
> mmusic@ietf.org https://www.ietf.org/mailman/listinfo/mmusic
> 

- -- 
Marc Petit-Huguenin
Email: marc@petit-huguenin.org
Blog: http://blog.marc.petit-huguenin.org
Profile: http://www.linkedin.com/in/petithug
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On 06.07.13, 00:45, Marc Petit-Huguenin wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA256
>
> On 07/05/2013 02:59 PM, Emil Ivov wrote:
>> On Fri, Jul 5, 2013 at 7:28 PM, Marc Petit-Huguenin <petithug@acm.org>
>> wrote: Hi,
>>
>> I did not receive much comments on my proposal to use STUN packets to
>> carry the new ICE candidates.  To summarize, instead of using the SDP in an
>> INFO and or UPDATE transaction, the new candidates can be sent on the same
>> path that the media will use by default (i.e. using the candidate that is
>> copied in the m= and c= line).  Because this candidate is chosen to always
>> succeed (using a TURN server), the candidate will always reach their
>> destination.  This will remove any problem in the signaling path, because
>> it is no longer used to update the candidates.
>>
>>> Here's my take: Default candidates are not guaranteed to work.
>
> Sorry, my summary was incorrect.  It is assumed that half-trickle is used (for
> the same reason than in draft-ivov-mmusic-trickle-ice-sip section 3), so the
> offerer sends the full set of candidates.

Are you suggesting that we would always need to do half trickle, even 
when we know the other party has support for trickle and can switch to full?
>
>>> Also, you only get to place one candidate as default. What if your peer
>>> is IPv6 only and you chose an IPv4 candidate?
>
> With the offerer full set of candidates, it is easy to find the IPv6 TURN
> candidate.
>
>> Or, what if you are both on the same LAN with
>>> limited Internet connectivity? Or what if the service you use doesn't
>>> even have a TURN server?
>
> Then what about sending the answerer candidates using all the offerer
> candidates available, but in reversed priority order?

Even then, endpoint dependent filtering (which seems to be the default 
these days) requires that both ends start sending packets simultaneously 
for server-reflexive addresses to work.

In the case above the trickled candidates wouldn't be able to reach the 
offerer because it wouldn't be sending anything toward the answerer.

>>> I am sure we can come up with many other similar situations.
>>
>>> The only transport that's guaranteed to work prior to ICE is signalling.
>>> There's nothing else that's as reliable as this. (If there was then we
>>> probably wouldn't even need ICE in the first place).
>
> ICE is less about reliability than about latency,

I believe it's always been about both. (Otherwise we would have probably 
stopped at STUN).

> i.e. so far nobody proposed
> to tunnel the media over SIP :-)

You'd be surprised ;)

https://java.net/projects/jain-sip-applet-phone

Emil
>
>>
>>> Emil
>>
>>> -- https://jitsi.org
>>
>>
>> Thanks.
>>
>> On 07/03/2013 07:13 AM, Emil Ivov wrote:
>>>>> Hey all,
>>>>>
>>>>> Christer, Enrico and I are preparing the next version of Trickle ICE
>>>>> for SIP. Now that discussions on BUNDLE and the plans seem to be
>>>>> winding down, we wanted to run a few questions by the working group.
>>>>>
>>>>> Q1: Making reliable provisional responses and PRACK mandatory.
>>>>> Obviously this would be nice to avoid, so the question is: is there a
>>>>> reasonable mechanism to achieve this (and by reasonable, we mean
>>>>> something that wouldn't be harder than implementing support for
>>>>> PRACK).
>>>>>
>>>>> There was some discussion about this back in April and there was a
>>>>> suggestion for implementing a 5245-style hack where the answerer
>>>>> basically resends the 180 until it knows that it has been received.
>>>>>
>>>>> 5245 uses connectivity checks for this (i.e. it stops 180
>>>>> retransmissions when the first connectivity check is received) but we
>>>>> don't have that option here since the 180 may contain either none or
>>>>> only host candidates so there are strong chances that no binding
>>>>> request would be received on them.
>>>>>
>>>>> Thomas also suggested a second option which would be to also use
>>>>> INFO requests with trickled candidates as an indication that 180 was
>>>>> received. This however wouldn't work with half trickle so we are
>>>>> basically back to vanilla ICE for all (non-re) INVITEs.
>>>>>
>>>>> Another option would be to mandate an INFO request with
>>>>> "end-of-candidates" in response to the 180, but that would be just
>>>>> the same as mandating PRACK support.
>>>>>
>>>>> Thomas also suggested that the answerer can start sending INFOs right
>>>>> after it sends its answer in the 180 and then it can just resend the
>>>>> 180 if the INFOs result in a 481 response.
>>>>>
>>>>> Personally I think this could potentially be made to work, but it
>>>>> would imply a level of complexity that considerably exceeds PRACK
>>>>> support.
>>>>>
>>>>> Opinions?
>>>>>
>>>>> Q2: How do we send INFOs? Are they blocking or do we just send them
>>>>> in parallel? If the latter, then what happens when an INFO fails
>>>>> because it is received out of order? Do we just tell the application
>>>>> to resend the candidates asap?
>>>>>
>>>>> This also leads to the following question:
>>>>>
>>>>> Q3: What exactly do we send in INFOs? Just the latest batch of
>>>>> freshly learned candidates or all candidates we've learned so far?
>>>>> Dale suggested that if we do this cumulatively we wouldn't need to
>>>>> worry about the case with the out-of-order INFOs from Q2 since the
>>>>> information gets resent anyway. A drawback here would obviously be
>>>>> that this adds more complexity for trickle ICE users (WebRTC
>>>>> applications specifically)
>>>>>
>>>>> A third option would be to allow both and leave it to the
>>>>> application.
>>>>>
>>>>> Comments are most welcome!
>>>>>
>>>>> Emil
>>>>>
>>
>>
>> _______________________________________________ mmusic mailing list
>> mmusic@ietf.org https://www.ietf.org/mailman/listinfo/mmusic
>>
>
> - --
> Marc Petit-Huguenin
> Email: marc@petit-huguenin.org
> Blog: http://blog.marc.petit-huguenin.org
> Profile: http://www.linkedin.com/in/petithug
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-- 
https://jitsi.org

From petithug@acm.org  Fri Jul  5 16:14:43 2013
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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On 07/05/2013 03:53 PM, Emil Ivov wrote:
> 
> 
> On 06.07.13, 00:45, Marc Petit-Huguenin wrote: On 07/05/2013 02:59 PM, Emil
> Ivov wrote:
>>>> On Fri, Jul 5, 2013 at 7:28 PM, Marc Petit-Huguenin
>>>> <petithug@acm.org> wrote: Hi,
>>>> 
>>>> I did not receive much comments on my proposal to use STUN packets
>>>> to carry the new ICE candidates.  To summarize, instead of using the
>>>> SDP in an INFO and or UPDATE transaction, the new candidates can be
>>>> sent on the same path that the media will use by default (i.e. using
>>>> the candidate that is copied in the m= and c= line).  Because this
>>>> candidate is chosen to always succeed (using a TURN server), the
>>>> candidate will always reach their destination.  This will remove any
>>>> problem in the signaling path, because it is no longer used to update
>>>> the candidates.
>>>> 
>>>>> Here's my take: Default candidates are not guaranteed to work.
> 
> Sorry, my summary was incorrect.  It is assumed that half-trickle is used
> (for the same reason than in draft-ivov-mmusic-trickle-ice-sip section 3),
> so the offerer sends the full set of candidates.
> 
>> Are you suggesting that we would always need to do half trickle, even
>> when we know the other party has support for trickle and can switch to
>> full?

No, I just am saying that the solution that I propose does not take in account
full-trickle.

> 
>>>>> Also, you only get to place one candidate as default. What if your
>>>>> peer is IPv6 only and you chose an IPv4 candidate?
> 
> With the offerer full set of candidates, it is easy to find the IPv6 TURN 
> candidate.
> 
>>>> Or, what if you are both on the same LAN with
>>>>> limited Internet connectivity? Or what if the service you use
>>>>> doesn't even have a TURN server?
> 
> Then what about sending the answerer candidates using all the offerer 
> candidates available, but in reversed priority order?
> 
>> Even then, endpoint dependent filtering (which seems to be the default
>> these days) requires that both ends start sending packets simultaneously
>> for server-reflexive addresses to work.
> 
>> In the case above the trickled candidates wouldn't be able to reach the
>> offerer because it wouldn't be sending anything toward the answerer.

Right.

> 
>>>>> I am sure we can come up with many other similar situations.
>>>> 
>>>>> The only transport that's guaranteed to work prior to ICE is
>>>>> signalling. There's nothing else that's as reliable as this. (If
>>>>> there was then we probably wouldn't even need ICE in the first
>>>>> place).
> 
> ICE is less about reliability than about latency,
> 
>> I believe it's always been about both. (Otherwise we would have probably
>> stopped at STUN).

You mean TURN?  As long as you do not care about latency, TURN is all you need
as it always works (and that's all we used for a long time, even if it was not
called TURN).  ICE was invented because latency was important - so STUN was
useful - but there is no reliable way to predict if it will work.

> 
> i.e. so far nobody proposed to tunnel the media over SIP :-)
> 
>> You'd be surprised ;)
> 
>> https://java.net/projects/jain-sip-applet-phone

Hmm, I do not see SIP as media transport in this project.  What I meant was
something like transporting the media in a SIP INFO.

> 
>> Emil
> 
>>>> 
>>>>> Emil
>>>> 
>>>>> -- https://jitsi.org
>>>> 
>>>> 
>>>> Thanks.
>>>> 
>>>> On 07/03/2013 07:13 AM, Emil Ivov wrote:
>>>>>>> Hey all,
>>>>>>> 
>>>>>>> Christer, Enrico and I are preparing the next version of
>>>>>>> Trickle ICE for SIP. Now that discussions on BUNDLE and the
>>>>>>> plans seem to be winding down, we wanted to run a few questions
>>>>>>> by the working group.
>>>>>>> 
>>>>>>> Q1: Making reliable provisional responses and PRACK mandatory. 
>>>>>>> Obviously this would be nice to avoid, so the question is: is
>>>>>>> there a reasonable mechanism to achieve this (and by
>>>>>>> reasonable, we mean something that wouldn't be harder than
>>>>>>> implementing support for PRACK).
>>>>>>> 
>>>>>>> There was some discussion about this back in April and there
>>>>>>> was a suggestion for implementing a 5245-style hack where the
>>>>>>> answerer basically resends the 180 until it knows that it has
>>>>>>> been received.
>>>>>>> 
>>>>>>> 5245 uses connectivity checks for this (i.e. it stops 180 
>>>>>>> retransmissions when the first connectivity check is received)
>>>>>>> but we don't have that option here since the 180 may contain
>>>>>>> either none or only host candidates so there are strong chances
>>>>>>> that no binding request would be received on them.
>>>>>>> 
>>>>>>> Thomas also suggested a second option which would be to also
>>>>>>> use INFO requests with trickled candidates as an indication
>>>>>>> that 180 was received. This however wouldn't work with half
>>>>>>> trickle so we are basically back to vanilla ICE for all
>>>>>>> (non-re) INVITEs.
>>>>>>> 
>>>>>>> Another option would be to mandate an INFO request with 
>>>>>>> "end-of-candidates" in response to the 180, but that would be
>>>>>>> just the same as mandating PRACK support.
>>>>>>> 
>>>>>>> Thomas also suggested that the answerer can start sending INFOs
>>>>>>> right after it sends its answer in the 180 and then it can just
>>>>>>> resend the 180 if the INFOs result in a 481 response.
>>>>>>> 
>>>>>>> Personally I think this could potentially be made to work, but
>>>>>>> it would imply a level of complexity that considerably exceeds
>>>>>>> PRACK support.
>>>>>>> 
>>>>>>> Opinions?
>>>>>>> 
>>>>>>> Q2: How do we send INFOs? Are they blocking or do we just send
>>>>>>> them in parallel? If the latter, then what happens when an INFO
>>>>>>> fails because it is received out of order? Do we just tell the
>>>>>>> application to resend the candidates asap?
>>>>>>> 
>>>>>>> This also leads to the following question:
>>>>>>> 
>>>>>>> Q3: What exactly do we send in INFOs? Just the latest batch of 
>>>>>>> freshly learned candidates or all candidates we've learned so
>>>>>>> far? Dale suggested that if we do this cumulatively we wouldn't
>>>>>>> need to worry about the case with the out-of-order INFOs from
>>>>>>> Q2 since the information gets resent anyway. A drawback here
>>>>>>> would obviously be that this adds more complexity for trickle
>>>>>>> ICE users (WebRTC applications specifically)
>>>>>>> 
>>>>>>> A third option would be to allow both and leave it to the 
>>>>>>> application.
>>>>>>> 
>>>>>>> Comments are most welcome!
>>>>>>> 
>>>>>>> Emil
>>>>>>> 
>>>> 
>>>> 
>>>> _______________________________________________ mmusic mailing list 
>>>> mmusic@ietf.org https://www.ietf.org/mailman/listinfo/mmusic
>>>> 
> 
>> 
> 

- -- 
Marc Petit-Huguenin
Email: marc@petit-huguenin.org
Blog: http://blog.marc.petit-huguenin.org
Profile: http://www.linkedin.com/in/petithug
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Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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On 7/3/13 1:06 AM, Mo Zanaty (mzanaty) wrote:
> To clarify, I have no plan to use or abuse Plan A. The implementations I'm most interested in are all closer to Plan B. However, I'm aware of widely deployed implementations (from other vendors) which are closer to Plan A in general, and specifically for the thumbnail example below (one m-line per thumbnail, all with the same port/PT). While I personally would not chose this, I see no reason to forbid it, because I recognize it is just personal preference.

Since there currently is no BUNDLE, I don't know how you can say an 
implementation is closer to any of the plans. Instead, you can only 
hypothesize on how such an application might use any of the bundling 
schemes if it became available.

	Thanks,
	Paul

> Mo
>
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Tuesday, July 02, 2013 6:40 PM
> To: Mo Zanaty (mzanaty)
> Cc: Christer Holmberg; mmusic@ietf.org
> Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>
> On 7/2/13 4:33 PM, Mo Zanaty (mzanaty) wrote:
>>> You could mandate that multiple m-lines of thumbnails must be combined into a single m-line with a=ssrc/max-ssrc/multiple-render/whatever.
>>
>> It seems you both want to mandate this (Plan B). Why? I see no reason for bundle to pick a plan. Let the plan wars rage outside bundle.
>
> So you want to use plan A *without* the association, so that the *only*
> purpose of the extra m-lines is to denote the number of flows?
>
> How would the receiver know if an incoming packet was associated with
> one of the flows defined by the m-lines?
>
> How would the receiver know if each ssrc should be treated as a
> different flow, or if appid should be used to define flows?
>
> I think you abuse plan A. Its different with plan B or No Plan. There
> you still need the ability to associate packets to m-lines, but that
> isn't sufficient to map them to flows. It is assumed that "extra"
> information, learned by different means, is used to sort out the flows.
>
> 	Thanks,
> 	Paul
>
>> Mo
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>> Sent: Tuesday, July 02, 2013 4:00 PM
>> To: Christer Holmberg
>> Cc: Mo Zanaty (mzanaty); mmusic@ietf.org
>> Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>>
>> +1
>> (Except that I *am* Paul)
>>
>> On 7/2/13 2:30 PM, Christer Holmberg wrote:
>>> Hi,
>>>
>>> I am not Paul, but just to verify that I understand his issue, the question is:
>>>
>>> Why do you need 4 m- lines for the thumbnails, when you could use 1 m- line?
>>>
>>> Regards,
>>>
>>> Christer
>>>
>>> -----Alkuperäinen viesti-----
>>> Lähettäjä: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] Puolesta Mo Zanaty (mzanaty)
>>> Lähetetty: 2. heinäkuuta 2013 19:09
>>> Vastaanottaja: Paul Kyzivat
>>> Kopio: mmusic@ietf.org
>>> Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>>>
>>> An attempt at "an answer that makes sense". (Just snippets, not full SDP.)
>>>
>>> m=video 10000 RTP/SAVPF 96
>>> a=rtpmap:96 H264/90000
>>> a=fmtp:96 profile-level-id=42001F <--- HD 720p30 main speaker
>>>
>>> m=video 10000 RTP/SAVPF 97
>>> a=rtpmap:97 H264/90000
>>> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>>>
>>> m=video 10000 RTP/SAVPF 97
>>> a=rtpmap:97 H264/90000
>>> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>>>
>>> m=video 10000 RTP/SAVPF 97
>>> a=rtpmap:97 H264/90000
>>> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>>>
>>> m=video 10000 RTP/SAVPF 97
>>> a=rtpmap:97 H264/90000
>>> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>>>
>>> No need to map thumbnail packets to specific m-lines, so reuse of PT=97 is fine. Just render the last 4 thumbnail sources received (in addition to the main speaker).
>>>
>>> You could mandate that PT must be unique to allow unique mapping.
>>>
>>> You could mandate that multiple m-lines of thumbnails must be combined into a single m-line with a=ssrc/max-ssrc/multiple-render/whatever.
>>>
>>> Which do you want to mandate and why?
>>>
>>> Cheers,
>>> Mo
>>>
>>>
>>> -----Original Message-----
>>> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>>> Sent: Tuesday, July 02, 2013 10:06 AM
>>> To: Mo Zanaty (mzanaty)
>>> Cc: mmusic@ietf.org
>>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>>>
>>> On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
>>>> Informative descriptions of possible mappings are fine, but nothing normative, please. Some applications won't care about the mapping, and others may care but use a different mapping from those described. The only normative statement you could make is something like: "If you care about the mapping, you MUST have a mapping mechanism. You MAY use the mechanism(s) described here, or some other mechanism." But why mandate such a tautology?
>>>>
>>>> The assumption below that multiple m-lines always require different m-line-specific processing of the packets also assumes Plan B, i.e. that multiple streams with similar processing should always be signaled with a single m-line (for example, using a=ssrc or max-ssrc). Plan A purists would use multiple m-lines even for multiple streams with the same processing. I don't think we want to force bundle to pick a plan.
>>>
>>> I'm going to keep asking this question until I get an answer that makes
>>> sense:
>>>
>>> Please explain to me a meaningful situation when there are multiple m-lines in a bundle and there is insufficient information to associate each packet with an m-line.
>>>
>>> The *point* of plan A is that there be an m-line per "flow", and that the offerer is enumerating the flows. So clearly it knows the mapping.
>>> If the answerer isn't capable of doing the same mapping, then the use of plan A has failed.
>>>
>>> The part that I want to be normative is that an O/A is not valid unless the mapping is possible with the information in the O/A, and that the offerer and the answerer would agree about the mapping of each packet. I don't care if they actually do the mapping, though if they are doing any sensible processing then they will be doing something *equivalent* to the mapping.
>>>
>>> 	Thanks,
>>> 	Paul
>>>
>>>> Mo
>>>>
>>>> -----Original Message-----
>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>>> Behalf Of Paul Kyzivat
>>>> Sent: Saturday, June 29, 2013 12:12 PM
>>>> To: mmusic@ietf.org
>>>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>>>>
>>>> On 6/29/13 8:50 AM, Colin Perkins wrote:
>>>>> Christer,
>>>>>
>>>>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
>>>>>> There has been some discussions about whether BUNDLE should mandate that users are mandated to always (no matter what transport protocols are used in the BUNDLE group) have a mechanism to map received data to an m- line, or whether it from a generic BUNDLE perspective should be optional - IF there would be cases where it's not needed.
>>>>>>
>>>>>> We haven't had that much discussion about it yet, so I will not ask
>>>>>> a DECISION question at this point, but I would really like to get
>>>>>> some input from people who have opinions about this :)
>>>>>
>>>>> My opinion: BUNDLE needs to mandate a single algorithm for mapping from RTP flows to m= lines, for those applications that care about such a mapping. I don't think it should mandate that all applications need to care.
>>>>
>>>> I just replied elsewhere on this.
>>>> My opinion turns out to be the exact opposite of yours: we need to
>>>> support multiple algorithms, and that all applications should care.
>>>>
>>>> I won't repeat the stuff about multiple algorithms.
>>>>
>>>> I will repeat what I have said about the need for being able to associate:
>>>>
>>>> There must have been *some* reason that two m-lines were used, rather
>>>> than just one. Each m-line heads a media section of the SDP that
>>>> contains a bunch of declarations. Something must be different in those
>>>> two media sections, or else a single media section would have been
>>>> enough. Presumably whatever it is that is different is intended to
>>>> affect how received packets are processed or interpreted. (If not,
>>>> again there is no need for it to be there.) If you can't associate the
>>>> packet with the m-line, then you don't know which of the multiple
>>>> interpretations to apply to the packet.
>>>>
>>>> Maybe there is some exception to this, though I haven't come up with
>>>> anything. If there is, then I hope someone will call it out. If so,
>>>> then the constraint can be tightened.
>>>>
>>>> 	Thanks,
>>>> 	Paul
>>>>
>>>>
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>>
>
>


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Cc: "mmusic_ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
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On 7/3/13 7:38 AM, Christer Holmberg wrote:
>
> Hi,
> I guess one advantage with 4 m- lines is that, if the remote endpoint
> does not support BUNDLE, it will allow to have a separate 5-tuple for
> each thumbnail stream, without having to send a "fallack" offer.
> When both endpoints do support BUNDLE it makes little sense, but when
> thinking about it I really wonder if there is a reason to forbid it.

Yes, perhaps it is an interesting case to consider.
I think there is still an issue, but maybe it needs to be refined.

ISTM that *if* there is a subset of bundled media-sections that are 
indistinguishable other than by position or the address-info that will 
become common when the bundle is accepted, then it is only necessary 
that there be a way to associate a media packet to the subset, not to an 
individual m-line within that subset. (This is because being able to do 
so would not yield any useful information.)

Unfortunately, trying to turn that into clear text in the draft will be 
a challenge. :-(

The point I am trying to make about this is that when there is something 
*different* about the media sections, then only by associating media 
packets to m-lines is is possible to associate that other info with the 
packet. For instance, suppose each of the bundled m-lines has an 
associated a=label attribute. Then it is necessary to be able to 
associate a packet to an m-line in order to know which label applies to 
it, and that can impact what application level processing is done.

	Thanks,
	Paul

> Regards,
> Christer
>
> Sent from */Windows/* using *TouchDown* (www.nitrodesk.com)
>
> -----Original Message-----
> *From:* Mo Zanaty (mzanaty) [mzanaty@cisco.com]
> *To:* Paul Kyzivat [pkyzivat@alum.mit.edu]
> *CC:* Christer Holmberg [christer.holmberg@ericsson.com];
> mmusic@ietf.org [mmusic@ietf.org]
> *Subject:* RE: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism
> to map received data to m- line?
> To clarify, I have no plan to use or abuse Plan A. The implementations
> I'm most interested in are all closer to Plan B. However, I'm aware of
> widely deployed implementations (from other vendors) which are closer to
> Plan A in general, and specifically for the thumbnail example below (one
> m-line per thumbnail, all with the same port/PT). While I personally
> would not chose this, I see no reason to forbid it, because I recognize
> it is just personal preference.
>
> Mo
>
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Tuesday, July 02, 2013 6:40 PM
> To: Mo Zanaty (mzanaty)
> Cc: Christer Holmberg; mmusic@ietf.org
> Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to
> map received data to m- line?
>
> On 7/2/13 4:33 PM, Mo Zanaty (mzanaty) wrote:
>>> You could mandate that multiple m-lines of thumbnails must be combined into a single m-line with a=ssrc/max-ssrc/multiple-render/whatever.
>>
>> It seems you both want to mandate this (Plan B). Why? I see no reason for bundle to pick a plan. Let the plan wars rage outside bundle.
>
> So you want to use plan A *without* the association, so that the *only*
> purpose of the extra m-lines is to denote the number of flows?
>
> How would the receiver know if an incoming packet was associated with
> one of the flows defined by the m-lines?
>
> How would the receiver know if each ssrc should be treated as a
> different flow, or if appid should be used to define flows?
>
> I think you abuse plan A. Its different with plan B or No Plan. There
> you still need the ability to associate packets to m-lines, but that
> isn't sufficient to map them to flows. It is assumed that "extra"
> information, learned by different means, is used to sort out the flows.
>
>          Thanks,
>          Paul
>
>> Mo
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>> Sent: Tuesday, July 02, 2013 4:00 PM
>> To: Christer Holmberg
>> Cc: Mo Zanaty (mzanaty); mmusic@ietf.org
>> Subject: Re: VS: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>>
>> +1
>> (Except that I *am* Paul)
>>
>> On 7/2/13 2:30 PM, Christer Holmberg wrote:
>>> Hi,
>>>
>>> I am not Paul, but just to verify that I understand his issue, the question is:
>>>
>>> Why do you need 4 m- lines for the thumbnails, when you could use 1 m- line?
>>>
>>> Regards,
>>>
>>> Christer
>>>
>>> -----Alkuperäinen viesti-----
>>> Lähettäjä: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] Puolesta Mo Zanaty (mzanaty)
>>> Lähetetty: 2. heinäkuuta 2013 19:09
>>> Vastaanottaja: Paul Kyzivat
>>> Kopio: mmusic@ietf.org
>>> Aihe: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>>>
>>> An attempt at "an answer that makes sense". (Just snippets, not full SDP.)
>>>
>>> m=video 10000 RTP/SAVPF 96
>>> a=rtpmap:96 H264/90000
>>> a=fmtp:96 profile-level-id=42001F <--- HD 720p30 main speaker
>>>
>>> m=video 10000 RTP/SAVPF 97
>>> a=rtpmap:97 H264/90000
>>> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>>>
>>> m=video 10000 RTP/SAVPF 97
>>> a=rtpmap:97 H264/90000
>>> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>>>
>>> m=video 10000 RTP/SAVPF 97
>>> a=rtpmap:97 H264/90000
>>> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>>>
>>> m=video 10000 RTP/SAVPF 97
>>> a=rtpmap:97 H264/90000
>>> a=fmtp:97 profile-level-id=42000C <--- SD 240p15 thumbnail a=recvonly
>>>
>>> No need to map thumbnail packets to specific m-lines, so reuse of PT=97 is fine. Just render the last 4 thumbnail sources received (in addition to the main speaker).
>>>
>>> You could mandate that PT must be unique to allow unique mapping.
>>>
>>> You could mandate that multiple m-lines of thumbnails must be combined into a single m-line with a=ssrc/max-ssrc/multiple-render/whatever.
>>>
>>> Which do you want to mandate and why?
>>>
>>> Cheers,
>>> Mo
>>>
>>>
>>> -----Original Message-----
>>> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>>> Sent: Tuesday, July 02, 2013 10:06 AM
>>> To: Mo Zanaty (mzanaty)
>>> Cc: mmusic@ietf.org
>>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>>>
>>> On 6/30/13 11:32 PM, Mo Zanaty (mzanaty) wrote:
>>>> Informative descriptions of possible mappings are fine, but nothing normative, please. Some applications won't care about the mapping, and others may care but use a different mapping from those described. The only normative statement you could make is something  like: "If you care about the mapping, you MUST have a mapping
> mechanism. You MAY use the mechanism(s) described here, or some other
> mechanism." But why mandate such a tautology?
>>>>
>>>> The assumption below that multiple m-lines always require different m-line-specific processing of the packets also assumes Plan B, i.e. that multiple streams with similar processing should always be signaled with a single m-line (for example, using a=ssrc  or max-ssrc). Plan A purists would use multiple m-lines even for
> multiple streams with the same processing. I don't think we want to
> force bundle to pick a plan.
>>>
>>> I'm going to keep asking this question until I get an answer that makes
>>> sense:
>>>
>>> Please explain to me a meaningful situation when there are multiple m-lines in a bundle and there is insufficient information to associate each packet with an m-line.
>>>
>>> The *point* of plan A is that there be an m-line per "flow", and that the offerer is enumerating the flows. So clearly it knows the mapping.
>>> If the answerer isn't capable of doing the same mapping, then the use of plan A has failed.
>>>
>>> The part that I want to be normative is that an O/A is not valid unless the mapping is possible with the information in the O/A, and that the offerer and the answerer would agree about the mapping of each packet. I don't care if they actually do the mapping,  though if they are doing any sensible processing then they will be
> doing something *equivalent* to the mapping.
>>>
>>>       Thanks,
>>>       Paul
>>>
>>>> Mo
>>>>
>>>> -----Original Message-----
>>>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>>>> Behalf Of Paul Kyzivat
>>>> Sent: Saturday, June 29, 2013 12:12 PM
>>>> To: mmusic@ietf.org
>>>> Subject: Re: [MMUSIC] BUNDLE DISCUSION: Always mandate mechanism to map received data to m- line?
>>>>
>>>> On 6/29/13 8:50 AM, Colin Perkins wrote:
>>>>> Christer,
>>>>>
>>>>> On 25 Jun 2013, at 12:16, Christer Holmberg wrote:
>>>>>> There has been some discussions about whether BUNDLE should mandate that users are mandated to always (no matter what transport protocols are used in the BUNDLE group) have a mechanism to map received data to an m- line, or whether it from a generic BUNDLE  perspective should be optional - IF there would be cases where it's
> not needed.
>>>>>>
>>>>>> We haven't had that much discussion about it yet, so I will not ask
>>>>>> a DECISION question at this point, but I would really like to get
>>>>>> some input from people who have opinions about this :)
>>>>>
>>>>> My opinion: BUNDLE needs to mandate a single algorithm for mapping from RTP flows to m= lines, for those applications that care about such a mapping. I don't think it should mandate that all applications need to care.
>>>>
>>>> I just replied elsewhere on this.
>>>> My opinion turns out to be the exact opposite of yours: we need to
>>>> support multiple algorithms, and that all applications should care.
>>>>
>>>> I won't repeat the stuff about multiple algorithms.
>>>>
>>>> I will repeat what I have said about the need for being able to associate:
>>>>
>>>> There must have been *some* reason that two m-lines were used, rather
>>>> than just one. Each m-line heads a media section of the SDP that
>>>> contains a bunch of declarations. Something must be different in those
>>>> two media sections, or else a single media section would have been
>>>> enough. Presumably whatever it is that is different is intended to
>>>> affect how received packets are processed or interpreted. (If not,
>>>> again there is no need for it to be there.) If you can't associate the
>>>> packet with the m-line, then you don't know which of the multiple
>>>> interpretations to apply to the packet.
>>>>
>>>> Maybe there is some exception to this, though I haven't come up with
>>>> anything. If there is, then I hope someone will call it out. If so,
>>>> then the constraint can be tightened.
>>>>
>>>>      Thanks,
>>>>      Paul
>>>>
>>>>
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>>>> mmusic mailing list
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>>
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From randy@qti.qualcomm.com  Sun Jul  7 15:43:34 2013
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To: Christian Groves <Christian.Groves@nteczone.com>, <mmusic@ietf.org>
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Subject: Re: [MMUSIC] Notes from Orlando human language draft discussion
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At 4:50 PM +1100 3/25/13, Christian Groves wrote:

>  Hello,
>
>  One thing that doesn't seem to be noted is an overlap with the work 
> of the CLUE WG. There has been some discussion in the CLUE working 
> group about adding attributes to captures to describe the language, 
> whether sub-titles are used, whether a stream comes from an 
> interpreter, etc. Where multiple streams are used I would expect 
> any work in this area should be aligned.
>
>  Regards, Christian

Hi Christian,

Thanks for the info.  Would it make sense to talk about this in 
Berlin?  I agree that if there is overlap in the approaches we should 
align them.

Also, I have an updated draft that was just uploaded.  It is now 
named 'draft-gellens-mmusic-negotiating-human-language-00' (adding 
-mmusic and hence reverting back to -00).

-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly selected tag: ---------------
If you can't annoy somebody there's little point in writing --Kingsley Amis

From ari.keranen@ericsson.com  Sun Jul  7 15:55:40 2013
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From: =?ISO-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
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Subject: [MMUSIC] Draft submission deadlines change
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FYI, since many of you may have missed this: "Please note that for IETF 
87, there is only one deadline for draft submission: Monday 15th July."

http://www.ietf.org/mail-archive/web/ietf-announce/current/msg11651.html

However, as the announcement says: "But please do note that working 
group chairs will continue to make smart decisions about what topics are 
worthwhile for discussing in a session in the upcoming meeting, and will 
also set their agendas in a timely manner and create deadlines for their 
working groups that must be adhered to. The earlier new drafts are 
submitted, the more time there is to talk about them on the mailing 
lists and consider them for the session agendas."

So please, submit rather early than late.


Thanks,
Ari

From thomas.stach@siemens-enterprise.com  Mon Jul  8 02:25:37 2013
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From: "Stach, Thomas" <thomas.stach@siemens-enterprise.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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A few thoughts/response inline

> -----Original Message-----
> From: Emil Ivov [mailto:emcho@jitsi.org]
> Sent: Friday, 05 July, 2013 16:13
> To: Stach, Thomas
> Cc: MMUSIC IETF WG
> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
>=20
> Hey Thomas
>=20
> On 04.07.13, 15:15, Stach, Thomas wrote:
> > Emil,
> > inline
> >
> >> -----Original Message-----
> >> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >> Behalf Of Emil Ivov
> >> Sent: Wednesday, 03 July, 2013 16:13
> >> To: MMUSIC IETF WG
> >> Subject: [MMUSIC] Trickle ICE for SIP Questions
> >>
> >> Hey all,
> >>
> >> Christer, Enrico and I are preparing the next version of Trickle ICE
> >> for
> >> SIP. Now that discussions on BUNDLE and the plans seem to be winding
> >> down, we wanted to run a few questions by the working group.
> >>
> >> Q1: Making reliable provisional responses and PRACK mandatory.
> >> Obviously
> >> this would be nice to avoid, so the question is: is there a
> reasonable
> >> mechanism to achieve this (and by reasonable, we mean something that
> >> wouldn't be harder than implementing support for PRACK).
> >>
> >> There was some discussion about this back in April and there was a
> >> suggestion for implementing a 5245-style hack where the answerer
> >> basically resends the 180 until it knows that it has been received.
> >>
> >> 5245 uses connectivity checks for this (i.e. it stops 180
> >> retransmissions when the first connectivity check is received) but
> we
> >> don't have that option here since the 180 may contain either none or
> >> only host candidates so there are strong chances that no binding
> >> request
> >> would be received on them.
> >>
> >> Thomas also suggested a second option which would be to also use
> INFO
> >> requests with trickled candidates as an indication that 180 was
> >> received. This however wouldn't work with half trickle so we are
> >> basically back to vanilla ICE for all (non-re) INVITEs.
> >>
> >> Another option would be to mandate an INFO request with
> >> "end-of-candidates" in response to the 180, but that would be just
> the
> >> same as mandating PRACK support.
> >>
> >> Thomas also suggested that the answerer can start sending INFOs
> right
> >> after it sends its answer in the 180 and then it can just resend the
> >> 180
> >> if the INFOs result in a 481 response.
> >>
> >> Personally I think this could potentially be made to work, but it
> would
> >> imply a level of complexity that considerably exceeds PRACK support.
> > [TS] I'm not convinced about the different level of complexity.
> > If you use PRACK you have SIP timer T1 running and would re-transmit
> the 180 in case that the PRACK is not received.
>=20
> Right.
>=20
> > Assuming that you want to wait with sending of the INFO until you saw
> the PRACK, would lead to delay in that case.
>=20
> True. Is this the only problem that we are trying to optimize? Save
> half
> an RTT? (This is a really a question. I am not trying to make a point.
> I
> am simply trying to understand why PRACK is an issue.)
[TS] Saving the half RTT is not my primary goal. When implementing ICE we g=
ot away without requiring PRACK.=20
I want to keep this property, since it allows=20
>=20
> > If you don't wait you would also have to treat a 481 response to the
> INFO.
>=20
> I am always a bit nervous when it comes to interpreting error
> responses.
[TS] We are actually specifying new behavior here.=20
Thus, we can specify the rule for the new info package and its application =
for trickle-ICE in way as exact as possible such that there would be no nee=
d for interpretation or nervousness.

> We are indeed likely to get a 481 when sending INFOs and the 180 hasn't
> yet made it to the remote side. However 481 might also imply that our
> dialog is simply dead for some reason and in this case 3261 says:
>=20
>=20
>    If the response for a request within a dialog is a 481
>    (Call/Transaction Does Not Exist) or a 408 (Request Timeout), the
> UAC
>    SHOULD terminate the dialog.
[TS] We are in the dialog establishing phase here and when the 180 was lost=
 we don't even have an early dialog.
So this rule does not apply. The 481 to the INFO would rather be the indica=
tor for this situation. As stated above we can prescribe that the 481 (and =
only the 481) is to be sent for the INFO and what happens next, i.e. retran=
smit the 180.
>=20
> > ... and you would detect quicker that the 180 was lost than by
> waiting for the PRACK.
>=20
> As said above: a lost 180 is not the only possible cause for a 408 in
> this case.
[TS] I assume you mean 481.
>=20
> > BTW: 5245 recommends to use PRACK. However it does not mandate it,
> but gives the alternative procedures as an option.
> > You could do the same here.
>=20
> Of course. I should have made that clearer: all alternative approaches
> were meant as something we could have in addition to the PRACK one.
>=20
> Also, keep in mind that the 180 carries an entire SDP answer. If INFOs
> are competing with 180s then they may potentially need to carry that
> too
> (or at least the ICE parts) and it would be good if we don't have to
> open that can of worms.
[TS] For the trickle-ICE application of the info-package I would just recom=
mend sending (all) the ICE candidates.
This eliminates the competition, you just take what you got last.
>=20
> >> Opinions?
> >>
> >> Q2: How do we send INFOs? Are they blocking or do we just send them
> in
> >> parallel? If the latter, then what happens when an INFO fails
> because
> >> it
> >> is received out of order? Do we just tell the application to resend
> the
> >> candidates asap?
> > [TS] I would go with the proposal below and send cumulatively. You
> also have the CSeq: header field to detect late arrivals.
> >
> >>
> >> This also leads to the following question:
> >>
> >> Q3: What exactly do we send in INFOs? Just the latest batch of
> freshly
> >> learned candidates or all candidates we've learned so far? Dale
> >> suggested that if we do this cumulatively we wouldn't need to worry
> >> about the case with the out-of-order INFOs from Q2 since the
> >> information
> >> gets resent anyway. A drawback here would obviously be that this
> adds
> >> more complexity for trickle ICE users (WebRTC applications
> >> specifically)
> > [TS] Again I'm not convinced about the complexity. Both peer know
> their candidates, what is the issue in putting all of them into the
> INFO?
>=20
> I am not sure how current WebRTC stacks handle provisioning of
> repeating
> candidates.=20
[TS] As Trickle-ICE is just being specified, we are in a position to specif=
y this handling.
>If they don't then it would be up to the receiving JS app,
> to sort our which are the new candidates and which were already there.
>=20
> The sending side would be similarly tricky for a JS app: it would need
> to construct SDP so that the a=3Dcandidate attributes appear under the
> right "a=3Dmid" line.
[TS] As the ICE/STUN logic resides in the browser not in the JS app, this w=
ould be a matter of the API definition to always provide all known candidat=
es towards the JS APP for sending. On the receiving side the JS APP would a=
lso just have to push all received candidates through the browser API.

Regards
Thomas
>=20
> >> A third option would be to allow both and leave it to the
> application.
> > [TS] This now adds complexity. ;-)
>=20
> Likely yes.
>=20
> Emil
>=20
> > I would just always send all candidates.
>=20
>=20
>=20
>=20
> > Regards
> > Thomas
> >>
> >> Comments are most welcome!
> >>
> >> Emil
> >>
> >> --
> >> https://jitsi.org
> >> _______________________________________________
> >> mmusic mailing list
> >> mmusic@ietf.org
> >> https://www.ietf.org/mailman/listinfo/mmusic
> >
>=20
> --
> https://jitsi.org

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From: "Stach, Thomas" <thomas.stach@siemens-enterprise.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Paul,=20

inline

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: Friday, 05 July, 2013 18:47
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
>=20
> On 7/5/13 10:12 AM, Emil Ivov wrote:
> > Hey Thomas
> >
> > On 04.07.13, 15:15, Stach, Thomas wrote:
> >> Emil,
> >> inline
> >>
> >>> -----Original Message-----
> >>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> >>> Behalf Of Emil Ivov
> >>> Sent: Wednesday, 03 July, 2013 16:13
> >>> To: MMUSIC IETF WG
> >>> Subject: [MMUSIC] Trickle ICE for SIP Questions
> >>>
> >>> Hey all,
> >>>
> >>> Christer, Enrico and I are preparing the next version of Trickle
> ICE
> >>> for
> >>> SIP. Now that discussions on BUNDLE and the plans seem to be
> winding
> >>> down, we wanted to run a few questions by the working group.
> >>>
> >>> Q1: Making reliable provisional responses and PRACK mandatory.
>=20
> Yes.
>=20
> >>> Obviously
> >>> this would be nice to avoid, so the question is: is there a
> reasonable
> >>> mechanism to achieve this (and by reasonable, we mean something
> that
> >>> wouldn't be harder than implementing support for PRACK).
> >>>
> >>> There was some discussion about this back in April and there was a
> >>> suggestion for implementing a 5245-style hack where the answerer
> >>> basically resends the 180 until it knows that it has been received.
> >>>
> >>> 5245 uses connectivity checks for this (i.e. it stops 180
> >>> retransmissions when the first connectivity check is received) but
> we
> >>> don't have that option here since the 180 may contain either none
> or
> >>> only host candidates so there are strong chances that no binding
> >>> request
> >>> would be received on them.
> >>>
> >>> Thomas also suggested a second option which would be to also use
> INFO
> >>> requests with trickled candidates as an indication that 180 was
> >>> received. This however wouldn't work with half trickle so we are
> >>> basically back to vanilla ICE for all (non-re) INVITEs.
> >>>
> >>> Another option would be to mandate an INFO request with
> >>> "end-of-candidates" in response to the 180, but that would be just
> the
> >>> same as mandating PRACK support.
> >>>
> >>> Thomas also suggested that the answerer can start sending INFOs
> right
> >>> after it sends its answer in the 180 and then it can just resend
> the
> >>> 180
> >>> if the INFOs result in a 481 response.
> >>>
> >>> Personally I think this could potentially be made to work, but it
> would
> >>> imply a level of complexity that considerably exceeds PRACK
> support.
> >> [TS] I'm not convinced about the different level of complexity.
> >> If you use PRACK you have SIP timer T1 running and would re-transmit
> >> the 180 in case that the PRACK is not received.
> >
> > Right.
> >
> >> Assuming that you want to wait with sending of the INFO until you
> saw
> >> the PRACK, would lead to delay in that case.
> >
> > True. Is this the only problem that we are trying to optimize? Save
> half
> > an RTT? (This is a really a question. I am not trying to make a
> point. I
> > am simply trying to understand why PRACK is an issue.)
>=20
> Yes, why is it such an issue to require PRACK?
[TS] I see potential for easier GW/SBC interworking with vanilla-ICE UAs, t=
hat did not implement PRACK
> It is a good thing to implement for many reasons.=20
[TS] There are also good reasons against PRACK. e.g. the much more complica=
ted state machine.
So in conjunction
> with
> defining other new functionality, why not require it?
[TS] We did not need to require PRACK for vanilla ICE, why require it now. =
There is a proposal on the table that solves the problem without the comple=
xity of PRACK.=20
As already said I don't object to recommending PRACK (as in 5245), I just d=
on't want it mandatory.=20

>=20
> >> If you don't wait you would also have to treat a 481 response to the
> >> INFO.
> >
> > I am always a bit nervous when it comes to interpreting error
> responses.
> > We are indeed likely to get a 481 when sending INFOs and the 180
> hasn't
> > yet made it to the remote side. However 481 might also imply that our
> > dialog is simply dead for some reason and in this case 3261 says:
> >
> >
> >    If the response for a request within a dialog is a 481
> >    (Call/Transaction Does Not Exist) or a 408 (Request Timeout), the
> UAC
> >    SHOULD terminate the dialog.
> >
> >> ... and you would detect quicker that the 180 was lost than by
> waiting
> >> for the PRACK.
>=20
> This just highlights how "delicate" things get if you start trying to
> work around the lack of PRACK. It might be possible to get something
> that seems to work, but its a house of cards.
[TS] Please refer to my reply to Emil, the rule above does not apply here.
>=20
> > As said above: a lost 180 is not the only possible cause for a 408 in
> > this case.
> >
> >> BTW: 5245 recommends to use PRACK. However it does not mandate it,
> but
> >> gives the alternative procedures as an option.
> >> You could do the same here.
> >
> > Of course. I should have made that clearer: all alternative
> approaches
> > were meant as something we could have in addition to the PRACK one.
> >
> > Also, keep in mind that the 180 carries an entire SDP answer. If
> INFOs
> > are competing with 180s then they may potentially need to carry that
> too
> > (or at least the ICE parts) and it would be good if we don't have to
> > open that can of worms.
>=20
> Yes. The trickle ICE INFOs are *relative* to what was set up in the
> O/A.
> In the absence of the answer the trickle ICE is problematic.
>=20
> >>> Opinions?
> >>>
> >>> Q2: How do we send INFOs? Are they blocking or do we just send them
> in
> >>> parallel? If the latter, then what happens when an INFO fails
> because
> >>> it
> >>> is received out of order? Do we just tell the application to resend
> the
> >>> candidates asap?
>=20
> *If* you are going to send in a non-blocking way then they should be
> cumulative.
>=20
> If you treat them as blocking, then they don't have to be, but it might
> still be better.
>=20
> >> [TS] I would go with the proposal below and send cumulatively. You
> >> also have the CSeq: header field to detect late arrivals.
>=20
> Yes, you can detect the late arrival. But you have at that point
> already
> acted on the one that should have been preceded by it.
[TS] What is the problem? The previous INFO had fewer candidates, you just =
got all additional candidates. =20
I can't see a case were a candidate got removed in a subsequent INFO.

>=20
> >>> This also leads to the following question:
> >>>
> >>> Q3: What exactly do we send in INFOs? Just the latest batch of
> freshly
> >>> learned candidates or all candidates we've learned so far? Dale
> >>> suggested that if we do this cumulatively we wouldn't need to worry
> >>> about the case with the out-of-order INFOs from Q2 since the
> >>> information
> >>> gets resent anyway.
>=20
> Exactly.
>=20
> >>> A drawback here would obviously be that this adds
> >>> more complexity for trickle ICE users (WebRTC applications
> >>> specifically)
> >> [TS] Again I'm not convinced about the complexity. Both peer know
> >> their candidates, what is the issue in putting all of them into the
> INFO?
> >
> > I am not sure how current WebRTC stacks handle provisioning of
> repeating
> > candidates. If they don't then it would be up to the receiving JS
> app,
> > to sort our which are the new candidates and which were already
> there.
> >
> > The sending side would be similarly tricky for a JS app: it would
> need
> > to construct SDP so that the a=3Dcandidate attributes appear under the
> > right "a=3Dmid" line.
>=20
> I don't know enough about ICE to comment specifically on this.
>=20
> But ISTM that this is analogous to what a SIP device must do to manage
> the target set of destinations for a request - it is required to keep
> track and not add the same address to the target set more than once.
>=20
> 	Thanks,
> 	Paul
>=20
> >>> A third option would be to allow both and leave it to the
> application.
> >> [TS] This now adds complexity. ;-)
> >
> > Likely yes.
> >
> > Emil
> >
> >> I would just always send all candidates.
> >
> >
> >
> >
> >> Regards
> >> Thomas
> >>>
> >>> Comments are most welcome!
> >>>
> >>> Emil
> >>>
> >>> --
> >>> https://jitsi.org
> >>> _______________________________________________
> >>> mmusic mailing list
> >>> mmusic@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/mmusic
> >>
> >
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From gunnar.hellstrom@omnitor.se  Mon Jul  8 03:09:30 2013
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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-00.txt - asymmetric preferences
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Randall,
It is good to see this shape up.

One initial comment:

The draft discourages in an infavourable way from specifying 
asymmetrical preferences.

In 7.1, it says:

"In an offer, the 'humintlang-send' values constitute a list in
    preference order (first is most preferred) of the languages the
    offerer wishes to send, and the 'humintlang-recv' values constitute a
    list in preference order of the languages the offerer wishes to
    receive.  Both SHOULD have the same values in the same order, because
    otherwise it is harder to match desired resources."


A typical example of a user who do not want to specify the same value in 
both directions is a hard-of-hearing user who prefers to speak English 
but might accept to type English text on the transmission side,  but 
prefers to read English text and do not have sufficient hearing to at 
all handle hearing spoken English on the receiving side.

In the audio media specification there would then be humintlang-send 
value English,  and nothing in the humintlang-recv.
While in the text media specification there would be humintlang-send 
value for English, that would need a very much lower preference than the 
audio send, and the humintlang-recv value for English.

Note also the need for an absolute preference indication. Or relative 
between media. This user will be quite disappointed to be forced to 
handle the call in text both ways.

With this background, the SHOULD in the cited section is not desirable. 
I suggest to delete that sentence.

There is a similar problem with the last paragraph in 7.1

"  When placing an emergency call, and in any other case where the
    language cannot be assumed from context, each media stream in an
    offer SHOULD specify both 'humintlang-send' and 'humintlang-recv'
    attributes (to avoid ambiguity), and both SHOULD be set to the same
    values in the same order."

Since users may have capability for only one direction of a modality, it is not reasonable to require them to
  specify a value in the direction they do not have a capability in.


So this paragraph should be deleted or modified.


Thanks

Gunnar










------------------------------------------------------------------------
Gunnar Hellström
Omnitor
gunnar.hellstrom@omnitor.se
+46708204288
On 2013-07-08 00:39, internet-drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>
>
> 	Title           : Negotiating Human Language Using SDP
> 	Author(s)       : Randall Gellens
> 	Filename        : draft-gellens-mmusic-negotiating-human-language-00.txt
> 	Pages           : 13
> 	Date            : 2013-07-07
>
> Abstract:
>     Users have various human (natural) language needs, abilities, and
>     preferences regarding spoken, written, and signed languages.  When
>     establishing interactive communication "calls" there needs to be a
>     way to communicate and ideally match (i.e., negotiate) the caller's
>     language preferences with the capabilities of the called party.  This
>     is especially important with emergency calls, where a call can be
>     routed to a Public Safety Answering Point (PSAP) or call taker
>     capable of communicating with the user, or a translator or relay
>     operator can be bridged into the call during setup, but this applies
>     to non-emergency calls as well (as an example, when calling a company
>     call center).
>
>     This document describes the need and expected use, and describes a
>     solution using new SDP stream attributes plus an optional SIP "hint."
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-gellens-mmusic-negotiating-human-language
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-gellens-mmusic-negotiating-human-language-00
>
>
> 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


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    <div class="moz-cite-prefix">Randall,<br>
      It is good to see this shape up.<br>
      <br>
      One initial comment:<br>
      <br>
      The draft discourages in an infavourable way from specifying
      asymmetrical preferences.<br>
      <br>
      In 7.1, it says:<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: auto; text-align: start; text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">"In an offer, the 'humintlang-send' values constitute a list in
   preference order (first is most preferred) of the languages the
   offerer wishes to send, and the 'humintlang-recv' values constitute a
   list in preference order of the languages the offerer wishes to
   receive.  Both SHOULD have the same values in the same order, because
   otherwise it is harder to match desired resources."</pre>
      <br>
      A typical example of a user who do not want to specify the same
      value in both directions is a hard-of-hearing user who prefers to
      speak English but might accept to type English text on the
      transmission side,&nbsp; but prefers to read English text and do not
      have sufficient hearing to at all handle hearing spoken English on
      the receiving side.<br>
      <br>
      In the audio media specification there would then be
      humintlang-send value English,&nbsp; and nothing in the
      humintlang-recv. <br>
      While in the text media specification there would be
      humintlang-send value for English, that would need a very much
      lower preference than the audio send, and the humintlang-recv
      value for English.<br>
      <br>
      Note also the need for an absolute preference indication. Or
      relative between media. This user will be quite disappointed to be
      forced to handle the call in text both ways.<br>
      <br>
      With this background, the SHOULD in the cited section is not
      desirable. I suggest to delete that sentence.<br>
      <br>
      There is a similar problem with the last paragraph in 7.1<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: auto; text-align: start; text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">"  When placing an emergency call, and in any other case where the
   language cannot be assumed from context, each media stream in an
   offer SHOULD specify both 'humintlang-send' and 'humintlang-recv'
   attributes (to avoid ambiguity), and both SHOULD be set to the same
   values in the same order."

Since users may have capability for only one direction of a modality, it is not reasonable to require them to
&nbsp;specify a value in the direction they do not have a capability in.
</pre>
      <br>
      So this paragraph should be deleted or modified.<br>
      <br>
      <br>
      Thanks<br>
      <br>
      Gunnar<br>
      <br>
      <br>
      <br>
      <br>
      <br>
      <br>
      <br>
      <br>
      <br>
      <br>
      <div class="moz-signature">
        <hr>
        Gunnar Hellstr&ouml;m<br>
        Omnitor<br>
        <a class="moz-txt-link-abbreviated" href="mailto:gunnar.hellstrom@omnitor.se">gunnar.hellstrom@omnitor.se</a><br>
        +46708204288<br>
      </div>
      On 2013-07-08 00:39, <a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a> wrote:<br>
    </div>
    <blockquote
      cite="mid:20130707223940.32193.45404.idtracker@ietfa.amsl.com"
      type="cite">
      <pre wrap="">
A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title           : Negotiating Human Language Using SDP
	Author(s)       : Randall Gellens
	Filename        : draft-gellens-mmusic-negotiating-human-language-00.txt
	Pages           : 13
	Date            : 2013-07-07

Abstract:
   Users have various human (natural) language needs, abilities, and
   preferences regarding spoken, written, and signed languages.  When
   establishing interactive communication "calls" there needs to be a
   way to communicate and ideally match (i.e., negotiate) the caller's
   language preferences with the capabilities of the called party.  This
   is especially important with emergency calls, where a call can be
   routed to a Public Safety Answering Point (PSAP) or call taker
   capable of communicating with the user, or a translator or relay
   operator can be bridged into the call during setup, but this applies
   to non-emergency calls as well (as an example, when calling a company
   call center).

   This document describes the need and expected use, and describes a
   solution using new SDP stream attributes plus an optional SIP "hint."


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

There's also a htmlized version available at:
<a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-gellens-mmusic-negotiating-human-language-00">http://tools.ietf.org/html/draft-gellens-mmusic-negotiating-human-language-00</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>

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From gunnar.hellstrom@omnitor.se  Mon Jul  8 05:08:48 2013
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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-00.txt - two modalities in video
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There is a good description of avoidance of using subtags in 7.1, third 
paragraph from the end.

It may be good to inform after that paragraph about the language tags 
that can be used for video.
There are separate language tags for sign languages.
So, for the video medium, that can be used to distinguish between two 
different usages of video; for sign language versus for assisting 
hearing with the view of the speaking human for lip-reading or for just 
emphasizing the understanding by a view of the speaker.

Insert a new paragraph to be the third from the end of 7.1:

For indication of preferences and capabilities of sign languages, the 
specific sign language tags of RFC 5646 SHALL be used in the video media 
specifications.
Preference and capability of a spoken language tag (i.e. not a sign 
language) indicated in the video media specification SHOULD be used for 
indicating a view of the speaking human that may be needed or preferred 
for speech-reading assisted hearing.

Regards

Gunnar
------------------------------------------------------------------------
Gunnar Hellström
Omnitor
gunnar.hellstrom@omnitor.se
+46708204288

>


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    <div class="moz-cite-prefix">There is a good description of
      avoidance of using subtags in 7.1, third paragraph from the end. <br>
      <br>
      It may be good to inform after that paragraph about the language
      tags that can be used for video. <br>
      There are separate language tags for sign languages.<br>
      So, for the video medium, that can be used to distinguish between
      two different usages of video; for sign language versus for
      assisting hearing with the view of the speaking human for
      lip-reading or for just emphasizing the understanding by a view of
      the speaker.<br>
      <br>
      Insert a new paragraph to be the third from the end of 7.1:<br>
      <br>
      For indication of preferences and capabilities of sign languages,
      the specific sign language tags of RFC 5646 SHALL be used in the
      video media specifications.<br>
      Preference and capability of a spoken language tag (i.e. not a
      sign language) indicated in the video media specification SHOULD
      be used for indicating a view of the speaking human that may be
      needed or preferred for speech-reading assisted hearing. <br>
      <br>
      Regards<br>
      <br>
      Gunnar<br>
      <div class="moz-signature">
        <hr>
        Gunnar Hellstr&ouml;m<br>
        Omnitor<br>
        <a class="moz-txt-link-abbreviated" href="mailto:gunnar.hellstrom@omnitor.se">gunnar.hellstrom@omnitor.se</a><br>
        +46708204288<br>
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From pkyzivat@alum.mit.edu  Mon Jul  8 08:53:45 2013
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Thomas,

On 7/8/13 5:24 AM, Stach, Thomas wrote:

> [TS] Saving the half RTT is not my primary goal. When implementing ICE we got away without requiring PRACK.
> I want to keep this property, since it allows

I get it that you think PRACK is hard to implement.
But hard compared to *what*? I suspect that this will introduce 
complexity elsewhere.

>>     If the response for a request within a dialog is a 481
>>     (Call/Transaction Does Not Exist) or a 408 (Request Timeout), the
>> UAC
>>     SHOULD terminate the dialog.
> [TS] We are in the dialog establishing phase here and when the 180 was lost we don't even have an early dialog.
> So this rule does not apply. The 481 to the INFO would rather be the indicator for this situation. As stated above we can prescribe that the 481 (and only the 481) is to be sent for the INFO and what happens next, i.e. retransmit the 180.

This is part of the house of cards that I was talking about.
You seem to be saying that a special rule should be be adopted, allowing 
INFO to be sent from INVITE-UAS before it is known that the dialog has 
been established. And then a special rule for processing 481 responses 
to INFO in this context.

>> Also, keep in mind that the 180 carries an entire SDP answer. If INFOs
>> are competing with 180s then they may potentially need to carry that
>> too
>> (or at least the ICE parts) and it would be good if we don't have to
>> open that can of worms.
> [TS] For the trickle-ICE application of the info-package I would just recommend sending (all) the ICE candidates.
> This eliminates the competition, you just take what you got last.

Certainly it will be simpler if all candidates are sent each time.
But I don't think that means you can just take what you got last. You 
still need to check if what you got is *older* than what you already 
have, and if so ignore it.

	Thanks,
	Paul

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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Just responding to things that haven't already been covered.

On 7/8/13 5:44 AM, Stach, Thomas wrote:

>> Yes, why is it such an issue to require PRACK?
> [TS] I see potential for easier GW/SBC interworking with vanilla-ICE UAs, that did not implement PRACK

I don't get it.

If you are talking about SBCs that terminate/relay media, then the leg 
doing trickle-ICE is separate from the leg doing vanilla-ICE. So what is 
the issue?

And if you have a signaling-only SBC, and one leg supports trickle-ICE 
while the other side supports only vanilla-ICE, then just use 
vanilla-ICE e2e. You won't gain anything by an intermediary trying to 
map trickle-ICE to vanilla-ICE.

>> It is a good thing to implement for many reasons.
> [TS] There are also good reasons against PRACK. e.g. the much more complicated state machine.

Yes, but once you have done it then it is there to use for all sorts of 
things. The alternative seems to be to instead introduce a lot of new 
complexity that is *only* applicable to trickle-ICE.

	Thanks,
	Paul

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Subject: [MMUSIC] Milestones changed for mmusic WG
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Changed milestone "Submit SDP extensions for Media Titles and Media
Bandwidth Capability Negotiations as Proposed Standard", set due date
to December 2012 from December 2012.

Changed milestone "Submit SDP extensions for audio media streams over
Circuit- Switched bearers as Proposed Standard", set due date to
December 2012 from December 2012.

Changed milestone "Submit revised RTSP spec for Proposed Standard",
set due date to December 2012 from December 2012.

Changed milestone "Submit SDP extensions to classify traffic as
Proposed Standard", set due date to February 2013 from February 2013.

Changed milestone "Submit SDP extensions for duplicated streams as
Proposed Standard", set due date to February 2013 from February 2013.

Changed milestone "Submit SDP considerations for G.723 Annex A and
G.729 Annex B as Proposed Standard", set due date to February 2013
from February 2013.

Changed milestone "Submit Considerations for using SDP offer/answer
with middleboxes for Informational", set due date to March 2013 from
March 2013.

Changed milestone "Submit revised SDP specification to IETF as
Proposed Standard", set due date to March 2013 from March 2013.

Changed milestone "Submit SDP extension for cross session stream
identification as Proposed Standard", set due date to March 2013 from
March 2013.

Changed milestone "Submit SCTP-Based Media Transport in SDP as
Proposed Standard.", set due date to April 2013 from April 2013.

Changed milestone "Submit a document describing current practices in
hosted NAT traversal as Informational", set due date to April 2013
from April 2013.

Changed milestone "Submit SDP extensions to negotiate multiplexed
media streams as Proposed Standard", set due date to June 2013 from
June 2013.

Changed milestone "Submit RTSP NAT considerations draft as a Proposed
Standard", set due date to July 2013 from January 2013.

Changed milestone "Submit a revision of ICE (RFC 5245) as Proposed
Standard", set due date to July 2013 from July 2013.

URL: http://datatracker.ietf.org/wg/mmusic/charter/

From ietf-secretariat-reply@ietf.org  Mon Jul  8 12:54:55 2013
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Subject: [MMUSIC] Milestones changed for mmusic WG
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Changed milestone "Submit SDP extensions for duplicated streams as
Proposed Standard", set due date to July 2013 from February 2013.

Changed milestone "Submit SDP considerations for G.723 Annex A and
G.729 Annex B as Proposed Standard", set due date to July 2013 from
February 2013.

Changed milestone "Submit a document describing current practices in
hosted NAT traversal as Informational", set due date to July 2013 from
April 2013.

Changed milestone "Submit SDP extensions to classify traffic as
Proposed Standard", set due date to September 2013 from February 2013.

Changed milestone "Submit SDP extensions to negotiate multiplexed
media streams as Proposed Standard", set due date to September 2013
from June 2013.

Changed milestone "Submit Considerations for using SDP offer/answer
with middleboxes for Informational", set due date to October 2013 from
March 2013.

Changed milestone "Submit revised SDP specification to IETF as
Proposed Standard", set due date to November 2013 from March 2013.

Changed milestone "Submit SCTP-Based Media Transport in SDP as
Proposed Standard.", set due date to November 2013 from April 2013.

Changed milestone "Submit SDP extension for cross session stream
identification as Proposed Standard", set due date to December 2013
from March 2013.

URL: http://datatracker.ietf.org/wg/mmusic/charter/

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Deleted milestone "Submit SDP extensions for Media Titles and Media
Bandwidth Capability Negotiations as Proposed Standard".

Deleted milestone "Submit SDP extensions for audio media streams over
Circuit- Switched bearers as Proposed Standard".

Deleted milestone "Submit revised RTSP spec for Proposed Standard".

Deleted milestone "Submit a revision of ICE (RFC 5245) as Proposed
Standard".

Added milestone "Submit Trickle ICE as Proposed Standard", due
December 2013.

Added milestone "Submit a revision of ICE (RFC 5245) as Proposed
Standard", due January 2014.

Added milestone "Submit SIP usage for Trickle ICE as Proposed
Standard", due March 2014.

URL: http://datatracker.ietf.org/wg/mmusic/charter/

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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-00.txt - asymmetric preferences
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On 2013-07-09 03:51, Randall Gellens wrote:
> Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-langua
> Hi Gunnar,
>
> Thanks for the quick review and comments.
>
>
> At 12:08 PM +0200 7/8/13, Gunnar Hellstrom wrote:
>
>> In 7.1, it says:
>> "In an offer, the 'humintlang-send' values constitute a list in
>>    preference order (first is most preferred) of the languages the
>>    offerer wishes to send, and the 'humintlang-recv' values constitute a
>>    list in preference order of the languages the offerer wishes to
>>    receive.  Both SHOULD have the same values in the same order, because
>>    otherwise it is harder to match desired resources."
>>
>> A typical example of a user who do not want to specify the same value 
>> in both directions is a hard-of-hearing user who prefers to speak 
>> English but might accept to type English text on the transmission 
>> side,  but prefers to read English text and do not have sufficient 
>> hearing to at all handle hearing spoken English on the receiving side.
>
> This is very similar to the use case included of a user who can hear 
> English but not speak; an audio and text stream are set up, both using 
> English.  The point of negotiating language is to get the call to a 
> callee who can communicate with the caller (e.g., a PSAP and call 
> taker who understand at least one language in common with the 
> caller).  The point is not to fully specify exactly how the media will 
> be used; there is no need to do so and it won't help get the call to 
> the right callee, but will make things much more complex.
>
> It is simpler and more robust to handle cases such as you describe and 
> the one in the document now by negotiating the needed media as two-way 
> streams.
Yes, two-way streams should usually be setup, on the media level. A 
media direction without a declared preference is usually valuable 
anyway. But that is not what the language specification aims at. The 
goal is to match caller, callee and possibly required assisting services 
to fill gaps between in preferences and capabilites between caller and 
callee.

I said that the proper declaration for this example would be:
In the audio media specification there would be humintlang-send value 
English,  and nothing in the humintlang-recv.
While in the text media specification there would be humintlang-send 
value for English, (that would need a very much lower preference than 
the audio send), and the humintlang-recv value for English.
>
>
You say that the declaration should instead be symmetrical:
In the audio media specification there would be humintlang-send value 
English,  and English in the humintlang-recv.

This is falsely declaring a preference, and cannot contribute to more 
successful matching.

If this person calls another person with the same characteristics, and 
the same declaration, a comparison of their preferences would result in 
connecting them together without assisting service, because the language 
specifications indicate that they happily both talk and hear.
Resulting in that they are left there without an opportunity to talk to 
each other.

Instead keeping the declaration to the truth and not have a value for 
audio humintlang-recv, makes it possible to evaluate the need, and 
provide a two-way captioned telephony service for the call.

Another point for truthful declaration is that the parties can get 
indications on how to behave in the call. If a call-center get a call 
from the deaf person in the example above with spoken English 
declaration both ways, and English text specification both ways, it is 
natural for the call center to try speaking a welcome phrase. There is 
no indication that typing will be needed. The calling person has no 
hearing capability and need therefore to explain with voice  "I am deaf, 
please type". The call starts with a disappointing confusion instead of 
a nice welcome in the appropriate language and modality that would be 
achievable if the declaration shows the real preferences and capabilities.

So, therefore I suggest to delete the sentences requiring symmetric 
humintlang declarations.

Thanks
Gunnar Hellström


>
> -- 
> Randall Gellens


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    <div class="moz-cite-prefix">On 2013-07-09 03:51, Randall Gellens
      wrote:<br>
    </div>
    <blockquote cite="mid:p0624060ece011b92fdac@%5B99.111.97.136%5D"
      type="cite">
      <style type="text/css"><!--
blockquote, dl, ul, ol, li { padding-top: 0 ; padding-bottom: 0 }
 --></style>
      <title>Re: [MMUSIC]
        draft-gellens-mmusic-negotiating-human-langua</title>
      <div>Hi Gunnar,</div>
      <div><br>
      </div>
      <div>Thanks for the quick review and comments.</div>
      <div><br>
      </div>
      <div><br>
      </div>
      <div>At 12:08 PM +0200 7/8/13, Gunnar Hellstrom wrote:</div>
      <div><br>
      </div>
      <blockquote type="cite" cite="">In 7.1, it says:<br>
      </blockquote>
      <blockquote type="cite" cite=""><tt>"In an offer, the
          'humintlang-send' values constitute a list in<br>
          &nbsp;&nbsp; preference order (first is most preferred) of the
          languages the<br>
          &nbsp;&nbsp; offerer wishes to send, and the 'humintlang-recv' values
          constitute a<br>
          &nbsp;&nbsp; list in preference order of the languages the offerer
          wishes to<br>
          &nbsp;&nbsp; receive.&nbsp; Both SHOULD have the same values in the
          same order, because<br>
          &nbsp;&nbsp; otherwise it is harder to match desired
          resources."</tt></blockquote>
      <blockquote type="cite" cite=""><br>
      </blockquote>
      <blockquote type="cite" cite="">A typical example of a user who do
        not
        want to specify the same value in both directions is a
        hard-of-hearing
        user who prefers to speak English but might accept to type
        English
        text on the transmission side,&nbsp; but prefers to read English text
        and do not have sufficient hearing to at all handle hearing
        spoken
        English on the receiving side.</blockquote>
      <div><br>
      </div>
      <div>This is very similar to the use case included of a user who
        can
        hear English but not speak; an audio and text stream are set up,
        both
        using English.&nbsp; The point of negotiating language is to get the
        call to a callee who can communicate with the caller (e.g., a
        PSAP and
        call taker who understand at least one language in common with
        the
        caller).&nbsp; The point is not to fully specify exactly how the
        media
        will be used; there is no need to do so and it won't help get
        the call
        to the right callee, but will make things much more complex.</div>
      <div><br>
      </div>
      <div>It is simpler and more robust to handle cases such as you
        describe and the one in the document now by negotiating the
        needed
        media as two-way streams.</div>
    </blockquote>
    Yes, two-way streams should usually be setup, on the media level. A
    media direction without a declared preference is usually valuable
    anyway. But that is not what the language specification aims at. The
    goal is to match caller, callee and possibly required assisting
    services to fill gaps between in preferences and capabilites between
    caller and callee.<br>
    <br>
    I said that the proper declaration for this example would be:<br>
    In the audio media specification there would be humintlang-send
    value English,&nbsp; and nothing in the humintlang-recv. <br>
    While in the text media specification there would be humintlang-send
    value for English, (that would need a very much lower preference
    than the audio send), and the humintlang-recv value for English.<br>
    <blockquote cite="mid:p0624060ece011b92fdac@%5B99.111.97.136%5D"
      type="cite">
      <div><br>
      </div>
      <div><br>
      </div>
    </blockquote>
    You say that the declaration should instead be symmetrical:<br>
    In the audio media specification there would be humintlang-send
    value English,&nbsp; and English in the humintlang-recv. <br>
    <br>
    This is falsely declaring a preference, and cannot contribute to
    more successful matching.<br>
    <br>
    If this person calls another person with the same characteristics,
    and the same declaration, a comparison of their preferences would
    result in connecting them together without assisting service,
    because the language specifications indicate that they happily both
    talk and hear. <br>
    Resulting in that they are left there without an opportunity to talk
    to each other. <br>
    <br>
    Instead keeping the declaration to the truth and not have a value
    for audio humintlang-recv, makes it possible to evaluate the need,
    and provide a two-way captioned telephony service for the call. <br>
    <br>
    Another point for truthful declaration is that the parties can get
    indications on how to behave in the call. If a call-center get a
    call from the deaf person in the example above with spoken English
    declaration both ways, and English text specification both ways, it
    is natural for the call center to try speaking a welcome phrase.
    There is no indication that typing will be needed. The calling
    person has no hearing capability and need therefore to explain with
    voice&nbsp; "I am deaf, please type". The call starts with a
    disappointing confusion instead of a nice welcome in the appropriate
    language and modality that would be achievable if the declaration
    shows the real preferences and capabilities.<br>
    <br>
    So, therefore I suggest to delete the sentences requiring symmetric
    humintlang declarations.<br>
    <br>
    Thanks<br>
    Gunnar Hellstr&ouml;m<br>
    <br>
    &nbsp;<br>
    <blockquote cite="mid:p0624060ece011b92fdac@%5B99.111.97.136%5D"
      type="cite">
      <div><br>
      </div>
      <x-sigsep>
        <pre>-- 
</pre>
      </x-sigsep>
      <div>Randall Gellens<br>
      </div>
    </blockquote>
    <br>
  </body>
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=20
=0A=
      This is very similar to the use case included of a user who=0A=
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=2C=0A=
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=0A=
      =0A=
      It is simpler and more robust to handle cases such as you=0A=
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        needed=0A=
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    =0A=
    [BA]  I think this example points out the limitations of a call routing=
 approach that relies solely on language. In this case=2C it is important f=
or the "hint" to cause the call to be routed to an endpoint that can handle=
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 Since both the audio and text media are set up using English in both direc=
tions=2C one might conclude that the "hint" need only express the language =
preference (English)=2C but that would not be sufficient to get a satisfact=
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<body class=3D'hmmessage'><div dir=3D'ltr'>&nbsp=3B<BR><blockquote cite=3D"=
mid:p0624060ece011b92fdac@%5B99.111.97.136%5D">=0A=
      <div>This is very similar to the use case included of a user who=0A=
        can=0A=
        hear English but not speak=3B an audio and text stream are set up=
=2C=0A=
        both=0A=
        using English.&nbsp=3B The point of negotiating language is to get =
the=0A=
        call to a callee who can communicate with the caller (e.g.=2C a=0A=
        PSAP and=0A=
        call taker who understand at least one language in common with=0A=
        the=0A=
        caller).&nbsp=3B The point is not to fully specify exactly how the=
=0A=
        media=0A=
        will be used=3B there is no need to do so and it won't help get=0A=
        the call=0A=
        to the right callee=2C but will make things much more complex.<br>=
=0A=
&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B </div>=0A=
      <div>It is simpler and more robust to handle cases such as you=0A=
        describe and the one in the document now by negotiating the=0A=
        needed=0A=
        media as two-way streams.</div>=0A=
    </blockquote><div>=0A=
    </div><div>[BA]&nbsp=3B I think this example points out the limitations=
 of a call routing approach that relies solely on language. In this case=2C=
 it is important for the "hint" to cause the call to be routed to an endpoi=
nt that can handle text media=2C since otherwise the caller will not be abl=
e to communicate.&nbsp=3B Since both the audio and text media are set up us=
ing English in both directions=2C&nbsp=3Bone might conclude that the&nbsp=
=3B"hint"&nbsp=3Bneed only express the language preference (English)=2C but=
 that would not be sufficient to get a satisfactory result.&nbsp=3B What ne=
eds to be expressed in the "hint" is *both* a preference for English and a =
requirement for text media support. </div><div>&nbsp=3B</div><div>=0A=
</div> 		 	   		  </div></body>
</html>=

--_a29b1acf-3497-4503-b87f-927eca5f359b_--

From Christian.Groves@nteczone.com  Tue Jul  9 00:13:29 2013
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Cc: "clue@ietf.org" <clue@ietf.org>, mmusic@ietf.org
Subject: Re: [MMUSIC] Notes from Orlando human language draft discussion
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Hello Randall,

I had a look at the draft with respect to CLUE. The main difference 
between the CLUE usage of language and what is being proposed is the 
ability for the sender to specify sending AND receiving preferences. The 
current CLUE model is that an each endpoint Advertises what it can do 
and then each consumer chooses from this list. This could still result 
in an asymmetric usage of language (just negotiated differently).

You may need to consider the case where there are multiple streams of 
the same media type with different languages. e.g. an audio conference 
where one stream is the main audio of the conference and there's a 
second stream with a translation/s. This is the type of thing CLUE is 
considering. I'm not sure how this fits in with your work?

Some other comments:

With regards to the SIP "hint" it wasn't clear to me how this matches 
the languages in "humintlang-send" and "humintlang-recv". Is it planned 
that both will be supported in the header?

The "TBD" paragraph in 7.3 SIP "hint" mentions (where the caller sets 
media and language) but in the final paragraph of the 7.3 it sounds like 
the "hint" is only for audio?

7.4 Silly states - How do the language sub-tags come into play here? For 
example: the endpoint might specify a language for an audio stream which 
may be valid. If the endpoint also specified a sub-tag with "script" I 
guess it makes it a silly state?

Regards, Christian

On 8/07/2013 8:40 AM, Randall Gellens wrote:
> At 4:50 PM +1100 3/25/13, Christian Groves wrote:
>
>>  Hello,
>>
>>  One thing that doesn't seem to be noted is an overlap with the work 
>> of the CLUE WG. There has been some discussion in the CLUE working 
>> group about adding attributes to captures to describe the language, 
>> whether sub-titles are used, whether a stream comes from an 
>> interpreter, etc. Where multiple streams are used I would expect any 
>> work in this area should be aligned.
>>
>>  Regards, Christian
>
> Hi Christian,
>
> Thanks for the info.  Would it make sense to talk about this in 
> Berlin?  I agree that if there is overlap in the approaches we should 
> align them.
>
> Also, I have an updated draft that was just uploaded.  It is now named 
> 'draft-gellens-mmusic-negotiating-human-language-00' (adding -mmusic 
> and hence reverting back to -00).
>


From gunnar.hellstrom@omnitor.se  Tue Jul  9 00:58:39 2013
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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-00.txt - preferences between media
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On 2013-07-09 08:56, Bernard Aboba wrote:
>
>     This is very similar to the use case included of a user who can
>     hear English but not speak; an audio and text stream are set up,
>     both using English.  The point of negotiating language is to get
>     the call to a callee who can communicate with the caller (e.g., a
>     PSAP and call taker who understand at least one language in common
>     with the caller).  The point is not to fully specify exactly how
>     the media will be used; there is no need to do so and it won't
>     help get the call to the right callee, but will make things much
>     more complex.
>     It is simpler and more robust to handle cases such as you describe
>     and the one in the document now by negotiating the needed media as
>     two-way streams.
>
> [BA]  I think this example points out the limitations of a call 
> routing approach that relies solely on language. In this case, it is 
> important for the "hint" to cause the call to be routed to an endpoint 
> that can handle text media, since otherwise the caller will not be 
> able to communicate.  Since both the audio and text media are set up 
> using English in both directions, one might conclude that 
> the "hint" need only express the language preference (English), but 
> that would not be sufficient to get a satisfactory result.  What needs 
> to be expressed in the "hint" is *both* a preference for English and a 
> requirement for text media support.
<GH>And the humintlang specification allows that if the requirement to 
have symmetric specifications per media is deleted ( as described in 
another mail ).

But there is also a need for absolute preference. Or relative order of 
preference between media.

It is easy to use a sign language example fort that situation.

A person prefers to use British sign language in both directions but is 
capable with some reluctance to use English text. That is natural. Text 
is always slower and less expressive than sign language or spoken language.

A call goes to a call center that declared spoken English and English 
text in both directions.
The result can be a direct connection using English text, or invocation 
of a sign relay service translating between British sign language and 
spoken English.

An ability to declare that the British sign language is preferred over 
the English text would result in the invocation of the sign relay 
service and make both sides happy with a call with good flow.

Without the relative preference between media, the call will likely 
result in the direct text communication that will work but with less 
satisfaction.

This can be solved either by an absolute preference indication  : 
British sing language *preferred*  , English text *managed*
Or by some order of preference between the media;  video *preferred* 
text *managed*

I think an absolute preference indication is easiest to handle. Media 
preference will collide with relative preference between languages in 
each medium.

I suggest that that is introduced with some suitable coding.

/Gunnar Hellström

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


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    <div class="moz-cite-prefix">On 2013-07-09 08:56, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote cite="mid:BLU169-W1392BC759582397692F5AAA93790@phx.gbl"
      type="cite">
      <style><!--
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      <div dir="ltr">&nbsp;<br>
        <blockquote cite="mid:p0624060ece011b92fdac@%5B99.111.97.136%5D">
          <div>This is very similar to the use case included of a user
            who can hear English but not speak; an audio and text stream
            are set up, both using English.&nbsp; The point of negotiating
            language is to get the call to a callee who can communicate
            with the caller (e.g., a PSAP and call taker who understand
            at least one language in common with the caller).&nbsp; The point
            is not to fully specify exactly how the media will be used;
            there is no need to do so and it won't help get the call to
            the right callee, but will make things much more complex.<br>
            &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </div>
          <div>It is simpler and more robust to handle cases such as you
            describe and the one in the document now by negotiating the
            needed media as two-way streams.</div>
        </blockquote>
        <div> </div>
        <div>[BA]&nbsp; I think this example points out the limitations of a
          call routing approach that relies solely on language. In this
          case, it is important for the "hint" to cause the call to be
          routed to an endpoint that can handle text media, since
          otherwise the caller will not be able to communicate.&nbsp; Since
          both the audio and text media are set up using English in both
          directions,&nbsp;one might conclude that the&nbsp;"hint"&nbsp;need only
          express the language preference (English), but that would not
          be sufficient to get a satisfactory result.&nbsp; What needs to be
          expressed in the "hint" is *both* a preference for English and
          a requirement for text media support. </div>
      </div>
    </blockquote>
    &lt;GH&gt;And the humintlang specification allows that if the
    requirement to have symmetric specifications per media is deleted (
    as described in another mail ).<br>
    <br>
    But there is also a need for absolute preference. Or relative order
    of preference between media.<br>
    <br>
    It is easy to use a sign language example fort that situation.<br>
    <br>
    A person prefers to use British sign language in both directions but
    is capable with some reluctance to use English text. That is
    natural. Text is always slower and less expressive than sign
    language or spoken language. <br>
    <br>
    A call goes to a call center that declared spoken English and
    English text in both directions.<br>
    The result can be a direct connection using English text, or
    invocation of a sign relay service translating between British sign
    language and spoken English. <br>
    <br>
    An ability to declare that the British sign language is preferred
    over the English text would result in the invocation of the sign
    relay service and make both sides happy with a call with good flow.<br>
    <br>
    Without the relative preference between media, the call will likely
    result in the direct text communication that will work but with less
    satisfaction.<br>
    <br>
    This can be solved either by an absolute preference indication&nbsp; :
    British sing language *preferred*&nbsp; , English text *managed*&nbsp;&nbsp; <br>
    Or by some order of preference between the media;&nbsp; video *preferred*
    text *managed*<br>
    <br>
    I think an absolute preference indication is easiest to handle.
    Media preference will collide with relative preference between
    languages in each medium.<br>
    <br>
    I suggest that that is introduced with some suitable coding. <br>
    <br>
    /Gunnar Hellstr&ouml;m<br>
    <br>
    <blockquote cite="mid:BLU169-W1392BC759582397692F5AAA93790@phx.gbl"
      type="cite">
      <div dir="ltr">
        <div>&nbsp;</div>
        <div>
        </div>
      </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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draft-gellens-mmusic-negotiating-human-langua</title></head><body>
<div>Hi Gunnar,</div>
<div><br></div>
<div>Thanks for the quick review and comments.</div>
<div><br></div>
<div><br></div>
<div>At 12:08 PM +0200 7/8/13, Gunnar Hellstrom wrote:</div>
<div><br></div>
<blockquote type="cite" cite>In 7.1, it says:<br>
</blockquote>
<blockquote type="cite" cite><tt>&quot;In an offer, the
'humintlang-send' values constitute a list in<br>
&nbsp;&nbsp; preference order (first is most preferred) of the
languages the<br>
&nbsp;&nbsp; offerer wishes to send, and the 'humintlang-recv' values
constitute a<br>
&nbsp;&nbsp; list in preference order of the languages the offerer
wishes to<br>
&nbsp;&nbsp; receive.&nbsp; Both SHOULD have the same values in the
same order, because<br>
&nbsp;&nbsp; otherwise it is harder to match desired
resources.&quot;</tt></blockquote>
<blockquote type="cite" cite><br></blockquote>
<blockquote type="cite" cite>A typical example of a user who do not
want to specify the same value in both directions is a hard-of-hearing
user who prefers to speak English but might accept to type English
text on the transmission side,&nbsp; but prefers to read English text
and do not have sufficient hearing to at all handle hearing spoken
English on the receiving side.</blockquote>
<div><br></div>
<div>This is very similar to the use case included of a user who can
hear English but not speak; an audio and text stream are set up, both
using English.&nbsp; The point of negotiating language is to get the
call to a callee who can communicate with the caller (e.g., a PSAP and
call taker who understand at least one language in common with the
caller).&nbsp; The point is not to fully specify exactly how the media
will be used; there is no need to do so and it won't help get the call
to the right callee, but will make things much more complex.</div>
<div><br></div>
<div>It is simpler and more robust to handle cases such as you
describe and the one in the document now by negotiating the needed
media as two-way streams.</div>
<div><br></div>
<div><br></div>
<div><br></div>
<x-sigsep><pre>-- 
</pre></x-sigsep>
<div>Randall Gellens<br>
Opinions are personal;&nbsp;&nbsp;&nbsp; facts are
suspect;&nbsp;&nbsp;&nbsp; I speak for myself only<br>
-------------- Randomly selected tag: ---------------<br>
... Any resemblance between the above views and those of my<br>
employer, my terminal, or the view out my window are purely<br>
coincidental. &nbsp;Any resemblance between the above and my own<br>
views is non-deterministic. &nbsp;The question of the existence of<br>
views in the absence of anyone to hold them is left as an<br>
exercise for the reader. &nbsp;The question of the existence of the<br>
reader is left as an exercise for the second god coefficient.<br>
(A discussion of non-orthogonal, non-integral polytheism<br>
is beyond the scope of this article.)<br>
</div></body>
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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-00.txt - two modalities in video
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<!doctype html public "-//W3C//DTD W3 HTML//EN">
<html><head><style type=3D"text/css"><!--
blockquote, dl, ul, ol, li { padding-top: 0 ; padding-bottom: 0 }
 --></style><title>Re: [MMUSIC]
draft-gellens-mmusic-negotiating-human-langua</title></head><body>
<div>Hi Gunnar,</div>
<div><br></div>
<div>Thanks again for the quick review and comments.</div>
<div><br></div>
<div>I agree it can be helpful to include a clarifying note that a
spoken language can be used with a video stream to indicate a request
to see the speaker.&nbsp; I'm not sure that normative language is
needed here but I have added informative text to both 7.1 and 7.4 (on
silly states).</div>
<div><br></div>
<div>At 2:08 PM +0200 7/8/13, Gunnar Hellstrom wrote:</div>
<div><br></div>
<blockquote type=3D"cite" cite>There is a good description of avoidance
of using subtags in 7.1, third paragraph from the end.<br>
<br>
It may be good to inform after that paragraph about the language tags
that can be used for video.<br>
There are separate language tags for sign languages.<br>
So, for the video medium, that can be used to distinguish between two
different usages of video; for sign language versus for assisting
hearing with the view of the speaking human for lip-reading or for
just emphasizing the understanding by a view of the speaker.<br>
<br>
Insert a new paragraph to be the third from the end of 7.1:<br>
<br>
=46or indication of preferences and capabilities of sign languages, the
specific sign language tags of RFC 5646 SHALL be used in the video
media specifications.<br>
Preference and capability of a spoken language tag (i.e. not a sign
language) indicated in the video media specification SHOULD be used
for indicating a view of the speaking human that may be needed or
preferred for speech-reading assisted hearing.<br>
<br>
Regards<br>
<br>
Gunnar<br>
</blockquote>
<blockquote type=3D"cite" cite>
<hr></blockquote>
<blockquote type=3D"cite" cite>Gunnar Hellstr=F6m<br>
Omnitor<br>
<a
href=3D"mailto:gunnar.hellstrom@omnitor.se">gunnar.hellstrom@omnitor.se</a
><br>
+46708204288<br>
</blockquote>
<blockquote type=3D"cite" cite><br>
<blockquote type=3D"cite" cite><br></blockquote>
</blockquote>
<blockquote type=3D"cite" cite><br></blockquote>
<blockquote type=3D"cite" cite><br>
_______________________________________________<br>
mmusic mailing list<br>
mmusic@ietf.org<br>
https://www.ietf.org/mailman/listinfo/mmusic</blockquote>
<div><br></div>
<div><br></div>
<div><br></div>
<x-sigsep><pre>-- 
</pre></x-sigsep>
<div>Randall Gellens<br>
Opinions are personal;&nbsp;&nbsp;&nbsp; facts are
suspect;&nbsp;&nbsp;&nbsp; I speak for myself only<br>
-------------- Randomly selected tag: ---------------<br>
Studied carefully, a modern payphone reveals itself as a little<br>
fortress, carefully designed and redesigned over generations, to resist<br>
coin, slugs, zaps of electricity, chunks of coin-shaped ice, prybars,<br>
magnets, lockpicks, blasting caps. Public pay- phones must survive in a<br>
world of unfriendly, greedy people, and a modern payphone is as<br>
exquisitely evolved as a cactus.<br>
 &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;--Bruce Sterling<br>
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From emil@sip-communicator.org  Tue Jul  9 05:44:09 2013
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Subject: [MMUSIC] Trickle ICE and frozen candidates
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Hey all,

In Orlando we discussed a few issues around trickle ICE without reaching 
any form of consensus so it would be great if we could do so here.

The most delicate is about handling frozen candidates with trickle ICE. 
I apologise for the long e-mail that's about to follow. It's just one of 
those gluey, hairy issues. The faster we get rid of it, the better, so 
please don't be intimidated by the text ;).

With vanilla ICE, we have the advantage of knowing that candidate pairs 
will be ordered the same way on both sides. Checks are executed in the 
same order, which allows NATs to open all the necessary holes in their 
filters.

The "frozen candidates" algorithm relies on this. It starts by 
unfreezing, on both sides, all the first pairs for all foundations. In 
other words, for an audio/video rtp/rtcp call, we normally start ICE 
processing with Audio.RTP and then unfreeze RTCP and Video when the 
first checks have succeeded (roughly put).

The problem with trickle ICE is that such an order is no longer 
guaranteed. It is hence possible that both agents obtain their STUN 
candidates in different order. For example, one of them first obtains 
Audio.RTP, while the other one first learns a mapping for Video RTCP.

Here's a more specific example: Imagine Alice and Bob both have a single 
host address each: A1 and B1 (both are NATed in different LANs). They 
both use ports 5000/5001 for Audio RTP/RTCP and ports 5002/5004 for 
Video RTP/RTCP.  They also both use a STUN server (no TURN for simplicity).

Now let's say that Alice gets (and trickles) a STUN candidate for her 
Audio.RTP base, while Bob first obtains one for RTCP.

For clarity, their check lists would hence look something like this:

Foundation:  A1B1	       A1B2	        A2B1	
Audio.RTP    A1:5000/B1:5000                    A2:5000/B1:5000	
Audio.RTCP   A1:5001/B1:5001   A1:5001/B2:5001		
				
Video.RTP    A1:5002/B1:5002			
Video.RTCP   A1:5003/B1:5003			

We now arrive at the essence: If we are to use freezing in a way that's 
similar to 5245, then Alice and Bob would both unfreeze the first pair 
they have for a given foundation and keep everyone else frozen: Alice 
will thus be sending STUN checks to Bob for Audio.RTCP, while Bob tries 
to conn. check Alice's Audio.RTP candidate.

While they do this, they will learn and trickle all the other candidates 
for these foundations but this wouldn't matter as they would have to 
remain frozen.

Both of the above checks and their entire foundations will hence fail 
since they won't be symmetric.

I hope the description makes sense.

So we basically have two ways to go about this:

1. Try to come up with some rules that avoid or fix the above situation. 
We could try to formulate text that basically starts the unfreezing only 
once we have symmetric candidates for all foundations.

2. We could start by unfreezing the first available pair for any 
foundation and then also unfreeze new pairs with a higher priority. New 
pairs with a lower priority will remain frozen until at least one higher 
priority pair succeeds.

3. We could also get rid of freezing and just start checks for anything 
we get. STUN transactions give us quite some leash in terms of time so 
that even if one of the peers starts checks for a given component 
earlier than the other, this should not be reason enough for failure. 
(Although it could lead to delays in some cases).

While 1 and especially 2 may be preferable in terms of efficiency, 
personally, I am worried that getting it right in terms of spec and 
implementation is non-trivial and this may become a common reason for 
interoperability problems. Therefore, 3 does seem like the simplest way 
to go.

Opinions?

Cheers,
Emil

-- 
https://jitsi.org

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Subject: [MMUSIC] 2 simple Trickle ICE questions (bike shed warning)
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Hey all,

In Orlando there were a couple of points with trickle ICE that triggered 
some discussion but that didn't lead to a specific decision. They are 
relatively simple and can hence be contentious, so hopefully, we could 
agree to focus on actual problems as opposed to "it just doesn't look 
that pretty" concerns.

Here goes:

Q1. Is it OK to keep using ::0 and port 9 in offers and answers that 
carry no other candidates.

Q2. Is it OK to mandate use of "mid" when mapping candidates to an m= 
line (as opposed to an m= line index or a label).

Emil

-- 
https://jitsi.org

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From martin.thomson@gmail.com  Tue Jul  9 12:45:46 2013
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Subject: Re: [MMUSIC] Trickle ICE and frozen candidates
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On 9 July 2013 05:44, Emil Ivov <emcho@jitsi.org> wrote:
> While 1 and especially 2 may be preferable in terms of efficiency,
> personally, I am worried that getting it right in terms of spec and
> implementation is non-trivial and this may become a common reason for
> interoperability problems. Therefore, 3 does seem like the simplest way to
> go.

You are going to want to ensure that both peers check in the same
order.  3 seems a little too loose for me.  I think that some sort of
mechanism is going to be needed to avoid problems with races, since
both peers will have different sets of candidate pairs at every point
in the process.  And signaling is slow enough for this to be a real
problem.

From ari.keranen@ericsson.com  Tue Jul  9 13:52:34 2013
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Cc: Flemming Andreasen <fandreas@cisco.com>
Subject: [MMUSIC] Minutes of the MMUSIC WG June 17th virtual interim meeting
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Hi all,

The draft minutes of the MMUSIC WG June 17th virtual interim meeting 
have been posted at:

http://www.ietf.org/proceedings/interim/2013/06/17/mmusic/minutes/minutes-interim-2013-mmusic-3

Please review the minutes and let us know of any comments or corrections.


Thanks,
Ari & Flemming (MMUSIC co-chairs)

From thomas.stach@siemens-enterprise.com  Tue Jul  9 14:20:48 2013
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From: "Stach, Thomas" <thomas.stach@siemens-enterprise.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: Monday, 08 July, 2013 17:53
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
>=20
> Thomas,
>=20
> On 7/8/13 5:24 AM, Stach, Thomas wrote:
>=20
> > [TS] Saving the half RTT is not my primary goal. When implementing
> ICE we got away without requiring PRACK.
> > I want to keep this property, since it allows
[TS] The remainder of the sentence got lost. It should have read "... since=
 it allows easier introduction RTCWEB client with trickle ICE into a SIP ne=
twork that never used PRACK."
>=20
> I get it that you think PRACK is hard to implement.
> But hard compared to *what*? I suspect that this will introduce
> complexity elsewhere.
[TS] With respect to this complexity I just see the 481 handling for the IN=
FO.=20
Compared to all the new states that I need when I implement PRACK (which ca=
n include a new offer as you know), I just have to keep the previously sent=
 provisional response together with the INFO request. The latter I have to =
keep anyhow for potential retransmissions.=20
If I get a 481, I resent the 18x and the INFO request.
BTW: Just form emphasis. This is only needed if there is a UDP hop somewher=
e, in case of TCP/TLS the 18x would never be lost.

>=20
> >>     If the response for a request within a dialog is a 481
> >>     (Call/Transaction Does Not Exist) or a 408 (Request Timeout),
> the
> >> UAC
> >>     SHOULD terminate the dialog.
> > [TS] We are in the dialog establishing phase here and when the 180
> was lost we don't even have an early dialog.
> > So this rule does not apply. The 481 to the INFO would rather be the
> indicator for this situation. As stated above we can prescribe that the
> 481 (and only the 481) is to be sent for the INFO and what happens
> next, i.e. retransmit the 180.
>=20
> This is part of the house of cards that I was talking about.
> You seem to be saying that a special rule should be be adopted,
> allowing
> INFO to be sent from INVITE-UAS before it is known that the dialog has
> been established. And then a special rule for processing 481 responses
> to INFO in this context.
[TS] Let me reject the comparison with a "house of cards". We are currently=
 specifying a new feature. There are no deployed implementations yet. Thus =
we can exactly specify what should happen at the UAC when it receives an ou=
t-of-dialog INFO because the dialog establishing unreliable 18x was lost, i=
.e. to send 481.=20
On the UAS side we can also specify what to do when the INFO gets a 481. I.=
e. resend the 18x and the INFO request.
I consider this closer to a solid foundation than to a house of cards.

Regards
Thomas

>=20
> >> Also, keep in mind that the 180 carries an entire SDP answer. If
> INFOs
> >> are competing with 180s then they may potentially need to carry that
> >> too
> >> (or at least the ICE parts) and it would be good if we don't have to
> >> open that can of worms.
> > [TS] For the trickle-ICE application of the info-package I would just
> recommend sending (all) the ICE candidates.
> > This eliminates the competition, you just take what you got last.
>=20
> Certainly it will be simpler if all candidates are sent each time.
> But I don't think that means you can just take what you got last. You
> still need to check if what you got is *older* than what you already
> have, and if so ignore it.
>=20
> 	Thanks,
> 	Paul
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From thomas.stach@siemens-enterprise.com  Tue Jul  9 14:21:33 2013
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From: "Stach, Thomas" <thomas.stach@siemens-enterprise.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
Thread-Index: AQHOd/eUM0XqJDBgnkG3edAtUHILLJlUd8BAgAGKMgCAACstAIAEXPYAgABOUoCAAgJgEA==
Date: Tue, 9 Jul 2013 21:21:05 +0000
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Paul,

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: Monday, 08 July, 2013 18:05
> To: Stach, Thomas
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
>=20
> Just responding to things that haven't already been covered.
>=20
> On 7/8/13 5:44 AM, Stach, Thomas wrote:
>=20
> >> Yes, why is it such an issue to require PRACK?
> > [TS] I see potential for easier GW/SBC interworking with vanilla-ICE
> UAs, that did not implement PRACK
>=20
> I don't get it.
>=20
> If you are talking about SBCs that terminate/relay media, then the leg
> doing trickle-ICE is separate from the leg doing vanilla-ICE. So what
> is
> the issue?

>=20
> And if you have a signaling-only SBC, and one leg supports trickle-ICE
> while the other side supports only vanilla-ICE, then just use
> vanilla-ICE e2e. You won't gain anything by an intermediary trying to
> map trickle-ICE to vanilla-ICE.
[TS]=20
If the trickle-ICE side uses PRACK and the vanilla ICE side does not, the i=
nterworking SBC does not only have to collect the trickled candidates and s=
end it on as a bunch once trickling finished. It also has to interwork PRAC=
K.
As you know the PRACK request can contain a new SDP offer or you could also=
 use UPDATE for another SDP O/A before the 200 OK to the INVITE request. Th=
is is rather nasty to interwork if not impossible.

>=20
> >> It is a good thing to implement for many reasons.
> > [TS] There are also good reasons against PRACK. e.g. the much more
> complicated state machine.
>=20
> Yes, but once you have done it then it is there to use for all sorts of
> things. The alternative seems to be to instead introduce a lot of new
> complexity that is *only* applicable to trickle-ICE.
>=20
> 	Thanks,
> 	Paul

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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-00.txt - asymmetric preferences
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At 11:56 PM -0700 7/8/13, Bernard Aboba wrote:

>
>
>>  This is very similar to the use case included of a user who can 
>> hear English but not speak; an audio and text stream are set up, 
>> both using English.  The point of negotiating language is to get 
>> the call to a callee who can communicate with the caller (e.g., a 
>> PSAP and call taker who understand at least one language in common 
>> with the caller).  The point is not to fully specify exactly how 
>> the media will be used; there is no need to do so and it won't 
>> help get the call to the right callee, but will make things much 
>> more complex.
>>
>>  It is simpler and more robust to handle cases such as you describe 
>> and the one in the document now by negotiating the needed media as 
>> two-way streams.
>>
>  [BA]  I think this example points out the limitations of a call 
> routing approach that relies solely on language. In this case, it 
> is important for the "hint" to cause the call to be routed to an 
> endpoint that can handle text media, since otherwise the caller 
> will not be able to communicate.  Since both the audio and text 
> media are set up using English in both directions, one might 
> conclude that the "hint" need only express the language preference 
> (English), but that would not be sufficient to get a satisfactory 
> result.  What needs to be expressed in the "hint" is *both* a 
> preference for English and a requirement for text media support.

[RG] In the case of emergency calls, policy based routing can look at 
media needs and with the proposal to include language in SDP, also 
language.  So for emergency calls, there is no need for a SIP "hint." 
The SIP "hint" is in the current draft because at the Orlando 
discussion it was suggested as a way to allow non-emergency calling 
facilities that lack the ability to examine SDP when determining call 
handling.  I agree that it isn't sufficient if the caller requests 
non-audio media.  I think currently, corporate call centers usually 
have a dedicated number for text communications, while for audio they 
resort to "press 1 for <default language>, 2 for <alternate language 
1>, ..." approaches.  So having just language in the hint would map 
to current usage.

-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly selected tag: ---------------
This is not a novel to be tossed aside lightly.  It should
be thrown with great force.               --Dorothy Parker

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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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On 7/9/13 5:20 PM, Stach, Thomas wrote:
>
>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>> Behalf Of Paul Kyzivat
>> Sent: Monday, 08 July, 2013 17:53
>> To: mmusic@ietf.org
>> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
>>
>> Thomas,
>>
>> On 7/8/13 5:24 AM, Stach, Thomas wrote:
>>
>>> [TS] Saving the half RTT is not my primary goal. When implementing
>> ICE we got away without requiring PRACK.
>>> I want to keep this property, since it allows
> [TS] The remainder of the sentence got lost. It should have read "... since it allows easier introduction RTCWEB client with trickle ICE into a SIP network that never used PRACK."
>>
>> I get it that you think PRACK is hard to implement.
>> But hard compared to *what*? I suspect that this will introduce
>> complexity elsewhere.
> [TS] With respect to this complexity I just see the 481 handling for the INFO.
> Compared to all the new states that I need when I implement PRACK (which can include a new offer as you know), I just have to keep the previously sent provisional response together with the INFO request. The latter I have to keep anyhow for potential retransmissions.
> If I get a 481, I resent the 18x and the INFO request.
> BTW: Just form emphasis. This is only needed if there is a UDP hop somewhere, in case of TCP/TLS the 18x would never be lost.

With PRACK, the work is implementing it as it is specified. And then you 
can use it everywhere, not just for trickle-ICE.

I'm not sure I understand your proposal for using info in the context, 
but I *think* you are asking for *special* handling of 481 for INFO, 
contrary to how specs call for it to be implemented. For one think, this 
would effectively be a revision to some existing specification. And 
maybe you only what this handling when INFO is used for trickle-ICE?

Remember, introducing this will often mean using an existing 
implementation of INFO for this purpose.

>>>>      If the response for a request within a dialog is a 481
>>>>      (Call/Transaction Does Not Exist) or a 408 (Request Timeout),
>> the
>>>> UAC
>>>>      SHOULD terminate the dialog.
>>> [TS] We are in the dialog establishing phase here and when the 180
>> was lost we don't even have an early dialog.
>>> So this rule does not apply. The 481 to the INFO would rather be the
>> indicator for this situation. As stated above we can prescribe that the
>> 481 (and only the 481) is to be sent for the INFO and what happens
>> next, i.e. retransmit the 180.
>>
>> This is part of the house of cards that I was talking about.
>> You seem to be saying that a special rule should be be adopted,
>> allowing
>> INFO to be sent from INVITE-UAS before it is known that the dialog has
>> been established. And then a special rule for processing 481 responses
>> to INFO in this context.
> [TS] Let me reject the comparison with a "house of cards". We are currently specifying a new feature. There are no deployed implementations yet. Thus we can exactly specify what should happen at the UAC when it receives an out-of-dialog INFO because the dialog establishing unreliable 18x was lost, i.e. to send 481.
> On the UAS side we can also specify what to do when the INFO gets a 481. I.e. resend the 18x and the INFO request.
> I consider this closer to a solid foundation than to a house of cards.

Lets first decide in you are proposing a special mechanism for use with 
trickle-ICE, or if you are proposing a revison (bug fix?) to INFO, or to 
3261 & 5057.

If you are proposing a change in how INFO works exclusively when used 
for trickle-ICE then I think it is a non-starter.

If you are proposing a revision regarding the handling of presumed-to-be 
in-dialog requests from UAS to UAC before the UAS has proof that the UAC 
knows of the dialog, then we can talk.

	Thanks,
	Paul

> Regards
> Thomas
>
>>
>>>> Also, keep in mind that the 180 carries an entire SDP answer. If
>> INFOs
>>>> are competing with 180s then they may potentially need to carry that
>>>> too
>>>> (or at least the ICE parts) and it would be good if we don't have to
>>>> open that can of worms.
>>> [TS] For the trickle-ICE application of the info-package I would just
>> recommend sending (all) the ICE candidates.
>>> This eliminates the competition, you just take what you got last.
>>
>> Certainly it will be simpler if all candidates are sent each time.
>> But I don't think that means you can just take what you got last. You
>> still need to check if what you got is *older* than what you already
>> have, and if so ignore it.
>>
>> 	Thanks,
>> 	Paul
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>


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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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On 7/9/13 5:21 PM, Stach, Thomas wrote:
> Paul,
>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>> Sent: Monday, 08 July, 2013 18:05
>> To: Stach, Thomas
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
>>
>> Just responding to things that haven't already been covered.
>>
>> On 7/8/13 5:44 AM, Stach, Thomas wrote:
>>
>>>> Yes, why is it such an issue to require PRACK?
>>> [TS] I see potential for easier GW/SBC interworking with vanilla-ICE
>> UAs, that did not implement PRACK
>>
>> I don't get it.
>>
>> If you are talking about SBCs that terminate/relay media, then the leg
>> doing trickle-ICE is separate from the leg doing vanilla-ICE. So what
>> is
>> the issue?
>
>>
>> And if you have a signaling-only SBC, and one leg supports trickle-ICE
>> while the other side supports only vanilla-ICE, then just use
>> vanilla-ICE e2e. You won't gain anything by an intermediary trying to
>> map trickle-ICE to vanilla-ICE.
> [TS]
> If the trickle-ICE side uses PRACK and the vanilla ICE side does not, the interworking SBC does not only have to collect the trickled candidates and send it on as a bunch once trickling finished. It also has to interwork PRACK.
> As you know the PRACK request can contain a new SDP offer or you could also use UPDATE for another SDP O/A before the 200 OK to the INVITE request. This is rather nasty to interwork if not impossible.

If trickle-ICE worked without PRACK, then the SBC *still* needs to 
collect up all the trickled candidates and send them when it can based 
on the limitations of vanilla-ICE. So the use of trickle-ICE provides no 
speedup compared to using vanilla-ICE e2e.

Also, it sounds like trickle-ICE is requiring changes in the ordering of 
how candidates are tried. If one side uses trickle-ICE and the other 
doesn't, then ISTM that may break.

So I'm still not seeing a case here when it makes sense to gateway 
between trickle-ICE and vanilla-ICE. There may be one, but I'm not 
seeing it.

	Thanks,
	Paul


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Subject: Re: [MMUSIC] Notes from Orlando human language draft discussion
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Hi Christian,

Thanks for the review and comments.


At 5:13 PM +1000 7/9/13, Christian Groves wrote:

>  I had a look at the draft with respect to CLUE. The main difference 
> between the CLUE usage of language and what is being proposed is 
> the ability for the sender to specify sending AND receiving 
> preferences. The current CLUE model is that an each endpoint 
> Advertises what it can do and then each consumer chooses from this 
> list. This could still result in an asymmetric usage of language 
> (just negotiated differently).

The draft doesn't try to accommodate asymmetric usage, since that 
usually can just happen without detailed technical backing.  The 
"-send" and "-receive" attributes are there because Harald warned 
that we'd be violating SDP otherwise.

If CLUE can allow the caller to indicate in the offer language needs 
for each stream, this to be taken into accounting in policy bvased 
routing, and the callee to respond to the offer with a language in 
common, then perhaps we don't need this proposal at all and can just 
drop it in favor of CLUE.  I admit at this point I am ignorant of 
CLUE (yes, clueless about CLUE).


>  You may need to consider the case where there are multiple streams 
> of the same media type with different languages. e.g. an audio 
> conference where one stream is the main audio of the conference and 
> there's a second stream with a translation/s. This is the type of 
> thing CLUE is considering. I'm not sure how this fits in with your 
> work?

This is a different, and more complex, scenario than what this 
proposal is trying to solve.  This proposal is focused on simpler 
cases where someone places an emergency call or calls a corporate 
call center.  Possibly this can be just a simple case of the more 
complex cases?

>
>  Some other comments:
>
>  With regards to the SIP "hint" it wasn't clear to me how this 
> matches the languages in "humintlang-send" and "humintlang-recv". 
> Is it planned that both will be supported in the header?

I suppose this depends on if the "-send" and "-recv" are really 
needed, and if so if they should be the same.

>
>  The "TBD" paragraph in 7.3 SIP "hint" mentions (where the caller 
> sets media and language) but in the final paragraph of the 7.3 it 
> sounds like the "hint" is only for audio?

Good catch -- this will need to be sorted out.

>
>  7.4 Silly states - How do the language sub-tags come into play 
> here? For example: the endpoint might specify a language for an 
> audio stream which may be valid. If the endpoint also specified a 
> sub-tag with "script" I guess it makes it a silly state?
>
>  Regards, Christian
>
>  On 8/07/2013 8:40 AM, Randall Gellens wrote:
>>  At 4:50 PM +1100 3/25/13, Christian Groves wrote:
>>
>>>   Hello,
>>>
>>>   One thing that doesn't seem to be noted is an overlap with the 
>>> work of the CLUE WG. There has been some discussion in the CLUE 
>>> working group about adding attributes to captures to describe the 
>>> language, whether sub-titles are used, whether a stream comes 
>>> from an interpreter, etc. Where multiple streams are used I would 
>>> expect any work in this area should be aligned.
>>>
>>>   Regards, Christian
>>
>>  Hi Christian,
>>
>>  Thanks for the info.  Would it make sense to talk about this in 
>> Berlin?  I agree that if there is overlap in the approaches we 
>> should align them.
>>
>>  Also, I have an updated draft that was just uploaded.  It is now 
>> named 'draft-gellens-mmusic-negotiating-human-language-00' (adding 
>> -mmusic and hence reverting back to -00).
>>
>
>  _______________________________________________
>  mmusic mailing list
>  mmusic@ietf.org
>  https://www.ietf.org/mailman/listinfo/mmusic


-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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A few cross mail replies:

On 09.07.13, 23:20, Stach, Thomas wrote:
> BTW: Just form emphasis. This is only needed if there is a UDP hop
> somewhere, in case of TCP/TLS the 18x would never be lost.

What does this change for anyone? Clients have no way of knowing whether
there would be UDP hops along the way so they would always need to be
ready for this option.

> [TS] As Trickle-ICE is just being specified, we are in a position to
> specify this handling.

While this is true, I see the following issue here: cumulative sending
is only necessary for SIP/UDP (which is quite a paradox because SIP/UDP
with ICE would break in a big number of cases).

> [TS] As the ICE/STUN logic resides in the browser not in the JS app,
> this would be a matter of the API definition to always provide all
> known candidates towards the JS APP for sending. On the receiving
> side the JS APP would also just have to push all received candidates
> through the browser API.

I find it strange that we would be imposing constraints on a new generic
API because of a broken use case that only exists with one usage
protocol. Especially given that the said use case (SIP+ICE over UDP) is
NOT going to work anyway because of MTUs and improper IP fragmentation
handling on various devices today.

>>> We are indeed likely to get a 481 when sending INFOs and the 180
>>> hasn't yet made it to the remote side. However 481 might also
>>> imply that our dialog is simply dead for some reason and in this
>>> case 3261 says:
>>>
>>>
>>> If the response for a request within a dialog is a 481
>>> (Call/Transaction Does Not Exist) or a 408 (Request Timeout),
>>> the UAC SHOULD terminate the dialog.
>
> [TS] We are in the dialog establishing phase here and when the 180
> was lost we don't even have an early dialog. So this rule does not
> apply.

I am not sure this is true. There *is* an early dialog at the side 
that's sending the 180. The INFO is a second request within that same 
dialog and 3261 is very clear that a 481 would lead to termination of 
the dialog. Specifying different behaviour here would mean that many 
developers may need to start fighting/changing their SIP stacks.

Cheers,
Emil

-- 
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Subject: Re: [MMUSIC] Notes from Orlando human language draft discussion
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On 2013-07-09 23:58, Randall Gellens wrote:
> This is a different, and more complex, scenario than what this 
> proposal is trying to solve.  This proposal is focused on simpler 
> cases where someone places an emergency call or calls a corporate call 
> center.  Possibly this can be just a simple case of the more complex 
> cases? 
My view is that it is intended to also cover user-to-user cases, where 
preferences are compared and a relay service (or language interpreter) 
can be invoked to bridge language gaps.

Gunnar



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Hey Martin,

On 09.07.13, 21:45, Martin Thomson wrote:
> On 9 July 2013 05:44, Emil Ivov <emcho@jitsi.org> wrote:
>> While 1 and especially 2 may be preferable in terms of efficiency,
>> personally, I am worried that getting it right in terms of spec and
>> implementation is non-trivial and this may become a common reason for
>> interoperability problems. Therefore, 3 does seem like the simplest way to
>> go.
>
> You are going to want to ensure that both peers check in the same
> order.

What exactly do you mean by "in the same order"? Do you mean that:

a) they should try and make sure that they are checking the candidate 
pair that is symmetric to the one their peer is testing and
b) all other components for the involved foundations should be frozen?

Do you have a better wording?

> 3 seems a little too loose for me.

It is indeed loose but I can't think of a reason why it wouldn't work 
even with several seconds of RTT. So, unless you are going to manually 
paste every new candidate into a mail and then do the the same on the 
remote side, you should probably be fine.

Ekr, I believe you mentioned you were doing this in Firefox? How is it 
working?

> I think that some sort of
> mechanism is going to be needed to avoid problems with races, since
> both peers will have different sets of candidate pairs at every point
> in the process.

OK. Do you mean something like 2. or do you have another mechanism in mind?

Emil


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From Christian.Groves@nteczone.com  Tue Jul  9 21:45:29 2013
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Hello Randall,

Please see my response below.

Regards, Christian

On 10/07/2013 7:58 AM, Randall Gellens wrote:
> Hi Christian,
>
> Thanks for the review and comments.
>
>
> At 5:13 PM +1000 7/9/13, Christian Groves wrote:
>
>>  I had a look at the draft with respect to CLUE. The main difference 
>> between the CLUE usage of language and what is being proposed is the 
>> ability for the sender to specify sending AND receiving preferences. 
>> The current CLUE model is that an each endpoint Advertises what it 
>> can do and then each consumer chooses from this list. This could 
>> still result in an asymmetric usage of language (just negotiated 
>> differently).
>
> The draft doesn't try to accommodate asymmetric usage, since that 
> usually can just happen without detailed technical backing.  The 
> "-send" and "-receive" attributes are there because Harald warned that 
> we'd be violating SDP otherwise.
[CNG] By asymmetric I mean that a different language could be used in 
the send and receive sections. Are you saying that the draft assumes the 
same language in the "-send" and "-receive" attributes?
>
> If CLUE can allow the caller to indicate in the offer language needs 
> for each stream, this to be taken into accounting in policy bvased 
> routing, and the callee to respond to the offer with a language in 
> common, then perhaps we don't need this proposal at all and can just 
> drop it in favor of CLUE.  I admit at this point I am ignorant of CLUE 
> (yes, clueless about CLUE).

[CNG] I think you still want your proposal. Endpoints shouldn't be 
compelled to use CLUE to indicate a language preference. The first CLUE 
advertisement will most likely come after the initial SIP SDP Offer so 
it would be difficult to route the call based on CLUE language information.
I'm still trying to understand what the relationship/mapping would be 
between the language specified as a SIP "hint", to the SDP "-send" and 
"-receive" attributes in an Initial SDP Offer to any subsequent CLUE 
signalling that includes a language tag.
For example: If an endpoint sends an initial SIP (with 'hint' with SDP 
Offer with a set of languages for a single media (and indicates CLUE 
support). Should the endpoint when sending a CLUE Advertisement only 
advertise captures that contain the negotiated languages? (Seeing "the 
hint" is at the call level) Or, should it treat any media described by 
CLUE as something completely new?

>
>
>>  You may need to consider the case where there are multiple streams 
>> of the same media type with different languages. e.g. an audio 
>> conference where one stream is the main audio of the conference and 
>> there's a second stream with a translation/s. This is the type of 
>> thing CLUE is considering. I'm not sure how this fits in with your work?
>
> This is a different, and more complex, scenario than what this 
> proposal is trying to solve.  This proposal is focused on simpler 
> cases where someone places an emergency call or calls a corporate call 
> center.  Possibly this can be just a simple case of the more complex 
> cases?

[CNG] I don't think this is a problem for the SDP attributes, I think 
the issue is more for the SIP "hint" if it is media type specific. For 
example if you set "LANG1" and "LANG2" against "audio" and you actually 
have two streams one of LANG1 and another of LANG2 and base your routing 
decision on this there is a chance that you route to an endpoint based 
on LANG2 but when the endpoint inspects the SDP it actually wants the 
LANG1 stream.

>
>>
>>  Some other comments:
>>
>>  With regards to the SIP "hint" it wasn't clear to me how this 
>> matches the languages in "humintlang-send" and "humintlang-recv". Is 
>> it planned that both will be supported in the header?
>
> I suppose this depends on if the "-send" and "-recv" are really 
> needed, and if so if they should be the same.

[CNG] OK I didn't understand that from the draft.

>
>>
>>  The "TBD" paragraph in 7.3 SIP "hint" mentions (where the caller 
>> sets media and language) but in the final paragraph of the 7.3 it 
>> sounds like the "hint" is only for audio?
>
> Good catch -- this will need to be sorted out.
>
>>
>>  7.4 Silly states - How do the language sub-tags come into play here? 
>> For example: the endpoint might specify a language for an audio 
>> stream which may be valid. If the endpoint also specified a sub-tag 
>> with "script" I guess it makes it a silly state?
>>
>>  Regards, Christian
>>
>>  On 8/07/2013 8:40 AM, Randall Gellens wrote:
>>>  At 4:50 PM +1100 3/25/13, Christian Groves wrote:
>>>
>>>>   Hello,
>>>>
>>>>   One thing that doesn't seem to be noted is an overlap with the 
>>>> work of the CLUE WG. There has been some discussion in the CLUE 
>>>> working group about adding attributes to captures to describe the 
>>>> language, whether sub-titles are used, whether a stream comes from 
>>>> an interpreter, etc. Where multiple streams are used I would expect 
>>>> any work in this area should be aligned.
>>>>
>>>>   Regards, Christian
>>>
>>>  Hi Christian,
>>>
>>>  Thanks for the info.  Would it make sense to talk about this in 
>>> Berlin?  I agree that if there is overlap in the approaches we 
>>> should align them.
>>>
>>>  Also, I have an updated draft that was just uploaded.  It is now 
>>> named 'draft-gellens-mmusic-negotiating-human-language-00' (adding 
>>> -mmusic and hence reverting back to -00).
>>>
>>
>>  _______________________________________________
>>  mmusic mailing list
>>  mmusic@ietf.org
>>  https://www.ietf.org/mailman/listinfo/mmusic
>
>


From Christian.Groves@nteczone.com  Tue Jul  9 21:49:23 2013
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I wouldn't have thought that it would be use case specific... i.e. i'd 
only include my language when I called an emergency service. I would 
think an endpoint would be configured with the information and then 
provide it whenever I made a call. Is that the intent?

Christian

On 10/07/2013 9:23 AM, Gunnar Hellstrom wrote:
> On 2013-07-09 23:58, Randall Gellens wrote:
>> This is a different, and more complex, scenario than what this 
>> proposal is trying to solve.  This proposal is focused on simpler 
>> cases where someone places an emergency call or calls a corporate 
>> call center.  Possibly this can be just a simple case of the more 
>> complex cases? 
> My view is that it is intended to also cover user-to-user cases, where 
> preferences are compared and a relay service (or language interpreter) 
> can be invoked to bridge language gaps.
>
> Gunnar
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-miscellaneous-caps-07.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           : Miscellaneous Capabilities Negotiation in the Session De=
scription Protocol (SDP)
	Author(s)       : Miguel A. Garcia-Martin
                          Simo Veikkolainen
                          Robert R. Gilman
	Filename        : draft-ietf-mmusic-sdp-miscellaneous-caps-07.txt
	Pages           : 22
	Date            : 2013-07-10

Abstract:
   SDP has been extended with a capability negotiation mechanism
   framework that allows the endpoints to negotiate transport protocols
   and attributes.  This framework has been extended with a media
   capabilities negotiation mechanism that allows endpoints to negotiate
   additional media-related capabilities.  This negotiation is embedded
   into the widely-used SDP offer/answer procedures.

   This memo extends the SDP capability negotiation framework to allow
   endpoints to negotiate three additional SDP capabilities.  In
   particular, this memo provides a mechanism to negotiate bandwidth
   ('b=3D' line), connection data ('c=3D' line), and session or media titles
   ('i=3D' line for each session or media).


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-miscellaneous-caps-07

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-miscellaneous-caps=
-07


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


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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-miscellaneous-caps-07.txt
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Dear WG,
We have submitted version -07 of the SDP Miscellaneous Capabilities draft. =
Changes requested during the IESG review are:
-	Added an example where two title capabilities are offered with different =
character encodings
-	Added a reference to RFC5939 SDP Capability Negotiation in the Introducti=
on section to remind implementers how supporting and/or requiring capabilit=
y extensions is done during Offer/Answer negotiation (by the "csup" and "cr=
eq" option tags).

On behalf of the authors,
Simo

https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-miscellaneous-caps=20
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-miscellaneous-caps-07=20
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-miscellaneous-caps=
-07=20

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 ext internet-drafts@ietf.org
Sent: 10. hein=E4kuuta 2013 12:17
To: i-d-announce@ietf.org
Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-miscellaneous-caps-07.t=
xt


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           : Miscellaneous Capabilities Negotiation in the Session De=
scription Protocol (SDP)
	Author(s)       : Miguel A. Garcia-Martin
                          Simo Veikkolainen
                          Robert R. Gilman
	Filename        : draft-ietf-mmusic-sdp-miscellaneous-caps-07.txt
	Pages           : 22
	Date            : 2013-07-10

Abstract:
   SDP has been extended with a capability negotiation mechanism
   framework that allows the endpoints to negotiate transport protocols
   and attributes.  This framework has been extended with a media
   capabilities negotiation mechanism that allows endpoints to negotiate
   additional media-related capabilities.  This negotiation is embedded
   into the widely-used SDP offer/answer procedures.

   This memo extends the SDP capability negotiation framework to allow
   endpoints to negotiate three additional SDP capabilities.  In
   particular, this memo provides a mechanism to negotiate bandwidth
   ('b=3D' line), connection data ('c=3D' line), and session or media title=
s
   ('i=3D' line for each session or media).


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-miscellaneous-caps-07

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-miscellaneous-caps=
-07


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

_______________________________________________
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mmusic@ietf.org
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From gavin.llewellyn@crocodilertc.net  Wed Jul 10 04:50:57 2013
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From: Gavin Llewellyn <gavin.llewellyn@crocodilertc.net>
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Subject: Re: [MMUSIC] [dispatch] FW: New Version Notification for draft-marcon-msrp-over-webrtc-data-channels-00.txt
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--089e0160b6cea8abbf04e126e33b
Content-Type: text/plain; charset=ISO-8859-1

As a co-author of draft-pd-msrp-webrtc-00, I support this new draft for
establishment of MSRP sessions between WebRTC endpoints.  The SDP
enhancements allowing re-use of the SCTP association is very much along the
lines of what we had in mind for the next version of our draft, but we have
not had time to complete it.  We will likely be implementing this as a
feature in our WebRTC library in the near future, as the DataChannel
implementations in browsers mature (support SCTP, reliable delivery, binary
data, etc).

Having said that, it would have been appreciated if the authors had
contacted Peter and/or myself to discuss the existing draft and its
limitations - we would have been happy to receive feedback and work
together to find the best solution going forward.

I cannot see the value in introducing DataChannel gateways as a means to
communicate with traditional TCP-based MSRP endpoints.  I believe this is
better catered for by the WebSocket-based solution, as described in
draft-pd-dispatch-msrp-websocket,
which already has client and server implementations, and is currently
in-use in a production environment.  Given the (intentional) similarities
between the WebSocket and DataChannel APIs, it is not difficult for the
client application to support both transports.  I would like to see a clear
justification for this aspect of the draft.

One minor correction: the opening paragraph of section 5.2.3 appears to
reference the wrong SDP syntax for sub-protocol-specific attributes; I
believe it should reference the syntax for the new "wdcsa" attribute.

Regards,
Gavin Llewellyn

--
Gavin Llewellyn
Principal Design Engineer
Crocodile RCS Ltd
GPG key: 0xF8F6FFF2


On 9 July 2013 20:32, Ejzak, Richard P (Richard) <
richard.ejzak@alcatel-lucent.com> wrote:

> Please note our proposed alternative to draft-pd-msrp-webrtc-00 for how to
> use DataChannel transport for MSRP.  Our draft does not place restrictions
> on the number of MSRP sessions that can be established within an SCTP
> association and describes alternatives for using a gateway to interconnect
> to existing MSRP endpoints.  The SDP extensions described are applicable to
> other potential DataChannel sub-protocols such as T.140, BFCP, T.38, etc.
>
> Richard
>
> -----Original Message-----
> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
> Sent: Monday, July 08, 2013 6:31 PM
> To: Ejzak, Richard P (Richard); MARCON, JEROME (JEROME)
> Subject: New Version Notification for
> draft-marcon-msrp-over-webrtc-data-channels-00.txt
>
>
> A new version of I-D, draft-marcon-msrp-over-webrtc-data-channels-00.txt
> has been successfully submitted by Jerome Marcon and posted to the IETF
> repository.
>
> Filename:        draft-marcon-msrp-over-webrtc-data-channels
> Revision:        00
> Title:           MSRP over WebRTC data channels
> Creation date:   2013-07-09
> Group:           Individual Submission
> Number of pages: 12
> URL:
> http://www.ietf.org/internet-drafts/draft-marcon-msrp-over-webrtc-data-channels-00.txt
> Status:
> http://datatracker.ietf.org/doc/draft-marcon-msrp-over-webrtc-data-channels
> Htmlized:
> http://tools.ietf.org/html/draft-marcon-msrp-over-webrtc-data-channels-00
>
>
> Abstract:
>    The Real-Time Communication in WEB-browsers (RTCWeb) working group is
>    charged to provide protocols to support direct interactive rich
>    communication using audio, video, and data between two peers' web-
>    browsers.  For the support of data communication, the RTCWeb working
>    group has in particular defined the concept of bi-directional data
>    channels over SCTP, where each data channel might be used to
>    transport other protocols, called sub-protocols.  This document
>    specifies how the Message Session Relay Protocol (MSRP) can be
>    instantiated as a WebRTC data channel sub-protocol, using the SDP
>    offer/exchange to negotiate out-of-band the sub-protocol specific
>    parameters.  Two network configurations are documented: a WebRTC end-
>    to-end configuration (connecting two MSRP over data channel
>    endpoints), and a gateway configuration (connecting an MSRP over data
>    channel endpoint with an MSRP over TCP endpoint).
>
>
>
>
> The IETF Secretariat
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

--089e0160b6cea8abbf04e126e33b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">As a co-author of=A0<span style=3D"font-family:arial,sans-=
serif;font-size:13px">draft-pd-msrp-webrtc-00, I support this new draft for=
 establishment of MSRP sessions between WebRTC endpoints. =A0The SDP enhanc=
ements allowing re-use of the SCTP association is very much along the lines=
 of what we had in mind for the next version of our draft, but we have not =
had time to complete it. =A0We will likely be implementing this as a featur=
e in our WebRTC library in the near future, as the DataChannel implementati=
ons in browsers mature (support SCTP, reliable delivery, binary data, etc).=
</span><div>

<span style=3D"font-family:arial,sans-serif;font-size:13px"><br></span></di=
v><div><span style=3D"font-family:arial,sans-serif;font-size:13px">Having s=
aid that, it would have been appreciated if the authors had contacted Peter=
 and/or myself to discuss the existing draft and its limitations - we would=
 have been happy to receive feedback and work together to find the best sol=
ution going forward.</span></div>

<div><span style=3D"font-family:arial,sans-serif;font-size:13px"><br></span=
></div><div><span style=3D"font-family:arial,sans-serif;font-size:13px">I c=
annot see the value in introducing DataChannel gateways as a means to commu=
nicate with traditional TCP-based MSRP endpoints. =A0I believe this is bett=
er catered for by the WebSocket-based solution, as described in=A0</span><f=
ont face=3D"arial, sans-serif">draft-pd-dispatch-msrp-websocket, which alre=
ady has client and server implementations, and is currently in-use in a pro=
duction environment. =A0Given the (intentional) similarities between the We=
bSocket and DataChannel APIs, it is not difficult for the client applicatio=
n to support both transports. =A0I would like to see a clear justification =
for this aspect of the draft.</font></div>

<div><font face=3D"arial, sans-serif"><br></font></div><div><font face=3D"a=
rial, sans-serif">One minor correction: the opening paragraph of section 5.=
2.3 appears to reference the wrong SDP syntax for sub-protocol-specific att=
ributes; I believe it should reference the syntax for the new &quot;</font>=
<span style=3D"white-space:pre-wrap">wdcsa&quot; attribute.</span></div>

<div><span style=3D"white-space:pre-wrap"><br></span></div><div><span style=
=3D"white-space:pre-wrap">Regards,</span></div><div><span style=3D"white-sp=
ace:pre-wrap">Gavin Llewellyn</span></div>
<div><br></div><div class=3D"gmail_extra"><div><div dir=3D"ltr"><div>--</di=
v>Gavin Llewellyn<div>Principal Design Engineer</div><div>Crocodile RCS Ltd=
</div><div>GPG key: 0xF8F6FFF2</div></div></div>
<br><br><div class=3D"gmail_quote">On 9 July 2013 20:32, Ejzak, Richard P (=
Richard) <span dir=3D"ltr">&lt;<a href=3D"mailto:richard.ejzak@alcatel-luce=
nt.com" target=3D"_blank">richard.ejzak@alcatel-lucent.com</a>&gt;</span> w=
rote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Please note our proposed alternative to draf=
t-pd-msrp-webrtc-00 for how to use DataChannel transport for MSRP. =A0Our d=
raft does not place restrictions on the number of MSRP sessions that can be=
 established within an SCTP association and describes alternatives for usin=
g a gateway to interconnect to existing MSRP endpoints. =A0The SDP extensio=
ns described are applicable to other potential DataChannel sub-protocols su=
ch as T.140, BFCP, T.38, etc.<br>


<br>
Richard<br>
<br>
-----Original Message-----<br>
From: <a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">interne=
t-drafts@ietf.org</a> [mailto:<a href=3D"mailto:internet-drafts@ietf.org" t=
arget=3D"_blank">internet-drafts@ietf.org</a>]<br>
Sent: Monday, July 08, 2013 6:31 PM<br>
To: Ejzak, Richard P (Richard); MARCON, JEROME (JEROME)<br>
Subject: New Version Notification for draft-marcon-msrp-over-webrtc-data-ch=
annels-00.txt<br>
<br>
<br>
A new version of I-D, draft-marcon-msrp-over-webrtc-data-channels-00.txt<br=
>
has been successfully submitted by Jerome Marcon and posted to the IETF rep=
ository.<br>
<br>
Filename: =A0 =A0 =A0 =A0draft-marcon-msrp-over-webrtc-data-channels<br>
Revision: =A0 =A0 =A0 =A000<br>
Title: =A0 =A0 =A0 =A0 =A0 MSRP over WebRTC data channels<br>
Creation date: =A0 2013-07-09<br>
Group: =A0 =A0 =A0 =A0 =A0 Individual Submission<br>
Number of pages: 12<br>
URL: =A0 =A0 =A0 =A0 =A0 =A0 <a href=3D"http://www.ietf.org/internet-drafts=
/draft-marcon-msrp-over-webrtc-data-channels-00.txt" target=3D"_blank">http=
://www.ietf.org/internet-drafts/draft-marcon-msrp-over-webrtc-data-channels=
-00.txt</a><br>


Status: =A0 =A0 =A0 =A0 =A0<a href=3D"http://datatracker.ietf.org/doc/draft=
-marcon-msrp-over-webrtc-data-channels" target=3D"_blank">http://datatracke=
r.ietf.org/doc/draft-marcon-msrp-over-webrtc-data-channels</a><br>
Htmlized: =A0 =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-marcon=
-msrp-over-webrtc-data-channels-00" target=3D"_blank">http://tools.ietf.org=
/html/draft-marcon-msrp-over-webrtc-data-channels-00</a><br>
<br>
<br>
Abstract:<br>
=A0 =A0The Real-Time Communication in WEB-browsers (RTCWeb) working group i=
s<br>
=A0 =A0charged to provide protocols to support direct interactive rich<br>
=A0 =A0communication using audio, video, and data between two peers&#39; we=
b-<br>
=A0 =A0browsers. =A0For the support of data communication, the RTCWeb worki=
ng<br>
=A0 =A0group has in particular defined the concept of bi-directional data<b=
r>
=A0 =A0channels over SCTP, where each data channel might be used to<br>
=A0 =A0transport other protocols, called sub-protocols. =A0This document<br=
>
=A0 =A0specifies how the Message Session Relay Protocol (MSRP) can be<br>
=A0 =A0instantiated as a WebRTC data channel sub-protocol, using the SDP<br=
>
=A0 =A0offer/exchange to negotiate out-of-band the sub-protocol specific<br=
>
=A0 =A0parameters. =A0Two network configurations are documented: a WebRTC e=
nd-<br>
=A0 =A0to-end configuration (connecting two MSRP over data channel<br>
=A0 =A0endpoints), and a gateway configuration (connecting an MSRP over dat=
a<br>
=A0 =A0channel endpoint with an MSRP over TCP endpoint).<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<br>
<br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.org</a=
><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</blockquote></div><br></div></div>

--089e0160b6cea8abbf04e126e33b--

From richard.ejzak@alcatel-lucent.com  Wed Jul 10 07:22:49 2013
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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: Gavin Llewellyn <gavin.llewellyn@crocodilertc.net>
Thread-Topic: [dispatch] FW: New Version Notification for draft-marcon-msrp-over-webrtc-data-channels-00.txt
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--_000_03FBA798AC24E3498B74F47FD082A92F3D802975US70UWXCHMBA04z_
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Hi Gavin,
Thanks for your support of our approach and for your comments.  It's great =
that you are planning an implementation along these lines.

I am including mmusic on this distribution to not exclude anyone who may be=
 interested, but please limit the distribution to dispatch only for subsequ=
ent emails in this thread.

Regarding why we chose to create a new draft: Our approach was sufficiently=
 different from yours that we felt a separate draft was a better idea in th=
e short term.  We can merge when it's appropriate to do so.  Even though we=
 have no significant difference of opinion regarding the SDP extensions nee=
ded, we clearly have different views regarding the value of a DataChannel g=
ateway.  It would be good to hear other opinions on this matter, as you are=
 clearly heavily committed to your preference.

Our preference for support of a DataChannel gateway model is based on the n=
umerous advantages of DataChannels.  Just a partial list: We have one trans=
port mechanism for all cases; we don't have to offer two options or perform=
 fallback/retry procedures on failure of one transport option; we can easil=
y multiplex all media and sub-protocols together; we can leverage the integ=
rated browser handling of multiple flows of different priority; we have to =
be able to interoperate with legacy tcp/tls endpoints anyway.

We look forward to continuing the debate.

BR, Richard

From: Gavin Llewellyn [mailto:gavin.llewellyn@crocodilertc.net]
Sent: Wednesday, July 10, 2013 6:51 AM
To: Ejzak, Richard P (Richard)
Cc: dispatch@ietf.org; mmusic@ietf.org WG
Subject: Re: [dispatch] FW: New Version Notification for draft-marcon-msrp-=
over-webrtc-data-channels-00.txt

As a co-author of draft-pd-msrp-webrtc-00, I support this new draft for est=
ablishment of MSRP sessions between WebRTC endpoints.  The SDP enhancements=
 allowing re-use of the SCTP association is very much along the lines of wh=
at we had in mind for the next version of our draft, but we have not had ti=
me to complete it.  We will likely be implementing this as a feature in our=
 WebRTC library in the near future, as the DataChannel implementations in b=
rowsers mature (support SCTP, reliable delivery, binary data, etc).

Having said that, it would have been appreciated if the authors had contact=
ed Peter and/or myself to discuss the existing draft and its limitations - =
we would have been happy to receive feedback and work together to find the =
best solution going forward.

I cannot see the value in introducing DataChannel gateways as a means to co=
mmunicate with traditional TCP-based MSRP endpoints.  I believe this is bet=
ter catered for by the WebSocket-based solution, as described in draft-pd-d=
ispatch-msrp-websocket, which already has client and server implementations=
, and is currently in-use in a production environment.  Given the (intentio=
nal) similarities between the WebSocket and DataChannel APIs, it is not dif=
ficult for the client application to support both transports.  I would like=
 to see a clear justification for this aspect of the draft.

One minor correction: the opening paragraph of section 5.2.3 appears to ref=
erence the wrong SDP syntax for sub-protocol-specific attributes; I believe=
 it should reference the syntax for the new "wdcsa" attribute.

Regards,
Gavin Llewellyn

--
Gavin Llewellyn
Principal Design Engineer
Crocodile RCS Ltd
GPG key: 0xF8F6FFF2

On 9 July 2013 20:32, Ejzak, Richard P (Richard) <richard.ejzak@alcatel-luc=
ent.com<mailto:richard.ejzak@alcatel-lucent.com>> wrote:
Please note our proposed alternative to draft-pd-msrp-webrtc-00 for how to =
use DataChannel transport for MSRP.  Our draft does not place restrictions =
on the number of MSRP sessions that can be established within an SCTP assoc=
iation and describes alternatives for using a gateway to interconnect to ex=
isting MSRP endpoints.  The SDP extensions described are applicable to othe=
r potential DataChannel sub-protocols such as T.140, BFCP, T.38, etc.

Richard

-----Original Message-----
From: internet-drafts@ietf.org<mailto:internet-drafts@ietf.org> [mailto:int=
ernet-drafts@ietf.org<mailto:internet-drafts@ietf.org>]
Sent: Monday, July 08, 2013 6:31 PM
To: Ejzak, Richard P (Richard); MARCON, JEROME (JEROME)
Subject: New Version Notification for draft-marcon-msrp-over-webrtc-data-ch=
annels-00.txt


A new version of I-D, draft-marcon-msrp-over-webrtc-data-channels-00.txt
has been successfully submitted by Jerome Marcon and posted to the IETF rep=
ository.

Filename:        draft-marcon-msrp-over-webrtc-data-channels
Revision:        00
Title:           MSRP over WebRTC data channels
Creation date:   2013-07-09
Group:           Individual Submission
Number of pages: 12
URL:             http://www.ietf.org/internet-drafts/draft-marcon-msrp-over=
-webrtc-data-channels-00.txt
Status:          http://datatracker.ietf.org/doc/draft-marcon-msrp-over-web=
rtc-data-channels
Htmlized:        http://tools.ietf.org/html/draft-marcon-msrp-over-webrtc-d=
ata-channels-00


Abstract:
   The Real-Time Communication in WEB-browsers (RTCWeb) working group is
   charged to provide protocols to support direct interactive rich
   communication using audio, video, and data between two peers' web-
   browsers.  For the support of data communication, the RTCWeb working
   group has in particular defined the concept of bi-directional data
   channels over SCTP, where each data channel might be used to
   transport other protocols, called sub-protocols.  This document
   specifies how the Message Session Relay Protocol (MSRP) can be
   instantiated as a WebRTC data channel sub-protocol, using the SDP
   offer/exchange to negotiate out-of-band the sub-protocol specific
   parameters.  Two network configurations are documented: a WebRTC end-
   to-end configuration (connecting two MSRP over data channel
   endpoints), and a gateway configuration (connecting an MSRP over data
   channel endpoint with an MSRP over TCP endpoint).




The IETF Secretariat

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


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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 Gavin,<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thanks for your support o=
f our approach and for your comments.&nbsp; It&#8217;s great that you are p=
lanning an implementation along these lines.<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">I am including mmusic on =
this distribution to not exclude anyone who may be interested, but please l=
imit the distribution to dispatch only for subsequent emails
 in this thread.<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">Regarding why we chose to=
 create a new draft: Our approach was sufficiently different from yours tha=
t we felt a separate draft was a better idea in the short
 term.&nbsp; We can merge when it&#8217;s appropriate to do so.&nbsp; Even =
though we have no significant difference of opinion regarding the SDP exten=
sions needed, we clearly have different views regarding the value of a Data=
Channel gateway.&nbsp; It would be good to hear other
 opinions on this matter, as you are clearly heavily committed to your pref=
erence.<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">Our preference for suppor=
t of a DataChannel gateway model is based on the numerous advantages of Dat=
aChannels.&nbsp; Just a partial list: We have one transport mechanism
 for all cases; we don&#8217;t have to offer two options or perform fallbac=
k/retry procedures on failure of one transport option; we can easily multip=
lex all media and sub-protocols together; we can leverage the integrated br=
owser handling of multiple flows of different
 priority; we have to be able to interoperate with legacy tcp/tls endpoints=
 anyway.
<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">We look forward to contin=
uing the debate.<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">BR, Richard<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Gavin Ll=
ewellyn [mailto:gavin.llewellyn@crocodilertc.net]
<br>
<b>Sent:</b> Wednesday, July 10, 2013 6:51 AM<br>
<b>To:</b> Ejzak, Richard P (Richard)<br>
<b>Cc:</b> dispatch@ietf.org; mmusic@ietf.org WG<br>
<b>Subject:</b> Re: [dispatch] FW: New Version Notification for draft-marco=
n-msrp-over-webrtc-data-channels-00.txt<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">As a co-author of&nbsp;<span style=3D"font-size:10.0=
pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">draft-pd-msrp-webr=
tc-00, I support this new draft for establishment of MSRP sessions between =
WebRTC endpoints. &nbsp;The SDP enhancements allowing re-use of the
 SCTP association is very much along the lines of what we had in mind for t=
he next version of our draft, but we have not had time to complete it. &nbs=
p;We will likely be implementing this as a feature in our WebRTC library in=
 the near future, as the DataChannel
 implementations in browsers mature (support SCTP, reliable delivery, binar=
y data, etc).</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Having said that, it would have been appr=
eciated if the authors had contacted Peter and/or myself to discuss the exi=
sting draft and its limitations - we would have been happy
 to receive feedback and work together to find the best solution going forw=
ard.</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">I cannot see the value in introducing Dat=
aChannel gateways as a means to communicate with traditional TCP-based MSRP=
 endpoints. &nbsp;I believe this is better catered for by the
 WebSocket-based solution, as described in&nbsp;</span><span style=3D"font-=
family:&quot;Arial&quot;,&quot;sans-serif&quot;">draft-pd-dispatch-msrp-web=
socket, which already has client and server implementations, and is current=
ly in-use in a production environment. &nbsp;Given the (intentional)
 similarities between the WebSocket and DataChannel APIs, it is not difficu=
lt for the client application to support both transports. &nbsp;I would lik=
e to see a clear justification for this aspect of the draft.</span><o:p></o=
:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,&quot;s=
ans-serif&quot;">One minor correction: the opening paragraph of section 5.2=
.3 appears to reference the wrong SDP syntax for sub-protocol-specific attr=
ibutes; I believe it should reference the syntax for the
 new &quot;</span>wdcsa&quot; attribute.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Gavin Llewellyn<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal">--<o:p></o:p></p>
</div>
<p class=3D"MsoNormal">Gavin Llewellyn<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">Principal Design Engineer<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Crocodile RCS Ltd<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">GPG key: 0xF8F6FFF2<o:p></o:p></p>
</div>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On 9 July 2013 20:32, Ejzak, Richard P (Richard) &lt=
;<a href=3D"mailto:richard.ejzak@alcatel-lucent.com" target=3D"_blank">rich=
ard.ejzak@alcatel-lucent.com</a>&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal">Please note our proposed alternative to draft-pd-msr=
p-webrtc-00 for how to use DataChannel transport for MSRP. &nbsp;Our draft =
does not place restrictions on the number of MSRP sessions that can be esta=
blished within an SCTP association and
 describes alternatives for using a gateway to interconnect to existing MSR=
P endpoints. &nbsp;The SDP extensions described are applicable to other pot=
ential DataChannel sub-protocols such as T.140, BFCP, T.38, etc.<br>
<br>
Richard<br>
<br>
-----Original Message-----<br>
From: <a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">interne=
t-drafts@ietf.org</a> [mailto:<a href=3D"mailto:internet-drafts@ietf.org" t=
arget=3D"_blank">internet-drafts@ietf.org</a>]<br>
Sent: Monday, July 08, 2013 6:31 PM<br>
To: Ejzak, Richard P (Richard); MARCON, JEROME (JEROME)<br>
Subject: New Version Notification for draft-marcon-msrp-over-webrtc-data-ch=
annels-00.txt<br>
<br>
<br>
A new version of I-D, draft-marcon-msrp-over-webrtc-data-channels-00.txt<br=
>
has been successfully submitted by Jerome Marcon and posted to the IETF rep=
ository.<br>
<br>
Filename: &nbsp; &nbsp; &nbsp; &nbsp;draft-marcon-msrp-over-webrtc-data-cha=
nnels<br>
Revision: &nbsp; &nbsp; &nbsp; &nbsp;00<br>
Title: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; MSRP over WebRTC data channels<br=
>
Creation date: &nbsp; 2013-07-09<br>
Group: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Individual Submission<br>
Number of pages: 12<br>
URL: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; <a href=3D"http://www.ietf.o=
rg/internet-drafts/draft-marcon-msrp-over-webrtc-data-channels-00.txt" targ=
et=3D"_blank">
http://www.ietf.org/internet-drafts/draft-marcon-msrp-over-webrtc-data-chan=
nels-00.txt</a><br>
Status: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<a href=3D"http://datatracker.iet=
f.org/doc/draft-marcon-msrp-over-webrtc-data-channels" target=3D"_blank">ht=
tp://datatracker.ietf.org/doc/draft-marcon-msrp-over-webrtc-data-channels</=
a><br>
Htmlized: &nbsp; &nbsp; &nbsp; &nbsp;<a href=3D"http://tools.ietf.org/html/=
draft-marcon-msrp-over-webrtc-data-channels-00" target=3D"_blank">http://to=
ols.ietf.org/html/draft-marcon-msrp-over-webrtc-data-channels-00</a><br>
<br>
<br>
Abstract:<br>
&nbsp; &nbsp;The Real-Time Communication in WEB-browsers (RTCWeb) working g=
roup is<br>
&nbsp; &nbsp;charged to provide protocols to support direct interactive ric=
h<br>
&nbsp; &nbsp;communication using audio, video, and data between two peers' =
web-<br>
&nbsp; &nbsp;browsers. &nbsp;For the support of data communication, the RTC=
Web working<br>
&nbsp; &nbsp;group has in particular defined the concept of bi-directional =
data<br>
&nbsp; &nbsp;channels over SCTP, where each data channel might be used to<b=
r>
&nbsp; &nbsp;transport other protocols, called sub-protocols. &nbsp;This do=
cument<br>
&nbsp; &nbsp;specifies how the Message Session Relay Protocol (MSRP) can be=
<br>
&nbsp; &nbsp;instantiated as a WebRTC data channel sub-protocol, using the =
SDP<br>
&nbsp; &nbsp;offer/exchange to negotiate out-of-band the sub-protocol speci=
fic<br>
&nbsp; &nbsp;parameters. &nbsp;Two network configurations are documented: a=
 WebRTC end-<br>
&nbsp; &nbsp;to-end configuration (connecting two MSRP over data channel<br=
>
&nbsp; &nbsp;endpoints), and a gateway configuration (connecting an MSRP ov=
er data<br>
&nbsp; &nbsp;channel endpoint with an MSRP over TCP endpoint).<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<br>
<br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.org</a=
><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/dispatch</a><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_03FBA798AC24E3498B74F47FD082A92F3D802975US70UWXCHMBA04z_--

From martin.thomson@gmail.com  Wed Jul 10 09:50:20 2013
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To: Emil Ivov <emcho@jitsi.org>
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Subject: Re: [MMUSIC] Trickle ICE and frozen candidates
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On 9 July 2013 18:58, Emil Ivov <emcho@jitsi.org> wrote:
> OK. Do you mean something like 2. or do you have another mechanism in mind?

I don't have any particular mechanism in mind, though 2 might be
workable (I initially read "priority" there, which isn't quite right,
I think that you mean something like "preference order", since it's
the component ordering that would matter).

From gunnar.hellstrom@omnitor.se  Wed Jul 10 22:35:06 2013
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Subject: Re: [MMUSIC] Notes from Orlando human language draft discussion
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On 2013-07-10 06:49, Christian Groves wrote:
> I wouldn't have thought that it would be use case specific... i.e. i'd 
> only include my language when I called an emergency service. I would 
> think an endpoint would be configured with the information and then 
> provide it whenever I made a call. Is that the intent?
>
> Christian
Yes, that is the intent.

And for both relay service invocation and any other assisting service 
invocation, it is best for the user to dial the destination, and get the 
assisting service invoked by an indication like this in the call, rather 
than being forced to first call an assisting service and with them 
discuss who is the final destination of the call.

The other party in the call may be another user, a call center, an 
emergency service or a conference bridge.

Gunnar

>
> On 10/07/2013 9:23 AM, Gunnar Hellstrom wrote:
>> On 2013-07-09 23:58, Randall Gellens wrote:
>>> This is a different, and more complex, scenario than what this 
>>> proposal is trying to solve.  This proposal is focused on simpler 
>>> cases where someone places an emergency call or calls a corporate 
>>> call center.  Possibly this can be just a simple case of the more 
>>> complex cases? 
>> My view is that it is intended to also cover user-to-user cases, 
>> where preferences are compared and a relay service (or language 
>> interpreter) can be invoked to bridge language gaps.
>>
>> Gunnar
>>
>>
>> _______________________________________________
>> 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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Hello,

My preference for the WebSocket solution for client/server interaction
stems simply from:
1) There are many MSRP use cases that are strictly client to/from server.
WebSockets are a far simpler mechanism for these use cases, is far simpler
to implement at both ends, and is arguably more efficient due to this
simplicity.
2) There are many MSRP use cases that do not involve real-time media, and
use of WebSockets enables these use cases to be satisfied with browsers (or
other environments) that do not support WebRTC at this time (and some of
these browsers/environments may never support WebRTC).  I am not happy
about requiring WebRTC support in a browser just for me to be able to use
MSRP there.

The requirements for peer-to-peer MSRP and client/server MSRP are quite
different.  Even when TCP/TLS is the transport these requirements are
covered by two separate RFCs (which were developed and published in
parallel).  RFC 4976 adds additional MSRP requests and procedures that are
simply not needed for peer-to-peer MSRP.

I have no objection in principle to existence of a "gateway" device as you
have described (although, for the reasons described above, I will continue
to use WebSockets in my production system when client/server interaction is
required), but to me it would make more sense to separate this function
into another document that updates RFC 4976.

When I started writing draft-pd-msrp-webrtc I envisaged this as simply an
update to RFC 4975 to enable a new transport between MSRP peers - and I
believe there is value in this simplicity.

Regards,

Peter


On 10 July 2013 15:22, Ejzak, Richard P (Richard) <
richard.ejzak@alcatel-lucent.com> wrote:

>  Hi Gavin,****
>
> Thanks for your support of our approach and for your comments.  It=92s gr=
eat
> that you are planning an implementation along these lines.****
>
> ** **
>
> I am including mmusic on this distribution to not exclude anyone who may
> be interested, but please limit the distribution to dispatch only for
> subsequent emails in this thread.****
>
> ** **
>
> Regarding why we chose to create a new draft: Our approach was
> sufficiently different from yours that we felt a separate draft was a
> better idea in the short term.  We can merge when it=92s appropriate to d=
o
> so.  Even though we have no significant difference of opinion regarding t=
he
> SDP extensions needed, we clearly have different views regarding the valu=
e
> of a DataChannel gateway.  It would be good to hear other opinions on thi=
s
> matter, as you are clearly heavily committed to your preference.****
>
> ** **
>
> Our preference for support of a DataChannel gateway model is based on the
> numerous advantages of DataChannels.  Just a partial list: We have one
> transport mechanism for all cases; we don=92t have to offer two options o=
r
> perform fallback/retry procedures on failure of one transport option; we
> can easily multiplex all media and sub-protocols together; we can leverag=
e
> the integrated browser handling of multiple flows of different priority; =
we
> have to be able to interoperate with legacy tcp/tls endpoints anyway. ***=
*
>
> ** **
>
> We look forward to continuing the debate.****
>
> ** **
>
> BR, Richard****
>
> ** **
>
> *From:* Gavin Llewellyn [mailto:gavin.llewellyn@crocodilertc.net]
> *Sent:* Wednesday, July 10, 2013 6:51 AM
>
> *To:* Ejzak, Richard P (Richard)
> *Cc:* dispatch@ietf.org; mmusic@ietf.org WG
> *Subject:* Re: [dispatch] FW: New Version Notification for
> draft-marcon-msrp-over-webrtc-data-channels-00.txt****
>
>  ** **
>
> As a co-author of draft-pd-msrp-webrtc-00, I support this new draft for
> establishment of MSRP sessions between WebRTC endpoints.  The SDP
> enhancements allowing re-use of the SCTP association is very much along t=
he
> lines of what we had in mind for the next version of our draft, but we ha=
ve
> not had time to complete it.  We will likely be implementing this as a
> feature in our WebRTC library in the near future, as the DataChannel
> implementations in browsers mature (support SCTP, reliable delivery, bina=
ry
> data, etc).****
>
> ** **
>
> Having said that, it would have been appreciated if the authors had
> contacted Peter and/or myself to discuss the existing draft and its
> limitations - we would have been happy to receive feedback and work
> together to find the best solution going forward.****
>
> ** **
>
> I cannot see the value in introducing DataChannel gateways as a means to
> communicate with traditional TCP-based MSRP endpoints.  I believe this is
> better catered for by the WebSocket-based solution, as described in draft=
-pd-dispatch-msrp-websocket,
> which already has client and server implementations, and is currently
> in-use in a production environment.  Given the (intentional) similarities
> between the WebSocket and DataChannel APIs, it is not difficult for the
> client application to support both transports.  I would like to see a cle=
ar
> justification for this aspect of the draft.****
>
> ** **
>
> One minor correction: the opening paragraph of section 5.2.3 appears to
> reference the wrong SDP syntax for sub-protocol-specific attributes; I
> believe it should reference the syntax for the new "wdcsa" attribute.****
>
> ** **
>
> Regards,****
>
> Gavin Llewellyn****
>
> ** **
>
> --****
>
> Gavin Llewellyn****
>
> Principal Design Engineer****
>
> Crocodile RCS Ltd****
>
> GPG key: 0xF8F6FFF2****
>
> ** **
>
> On 9 July 2013 20:32, Ejzak, Richard P (Richard) <
> richard.ejzak@alcatel-lucent.com> wrote:****
>
> Please note our proposed alternative to draft-pd-msrp-webrtc-00 for how t=
o
> use DataChannel transport for MSRP.  Our draft does not place restriction=
s
> on the number of MSRP sessions that can be established within an SCTP
> association and describes alternatives for using a gateway to interconnec=
t
> to existing MSRP endpoints.  The SDP extensions described are applicable =
to
> other potential DataChannel sub-protocols such as T.140, BFCP, T.38, etc.
>
> Richard
>
> -----Original Message-----
> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
> Sent: Monday, July 08, 2013 6:31 PM
> To: Ejzak, Richard P (Richard); MARCON, JEROME (JEROME)
> Subject: New Version Notification for
> draft-marcon-msrp-over-webrtc-data-channels-00.txt
>
>
> A new version of I-D, draft-marcon-msrp-over-webrtc-data-channels-00.txt
> has been successfully submitted by Jerome Marcon and posted to the IETF
> repository.
>
> Filename:        draft-marcon-msrp-over-webrtc-data-channels
> Revision:        00
> Title:           MSRP over WebRTC data channels
> Creation date:   2013-07-09
> Group:           Individual Submission
> Number of pages: 12
> URL:
> http://www.ietf.org/internet-drafts/draft-marcon-msrp-over-webrtc-data-ch=
annels-00.txt
> Status:
> http://datatracker.ietf.org/doc/draft-marcon-msrp-over-webrtc-data-channe=
ls
> Htmlized:
> http://tools.ietf.org/html/draft-marcon-msrp-over-webrtc-data-channels-00
>
>
> Abstract:
>    The Real-Time Communication in WEB-browsers (RTCWeb) working group is
>    charged to provide protocols to support direct interactive rich
>    communication using audio, video, and data between two peers' web-
>    browsers.  For the support of data communication, the RTCWeb working
>    group has in particular defined the concept of bi-directional data
>    channels over SCTP, where each data channel might be used to
>    transport other protocols, called sub-protocols.  This document
>    specifies how the Message Session Relay Protocol (MSRP) can be
>    instantiated as a WebRTC data channel sub-protocol, using the SDP
>    offer/exchange to negotiate out-of-band the sub-protocol specific
>    parameters.  Two network configurations are documented: a WebRTC end-
>    to-end configuration (connecting two MSRP over data channel
>    endpoints), and a gateway configuration (connecting an MSRP over data
>    channel endpoint with an MSRP over TCP endpoint).
>
>
>
>
> The IETF Secretariat
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch****
>
> ** **
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
>


--=20
Peter Dunkley
Technical Director
Crocodile RCS Ltd

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<div dir=3D"ltr"><div style=3D"font-family:arial,sans-serif;font-size:13px"=
><div>Hello,</div><div><br></div>My preference for the WebSocket solution f=
or client/server interaction stems simply from:<div>1) There are many MSRP =
use cases that are strictly client to/from server. WebSockets are a far sim=
pler mechanism for these use cases, is far simpler to implement at both end=
s, and is arguably more efficient due to this simplicity.</div>
<div>2) There are many MSRP use cases that do not involve real-time media, =
and use of WebSockets enables these use cases to be satisfied with browsers=
 (or other environments) that do not support WebRTC at this time (and some =
of these browsers/environments may never support WebRTC). =A0I am not happy=
 about requiring WebRTC support in a browser just for me to be able to use =
MSRP there.</div>
<div><br></div><div>The requirements for peer-to-peer MSRP and client/serve=
r MSRP are quite different. =A0Even when TCP/TLS is the transport these req=
uirements are covered by two separate RFCs (which were developed and publis=
hed in parallel). =A0RFC 4976 adds additional MSRP requests and procedures =
that are simply not needed for peer-to-peer MSRP.</div>
<div><br></div><div>I have no objection in principle to existence of a &quo=
t;gateway&quot; device as you have described (although, for the reasons des=
cribed above, I will continue to use WebSockets in my production system whe=
n client/server interaction is required), but to me it would make more sens=
e to separate this function into another document that updates RFC 4976.</d=
iv>
<div><br></div><div>When I started writing draft-pd-msrp-webrtc I envisaged=
 this as simply an update to RFC 4975 to enable a new transport between MSR=
P peers - and I believe there is value in this simplicity.</div></div><div =
style=3D"font-family:arial,sans-serif;font-size:13px">
<br></div><div style=3D"font-family:arial,sans-serif;font-size:13px">Regard=
s,</div><div style=3D"font-family:arial,sans-serif;font-size:13px"><br></di=
v><div style=3D"font-family:arial,sans-serif;font-size:13px">Peter</div></d=
iv>
<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On 10 July 20=
13 15:22, Ejzak, Richard P (Richard) <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:richard.ejzak@alcatel-lucent.com" target=3D"_blank">richard.ejzak@alcatel=
-lucent.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi Gavin,<u></u><u></u></=
span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Thanks for your support o=
f our approach and for your comments.=A0 It=92s great that you are planning=
 an implementation along these lines.<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">I am including mmusic on =
this distribution to not exclude anyone who may be interested, but please l=
imit the distribution to dispatch only for subsequent emails
 in this thread.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Regarding why we chose to=
 create a new draft: Our approach was sufficiently different from yours tha=
t we felt a separate draft was a better idea in the short
 term.=A0 We can merge when it=92s appropriate to do so.=A0 Even though we =
have no significant difference of opinion regarding the SDP extensions need=
ed, we clearly have different views regarding the value of a DataChannel ga=
teway.=A0 It would be good to hear other
 opinions on this matter, as you are clearly heavily committed to your pref=
erence.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Our preference for suppor=
t of a DataChannel gateway model is based on the numerous advantages of Dat=
aChannels.=A0 Just a partial list: We have one transport mechanism
 for all cases; we don=92t have to offer two options or perform fallback/re=
try procedures on failure of one transport option; we can easily multiplex =
all media and sub-protocols together; we can leverage the integrated browse=
r handling of multiple flows of different
 priority; we have to be able to interoperate with legacy tcp/tls endpoints=
 anyway.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">We look forward to contin=
uing the debate.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">BR, Richard<u></u><u></u>=
</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Gavin Ll=
ewellyn [mailto:<a href=3D"mailto:gavin.llewellyn@crocodilertc.net" target=
=3D"_blank">gavin.llewellyn@crocodilertc.net</a>]
<br>
<b>Sent:</b> Wednesday, July 10, 2013 6:51 AM</span></p><div class=3D"im"><=
br>
<b>To:</b> Ejzak, Richard P (Richard)<br>
</div><b>Cc:</b> <a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dis=
patch@ietf.org</a>; <a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mm=
usic@ietf.org</a> WG<br>
<b>Subject:</b> Re: [dispatch] FW: New Version Notification for draft-marco=
n-msrp-over-webrtc-data-channels-00.txt<u></u><u></u><p></p>
</div>
</div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<div>
<p class=3D"MsoNormal">As a co-author of=A0<span style=3D"font-size:10.0pt;=
font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">draft-pd-msrp-webrtc-=
00, I support this new draft for establishment of MSRP sessions between Web=
RTC endpoints. =A0The SDP enhancements allowing re-use of the
 SCTP association is very much along the lines of what we had in mind for t=
he next version of our draft, but we have not had time to complete it. =A0W=
e will likely be implementing this as a feature in our WebRTC library in th=
e near future, as the DataChannel
 implementations in browsers mature (support SCTP, reliable delivery, binar=
y data, etc).</span><u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">Having said that, it would have been appr=
eciated if the authors had contacted Peter and/or myself to discuss the exi=
sting draft and its limitations - we would have been happy
 to receive feedback and work together to find the best solution going forw=
ard.</span><u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">I cannot see the value in introducing Dat=
aChannel gateways as a means to communicate with traditional TCP-based MSRP=
 endpoints. =A0I believe this is better catered for by the
 WebSocket-based solution, as described in=A0</span><span style=3D"font-fam=
ily:&quot;Arial&quot;,&quot;sans-serif&quot;">draft-pd-dispatch-msrp-websoc=
ket, which already has client and server implementations, and is currently =
in-use in a production environment. =A0Given the (intentional)
 similarities between the WebSocket and DataChannel APIs, it is not difficu=
lt for the client application to support both transports. =A0I would like t=
o see a clear justification for this aspect of the draft.</span><u></u><u><=
/u></p>

</div>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,&quot;s=
ans-serif&quot;">One minor correction: the opening paragraph of section 5.2=
.3 appears to reference the wrong SDP syntax for sub-protocol-specific attr=
ibutes; I believe it should reference the syntax for the
 new &quot;</span>wdcsa&quot; attribute.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Regards,<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Gavin Llewellyn<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal">--<u></u><u></u></p>
</div>
<p class=3D"MsoNormal">Gavin Llewellyn<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">Principal Design Engineer<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Crocodile RCS Ltd<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">GPG key: 0xF8F6FFF2<u></u><u></u></p>
</div>
</div>
</div><div><div class=3D"h5">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On 9 July 2013 20:32, Ejzak, Richard P (Richard) &lt=
;<a href=3D"mailto:richard.ejzak@alcatel-lucent.com" target=3D"_blank">rich=
ard.ejzak@alcatel-lucent.com</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">Please note our proposed alternative to draft-pd-msr=
p-webrtc-00 for how to use DataChannel transport for MSRP. =A0Our draft doe=
s not place restrictions on the number of MSRP sessions that can be establi=
shed within an SCTP association and
 describes alternatives for using a gateway to interconnect to existing MSR=
P endpoints. =A0The SDP extensions described are applicable to other potent=
ial DataChannel sub-protocols such as T.140, BFCP, T.38, etc.<br>
<br>
Richard<br>
<br>
-----Original Message-----<br>
From: <a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">interne=
t-drafts@ietf.org</a> [mailto:<a href=3D"mailto:internet-drafts@ietf.org" t=
arget=3D"_blank">internet-drafts@ietf.org</a>]<br>
Sent: Monday, July 08, 2013 6:31 PM<br>
To: Ejzak, Richard P (Richard); MARCON, JEROME (JEROME)<br>
Subject: New Version Notification for draft-marcon-msrp-over-webrtc-data-ch=
annels-00.txt<br>
<br>
<br>
A new version of I-D, draft-marcon-msrp-over-webrtc-data-channels-00.txt<br=
>
has been successfully submitted by Jerome Marcon and posted to the IETF rep=
ository.<br>
<br>
Filename: =A0 =A0 =A0 =A0draft-marcon-msrp-over-webrtc-data-channels<br>
Revision: =A0 =A0 =A0 =A000<br>
Title: =A0 =A0 =A0 =A0 =A0 MSRP over WebRTC data channels<br>
Creation date: =A0 2013-07-09<br>
Group: =A0 =A0 =A0 =A0 =A0 Individual Submission<br>
Number of pages: 12<br>
URL: =A0 =A0 =A0 =A0 =A0 =A0 <a href=3D"http://www.ietf.org/internet-drafts=
/draft-marcon-msrp-over-webrtc-data-channels-00.txt" target=3D"_blank">
http://www.ietf.org/internet-drafts/draft-marcon-msrp-over-webrtc-data-chan=
nels-00.txt</a><br>
Status: =A0 =A0 =A0 =A0 =A0<a href=3D"http://datatracker.ietf.org/doc/draft=
-marcon-msrp-over-webrtc-data-channels" target=3D"_blank">http://datatracke=
r.ietf.org/doc/draft-marcon-msrp-over-webrtc-data-channels</a><br>
Htmlized: =A0 =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-marcon=
-msrp-over-webrtc-data-channels-00" target=3D"_blank">http://tools.ietf.org=
/html/draft-marcon-msrp-over-webrtc-data-channels-00</a><br>
<br>
<br>
Abstract:<br>
=A0 =A0The Real-Time Communication in WEB-browsers (RTCWeb) working group i=
s<br>
=A0 =A0charged to provide protocols to support direct interactive rich<br>
=A0 =A0communication using audio, video, and data between two peers&#39; we=
b-<br>
=A0 =A0browsers. =A0For the support of data communication, the RTCWeb worki=
ng<br>
=A0 =A0group has in particular defined the concept of bi-directional data<b=
r>
=A0 =A0channels over SCTP, where each data channel might be used to<br>
=A0 =A0transport other protocols, called sub-protocols. =A0This document<br=
>
=A0 =A0specifies how the Message Session Relay Protocol (MSRP) can be<br>
=A0 =A0instantiated as a WebRTC data channel sub-protocol, using the SDP<br=
>
=A0 =A0offer/exchange to negotiate out-of-band the sub-protocol specific<br=
>
=A0 =A0parameters. =A0Two network configurations are documented: a WebRTC e=
nd-<br>
=A0 =A0to-end configuration (connecting two MSRP over data channel<br>
=A0 =A0endpoints), and a gateway configuration (connecting an MSRP over dat=
a<br>
=A0 =A0channel endpoint with an MSRP over TCP endpoint).<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<br>
<br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ietf.org</a=
><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/dispatch</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div></div></div>
</div>
</div>
</div>
</div>

<br>_______________________________________________<br>
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<a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
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">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
<br></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><div dir=
=3D"ltr"><div><font face=3D"courier new, monospace">Peter Dunkley</font></d=
iv><div><font face=3D"courier new, monospace">Technical Director</font></di=
v><div>
<font face=3D"courier new, monospace">Crocodile RCS Ltd</font></div></div>
</div>

--089e013cc1c4578b7304e1297d42--

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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-duplication-grouping-03.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           : Duplication Grouping Semantics in the Session Descriptio=
n Protocol
	Author(s)       : Ali Begen
                          Yiqun Cai
                          Heidi Ou
	Filename        : draft-ietf-mmusic-duplication-grouping-03.txt
	Pages           : 9
	Date            : 2013-07-11

Abstract:
   Packet loss is undesirable for real-time multimedia sessions, but can
   occur due to congestion, or other unplanned network outages.  This is
   especially true for IP multicast networks, where packet loss patterns
   can vary greatly between receivers.  One technique that can be used
   to recover from packet loss without incurring unbounded delay for all
   the receivers is to duplicate the packets and send them in separate
   redundant streams.  This document defines the semantics for grouping
   redundant streams in the Session Description Protocol (SDP).  The
   semantics defined in this document are to be used with the SDP
   Grouping Framework.  SSRC-level (Synchronization Source) grouping
   semantics are also defined in this document for RTP streams using
   SSRC multiplexing.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-duplication-grouping-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-duplication-grouping-03


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


From pkyzivat@alum.mit.edu  Thu Jul 11 11:36:24 2013
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Subject: Re: [MMUSIC] Notes from Orlando human language draft discussion
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On 7/11/13 1:34 AM, Gunnar Hellstrom wrote:
> On 2013-07-10 06:49, Christian Groves wrote:
>> I wouldn't have thought that it would be use case specific... i.e. i'd
>> only include my language when I called an emergency service. I would
>> think an endpoint would be configured with the information and then
>> provide it whenever I made a call. Is that the intent?
>>
>> Christian
> Yes, that is the intent.
>
> And for both relay service invocation and any other assisting service
> invocation, it is best for the user to dial the destination, and get the
> assisting service invoked by an indication like this in the call, rather
> than being forced to first call an assisting service and with them
> discuss who is the final destination of the call.

Viewed this way, it is similar to recognizing the need for, and 
recruiting, a transcoder. (See RFCs 5369 & 5370.)

	Thanks,
	Paul

> The other party in the call may be another user, a call center, an
> emergency service or a conference bridge.
>
> Gunnar
>
>>
>> On 10/07/2013 9:23 AM, Gunnar Hellstrom wrote:
>>> On 2013-07-09 23:58, Randall Gellens wrote:
>>>> This is a different, and more complex, scenario than what this
>>>> proposal is trying to solve.  This proposal is focused on simpler
>>>> cases where someone places an emergency call or calls a corporate
>>>> call center.  Possibly this can be just a simple case of the more
>>>> complex cases?
>>> My view is that it is intended to also cover user-to-user cases,
>>> where preferences are compared and a relay service (or language
>>> interpreter) can be invoked to bridge language gaps.
>>>
>>> Gunnar
>>>
>>>
>>> _______________________________________________
>>> 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] Notes from Orlando human language draft discussion
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On 2013-07-11 20:36, Paul Kyzivat wrote:
>>
>>
>> And for both relay service invocation and any other assisting service
>> invocation, it is best for the user to dial the destination, and get the
>> assisting service invoked by an indication like this in the call, rather
>> than being forced to first call an assisting service and with them
>> discuss who is the final destination of the call.
>
> Viewed this way, it is similar to recognizing the need for, and 
> recruiting, a transcoder. (See RFCs 5369 & 5370.)
Yes, that is a useful model. Especially the bridge, because there you 
can let all three parties take part of common media when that makes sense.

/Gunnar

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Cc: MMUSIC IETF WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] Trickle ICE and frozen candidates
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Hey Martin, all,

On 10.07.13, 18:50, Martin Thomson wrote:
> On 9 July 2013 18:58, Emil Ivov <emcho@jitsi.org> wrote:
>> OK. Do you mean something like 2. or do you have another mechanism in mind?
>
> I don't have any particular mechanism in mind, though 2 might be
> workable (I initially read "priority" there, which isn't quite right,
> I think that you mean something like "preference order", since it's
> the component ordering that would matter).

Yes and the idea was that component ordering is generally the only thing 
that would influence "priority" within a single foundation. That said, I 
agree there's no reason not to say this directly.

So to summarize. The algorithm would look like this:

We start by unfreezing the first available pair for a foundation. Any 
new pairs would be:

a) automatically unfrozen if they belong to a component that precedes 
that of an unfrozen pair.

b) automatically unfrozen if other checks in that foundation already 
succeeded

c) kept frozen and then handled by regular 5245 unfreezing if they 
belong to components that come after the last currently unfrozen but not 
yet completed check

d) kept frozen if checks for unfrozen pairs in this same foundation have 
already failed.

Obviously "d" is kind of murky. We need it in order for the freezing 
algorithm to have any sense but it could still lead to race conditions 
and premature failures.

So, at least now this doesn't seem like a great way to go forward.

Therefore, it seems to me that the only way to keep "freezing" is to 
mandate that candidates be trickled by component order. This way we 
would be able to pretty much reuse the vanilla ICE algorithm.

Thoughts?

Emil


-- 
https://jitsi.org

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At 2:49 PM +1000 7/10/13, Christian Groves wrote:

>  I wouldn't have thought that it would be use case specific... i.e. 
> i'd only include my language when I called an emergency service. I 
> would think an endpoint would be configured with the information 
> and then provide it whenever I made a call. Is that the intent?

I expect so, but then, I also didn't expect it to be used much in the 
person-to-person case since usually language is implied by context 
when calling an individual.  I expect it to be used when a person is 
calling an call center (corporate or emergency).  To put it another 
way, I expect a personal client to always set language on outbound 
calls, but in most cases to ignore it on inbound calls.  I'd expect a 
corporate call center, as well as emergency call handling facilities, 
to pay attention.  I should make this more clear in the draft.

-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly selected tag: ---------------
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From gunnar.hellstrom@omnitor.se  Sat Jul 13 11:04:55 2013
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On 2013-07-13 10:07, Randall Gellens wrote:
> At 2:49 PM +1000 7/10/13, Christian Groves wrote:
>
>>  I wouldn't have thought that it would be use case specific... i.e. 
>> i'd only include my language when I called an emergency service. I 
>> would think an endpoint would be configured with the information and 
>> then provide it whenever I made a call. Is that the intent?
>
> I expect so, but then, I also didn't expect it to be used much in the 
> person-to-person case since usually language is implied by context 
> when calling an individual.  I expect it to be used when a person is 
> calling an call center (corporate or emergency).  To put it another 
> way, I expect a personal client to always set language on outbound 
> calls, but in most cases to ignore it on inbound calls.  I'd expect a 
> corporate call center, as well as emergency call handling facilities, 
> to pay attention.  I should make this more clear in the draft.
>
I expect the setting to be retrieved from a personal profile (likely but 
not necessarily stored in connection with other user information and 
account information) and used in every call.

/Gunnar

From rg+ietf@qualcomm.com  Sun Jul 14 02:06:13 2013
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Subject: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-01.txt
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Hi all,

I've tweaked the human language draft:

    o  Relaxed language on setting -send and -receive to same values;
       added text on leaving on empty to indicate asymmetric usage.

    o  Added text that clients on behalf of end users are expected to set
       the attributes on outgoing calls and ignore on incoming calls
       while systems on behalf of call centers and PSAPs are expected to
       take the attributes into account when processing incoming calls.

-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly selected tag: ---------------
The mind of a bigot is like the pupil of the eye. The more light you
shine on it, the more it contracts.      --Oliver Wendell Holmes, Jr.

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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-traffic-class-for-sdp-04.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 Session Description Protocol (SDP) 'trafficclass' At=
tribute
	Author(s)       : James Polk
                          Subha Dhesikan
                          Paul E. Jones
	Filename        : draft-ietf-mmusic-traffic-class-for-sdp-04.txt
	Pages           : 28
	Date            : 2013-07-14

Abstract:
   This document proposes a new Session Description Protocol (SDP)
   attribute to identify the traffic class a session is requesting
   in its offer/answer exchange.




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

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

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-traffic-class-for-sdp-=
04


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


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Subject: [MMUSIC] Updated the draft-ietf-mmusic-traffic-class-for-sdp draft
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MMUSIC

I have updated the trafficclass label attribute draft per the WG's 
feedback out of the Orlando meeting. I don't believe there is 
anything controversial remaining in this draft in its present state, 
and I consider the draft done - barring additional feedback.

James

++++++++++
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 Session Description Protocol (SDP) 
'trafficclass' Attribute
	Author(s)       : James Polk
                           Subha Dhesikan
                           Paul E. Jones
	Filename        : draft-ietf-mmusic-traffic-class-for-sdp-04.txt
	Pages           : 28
	Date            : 2013-07-14

Abstract:
    This document proposes a new Session Description Protocol (SDP)
    attribute to identify the traffic class a session is requesting
    in its offer/answer exchange.

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

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

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


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


From gunnar.hellstrom@omnitor.se  Sun Jul 14 15:15:26 2013
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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-01.txt
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Randall,

Yes, version 01 is a step forward.


Comments:
1. In 7.1 you now say:

When placing an emergency call, and in any other case where the
    language cannot be assumed from context, each media stream in an
    offer SHOULD specify one or both 'humintlang-send' and 'humintlang-
    recv' attributes (to avoid ambiguity).


This might be a good advice. But to what extent SHOULD it be followed?

Some media might be included in SDP without the user having any interest 
of using it for language communication.
Video may be included without the person really preferring it for spoken 
language understanding enhancement but just for pleasure or seeing the 
scene of the other user.

Do you mean that such cases should also be specified with humintlang but 
with empty attributes?
As a clear indication that the user is not interested in using that 
medium for language transfer.
That could make sense. That is different from not providing any 
humintlang at all, that should be understood as undefined language 
preferences.


But there are media streams specified by SDP where a language preference 
specification does not make sense. There are many application and 
message media specifications that are not used for any human language 
communication. I think for example that there are media specifications 
for camera control.
Is the SHOULD a sufficient weakening of the requirement, so that the 
reader can skip using humintlang for media where it does not make sense?
If not, expand "each media stream"  to: "each media stream in which 
human language communication might occur".



2. You have also introduced this paragraph in 7.1:
"

Clients acting on behalf of end users are expected to set one or both
    'humintlang-send' and 'humintlang-recv' attributes on each media
    stream in an offer when placing an outgoing session but ignore the
    attributes when receiving incoming calls.  Systems acting on behalf
    of call centers and PSAPs are expected to take into account the
    values when processing inbound calls.
"

What is the thought behind this?

I see the humintlang specification equally valuable in outgoing as incoming calls, and for end-users as well as call-centers and PSAPs.
If a user gets an incoming call, with humintlang specifications, it can compare with its own profile and decide to invoke a
relay service or interpreter servicde to cover some differences in language preferences, or just prepare for what media, modalities and
languages to use in the call..
I do not see the background for this paragraph, and think it will be better to delete it.


3. Absolute preference is still needed.
I described earlier the need for an absolute preference indication. But I did not provide any coding proposal.
I suggest to add something at the end of the language tag. Regardless if it gets before or after the * if it is there.
Can a "+" be used to indicate a strongly preferred language and modality, no extra specification a managed language and modality
and a "-" a non-preferred, but possible language and modality.
  
( needs to be checked against syntax of language tags.)

  
Regards

Gunnar

------------------------------------------------------------------------
Gunnar Hellström
Omnitor
gunnar.hellstrom@omnitor.se
+46708204288
On 2013-07-14 11:05, Randall Gellens wrote:
> Hi all,
>
> I've tweaked the human language draft:
>
>    o  Relaxed language on setting -send and -receive to same values;
>       added text on leaving on empty to indicate asymmetric usage.
>
>    o  Added text that clients on behalf of end users are expected to set
>       the attributes on outgoing calls and ignore on incoming calls
>       while systems on behalf of call centers and PSAPs are expected to
>       take the attributes into account when processing incoming calls.
>


--------------060203030902050504080103
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: 7bit

<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    <div class="moz-cite-prefix">Randall,<br>
      <br>
      Yes, version 01 is a step forward.<br>
      <br>
      <br>
      Comments:<br>
      1. In 7.1 you now say:<br>
      <pre style="font-family: monospace; line-height: 1.2em; margin: 0px; color: rgb(0, 0, 0); font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">When placing an emergency call, and in any other case where the
   language cannot be assumed from context, each media stream in an
   offer SHOULD specify one or both 'humintlang-send' and 'humintlang-
   recv' attributes (to avoid ambiguity).</pre>
      <br>
      This might be a good advice. But to what extent SHOULD it be
      followed?<br>
      <br>
      Some media might be included in SDP without the user having any
      interest of using it for language communication. <br>
      Video may be included without the person really preferring it for
      spoken language understanding enhancement but just for pleasure or
      seeing the scene of the other user.<br>
      <br>
      Do you mean that such cases should also be specified with
      humintlang but with empty attributes?<br>
      As a clear indication that the user is not interested in using
      that medium for language transfer. <br>
      That could make sense. That is different from not providing any
      humintlang at all, that should be understood as undefined language
      preferences.<br>
      <br>
      <br>
      But there are media streams specified by SDP where a language
      preference specification does not make sense. There are many
      application and message media specifications that are not used for
      any human language communication. I think for example that there
      are media specifications for camera control. <br>
      Is the SHOULD a sufficient weakening of the requirement, so that
      the reader can skip using humintlang for media where it does not
      make sense?<br>
      If not, expand "each media stream"&nbsp; to: "each media stream in
      which human language communication might occur".<br>
      <br>
      <br>
      <br>
      2. You have also introduced this paragraph in 7.1:<br>
      "<br>
      <pre style="font-family: monospace; line-height: 1.2em; margin: 0px; color: rgb(0, 0, 0); font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">Clients acting on behalf of end users are expected to set one or both
   'humintlang-send' and 'humintlang-recv' attributes on each media
   stream in an offer when placing an outgoing session but ignore the
   attributes when receiving incoming calls.  Systems acting on behalf
   of call centers and PSAPs are expected to take into account the
   values when processing inbound calls.
"

What is the thought behind this?

I see the humintlang specification equally valuable in outgoing as incoming calls, and for end-users as well as call-centers and PSAPs.
If a user gets an incoming call, with humintlang specifications, it can compare with its own profile and decide to invoke a 
relay service or interpreter servicde to cover some differences in language preferences, or just prepare for what media, modalities and 
languages to use in the call..
I do not see the background for this paragraph, and think it will be better to delete it.


3. Absolute preference is still needed.
I described earlier the need for an absolute preference indication. But I did not provide any coding proposal.
I suggest to add something at the end of the language tag. Regardless if it gets before or after the * if it is there.
Can a "+" be used to indicate a strongly preferred language and modality, no extra specification a managed language and modality 
and a "-" a non-preferred, but possible language and modality.
&nbsp;
( needs to be checked against syntax of language tags.)

&nbsp;
Regards

Gunnar
</pre>
      <div class="moz-signature">
        <hr>
        Gunnar Hellstr&ouml;m<br>
        Omnitor<br>
        <a class="moz-txt-link-abbreviated" href="mailto:gunnar.hellstrom@omnitor.se">gunnar.hellstrom@omnitor.se</a><br>
        +46708204288<br>
      </div>
      On 2013-07-14 11:05, Randall Gellens wrote:<br>
    </div>
    <blockquote cite="mid:p06240602ce081a3d3f8c@%5B10.121.65.183%5D"
      type="cite">Hi all,
      <br>
      <br>
      I've tweaked the human language draft:
      <br>
      <br>
      &nbsp;&nbsp; o&nbsp; Relaxed language on setting -send and -receive to same
      values;
      <br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; added text on leaving on empty to indicate asymmetric usage.
      <br>
      <br>
      &nbsp;&nbsp; o&nbsp; Added text that clients on behalf of end users are expected
      to set
      <br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the attributes on outgoing calls and ignore on incoming
      calls
      <br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; while systems on behalf of call centers and PSAPs are
      expected to
      <br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; take the attributes into account when processing incoming
      calls.
      <br>
      <br>
    </blockquote>
    <br>
  </body>
</html>

--------------060203030902050504080103--

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Subject: Re: [MMUSIC] Notes from Orlando human language draft discussion
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Hello Randall and Gunnar,

I think Randall's clarification would be good to have in the draft. I 
think that when a caller support the language it is sent for every call. 
Whether the language is used by a receiver depends on the receiver's 
local capability.


Regards, Christian


On 14/07/2013 4:04 AM, Gunnar Hellstrom wrote:
> On 2013-07-13 10:07, Randall Gellens wrote:
>> At 2:49 PM +1000 7/10/13, Christian Groves wrote:
>>
>>>  I wouldn't have thought that it would be use case specific... i.e. 
>>> i'd only include my language when I called an emergency service. I 
>>> would think an endpoint would be configured with the information and 
>>> then provide it whenever I made a call. Is that the intent?
>>
>> I expect so, but then, I also didn't expect it to be used much in the 
>> person-to-person case since usually language is implied by context 
>> when calling an individual.  I expect it to be used when a person is 
>> calling an call center (corporate or emergency).  To put it another 
>> way, I expect a personal client to always set language on outbound 
>> calls, but in most cases to ignore it on inbound calls.  I'd expect a 
>> corporate call center, as well as emergency call handling facilities, 
>> to pay attention.  I should make this more clear in the draft.
>>
> I expect the setting to be retrieved from a personal profile (likely 
> but not necessarily stored in connection with other user information 
> and account information) and used in every call.
>
> /Gunnar
> _______________________________________________
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From Christian.Groves@nteczone.com  Sun Jul 14 19:20:58 2013
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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-01.txt
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Hello Randall,

Please see my comment below.

..snip..
On 15/07/2013 8:14 AM, Gunnar Hellstrom wrote:
> ...
>
> 2. You have also introduced this paragraph in 7.1:
> "
> Clients acting on behalf of end users are expected to set one or both
>     'humintlang-send' and 'humintlang-recv' attributes on each media
>     stream in an offer when placing an outgoing session but ignore the
>     attributes when receiving incoming calls.  Systems acting on behalf
>     of call centers and PSAPs are expected to take into account the
>     values when processing inbound calls.
> "
>
> What is the thought behind this?
>
> I see the humintlang specification equally valuable in outgoing as incoming calls, and for end-users as well as call-centers and PSAPs.
> If a user gets an incoming call, with humintlang specifications, it can compare with its own profile and decide to invoke a
> relay service or interpreter servicde to cover some differences in language preferences, or just prepare for what media, modalities and
> languages to use in the call..
> I do not see the background for this paragraph, and think it will be better to delete it.
>
>
>
[CNG] I guess this is a question of scope. I would think that the draft 
would just describe the syntax and procedures of this new language tag 
in a generic manner. The application level behaviour of call centres and 
PSAPs would be defined elsewhere?

Regards, Christian

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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 G723 Annex A and G729 An=
nex B
	Author(s)       : Muthu Arul Mozhi Perumal
                          Parthasarathi Ravindran
	Filename        : draft-ietf-mmusic-sdp-g723-g729-04.txt
	Pages           : 8
	Date            : 2013-07-14

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 Session Description
   protocol(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-g729-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-g723-g729-04


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


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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
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This updated version addresses all comments received during WGLC.

- Muthu, Partha

-----Original Message-----
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On Behalf Of internet-drafts@ietf.org
Sent: Monday, July 15, 2013 10:13 AM
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Subject: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.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           : Offer/Answer Considerations for G723 Annex A and G729 An=
nex B
	Author(s)       : Muthu Arul Mozhi Perumal
                          Parthasarathi Ravindran
	Filename        : draft-ietf-mmusic-sdp-g723-g729-04.txt
	Pages           : 8
	Date            : 2013-07-14

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 Session Description
   protocol(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-g729-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-g723-g729-04

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Subject: Re: [MMUSIC] Review Request - draft-nandakumar-mmusic-sdp-mux-attributes-02 - (RFC3407 Related)
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Section 5.20 (RFC 3407), refers to the following attributes with their 
category shown in brackets (note: changed "csdc" to "cdsc"):

a=sqn (NORMAL)
a=cdsc (TBD)
a=cpar (TBD)
a=cparmin (TBD)
a=cparmax (TBD)

I agree with the "sqn" category of NORMAL.

For the rest of them, the challenge we have is that they are essentially 
what we could call "encapsulating" attributes, i.e. they simply encode 
attributes or other parameters. This leads to a problem that may apply 
to other attributes as well: Currently, the mux mechanism seeems to only 
distinguish attributes at the attribute name level, however there are 
attributes where you have the same attribute-name appear many times, but 
further parameters in the attribute distinguish these (and as per the 
above, this distinction may in turn lead to different categories). Also 
note, that these attributes to not actually negotiate use of their 
encapsulated values; a separate signaling mechanism, e.g. O/A exchange,  
would be needed for this (as noted in RFC 3407, Section 2).

For "cdsc", I'd be inclined to put it in category "NORMAL", but for the 
rest of them, I'm not sure any of the current categories match, since it 
really depends on what was encapsulated in them (note that both bandwith 
and attributes can be encapsulated in them).

Thanks

-- Flemming


On 4/26/13 5:24 PM, Suhas Nandakumar wrote:
> Hello Flemming
> This is a request for reviewing the following sections
> 5.20
>
> of the draft
> draft-nandakumar-mmusic-sdp-mux-attributes-02
>
> It will be greatly appreciated if you could inform on your willingness 
> for reviewing the aforementioned sessions.
>
> It would be great to have your review feedback/comments/inputs before 
> June 9th 2013.
>
> The document is available here:
> http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-attributes-02
>
>
> Thanks in Advance
>
> Regards
> Suhas Nandakumar


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    Section 5.20 (RFC 3407), refers to the following attributes with
    their category shown in brackets (note: changed "csdc" to "cdsc"):<br>
    <br>
    a=sqn (NORMAL)<br>
    a=cdsc (TBD)<br>
    a=cpar (TBD)<br>
    a=cparmin (TBD)<br>
    a=cparmax (TBD)<br>
    <br>
    I agree with the "sqn" category of NORMAL. <br>
    <br>
    For the rest of them, the challenge we have is that they are
    essentially what we could call "encapsulating" attributes, i.e. they
    simply encode attributes or other parameters. This leads to a
    problem that may apply to other attributes as well: Currently, the
    mux mechanism seeems to only distinguish attributes at the attribute
    name level, however there are attributes where you have the same
    attribute-name appear many times, but further parameters in the
    attribute distinguish these (and as per the above, this distinction
    may in turn lead to different categories). Also note, that these
    attributes to not actually negotiate use of their encapsulated
    values; a separate signaling mechanism, e.g. O/A exchange,&nbsp; would be
    needed for this (as noted in RFC 3407, Section 2). <br>
    <br>
    For "cdsc", I'd be inclined to put it in category "NORMAL", but for
    the rest of them, I'm not sure any of the current categories match,
    since it really depends on what was encapsulated in them (note that
    both bandwith and attributes can be encapsulated in them). <br>
    <br>
    Thanks <br>
    <br>
    -- Flemming <br>
    <br>
    <br>
    <div class="moz-cite-prefix">On 4/26/13 5:24 PM, Suhas Nandakumar
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAMRcRGRhwp74W8hBtxMw6jWiXsPmOJS+PBnJ2=pEkmAfVCXU3A@mail.gmail.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <div dir="ltr">Hello Flemming
        <div>&nbsp;&nbsp;</div>
        <div>This is a request for reviewing the following sections</div>
        <div>&nbsp;</div>
        <div>5.20</div>
        <div><br>
        </div>
        <div>of the draft<br>
        </div>
        <div>&nbsp; &nbsp;</div>
        <div>draft-nandakumar-mmusic-sdp-mux-attributes-02</div>
        <div>&nbsp;&nbsp;&nbsp;<br>
        </div>
        <div>It will be greatly appreciated if you could inform on your
          willingness for reviewing the aforementioned sessions.</div>
        <div><br>
        </div>
        <div>It would be great to have your review
          feedback/comments/inputs before June 9th 2013.</div>
        <div><br>
        </div>
        <div>The document is available here:</div>
        <div><a moz-do-not-send="true"
href="http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-attributes-02"
            target="_blank">http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-attributes-02</a><br>
        </div>
        <div><br>
        </div>
        <div><br>
        </div>
        <div>Thanks in Advance</div>
        <div><br>
        </div>
        <div>Regards</div>
        <div>Suhas Nandakumar</div>
      </div>
    </blockquote>
    <br>
  </body>
</html>

--------------090708050004090607080703--

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Date: Sun, 14 Jul 2013 21:12:10 -0400
From: Flemming Andreasen <fandreas@cisco.com>
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Cc: mmusic WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] Review Request- draft-nandakumar-mmusic-sdp-mux-attributes-02 - (RFC5939 Related)
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Section 5.28 defines the following attributes with the category shown in 
brackets:

a=pcfg (SPECIAL)
a=acfg (SPECIAL)
a=csup (SPECIAL)
a=creq (SPECIAL)
a=acap (SPECIAL)
a=tcap (IDENTICAL)

I think the definition of "SPECIAL" needs to be changed and maybe we 
need two versions of it: A "SPECIAL" for existing attributes, in which 
case the mux-attributes draft will have to define the handling that is 
needed, and a "SPECIAL NEW", which would essentially impose a new 
requirement on all new SDP attribute definitions to have one of the 
mux-attribute categories assigned to them, and if it's "SPECIAL NEW", 
then define the handling with muxing.

Moving on to the categories suggested above:

On the "tcap" one, it's debatable whether they need to be "IDENTICAL" 
since they are purely capabilities and as such not being used at this 
point in time; the "pcfg" atttribute is what would instantiate their 
use. Also, repeating the comment from the RFC 3407 attributes, we have 
an example of an attribute where there can be multiple different values 
(on purpose) and they are all equally valid. In this case, the 
negotiation process would end up picking one of them, and what you 
probably want for mux-attributes is the negotiation process to pick the 
same one for all the multiplexed m-lines; this seems well suited for a 
"SPECIAL" description (there are probably several other nuances to be 
covered). Similar considerations apply for "acap".

For the "csup" and "creq" ones, I don't think there is anything 
"SPECIAL". They each list one or more option tags supported or required 
for capability negotiation to be used, and I believe they can just 
continue to do so when doing multi-plexing. In other words, I think 
"NORMAL" would be right for these.

For "acfg" and "pcfg" I agree that further description on how they would 
be used with multiplexing would be needed (need to drive consistent 
negotiation across the multi-plexed m-lines).

Thanks

-- Flemming



On 4/26/13 5:31 PM, Suhas Nandakumar wrote:
> Hello Flemming
> This is a request for reviewing the following sections
> 5.28
>
> of the draft
> draft-nandakumar-mmusic-sdp-mux-attributes-02
>
> It will be greatly appreciated if you could inform on your willingness 
> for reviewing the aforementioned sessions.
>
> It would be great to have your review feedback/comments/input before 
> June 9th 2013.
>
> The document is available here:
> http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-attributes-02
>
>
> Thanks in Advance
>
> Regards
> Suhas Nandakumar


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    Section 5.28 defines the following attributes with the category
    shown in brackets:<br>
    <br>
    a=pcfg (SPECIAL)<br>
    a=acfg (SPECIAL)<br>
    a=csup (SPECIAL)<br>
    a=creq (SPECIAL)<br>
    a=acap (SPECIAL)<br>
    a=tcap (IDENTICAL)<br>
    <br>
    I think the definition of "SPECIAL" needs to be changed and maybe we
    need two versions of it: A "SPECIAL" for existing attributes, in
    which case the mux-attributes draft will have to define the handling
    that is needed, and a "SPECIAL NEW", which would essentially impose
    a new requirement on all new SDP attribute definitions to have one
    of the mux-attribute categories assigned to them, and if it's
    "SPECIAL NEW", then define the handling with muxing. <br>
    <br>
    Moving on to the categories suggested above: <br>
    <br>
    On the "tcap" one, it's debatable whether they need to be
    "IDENTICAL" since they are purely capabilities and as such not being
    used at this point in time; the "pcfg" atttribute is what would
    instantiate their use. Also, repeating the comment from the RFC 3407
    attributes, we have an example of an attribute where there can be
    multiple different values (on purpose) and they are all equally
    valid. In this case, the negotiation process would end up picking
    one of them, and what you probably want for mux-attributes is the
    negotiation process to pick the same one for all the multiplexed
    m-lines; this seems well suited for a "SPECIAL" description (there
    are probably several other nuances to be covered). Similar
    considerations apply for "acap". <br>
    <br>
    For the "csup" and "creq" ones, I don't think there is anything
    "SPECIAL". They each list one or more option tags supported or
    required for capability negotiation to be used, and I believe they
    can just continue to do so when doing multi-plexing. In other words,
    I think "NORMAL" would be right for these. <br>
    <br>
    For "acfg" and "pcfg" I agree that further description on how they
    would be used with multiplexing would be needed (need to drive
    consistent negotiation across the multi-plexed m-lines). <br>
    <br>
    Thanks <br>
    <br>
    -- Flemming <br>
    <br>
    <br>
    <br>
    <div class="moz-cite-prefix">On 4/26/13 5:31 PM, Suhas Nandakumar
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAMRcRGTGDbHhXmJKYVph-xBE8AOViwXaaAhfmEFiyX91XwFiKA@mail.gmail.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <div dir="ltr">Hello Flemming
        <div>&nbsp;&nbsp;</div>
        <div>This is a request for reviewing the following sections</div>
        <div>&nbsp;</div>
        <div>5.28</div>
        <div><br>
        </div>
        <div>of the draft<br>
        </div>
        <div>&nbsp; &nbsp;</div>
        <div>draft-nandakumar-mmusic-sdp-mux-attributes-02</div>
        <div>&nbsp;&nbsp;&nbsp;<br>
        </div>
        <div>It will be greatly appreciated if you could inform on your
          willingness for reviewing the aforementioned sessions.</div>
        <div><br>
        </div>
        <div>It would be great to have your review
          feedback/comments/input before June 9th 2013.</div>
        <div><br>
        </div>
        <div>The document is available here:</div>
        <div><a moz-do-not-send="true"
href="http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-attributes-02"
            target="_blank">http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-attributes-02</a><br>
        </div>
        <div><br>
        </div>
        <div><br>
        </div>
        <div>Thanks in Advance</div>
        <div><br>
        </div>
        <div>Regards</div>
        <div>Suhas Nandakumar</div>
      </div>
    </blockquote>
    <br>
  </body>
</html>

--------------020306050508010102050601--

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From: Suhas Nandakumar <suhasietf@gmail.com>
To: fandreas@cisco.com
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Subject: Re: [MMUSIC] Review Request - draft-nandakumar-mmusic-sdp-mux-attributes-02 - (RFC3407 Related)
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Thanks Flemming for the review inputs.

I was wondering does assigning SPECIAL to encapsulating attributes and
updating the Notes section to reflect that one should look into the
categorization of encapsulated attribute for assessing the behavior

Please let me know if this is the right way to go.

Cheers
Suhas

On Fri, Apr 26, 2013 at 2:24 PM, Suhas Nandakumar <suhasietf@gmail.com>wrote:

> Hello Flemming
>
> This is a request for reviewing the following sections
>
> 5.20
>
> of the draft
>
> draft-nandakumar-mmusic-sdp-mux-attributes-02
>
> It will be greatly appreciated if you could inform on your willingness for
> reviewing the aforementioned sessions.
>
> It would be great to have your review feedback/comments/inputs before June
> 9th 2013.
>
> The document is available here:
> http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-attributes-02
>
>
> Thanks in Advance
>
> Regards
> Suhas Nandakumar
>

--e89a8f6474fd04847604e1868caa
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Thanks Flemming for the review inputs.=A0<div><br></div><div>I was wonderin=
g does assigning SPECIAL to encapsulating attributes and updating the Notes=
 section to reflect that one should look into the categorization of encapsu=
lated attribute for assessing the behavior<div>
<br></div><div>Please let me know if this is the right way to go.</div><div=
><br></div><div>Cheers</div><div>Suhas<br><div><div><div><div><br><div clas=
s=3D"gmail_quote">On Fri, Apr 26, 2013 at 2:24 PM, Suhas Nandakumar <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:suhasietf@gmail.com" target=3D"_blank">suh=
asietf@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr">Hello Flemming<div>=A0=A0</=
div><div>This is a request for reviewing the following sections</div><div>=
=A0</div>
<div>5.20</div><div><br></div><div>of the draft<br></div><div>=A0 =A0</div>=
<div>draft-nandakumar-mmusic-sdp-mux-attributes-02</div>



<div>=A0=A0=A0<br></div><div>It will be greatly appreciated if you could in=
form on your willingness for reviewing the aforementioned sessions.</div><d=
iv><br></div><div>It would be great to have your review feedback/comments/i=
nputs before June 9th 2013.</div>




<div><br></div><div>The document is available here:</div><div><a href=3D"ht=
tp://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-attributes-02" tar=
get=3D"_blank">http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-a=
ttributes-02</a><br>




</div><div><br></div><div><br></div><div>Thanks in Advance</div><div><br></=
div><div>Regards</div><span class=3D"HOEnZb"><font color=3D"#888888"><div>S=
uhas Nandakumar</div></font></span></div>
</blockquote></div><br></div></div></div></div></div></div>

--e89a8f6474fd04847604e1868caa--

From gunnar.hellstrom@omnitor.se  Sun Jul 14 23:40:37 2013
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From: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-01.txt - more use cases
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I suggest to add a couple of use cases.


4.x.  Deaf French sign language user calling another deaf French sign language user.

Someone who mainly uses French sign language (FSL) places a call.
The call offers a video stream listing FSL in both directions with highest preference and also
real-time text stream on low preference level for French in both directions.
Audio is also included but without language indications.
The call destination turns out to be another user of FSL with the same media and language settings.
The called client can find that the language preferences match, so the same language indications are
sent in the SDP answer, and the call can be connected without connection of any relay service.
The users can sign directly with each other in the video streams, and occasionally use the real-time
text streams for data that need exact spelling or noted.


4.y.  Spoken Japanese user calling a Spanish speaking person.

Someone who speaks Japanese and is subscriber of an on-line interpreting service places a call.
The call offers audio streams using Japanese in both directions on high preference level, video
with spoken Japanese with no specific preference and both message-wise and real-time text in
Japanese in both directions.
This user is customer of an on-line interpreting service. The service adds its interpreting
language capabilities to the SDP on lower level than the originator, and conveys the call to the destination.
The destination is a Spanish speaking person. The called terminal detects that Spanish is among the possible
languages even if not highest preference, so that call is answered with indication of spoken Spanish at high
priority level and Spanish text messaging in both directions at low priority level.
The on-line interpreting service detects that their translation between Japanese and Spanish is needed to
make the call work, so they allocate an interpreter to the call, and answers the call with an indication
of Japanese towards the caller.
Information that the call will be soon supported by a Japanese/Spanish interpreter is provided in the
appropriate languages in the text message connections.
The users can communicate in their own languages supported by the interpreter.


-------------------------------
More might be needed to clarify some more of the endless possible variations and benefit.





------------------------------------------------------------------------
Gunnar Hellström
Omnitor
gunnar.hellstrom@omnitor.se
+46708204288
On 2013-07-15 04:11, Christian Groves wrote:
> Hello Randall and Gunnar,
>
> I think Randall's clarification would be good to have in the draft. I 
> think that when a caller support the language it is sent for every 
> call. Whether the language is used by a receiver depends on the 
> receiver's local capability.
>
>
> Regards, Christian
>
>
> On 14/07/2013 4:04 AM, Gunnar Hellstrom wrote:
>> On 2013-07-13 10:07, Randall Gellens wrote:
>>> At 2:49 PM +1000 7/10/13, Christian Groves wrote:
>>>
>>>>  I wouldn't have thought that it would be use case specific... i.e. 
>>>> i'd only include my language when I called an emergency service. I 
>>>> would think an endpoint would be configured with the information 
>>>> and then provide it whenever I made a call. Is that the intent?
>>>
>>> I expect so, but then, I also didn't expect it to be used much in 
>>> the person-to-person case since usually language is implied by 
>>> context when calling an individual.  I expect it to be used when a 
>>> person is calling an call center (corporate or emergency).  To put 
>>> it another way, I expect a personal client to always set language on 
>>> outbound calls, but in most cases to ignore it on inbound calls.  
>>> I'd expect a corporate call center, as well as emergency call 
>>> handling facilities, to pay attention.  I should make this more 
>>> clear in the draft.
>>>
>> I expect the setting to be retrieved from a personal profile (likely 
>> but not necessarily stored in connection with other user information 
>> and account information) and used in every call.
>>
>> /Gunnar
>> _______________________________________________
>> 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


--------------070009030701040206060904
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: 7bit

<html>
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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
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    <div class="moz-cite-prefix">I suggest to add a couple of use cases.<br>
      <br>
      <br>
      <pre style="font-family: monospace; line-height: 1.2em; margin: 0px; color: rgb(0, 0, 0); font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;"><span class="m_h" style="font-family: arial; font-weight: bold;">4.x.  Deaf French sign language user calling another deaf French sign language user.</span>

Someone who mainly uses French sign language (FSL) places a call.  
The call offers a video stream listing FSL in both directions with highest preference and also 
real-time text stream on low preference level for French in both directions. 
Audio is also included but without language indications.
The call destination turns out to be another user of FSL with the same media and language settings.
The called client can find that the language preferences match, so the same language indications are 
sent in the SDP answer, and the call can be connected without connection of any relay service.
The users can sign directly with each other in the video streams, and occasionally use the real-time
text streams for data that need exact spelling or noted.


<span class="m_h" style="font-family: arial; font-weight: bold;">4.y.  Spoken Japanese user calling a Spanish speaking person.</span>

Someone who speaks Japanese and is subscriber of an on-line interpreting service places a call.  
The call offers audio streams using Japanese in both directions on high preference level, video 
with spoken Japanese with no specific preference and both message-wise and real-time text in 
Japanese in both directions.
This user is customer of an on-line interpreting service. The service adds its interpreting 
language capabilities to the SDP on lower level than the originator, and conveys the call to the destination.
The destination is a Spanish speaking person. The called terminal detects that Spanish is among the possible 
languages even if not highest preference, so that call is answered with indication of spoken Spanish at high 
priority level and Spanish text messaging in both directions at low priority level.
The on-line interpreting service detects that their translation between Japanese and Spanish is needed to
make the call work, so they allocate an interpreter to the call, and answers the call with an indication 
of Japanese towards the caller.
Information that the call will be soon supported by a Japanese/Spanish interpreter is provided in the 
appropriate languages in the text message connections.
The users can communicate in their own languages supported by the interpreter.


-------------------------------
More might be needed to clarify some more of the endless possible variations and benefit.




</pre>
      <br>
      <div class="moz-signature">
        <hr>
        Gunnar Hellstr&ouml;m<br>
        Omnitor<br>
        <a class="moz-txt-link-abbreviated" href="mailto:gunnar.hellstrom@omnitor.se">gunnar.hellstrom@omnitor.se</a><br>
        +46708204288<br>
      </div>
      On 2013-07-15 04:11, Christian Groves wrote:<br>
    </div>
    <blockquote cite="mid:51E35A53.5030502@nteczone.com" type="cite">Hello
      Randall and Gunnar,
      <br>
      <br>
      I think Randall's clarification would be good to have in the
      draft. I think that when a caller support the language it is sent
      for every call. Whether the language is used by a receiver depends
      on the receiver's local capability.
      <br>
      <br>
      <br>
      Regards, Christian
      <br>
      <br>
      <br>
      On 14/07/2013 4:04 AM, Gunnar Hellstrom wrote:
      <br>
      <blockquote type="cite">On 2013-07-13 10:07, Randall Gellens
        wrote:
        <br>
        <blockquote type="cite">At 2:49 PM +1000 7/10/13, Christian
          Groves wrote:
          <br>
          <br>
          <blockquote type="cite">&nbsp;I wouldn't have thought that it would
            be use case specific... i.e. i'd only include my language
            when I called an emergency service. I would think an
            endpoint would be configured with the information and then
            provide it whenever I made a call. Is that the intent?
            <br>
          </blockquote>
          <br>
          I expect so, but then, I also didn't expect it to be used much
          in the person-to-person case since usually language is implied
          by context when calling an individual.&nbsp; I expect it to be used
          when a person is calling an call center (corporate or
          emergency).&nbsp; To put it another way, I expect a personal client
          to always set language on outbound calls, but in most cases to
          ignore it on inbound calls.&nbsp; I'd expect a corporate call
          center, as well as emergency call handling facilities, to pay
          attention.&nbsp; I should make this more clear in the draft.
          <br>
          <br>
        </blockquote>
        I expect the setting to be retrieved from a personal profile
        (likely but not necessarily stored in connection with other user
        information and account information) and used in every call.
        <br>
        <br>
        /Gunnar
        <br>
        _______________________________________________
        <br>
        mmusic mailing list
        <br>
        <a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
        <br>
        <a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
        <br>
        <br>
      </blockquote>
      <br>
      _______________________________________________
      <br>
      mmusic mailing list
      <br>
      <a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
      <br>
      <a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
      <br>
    </blockquote>
    <br>
  </body>
</html>

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From: "Avasarala, Ranjit (NSN - IN/Bangalore)" <ranjit.avasarala@nsn.com>
To: "ext Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>, "mmusic@ietf.org" <mmusic@ietf.org>
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
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Hi Muthu

I have following comments
1. RFC 4856 defines "annexa" and "annexb" as optional parameters. So I agre=
e there could be misinpterpretations.
2. The proposed draft should make these parameters as mandatory, otherwise =
the implementations would continue to ignore the parameters.
3. Update RFC 4856 moving "annexa" and "annexb" parameters from "optional" =
to "required".

Regards
Ranjit



-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 ext Muthu Arul Mozhi Perumal (mperumal)
Sent: Monday, July 15, 2013 10:21 AM
To: mmusic@ietf.org
Subject: [MMUSIC] FW: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt

This updated version addresses all comments received during WGLC.

- Muthu, Partha

-----Original Message-----
From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org] =
On Behalf Of internet-drafts@ietf.org
Sent: Monday, July 15, 2013 10:13 AM
To: i-d-announce@ietf.org
Cc: mmusic@ietf.org
Subject: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.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           : Offer/Answer Considerations for G723 Annex A and G729 An=
nex B
	Author(s)       : Muthu Arul Mozhi Perumal
                          Parthasarathi Ravindran
	Filename        : draft-ietf-mmusic-sdp-g723-g729-04.txt
	Pages           : 8
	Date            : 2013-07-14

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 Session Description
   protocol(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-g729-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-g723-g729-04

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

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From: <aurelien.sollaud@orange.com>
To: "Avasarala, Ranjit (NSN - IN/Bangalore)" <ranjit.avasarala@nsn.com>, "ext Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
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Hi

A lot of media subtypes have optional parameters, and as long as a default =
value is defined, there is no issue.
The problem here is not that annexa/annexb parameters are optional, but tha=
t the offer/answer procedure was not defined for these media subtypes.

-----Message d'origine-----
De=A0: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] De la part =
de Avasarala, Ranjit (NSN - IN/Bangalore)
Envoy=E9=A0: lundi 15 juillet 2013 09:08
=C0=A0: ext Muthu Arul Mozhi Perumal (mperumal); mmusic@ietf.org
Objet=A0: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt

Hi Muthu

I have following comments
1. RFC 4856 defines "annexa" and "annexb" as optional parameters. So I agre=
e there could be misinpterpretations.
2. The proposed draft should make these parameters as mandatory, otherwise =
the implementations would continue to ignore the parameters.
3. Update RFC 4856 moving "annexa" and "annexb" parameters from "optional" =
to "required".

Regards
Ranjit



-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 ext Muthu Arul Mozhi Perumal (mperumal)
Sent: Monday, July 15, 2013 10:21 AM
To: mmusic@ietf.org
Subject: [MMUSIC] FW: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt

This updated version addresses all comments received during WGLC.

- Muthu, Partha

-----Original Message-----
From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org] =
On Behalf Of internet-drafts@ietf.org
Sent: Monday, July 15, 2013 10:13 AM
To: i-d-announce@ietf.org
Cc: mmusic@ietf.org
Subject: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.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           : Offer/Answer Considerations for G723 Annex A and G729 An=
nex B
	Author(s)       : Muthu Arul Mozhi Perumal
                          Parthasarathi Ravindran
	Filename        : draft-ietf-mmusic-sdp-g723-g729-04.txt
	Pages           : 8
	Date            : 2013-07-14

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 Session Description
   protocol(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-g729-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-g723-g729-04

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

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From mperumal@cisco.com  Mon Jul 15 07:29:07 2013
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "aurelien.sollaud@orange.com" <aurelien.sollaud@orange.com>, "Avasarala, Ranjit (NSN - IN/Bangalore)" <ranjit.avasarala@nsn.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
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Agree with Aurelien. In addition, the draft explicitly specifies the behavi=
or for the default case:

   When the offer or answer has G723 and the annexa parameter is absent,
   the offerer or answerer knows that it has implied the default
   "annexa=3Dyes".  This is because the annexa attribute is part of the
   original registration of audio/G723 [RFC4856].  All implementations
   that support G723 understand the annexa attribute.  Hence, this case
   MUST be considered as if the offer or answer has G723 with
   annexa=3Dyes.

This I think should suffice to make sure implementations don't ignore it. S=
o, making these optional parameters mandatory optional isn't needed (and pr=
obably isn't right).

Muthu

|-----Original Message-----
|From: aurelien.sollaud@orange.com [mailto:aurelien.sollaud@orange.com]
|Sent: Monday, July 15, 2013 2:48 PM
|To: Avasarala, Ranjit (NSN - IN/Bangalore); Muthu Arul Mozhi Perumal (mper=
umal); mmusic@ietf.org
|Subject: RE: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
|
|Hi
|
|A lot of media subtypes have optional parameters, and as long as a default=
 value is defined, there is
|no issue.
|The problem here is not that annexa/annexb parameters are optional, but th=
at the offer/answer
|procedure was not defined for these media subtypes.
|
|-----Message d'origine-----
|De=A0: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] De la part=
 de Avasarala, Ranjit (NSN -
|IN/Bangalore)
|Envoy=E9=A0: lundi 15 juillet 2013 09:08
|=C0=A0: ext Muthu Arul Mozhi Perumal (mperumal); mmusic@ietf.org
|Objet=A0: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
|
|Hi Muthu
|
|I have following comments
|1. RFC 4856 defines "annexa" and "annexb" as optional parameters. So I agr=
ee there could be
|misinpterpretations.
|2. The proposed draft should make these parameters as mandatory, otherwise=
 the implementations would
|continue to ignore the parameters.
|3. Update RFC 4856 moving "annexa" and "annexb" parameters from "optional"=
 to "required".
|
|Regards
|Ranjit
|
|
|
|-----Original Message-----
|From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf O=
f ext Muthu Arul Mozhi
|Perumal (mperumal)
|Sent: Monday, July 15, 2013 10:21 AM
|To: mmusic@ietf.org
|Subject: [MMUSIC] FW: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
|
|This updated version addresses all comments received during WGLC.
|
|- Muthu, Partha
|
|-----Original Message-----
|From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org]=
 On Behalf Of internet-
|drafts@ietf.org
|Sent: Monday, July 15, 2013 10:13 AM
|To: i-d-announce@ietf.org
|Cc: mmusic@ietf.org
|Subject: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
|
|A New Internet-Draft is available from the on-line Internet-Drafts directo=
ries.
| This draft is a work item of the Multiparty Multimedia Session Control Wo=
rking Group of the IETF.
|
|	Title           : Offer/Answer Considerations for G723 Annex A and G729 A=
nnex B
|	Author(s)       : Muthu Arul Mozhi Perumal
|                          Parthasarathi Ravindran
|	Filename        : draft-ietf-mmusic-sdp-g723-g729-04.txt
|	Pages           : 8
|	Date            : 2013-07-14
|
|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 Session Description
|   protocol(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-g729-04
|
|A diff from the previous version is available at:
|http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-g723-g729-04
|
|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.i=
etf.org/ietf/1shadow-
|sites.txt
|_______________________________________________
|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
|
|__________________________________________________________________________=
____________________________
|___________________
|
|Ce message et ses pieces jointes peuvent contenir des informations confide=
ntielles ou privilegiees et
|ne doivent donc
|pas etre diffuses, exploites ou copies sans autorisation. Si vous avez rec=
u ce message par erreur,
|veuillez le signaler
|a l'expediteur et le detruire ainsi que les pieces jointes. Les messages e=
lectroniques etant
|susceptibles d'alteration,
|Orange decline toute responsabilite si ce message a ete altere, deforme ou=
 falsifie. Merci.
|
|This message and its attachments may contain confidential or privileged in=
formation that may be
|protected by law;
|they should not be distributed, used or copied without authorisation.
|If you have received this email in error, please notify the sender and del=
ete this message and its
|attachments.
|As emails may be altered, Orange is not liable for messages that have been=
 modified, changed or
|falsified.
|Thank you.


From suhasietf@gmail.com  Mon Jul 15 08:40:50 2013
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--089e0122e9c62a5a2a04e18eafda
Content-Type: text/plain; charset=ISO-8859-1

Hello All

   An updated version of the SDP Attributes Mux draft has been submitted
with reviews for close to 50% of the attributes completed. Thanks every one
for their valuable inputs. We are half-way there :)


Cheers
Suhas Nandakumar

A new version of I-D, draft-nandakumar-mmusic-sdp-mux-attributes-03.txt
has been successfully submitted by Suhas Nandakumar and posted to the
IETF repository.

Filename:        draft-nandakumar-mmusic-sdp-mux-attributes
Revision:        03
Title:           A Framework for SDP Attributes when Multiplexing
Creation date:   2013-07-15
Group:           Individual Submission
Number of pages: 56
URL:
http://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-sdp-mux-attributes-03.txt
Status:
http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-sdp-mux-attributes
Htmlized:
http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-attributes-03
Diff:
http://www.ietf.org/rfcdiff?url2=draft-nandakumar-mmusic-sdp-mux-attributes-03

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

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

--089e0122e9c62a5a2a04e18eafda
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div>Hello All</div><div><br></div><div>=A0 =A0An updated version of the SD=
P Attributes Mux draft has been submitted with reviews for close to 50% of =
the attributes completed. Thanks every one for their valuable inputs. We ar=
e half-way there :)</div>
<div><br></div><div><br></div><div>Cheers</div><div>Suhas Nandakumar</div><=
div><br></div><div>A new version of I-D, draft-nandakumar-mmusic-sdp-mux-at=
tributes-03.txt</div><div>has been successfully submitted by Suhas Nandakum=
ar and posted to the</div>
<div>IETF repository.</div><div><br></div><div>Filename: =A0 =A0 =A0 =A0dra=
ft-nandakumar-mmusic-sdp-mux-attributes</div><div>Revision: =A0 =A0 =A0 =A0=
03</div><div>Title: =A0 =A0 =A0 =A0 =A0 A Framework for SDP Attributes when=
 Multiplexing</div><div>
Creation date: =A0 2013-07-15</div><div>Group: =A0 =A0 =A0 =A0 =A0 Individu=
al Submission</div><div>Number of pages: 56</div><div>URL: =A0 =A0 =A0 =A0 =
=A0 =A0 <a href=3D"http://www.ietf.org/internet-drafts/draft-nandakumar-mmu=
sic-sdp-mux-attributes-03.txt">http://www.ietf.org/internet-drafts/draft-na=
ndakumar-mmusic-sdp-mux-attributes-03.txt</a></div>
<div>Status: =A0 =A0 =A0 =A0 =A0<a href=3D"http://datatracker.ietf.org/doc/=
draft-nandakumar-mmusic-sdp-mux-attributes">http://datatracker.ietf.org/doc=
/draft-nandakumar-mmusic-sdp-mux-attributes</a></div><div>Htmlized: =A0 =A0=
 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-m=
ux-attributes-03">http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mu=
x-attributes-03</a></div>
<div>Diff: =A0 =A0 =A0 =A0 =A0 =A0<a href=3D"http://www.ietf.org/rfcdiff?ur=
l2=3Ddraft-nandakumar-mmusic-sdp-mux-attributes-03">http://www.ietf.org/rfc=
diff?url2=3Ddraft-nandakumar-mmusic-sdp-mux-attributes-03</a></div><div><br=
></div><div>Abstract:</div>
<div>=A0 =A0The Session Description Protocol (SDP) provides mechanisms to</=
div><div>=A0 =A0describe attributes of multimedia sessions and of individua=
l media</div><div>=A0 =A0streams (e.g., Real-time Transport Protocol (RTP) =
sessions) within a</div>
<div>=A0 =A0multimedia session. =A0In the RTCWeb WG, there is a need to use=
 a</div><div>=A0 =A0single 5-tuple for sending and receiving media associat=
ed with</div><div>=A0 =A0multiple media descriptions (&quot;m=3D&quot; line=
s). =A0Such a requirement has</div>
<div>=A0 =A0raised concerns over the semantic implications of the SDP attri=
butes</div><div>=A0 =A0associated with the RTP Sessions multiplexed over a =
single transport</div><div>=A0 =A0layer flow.</div><div><br></div><div>=A0 =
=A0The scope of this specification is to provide a framework for</div>
<div>=A0 =A0analyzing the multiplexing characteristics of SDP attributes. =
=A0The</div><div>=A0 =A0specification also categorizes existing attributes =
based on the</div><div>=A0 =A0framework described herein.</div>

--089e0122e9c62a5a2a04e18eafda--

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

	Title           : Using Interactive Connectivity Establishment (ICE) with =
Session Description Protocol (SDP) offer/answer and Session Initiation Prot=
ocol (SIP)
	Author(s)       : Marc Petit-Huguenin
                          Ari Keranen
	Filename        : draft-ietf-mmusic-ice-sip-sdp-00.txt
	Pages           : 40
	Date            : 2013-07-15

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


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

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


Internet-Drafts are also available by anonymous FTP at:
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Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-rfc5245bis-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           : Interactive Connectivity Establishment (ICE): A Protocol=
 for Network Address Translator (NAT) Traversal for Offer/Answer Protocols
	Author(s)       : Ari Keranen
                          Jonathan Rosenberg
	Filename        : draft-ietf-mmusic-rfc5245bis-00.txt
	Pages           : 89
	Date            : 2013-07-15

Abstract:
   This document describes a protocol for Network Address Translator
   (NAT) traversal for UDP-based multimedia sessions established with
   the offer/answer model.  This protocol is called Interactive
   Connectivity Establishment (ICE).  ICE makes use of the Session
   Traversal Utilities for NAT (STUN) protocol and its extension,
   Traversal Using Relay NAT (TURN).  ICE can be used by any protocol
   utilizing the offer/answer model, such as the Session Initiation
   Protocol (SIP).

   This document obsoletes RFC 5245.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-rfc5245bis

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-rfc5245bis-00


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Cc: mmusic WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] Review Request - draft-nandakumar-mmusic-sdp-mux-attributes-02 - (RFC3407 Related)
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I'd lean towards a new category for these kinds of attributes.

-- Flemming


On 7/15/13 1:58 AM, Suhas Nandakumar wrote:
> Thanks Flemming for the review inputs.
>
> I was wondering does assigning SPECIAL to encapsulating attributes and 
> updating the Notes section to reflect that one should look into the 
> categorization of encapsulated attribute for assessing the behavior
>
> Please let me know if this is the right way to go.
>
> Cheers
> Suhas
>
> On Fri, Apr 26, 2013 at 2:24 PM, Suhas Nandakumar <suhasietf@gmail.com 
> <mailto:suhasietf@gmail.com>> wrote:
>
>     Hello Flemming
>     This is a request for reviewing the following sections
>     5.20
>
>     of the draft
>     draft-nandakumar-mmusic-sdp-mux-attributes-02
>
>     It will be greatly appreciated if you could inform on your
>     willingness for reviewing the aforementioned sessions.
>
>     It would be great to have your review feedback/comments/inputs
>     before June 9th 2013.
>
>     The document is available here:
>     http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-attributes-02
>
>
>     Thanks in Advance
>
>     Regards
>     Suhas Nandakumar
>
>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    I'd lean towards a new category for these kinds of attributes. <br>
    <br>
    -- Flemming <br>
    <br>
    <br>
    <div class="moz-cite-prefix">On 7/15/13 1:58 AM, Suhas Nandakumar
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAMRcRGSaJaF4tSXKuOD+6-HDBG8hpD0P1gQym2jzhHxucfjqmA@mail.gmail.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      Thanks Flemming for the review inputs.&nbsp;
      <div><br>
      </div>
      <div>I was wondering does assigning SPECIAL to encapsulating
        attributes and updating the Notes section to reflect that one
        should look into the categorization of encapsulated attribute
        for assessing the behavior
        <div>
          <br>
        </div>
        <div>Please let me know if this is the right way to go.</div>
        <div><br>
        </div>
        <div>Cheers</div>
        <div>Suhas<br>
          <div>
            <div>
              <div>
                <div><br>
                  <div class="gmail_quote">On Fri, Apr 26, 2013 at 2:24
                    PM, Suhas Nandakumar <span dir="ltr">&lt;<a
                        moz-do-not-send="true"
                        href="mailto:suhasietf@gmail.com"
                        target="_blank">suhasietf@gmail.com</a>&gt;</span>
                    wrote:<br>
                    <blockquote class="gmail_quote" style="margin:0 0 0
                      .8ex;border-left:1px #ccc solid;padding-left:1ex">
                      <div dir="ltr">Hello Flemming
                        <div>&nbsp;&nbsp;</div>
                        <div>This is a request for reviewing the
                          following sections</div>
                        <div>&nbsp;</div>
                        <div>5.20</div>
                        <div><br>
                        </div>
                        <div>of the draft<br>
                        </div>
                        <div>&nbsp; &nbsp;</div>
                        <div>draft-nandakumar-mmusic-sdp-mux-attributes-02</div>
                        <div>&nbsp;&nbsp;&nbsp;<br>
                        </div>
                        <div>It will be greatly appreciated if you could
                          inform on your willingness for reviewing the
                          aforementioned sessions.</div>
                        <div><br>
                        </div>
                        <div>It would be great to have your review
                          feedback/comments/inputs before June 9th 2013.</div>
                        <div><br>
                        </div>
                        <div>The document is available here:</div>
                        <div><a moz-do-not-send="true"
href="http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-attributes-02"
                            target="_blank">http://tools.ietf.org/html/draft-nandakumar-mmusic-sdp-mux-attributes-02</a><br>
                        </div>
                        <div><br>
                        </div>
                        <div><br>
                        </div>
                        <div>Thanks in Advance</div>
                        <div><br>
                        </div>
                        <div>Regards</div>
                        <span class="HOEnZb"><font color="#888888">
                            <div>Suhas Nandakumar</div>
                          </font></span></div>
                    </blockquote>
                  </div>
                  <br>
                </div>
              </div>
            </div>
          </div>
        </div>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
mmusic mailing list
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</pre>
    </blockquote>
    <br>
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From: Suhas Nandakumar <suhasietf@gmail.com>
To: mmusic WG <mmusic@ietf.org>, rtcweb@ietf.org
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Subject: [MMUSIC] New Draft: draft-nandakumar-mmusic-rtcweb-grouping-00.txt
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--089e013d0f806648e504e1900c34
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Hello All

   A new draft has been submitted to describe Simulcast streams within SDP
based on RFC5576 ssrc-grouping semantics.

Please have a read and let me know your thoughts

Cheers
Suhas

A new version of I-D, draft-nandakumar-mmusic-rtcweb-grouping-00.txt
has been successfully submitted by Suhas Nandakumar and posted to the
IETF repository.

Filename:        draft-nandakumar-mmusic-rtcweb-grouping
Revision:        00
Title:           SSRC Group Based Simulcast Signaling
Creation date:   2013-07-15
Group:           Individual Submission
Number of pages: 8
URL:
http://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-rtcweb-grouping=
-00.txt<https://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiMHIouDdKX4c=
q8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&URL=3Dhttp%3a%2f%2fwww.=
ietf.org%2finternet-drafts%2fdraft-nandakumar-mmusic-rtcweb-grouping-00.txt=
>
Status:
http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-rtcweb-grouping<htt=
ps://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiMHIouDdKX4cq8VVdAIUowN=
DR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&URL=3Dhttp%3a%2f%2fdatatracker.iet=
f.org%2fdoc%2fdraft-nandakumar-mmusic-rtcweb-grouping>
Htmlized:
http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-00<https=
://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiMHIouDdKX4cq8VVdAIUowNDR=
3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&URL=3Dhttp%3a%2f%2ftools.ietf.org%2f=
html%2fdraft-nandakumar-mmusic-rtcweb-grouping-00>


Abstract:
   In some applications it may be necessary to send multiple media
   encodings corresponding to a media source with in independent RTP
   media streams.  This is called Simulcast.  This document discusses a
   framework for describing simulcast media streams in SDP and also
   defines semantics to express relationship amongst them.  The
   semantics defined in this document are to be used with the source
   specific grouping framework defined in the [RFC5576]

--089e013d0f806648e504e1900c34
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div><span style=3D"font-family:monospace;font-size:16px">Hello All</span><=
/div><div><span style=3D"font-family:monospace;font-size:16px"><br></span><=
/div><div><span style=3D"font-family:monospace;font-size:16px">=A0 =A0A new=
 draft has been submitted to describe Simulcast streams within SDP based on=
 RFC5576 ssrc-grouping semantics.</span></div>
<div><span style=3D"font-family:monospace;font-size:16px"><br></span></div>=
<div><span style=3D"font-family:monospace;font-size:16px">Please have a rea=
d and let me know your thoughts</span></div><div><span style=3D"font-family=
:monospace;font-size:16px"><br>
</span></div><div><span style=3D"font-family:monospace;font-size:16px">Chee=
rs</span></div><div><span style=3D"font-family:monospace;font-size:16px">Su=
has</span></div><span style=3D"font-family:monospace;font-size:16px"><div><=
span style=3D"font-family:monospace;font-size:16px"><br>
</span></div>A new version of I-D, draft-nandakumar-mmusic-rtcweb-grouping-=
00.txt</span><br style=3D"font-family:monospace;font-size:16px"><span style=
=3D"font-family:monospace;font-size:16px">has been successfully submitted b=
y Suhas Nandakumar and posted to the</span><br style=3D"font-family:monospa=
ce;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">IETF repository.</span=
><br style=3D"font-family:monospace;font-size:16px"><br style=3D"font-famil=
y:monospace;font-size:16px"><span style=3D"font-family:monospace;font-size:=
16px">Filename:=A0=A0=A0=A0=A0=A0=A0 draft-nandakumar-mmusic-rtcweb-groupin=
g</span><br style=3D"font-family:monospace;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">Revision:=A0=A0=A0=A0=
=A0=A0=A0 00</span><br style=3D"font-family:monospace;font-size:16px"><span=
 style=3D"font-family:monospace;font-size:16px">Title:=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0 SSRC Group Based Simulcast Signaling</span><br style=3D"font-fami=
ly:monospace;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">Creation date:=A0=A0 2=
013-07-15</span><br style=3D"font-family:monospace;font-size:16px"><span st=
yle=3D"font-family:monospace;font-size:16px">Group:=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0 Individual Submission</span><br style=3D"font-family:monospace;font-=
size:16px">
<span style=3D"font-family:monospace;font-size:16px">Number of pages: 8</sp=
an><br style=3D"font-family:monospace;font-size:16px"><span style=3D"font-f=
amily:monospace;font-size:16px">URL:=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
</span><a href=3D"https://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiM=
HIouDdKX4cq8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&amp;URL=3Dhtt=
p%3a%2f%2fwww.ietf.org%2finternet-drafts%2fdraft-nandakumar-mmusic-rtcweb-g=
rouping-00.txt" target=3D"_blank" style=3D"font-family:monospace;font-size:=
16px">http://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-rtcweb-gr=
ouping-00.txt</a><br style=3D"font-family:monospace;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">Status:=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0</span><a href=3D"https://mail.cisco.com/owa/redir.aspx?C=3D=
KlIQ10YF-Eu3HiMHIouDdKX4cq8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY=
.&amp;URL=3Dhttp%3a%2f%2fdatatracker.ietf.org%2fdoc%2fdraft-nandakumar-mmus=
ic-rtcweb-grouping" target=3D"_blank" style=3D"font-family:monospace;font-s=
ize:16px">http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-rtcweb-gr=
ouping</a><br style=3D"font-family:monospace;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">Htmlized:=A0=A0=A0=A0=
=A0=A0=A0=A0</span><a href=3D"https://mail.cisco.com/owa/redir.aspx?C=3DKlI=
Q10YF-Eu3HiMHIouDdKX4cq8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&a=
mp;URL=3Dhttp%3a%2f%2ftools.ietf.org%2fhtml%2fdraft-nandakumar-mmusic-rtcwe=
b-grouping-00" target=3D"_blank" style=3D"font-family:monospace;font-size:1=
6px">http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-00<=
/a><br style=3D"font-family:monospace;font-size:16px">
<br style=3D"font-family:monospace;font-size:16px"><br style=3D"font-family=
:monospace;font-size:16px"><span style=3D"font-family:monospace;font-size:1=
6px">Abstract:</span><br style=3D"font-family:monospace;font-size:16px"><sp=
an style=3D"font-family:monospace;font-size:16px">=A0=A0 In some applicatio=
ns it may be necessary to send multiple media</span><br style=3D"font-famil=
y:monospace;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">=A0=A0 encodings corre=
sponding to a media source with in independent RTP</span><br style=3D"font-=
family:monospace;font-size:16px"><span style=3D"font-family:monospace;font-=
size:16px">=A0=A0 media streams.=A0 This is called Simulcast.=A0 This docum=
ent discusses a</span><br style=3D"font-family:monospace;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">=A0=A0 framework for d=
escribing simulcast media streams in SDP and also</span><br style=3D"font-f=
amily:monospace;font-size:16px"><span style=3D"font-family:monospace;font-s=
ize:16px">=A0=A0 defines semantics to express relationship amongst them.=A0=
 The</span><br style=3D"font-family:monospace;font-size:16px">
<span style=3D"font-family:monospace;font-size:16px">=A0=A0 semantics defin=
ed in this document are to be used with the source</span><br style=3D"font-=
family:monospace;font-size:16px"><span style=3D"font-family:monospace;font-=
size:16px">=A0=A0 specific grouping framework defined in the [RFC5576]</spa=
n>

--089e013d0f806648e504e1900c34--

From ari.keranen@ericsson.com  Mon Jul 15 10:48:10 2013
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-rfc5245bis-00.txt
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Folks,

We've submitted the ICE drafts as WG documents, as agreed at the 
previous meeting. They also address the issues that were discussed 
during the meeting and/or on the list:
- clarified the use of short term credentials in connectivity checks
- not using deprecated IPv6 addresses or loop back addresses for candidates
- pairing IPv6 link-locals only with link-locals (wrt ULAs still more 
thinking and discussion is needed)
- ICE options for both session and media level 
(draft-ietf-mmusic-ice-sip-sdp-00)


Cheers,
Ari

On 7/15/13 7:48 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           : Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols
> 	Author(s)       : Ari Keranen
>                            Jonathan Rosenberg
> 	Filename        : draft-ietf-mmusic-rfc5245bis-00.txt
> 	Pages           : 89
> 	Date            : 2013-07-15
>
> Abstract:
>     This document describes a protocol for Network Address Translator
>     (NAT) traversal for UDP-based multimedia sessions established with
>     the offer/answer model.  This protocol is called Interactive
>     Connectivity Establishment (ICE).  ICE makes use of the Session
>     Traversal Utilities for NAT (STUN) protocol and its extension,
>     Traversal Using Relay NAT (TURN).  ICE can be used by any protocol
>     utilizing the offer/answer model, such as the Session Initiation
>     Protocol (SIP).
>
>     This document obsoletes RFC 5245.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-mmusic-rfc5245bis
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-mmusic-rfc5245bis-00
>
>
> 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
>


From emcho@sip-communicator.org  Mon Jul 15 11:05:07 2013
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From: Emil Ivov <emcho@jitsi.org>
Date: Mon, 15 Jul 2013 20:04:34 +0200
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To: =?ISO-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-rfc5245bis-00.txt
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Very minor comment:

On Mon, Jul 15, 2013 at 7:48 PM, Ari Ker=E4nen <ari.keranen@ericsson.com> w=
rote:
> Folks,
>
> We've submitted the ICE drafts as WG documents, as agreed at the previous
> meeting. They also address the issues that were discussed during the meet=
ing
> and/or on the list:
> - clarified the use of short term credentials in connectivity checks
> - not using deprecated IPv6 addresses or loop back addresses for candidat=
es

Wouldn't it make sense to allow pairing loopback addresses with other
loopback addresses? The main use case that comes to mind is debug
sessions where both the transmitter and the receiver are on the same
host?

Emil

> - pairing IPv6 link-locals only with link-locals (wrt ULAs still more
> thinking and discussion is needed)
> - ICE options for both session and media level
> (draft-ietf-mmusic-ice-sip-sdp-00)
>
>
> Cheers,
> Ari
>
>
> On 7/15/13 7:48 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           : Interactive Connectivity Establishment (ICE): =
A
>> Protocol for Network Address Translator (NAT) Traversal for Offer/Answer
>> Protocols
>>         Author(s)       : Ari Keranen
>>                            Jonathan Rosenberg
>>         Filename        : draft-ietf-mmusic-rfc5245bis-00.txt
>>         Pages           : 89
>>         Date            : 2013-07-15
>>
>> Abstract:
>>     This document describes a protocol for Network Address Translator
>>     (NAT) traversal for UDP-based multimedia sessions established with
>>     the offer/answer model.  This protocol is called Interactive
>>     Connectivity Establishment (ICE).  ICE makes use of the Session
>>     Traversal Utilities for NAT (STUN) protocol and its extension,
>>     Traversal Using Relay NAT (TURN).  ICE can be used by any protocol
>>     utilizing the offer/answer model, such as the Session Initiation
>>     Protocol (SIP).
>>
>>     This document obsoletes RFC 5245.
>>
>>
>> The IETF datatracker status page for this draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-mmusic-rfc5245bis
>>
>> There's also a htmlized version available at:
>> http://tools.ietf.org/html/draft-ietf-mmusic-rfc5245bis-00
>>
>>
>> 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



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

From suhasietf@gmail.com  Mon Jul 15 11:34:49 2013
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Date: Mon, 15 Jul 2013 11:34:46 -0700
Message-ID: <CAMRcRGRwMKyrkYc1=vhx3DxQXvAboo8h1TpL2fDd0Xfjndbhvg@mail.gmail.com>
From: Suhas Nandakumar <suhasietf@gmail.com>
To: mmusic WG <mmusic@ietf.org>, rtcweb@ietf.org
Content-Type: multipart/alternative; boundary=047d7bf10aea6b3ecd04e1911dd1
Subject: Re: [MMUSIC] New Draft: draft-nandakumar-mmusic-rtcweb-grouping-00.txt
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--047d7bf10aea6b3ecd04e1911dd1
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The links in my earlier email seems to have mail.cisco.com redirects and
hence I am including the direct links for the same below

Filename:        draft-nandakumar-mmusic-rtcweb-grouping
Revision:         00
Title:              SSRC Group Based Simulcast Signaling
Creation date:   2013-07-15
Group:            Individual Submission
Number of pages: 8
URL:
http://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-rtcweb-grouping=
-00.txt
Status:
http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-rtcweb-grouping
Htmlized:
http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-00

Sorry for the inconvenience

Cheers
Suhas
PS: Thanks Martin and Ethan for catching this.


On Mon, Jul 15, 2013 at 10:18 AM, Suhas Nandakumar <suhasietf@gmail.com>wro=
te:

> Hello All
>
>    A new draft has been submitted to describe Simulcast streams within SD=
P
> based on RFC5576 ssrc-grouping semantics.
>
> Please have a read and let me know your thoughts
>
> Cheers
> Suhas
>
> A new version of I-D, draft-nandakumar-mmusic-rtcweb-grouping-00.txt
> has been successfully submitted by Suhas Nandakumar and posted to the
> IETF repository.
>
> Filename:        draft-nandakumar-mmusic-rtcweb-grouping
> Revision:        00
> Title:           SSRC Group Based Simulcast Signaling
> Creation date:   2013-07-15
> Group:           Individual Submission
> Number of pages: 8
> URL:
> http://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-rtcweb-groupi=
ng-00.txt<https://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiMHIouDdKX=
4cq8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&URL=3Dhttp%3a%2f%2fww=
w.ietf.org%2finternet-drafts%2fdraft-nandakumar-mmusic-rtcweb-grouping-00.t=
xt>
> Status:
> http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-rtcweb-grouping<h=
ttps://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiMHIouDdKX4cq8VVdAIUo=
wNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&URL=3Dhttp%3a%2f%2fdatatracker.i=
etf.org%2fdoc%2fdraft-nandakumar-mmusic-rtcweb-grouping>
> Htmlized:
> http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-00<htt=
ps://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiMHIouDdKX4cq8VVdAIUowN=
DR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&URL=3Dhttp%3a%2f%2ftools.ietf.org%=
2fhtml%2fdraft-nandakumar-mmusic-rtcweb-grouping-00>
>
>
> Abstract:
>    In some applications it may be necessary to send multiple media
>    encodings corresponding to a media source with in independent RTP
>    media streams.  This is called Simulcast.  This document discusses a
>    framework for describing simulcast media streams in SDP and also
>    defines semantics to express relationship amongst them.  The
>    semantics defined in this document are to be used with the source
>    specific grouping framework defined in the [RFC5576]

--047d7bf10aea6b3ecd04e1911dd1
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div><br></div><div><br></div>The links in my earlier email seems to have <=
a href=3D"http://mail.cisco.com" target=3D"_blank">mail.cisco.com</a> redir=
ects and hence I am including the direct links for the same below<div><br><=
/div>
<div><span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-s=
ize:13px;background-color:rgb(255,255,255)">Filename: =A0 =A0 =A0 =A0draft-=
nandakumar-mmusic-</span><span style=3D"color:rgb(34,34,34);font-family:ari=
al,sans-serif;font-size:13px;background-color:rgb(255,255,255)">rtcweb-grou=
ping</span><br style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;fo=
nt-size:13px;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">Revision: =A0 =A0 =A0 =A0 00</span><=
br style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13px=
;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">Title: =A0 =A0 =A0 =A0 =A0 =A0 =A0SS=
RC Group Based Simulcast Signaling</span><br style=3D"color:rgb(34,34,34);f=
ont-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255=
)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">Creation date: =A0=A0</span><span cl=
ass=3D"aBn" tabindex=3D"0" style=3D"border-bottom-width:1px;border-bottom-s=
tyle:dashed;border-bottom-color:rgb(204,204,204);color:rgb(34,34,34);font-f=
amily:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)"><s=
pan class=3D"aQJ" style>2013-07-15</span></span><br style=3D"color:rgb(34,3=
4,34);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,=
255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">Group: =A0 =A0 =A0 =A0 =A0 =A0Indivi=
dual Submission</span><br style=3D"color:rgb(34,34,34);font-family:arial,sa=
ns-serif;font-size:13px;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">Number of pages: 8</span><br style=
=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13px;backgro=
und-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">URL: =A0 =A0 =A0 =A0 =A0 =A0=A0</spa=
n><a href=3D"http://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-rt=
cweb-grouping-00.txt" target=3D"_blank" style=3D"color:rgb(17,85,204);font-=
family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">h=
ttp://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-rtcweb-grouping-=
00.txt</a><br style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;fon=
t-size:13px;background-color:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">Status: =A0 =A0 =A0 =A0 =A0</span><a=
 href=3D"http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-rtcweb-gro=
uping" target=3D"_blank" style=3D"color:rgb(17,85,204);font-family:arial,sa=
ns-serif;font-size:13px;background-color:rgb(255,255,255)">http://datatrack=
er.ietf.org/doc/draft-nandakumar-mmusic-rtcweb-grouping</a><br style=3D"col=
or:rgb(34,34,34);font-family:arial,sans-serif;font-size:13px;background-col=
or:rgb(255,255,255)">
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:1=
3px;background-color:rgb(255,255,255)">Htmlized: =A0 =A0 =A0 =A0</span><a h=
ref=3D"http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-0=
0" target=3D"_blank" style=3D"color:rgb(17,85,204);font-family:arial,sans-s=
erif;font-size:13px;background-color:rgb(255,255,255)">http://tools.ietf.or=
g/html/draft-nandakumar-mmusic-rtcweb-grouping-00</a><br style=3D"color:rgb=
(34,34,34);font-family:arial,sans-serif;font-size:13px;background-color:rgb=
(255,255,255)">
</div>
<div><br></div><div>Sorry for the inconvenience</div><div><br></div><div>Ch=
eers</div><div>Suhas<br>PS: Thanks Martin and Ethan for catching this.</div=
><div><br></div><div><br><div class=3D"gmail_quote">On Mon, Jul 15, 2013 at=
 10:18 AM, Suhas Nandakumar <span dir=3D"ltr">&lt;<a href=3D"mailto:suhasie=
tf@gmail.com" target=3D"_blank">suhasietf@gmail.com</a>&gt;</span> wrote:<b=
r>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div><span style=3D"font-family:monospace;fo=
nt-size:16px">Hello All</span></div><div><span style=3D"font-family:monospa=
ce;font-size:16px"><br>

</span></div><div><span style=3D"font-family:monospace;font-size:16px">=A0 =
=A0A new draft has been submitted to describe Simulcast streams within SDP =
based on RFC5576 ssrc-grouping semantics.</span></div>
<div><span style=3D"font-family:monospace;font-size:16px"><br></span></div>=
<div><span style=3D"font-family:monospace;font-size:16px">Please have a rea=
d and let me know your thoughts</span></div><div><span style=3D"font-family=
:monospace;font-size:16px"><br>


</span></div><div><span style=3D"font-family:monospace;font-size:16px">Chee=
rs</span></div><div><span style=3D"font-family:monospace;font-size:16px">Su=
has</span></div><span style=3D"font-family:monospace;font-size:16px"><div><=
span style=3D"font-family:monospace;font-size:16px"><br>


</span></div>A new version of I-D, draft-nandakumar-mmusic-rtcweb-grouping-=
00.txt</span><br style=3D"font-family:monospace;font-size:16px"><span style=
=3D"font-family:monospace;font-size:16px">has been successfully submitted b=
y Suhas Nandakumar and posted to the</span><br style=3D"font-family:monospa=
ce;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">IETF repository.</span=
><br style=3D"font-family:monospace;font-size:16px"><br style=3D"font-famil=
y:monospace;font-size:16px"><span style=3D"font-family:monospace;font-size:=
16px">Filename:=A0=A0=A0=A0=A0=A0=A0 draft-nandakumar-mmusic-rtcweb-groupin=
g</span><br style=3D"font-family:monospace;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">Revision:=A0=A0=A0=A0=
=A0=A0=A0 00</span><br style=3D"font-family:monospace;font-size:16px"><span=
 style=3D"font-family:monospace;font-size:16px">Title:=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0 SSRC Group Based Simulcast Signaling</span><br style=3D"font-fami=
ly:monospace;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">Creation date:=A0=A0 2=
013-07-15</span><br style=3D"font-family:monospace;font-size:16px"><span st=
yle=3D"font-family:monospace;font-size:16px">Group:=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0 Individual Submission</span><br style=3D"font-family:monospace;font-=
size:16px">


<span style=3D"font-family:monospace;font-size:16px">Number of pages: 8</sp=
an><br style=3D"font-family:monospace;font-size:16px"><span style=3D"font-f=
amily:monospace;font-size:16px">URL:=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
</span><a href=3D"https://mail.cisco.com/owa/redir.aspx?C=3DKlIQ10YF-Eu3HiM=
HIouDdKX4cq8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&amp;URL=3Dhtt=
p%3a%2f%2fwww.ietf.org%2finternet-drafts%2fdraft-nandakumar-mmusic-rtcweb-g=
rouping-00.txt" style=3D"font-family:monospace;font-size:16px" target=3D"_b=
lank">http://www.ietf.org/internet-drafts/draft-nandakumar-mmusic-rtcweb-gr=
ouping-00.txt</a><br style=3D"font-family:monospace;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">Status:=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0</span><a href=3D"https://mail.cisco.com/owa/redir.aspx?C=3D=
KlIQ10YF-Eu3HiMHIouDdKX4cq8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY=
.&amp;URL=3Dhttp%3a%2f%2fdatatracker.ietf.org%2fdoc%2fdraft-nandakumar-mmus=
ic-rtcweb-grouping" style=3D"font-family:monospace;font-size:16px" target=
=3D"_blank">http://datatracker.ietf.org/doc/draft-nandakumar-mmusic-rtcweb-=
grouping</a><br style=3D"font-family:monospace;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">Htmlized:=A0=A0=A0=A0=
=A0=A0=A0=A0</span><a href=3D"https://mail.cisco.com/owa/redir.aspx?C=3DKlI=
Q10YF-Eu3HiMHIouDdKX4cq8VVdAIUowNDR3V500RRSx6WMo4z3rBmcn7h2cdn9nodpYC6pY.&a=
mp;URL=3Dhttp%3a%2f%2ftools.ietf.org%2fhtml%2fdraft-nandakumar-mmusic-rtcwe=
b-grouping-00" style=3D"font-family:monospace;font-size:16px" target=3D"_bl=
ank">http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-00<=
/a><br style=3D"font-family:monospace;font-size:16px">


<br style=3D"font-family:monospace;font-size:16px"><br style=3D"font-family=
:monospace;font-size:16px"><span style=3D"font-family:monospace;font-size:1=
6px">Abstract:</span><br style=3D"font-family:monospace;font-size:16px"><sp=
an style=3D"font-family:monospace;font-size:16px">=A0=A0 In some applicatio=
ns it may be necessary to send multiple media</span><br style=3D"font-famil=
y:monospace;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">=A0=A0 encodings corre=
sponding to a media source with in independent RTP</span><br style=3D"font-=
family:monospace;font-size:16px"><span style=3D"font-family:monospace;font-=
size:16px">=A0=A0 media streams.=A0 This is called Simulcast.=A0 This docum=
ent discusses a</span><br style=3D"font-family:monospace;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">=A0=A0 framework for d=
escribing simulcast media streams in SDP and also</span><br style=3D"font-f=
amily:monospace;font-size:16px"><span style=3D"font-family:monospace;font-s=
ize:16px">=A0=A0 defines semantics to express relationship amongst them.=A0=
 The</span><br style=3D"font-family:monospace;font-size:16px">


<span style=3D"font-family:monospace;font-size:16px">=A0=A0 semantics defin=
ed in this document are to be used with the source</span><br style=3D"font-=
family:monospace;font-size:16px"><span style=3D"font-family:monospace;font-=
size:16px">=A0=A0 specific grouping framework defined in the [RFC5576]</spa=
n>
</blockquote></div><br></div>

--047d7bf10aea6b3ecd04e1911dd1--

From emil@sip-communicator.org  Mon Jul 15 11:44:36 2013
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Date: Mon, 15 Jul 2013 20:44:28 +0200
From: Emil Ivov <emcho@jitsi.org>
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Subject: [MMUSIC] Semantics of an "m=" line
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Hey all,

Just submitted a new draft that should server as a basis for discussion 
on "m=" line semantics in offer answer:

http://tools.ietf.org/html/draft-ivov-mmusic-multiple-sources-00

The goal of the draft is to discuss the use of "m=" lines by 
Offer/Answer according to their original design in SDP: as envelopes for 
RTP sessions with any number of SSRCs.

In many ways, this new document describes the basis of "No Plan" but in 
a generic way, not related to WebRTC.

Feedback is welcome and hopefully we will be able to also discuss this 
in Berlin.

Cheers,
Emil
-- 
https://jitsi.org

From adam@nostrum.com  Mon Jul 15 12:05:35 2013
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: [MMUSIC] Unified Plan for SDP Handling
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[Cross-posting to RTCWEB; follow-ups to MMUSIC, please]

After significant work, Justin, Martin and I have managed to produce a 
compromise plan that provides a high degree of interoperability with 
existing devices (and future non-WebRTC devices) while not being 
excessively onerous for WebRTC implementations or applications that use 
them. It's been a tricky balancing act, but I think we've found a good 
mix between the two that can form a solid basis for the working group to 
move forward.

Rather than summarize the key points of the document in this email, I 
direct interested parties to section 2 of the document, which summarizes 
the key aspects of the plan in eight relatively concise bullet points.

I apologize for the late publication date of this document -- there's 
actually been a lot more work put into coming up with a unified draft 
than I originally anticipated, and the production of this document took 
at least two weeks longer than I expected it to.

Note that this document is intended to be a plan for the work to be done 
in this area, and not a specification in itself. The intention is that 
its contents are used as the basis for work in several other drafts -- 
some new, some not -- that form the corpus of work necessary for RTCWEB 
(and potentially CLUE) to move forward. Except in rare cases, the 
document does not attempt to explicitly call out venues or documents for 
such work, as we (or, at the very least, I) anticipate guidance from the 
various working group chairs to assist in such decisions.

Comments prior to Berlin would be very helpful, although this will 
clearly be a point of significant discussion at the face-to-face meeting.

Document link:

http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt

/a

From internet-drafts@ietf.org  Mon Jul 15 12:12:40 2013
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Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-latching-03.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-03.txt
	Pages           : 14
	Date            : 2013-07-15

Abstract:
   This document describes behavior of signalling intermediaries in
   Real-Time Communication (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-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-latching-03


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


From pkyzivat@alum.mit.edu  Mon Jul 15 13:01:43 2013
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Subject: Re: [MMUSIC] Semantics of an "m=" line
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Emil,

This is a useful draft. It makes some good point. I have some 
comments/questions:

Section 1:

    o  the addition or removal of simulcast or layered streams must be
       possible without the need for Offer/Answer exchanges beyond the
       initial declaration of such capabilities for either direction.

ISTM that simulcast needs more discussion - it is nuanced. E.g. the 
following are all things one might like to say about simulcast:
- I am sending the same thing in multiple formats. You can pick
   the one(s) you want to receive. (Especially for broadcast/multicast)

- I am preparted to send the same thing in multiple formats.
   You can pick the one(s) you want me to send.

- I want to receive a single source in multiple formats.

Certainly there is need for the simulcast indication in SDP for the 
multicast case. And probably in some of the others. But I am not 
convinced it is always necessary for simulcast to be indicated in SDP. 
In some cases, there may be no need for the receiver to know that two 
streams it is receiving happen to be simulcasts on one another. And in 
other cases it will already know by some means. (Which is your point.)

Also Section 1:

    o  application specific signalling should be used to cover most
       semantics following call establishment, such as adding, removing
       or identifying SSRCs;

I think this is the controversial point. In *some* cases, where the 
semantics are complex and dynamic, this makes sense. But there are no 
doubt many cases that are relatively simple, where its much more 
convenient to do something in SDP than it is to define a new application 
channel to negotiate this.

Section 4: Demuxing and Identifying Streams

The demux based solely on PT is fine if bundling is only used to combine 
m-lines with different media types. In that case there will be at most 
one bundled m-line per top level mime type. And this satisfies my 
requirement that it always be possible to map a packet to an m-line.

But if bundling is used for any other reason, then the number of likely 
m-lines goes up, and the acceptability of unique PTs goes down. Then the 
discussion changes.

I don't think the description of what must be done is quite right. You 
don't quite say what a depacketizing/decoding/processing chain *is*. Do 
you intend this as a *type* of chain, or an *instance* of a chain? I 
think it has to be *instance*. E.g. there may be two VP8 chains, with 
all the same settings, but processing different flows. Simplistically, 
you might say that the *type* of chain can be determined by the PT, and 
the *instance* of that chain can be determined by the SSRC.

But typically such chains would be assigned based on where they are 
intended to be rendered. (Equal number of chains and renderers.) So when 
a new SSRC is received, and there is no matching chain, then either it 
must be dropped, or else some existing chain must be reconfigured to 
process this SSRC. This may need to happen without any additional out of 
band info to help. How do you imagine this being done? (In simple cases 
this might be done by *policy*. But I don't know if that is enough.) Is 
it your position that if this can't be done by policy then there must be 
some application-specific signaling mechanism to do it?

	Thanks,
	Paul

On 7/15/13 2:44 PM, Emil Ivov wrote:
> Hey all,
>
> Just submitted a new draft that should server as a basis for discussion
> on "m=" line semantics in offer answer:
>
> http://tools.ietf.org/html/draft-ivov-mmusic-multiple-sources-00
>
> The goal of the draft is to discuss the use of "m=" lines by
> Offer/Answer according to their original design in SDP: as envelopes for
> RTP sessions with any number of SSRCs.
>
> In many ways, this new document describes the basis of "No Plan" but in
> a generic way, not related to WebRTC.
>
> Feedback is welcome and hopefully we will be able to also discuss this
> in Berlin.
>
> Cheers,
> Emil


From roman@telurix.com  Mon Jul 15 13:46:57 2013
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Subject: Re: [MMUSIC] [rtcweb] Unified Plan for SDP Handling
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Quick nit on the draft: In initial examples with "group:BUNDLE S1 S2 S3"
mid attributes should be "a=mid:S1", "a=mid:S2", and "a=mid:S3", not "a=mid:1",
"a=mid:2", "a=mid:3"
_____________
Roman Shpount

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Quick nit on the draft: In initial examples with &quot;<span style=3D"white=
-space:pre-wrap">group:BUNDLE S1 S2 S3&quot; mid attributes should be &quot=
;</span><span style=3D"white-space:pre-wrap">a=3Dmid:S1&quot;, &quot;</span=
><span style=3D"white-space:pre-wrap">a=3Dmid:S2&quot;, and </span><span st=
yle=3D"white-space:pre-wrap">&quot;</span><span style=3D"white-space:pre-wr=
ap">a=3Dmid:S3&quot;, not </span><span style=3D"white-space:pre-wrap">&quot=
;</span><span style=3D"white-space:pre-wrap">a=3Dmid:1&quot;, </span><span =
style=3D"white-space:pre-wrap">&quot;</span><span style=3D"white-space:pre-=
wrap">a=3Dmid:2&quot;, </span><span style=3D"white-space:pre-wrap">&quot;</=
span><span style=3D"white-space:pre-wrap">a=3Dmid:3&quot;</span><div>
_____________<br>Roman Shpount</div>
<br><br>

--047d7bb04c90a6134504e192f53c--

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From: Suhas Nandakumar <suhasietf@gmail.com>
To: mmusic WG <mmusic@ietf.org>, rtcweb@ietf.org
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Subject: [MMUSIC] New Version Notification for draft-nandakumar-rtcweb-sdp-02.txt
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--089e013d0f800b62fb04e19302da
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Hello All

  An updated version of the draft showcasing SDP examples for the most
common WebRTC use-cases has be submitted.

I would recommend PDF version since that shows embedded images.

Please let us know your thoughts.

Cheers
Suhas

Filename:        draft-nandakumar-rtcweb-sdp
Revision:        02
Title:           SDP for the WebRTC
Creation date:   2013-07-12
Group:           Individual Submission
Number of pages: 64
URL:
http://www.ietf.org/internet-drafts/draft-nandakumar-rtcweb-sdp-02.txt
Status:          http://datatracker.ietf.org/doc/draft-nandakumar-rtcweb-sdp
Htmlized:        http://tools.ietf.org/html/draft-nandakumar-rtcweb-sdp-02
Diff:
http://www.ietf.org/rfcdiff?url2=draft-nandakumar-rtcweb-sdp-02

Abstract:
   The Web Real-Time Communication (WebRTC) [WEBRTC] working group is
   charged to provide protocol support for direct interactive rich
   communication using audio,video and data between two peers' web
   browsers.  With in the WebRTC framework, Session Description protocol
   (SDP) [RFC4566] is used for negotiating session capabilities between
   the peers.  Such a negotiataion happens based on the SDP Offer/Answer
   exchange mechanism described in the RFC 3264 [RFC3264].

   This document serves a introductory purpose in describing the role of
   SDP for the most common WebRTC use-cases.

   This SDP examples provided in this document is still a work in
   progress, but aims to align closest to the evolving standards.




The IETF Secretariat

--089e013d0f800b62fb04e19302da
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Hello All<div><br></div><div>=A0 An updated version of the draft showcasing=
 SDP examples for the most common WebRTC use-cases has be submitted.</div><=
div><br></div><div>I would recommend PDF version since that shows embedded =
images.</div>
<div><br></div><div>Please let us know your thoughts.</div><div><br></div><=
div>Cheers</div><div>Suhas<br><div class=3D"gmail_quote"><br>Filename: =A0 =
=A0 =A0 =A0draft-nandakumar-rtcweb-sdp<br>
Revision: =A0 =A0 =A0 =A002<br>
Title: =A0 =A0 =A0 =A0 =A0 SDP for the WebRTC<br>
Creation date: =A0 2013-07-12<br>
Group: =A0 =A0 =A0 =A0 =A0 Individual Submission<br>
Number of pages: 64<br>
URL: =A0 =A0 =A0 =A0 =A0 =A0 <a href=3D"http://www.ietf.org/internet-drafts=
/draft-nandakumar-rtcweb-sdp-02.txt" target=3D"_blank">http://www.ietf.org/=
internet-drafts/draft-nandakumar-rtcweb-sdp-02.txt</a><br>
Status: =A0 =A0 =A0 =A0 =A0<a href=3D"http://datatracker.ietf.org/doc/draft=
-nandakumar-rtcweb-sdp" target=3D"_blank">http://datatracker.ietf.org/doc/d=
raft-nandakumar-rtcweb-sdp</a><br>
Htmlized: =A0 =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-nandak=
umar-rtcweb-sdp-02" target=3D"_blank">http://tools.ietf.org/html/draft-nand=
akumar-rtcweb-sdp-02</a><br>
Diff: =A0 =A0 =A0 =A0 =A0 =A0<a href=3D"http://www.ietf.org/rfcdiff?url2=3D=
draft-nandakumar-rtcweb-sdp-02" target=3D"_blank">http://www.ietf.org/rfcdi=
ff?url2=3Ddraft-nandakumar-rtcweb-sdp-02</a><br>
<br>
Abstract:<br>
=A0 =A0The Web Real-Time Communication (WebRTC) [WEBRTC] working group is<b=
r>
=A0 =A0charged to provide protocol support for direct interactive rich<br>
=A0 =A0communication using audio,video and data between two peers&#39; web<=
br>
=A0 =A0browsers. =A0With in the WebRTC framework, Session Description proto=
col<br>
=A0 =A0(SDP) [RFC4566] is used for negotiating session capabilities between=
<br>
=A0 =A0the peers. =A0Such a negotiataion happens based on the SDP Offer/Ans=
wer<br>
=A0 =A0exchange mechanism described in the RFC 3264 [RFC3264].<br>
<br>
=A0 =A0This document serves a introductory purpose in describing the role o=
f<br>
=A0 =A0SDP for the most common WebRTC use-cases.<br>
<br>
=A0 =A0This SDP examples provided in this document is still a work in<br>
=A0 =A0progress, but aims to align closest to the evolving standards.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div>

--089e013d0f800b62fb04e19302da--

From iesg-secretary@ietf.org  Mon Jul 15 15:15:43 2013
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Subject: [MMUSIC] Protocol Action: 'Miscellaneous Capabilities Negotiation in the	Session Description Protocol (SDP)' to Proposed Standard	(draft-ietf-mmusic-sdp-miscellaneous-caps-07.txt)
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The IESG has approved the following document:
- 'Miscellaneous Capabilities Negotiation in the Session Description
   Protocol (SDP)'
  (draft-ietf-mmusic-sdp-miscellaneous-caps-07.txt) as Proposed Standard

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

The IESG contact persons are Gonzalo Camarillo and Richard Barnes.

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




*Technical Summary*

SDP has been extended with a capability negotiation mechanism based on
SDP Offer/Answer, which allows endpoints to negotiate transport
protocols, attributes as well additional media-related capabilities.

This document extends the SDP capability negotiation framework to allow
endpoints to negotiate three additional SDP capabilities. In particular,
this document provides a mechanism to negotiate bandwidth ('b=' line),
connection data ('c=' line), and titles ('i=' line for each session or
media) in SDP.

*Working Group Summary*

The document was adopted as a WG item in March 2012 and is an external
and long-standing dependency for 3GPP. The mechanisms in the document
are fairly straightforward and did not receive a lot of discussion in
the WG subsequently, and a WGLC was issued on the -02 version of the
draft in October 2012. There were no major comments received but a chair
review let to minor updates in the document resulting in the -04 version
in March 2013. At this point, several people noted lack of clarity wrt
potential overlap with ICE (RFC 5245) resulting in a somewhat
controversial debate on the mailing list about what the WG had agreed to
work on and timeliness of feedback on documents that had already been
through WGLC. See threads in

http://www.ietf.org/mail-archive/web/mmusic/current/msg10471.html
http://www.ietf.org/mail-archive/web/mmusic/current/msg10573.html

The chairs and document authors have subsequently worked to address the
concerns. While there has been limited explicit feedback on the current
wording, we have requested feedback several times, and lastly in
http://www.ietf.org/mail-archive/web/mmusic/current/msg10962.html which
resulted in no further comments received. Consequently, we believe all
concerns have been addressed at this point.

*Document Quality*

There are currently no known implementations of the draft, however the
draft is a dependency for 3GPP, so future implementations are expected.

Earlier versions of the draft have been reviewed by a couple of WG
participants and the document shepherd has reviewed the last several
versions in detail. As noted above, the connection data capability part
of the document has received detailed feedback more recently as well.
The major contributors to these as well as earlier discussions are
listed in the Acknowledgements section of the document.

*Personnel*

Document Shepherd: Flemming Andreasen
Responsible AD: Gonzalo Camarillo

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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-rfc5245bis-00.txt
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Hi Emil,

On 7/15/13 9:04 PM, Emil Ivov wrote:
> On Mon, Jul 15, 2013 at 7:48 PM, Ari Keränen <ari.keranen@ericsson.com> wrote:
>> We've submitted the ICE drafts as WG documents, as agreed at the previous
>> meeting. They also address the issues that were discussed during the meeting
>> and/or on the list:
>> - clarified the use of short term credentials in connectivity checks
>> - not using deprecated IPv6 addresses or loop back addresses for candidates
>
> Wouldn't it make sense to allow pairing loopback addresses with other
> loopback addresses? The main use case that comes to mind is debug
> sessions where both the transmitter and the receiver are on the same
> host?

Even on the same host debugging you should be able to use any IP address 
of the host for that purpose. That said, of course if you *only* have 
the loopback interface, then this might be useful. Perhaps it should be 
made "SHOULD NOT" strength?


Cheers,
Ari (as individual)

From emil@sip-communicator.org  Tue Jul 16 03:35:40 2013
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Date: Tue, 16 Jul 2013 12:35:31 +0200
From: Emil Ivov <emcho@jitsi.org>
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Cc: mmusic <mmusic@ietf.org>
Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-rfc5245bis-00.txt
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On 16.07.13, 09:07, Ari Ker=E4nen wrote:
> Hi Emil,
>
> On 7/15/13 9:04 PM, Emil Ivov wrote:
>> On Mon, Jul 15, 2013 at 7:48 PM, Ari Ker=E4nen <ari.keranen@ericsson.c=
om> wrote:
>>> We've submitted the ICE drafts as WG documents, as agreed at the prev=
ious
>>> meeting. They also address the issues that were discussed during the =
meeting
>>> and/or on the list:
>>> - clarified the use of short term credentials in connectivity checks
>>> - not using deprecated IPv6 addresses or loop back addresses for cand=
idates
>>
>> Wouldn't it make sense to allow pairing loopback addresses with other
>> loopback addresses? The main use case that comes to mind is debug
>> sessions where both the transmitter and the receiver are on the same
>> host?
>
> Even on the same host debugging you should be able to use any IP addres=
s
> of the host for that purpose. That said, of course if you *only* have
> the loopback interface, then this might be useful.

That's the case I was referring to. Someone made a comment in Orlando=20
about how loopback addresses had helped them debug a WebRTC app while in =

the plane.

> Perhaps it should be made "SHOULD NOT" strength?

I think "MAY choose not to pair" would work even better. It's mostly an=20
implementation choice and choosing one way or the other doesn't=20
profoundly break things.

Emil

--=20
https://jitsi.org


From ari.keranen@ericsson.com  Tue Jul 16 04:25:55 2013
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On 7/16/13 1:35 PM, Emil Ivov wrote:
> On 16.07.13, 09:07, Ari Keränen wrote:
>> Hi Emil,
>>
>> On 7/15/13 9:04 PM, Emil Ivov wrote:
>>> On Mon, Jul 15, 2013 at 7:48 PM, Ari Keränen
>>> <ari.keranen@ericsson.com> wrote:
>>>> We've submitted the ICE drafts as WG documents, as agreed at the
>>>> previous
>>>> meeting. They also address the issues that were discussed during the
>>>> meeting
>>>> and/or on the list:
>>>> - clarified the use of short term credentials in connectivity checks
>>>> - not using deprecated IPv6 addresses or loop back addresses for
>>>> candidates
>>>
>>> Wouldn't it make sense to allow pairing loopback addresses with other
>>> loopback addresses? The main use case that comes to mind is debug
>>> sessions where both the transmitter and the receiver are on the same
>>> host?
>>
>> Even on the same host debugging you should be able to use any IP address
>> of the host for that purpose. That said, of course if you *only* have
>> the loopback interface, then this might be useful.
>
> That's the case I was referring to. Someone made a comment in Orlando
> about how loopback addresses had helped them debug a WebRTC app while in
> the plane.

I wonder how common use case that's gonna be :)

>> Perhaps it should be made "SHOULD NOT" strength?
>
> I think "MAY choose not to pair" would work even better. It's mostly an
> implementation choice and choosing one way or the other doesn't
> profoundly break things.

While it doesn't profoundly break things, it may cause significant delay 
and hence I think it's a bad idea. If the default behavior would end up 
having the loopback address in the offers/answers and pairing it with 
all other candidates ("since it's only MAY strength - I don't need to 
implement that check"), that would generate a whole bunch of 
high-priority pairs that are almost 100% sure to fail. Moreover, both 
endpoints should do the same wrt pairing to keep the checks in sync, so 
IMHO it should be fairly strong do or don't (hence at least SHOULD).


Cheers,
Ari

From keith.drage@alcatel-lucent.com  Tue Jul 16 07:58:33 2013
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "clue@ietf.org" <clue@ietf.org>, "dispatch@ietf.org" <dispatch@ietf.org>, "avt@ietf.org" <avt@ietf.org>
Thread-Topic: Taxonomy
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Date: Tue, 16 Jul 2013 14:58:24 +0000
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Subject: [MMUSIC] FW: Taxonomy
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Extensive cross-post do not reply to this email.

For information. We do expect participation input from all the listed worki=
ng groups (and possibly more).

Keith

-----Original Message-----
From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On Behalf Of=
 DRAGE, Keith (Keith)
Sent: 16 July 2013 15:53
To: avtext@ietf.org
Subject: [avtext] Taxonomy

(As AVTEXT WG cochair)

The latest version of the RTP taxonomy draft has been submitted.

https://datatracker.ietf.org/doc/draft-lennox-raiarea-rtp-grouping-taxonomy=
/=20

This document was allocated to AVTEXT by DISPATCH, and we have created a mi=
lestone for it in AVTEXT.

This draft is not yet a working group document.

We do expect to spend some significant meeting time discussing the contents=
 in the AVTEXT meeting in Berlin, so please start raising your issues on th=
e AVTEXT list.

Additionally, if there are counter proposals existing that the AVTEXT chair=
s would not otherwise be aware of, then please identify them to the AVTEXT =
chairs / list.

Regards

Keith
_______________________________________________
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From ron.even.tlv@gmail.com  Tue Jul 16 13:30:31 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: <mmusic@ietf.org>
References: <51E447C8.1000701@nostrum.com>
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling
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Adam,
For correlation when the SSRC is not known in the SDP look at
http://tools.ietf.org/id/draft-even-mmusic-multiple-streams-02.txt and the
option to use SDES and not only RTP header extension.

BTW: there are other cases when the SSRC is not known using SDP before the
RTP stream arrives like when an MCU switch the video. Note that SDES is
distributed to the session when the source joins the RTP session which can
be long before the RTP media is sent to a specific receiver in the group.

Roni

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Adam Roach
> Sent: 15 July, 2013 10:05 PM
> To: mmusic@ietf.org
> Cc: rtcweb@ietf.org
> Subject: [MMUSIC] Unified Plan for SDP Handling
> 
> [Cross-posting to RTCWEB; follow-ups to MMUSIC, please]
> 
> After significant work, Justin, Martin and I have managed to produce a
> compromise plan that provides a high degree of interoperability with
existing
> devices (and future non-WebRTC devices) while not being excessively
> onerous for WebRTC implementations or applications that use them. It's
> been a tricky balancing act, but I think we've found a good mix between
the
> two that can form a solid basis for the working group to move forward.
> 
> Rather than summarize the key points of the document in this email, I
direct
> interested parties to section 2 of the document, which summarizes the key
> aspects of the plan in eight relatively concise bullet points.
> 
> I apologize for the late publication date of this document -- there's
actually
> been a lot more work put into coming up with a unified draft than I
originally
> anticipated, and the production of this document took at least two weeks
> longer than I expected it to.
> 
> Note that this document is intended to be a plan for the work to be done
in
> this area, and not a specification in itself. The intention is that its
contents are
> used as the basis for work in several other drafts -- some new, some not
--
> that form the corpus of work necessary for RTCWEB (and potentially CLUE)
to
> move forward. Except in rare cases, the document does not attempt to
> explicitly call out venues or documents for such work, as we (or, at the
very
> least, I) anticipate guidance from the various working group chairs to
assist in
> such decisions.
> 
> Comments prior to Berlin would be very helpful, although this will clearly
be a
> point of significant discussion at the face-to-face meeting.
> 
> Document link:
> 
> http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt
> 
> /a
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From adam@nostrum.com  Tue Jul 16 13:39:02 2013
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling
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On 7/16/13 15:28, Roni Even wrote:
> Adam,
> For correlation when the SSRC is not known in the SDP look at
> http://tools.ietf.org/id/draft-even-mmusic-multiple-streams-02.txt and the
> option to use SDES and not only RTP header extension.

Roni:

Thanks for the pointer. That's an interesting approach; however, for 
RTCWEB at least, need a solution that also works with DTLS-SRTP.

/a

From ron.even.tlv@gmail.com  Tue Jul 16 14:50:42 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
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Date: Wed, 17 Jul 2013 00:48:50 +0300
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling - multiple media sources
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Hi,
Some comments. 
In section 2 bullet 2 is changing RFC 5576 semantics and not for good. Why
do you want more than one SSRC for m-line but allow only one at a time, is
this for the simulcast case?

Others which are not on section 2

1. The document talks about the case when there are multiple participants
each sending multiple streams as mentioned in section 1.1.1 and the receiver
may want to select among all these sources as discussed in section 1.1.2.
The document focus on the point to point case and does not address the issue
of how all the m-lines from all participants are distributes and how a late
joiner gets all this information.

2. In the examples using the ssrc attribute note that according to RFC 5576
section 4.1 you must provide for each SSRC the CNAME attribute

3. The simulcast example and description allows the case of sending one of
the resolutions at a time but in simulcast the source may be able to send
multiple resolutions at the same time. To further complicate this case the
offer can support three resolutions but can only send any two at a time
which the receiver can select. How do you signal this, my view is with the
maxsendssrc attribute.

4. in section 3.2.1.1 "as well as a 16-bit identifier (expressed as an
integer) used for correlation;" I think that it must be a byte string or
token not integer

Roni



> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Adam Roach
> Sent: 15 July, 2013 10:05 PM
> To: mmusic@ietf.org
> Cc: rtcweb@ietf.org
> Subject: [MMUSIC] Unified Plan for SDP Handling
> 
> [Cross-posting to RTCWEB; follow-ups to MMUSIC, please]
> 
> After significant work, Justin, Martin and I have managed to produce a
> compromise plan that provides a high degree of interoperability with
existing
> devices (and future non-WebRTC devices) while not being excessively
> onerous for WebRTC implementations or applications that use them. It's
> been a tricky balancing act, but I think we've found a good mix between
the
> two that can form a solid basis for the working group to move forward.
> 
> Rather than summarize the key points of the document in this email, I
direct
> interested parties to section 2 of the document, which summarizes the key
> aspects of the plan in eight relatively concise bullet points.
> 
> I apologize for the late publication date of this document -- there's
actually
> been a lot more work put into coming up with a unified draft than I
originally
> anticipated, and the production of this document took at least two weeks
> longer than I expected it to.
> 
> Note that this document is intended to be a plan for the work to be done
in
> this area, and not a specification in itself. The intention is that its
contents are
> used as the basis for work in several other drafts -- some new, some not
--
> that form the corpus of work necessary for RTCWEB (and potentially CLUE)
to
> move forward. Except in rare cases, the document does not attempt to
> explicitly call out venues or documents for such work, as we (or, at the
very
> least, I) anticipate guidance from the various working group chairs to
assist in
> such decisions.
> 
> Comments prior to Berlin would be very helpful, although this will clearly
be a
> point of significant discussion at the face-to-face meeting.
> 
> Document link:
> 
> http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt
> 
> /a
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From emil@sip-communicator.org  Tue Jul 16 14:52:20 2013
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Subject: Re: [MMUSIC] Semantics of an "m=" line
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Hey Paul,

On 15.07.13, 22:01, Paul Kyzivat wrote:
> Emil,
>
> This is a useful draft. It makes some good point. I have some
> comments/questions:
>
> Section 1:
>
>      o  the addition or removal of simulcast or layered streams must be
>         possible without the need for Offer/Answer exchanges beyond the
>         initial declaration of such capabilities for either direction.
>
> ISTM that simulcast needs more discussion - it is nuanced.

Completely agreed.

> E.g. the
> following are all things one might like to say about simulcast:
> - I am sending the same thing in multiple formats. You can pick
>     the one(s) you want to receive. (Especially for broadcast/multicast)
>
> - I am preparted to send the same thing in multiple formats.
>     You can pick the one(s) you want me to send.
>
> - I want to receive a single source in multiple formats.
>
> Certainly there is need for the simulcast indication in SDP for the
> multicast case. And probably in some of the others. But I am not
> convinced it is always necessary for simulcast to be indicated in SDP.
> In some cases, there may be no need for the receiver to know that two
> streams it is receiving happen to be simulcasts on one another. And in
> other cases it will already know by some means. (Which is your point.)

Yes.

> Also Section 1:
>
>      o  application specific signalling should be used to cover most
>         semantics following call establishment, such as adding, removing
>         or identifying SSRCs;
>
> I think this is the controversial point. In *some* cases, where the
> semantics are complex and dynamic, this makes sense. But there are no
> doubt many cases that are relatively simple, where its much more
> convenient to do something in SDP than it is to define a new application
> channel to negotiate this.

I agree and the draft doesn't try to argue against this. I suppose in 
these cases SDP can be viewed as *the* application specific signalling.

When doing this however, it would make sense to have a fallback 
strategy. That is, unless I know that the remote party supports these 
stream control semantics, I most not rely on them and I stick to the 
basic offer/answer exchange. That's what we generally do with new 
Offer/Answer features, right?

> Section 4: Demuxing and Identifying Streams
>
> The demux based solely on PT is fine if bundling is only used to combine
> m-lines with different media types. In that case there will be at most
> one bundled m-line per top level mime type. And this satisfies my
> requirement that it always be possible to map a packet to an m-line.
>
> But if bundling is used for any other reason, then the number of likely
> m-lines goes up, and the acceptability of unique PTs goes down. Then the
> discussion changes.

I agree, with the same comment as above: you should only use multiple 
"m=" lines for the same media type if you know that your peer would 
understand what this means. If not then you stick to one "m=" line per 
type and add an option indicating you support a multi-line mechanism.

This multi-line mechanism, like for example Plan A (a.k.a. "unified 
plan"), will then define demux procedures that work with it.  Does this 
make sense?

> I don't think the description of what must be done is quite right. You
> don't quite say what a depacketizing/decoding/processing chain *is*. Do
> you intend this as a *type* of chain, or an *instance* of a chain?

I am not sure I understand how you define type and instance. Do you get 
one chain type per media type? If so, then I meant type.

> I
> think it has to be *instance*. E.g. there may be two VP8 chains, with
> all the same settings, but processing different flows. Simplistically,
> you might say that the *type* of chain can be determined by the PT, and
> the *instance* of that chain can be determined by the SSRC.

Yes agreed. The point of the draft is that you use PT to find a chain 
type. You can then use whatever you want (e.g. SSRC) to map to a 
specific instance. That part does  not need to happen in SDP though.

> But typically such chains would be assigned based on where they are
> intended to be rendered. (Equal number of chains and renderers.) So when
> a new SSRC is received, and there is no matching chain, then either it
> must be dropped, or else some existing chain must be reconfigured to
> process this SSRC. This may need to happen without any additional out of
> band info to help. How do you imagine this being done? (In simple cases
> this might be done by *policy*. But I don't know if that is enough.) Is
> it your position that if this can't be done by policy then there must be
> some application-specific signaling mechanism to do it?

Yes exactly. Note in this case application-specific does not have to 
mean SIP or XMPP. It can also be RTP header extensions, a semi-dynamic 
policy downloadable over http or even SDP (although, as Roni mentioned, 
you can have the same problem there).

Emil
>
> 	Thanks,
> 	Paul
>
> On 7/15/13 2:44 PM, Emil Ivov wrote:
>> Hey all,
>>
>> Just submitted a new draft that should server as a basis for discussion
>> on "m=" line semantics in offer answer:
>>
>> http://tools.ietf.org/html/draft-ivov-mmusic-multiple-sources-00
>>
>> The goal of the draft is to discuss the use of "m=" lines by
>> Offer/Answer according to their original design in SDP: as envelopes for
>> RTP sessions with any number of SSRCs.
>>
>> In many ways, this new document describes the basis of "No Plan" but in
>> a generic way, not related to WebRTC.
>>
>> Feedback is welcome and hopefully we will be able to also discuss this
>> in Berlin.
>>
>> Cheers,
>> Emil
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

-- 
https://jitsi.org

From jonathan@vidyo.com  Tue Jul 16 15:05:51 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Adam Roach <adam@nostrum.com>
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On Jul 16, 2013, at 4:38 PM, Adam Roach <adam@nostrum.com> wrote:

> On 7/16/13 15:28, Roni Even wrote:
>> Adam,
>> For correlation when the SSRC is not known in the SDP look at
>> http://tools.ietf.org/id/draft-even-mmusic-multiple-streams-02.txt and t=
he
>> option to use SDES and not only RTP header extension.
>=20
> Roni:
>=20
> Thanks for the pointer. That's an interesting approach; however, for=20
> RTCWEB at least, need a solution that also works with DTLS-SRTP.


Acronym collision: This uses RTCP Source Description packets, not SDP Secur=
ity Descriptions.

I think the (related) app-id approach described in http://tools.ietf.org/ht=
ml/draft-even-mmusic-application-token-00.txt will also be of interest.

--
Jonathan Lennox
jonathan@vidyo.com



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Hi Roni,

On 16 July 2013 14:48, Roni Even <ron.even.tlv@gmail.com> wrote:
> Hi,
> Some comments.
> In section 2 bullet 2 is changing RFC 5576 semantics and not for good. Why
> do you want more than one SSRC for m-line but allow only one at a time, is
> this for the simulcast case?

The text includes "Absent any other signaled extension, [...]" for
exactly that reason.  The plan is to include a specific extension for
simulcast, layering and FEC that would avoid this specific problem.

> Others which are not on section 2
>
> 1. The document talks about the case when there are multiple participants
> each sending multiple streams as mentioned in section 1.1.1 and the receiver
> may want to select among all these sources as discussed in section 1.1.2.
> The document focus on the point to point case and does not address the issue
> of how all the m-lines from all participants are distributes and how a late
> joiner gets all this information.

The assumption is that there is always a point-to-point leg that is
being negotiated.  That doesn't remove the need to combine information
from a multi-party session into a single description.

> 3. The simulcast example and description allows the case of sending one of
> the resolutions at a time but in simulcast the source may be able to send
> multiple resolutions at the same time. To further complicate this case the
> offer can support three resolutions but can only send any two at a time
> which the receiver can select. How do you signal this, my view is with the
> maxsendssrc attribute.

(Full disclosure, I didn't keep track of the changes in this area
effectively, my views may differ from Adam and Justin on this point.)

The combination of FID grouping and the newly defined SIMULCAST
grouping might provide enough information.  In that way, I can suggest
that I don't want to see both A and C at the same time, even though
A+B or B+C might be OK.

I'd rather not rely on maxsendssrc, which is based on a very different
set assumption on the meaning of an m-line to what this document
describes.

> 4. in section 3.2.1.1 "as well as a 16-bit identifier (expressed as an
> integer) used for correlation;" I think that it must be a byte string or
> token not integer

There have been serious concerns raised regarding size of packets,
even though the header extension is not sent on many packets.  While
it's true that a 16-bit identifier is restrictive, but you'll need to
motivate a size increase a little better than that.

From adam@nostrum.com  Tue Jul 16 15:25:50 2013
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On 7/16/13 16:48, Roni Even wrote:
> Hi,
> Some comments.
> In section 2 bullet 2 is changing RFC 5576 semantics and not for good. Why
> do you want more than one SSRC for m-line but allow only one at a time, is
> this for the simulcast case?

No, simulcast is signaled explicitly (I will point out and emphasize the 
phrase "absent any other signaled extension"). This is "default 
behavior" is intended to accommodate MCUs that perform packet switching 
for RTP (and similar architectures that require different timing and/or 
sequence number spaces on a single stream).

> Others which are not on section 2
>
> 1. The document talks about the case when there are multiple participants
> each sending multiple streams as mentioned in section 1.1.1 and the receiver
> may want to select among all these sources as discussed in section 1.1.2.
> The document focus on the point to point case and does not address the issue
> of how all the m-lines from all participants are distributes and how a late
> joiner gets all this information.

The intention is that these changes to the session be explicitly 
described via SDP.

> 2. In the examples using the ssrc attribute note that according to RFC 5576
> section 4.1 you must provide for each SSRC the CNAME attribute

I think we had that in some early examples, but they were later removed. 
I agree that the text in 5576 requires its presence, and see no reason 
not to add it to the examples.

> 3. The simulcast example and description allows the case of sending one of
> the resolutions at a time but in simulcast the source may be able to send
> multiple resolutions at the same time.

The intention is definitely that sending multiple resolutions in 
parallel would be facilitated. Do you see any specific text prohibiting 
doing so, or is this fallout from your misunderstanding of section 2 
bullet 2?

> 4. in section 3.2.1.1 "as well as a 16-bit identifier (expressed as an
> integer) used for correlation;" I think that it must be a byte string or
> token not integer

Do you have some rationale for your assertion?

The goal here is to limit expansion of the RTP packets as much as 
possible. Given that there is a natural limit of 65536 streams in a 
single RTP session, 16 bits for a correlator seems sufficient.

/a

From adam@nostrum.com  Tue Jul 16 15:36:15 2013
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On 7/16/13 17:25, Adam Roach wrote:
> The goal here is to limit expansion of the RTP packets as much as 
> possible. Given that there is a natural limit of 65536 streams in a 
> single RTP session, 16 bits for a correlator seems sufficient.

Gack. I had crosswired some bytefield lengths in my brain here. SSRCs 
are 32 bits, not 16. I'm game for increasing the size to 32 bits for the 
correlator if we really think we need more than 64k streams in a session 
-- but I'm going to ask for some justification if someone makes that 
argument.

/a

From jonathan@vidyo.com  Tue Jul 16 15:46:37 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Martin Thomson <martin.thomson@gmail.com>
Date: Tue, 16 Jul 2013 18:46:29 -0400
Thread-Topic: [MMUSIC] Unified Plan for SDP Handling - multiple media sources
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling - multiple media sources
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On Jul 16, 2013, at 6:12 PM, Martin Thomson <martin.thomson@gmail.com> wrot=
e:

>> 4. in section 3.2.1.1 "as well as a 16-bit identifier (expressed as an
>> integer) used for correlation;" I think that it must be a byte string or
>> token not integer
>=20
> There have been serious concerns raised regarding size of packets,
> even though the header extension is not sent on many packets.  While
> it's true that a 16-bit identifier is restrictive, but you'll need to
> motivate a size increase a little better than that.

Have you read http://tools.ietf.org/html/draft-even-mmusic-application-toke=
n-00.txt ?  This is a more fleshed-out version of the same idea, as I under=
stand the idea in Unified, written to be agnostic between Plans A / B / No =
Plan.

The important difference the version of the idea presented in application-t=
oken, and the version in Unified is, I think, to the fact that once you hav=
e the correlator (app-id) signaled in SDP, you no longer *need* the SSRC to=
 also be signaled -- the correlation can be done entirely based on the app-=
id, if you so desire.  This gives you a lot more flexibility, with less nee=
d for a fresh offer-answer when something changes.

In app-id, we chose to use a token, rather than an integer, for this field.=
  One important reason for this is that it's useful to be able to send this=
 identifier in RTCP SDES[*] packets, so you can pre-announce correlations b=
efore sending media; however, SDES fields are defined by RFC 3550 as UTF-8 =
strings.  (There are also some RTP architectural-cleanliness reasons for ha=
ving the primary location for the correlator be in RTCP, with the header ex=
tension being a rapid-convergence optimization.)=20

[*note: RTCP "SDES" (Source Description) packets are entirely unrelated to =
SDP "SDES" (Security Description) SRTP key exchange; this is an unfortunate=
 acronym collision.]

Both RFC 5285 header extensions and RTCP SDES fields are length-delimited, =
so making the app identifiers be variable-length has no intrinsic packet ov=
erhead cost.  A sender is always free to choose short identifiers.  (Probab=
ly the spec should make short identifiers RECOMMENDED.)  It adds a bit of i=
mplementation complexity, but it doesn't seem like very much compared with =
all the rest of an RTP pipeline.

--
Jonathan Lennox
jonathan@vidyo.com



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Subject: Re: [MMUSIC] Unified Plan for SDP Handling - multiple media sources
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On 16 July 2013 15:46, Jonathan Lennox <jonathan@vidyo.com> wrote:
> Have you read http://tools.ietf.org/html/draft-even-mmusic-application-token-00.txt ?  This is a more fleshed-out version of the same idea, as I understand the idea in Unified, written to be agnostic between Plans A / B / No Plan.

I have read that draft.  I don't see any major problems with it.  That
is, aside from some serious ugliness when it comes to no-plan to the
extent that I don't believe it to be practical (if I've understood the
intent).

My current leaning is toward a (small) number if the unified plan is
chosen, simply because so much else depends on the existence of
signaling; a no-plan-style approach would provide greater motivation
for having all the bits present.  That said, I don't have enough
information yet to be able to make the header extension size vs. need
for signaling trade-off.

From bernard_aboba@hotmail.com  Tue Jul 16 16:21:51 2013
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling
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The app-id idea is useful for multiple reasons.  One reason similar RTP ext=
ensions are used today is to avoid having to parse the payload to figure ou=
t what the stream is (e.g. looking for parameter sets or SEI NAL units).  I=
t is cleaner to solve this in as generic rather than a codec-specific way. =
 I see no conflict with no-plan=3B in fact=2C I believe no plan suggested a=
n RTP extension=2C no?

Jonathan Lennox <jonathan@vidyo.com> wrote:


On Jul 16=2C 2013=2C at 4:38 PM=2C Adam Roach <adam@nostrum.com> wrote:

> On 7/16/13 15:28=2C Roni Even wrote:
>> Adam=2C
>> For correlation when the SSRC is not known in the SDP look at
>> http://tools.ietf.org/id/draft-even-mmusic-multiple-streams-02.txt and t=
he
>> option to use SDES and not only RTP header extension.
>
> Roni:
>
> Thanks for the pointer. That's an interesting approach=3B however=2C for
> RTCWEB at least=2C need a solution that also works with DTLS-SRTP.


Acronym collision: This uses RTCP Source Description packets=2C not SDP Sec=
urity Descriptions.

I think the (related) app-id approach described in http://tools.ietf.org/ht=
ml/draft-even-mmusic-application-token-00.txt will also be of interest.

--
Jonathan Lennox
jonathan@vidyo.com


_______________________________________________
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling
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On 17.07.13, 01:21, Bernard Aboba wrote:
> The app-id idea is useful for multiple reasons.  One reason similar
> RTP extensions are used today is to avoid having to parse the payload
> to figure out what the stream is (e.g. looking for parameter sets or
> SEI NAL units).  It is cleaner to solve this in as generic rather
> than a codec-specific way.  I see no conflict with no-plan;

+1

> in fact, I believe no plan suggested an RTP extension, no?

It did indeed.

Emil

> Jonathan Lennox <jonathan@vidyo.com> wrote:
>
>
> On Jul 16, 2013, at 4:38 PM, Adam Roach <adam@nostrum.com> wrote:
>
>> On 7/16/13 15:28, Roni Even wrote:
>>> Adam, For correlation when the SSRC is not known in the SDP look
>>> at
>>> http://tools.ietf.org/id/draft-even-mmusic-multiple-streams-02.txt
>>> and the option to use SDES and not only RTP header extension.
>>
>> Roni:
>>
>> Thanks for the pointer. That's an interesting approach; however,
>> for RTCWEB at least, need a solution that also works with
>> DTLS-SRTP.
>
>
> Acronym collision: This uses RTCP Source Description packets, not SDP
> Security Descriptions.
>
> I think the (related) app-id approach described in
> http://tools.ietf.org/html/draft-even-mmusic-application-token-00.txt
> will also be of interest.
>
> -- Jonathan Lennox jonathan@vidyo.com
>
>
> _______________________________________________ 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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Yes=2C it is annoying when new docs attempt to redefine behavior in existin=
g deployed RFCs.  RFC 5576 has multiple implementations today=2C and none o=
f them have a one-at-a-time restriction.

The problem with max-send-ssrc is that it conflates ssrcs with streams. If =
you have layering or simulcast you need to have something like max-streams =
and max-sources. max-send-ssrc is confusing.

Roni Even <ron.even.tlv@gmail.com> wrote:

Hi=2C
Some comments.
In section 2 bullet 2 is changing RFC 5576 semantics and not for good. Why
do you want more than one SSRC for m-line but allow only one at a time=2C i=
s
this for the simulcast case?

Others which are not on section 2

1. The document talks about the case when there are multiple participants
each sending multiple streams as mentioned in section 1.1.1 and the receive=
r
may want to select among all these sources as discussed in section 1.1.2.
The document focus on the point to point case and does not address the issu=
e
of how all the m-lines from all participants are distributes and how a late
joiner gets all this information.

2. In the examples using the ssrc attribute note that according to RFC 5576
section 4.1 you must provide for each SSRC the CNAME attribute

3. The simulcast example and description allows the case of sending one of
the resolutions at a time but in simulcast the source may be able to send
multiple resolutions at the same time. To further complicate this case the
offer can support three resolutions but can only send any two at a time
which the receiver can select. How do you signal this=2C my view is with th=
e
maxsendssrc attribute.

4. in section 3.2.1.1 "as well as a 16-bit identifier (expressed as an
integer) used for correlation=3B" I think that it must be a byte string or
token not integer

Roni



> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Adam Roach
> Sent: 15 July=2C 2013 10:05 PM
> To: mmusic@ietf.org
> Cc: rtcweb@ietf.org
> Subject: [MMUSIC] Unified Plan for SDP Handling
>
> [Cross-posting to RTCWEB=3B follow-ups to MMUSIC=2C please]
>
> After significant work=2C Justin=2C Martin and I have managed to produce =
a
> compromise plan that provides a high degree of interoperability with
existing
> devices (and future non-WebRTC devices) while not being excessively
> onerous for WebRTC implementations or applications that use them. It's
> been a tricky balancing act=2C but I think we've found a good mix between
the
> two that can form a solid basis for the working group to move forward.
>
> Rather than summarize the key points of the document in this email=2C I
direct
> interested parties to section 2 of the document=2C which summarizes the k=
ey
> aspects of the plan in eight relatively concise bullet points.
>
> I apologize for the late publication date of this document -- there's
actually
> been a lot more work put into coming up with a unified draft than I
originally
> anticipated=2C and the production of this document took at least two week=
s
> longer than I expected it to.
>
> Note that this document is intended to be a plan for the work to be done
in
> this area=2C and not a specification in itself. The intention is that its
contents are
> used as the basis for work in several other drafts -- some new=2C some no=
t
--
> that form the corpus of work necessary for RTCWEB (and potentially CLUE)
to
> move forward. Except in rare cases=2C the document does not attempt to
> explicitly call out venues or documents for such work=2C as we (or=2C at =
the
very
> least=2C I) anticipate guidance from the various working group chairs to
assist in
> such decisions.
>
> Comments prior to Berlin would be very helpful=2C although this will clea=
rly
be a
> point of significant discussion at the face-to-face meeting.
>
> Document link:
>
> http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt
>
> /a
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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

From bernard_aboba@hotmail.com  Tue Jul 16 16:43:54 2013
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From: Bernard Aboba <bernard_aboba@hotmail.com>
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling
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IMHO=2C correlation needs to be supported in an RTP extension as well as RT=
CP SDES=2C regardless of the SDP discussion. In WebRTC we shouldn't be wedd=
ed to SDP signalling and even if we are=2C the dynamism implied by simulcas=
t and layered coding can't be accommodated in signalling.

Roni Even <ron.even.tlv@gmail.com> wrote:

Adam=2C
For correlation when the SSRC is not known in the SDP look at
http://tools.ietf.org/id/draft-even-mmusic-multiple-streams-02.txt and the
option to use SDES and not only RTP header extension.

BTW: there are other cases when the SSRC is not known using SDP before the
RTP stream arrives like when an MCU switch the video. Note that SDES is
distributed to the session when the source joins the RTP session which can
be long before the RTP media is sent to a specific receiver in the group.

Roni

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Adam Roach
> Sent: 15 July=2C 2013 10:05 PM
> To: mmusic@ietf.org
> Cc: rtcweb@ietf.org
> Subject: [MMUSIC] Unified Plan for SDP Handling
>
> [Cross-posting to RTCWEB=3B follow-ups to MMUSIC=2C please]
>
> After significant work=2C Justin=2C Martin and I have managed to produce =
a
> compromise plan that provides a high degree of interoperability with
existing
> devices (and future non-WebRTC devices) while not being excessively
> onerous for WebRTC implementations or applications that use them. It's
> been a tricky balancing act=2C but I think we've found a good mix between
the
> two that can form a solid basis for the working group to move forward.
>
> Rather than summarize the key points of the document in this email=2C I
direct
> interested parties to section 2 of the document=2C which summarizes the k=
ey
> aspects of the plan in eight relatively concise bullet points.
>
> I apologize for the late publication date of this document -- there's
actually
> been a lot more work put into coming up with a unified draft than I
originally
> anticipated=2C and the production of this document took at least two week=
s
> longer than I expected it to.
>
> Note that this document is intended to be a plan for the work to be done
in
> this area=2C and not a specification in itself. The intention is that its
contents are
> used as the basis for work in several other drafts -- some new=2C some no=
t
--
> that form the corpus of work necessary for RTCWEB (and potentially CLUE)
to
> move forward. Except in rare cases=2C the document does not attempt to
> explicitly call out venues or documents for such work=2C as we (or=2C at =
the
very
> least=2C I) anticipate guidance from the various working group chairs to
assist in
> such decisions.
>
> Comments prior to Berlin would be very helpful=2C although this will clea=
rly
be a
> point of significant discussion at the face-to-face meeting.
>
> Document link:
>
> http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt
>
> /a
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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

From adam@nostrum.com  Tue Jul 16 21:13:04 2013
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling - multiple media sources
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On 7/16/13 18:29, Bernard Aboba wrote:
> Yes, it is annoying when new docs attempt to redefine behavior in existing deployed RFCs.

And it's annoying when people try to comment on a proposal based solely 
on the mailing list discussion as opposed to reading the document in 
question.

Let's set the mudslinging aside for a moment.

>    RFC 5576 has multiple implementations today, and none of them have a one-at-a-time restriction.

Roni was confused, and you're riffing off his confusion. His comment was 
on the second bullet of section 2, which tries to clear up historical 
ambiguity by defining clear semantics for what it means to have multiple 
SSRCs specified in a single m-line IN THE ABSENCE of any signaling 
indicating their purpose.

Now, this clarification is nearly meaningless outside the context of the 
*first* bullet of section 2, which explains that each m-line corresponds 
to exactly one media stream track. I suppose it is this context you are 
missing, because I've worked with you in the past and know that you're 
smart enough to connect these dots and arrive at the conclusion that the 
second bullet, combined with the constraint imposed by the first bullet, 
is making things better, not worse.

In case we need to connect the dots for other working group 
participants, I'll be explicit. There are quite a number of use cases in 
which it is perfectly reasonable to have several SSRCs associated with a 
single media stream track, all of which are sent simultaneously. 
However, without explicit signaling (" Absent any other signaled 
extension", to use the exact phrase from the draft), there is very 
little hope of recipients having the first clue about what to do with 
these simultaneous streams all associated with the same media stream 
track. So we don't allow that, in general.

There's an exception, and it ties back to legacy use. In the 
preponderance of legacy cases in which a single m-line (historically, 
transport 5-tuple) was associated with multiple SSRCs, these multiple 
streams were sent in an exclusive (that is, non-parallel, one-at-a-time) 
fashion, and were meant to replace each other [1]. To maximize 
interoperability with legacy systems, the draft takes this 
(traditionally non-signaled) behavior and makes it the default when 
multiple SSRCs are present but not otherwise explained by the signaling. 
The bullet we're discussing requires signaling for all other uses; and, 
in doing so, effectively forbids multiple SSRCs only in the cases where 
sending them would be undefined or ambiguous.

If you don't think this course of action is a wise decision, please 
weigh in on the semantics you think should be implied when several SSRCs 
are present for a single media stream track without explicit signaling 
about their purpose. It's certainly something that I'd be happy to 
engage in useful discussion around. We had what I think is sound logic 
behind our choice, but there's no way we thought of everything. Your 
insights may be useful. However, simply sniping that you don't like this 
aspect of the proposal is counter to the goal (which I hope you share) 
of making progress.

/a

___
[1] I recognize that this observation about common historical behavior 
is based on my own experiences dealing with an admittedly limited set of 
products, and does not represent an exhaustive survey of the entire 
implementaion space. I'd love to hear what other people's observations 
of implementations' intended semantics when multiple SSRCs are present 
for a single logical stream without any signaling as to their purpose.

--------------070802000300070903020003
Content-Type: text/html; charset=ISO-8859-1
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<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 7/16/13 18:29, Bernard Aboba wrote:<br>
    </div>
    <blockquote cite="mid:BLU401-EAS46E4A7B1B6E70D598F788B93600@phx.gbl"
      type="cite">
      <pre wrap="">Yes, it is annoying when new docs attempt to redefine behavior in existing deployed RFCs.</pre>
    </blockquote>
    <br>
    And it's annoying when people try to comment on a proposal based
    solely on the mailing list discussion as opposed to reading the
    document in question.<br>
    <br>
    Let's set the mudslinging aside for a moment.<br>
    <br>
    <blockquote cite="mid:BLU401-EAS46E4A7B1B6E70D598F788B93600@phx.gbl"
      type="cite">
      <pre wrap="">  RFC 5576 has multiple implementations today, and none of them have a one-at-a-time restriction.
</pre>
    </blockquote>
    <br>
    Roni was confused, and you're riffing off his confusion. His comment
    was on the second bullet of section 2, which tries to clear up
    historical ambiguity by defining clear semantics for what it means
    to have multiple SSRCs specified in a single m-line IN THE ABSENCE
    of any signaling indicating their purpose.<br>
    <br>
    Now, this clarification is nearly meaningless outside the context of
    the *first* bullet of section 2, which explains that each m-line
    corresponds to exactly one media stream track. I suppose it is this
    context you are missing, because I've worked with you in the past
    and know that you're smart enough to connect these dots and arrive
    at the conclusion that the second bullet, combined with the
    constraint imposed by the first bullet, is making things better, not
    worse.<br>
    <br>
    In case we need to connect the dots for other working group
    participants, I'll be explicit. There are quite a number of use
    cases in which it is perfectly reasonable to have several SSRCs
    associated with a single media stream track, all of which are sent
    simultaneously. However, without explicit signaling ("
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    Absent any other signaled extension", to use the exact phrase from
    the draft), there is very little hope of recipients having the first
    clue about what to do with these simultaneous streams all associated
    with the same media stream track. So we don't allow that, in
    general.<br>
    <br>
    There's an exception, and it ties back to legacy use. In the
    preponderance of legacy cases in which a single m-line
    (historically, transport 5-tuple) was associated with multiple
    SSRCs, these multiple streams were sent in an exclusive (that is,
    non-parallel, one-at-a-time) fashion, and were meant to replace each
    other [1]. To maximize interoperability with legacy systems, the
    draft takes this (traditionally non-signaled) behavior and makes it
    the default when multiple SSRCs are present but not otherwise
    explained by the signaling. The bullet we're discussing requires
    signaling for all other uses; and, in doing so, effectively forbids
    multiple SSRCs only in the cases where sending them would be
    undefined or ambiguous.<br>
    <br>
    If you don't think this course of action is a wise decision, please
    weigh in on the semantics you think should be implied when several
    SSRCs are present for a single media stream track without explicit
    signaling about their purpose. It's certainly something that I'd be
    happy to engage in useful discussion around. We had what I think is
    sound logic behind our choice, but there's no way we thought of
    everything. Your insights may be useful. However, simply sniping
    that you don't like this aspect of the proposal is counter to the
    goal (which I hope you share) of making progress.<br>
    <br>
    /a<br>
    <br>
    ___<br>
    [1] I recognize that this observation about common historical
    behavior is based on my own experiences dealing with an admittedly
    limited set of products, and does not represent an exhaustive survey
    of the entire implementaion space. I'd love to hear what other
    people's observations of implementations' intended semantics when
    multiple SSRCs are present for a single logical stream without any
    signaling as to their purpose.<br>
  </body>
</html>

--------------070802000300070903020003--

From ron.even.tlv@gmail.com  Tue Jul 16 21:18:37 2013
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Jonathan Lennox'" <jonathan@vidyo.com>, "'Adam Roach'" <adam@nostrum.com>
References: <51E447C8.1000701@nostrum.com> <009001ce8263$0ed498b0$2c7dca10$@gmail.com> <51E5AF5A.9050405@nostrum.com> <EEE6BE4D-A231-4731-B824-B035BA099479@vidyo.com>
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Date: Wed, 17 Jul 2013 07:16:44 +0300
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling
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Adam,
I gave the wrong link before. I meant the one Jonathan pointed to
Roni

> -----Original Message-----
> From: Jonathan Lennox [mailto:jonathan@vidyo.com]
> Sent: 17 July, 2013 1:06 AM
> To: Adam Roach
> Cc: Roni Even; mmusic@ietf.org
> Subject: Re: [MMUSIC] Unified Plan for SDP Handling
> 
> 
> On Jul 16, 2013, at 4:38 PM, Adam Roach <adam@nostrum.com> wrote:
> 
> > On 7/16/13 15:28, Roni Even wrote:
> >> Adam,
> >> For correlation when the SSRC is not known in the SDP look at
> >> http://tools.ietf.org/id/draft-even-mmusic-multiple-streams-02.txt
> >> and the option to use SDES and not only RTP header extension.
> >
> > Roni:
> >
> > Thanks for the pointer. That's an interesting approach; however, for
> > RTCWEB at least, need a solution that also works with DTLS-SRTP.
> 
> 
> Acronym collision: This uses RTCP Source Description packets, not SDP
> Security Descriptions.
> 
> I think the (related) app-id approach described in
> http://tools.ietf.org/html/draft-even-mmusic-application-token-00.txt will
> also be of interest.
> 
> --
> Jonathan Lennox
> jonathan@vidyo.com



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Subject: Re: [MMUSIC] Unified Plan for SDP Handling - multiple media sources
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On 7/16/13 17:46, Jonathan Lennox wrote:
> Have you readhttp://tools.ietf.org/html/draft-even-mmusic-application-token-00.txt  ?  This is a more fleshed-out version of the same idea, as I understand the idea in Unified, written to be agnostic between Plans A / B / No Plan.

Thanks for the pointer -- I had missed this.

On a first read, this proposal seems to be backwards from what we need. 
The token scoping appears to be summed up by the following passage:

    The token is chosen by the sender, and represents the RTP stream that
    will be sent to the receiver.


What we need is a token chosen by the *receiver* representing what will 
be sent to the receiver. The crux of the matter is that the recipient 
may start receiving an RTP stream prior to receiving any signaling from 
the far end that can be used to correlate a token to the stream. We need 
to include something in an offer that allows us to correlate streams 
*before* the answer arrives.

That's not to say we couldn't add such functionality to this draft. But, 
unless I've completely misunderstood the meaning of the passage I quote 
above, the current application token proposal doesn't solve the clipping 
problem that we're trying to avoid.

/a

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    <div class="moz-cite-prefix">On 7/16/13 17:46, Jonathan Lennox
      wrote:<br>
    </div>
    <blockquote
      cite="mid:AE206958-4085-4B6C-A747-D4D60C5CD7FE@vidyo.com"
      type="cite">
      <pre wrap="">Have you read <a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-even-mmusic-application-token-00.txt">http://tools.ietf.org/html/draft-even-mmusic-application-token-00.txt</a> ?  This is a more fleshed-out version of the same idea, as I understand the idea in Unified, written to be agnostic between Plans A / B / No Plan.</pre>
    </blockquote>
    <br>
    Thanks for the pointer -- I had missed this.<br>
    <br>
    On a first read, this proposal seems to be backwards from what we
    need. The token scoping appears to be summed up by the following
    passage:<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <pre class="newpage">   The token is chosen by the sender, and represents the RTP stream that
   will be sent to the receiver.</pre>
    <br>
    What we need is a token chosen by the *receiver* representing what
    will be sent to the receiver. The crux of the matter is that the
    recipient may start receiving an RTP stream prior to receiving any
    signaling from the far end that can be used to correlate a token to
    the stream. We need to include something in an offer that allows us
    to correlate streams *before* the answer arrives.<br>
    <br>
    That's not to say we couldn't add such functionality to this draft.
    But, unless I've completely misunderstood the meaning of the passage
    I quote above, the current application token proposal doesn't solve
    the clipping problem that we're trying to avoid.<br>
    <br>
    /a<br>
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Adam Roach'" <adam@nostrum.com>
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Hi,
> 
> > 4. in section 3.2.1.1 "as well as a 16-bit identifier (expressed as an
> > integer) used for correlation;" I think that it must be a byte string
> > or token not integer
> 
> Do you have some rationale for your assertion?
> 
> The goal here is to limit expansion of the RTP packets as much as
possible.
> Given that there is a natural limit of 65536 streams in a single RTP
session, 16
> bits for a correlator seems sufficient.
> 
> /a
[Roni Even] This is the syntax of the header extension and
extensionattributes in RFC5285. It is a byte string and not an integer



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Indeed=2C unless the RTP extension is self-describing it has to be chosen b=
y the receiver in order to provide guidance to the receiver before an answe=
r arrives.  If it is opaque and chosen by the sender it isn't all that diff=
erent from an SSRC chosen by the sender and potentially signalled in the an=
swer.

Adam Roach <adam@nostrum.com> wrote:

On 7/16/13 17:46=2C Jonathan Lennox wrote:
> Have you readhttp://tools.ietf.org/html/draft-even-mmusic-application-tok=
en-00.txt  ?  This is a more fleshed-out version of the same idea=2C as I u=
nderstand the idea in Unified=2C written to be agnostic between Plans A / B=
 / No Plan.

Thanks for the pointer -- I had missed this.

On a first read=2C this proposal seems to be backwards from what we need.
The token scoping appears to be summed up by the following passage:

    The token is chosen by the sender=2C and represents the RTP stream that
    will be sent to the receiver.


What we need is a token chosen by the *receiver* representing what will
be sent to the receiver. The crux of the matter is that the recipient
may start receiving an RTP stream prior to receiving any signaling from
the far end that can be used to correlate a token to the stream. We need
to include something in an offer that allows us to correlate streams
*before* the answer arrives.

That's not to say we couldn't add such functionality to this draft. But=2C
unless I've completely misunderstood the meaning of the passage I quote
above=2C the current application token proposal doesn't solve the clipping
problem that we're trying to avoid.

/a

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<div class=3D"PlainText">Indeed=2C unless the RTP extension is self-describ=
ing it has to be chosen by the receiver in order to provide guidance to the=
 receiver before an answer arrives.&nbsp=3B If it is opaque and chosen by t=
he sender it isn't all that different from
 an SSRC chosen by the sender and potentially signalled in the answer. <br>
<br>
Adam Roach &lt=3Badam@nostrum.com&gt=3B wrote:<br>
<br>
</div>
<div style=3D"background-color:#FFFFFF">
<div class=3D"x_moz-cite-prefix">On 7/16/13 17:46=2C Jonathan Lennox wrote:=
<br>
</div>
<blockquote type=3D"cite">
<pre>Have you read <a class=3D"x_moz-txt-link-freetext" href=3D"http://tool=
s.ietf.org/html/draft-even-mmusic-application-token-00.txt">http://tools.ie=
tf.org/html/draft-even-mmusic-application-token-00.txt</a> ?  This is a mor=
e fleshed-out version of the same idea=2C as I understand the idea in Unifi=
ed=2C written to be agnostic between Plans A / B / No Plan.</pre>
</blockquote>
<br>
Thanks for the pointer -- I had missed this.<br>
<br>
On a first read=2C this proposal seems to be backwards from what we need. T=
he token scoping appears to be summed up by the following passage:<br>
<br>
<pre class=3D"x_newpage">   The token is chosen by the sender=2C and repres=
ents the RTP stream that
   will be sent to the receiver.</pre>
<br>
What we need is a token chosen by the *receiver* representing what will be =
sent to the receiver. The crux of the matter is that the recipient may star=
t receiving an RTP stream prior to receiving any signaling from the far end=
 that can be used to correlate a
 token to the stream. We need to include something in an offer that allows =
us to correlate streams *before* the answer arrives.<br>
<br>
That's not to say we couldn't add such functionality to this draft. But=2C =
unless I've completely misunderstood the meaning of the passage I quote abo=
ve=2C the current application token proposal doesn't solve the clipping pro=
blem that we're trying to avoid.<br>
<br>
/a<br>
</div>
</body>
</html>

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Adam Roach'" <adam@nostrum.com>
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Date: Wed, 17 Jul 2013 08:34:34 +0300
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling - multiple media sources
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I think that the problem in bullet 2 of section 2  when reading bullet 1 is
that it is not clear. Bullet 1 says "nor may a single m-line    represent
multiple media stream tracks".  Since there is no definition of media stream
track in this document and of source I looked at section 2.3 of
http://tools.ietf.org/html/draft-lennox-raiarea-rtp-grouping-taxonomy-01
which tries to clarify the terminology based on existing documents. 
To me even though bullet 2 says "in the absent of other signaled extension"
it will be not correct according to bullet 1 to have one m-line that have
multiple sources (according to my reading of the taxonomy source and media
stream track are equivalent)

As for "although senders can   switch between the SSRCs as frequently as
desired, only one    should be sent at any given time." It got me confused,
in simulcast  does it mean that a sender can send one packet of one
resolution followed by a packet of a second resolution each has a unique
SSRC but they should have a different timestamp. What is "at any given
time"?

Roni

> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: 17 July, 2013 1:26 AM
> To: Roni Even
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] Unified Plan for SDP Handling - multiple media
> sources
> 
> On 7/16/13 16:48, Roni Even wrote:
> > Hi,
> > Some comments.
> > In section 2 bullet 2 is changing RFC 5576 semantics and not for good.
> > Why do you want more than one SSRC for m-line but allow only one at a
> > time, is this for the simulcast case?
> 
> No, simulcast is signaled explicitly (I will point out and emphasize the
phrase
> "absent any other signaled extension"). This is "default behavior" is
intended
> to accommodate MCUs that perform packet switching for RTP (and similar
> architectures that require different timing and/or sequence number spaces
> on a single stream).
> 
> > Others which are not on section 2
> >
> /a


From adam@nostrum.com  Tue Jul 16 22:52:32 2013
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling - multiple media sources
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On Jul 17, 2013, at 0:34, "Roni Even" <ron.even.tlv@gmail.com> wrote:

> As for "although senders can   switch between the SSRCs as frequently as
> desired, only one    should be sent at any given time." It got me confused=
,
> in simulcast...


This bullet has no bearing on simulcast behavior, other than to say that sim=
ulcast must be explicitly signaled.

/a=

From keith.drage@alcatel-lucent.com  Wed Jul 17 04:07:01 2013
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "clue@ietf.org" <clue@ietf.org>, "dispatch@ietf.org" <dispatch@ietf.org>, "avt@ietf.org" <avt@ietf.org>
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Breaking my own rule about cross posting...

Just wanted to add is that this document would not be retrospective, i.e. w=
e would not start rewriting the existing published RFCs to take into accoun=
t the new terminology.

The expectation however is that documents going forward from other working =
groups would use the new terminology rather than other terms.=20

Obviously how this is handled is up to the individual working groups, but u=
sing the new terminology will be the only good way of making your results u=
nderstandable outside your specific field.

So that is why your review and input is important.

Regards

Keith

> -----Original Message-----
> From: DRAGE, Keith (Keith)
> Sent: 16 July 2013 15:58
> To: mmusic@ietf.org; rtcweb@ietf.org; clue@ietf.org; dispatch@ietf.org;
> avt@ietf.org
> Subject: FW: Taxonomy
>=20
> Extensive cross-post do not reply to this email.
>=20
> For information. We do expect participation input from all the listed
> working groups (and possibly more).
>=20
> Keith
>=20
> -----Original Message-----
> From: avtext-bounces@ietf.org [mailto:avtext-bounces@ietf.org] On Behalf
> Of DRAGE, Keith (Keith)
> Sent: 16 July 2013 15:53
> To: avtext@ietf.org
> Subject: [avtext] Taxonomy
>=20
> (As AVTEXT WG cochair)
>=20
> The latest version of the RTP taxonomy draft has been submitted.
>=20
> https://datatracker.ietf.org/doc/draft-lennox-raiarea-rtp-grouping-
> taxonomy/
>=20
> This document was allocated to AVTEXT by DISPATCH, and we have created a
> milestone for it in AVTEXT.
>=20
> This draft is not yet a working group document.
>=20
> We do expect to spend some significant meeting time discussing the
> contents in the AVTEXT meeting in Berlin, so please start raising your
> issues on the AVTEXT list.
>=20
> Additionally, if there are counter proposals existing that the AVTEXT
> chairs would not otherwise be aware of, then please identify them to the
> AVTEXT chairs / list.
>=20
> Regards
>=20
> Keith
> _______________________________________________
> avtext mailing list
> avtext@ietf.org
> https://www.ietf.org/mailman/listinfo/avtext

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 17 Jul 2013 13:26:47 +0200
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Subject: Re: [MMUSIC] [dispatch] Taxonomy
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2013/7/17 DRAGE, Keith (Keith) <keith.drage@alcatel-lucent.com>:
> The expectation however is that documents going forward from other workin=
g groups would use the new terminology rather than other terms.
>
> Obviously how this is handled is up to the individual working groups, but=
 using the new terminology will be the only good way of making your results=
 understandable outside your specific field.
>
> So that is why your review and input is important.

Hi Drage,

First of I'm sorry but I'm not subscribed to MMUSIC nor CLUE, so let
me please reply here (it is just a single topic):

I'm really annoying after reading the draft section 2.4 "Media Stream":

  http://tools.ietf.org/html/draft-lennox-raiarea-rtp-grouping-taxonomy-01#=
section-2.4

------------------------
      Each Media Stream is identified by a unique Synchronization source
      (SSRC) [RFC3550] that is carried in every RTP and Real-time
      Transport Control Protocol (RTCP) packet header.
------------------------

In W3C we have "MediaStream" and "MediaStreamTrack", and
"MediaStreamTrack" is exactly what the draft above defines as "Media
Stream".

So let me say that this is the most confusing terminology it can be.



--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [MMUSIC] Unified Plan for SDP Handling
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On 7/15/13 9:06 PM, Adam Roach wrote:=0A=
> [Cross-posting to RTCWEB; follow-ups to MMUSIC, please]=0A=
>=0A=
> After significant work, Justin, Martin and I have managed to produce a=0A=
> compromise plan that provides a high degree of interoperability with=0A=
> existing devices (and future non-WebRTC devices) while not being=0A=
> excessively onerous for WebRTC implementations or applications that use=
=0A=
> them. It's been a tricky balancing act, but I think we've found a good=0A=
> mix between the two that can form a solid basis for the working group to=
=0A=
> move forward.=0A=
=0A=
Thanks for doing this work, I hope that we now finally can get going on =0A=
specifying this up (and I hope it won't take too long to do so!).=0A=
=0A=
I think this looks like a really good start, and there is a lot I like =0A=
(e.g. the independent signaling per MediaStreamTrack).=0A=
=0A=
I have a couple of specific questions/comments, coming from a very =0A=
WebRTC centric (sorry for that) perspective:=0A=
=0A=
* Simulcast: In the current API model, MediaStreamTrack is about the =0A=
lowest you can get to. And the same source (camera, microphone) can =0A=
source several MediaStreamTrack - and each of those can use individual =0A=
constraints for width, height, framerate, etc. So in a WebRTC context, =0A=
there seem to me we would not need the simulcast solution you have =0A=
included. The different resolutions can be sent as different =0A=
MediaStreamTrack's - meaning they would be separate m-lines.=0A=
=0A=
* Direction attribute: The current WebRTC API is "sender" oriented: the =0A=
application can select MediaStream(Track)'s to send. Would it for WebRTC =
=0A=
make sense to have all m-lines being unidirectional (send/recvonly)? (If =
=0A=
they should be sendrecv we also need to specify when a return track =0A=
should reuse an existing m-line, and when a new m-line should be created.)=
=0A=
=0A=
Stefan=0A=
=0A=
>=0A=
> Rather than summarize the key points of the document in this email, I=0A=
> direct interested parties to section 2 of the document, which summarizes=
=0A=
> the key aspects of the plan in eight relatively concise bullet points.=0A=
>=0A=
> I apologize for the late publication date of this document -- there's=0A=
> actually been a lot more work put into coming up with a unified draft=0A=
> than I originally anticipated, and the production of this document took=
=0A=
> at least two weeks longer than I expected it to.=0A=
>=0A=
> Note that this document is intended to be a plan for the work to be done=
=0A=
> in this area, and not a specification in itself. The intention is that=0A=
> its contents are used as the basis for work in several other drafts --=0A=
> some new, some not -- that form the corpus of work necessary for RTCWEB=
=0A=
> (and potentially CLUE) to move forward. Except in rare cases, the=0A=
> document does not attempt to explicitly call out venues or documents for=
=0A=
> such work, as we (or, at the very least, I) anticipate guidance from the=
=0A=
> various working group chairs to assist in such decisions.=0A=
>=0A=
> Comments prior to Berlin would be very helpful, although this will=0A=
> clearly be a point of significant discussion at the face-to-face meeting.=
=0A=
>=0A=
> Document link:=0A=
>=0A=
> http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt=0A=
>=0A=
> /a=0A=
> _______________________________________________=0A=
> mmusic mailing list=0A=
> mmusic@ietf.org=0A=
> https://www.ietf.org/mailman/listinfo/mmusic=0A=
>=0A=
=0A=

From pkyzivat@alum.mit.edu  Wed Jul 17 09:38:29 2013
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Subject: Re: [MMUSIC] [rtcweb] Unified Plan for SDP Handling
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Adam,

While I see there can be some value to extending O/A to allow for 
smaller critical sections, I fear that it is going to be very messy to 
get "right". Some messy things:

- Glare between a partial O/A and a full O/A
- Is a partial O/A permitted in a sip reINVITE? Can you have
   an offerless reINVITE where the response contains a partial offer?
   Or can these only be done in UPDATE?
- In SIP, processing of concurrent reINVITEs/UPDATEs containing
   partial SDP updates. This seems to already be a
- How to assign version numbers to the full SDP as a result
   of partial updates
- I'm not sure your algorithm for resolving glare when adding
   streams works in all cases. (More below on this.)

These (and ones I haven't thought of) *may* all be resolvable. But this 
certainly complicates the already complex O/A process - maybe to the 
point that many get it wrong. (I don't want to write rfc6337bis!)

My concern with the glare resolution algorithm is that for it to work 
both sides must realize that glare occurred. And I think there are 
common sequences when one side won't realize it.

I'm reluctant to buy into a scheme that depends on this without knowing 
it it will be implementable.

	Thanks,
	Paul


From jonathan@vidyo.com  Wed Jul 17 10:07:55 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
Date: Wed, 17 Jul 2013 13:07:41 -0400
Thread-Topic: [MMUSIC] Unified Plan for SDP Handling
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling
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On Jul 16, 2013, at 7:21 PM, Bernard Aboba <bernard_aboba@hotmail.com> wrot=
e:

> The app-id idea is useful for multiple reasons.  One reason similar RTP e=
xtensions are used today is to avoid having to parse the payload to figure =
out what the stream is (e.g. looking for parameter sets or SEI NAL units). =
=20

That sounds like an interesting idea, but unless I'm misunderstanding you, =
that sounds very different from what app-id (or the correlator, in Unified)=
 is doing.  Those are for identifying streams, whereas this sounds more lik=
e identifying the sub-structure of a stream (i.e. which packets contain par=
ameter sets or SEIs).

I could imagine that being useful, but it doesn't sound very related to str=
eam correlation.

> It is cleaner to solve this in as generic rather than a codec-specific wa=
y.  I see no conflict with no-plan; in fact, I believe no plan suggested an=
 RTP extension, no?
>=20
> Jonathan Lennox <jonathan@vidyo.com> wrote:
>=20
>=20
> On Jul 16, 2013, at 4:38 PM, Adam Roach <adam@nostrum.com> wrote:
>=20
>> On 7/16/13 15:28, Roni Even wrote:
>>> Adam,
>>> For correlation when the SSRC is not known in the SDP look at
>>> http://tools.ietf.org/id/draft-even-mmusic-multiple-streams-02.txt and =
the
>>> option to use SDES and not only RTP header extension.
>>=20
>> Roni:
>>=20
>> Thanks for the pointer. That's an interesting approach; however, for
>> RTCWEB at least, need a solution that also works with DTLS-SRTP.
>=20
>=20
> Acronym collision: This uses RTCP Source Description packets, not SDP Sec=
urity Descriptions.
>=20
> I think the (related) app-id approach described in http://tools.ietf.org/=
html/draft-even-mmusic-application-token-00.txt will also be of interest.
>=20
> --
> Jonathan Lennox
> jonathan@vidyo.com
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

--
Jonathan Lennox
jonathan@vidyo.com



From adam@nostrum.com  Wed Jul 17 10:07:59 2013
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Subject: Re: [MMUSIC] [rtcweb] Unified Plan for SDP Handling
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On 7/17/13 11:38, Paul Kyzivat wrote:
> Adam,
>
> While I see there can be some value to extending O/A to allow for 
> smaller critical sections, I fear that it is going to be very messy to 
> get "right". Some messy things:
>
> - Glare between a partial O/A and a full O/A

...is glare. Resolve as normal. For SIP, that means timers.

> - Is a partial O/A permitted in a sip reINVITE? Can you have
>   an offerless reINVITE where the response contains a partial offer?
>   Or can these only be done in UPDATE?

No, no, and no. In SIP, the protocol machinery would not allow INVITE or 
UPDATE to complete concurrently in opposite directions for any amount of 
overlap. We aren't proposing a change to SIP state machines here, only 
to the offer/answer model. This necessarily means that we need to carry 
partial offers and partial answers in something else -- probably INFO.

> - In SIP, processing of concurrent reINVITEs/UPDATEs containing
>   partial SDP updates. This seems to already be a

You left this unfinished, but I *think* it was going to make the same 
point I did in my response above.

> - How to assign version numbers to the full SDP as a result
>   of partial updates

Section 3.4.1: "... the SDP version is incremented by one..."


> - I'm not sure your algorithm for resolving glare when adding
>   streams works in all cases. (More below on this.)
>
> These (and ones I haven't thought of) *may* all be resolvable. But 
> this certainly complicates the already complex O/A process - maybe to 
> the point that many get it wrong. (I don't want to write rfc6337bis!)
>
> My concern with the glare resolution algorithm is that for it to work 
> both sides must realize that glare occurred. And I think there are 
> common sequences when one side won't realize it.

I'm not saying I didn't miss it, but I did work through as many 
permutations as I could think of, and none of them are left out by the 
algorithm described in section 3.4.1.

That said, I did my analysis in the context of WebRTC, and assumed a 
reliable, in-order delivery channel for messages. I recognize that we 
don't have that for SIP, and that messages in the same direction can be 
reordered. I admit that this does raise some interesting scenarios that 
may require additional work to resolve.

> I'm reluctant to buy into a scheme that depends on this without 
> knowing it it will be implementable.

I understand your reluctance, and don't want to gloss over it. This is 
but one plank in the overall plan, and it represents what we think is 
the best (but not only) solution to this specific requirement. If the 
issues prove intractable, we could fall back to an alternate scheme, 
such as the designated-offerer I describe in 
draft-roach-rtcweb-glareless-add-00, or the token-passing scheme 
described by Ted Hardie (cf. 
http://www.ietf.org/mail-archive/web/rtcweb/current/msg07361.html).

/a

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    <div class="moz-cite-prefix">On 7/17/13 11:38, Paul Kyzivat wrote:<br>
    </div>
    <blockquote cite="mid:51E6C86C.5020101@alum.mit.edu" type="cite">Adam,
      <br>
      <br>
      While I see there can be some value to extending O/A to allow for
      smaller critical sections, I fear that it is going to be very
      messy to get "right". Some messy things:
      <br>
      <br>
      - Glare between a partial O/A and a full O/A</blockquote>
    <br>
    ...is glare. Resolve as normal. For SIP, that means timers.<br>
    <br>
    <blockquote cite="mid:51E6C86C.5020101@alum.mit.edu" type="cite">
      - Is a partial O/A permitted in a sip reINVITE? Can you have
      <br>
      &nbsp; an offerless reINVITE where the response contains a partial
      offer?
      <br>
      &nbsp; Or can these only be done in UPDATE?
      <br>
    </blockquote>
    <br>
    No, no, and no. In SIP, the protocol machinery would not allow
    INVITE or UPDATE to complete concurrently in opposite directions for
    any amount of overlap. We aren't proposing a change to SIP state
    machines here, only to the offer/answer model. This necessarily
    means that we need to carry partial offers and partial answers in
    something else -- probably INFO.<br>
    <br>
    <blockquote cite="mid:51E6C86C.5020101@alum.mit.edu" type="cite">-
      In SIP, processing of concurrent reINVITEs/UPDATEs containing
      <br>
      &nbsp; partial SDP updates. This seems to already be a
      <br>
    </blockquote>
    <br>
    You left this unfinished, but I *think* it was going to make the
    same point I did in my response above.<br>
    <br>
    <blockquote cite="mid:51E6C86C.5020101@alum.mit.edu" type="cite">-
      How to assign version numbers to the full SDP as a result
      <br>
      &nbsp; of partial updates
      <br>
    </blockquote>
    <br>
    Section 3.4.1: "...
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    the SDP version is incremented by one..."<br>
    <br>
    <br>
    <blockquote cite="mid:51E6C86C.5020101@alum.mit.edu" type="cite">-
      I'm not sure your algorithm for resolving glare when adding
      <br>
      &nbsp; streams works in all cases. (More below on this.)
      <br>
      <br>
      These (and ones I haven't thought of) *may* all be resolvable. But
      this certainly complicates the already complex O/A process - maybe
      to the point that many get it wrong. (I don't want to write
      rfc6337bis!)
      <br>
      <br>
      My concern with the glare resolution algorithm is that for it to
      work both sides must realize that glare occurred. And I think
      there are common sequences when one side won't realize it.
      <br>
    </blockquote>
    <br>
    I'm not saying I didn't miss it, but I did work through as many
    permutations as I could think of, and none of them are left out by
    the algorithm described in section 3.4.1.<br>
    <br>
    That said, I did my analysis in the context of WebRTC, and assumed a
    reliable, in-order delivery channel for messages. I recognize that
    we don't have that for SIP, and that messages in the same direction
    can be reordered. I admit that this does raise some interesting
    scenarios that may require additional work to resolve.<br>
    <br>
    <blockquote cite="mid:51E6C86C.5020101@alum.mit.edu" type="cite">
      I'm reluctant to buy into a scheme that depends on this without
      knowing it it will be implementable.
      <br>
    </blockquote>
    <br>
    I understand your reluctance, and don't want to gloss over it. This
    is but one plank in the overall plan, and it represents what we
    think is the best (but not only) solution to this specific
    requirement. If the issues prove intractable, we could fall back to
    an alternate scheme, such as the designated-offerer I describe in
    draft-roach-rtcweb-glareless-add-00, or the token-passing scheme
    described by Ted Hardie (cf.
    <a class="moz-txt-link-freetext" href="http://www.ietf.org/mail-archive/web/rtcweb/current/msg07361.html">http://www.ietf.org/mail-archive/web/rtcweb/current/msg07361.html</a>).<br>
    <br>
    /a<br>
  </body>
</html>

--------------070101020305040305030907--

From adam@nostrum.com  Wed Jul 17 13:23:55 2013
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling
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On 7/17/13 09:08, Stefan Håkansson LK wrote:
> I have a couple of specific questions/comments, coming from a very
> WebRTC centric (sorry for that) perspective:
>
> * Simulcast: In the current API model, MediaStreamTrack is about the
> lowest you can get to. And the same source (camera, microphone) can
> source several MediaStreamTrack - and each of those can use individual
> constraints for width, height, framerate, etc. So in a WebRTC context,
> there seem to me we would not need the simulcast solution you have
> included. The different resolutions can be sent as different
> MediaStreamTrack's - meaning they would be separate m-lines.

That's what we meant to convey with this (section 2, bullet 1): "This 
does not preclude 'application level' simulcasting; i.e., the creation 
of multiple media stream tracks from a single source."

It's true that we don't broach the issue of how to indicate the need for 
simulcast via a WebRTC javascript API. I think this is something that 
would be future work, probably in the W3C. I'm going to leave any 
further explanation to Justin, as most of the simulcast ideas in the 
document came from him.

> * Direction attribute: The current WebRTC API is "sender" oriented: the
> application can select MediaStream(Track)'s to send. Would it for WebRTC
> make sense to have all m-lines being unidirectional (send/recvonly)?

I understand how this makes things cleaner in general. The issue I have 
is that it interacts poorly with legacy uses.

>   (If
> they should be sendrecv we also need to specify when a return track
> should reuse an existing m-line, and when a new m-line should be created.)
>

I agree that this could stand to be spelled out more clearly. I do not 
think that doing so will prove particularly controversial.

/a

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From: Jonathan Lennox <jonathan@vidyo.com>
To: Adam Roach <adam@nostrum.com>
Date: Wed, 17 Jul 2013 17:40:11 -0400
Thread-Topic: [MMUSIC] Unified Plan for SDP Handling - multiple media sources
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On Jul 17, 2013, at 12:24 AM, Adam Roach <adam@nostrum.com> wrote:

> What we need is a token chosen by the *receiver* representing what will b=
e sent to the receiver. The crux of the matter is that the recipient may st=
art receiving an RTP stream prior to receiving any signaling from the far e=
nd that can be used to correlate a token to the stream. We need to include =
something in an offer that allows us to correlate streams *before* the answ=
er arrives.
>=20
> That's not to say we couldn't add such functionality to this draft. But, =
unless I've completely misunderstood the meaning of the passage I quote abo=
ve, the current application token proposal doesn't solve the clipping probl=
em that we're trying to avoid.

Ah, interesting.  The problem case is an offerer offering a new bundled sen=
drecv or recvonly m-line, where the answerer can start sending media for it=
 immediately, outracing the answer?

Yes, I see how a receiver-chosen correlator solves that case.

That said, what I still don't see is, if you have the correlator, why you t=
hen actually need to send the a=3Dssrc in the SDP at all? The correlator sh=
ould be sufficient for disambiguation, and it lets the sender switch the ss=
rc it's sending for an m-line without needing a fresh offer/answer.

(The use case I'm thinking about here is something like the CLUE "dynamic c=
apture" in the context of what Magnus named the "Source-projecting mixer" t=
opology -- where the ssrc sent for a given m-line switches dynamically and =
quickly, e.g. with loudest-speaker switching.)

--
Jonathan Lennox
jonathan@vidyo.com



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On 7/17/13 16:40, Jonathan Lennox wrote:
> That said, what I still don't see is, if you have the correlator, why you then actually need to send the a=ssrc in the SDP at all? The correlator should be sufficient for disambiguation, and it lets the sender switch the ssrc it's sending for an m-line without needing a fresh offer/answer.

That might be worth exploring. I honestly haven't given it a whole lot 
of thought. Perhaps Justin or Martin have a more well-formed opinion on 
the topic?

/a

From fandreas@cisco.com  Wed Jul 17 14:49:36 2013
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The MMUSIC draft agenda for IETF 87 has been posted at:

https://datatracker.ietf.org/meeting/87/agenda/mmusic/

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

Thanks

     Ari & Flemming

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Subject: Re: [MMUSIC] Unified Plan for SDP Handling - multiple media sources
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On 17 July 2013 14:44, Adam Roach <adam@nostrum.com> wrote:
> On 7/17/13 16:40, Jonathan Lennox wrote:
>>
>> That said, what I still don't see is, if you have the correlator, why you
>> then actually need to send the a=ssrc in the SDP at all? The correlator
>> should be sufficient for disambiguation, and it lets the sender switch the
>> ssrc it's sending for an m-line without needing a fresh offer/answer.
>
>
> That might be worth exploring. I honestly haven't given it a whole lot of
> thought. Perhaps Justin or Martin have a more well-formed opinion on the
> topic?

I did consider this, but there are several reasons that I prefer to
have signaling.  The main one being that it is the only guaranteed way
to get this information through.  In the happy cases, the RTP header
is enough, but if you are losing packets at session startup (a real
possibility in some configurations), or RTP header extensions are
being dropped, or you need to communicate about an in progress stream
and can't guarantee that you can cause the sender to re-send a few
packets with the header extension, or ...

Basically, there are too many ways for the RTP extension to break.
Not enough that it isn't hugely valuable for dealing with the race,
but it can't be the only mechanism that you rely on.

From bernard_aboba@hotmail.com  Wed Jul 17 20:35:15 2013
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling - multiple media sources
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Martin said:=20

> I did consider this=2C but there are several reasons that I prefer to hav=
e signaling. =20
=20
[BA] The RTP Extension is by definition=2C optional.  So it cannot be used =
instead of initial signaling=2C because of reliability issues (same issue a=
pplies with SDES packets) and because it might not be there.  However=2C in=
 the event of an SSRC collision it can be useful.  Also=2C  RTP extensions =
are deployed today to label temporal=2C spatial or quality scalability stre=
ams=2C so that the SRTP packets don't have to be decrypted and parsed befor=
e figuring out what layer they pertain to.=20
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>Martin said: <BR><br>&gt=3B I di=
d consider this=2C but there are several reasons that I prefer to&nbsp=3Bha=
ve signaling.&nbsp=3B <BR>&nbsp=3B<BR>[BA] The RTP Extension is by definiti=
on=2C optional.&nbsp=3B So it cannot be used instead of initial signaling=
=2C because of reliability issues (same issue applies with SDES packets) an=
d because it might not be there.&nbsp=3B&nbsp=3BHowever=2C in the event of =
an SSRC collision&nbsp=3Bit can be useful. &nbsp=3BAlso=2C&nbsp=3B RTP exte=
nsions are deployed today to label temporal=2C spatial or quality scalabili=
ty streams=2C so that the SRTP packets don't have to be decrypted and parse=
d before figuring out what layer they pertain to. <BR> 		 	   		  </div></b=
ody>
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From harald@alvestrand.no  Thu Jul 18 02:40:49 2013
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Subject: [MMUSIC] Review of unified-plan-00
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First of all, let me say that I'm very happy to see a plan that the 
people who have been voicing the strongest opinions in this debate are 
willing to stand behind.
I believe, extremely strongly, that it is better to move forward on one 
path, no matter what it is (as long as it's not obviously broken), than 
to continue to debate. This topic has long been needing a decision.

Thus, I stand behind the technology proposed, and have no wish to change 
it. The purpose of this review is to help make sure it's fully 
described, and that all parties understand it the same way.

I have ignored language-only issues at this time.

Introduction, paragraph 2:

    The standard way of encoding this information in SDP is to have each
    RTP flow (i.e., SSRC) appear on its own m-line.

This is not strictly true, since the proposal continues to use each 
m-line for up to two flows, one in each direction, and some of the . 
Suggested rephrase:

   One common way of encoding this information uses each M-line to
   describe one media source in each direction.

"media source" is as defined in 
draft-westerlund-avtcore-multiplex-architecture-03 section 2.1. The term 
is used elsewhere in the document, and seems to have this meaning. "One 
common" is a reflection of my belief that practice has not been uniform 
in this area.

3.1, Bundle-only M-lines

This mechanism requires sending M-lines with port zero as part of a 
bundle in an SDP OFFER.
It needs to note that this specification modifies RFC 5888 section 9.2 
(the grouping framework) by allowing such M-lines to be part of a group 
in an offer (but I do not think it needs to be allowed in an answer).

4.1 Simple example

"It also shows unique payload across the audio and video m=lines for the 
Answerer that does not support BUNDLE semantics." - this doesn't make 
sense to me; unique payloads are needed for answerers that support 
BUNDLE semantics. Copy/paste error?

Would be consistent if it said "for the Answerer that supports BUNDLE 
semantics".

4.4 Multiple Video with Simulcast

If it's intended that draft-westerlund-avtcore-rtp-simulcast is used, 
the ssrc-group lines should probably say SCS rather than SIMULCAST.


And now for some stuff that isn't in the document, but might want to be:

More examples

There should probably be one example where the offer contains 1 audio 
and 2 video streams, and the answer contains 1 audio and 1 video stream 
(fewer streams in response than in request); I believe that this answer 
would then contain "a=recvonly" and no "a=msid" attribute on the 3rd 
m-line, but it's nice to be sure.

Interaction with a=content

Previously, we had discussed having a "content" attribute on 
MediaStreamTracks that would make a track link to an m-line with the 
same "a=content" attribute. If this is still expected to work, it might 
be worthy of inclusion here.

I think that's all for this round of review. Thanks for getting it out!

             Harald





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On 07/17/2013 11:49 PM, Flemming Andreasen wrote:
> The MMUSIC draft agenda for IETF 87 has been posted at:
>
> https://datatracker.ietf.org/meeting/87/agenda/mmusic/
>
> Please take a look and let us know if you have any questions or comments.

Hi,

I note that AVTEXT's posted agenda:

https://datatracker.ietf.org/meeting/87/agenda/avtext/

has them meeting on Tuesday at 1700, and they want to consider the 
result of MMUSIC's discussions on draft-roach-mmusic-unified-plan - 
which according to

https://datatracker.ietf.org/meeting/87/agenda/mmusic/

will take place on Wednesday, making this a little bit difficult absent 
temporal manipulation.

Perhaps the chairs should consult on whether it's worth swapping the two 
days of MMUSIC discussion in order to accomodate the AVTEXT linkage?

>
> Thanks
>
>     Ari & Flemming
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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    <div class="moz-cite-prefix">On 07/17/2013 11:49 PM, Flemming
      Andreasen wrote:<br>
    </div>
    <blockquote cite="mid:51E71167.3020203@cisco.com" type="cite">The
      MMUSIC draft agenda for IETF 87 has been posted at:
      <br>
      <br>
      <a class="moz-txt-link-freetext" href="https://datatracker.ietf.org/meeting/87/agenda/mmusic/">https://datatracker.ietf.org/meeting/87/agenda/mmusic/</a>
      <br>
      <br>
      Please take a look and let us know if you have any questions or
      comments.
      <br>
    </blockquote>
    <br>
    Hi,<br>
    <br>
    I note that AVTEXT's posted agenda:<br>
    <br>
    <a class="moz-txt-link-freetext" href="https://datatracker.ietf.org/meeting/87/agenda/avtext/">https://datatracker.ietf.org/meeting/87/agenda/avtext/</a><br>
    <br>
    has them meeting on Tuesday at 1700, and they want to consider the
    result of MMUSIC's discussions on draft-roach-mmusic-unified-plan -
    which according to<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <a href="https://datatracker.ietf.org/meeting/87/agenda/mmusic/">https://datatracker.ietf.org/meeting/87/agenda/mmusic/</a><br>
    <br>
    will take place on Wednesday, making this a little bit difficult
    absent temporal manipulation.<br>
    <br>
    Perhaps the chairs should consult on whether it's worth swapping the
    two days of MMUSIC discussion in order to accomodate the AVTEXT
    linkage?<br>
    <br>
    <blockquote cite="mid:51E71167.3020203@cisco.com" type="cite">
      <br>
      Thanks
      <br>
      <br>
      &nbsp;&nbsp;&nbsp; Ari &amp; Flemming
      <br>
      _______________________________________________
      <br>
      mmusic mailing list
      <br>
      <a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
      <br>
      <a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>
      <br>
    </blockquote>
    <br>
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From keith.drage@alcatel-lucent.com  Thu Jul 18 05:46:18 2013
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] MMUSIC Draft Agenda for IETF 87
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Date: Thu, 18 Jul 2013 12:46:00 +0000
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Ideally, AVTEXT (as in the AVTEXT chairs) would have liked the MMUSIC discu=
ssion first, but I suspect they are dealing with a whole lot of other sched=
uling problems.

I'd also note that the AVTEXT agenda point relates to one very small part o=
f this discussion.

There have been a number of mentions of the need for RTP header extensions,=
 which are therefore within the scope of AVTEXT. Timescales are getting sho=
rter and shorter for this to happen, so we want to open up some information=
 exchange about:

1)	Is this a sensible usage of a RTP header extension;
2)	What we need to do (and that is "we" as in IETF as a whole) to make this=
 happen.

I may be wrong, and if need be we'll have one slide per solution, but I sus=
pect all the discussions that identify a need for an RTP header extension a=
re trying to address a similar problem. It may be that we can corner the va=
rious authors into a hallway discussion on Monday and get them to come up w=
ith a single idea of what is needed.

That means we don't necessarily need the MMUSIC discussion to happen first,=
 even if it makes any AVTEXT discussion results conditional on MMUSIC makin=
g some form of agreement.

The AVTEXT chairs would welcome any input to this.

Regards

Keith

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of Harald Alvestrand
> Sent: 18 July 2013 13:26
> To: mmusic@ietf.org
> Subject: Re: [MMUSIC] MMUSIC Draft Agenda for IETF 87
>=20
> On 07/17/2013 11:49 PM, Flemming Andreasen wrote:
>=20
>=20
> 	The MMUSIC draft agenda for IETF 87 has been posted at:
>=20
> 	https://datatracker.ietf.org/meeting/87/agenda/mmusic/
>=20
> 	Please take a look and let us know if you have any questions or
> comments.
>=20
>=20
>=20
> Hi,
>=20
> I note that AVTEXT's posted agenda:
>=20
> https://datatracker.ietf.org/meeting/87/agenda/avtext/
>=20
> has them meeting on Tuesday at 1700, and they want to consider the result
> of MMUSIC's discussions on draft-roach-mmusic-unified-plan - which
> according to
>=20
> https://datatracker.ietf.org/meeting/87/agenda/mmusic/
>=20
> will take place on Wednesday, making this a little bit difficult absent
> temporal manipulation.
>=20
> Perhaps the chairs should consult on whether it's worth swapping the two
> days of MMUSIC discussion in order to accomodate the AVTEXT linkage?
>=20
>=20
>=20
>=20
> 	Thanks
>=20
> 	    Ari & Flemming
> 	_______________________________________________
> 	mmusic mailing list
> 	mmusic@ietf.org
> 	https://www.ietf.org/mailman/listinfo/mmusic
>=20
>=20


From keith.drage@alcatel-lucent.com  Thu Jul 18 05:55:13 2013
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] MMUSIC Draft Agenda for IETF 87
Thread-Index: AQHOg7IEquvwH4osDkel9YL+QyfPsplqXp2AgAAFdhA=
Date: Thu, 18 Jul 2013 12:54:40 +0000
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Just to also note that we haven't directly yet opened up a discussion with =
the MMUSIC chairs on this, AVTEXT chairs were busy working out what they wa=
nted to cover last night before we hit the agenda deadline.

Keith

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
> Of DRAGE, Keith (Keith)
> Sent: 18 July 2013 13:46
> To: Harald Alvestrand; mmusic@ietf.org
> Subject: Re: [MMUSIC] MMUSIC Draft Agenda for IETF 87
>=20
> Ideally, AVTEXT (as in the AVTEXT chairs) would have liked the MMUSIC
> discussion first, but I suspect they are dealing with a whole lot of othe=
r
> scheduling problems.
>=20
> I'd also note that the AVTEXT agenda point relates to one very small part
> of this discussion.
>=20
> There have been a number of mentions of the need for RTP header extension=
s,
> which are therefore within the scope of AVTEXT. Timescales are getting
> shorter and shorter for this to happen, so we want to open up some
> information exchange about:
>=20
> 1)	Is this a sensible usage of a RTP header extension;
> 2)	What we need to do (and that is "we" as in IETF as a whole) to make
> this happen.
>=20
> I may be wrong, and if need be we'll have one slide per solution, but I
> suspect all the discussions that identify a need for an RTP header
> extension are trying to address a similar problem. It may be that we can
> corner the various authors into a hallway discussion on Monday and get
> them to come up with a single idea of what is needed.
>=20
> That means we don't necessarily need the MMUSIC discussion to happen firs=
t,
> even if it makes any AVTEXT discussion results conditional on MMUSIC
> making some form of agreement.
>=20
> The AVTEXT chairs would welcome any input to this.
>=20
> Regards
>=20
> Keith
>=20
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behal=
f
> > Of Harald Alvestrand
> > Sent: 18 July 2013 13:26
> > To: mmusic@ietf.org
> > Subject: Re: [MMUSIC] MMUSIC Draft Agenda for IETF 87
> >
> > On 07/17/2013 11:49 PM, Flemming Andreasen wrote:
> >
> >
> > 	The MMUSIC draft agenda for IETF 87 has been posted at:
> >
> > 	https://datatracker.ietf.org/meeting/87/agenda/mmusic/
> >
> > 	Please take a look and let us know if you have any questions or
> > comments.
> >
> >
> >
> > Hi,
> >
> > I note that AVTEXT's posted agenda:
> >
> > https://datatracker.ietf.org/meeting/87/agenda/avtext/
> >
> > has them meeting on Tuesday at 1700, and they want to consider the
> result
> > of MMUSIC's discussions on draft-roach-mmusic-unified-plan - which
> > according to
> >
> > https://datatracker.ietf.org/meeting/87/agenda/mmusic/
> >
> > will take place on Wednesday, making this a little bit difficult absent
> > temporal manipulation.
> >
> > Perhaps the chairs should consult on whether it's worth swapping the tw=
o
> > days of MMUSIC discussion in order to accomodate the AVTEXT linkage?
> >
> >
> >
> >
> > 	Thanks
> >
> > 	    Ari & Flemming
> > 	_______________________________________________
> > 	mmusic mailing list
> > 	mmusic@ietf.org
> > 	https://www.ietf.org/mailman/listinfo/mmusic
> >
> >
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From thomas.belling@nsn.com  Thu Jul 18 06:06:17 2013
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From: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>, ext Adam Roach <adam@nostrum.com>
Thread-Topic: [MMUSIC] Unified Plan for SDP Handling - msid attribute and media stream track
Thread-Index: AQHOg7eP8agj5MucQUyNspbOl23nzA==
Date: Thu, 18 Jul 2013 13:06:04 +0000
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling - msid attribute and media stream track
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Dear Adam, all,


good to see significant progress.


One aspect I still struggle to understand is the need for the "msid" attrib=
ute.



The draft says:

2.  Solution Overview

   At a high level, the solution described in this document can be
   summarized as follows:

   1.  Each media stream track is represented by its own unique m-line.
       This is a strict one-to-one mapping; a single media stream track
       cannot be spread across several m-lines, nor may a single m-line
       represent multiple media stream tracks. ...

   3.  Each m-line contains an MSID value to correlate it with a Media
       Stream ID and the Media Stream Track ID.


3.2.2.  Correlating Media Stream Tracks with m-lines

   Media Stream Tracks IDs are correlated with M-Lines directly by
   including an MSID in each m-line.  The MSID also provides the Media
   Stream ID.  (Note the format of the MSID used here is slightly
   different than what was proposed in the current MSID draft as that
   draft assumed multiple tracks in a single m-line and this proposal
   moves to a solution where there is a one to one relation between the
   Track and MSID.  This work assumes the MSID draft will be updated to
   match the syntax used user which simply provides the value of the
   MediaStream ID and MediaStreamTrack ID on an "a=3Dmsid" line.  )


The draft lacks a related references, but seems to refer to draft-ietf-mmus=
ic-msid-00. However, it is assuming some not yet documented modifications o=
f that draft, which does not simplify the understanding (hopefully only a p=
roblem for a short time). What you have in mind seems no longer an extensio=
n of the a=3Dssrc attribute but an own SDP attribute.=20


Unfortunately there is no definition of media stream track.

More fundamentally, do we require the term "media stream track" at all (rat=
her than simply speaking about m-lines), if there is a one-to-one mapping t=
o a unique m-line?
And do we require an extra identifier for it, in addition of the position (=
that could also be expressed by an ordinal number in some separate protocol=
 or for cross-referencing within SDP) of the m-line in the SDP?
I suspect the need for a new identifier of the m-line/media stream track ha=
s something to do with partial SDP offers? But could the identifier not sti=
ll be the ordinal number with respect to the "full" SDP (or the next free n=
umber if an m-line is added), and be restricted to partial offers?


The msid attribute also assings a "media stream ID", but your draft does no=
t describe at all what that ID would be used for.



Thomas



----------------------------------=20
Dr. Thomas Belling=20
3GPP Standardisation
Nokia Siemens Networks=20




-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 ext Adam Roach
Sent: Monday, July 15, 2013 9:05 PM
To: mmusic@ietf.org
Cc: rtcweb@ietf.org
Subject: [MMUSIC] Unified Plan for SDP Handling

[Cross-posting to RTCWEB; follow-ups to MMUSIC, please]

After significant work, Justin, Martin and I have managed to produce a comp=
romise plan that provides a high degree of interoperability with existing d=
evices (and future non-WebRTC devices) while not being excessively onerous =
for WebRTC implementations or applications that use them. It's been a trick=
y balancing act, but I think we've found a good mix between the two that ca=
n form a solid basis for the working group to move forward.

Rather than summarize the key points of the document in this email, I direc=
t interested parties to section 2 of the document, which summarizes the key=
 aspects of the plan in eight relatively concise bullet points.

I apologize for the late publication date of this document -- there's actua=
lly been a lot more work put into coming up with a unified draft than I ori=
ginally anticipated, and the production of this document took at least two =
weeks longer than I expected it to.

Note that this document is intended to be a plan for the work to be done in=
 this area, and not a specification in itself. The intention is that its co=
ntents are used as the basis for work in several other drafts -- some new, =
some not -- that form the corpus of work necessary for RTCWEB (and potentia=
lly CLUE) to move forward. Except in rare cases, the document does not atte=
mpt to explicitly call out venues or documents for such work, as we (or, at=
 the very least, I) anticipate guidance from the various working group chai=
rs to assist in such decisions.

Comments prior to Berlin would be very helpful, although this will clearly =
be a point of significant discussion at the face-to-face meeting.

Document link:

http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt

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

From harald@alvestrand.no  Thu Jul 18 06:41:08 2013
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling - msid attribute and media stream track
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On 07/18/2013 03:06 PM, Belling, Thomas (NSN - DE/Munich) wrote:
> Dear Adam, all,
>
>
> good to see significant progress.
>
>
> One aspect I still struggle to understand is the need for the "msid" attribute.
>
>
>
> The draft says:
>
> 2.  Solution Overview
>
>     At a high level, the solution described in this document can be
>     summarized as follows:
>
>     1.  Each media stream track is represented by its own unique m-line.
>         This is a strict one-to-one mapping; a single media stream track
>         cannot be spread across several m-lines, nor may a single m-line
>         represent multiple media stream tracks. ...
>
>     3.  Each m-line contains an MSID value to correlate it with a Media
>         Stream ID and the Media Stream Track ID.
>
>
> 3.2.2.  Correlating Media Stream Tracks with m-lines
>
>     Media Stream Tracks IDs are correlated with M-Lines directly by
>     including an MSID in each m-line.  The MSID also provides the Media
>     Stream ID.  (Note the format of the MSID used here is slightly
>     different than what was proposed in the current MSID draft as that
>     draft assumed multiple tracks in a single m-line and this proposal
>     moves to a solution where there is a one to one relation between the
>     Track and MSID.  This work assumes the MSID draft will be updated to
>     match the syntax used user which simply provides the value of the
>     MediaStream ID and MediaStreamTrack ID on an "a=msid" line.  )
>
>
> The draft lacks a related references, but seems to refer to draft-ietf-mmusic-msid-00. However, it is assuming some not yet documented modifications of that draft, which does not simplify the understanding (hopefully only a problem for a short time). What you have in mind seems no longer an extension of the a=ssrc attribute but an own SDP attribute.

I'll fix the MSID draft once I-D submission reopens. Yes, when an MSID 
is to be assigned for a whole M-line, it needs to be its own attribute, 
not an a=ssrc attribute (and that is the only change that needs to be 
made, I think).
>
>
> Unfortunately there is no definition of media stream track.
>
> More fundamentally, do we require the term "media stream track" at all (rather than simply speaking about m-lines), if there is a one-to-one mapping to a unique m-line?

I think it would actually be better if this draft talked in terms of 
media sources (as defined in the taxonomy draft I noted in my review), 
and just mentioned that in the case of WebRTC, a media source 
corresponds to the JavaScript object called MediaStreamTrack.

> And do we require an extra identifier for it, in addition of the position (that could also be expressed by an ordinal number in some separate protocol or for cross-referencing within SDP) of the m-line in the SDP?
> I suspect the need for a new identifier of the m-line/media stream track has something to do with partial SDP offers? But could the identifier not still be the ordinal number with respect to the "full" SDP (or the next free number if an m-line is added), and be restricted to partial offers?
>
>
> The msid attribute also assings a "media stream ID", but your draft does not describe at all what that ID would be used for.

The ID of a WEBRTC MediaStream is defined in 
http://dev.w3.org/2011/webrtc/editor/getusermedia.html#attributes
>
>
>
> Thomas
>
>
>
> ----------------------------------
> Dr. Thomas Belling
> 3GPP Standardisation
> Nokia Siemens Networks
>
>
>
>
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of ext Adam Roach
> Sent: Monday, July 15, 2013 9:05 PM
> To: mmusic@ietf.org
> Cc: rtcweb@ietf.org
> Subject: [MMUSIC] Unified Plan for SDP Handling
>
> [Cross-posting to RTCWEB; follow-ups to MMUSIC, please]
>
> After significant work, Justin, Martin and I have managed to produce a compromise plan that provides a high degree of interoperability with existing devices (and future non-WebRTC devices) while not being excessively onerous for WebRTC implementations or applications that use them. It's been a tricky balancing act, but I think we've found a good mix between the two that can form a solid basis for the working group to move forward.
>
> Rather than summarize the key points of the document in this email, I direct interested parties to section 2 of the document, which summarizes the key aspects of the plan in eight relatively concise bullet points.
>
> I apologize for the late publication date of this document -- there's actually been a lot more work put into coming up with a unified draft than I originally anticipated, and the production of this document took at least two weeks longer than I expected it to.
>
> Note that this document is intended to be a plan for the work to be done in this area, and not a specification in itself. The intention is that its contents are used as the basis for work in several other drafts -- some new, some not -- that form the corpus of work necessary for RTCWEB (and potentially CLUE) to move forward. Except in rare cases, the document does not attempt to explicitly call out venues or documents for such work, as we (or, at the very least, I) anticipate guidance from the various working group chairs to assist in such decisions.
>
> Comments prior to Berlin would be very helpful, although this will clearly be a point of significant discussion at the face-to-face meeting.
>
> Document link:
>
> http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt
>
> /a
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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    <div class="moz-cite-prefix">On 07/18/2013 03:06 PM, Belling, Thomas
      (NSN - DE/Munich) wrote:<br>
    </div>
    <blockquote
cite="mid:BDBE1A97E84675488F72A48C23811F351179A4@DEMUMBX001.nsn-intra.net"
      type="cite">
      <pre wrap="">Dear Adam, all,


good to see significant progress.


One aspect I still struggle to understand is the need for the "msid" attribute.



The draft says:

2.  Solution Overview

   At a high level, the solution described in this document can be
   summarized as follows:

   1.  Each media stream track is represented by its own unique m-line.
       This is a strict one-to-one mapping; a single media stream track
       cannot be spread across several m-lines, nor may a single m-line
       represent multiple media stream tracks. ...

   3.  Each m-line contains an MSID value to correlate it with a Media
       Stream ID and the Media Stream Track ID.


3.2.2.  Correlating Media Stream Tracks with m-lines

   Media Stream Tracks IDs are correlated with M-Lines directly by
   including an MSID in each m-line.  The MSID also provides the Media
   Stream ID.  (Note the format of the MSID used here is slightly
   different than what was proposed in the current MSID draft as that
   draft assumed multiple tracks in a single m-line and this proposal
   moves to a solution where there is a one to one relation between the
   Track and MSID.  This work assumes the MSID draft will be updated to
   match the syntax used user which simply provides the value of the
   MediaStream ID and MediaStreamTrack ID on an "a=msid" line.  )


The draft lacks a related references, but seems to refer to draft-ietf-mmusic-msid-00. However, it is assuming some not yet documented modifications of that draft, which does not simplify the understanding (hopefully only a problem for a short time). What you have in mind seems no longer an extension of the a=ssrc attribute but an own SDP attribute. </pre>
    </blockquote>
    <br>
    I'll fix the MSID draft once I-D submission reopens. Yes, when an
    MSID is to be assigned for a whole M-line, it needs to be its own
    attribute, not an a=ssrc attribute (and that is the only change that
    needs to be made, I think).<br>
    <blockquote
cite="mid:BDBE1A97E84675488F72A48C23811F351179A4@DEMUMBX001.nsn-intra.net"
      type="cite">
      <pre wrap="">


Unfortunately there is no definition of media stream track.

More fundamentally, do we require the term "media stream track" at all (rather than simply speaking about m-lines), if there is a one-to-one mapping to a unique m-line?</pre>
    </blockquote>
    <br>
    I think it would actually be better if this draft talked in terms of
    media sources (as defined in the taxonomy draft I noted in my
    review), and just mentioned that in the case of WebRTC, a media
    source corresponds to the JavaScript object called MediaStreamTrack.<br>
    <br>
    <blockquote
cite="mid:BDBE1A97E84675488F72A48C23811F351179A4@DEMUMBX001.nsn-intra.net"
      type="cite">
      <pre wrap="">
And do we require an extra identifier for it, in addition of the position (that could also be expressed by an ordinal number in some separate protocol or for cross-referencing within SDP) of the m-line in the SDP?
I suspect the need for a new identifier of the m-line/media stream track has something to do with partial SDP offers? But could the identifier not still be the ordinal number with respect to the "full" SDP (or the next free number if an m-line is added), and be restricted to partial offers?


The msid attribute also assings a "media stream ID", but your draft does not describe at all what that ID would be used for.</pre>
    </blockquote>
    <br>
    The ID of a WEBRTC MediaStream is defined in
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <a
      href="http://dev.w3.org/2011/webrtc/editor/getusermedia.html#attributes">http://dev.w3.org/2011/webrtc/editor/getusermedia.html#attributes</a>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <blockquote
cite="mid:BDBE1A97E84675488F72A48C23811F351179A4@DEMUMBX001.nsn-intra.net"
      type="cite">
      <pre wrap="">



Thomas



---------------------------------- 
Dr. Thomas Belling 
3GPP Standardisation
Nokia Siemens Networks 




-----Original Message-----
From: <a class="moz-txt-link-abbreviated" href="mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a> [<a class="moz-txt-link-freetext" href="mailto:mmusic-bounces@ietf.org">mailto:mmusic-bounces@ietf.org</a>] On Behalf Of ext Adam Roach
Sent: Monday, July 15, 2013 9:05 PM
To: <a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
Cc: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
Subject: [MMUSIC] Unified Plan for SDP Handling

[Cross-posting to RTCWEB; follow-ups to MMUSIC, please]

After significant work, Justin, Martin and I have managed to produce a compromise plan that provides a high degree of interoperability with existing devices (and future non-WebRTC devices) while not being excessively onerous for WebRTC implementations or applications that use them. It's been a tricky balancing act, but I think we've found a good mix between the two that can form a solid basis for the working group to move forward.

Rather than summarize the key points of the document in this email, I direct interested parties to section 2 of the document, which summarizes the key aspects of the plan in eight relatively concise bullet points.

I apologize for the late publication date of this document -- there's actually been a lot more work put into coming up with a unified draft than I originally anticipated, and the production of this document took at least two weeks longer than I expected it to.

Note that this document is intended to be a plan for the work to be done in this area, and not a specification in itself. The intention is that its contents are used as the basis for work in several other drafts -- some new, some not -- that form the corpus of work necessary for RTCWEB (and potentially CLUE) to move forward. Except in rare cases, the document does not attempt to explicitly call out venues or documents for such work, as we (or, at the very least, I) anticipate guidance from the various working group chairs to assist in such decisions.

Comments prior to Berlin would be very helpful, although this will clearly be a point of significant discussion at the face-to-face meeting.

Document link:

<a class="moz-txt-link-freetext" href="http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt">http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt</a>

/a
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</pre>
    </blockquote>
    <br>
  </body>
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From: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>
To: ext Harald Alvestrand <harald@alvestrand.no>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Unified Plan for SDP Handling - msid attribute and media stream track
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Thanks Harald,

I understand the Media Stream ID and Media Stream Track ID appear in the W3=
C API.

However, this does not necessarily mean they need to be signaled end-to-end=
 (rather than being only of local significance for the interactions between=
 the JavaScript application and the API, and being mapped to/from some othe=
r SDP properties such as the ordinal number of the SDP m-line).

If something important was missing in SDP up to now, we certainly need to a=
dd it. But we should not duplicate information. This is what I try to find =
out. And obviously, requiring mandatory SDP extensions does not simplify in=
teroperability with existing SDP peers.

Thomas

From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 ext Harald Alvestrand
Sent: Thursday, July 18, 2013 3:41 PM
To: mmusic@ietf.org
Subject: Re: [MMUSIC] Unified Plan for SDP Handling - msid attribute and me=
dia stream track

On 07/18/2013 03:06 PM, Belling, Thomas (NSN - DE/Munich) wrote:
...



And do we require an extra identifier for it, in addition of the position (=
that could also be expressed by an ordinal number in some separate protocol=
 or for cross-referencing within SDP) of the m-line in the SDP?

I suspect the need for a new identifier of the m-line/media stream track ha=
s something to do with partial SDP offers? But could the identifier not sti=
ll be the ordinal number with respect to the "full" SDP (or the next free n=
umber if an m-line is added), and be restricted to partial offers?





The msid attribute also assings a "media stream ID", but your draft does no=
t describe at all what that ID would be used for.

The ID of a WEBRTC MediaStream is defined in http://dev.w3.org/2011/webrtc/=
editor/getusermedia.html#attributes







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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">Thanks Harald,<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 lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I understa=
nd the Media Stream ID and Media Stream Track ID appear in the W3C API.<o:p=
></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">However, t=
his does not necessarily mean they need to be signaled end-to-end (rather t=
han being only of local significance for the interactions
 between the JavaScript application and the API, and being mapped to/from s=
ome other SDP properties such as the ordinal number of the SDP m-line).<o:p=
></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">If somethi=
ng important was missing in SDP up to now, we certainly need to add it. But=
 we should not duplicate information. This is what I try to
 find out. And obviously, requiring mandatory SDP extensions does not simpl=
ify interoperability with existing SDP peers.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thomas<o:p=
></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:=
</span></b><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext"> mmusic-bounces@ietf=
.org [mailto:mmusic-bounces@ietf.org]
<b>On Behalf Of </b>ext Harald Alvestrand<br>
<b>Sent:</b> Thursday, July 18, 2013 3:41 PM<br>
<b>To:</b> mmusic@ietf.org<br>
<b>Subject:</b> Re: [MMUSIC] Unified Plan for SDP Handling - msid attribute=
 and media stream track<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On 07/18/2013 03:06 PM, Belling, Thomas (NSN - DE/Mu=
nich) wrote:<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&#8230;</span><o:p></o=
:p></p>
<pre><o:p>&nbsp;</o:p></pre>
<pre>And do we require an extra identifier for it, in addition of the posit=
ion (that could also be expressed by an ordinal number in some separate pro=
tocol or for cross-referencing within SDP) of the m-line in the SDP?<o:p></=
o:p></pre>
<pre>I suspect the need for a new identifier of the m-line/media stream tra=
ck has something to do with partial SDP offers? But could the identifier no=
t still be the ordinal number with respect to the &quot;full&quot; SDP (or =
the next free number if an m-line is added), and be restricted to partial o=
ffers?<o:p></o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre>The msid attribute also assings a &quot;media stream ID&quot;, but you=
r draft does not describe at all what that ID would be used for.<o:p></o:p>=
</pre>
<p class=3D"MsoNormal"><br>
The ID of a WEBRTC MediaStream is defined in <a href=3D"http://dev.w3.org/2=
011/webrtc/editor/getusermedia.html#attributes">
http://dev.w3.org/2011/webrtc/editor/getusermedia.html#attributes</a> <o:p>=
</o:p></p>
<pre><o:p>&nbsp;</o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
<pre><o:p>&nbsp;</o:p></pre>
</div>
</body>
</html>

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From harald@alvestrand.no  Thu Jul 18 07:51:31 2013
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Unified Plan for SDP Handling - msid attribute and media stream track
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On 07/18/2013 04:21 PM, Belling, Thomas (NSN - DE/Munich) wrote:
>
> Thanks Harald,
>
> I understand the Media Stream ID and Media Stream Track ID appear in 
> the W3C API.
>
> However, this does not necessarily mean they need to be signaled 
> end-to-end (rather than being only of local significance for the 
> interactions between the JavaScript application and the API, and being 
> mapped to/from some other SDP properties such as the ordinal number of 
> the SDP m-line).
>

Well... if we want to have the same ID attached to the same 
MediaStreamTrack at both ends, the ID has to be carried somehow.

If there's no need for the ID to be the same, we don't need it. But so 
far, I think the consensus in the W3C WG is that it needs to be the same.

> If something important was missing in SDP up to now, we certainly need 
> to add it. But we should not duplicate information. This is what I try 
> to find out. And obviously, requiring mandatory SDP extensions does 
> not simplify interoperability with existing SDP peers.
>

That's why the MSID spec has a section on what to do when you get tracks 
without MSIDs.

> Thomas
>
> *From:*mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] *On 
> Behalf Of *ext Harald Alvestrand
> *Sent:* Thursday, July 18, 2013 3:41 PM
> *To:* mmusic@ietf.org
> *Subject:* Re: [MMUSIC] Unified Plan for SDP Handling - msid attribute 
> and media stream track
>
> On 07/18/2013 03:06 PM, Belling, Thomas (NSN - DE/Munich) wrote:
>
> ...
>
>   
> And do we require an extra identifier for it, in addition of the position (that could also be expressed by an ordinal number in some separate protocol or for cross-referencing within SDP) of the m-line in the SDP?
> I suspect the need for a new identifier of the m-line/media stream track has something to do with partial SDP offers? But could the identifier not still be the ordinal number with respect to the "full" SDP (or the next free number if an m-line is added), and be restricted to partial offers?
>   
>   
> The msid attribute also assings a "media stream ID", but your draft does not describe at all what that ID would be used for.
>
>
> The ID of a WEBRTC MediaStream is defined in 
> http://dev.w3.org/2011/webrtc/editor/getusermedia.html#attributes
>
>   
>   
>   


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    <div class="moz-cite-prefix">On 07/18/2013 04:21 PM, Belling, Thomas
      (NSN - DE/Munich) wrote:<br>
    </div>
    <blockquote
cite="mid:BDBE1A97E84675488F72A48C23811F351179DE@DEMUMBX001.nsn-intra.net"
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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">Thanks
            Harald,<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"
            lang="EN-US">I understand the Media Stream ID and Media
            Stream Track ID appear in the W3C API.<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"
            lang="EN-US"><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"
            lang="EN-US">However, this does not necessarily mean they
            need to be signaled end-to-end (rather than being only of
            local significance for the interactions between the
            JavaScript application and the API, and being mapped to/from
            some other SDP properties such as the ordinal number of the
            SDP m-line).</span></p>
      </div>
    </blockquote>
    <br>
    Well... if we want to have the same ID attached to the same
    MediaStreamTrack at both ends, the ID has to be carried somehow.<br>
    <br>
    If there's no need for the ID to be the same, we don't need it. But
    so far, I think the consensus in the W3C WG is that it needs to be
    the same.<br>
    <br>
    <blockquote
cite="mid:BDBE1A97E84675488F72A48C23811F351179DE@DEMUMBX001.nsn-intra.net"
      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"
            lang="EN-US"><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"
            lang="EN-US"><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"
            lang="EN-US">If something important was missing in SDP up to
            now, we certainly need to add it. But we should not
            duplicate information. This is what I try to find out. And
            obviously, requiring mandatory SDP extensions does not
            simplify interoperability with existing SDP peers.</span></p>
      </div>
    </blockquote>
    <br>
    That's why the MSID spec has a section on what to do when you get
    tracks without MSIDs.<br>
    <br>
    <blockquote
cite="mid:BDBE1A97E84675488F72A48C23811F351179DE@DEMUMBX001.nsn-intra.net"
      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"
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        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><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"
            lang="EN-US">Thomas<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"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <div>
          <div style="border:none;border-top:solid #B5C4DF
            1.0pt;padding:3.0pt 0cm 0cm 0cm">
            <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext"
                  lang="EN-US">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext"
                lang="EN-US"> <a class="moz-txt-link-abbreviated" href="mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a>
                [<a class="moz-txt-link-freetext" href="mailto:mmusic-bounces@ietf.org">mailto:mmusic-bounces@ietf.org</a>]
                <b>On Behalf Of </b>ext Harald Alvestrand<br>
                <b>Sent:</b> Thursday, July 18, 2013 3:41 PM<br>
                <b>To:</b> <a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
                <b>Subject:</b> Re: [MMUSIC] Unified Plan for SDP
                Handling - msid attribute and media stream track<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <div>
          <p class="MsoNormal">On 07/18/2013 03:06 PM, Belling, Thomas
            (NSN - DE/Munich) wrote:<o:p></o:p></p>
        </div>
        <p class="MsoNormal"><span style="color:#1F497D">&#8230;</span><o:p></o:p></p>
        <pre><o:p>&nbsp;</o:p></pre>
        <pre>And do we require an extra identifier for it, in addition of the position (that could also be expressed by an ordinal number in some separate protocol or for cross-referencing within SDP) of the m-line in the SDP?<o:p></o:p></pre>
        <pre>I suspect the need for a new identifier of the m-line/media stream track has something to do with partial SDP offers? But could the identifier not still be the ordinal number with respect to the "full" SDP (or the next free number if an m-line is added), and be restricted to partial offers?<o:p></o:p></pre>
        <pre><o:p>&nbsp;</o:p></pre>
        <pre><o:p>&nbsp;</o:p></pre>
        <pre>The msid attribute also assings a "media stream ID", but your draft does not describe at all what that ID would be used for.<o:p></o:p></pre>
        <p class="MsoNormal"><br>
          The ID of a WEBRTC MediaStream is defined in <a
            moz-do-not-send="true"
            href="http://dev.w3.org/2011/webrtc/editor/getusermedia.html#attributes">
http://dev.w3.org/2011/webrtc/editor/getusermedia.html#attributes</a> <o:p></o:p></p>
        <pre><o:p>&nbsp;</o:p></pre>
        <pre><o:p>&nbsp;</o:p></pre>
        <pre><o:p>&nbsp;</o:p></pre>
      </div>
    </blockquote>
    <br>
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From thomas.belling@nsn.com  Thu Jul 18 08:18:36 2013
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From: "Belling, Thomas (NSN - DE/Munich)" <thomas.belling@nsn.com>
To: ext Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [MMUSIC] Unified Plan for SDP Handling - msid attribute and media stream track
Thread-Index: AQHOg7eYiqJibBkYeUq8/d0kMSi315lqhQeDgAACXAA=
Date: Thu, 18 Jul 2013 15:18:24 +0000
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling - msid attribute and media stream track
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That is where my confusion starts:
The MSID draft, and also your related backward compatibility procedures, as=
sumes that multiple tracks can be described by one m-line.
This is no longer true.

For the MediaStreamTrack ID, I still suspect that a standardized mapping to=
 m-line ordinal numbers could now be sufficient to have the same IDs at bot=
h sides


For the MediaStream, the API description says:
Each MediaStream<http://dev.w3.org/2011/webrtc/editor/getusermedia.html#idl=
-def-MediaStream> object can contain zero or more tracks, in particular aud=
io and video tracks. All tracks in a MediaStream are intended to be synchro=
nized when rendered. Different MediaStreams do not need to be synchronized.

This sounds like a lipsynch group in SDP.
The group name can then serve as MediaStrem ID.  Each m-line outside such a=
 group would probably have a separate MediaStream,  and the m-line ordinal =
number could then serve as MediaStream ID.



From: ext Harald Alvestrand [mailto:harald@alvestrand.no]
Sent: Thursday, July 18, 2013 4:51 PM
To: Belling, Thomas (NSN - DE/Munich)
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Unified Plan for SDP Handling - msid attribute and me=
dia stream track

On 07/18/2013 04:21 PM, Belling, Thomas (NSN - DE/Munich) wrote:
Thanks Harald,

I understand the Media Stream ID and Media Stream Track ID appear in the W3=
C API.

However, this does not necessarily mean they need to be signaled end-to-end=
 (rather than being only of local significance for the interactions between=
 the JavaScript application and the API, and being mapped to/from some othe=
r SDP properties such as the ordinal number of the SDP m-line).

Well... if we want to have the same ID attached to the same MediaStreamTrac=
k at both ends, the ID has to be carried somehow.

If there's no need for the ID to be the same, we don't need it. But so far,=
 I think the consensus in the W3C WG is that it needs to be the same.



If something important was missing in SDP up to now, we certainly need to a=
dd it. But we should not duplicate information. This is what I try to find =
out. And obviously, requiring mandatory SDP extensions does not simplify in=
teroperability with existing SDP peers.

That's why the MSID spec has a section on what to do when you get tracks wi=
thout MSIDs.



Thomas

From: mmusic-bounces@ietf.org<mailto:mmusic-bounces@ietf.org> [mailto:mmusi=
c-bounces@ietf.org] On Behalf Of ext Harald Alvestrand
Sent: Thursday, July 18, 2013 3:41 PM
To: mmusic@ietf.org<mailto:mmusic@ietf.org>
Subject: Re: [MMUSIC] Unified Plan for SDP Handling - msid attribute and me=
dia stream track

On 07/18/2013 03:06 PM, Belling, Thomas (NSN - DE/Munich) wrote:
...



And do we require an extra identifier for it, in addition of the position (=
that could also be expressed by an ordinal number in some separate protocol=
 or for cross-referencing within SDP) of the m-line in the SDP?

I suspect the need for a new identifier of the m-line/media stream track ha=
s something to do with partial SDP offers? But could the identifier not sti=
ll be the ordinal number with respect to the "full" SDP (or the next free n=
umber if an m-line is added), and be restricted to partial offers?





The msid attribute also assings a "media stream ID", but your draft does no=
t describe at all what that ID would be used for.

The ID of a WEBRTC MediaStream is defined in http://dev.w3.org/2011/webrtc/=
editor/getusermedia.html#attributes








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<body bgcolor=3D"white" lang=3D"DE" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">That is wh=
ere my confusion starts:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">The MSID d=
raft, and also your related backward compatibility procedures, assumes that=
 multiple tracks can be described by one m-line.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">This is no=
 longer true.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">For the Me=
diaStreamTrack ID, I still suspect that a standardized mapping to m-line or=
dinal numbers could now be sufficient to have the same IDs
 at both sides<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">For the Me=
diaStream, the API description says:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-family:&quot;Aria=
l&quot;,&quot;sans-serif&quot;">Each
</span><code><span lang=3D"EN-US" style=3D"font-size:10.0pt"><a href=3D"htt=
p://dev.w3.org/2011/webrtc/editor/getusermedia.html#idl-def-MediaStream"><b=
><span style=3D"color:orangered;text-decoration:none">MediaStream</span></b=
></a></span></code><span lang=3D"EN-US" style=3D"font-family:&quot;Arial&qu=
ot;,&quot;sans-serif&quot;">
 object can contain zero or more tracks, in particular audio and video trac=
ks. All tracks in a MediaStream are intended to be synchronized when render=
ed. Different MediaStreams do not need to be synchronized.</span><span lang=
=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;=
sans-serif&quot;;color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">This sound=
s like a lipsynch group in SDP.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">The group =
name can then serve as MediaStrem ID. &nbsp;Each m-line outside such a grou=
p would probably have a separate MediaStream, &nbsp;and the m-line ordinal
 number could then serve as MediaStream ID.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:=
</span></b><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext"> ext Harald Alvestra=
nd [mailto:harald@alvestrand.no]
<br>
<b>Sent:</b> Thursday, July 18, 2013 4:51 PM<br>
<b>To:</b> Belling, Thomas (NSN - DE/Munich)<br>
<b>Cc:</b> mmusic@ietf.org<br>
<b>Subject:</b> Re: [MMUSIC] Unified Plan for SDP Handling - msid attribute=
 and media stream track<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On 07/18/2013 04:21 PM, Belling, Thomas (NSN - DE/Mu=
nich) wrote:<o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thanks Harald,</span><o:p=
></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I understa=
nd the Media Stream ID and Media Stream Track ID appear in the W3C API.</sp=
an><o:p></o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">However, t=
his does not necessarily mean they need to be signaled end-to-end (rather t=
han being only of local significance for the interactions
 between the JavaScript application and the API, and being mapped to/from s=
ome other SDP properties such as the ordinal number of the SDP m-line).</sp=
an><o:p></o:p></p>
</blockquote>
<p class=3D"MsoNormal"><br>
Well... if we want to have the same ID attached to the same MediaStreamTrac=
k at both ends, the ID has to be carried somehow.<br>
<br>
If there's no need for the ID to be the same, we don't need it. But so far,=
 I think the consensus in the W3C WG is that it needs to be the same.<br>
<br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">If somethi=
ng important was missing in SDP up to now, we certainly need to add it. But=
 we should not duplicate information. This is what I try to
 find out. And obviously, requiring mandatory SDP extensions does not simpl=
ify interoperability with existing SDP peers.</span><o:p></o:p></p>
<p class=3D"MsoNormal"><br>
That's why the MSID spec has a section on what to do when you get tracks wi=
thout MSIDs.<br>
<br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thomas</sp=
an><o:p></o:p></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:=
</span></b><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">
<a href=3D"mailto:mmusic-bounces@ietf.org">mmusic-bounces@ietf.org</a> [<a =
href=3D"mailto:mmusic-bounces@ietf.org">mailto:mmusic-bounces@ietf.org</a>]
<b>On Behalf Of </b>ext Harald Alvestrand<br>
<b>Sent:</b> Thursday, July 18, 2013 3:41 PM<br>
<b>To:</b> <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<b>Subject:</b> Re: [MMUSIC] Unified Plan for SDP Handling - msid attribute=
 and media stream track</span><o:p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">On 07/18/2013 03:06 PM, Belling, Thomas (NSN - DE/Mu=
nich) wrote:<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&#8230;</span><o:p></o=
:p></p>
<pre>&nbsp;<o:p></o:p></pre>
<pre>And do we require an extra identifier for it, in addition of the posit=
ion (that could also be expressed by an ordinal number in some separate pro=
tocol or for cross-referencing within SDP) of the m-line in the SDP?<o:p></=
o:p></pre>
<pre>I suspect the need for a new identifier of the m-line/media stream tra=
ck has something to do with partial SDP offers? But could the identifier no=
t still be the ordinal number with respect to the &quot;full&quot; SDP (or =
the next free number if an m-line is added), and be restricted to partial o=
ffers?<o:p></o:p></pre>
<pre>&nbsp;<o:p></o:p></pre>
<pre>&nbsp;<o:p></o:p></pre>
<pre>The msid attribute also assings a &quot;media stream ID&quot;, but you=
r draft does not describe at all what that ID would be used for.<o:p></o:p>=
</pre>
<p class=3D"MsoNormal"><br>
The ID of a WEBRTC MediaStream is defined in <a href=3D"http://dev.w3.org/2=
011/webrtc/editor/getusermedia.html#attributes">
http://dev.w3.org/2011/webrtc/editor/getusermedia.html#attributes</a> <o:p>=
</o:p></p>
<pre>&nbsp;<o:p></o:p></pre>
<pre>&nbsp;<o:p></o:p></pre>
<pre>&nbsp;<o:p></o:p></pre>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

--_000_BDBE1A97E84675488F72A48C23811F35117A17DEMUMBX001nsnintr_--

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<!doctype html public "-//W3C//DTD W3 HTML//EN">
<html><head><style type=3D"text/css"><!--
blockquote, dl, ul, ol, li { padding-top: 0 ; padding-bottom: 0 }
 --></style><title>Re: [MMUSIC]
draft-gellens-mmusic-negotiating-human-langua</title></head><body>
<div>At 12:14 AM +0200 7/15/13, Gunnar Hellstrom wrote:</div>
<div><br></div>
<blockquote type=3D"cite" cite>Randall,<br>
<br>
Yes, version 01 is a step forward.<br>
<br>
<br>
Comments:<br>
1. In 7.1 you now say:</blockquote>
<blockquote type=3D"cite" cite><tt>When placing an emergency call, and
in any other case where the<br>
&nbsp;&nbsp; language cannot be assumed from context, each media
stream in an<br>
&nbsp;&nbsp; offer SHOULD specify one or both 'humintlang-send' and
'humintlang-<br>
&nbsp;&nbsp; recv' attributes (to avoid ambiguity).</tt></blockquote>
<blockquote type=3D"cite" cite><br>
This might be a good advice. But to what extent SHOULD it be
followed?<br>
<br>
Some media might be included in SDP without the user having any
interest of using it for language communication.<br>
Video may be included without the person really preferring it for
spoken language understanding enhancement but just for pleasure or
seeing the scene of the other user.<br>
<br>
Do you mean that such cases should also be specified with humintlang
but with empty attributes?<br>
As a clear indication that the user is not interested in using that
medium for language transfer.<br>
That could make sense. That is different from not providing any
humintlang at all, that should be understood as undefined language
preferences.<br>
<br>
<br>
But there are media streams specified by SDP where a language
preference specification does not make sense. There are many
application and message media specifications that are not used for any
human language communication. I think for example that there are media
specifications for camera control.<br>
Is the SHOULD a sufficient weakening of the requirement, so that the
reader can skip using humintlang for media where it does not make
sense?<br>
If not, expand &quot;each media stream&quot;&nbsp; to: &quot;each
media stream in which human language communication might
occur&quot;.</blockquote>
<div><br></div>
<div>[RG] I have a -02 (which I can submit when the submission window
re-opens) in which I've attempted to clarify this by adding:</div>
<div>&nbsp;&nbsp; In cases where a media is</div>
<div>&nbsp;&nbsp; not primarily intended for language (for example, a
video or audio<br>
&nbsp;&nbsp; stream intended for background only) both SHOULD be
unset.</div>
<div><br></div>
<div>However, I'm not convinced that in reality clients will know how
the user intends to use various streams.&nbsp; I'd like to hear more
discussion on this point.</div>
<div><br></div>
<div><br></div>
<blockquote type=3D"cite" cite>2. You have also introduced this
paragraph in 7.1:<br>
&quot;</blockquote>
<blockquote type=3D"cite" cite><tt>Clients acting on behalf of end users
are expected to set one or both<br>
&nbsp;&nbsp; 'humintlang-send' and 'humintlang-recv' attributes on
each media<br>
&nbsp;&nbsp; stream in an offer when placing an outgoing session but
ignore the<br>
&nbsp;&nbsp; attributes when receiving incoming calls.&nbsp; Systems
acting on behalf<br>
&nbsp;&nbsp; of call centers and PSAPs are expected to take into
account the<br>
&nbsp;&nbsp; values when processing inbound calls.<br>
&quot;<br>
<br>
What is the thought behind this?<br>
<br>
I see the humintlang specification equally valuable in outgoing as
incoming calls, and for end-users as well as call-centers and
PSAPs.<br>
If a user gets an incoming call, with humintlang specifications, it
can compare with its own profile and decide to invoke a<br>
relay service or interpreter servicde to cover some differences in
language preferences, or just prepare for what media, modalities
and<br>
languages to use in the call..<br>
I do not see the background for this paragraph, and think it will be
better to delete it.</tt></blockquote>
<div><br></div>
<div>[RG] The reasoning is an attempt to not overly-complicate call
set-up with requirements that in reality will be difficult for clients
to handle in a reasonable way.&nbsp; I'd like to hear more views on
deleting it or modifying it.</div>
<div><br></div>
<blockquote type=3D"cite" cite><tt><br>
<br>
3. Absolute preference is still needed.<br>
I described earlier the need for an absolute preference indication.
But I did not provide any coding proposal.<br>
I suggest to add something at the end of the language tag. Regardless
if it gets before or after the * if it is there.<br>
Can a &quot;+&quot; be used to indicate a strongly preferred language
and modality, no extra specification a managed language and
modality<br>
and a &quot;-&quot; a non-preferred, but possible language and
modality.<br>
&nbsp;<br>
( needs to be checked against syntax of language tags.)<br>
<br>
&nbsp;<br>
Regards<br>
<br>
Gunnar</tt></blockquote>
<blockquote type=3D"cite" cite>
<hr></blockquote>
<blockquote type=3D"cite" cite>Gunnar Hellstr=F6m</blockquote>
<blockquote type=3D"cite" cite>Omnitor<br>
<a
href=3D"mailto:gunnar.hellstrom@omnitor.se">gunnar.hellstrom@omnitor.se</a
><br>
+46708204288<br>
</blockquote>
<blockquote type=3D"cite" cite>On 2013-07-14 11:05, Randall Gellens
wrote:<br>
<blockquote type=3D"cite" cite>Hi all,<br>
<br>
I've tweaked the human language draft:<br>
<br>
&nbsp;&nbsp; o&nbsp; Relaxed language on setting -send and -receive to
same values;<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; added text on leaving on empty to
indicate asymmetric usage.<br>
<br>
&nbsp;&nbsp; o&nbsp; Added text that clients on behalf of end users
are expected to set<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the attributes on outgoing calls and
ignore on incoming calls<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; while systems on behalf of call centers
and PSAPs are expected to<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; take the attributes into account when
processing incoming calls.<br>
</blockquote>
</blockquote>
<blockquote type=3D"cite" cite><br></blockquote>
<blockquote type=3D"cite" cite><br>
_______________________________________________<br>
mmusic mailing list<br>
mmusic@ietf.org<br>
https://www.ietf.org/mailman/listinfo/mmusic</blockquote>
<div><br></div>
<div><br></div>
<x-sigsep><pre>-- 
</pre></x-sigsep>
<div><font color=3D"#000000">Randall Gellens<br>
Opinions are personal;&nbsp;&nbsp;&nbsp; facts are
suspect;&nbsp;&nbsp;&nbsp; I speak for myself only<br>
-------------- Randomly selected tag: ---------------<br>
Books have to be read (worse luck it takes so long a time). &nbsp;It is the<br>
only way of discovering what they contain. &nbsp;A few savage tribes eat<br>
them, but reading is the only method of assimilation revealed to the<br>
West.<br>
 &nbsp;&nbsp;&nbsp;--E.M. Forster<br>
</font></div></body>
</html>

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Subject: Re: [MMUSIC] MMUSIC Draft Agenda for IETF 87
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On 17 July 2013 14:49, Flemming Andreasen <fandreas@cisco.com> wrote:
> The MMUSIC draft agenda for IETF 87

... unwrapped:

TUESDAY, July 30, 2013 (Bellevue), 13:00-15:00 (120 mins)
==========================================================

13:00-13:15 Introduction and Status Update (Chairs)

13:15-13:25, draft-wing-mmusic-ice-mobility-04 (10 mins)
   Mobility with ICE (MICE)

13:25-13:40, draft-reddy-mmusic-ice-happy-eyeballs-01 (15 mins)
   Happy Eyeballs Extension for ICE

13:40-13:55, draft-gellens-mmusic-negotiating-human-language-01 (15 mins)
   Negotiating Human Language Using SDP

13:55-14:10, draft-ivov-mmusic-trickle-ice-01 (15 mins)
            draft-ivov-mmusic-trickle-ice-sip-00
   Trickle ICE: Incremental Provisioning of Candidates for the
Interactive Connectivity Establishment (ICE) Protocol
   A Session Initiation Protocol (SIP) usage for Trickle ICE

14:10-14:25, draft-even-mmusic-application-token-00 (15 mins)
   The Session Description Protocol (SDP) Application Token Attribute

14:25-14:40, draft-ivov-mmusic-multiple-sources-00 (15 mins)
   Economical Use of the Offer/Answer Model in Sessions with Multiple
Media Sources

14:40-14:50, draft-nandakumar-mmusic-sdp-mux-attributes-03 (10 mins)
   A Framework for SDP Attributes when Multiplexing

14:50-15:00, draft-martinsen-mmusic-malice-00 (10 mins)
   Meta-data Attribute signaLling with ICE


WEDNESDAY, July 31, 2013 (Charlottenburg 2/3), 13:00-15:00 (120 mins)
======================================================================

13:00-13:40, draft-ietf-mmusic-sdp-bundle-negotiation-04 (40 mins)
   Multiplexing Negotiation Using Session Description Protocol (SDP)
Port Numbers

13:40-14:20, draft-roach-mmusic-unified-plan-00 (40 mins)
   A Unified Plan for Using SDP with Large Numbers of Media Flows

14:20-15:00 Plan Discussion and Decision on Path Forward (40 mins)



Overflow (draft presentation requests with no agenda time)
----------------------------------------------------------
draft-marcon-msrp-over-webrtc-data-channels-00
draft-nandakumar-mmusic-rtcweb-grouping-00
draft-nandakumar-rtcweb-sdp-02

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Hi Christian,

How about this text:

    Clients acting on behalf of end users are expected to set one or both
    'humintlang-send' and 'humintlang-recv' attributes on each media
    stream primarily intended for human communication in an offer when
    placing an outgoing session, but either ignore or take into
    consideration the attributes when receiving incoming calls, based on
    local configuration and capabilities.  Systems acting on behalf of
    call centers and PSAPs are expected to take into account the values
    when processing inbound calls.


At 12:11 PM +1000 7/15/13, Christian Groves wrote:

>  Hello Randall and Gunnar,
>
>  I think Randall's clarification would be good to have in the draft. 
> I think that when a caller support the language it is sent for 
> every call. Whether the language is used by a receiver depends on 
> the receiver's local capability.
>
>
>  Regards, Christian
>
>
>  On 14/07/2013 4:04 AM, Gunnar Hellstrom wrote:
>>  On 2013-07-13 10:07, Randall Gellens wrote:
>>>  At 2:49 PM +1000 7/10/13, Christian Groves wrote:
>>>
>>>>   I wouldn't have thought that it would be use case specific... 
>>>> i.e. i'd only include my language when I called an emergency 
>>>> service. I would think an endpoint would be configured with the 
>>>> information and then provide it whenever I made a call. Is that 
>>>> the intent?
>>>
>>>  I expect so, but then, I also didn't expect it to be used much in 
>>> the person-to-person case since usually language is implied by 
>>> context when calling an individual.  I expect it to be used when 
>>> a person is calling an call center (corporate or emergency).  To 
>>> put it another way, I expect a personal client to always set 
>>> language on outbound calls, but in most cases to ignore it on 
>>> inbound calls.  I'd expect a corporate call center, as well as 
>>> emergency call handling facilities, to pay attention.  I should 
>>> make this more clear in the draft.
>>>
>>  I expect the setting to be retrieved from a personal profile 
>> (likely but not necessarily stored in connection with other user 
>> information and account information) and used in every call.
>>
>>  /Gunnar
>>  _______________________________________________
>>  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


-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly selected tag: ---------------
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To: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>, <mmusic@ietf.org>
From: Randall Gellens <randy@qti.qualcomm.com>
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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-01.txt - more use cases
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<!doctype html public "-//W3C//DTD W3 HTML//EN">
<html><head><style type=3D"text/css"><!--
blockquote, dl, ul, ol, li { padding-top: 0 ; padding-bottom: 0 }
 --></style><title>Re: [MMUSIC]
draft-gellens-mmusic-negotiating-human-langua</title></head><body>
<div>Hi Gunnar,</div>
<div><br></div>
<div>I am concerned that getting into person-to-person usage will
overly complicate the draft with no real-world benefit, since
person-to-person communication has context and implied language/media,
as stated in the draft.&nbsp; The draft is trying to solve a more
focused real-world problem where there is no context.</div>
<div><br></div>
<div>At 8:40 AM +0200 7/15/13, Gunnar Hellstrom wrote:</div>
<div><br></div>
<blockquote type=3D"cite" cite>I suggest to add a couple of use
cases.<br>
</blockquote>
<blockquote type=3D"cite" cite><tt>4.x.&nbsp; Deaf French sign language
user calling another deaf French sign language user.<br>
<br>
Someone who mainly uses French sign language (FSL) places a
call.&nbsp;<br>
The call offers a video stream listing FSL in both directions with
highest preference and also<br>
real-time text stream on low preference level for French in both
directions.<br>
Audio is also included but without language indications.<br>
The call destination turns out to be another user of FSL with the same
media and language settings.<br>
The called client can find that the language preferences match, so the
same language indications are<br>
sent in the SDP answer, and the call can be connected without
connection of any relay service.<br>
The users can sign directly with each other in the video streams, and
occasionally use the real-time<br>
text streams for data that need exact spelling or noted.<br>
<br>
<br>
4.y.&nbsp; Spoken Japanese user calling a Spanish speaking person.<br>
<br>
Someone who speaks Japanese and is subscriber of an on-line
interpreting service places a call.&nbsp;<br>
The call offers audio streams using Japanese in both directions on
high preference level, video<br>
with spoken Japanese with no specific preference and both message-wise
and real-time text in<br>
Japanese in both directions.<br>
This user is customer of an on-line interpreting service. The service
adds its interpreting<br>
language capabilities to the SDP on lower level than the originator,
and conveys the call to the destination.<br>
The destination is a Spanish speaking person. The called terminal
detects that Spanish is among the possible<br>
languages even if not highest preference, so that call is answered
with indication of spoken Spanish at high<br>
priority level and Spanish text messaging in both directions at low
priority level.<br>
The on-line interpreting service detects that their translation
between Japanese and Spanish is needed to<br>
make the call work, so they allocate an interpreter to the call, and
answers the call with an indication<br>
of Japanese towards the caller.<br>
Information that the call will be soon supported by a Japanese/Spanish
interpreter is provided in the<br>
appropriate languages in the text message connections.<br>
The users can communicate in their own languages supported by the
interpreter.<br>
<br>
<br>
-------------------------------<br>
More might be needed to clarify some more of the endless possible
variations and benefit.<br>
<br>
<br>
</tt></blockquote>
<blockquote type=3D"cite" cite><br>
<br>
</blockquote>
<blockquote type=3D"cite" cite>
<hr></blockquote>
<blockquote type=3D"cite" cite>Gunnar Hellstr=9Am<br>
Omnitor<br>
<a
href=3D"mailto:gunnar.hellstrom@omnitor.se">gunnar.hellstrom@omnitor.se</a
><br>
+46708204288<br>
</blockquote>
<blockquote type=3D"cite" cite>On 2013-07-15 04:11, Christian Groves
wrote:<br>
<blockquote type=3D"cite" cite>Hello Randall and Gunnar,<br>
<br>
I think Randall's clarification would be good to have in the draft. I
think that when a caller support the language it is sent for every
call. Whether the language is used by a receiver depends on the
receiver's local capability.<br>
<br>
<br>
Regards, Christian<br>
<br>
<br>
On 14/07/2013 4:04 AM, Gunnar Hellstrom wrote:<br>
<blockquote type=3D"cite" cite>On 2013-07-13 10:07, Randall Gellens
wrote:<br>
<blockquote type=3D"cite" cite>At 2:49 PM +1000 7/10/13, Christian
Groves wrote:<br>
<blockquote type=3D"cite" cite>&nbsp;I wouldn't have thought that it
would be use case specific... i.e. i'd only include my language when I
called an emergency service. I would think an endpoint would be
configured with the information and then provide it whenever I made a
call. Is that the intent?<br>
</blockquote>
</blockquote>
<blockquote type=3D"cite" cite><br>
I expect so, but then, I also didn't expect it to be used much in the
person-to-person case since usually language is implied by context
when calling an individual.&nbsp; I expect it to be used when a person
is calling an call center (corporate or emergency).&nbsp; To put it
another way, I expect a personal client to always set language on
outbound calls, but in most cases to ignore it on inbound calls.&nbsp;
I'd expect a corporate call center, as well as emergency call handling
facilities, to pay attention.&nbsp; I should make this more clear in
the draft.</blockquote>
<blockquote type=3D"cite" cite><br></blockquote>
</blockquote>
<blockquote type=3D"cite" cite>I expect the setting to be retrieved from
a personal profile (likely but not necessarily stored in connection
with other user information and account information) and used in every
call.<br>
<br>
/Gunnar<br>
_______________________________________________<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.org/mailman/listinfo/mmusic</a><br>
</blockquote>
</blockquote>
<blockquote type=3D"cite" cite><br>
_______________________________________________<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.org/mailman/listinfo/mmusic</a><br>
</blockquote>
</blockquote>
<blockquote type=3D"cite" cite><br></blockquote>
<blockquote type=3D"cite" cite><br>
_______________________________________________<br>
mmusic mailing list<br>
mmusic@ietf.org<br>
https://www.ietf.org/mailman/listinfo/mmusic</blockquote>
<div><br></div>
<div><br></div>
<x-sigsep><pre>-- 
</pre></x-sigsep>
<div><font color=3D"#000000">Randall Gellens<br>
Opinions are personal;&nbsp;&nbsp;&nbsp; facts are
suspect;&nbsp;&nbsp;&nbsp; I speak for myself only<br>
-------------- Randomly selected tag: ---------------<br>
If you make people think they're thinking, they'll love you; but if you<br>
really make them think they'll hate you.<br>
</font></div></body>
</html>

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At 12:20 PM +1000 7/15/13, Christian Groves wrote:

>  Hello Randall,
>
>  Please see my comment below.
>
>  ..snip..
>  On 15/07/2013 8:14 AM, Gunnar Hellstrom wrote:
>>  ...
>>
>>  2. You have also introduced this paragraph in 7.1:
>>  "
>>  Clients acting on behalf of end users are expected to set one or both
>>      'humintlang-send' and 'humintlang-recv' attributes on each media
>>      stream in an offer when placing an outgoing session but ignore the
>>      attributes when receiving incoming calls.  Systems acting on behalf
>>      of call centers and PSAPs are expected to take into account the
>>      values when processing inbound calls.
>>  "
>>
>>  What is the thought behind this?
>>
>>  I see the humintlang specification equally valuable in outgoing as 
>> incoming calls, and for end-users as well as call-centers and 
>> PSAPs.
>>  If a user gets an incoming call, with humintlang specifications, 
>> it can compare with its own profile and decide to invoke a
>>  relay service or interpreter servicde to cover some differences in 
>> language preferences, or just prepare for what media, modalities 
>> and
>>  languages to use in the call..
>>  I do not see the background for this paragraph, and think it will 
>> be better to delete it.
>>
>>
>  [CNG] I guess this is a question of scope. I would think that the 
> draft would just describe the syntax and procedures of this new 
> language tag in a generic manner. The application level behaviour 
> of call centres and PSAPs would be defined elsewhere?

[RG] That would be fine with me; my attempt in adding the text using 
descriptive rather than normative language was to clarify the 
intended use without imposing mandates on call centers and PSAPs 
(which, as you say, will be done elsewhere).

-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly selected tag: ---------------
Mythology, n.:
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origin, early history, heroes, deities and so forth, as
distinguished from the true accounts which it invents later.
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From: Martin Thomson <martin.thomson@gmail.com>
To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [MMUSIC] Review of unified-plan-00
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On 18 July 2013 02:40, Harald Alvestrand <harald@alvestrand.no> wrote:
>    The standard way of encoding this information in SDP is to have each
>    RTP flow (i.e., SSRC) appear on its own m-line.
>
> This is not strictly true, since the proposal continues to use each m-line
> for up to two flows, one in each direction, and some of the . Suggested
> rephrase:
>
>   One common way of encoding this information uses each M-line to
>   describe one media source in each direction.

That's fine with me.  It seems that there is some contention around
what an m-line actually is, and this has no substantive impact.  So
weasel away.

> It needs to note that this specification modifies RFC 5888 section 9.2

Yes.  We've talked about that lots, but we missed it somehow.

> 4.1 Simple example
>
> "It also shows unique payload across the audio and video m=lines for the
> Answerer that does not support BUNDLE semantics." - this doesn't make sense
> to me; unique payloads are needed for answerers that support BUNDLE
> semantics. Copy/paste error?
>
> Would be consistent if it said "for the Answerer that supports BUNDLE
> semantics".

Actually, this has nothing to do with BUNDLE.  It assumes that you
support BUNDLE, but might not support the RTP header extension and
might want to have media play out at the offerer prior to receiving an
answer.

> 4.4 Multiple Video with Simulcast
>
> If it's intended that draft-westerlund-avtcore-rtp-simulcast is used, the
> ssrc-group lines should probably say SCS rather than SIMULCAST.

Maybe this is a veiled comment along the lines of : "SCS" is a
strangely perverse choice of label, why don't you call a spade a spade
:)

> Interaction with a=content
>
> Previously, we had discussed having a "content" attribute on
> MediaStreamTracks that would make a track link to an m-line with the same
> "a=content" attribute. If this is still expected to work, it might be worthy
> of inclusion here.

The purpose to which a=content was put in plan B isn't really
necessary any more.  No actual feature would depend on this, so don't
see a particular reason to mention this in the plan.

Maybe this is one of the SDP mutations we've been looking for.
Perhaps you can take this to the W3C.

From gunnar.hellstrom@omnitor.se  Thu Jul 18 13:24:22 2013
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Date: Thu, 18 Jul 2013 22:23:35 +0200
From: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
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To: Randall Gellens <randy@qti.qualcomm.com>
References: <p0624061acd691d9b38bc@dhcp-42ec.meeting.ietf.org> <B0295A41-BF2B-4520-8495-7A88B4803F30@acmepacket.com> <5144E4E5.4030801@omnitor.se> <03559DEF-E442-4118-A616-8D7E42D1F22A@acmepacket.com> <5145729C.6030905@omnitor.se> <514FE5A9.3060900@nteczone.com> <p06240615cdfea836f364@[99.111.97.136]> <51DBB813.4060607@nteczone.com> <p0624060ece02360a322d@[99.111.97.136]>	<51DC9B66.6090302@omnitor.se> <51DCE7CB.7080205@nteczone.com> <p06240603ce06ba958db4@[10.121.65.51]>	<51E196B0.5080406@omnitor.se> <51E35A53.5030502@nteczone.com> <51E39957.3080004@omnitor.se> <p06240604ce0dc6237036@[99.111.97.136]>
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-01.txt - more use cases
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Randall,
In my view the draft aims at solving the difference between the 
user-to-user call with known similar context and the situation when the 
user-to-user call has dissimilar contexts and need support.

The call center gets invoked because of the difference between the users 
in order to fill the gap.

Of course there are other situations where it can be useful, but this is 
the main application that I see.

In USA, it is called to dial-around, when the users are connected 
without connection of an assisting service, and use of relay service 
when assistance is needed and invoked.
In Sweden we call the concept Call-direct.
The function is mentioned and demonstrated a number of times 
Internationally, and this is an opportunity to get proper standardized 
support for it.

Users interested in this service will have the language settings. It is 
very valid to do the comparison between the users and invoke services if 
needed.
For users who have not set the language indicators, there will need to 
be a place for assumptions in the evaluation logic.


And. Most important, I think we should follow Christian's advice to get 
a way to include language indications specified here, and the ways to 
use it specified in other specifications.

But in order to get the indications useful, we must settle some use 
cases. So there we have a risk of including assumptions of how the 
indications can be used.

Gunnar






------------------------------------------------------------------------
Gunnar Hellström
Omnitor
gunnar.hellstrom@omnitor.se
+46708204288
On 2013-07-18 18:21, Randall Gellens wrote:
> Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-langua
> Hi Gunnar,
>
> I am concerned that getting into person-to-person usage will overly 
> complicate the draft with no real-world benefit, since 
> person-to-person communication has context and implied language/media, 
> as stated in the draft.  The draft is trying to solve a more focused 
> real-world problem where there is no context.
>
> At 8:40 AM +0200 7/15/13, Gunnar Hellstrom wrote:
>
>> I suggest to add a couple of use cases.
>> 4.x.  Deaf French sign language user calling another deaf French sign 
>> language user.
>>
>> Someone who mainly uses French sign language (FSL) places a call.
>> The call offers a video stream listing FSL in both directions with 
>> highest preference and also
>> real-time text stream on low preference level for French in both 
>> directions.
>> Audio is also included but without language indications.
>> The call destination turns out to be another user of FSL with the 
>> same media and language settings.
>> The called client can find that the language preferences match, so 
>> the same language indications are
>> sent in the SDP answer, and the call can be connected without 
>> connection of any relay service.
>> The users can sign directly with each other in the video streams, and 
>> occasionally use the real-time
>> text streams for data that need exact spelling or noted.
>>
>>
>> 4.y.  Spoken Japanese user calling a Spanish speaking person.
>>
>> Someone who speaks Japanese and is subscriber of an on-line 
>> interpreting service places a call.
>> The call offers audio streams using Japanese in both directions on 
>> high preference level, video
>> with spoken Japanese with no specific preference and both 
>> message-wise and real-time text in
>> Japanese in both directions.
>> This user is customer of an on-line interpreting service. The service 
>> adds its interpreting
>> language capabilities to the SDP on lower level than the originator, 
>> and conveys the call to the destination.
>> The destination is a Spanish speaking person. The called terminal 
>> detects that Spanish is among the possible
>> languages even if not highest preference, so that call is answered 
>> with indication of spoken Spanish at high
>> priority level and Spanish text messaging in both directions at low 
>> priority level.
>> The on-line interpreting service detects that their translation 
>> between Japanese and Spanish is needed to
>> make the call work, so they allocate an interpreter to the call, and 
>> answers the call with an indication
>> of Japanese towards the caller.
>> Information that the call will be soon supported by a 
>> Japanese/Spanish interpreter is provided in the
>> appropriate languages in the text message connections.
>> The users can communicate in their own languages supported by the 
>> interpreter.
>>
>>
>> -------------------------------
>> More might be needed to clarify some more of the endless possible 
>> variations and benefit.
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>> Gunnar Hellstrs(m
>> Omnitor
>> gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se>
>> +46708204288
>> On 2013-07-15 04:11, Christian Groves wrote:
>>> Hello Randall and Gunnar,
>>>
>>> I think Randall's clarification would be good to have in the draft. 
>>> I think that when a caller support the language it is sent for every 
>>> call. Whether the language is used by a receiver depends on the 
>>> receiver's local capability.
>>>
>>>
>>> Regards, Christian
>>>
>>>
>>> On 14/07/2013 4:04 AM, Gunnar Hellstrom wrote:
>>>> On 2013-07-13 10:07, Randall Gellens wrote:
>>>>> At 2:49 PM +1000 7/10/13, Christian Groves wrote:
>>>>>>  I wouldn't have thought that it would be use case specific... 
>>>>>> i.e. i'd only include my language when I called an emergency 
>>>>>> service. I would think an endpoint would be configured with the 
>>>>>> information and then provide it whenever I made a call. Is that 
>>>>>> the intent?
>>>>>
>>>>> I expect so, but then, I also didn't expect it to be used much in 
>>>>> the person-to-person case since usually language is implied by 
>>>>> context when calling an individual.  I expect it to be used when a 
>>>>> person is calling an call center (corporate or emergency).  To put 
>>>>> it another way, I expect a personal client to always set language 
>>>>> on outbound calls, but in most cases to ignore it on inbound 
>>>>> calls. I'd expect a corporate call center, as well as emergency 
>>>>> call handling facilities, to pay attention.  I should make this 
>>>>> more clear in the draft.
>>>>>
>>>> I expect the setting to be retrieved from a personal profile 
>>>> (likely but not necessarily stored in connection with other user 
>>>> information and account information) and used in every call.
>>>>
>>>> /Gunnar
>>>> _______________________________________________
>>>> 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
>> https://www.ietf.org/mailman/listinfo/mmusic
>
>
> -- 
> Randall Gellens
> Opinions are personal;    facts are suspect;    I speak for myself only
> -------------- Randomly selected tag: ---------------
> If you make people think they're thinking, they'll love you; but if you
> really make them think they'll hate you.


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    <div class="moz-cite-prefix">Randall,<br>
      In my view the draft aims at solving the difference between the
      user-to-user call with known similar context and the situation
      when the user-to-user call has dissimilar contexts and need
      support.<br>
      <br>
      The call center gets invoked because of the difference between the
      users in order to fill the gap.<br>
      <br>
      Of course there are other situations where it can be useful, but
      this is the main application that I see.<br>
      <br>
      In USA, it is called to dial-around, when the users are connected
      without connection of an assisting service, and use of relay
      service when assistance is needed and invoked.<br>
      In Sweden we call the concept Call-direct. <br>
      The function is mentioned and demonstrated a number of times
      Internationally, and this is an opportunity to get proper
      standardized support for it.<br>
      <br>
      Users interested in this service will have the language settings.
      It is very valid to do the comparison between the users and invoke
      services if needed. <br>
      For users who have not set the language indicators, there will
      need to be a place for assumptions in the evaluation logic.<br>
      <br>
      <br>
      And. Most important, I think we should follow Christian's advice
      to get a way to include language indications specified here, and
      the ways to use it specified in other specifications.<br>
      <br>
      But in order to get the indications useful, we must settle some
      use cases. So there we have a risk of including assumptions of how
      the indications can be used. <br>
      <br>
      Gunnar<br>
      <br>
      &nbsp;<br>
      <br>
      <br>
      <br>
      <br>
      <div class="moz-signature">
        <hr>
        Gunnar Hellstr&ouml;m<br>
        Omnitor<br>
        <a class="moz-txt-link-abbreviated" href="mailto:gunnar.hellstrom@omnitor.se">gunnar.hellstrom@omnitor.se</a><br>
        +46708204288<br>
      </div>
      On 2013-07-18 18:21, Randall Gellens wrote:<br>
    </div>
    <blockquote cite="mid:p06240604ce0dc6237036@%5B99.111.97.136%5D"
      type="cite">
      <style type="text/css"><!--
blockquote, dl, ul, ol, li { padding-top: 0 ; padding-bottom: 0 }
 --></style>
      <title>Re: [MMUSIC]
        draft-gellens-mmusic-negotiating-human-langua</title>
      <div>Hi Gunnar,</div>
      <div><br>
      </div>
      <div>I am concerned that getting into person-to-person usage will
        overly complicate the draft with no real-world benefit, since
        person-to-person communication has context and implied
        language/media,
        as stated in the draft.&nbsp; The draft is trying to solve a more
        focused real-world problem where there is no context.</div>
      <div><br>
      </div>
      <div>At 8:40 AM +0200 7/15/13, Gunnar Hellstrom wrote:</div>
      <div><br>
      </div>
      <blockquote type="cite" cite="">I suggest to add a couple of use
        cases.<br>
      </blockquote>
      <blockquote type="cite" cite=""><tt>4.x.&nbsp; Deaf French sign
          language
          user calling another deaf French sign language user.<br>
          <br>
          Someone who mainly uses French sign language (FSL) places a
          call.&nbsp;<br>
          The call offers a video stream listing FSL in both directions
          with
          highest preference and also<br>
          real-time text stream on low preference level for French in
          both
          directions.<br>
          Audio is also included but without language indications.<br>
          The call destination turns out to be another user of FSL with
          the same
          media and language settings.<br>
          The called client can find that the language preferences
          match, so the
          same language indications are<br>
          sent in the SDP answer, and the call can be connected without
          connection of any relay service.<br>
          The users can sign directly with each other in the video
          streams, and
          occasionally use the real-time<br>
          text streams for data that need exact spelling or noted.<br>
          <br>
          <br>
          4.y.&nbsp; Spoken Japanese user calling a Spanish speaking person.<br>
          <br>
          Someone who speaks Japanese and is subscriber of an on-line
          interpreting service places a call.&nbsp;<br>
          The call offers audio streams using Japanese in both
          directions on
          high preference level, video<br>
          with spoken Japanese with no specific preference and both
          message-wise
          and real-time text in<br>
          Japanese in both directions.<br>
          This user is customer of an on-line interpreting service. The
          service
          adds its interpreting<br>
          language capabilities to the SDP on lower level than the
          originator,
          and conveys the call to the destination.<br>
          The destination is a Spanish speaking person. The called
          terminal
          detects that Spanish is among the possible<br>
          languages even if not highest preference, so that call is
          answered
          with indication of spoken Spanish at high<br>
          priority level and Spanish text messaging in both directions
          at low
          priority level.<br>
          The on-line interpreting service detects that their
          translation
          between Japanese and Spanish is needed to<br>
          make the call work, so they allocate an interpreter to the
          call, and
          answers the call with an indication<br>
          of Japanese towards the caller.<br>
          Information that the call will be soon supported by a
          Japanese/Spanish
          interpreter is provided in the<br>
          appropriate languages in the text message connections.<br>
          The users can communicate in their own languages supported by
          the
          interpreter.<br>
          <br>
          <br>
          -------------------------------<br>
          More might be needed to clarify some more of the endless
          possible
          variations and benefit.<br>
          <br>
          <br>
        </tt></blockquote>
      <blockquote type="cite" cite=""><br>
        <br>
      </blockquote>
      <blockquote type="cite" cite="">
        <hr></blockquote>
      <blockquote type="cite" cite="">Gunnar Hellstr&#353;m<br>
        Omnitor<br>
        <a moz-do-not-send="true"
          href="mailto:gunnar.hellstrom@omnitor.se">gunnar.hellstrom@omnitor.se</a><br>
        +46708204288<br>
      </blockquote>
      <blockquote type="cite" cite="">On 2013-07-15 04:11, Christian
        Groves
        wrote:<br>
        <blockquote type="cite" cite="">Hello Randall and Gunnar,<br>
          <br>
          I think Randall's clarification would be good to have in the
          draft. I
          think that when a caller support the language it is sent for
          every
          call. Whether the language is used by a receiver depends on
          the
          receiver's local capability.<br>
          <br>
          <br>
          Regards, Christian<br>
          <br>
          <br>
          On 14/07/2013 4:04 AM, Gunnar Hellstrom wrote:<br>
          <blockquote type="cite" cite="">On 2013-07-13 10:07, Randall
            Gellens
            wrote:<br>
            <blockquote type="cite" cite="">At 2:49 PM +1000 7/10/13,
              Christian
              Groves wrote:<br>
              <blockquote type="cite" cite="">&nbsp;I wouldn't have thought
                that it
                would be use case specific... i.e. i'd only include my
                language when I
                called an emergency service. I would think an endpoint
                would be
                configured with the information and then provide it
                whenever I made a
                call. Is that the intent?<br>
              </blockquote>
            </blockquote>
            <blockquote type="cite" cite=""><br>
              I expect so, but then, I also didn't expect it to be used
              much in the
              person-to-person case since usually language is implied by
              context
              when calling an individual.&nbsp; I expect it to be used when a
              person
              is calling an call center (corporate or emergency).&nbsp; To
              put it
              another way, I expect a personal client to always set
              language on
              outbound calls, but in most cases to ignore it on inbound
              calls.&nbsp;
              I'd expect a corporate call center, as well as emergency
              call handling
              facilities, to pay attention.&nbsp; I should make this more
              clear in
              the draft.</blockquote>
            <blockquote type="cite" cite=""><br>
            </blockquote>
          </blockquote>
          <blockquote type="cite" cite="">I expect the setting to be
            retrieved from
            a personal profile (likely but not necessarily stored in
            connection
            with other user information and account information) and
            used in every
            call.<br>
            <br>
            /Gunnar<br>
            _______________________________________________<br>
            mmusic mailing list<br>
            <a moz-do-not-send="true" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
            <a moz-do-not-send="true"
              href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
          </blockquote>
        </blockquote>
        <blockquote type="cite" cite=""><br>
          _______________________________________________<br>
          mmusic mailing list<br>
          <a moz-do-not-send="true" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
          <a moz-do-not-send="true"
            href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
        </blockquote>
      </blockquote>
      <blockquote type="cite" cite=""><br>
      </blockquote>
      <blockquote type="cite" cite=""><br>
        _______________________________________________<br>
        mmusic mailing list<br>
        <a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
        <a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a></blockquote>
      <div><br>
      </div>
      <div><br>
      </div>
      <x-sigsep>
        <pre>-- 
</pre>
      </x-sigsep>
      <div><font color="#000000">Randall Gellens<br>
          Opinions are personal;&nbsp;&nbsp;&nbsp; facts are
          suspect;&nbsp;&nbsp;&nbsp; I speak for myself only<br>
          -------------- Randomly selected tag: ---------------<br>
          If you make people think they're thinking, they'll love you;
          but if you<br>
          really make them think they'll hate you.<br>
        </font></div>
    </blockquote>
    <br>
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From harald@alvestrand.no  Thu Jul 18 14:04:37 2013
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Subject: Re: [MMUSIC] Review of unified-plan-00
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On 07/18/2013 07:06 PM, Martin Thomson wrote:
> On 18 July 2013 02:40, Harald Alvestrand <harald@alvestrand.no> wrote:
>>    The standard way of encoding this information in SDP is to have each
>>    RTP flow (i.e., SSRC) appear on its own m-line.
>>
>> This is not strictly true, since the proposal continues to use each m-line
>> for up to two flows, one in each direction, and some of the . Suggested
>> rephrase:
>>
>>   One common way of encoding this information uses each M-line to
>>   describe one media source in each direction.
> That's fine with me.  It seems that there is some contention around
> what an m-line actually is, and this has no substantive impact.  So
> weasel away.
>
>> It needs to note that this specification modifies RFC 5888 section 9.2
> Yes.  We've talked about that lots, but we missed it somehow.
>
>> 4.1 Simple example
>>
>> "It also shows unique payload across the audio and video m=lines for the
>> Answerer that does not support BUNDLE semantics." - this doesn't make sense
>> to me; unique payloads are needed for answerers that support BUNDLE
>> semantics. Copy/paste error?
>>
>> Would be consistent if it said "for the Answerer that supports BUNDLE
>> semantics".
> Actually, this has nothing to do with BUNDLE.  It assumes that you
> support BUNDLE, but might not support the RTP header extension and
> might want to have media play out at the offerer prior to receiving an
> answer.

Since all the sentences talking about it have the word BUNDLE in them, I
have to assume it has *something* to do with BUNDLE. It's just that the
sign bit seemed wrong.

>
>> 4.4 Multiple Video with Simulcast
>>
>> If it's intended that draft-westerlund-avtcore-rtp-simulcast is used, the
>> ssrc-group lines should probably say SCS rather than SIMULCAST.
> Maybe this is a veiled comment along the lines of : "SCS" is a
> strangely perverse choice of label, why don't you call a spade a spade
> :)

I'd be happy to argue for use of a more readable label :-)

>> Interaction with a=content
>>
>> Previously, we had discussed having a "content" attribute on
>> MediaStreamTracks that would make a track link to an m-line with the same
>> "a=content" attribute. If this is still expected to work, it might be worthy
>> of inclusion here.
> The purpose to which a=content was put in plan B isn't really
> necessary any more.  No actual feature would depend on this, so don't
> see a particular reason to mention this in the plan.
>
> Maybe this is one of the SDP mutations we've been looking for.
> Perhaps you can take this to the W3C.

Cullen claimed at one point that a=content was necessary to make
interoperability with some class of devices happen - it may be totally
orthogonal to this specification, in which case I'm happy to discus that
on a different thread :-)



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Subject: Re: [MMUSIC] Review of unified-plan-00
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On 18 July 2013 14:04, Harald Alvestrand <harald@alvestrand.no> wrote:
>>> 4.1 Simple example
>>>
>>> "It also shows unique payload across the audio and video m=lines for the
>>> Answerer that does not support BUNDLE semantics." - this doesn't make sense
>>> to me; unique payloads are needed for answerers that support BUNDLE
>>> semantics. Copy/paste error?
>>>
>>> Would be consistent if it said "for the Answerer that supports BUNDLE
>>> semantics".
>> Actually, this has nothing to do with BUNDLE.  It assumes that you
>> support BUNDLE, but might not support the RTP header extension and
>> might want to have media play out at the offerer prior to receiving an
>> answer.
>
> Since all the sentences talking about it have the word BUNDLE in them, I
> have to assume it has *something* to do with BUNDLE. It's just that the
> sign bit seemed wrong.

There is an initial assumption given (one stream per m-line).

With that assumption stated, a unique payload type is a way to do the
correlation thing.  That only causes problems when BUNDLE is used.
Without BUNDLE, you have a 5-tuple, which is probably adequate.

I'll have to let Adam speak to why the draft says what it says here,
he was following the discussion closer than I.

From gunnar.hellstrom@omnitor.se  Thu Jul 18 14:19:55 2013
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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-01.txt
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On 2013-07-18 18:16, Randall Gellens wrote:
>>>  2. You have also introduced this paragraph in 7.1:
>>>  "
>>>  Clients acting on behalf of end users are expected to set one or both
>>>      'humintlang-send' and 'humintlang-recv' attributes on each media
>>>      stream in an offer when placing an outgoing session but ignore the
>>>      attributes when receiving incoming calls.  Systems acting on 
>>> behalf
>>>      of call centers and PSAPs are expected to take into account the
>>>      values when processing inbound calls.
>>>  "
>>>
>>>  What is the thought behind this?
>>>
>>>  I see the humintlang specification equally valuable in outgoing as 
>>> incoming calls, and for end-users as well as call-centers and PSAPs.
>>>  If a user gets an incoming call, with humintlang specifications, it 
>>> can compare with its own profile and decide to invoke a
>>>  relay service or interpreter servicde to cover some differences in 
>>> language preferences, or just prepare for what media, modalities and
>>>  languages to use in the call..
>>>  I do not see the background for this paragraph, and think it will 
>>> be better to delete it.
>>>
>>>
>>  [CNG] I guess this is a question of scope. I would think that the 
>> draft would just describe the syntax and procedures of this new 
>> language tag in a generic manner. The application level behaviour of 
>> call centres and PSAPs would be defined elsewhere?
>
> [RG] That would be fine with me; my attempt in adding the text using 
> descriptive rather than normative language was to clarify the intended 
> use without imposing mandates on call centers and PSAPs (which, as you 
> say, will be done elsewhere).
But since your assumption of use was far from mine, I hope that you 
accept to delete the paragraph.


Or change it to:

  "
  Clients are expected to set one or both
      'humintlang-send' and 'humintlang-recv' attributes on each media
      stream in an offer and answer for which the users has any 
preferences or capabilities for use in human communication. Systems are 
expected to take into account the
      values when processing inbound calls or when processing the 
answers on outbound calls. The values can also be used for call handling 
decisions by service providers.
  "


Regards

Gunnar


From gunnar.hellstrom@omnitor.se  Thu Jul 18 14:28:34 2013
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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-01.txt - more use cases
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On 2013-07-18 18:21, Randall Gellens wrote:
> Hi Gunnar,
>
> I am concerned that getting into person-to-person usage will overly 
> complicate the draft with no real-world benefit, since 
> person-to-person communication has context and implied language/media, 
> as stated in the draft.
This is not true anymore in the multimedia communication area. Video may 
be included in a call because the user just want to see the other call 
participant or it is included because the user urgently prefers to get a 
sign language commuication.

Two persons urgently wanting to use British Sign Language should have a 
chance to find that both support and prefer that, and they should 
connect without a relay service.
The evaluation can be made by the answering part, the calling part on 
the answer if the intication is prpoerly inserted, or by a service 
provider in betweeen the users.

But if the answering part ignores the indications, nothing is gained.

Gunnar



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On Thu, Jul 18, 2013 at 2:40 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

> First of all, let me say that I'm very happy to see a plan that the people
> who have been voicing the strongest opinions in this debate are willing to
> stand behind.
> I believe, extremely strongly, that it is better to move forward on one
> path, no matter what it is (as long as it's not obviously broken), than to
> continue to debate. This topic has long been needing a decision.
>
> Thus, I stand behind the technology proposed, and have no wish to change
> it. The purpose of this review is to help make sure it's fully described,
> and that all parties understand it the same way.
>
> I have ignored language-only issues at this time.
>
> Introduction, paragraph 2:
>
>    The standard way of encoding this information in SDP is to have each
>    RTP flow (i.e., SSRC) appear on its own m-line.
>
> This is not strictly true, since the proposal continues to use each m-line
> for up to two flows, one in each direction, and some of the . Suggested
> rephrase:
>
>   One common way of encoding this information uses each M-line to
>   describe one media source in each direction.
>
> "media source" is as defined in draft-westerlund-avtcore-**multiplex-architecture-03
> section 2.1. The term is used elsewhere in the document, and seems to have
> this meaning. "One common" is a reflection of my belief that practice has
> not been uniform in this area.
>
> 3.1, Bundle-only M-lines
>
> This mechanism requires sending M-lines with port zero as part of a bundle
> in an SDP OFFER.
> It needs to note that this specification modifies RFC 5888 section 9.2
> (the grouping framework) by allowing such M-lines to be part of a group in
> an offer (but I do not think it needs to be allowed in an answer).
>
> 4.1 Simple example
>
> "It also shows unique payload across the audio and video m=lines for the
> Answerer that does not support BUNDLE semantics." - this doesn't make sense
> to me; unique payloads are needed for answerers that support BUNDLE
> semantics. Copy/paste error?
>
> Would be consistent if it said "for the Answerer that supports BUNDLE
> semantics".
>
> 4.4 Multiple Video with Simulcast
>
> If it's intended that draft-westerlund-avtcore-rtp-**simulcast is used,
> the ssrc-group lines should probably say SCS rather than SIMULCAST.
>
>

Regarding the Simulcast ssrc-group based signaling, the following draft
explains its usage:
http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-00.

The idea is very similar to the Unified Plan in that , ssrc-group is used
to signal the ssrcs for a given MediaStreamTrack as part of Simulcast. It
also talks about using receiver selection of the Simulcast sources offered
by the Sender via
draft-lennox-mmusic-sdp-source-selection-05<http://tools.ietf.org/html/draft-lennox-mmusic-sdp-source-selection-05>
.



>
> And now for some stuff that isn't in the document, but might want to be:
>
> More examples
>
> There should probably be one example where the offer contains 1 audio and
> 2 video streams, and the answer contains 1 audio and 1 video stream (fewer
> streams in response than in request); I believe that this answer would then
> contain "a=recvonly" and no "a=msid" attribute on the 3rd m-line, but it's
> nice to be sure.
>
> Interaction with a=content
>
> Previously, we had discussed having a "content" attribute on
> MediaStreamTracks that would make a track link to an m-line with the same
> "a=content" attribute. If this is still expected to work, it might be
> worthy of inclusion here.
>
> I think that's all for this round of review. Thanks for getting it out!
>
>             Harald
>
>
>
>
> ______________________________**_________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/**listinfo/mmusic<https://www.ietf.org/mailman/listinfo/mmusic>
>

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<br><br><div class=3D"gmail_quote">On Thu, Jul 18, 2013 at 2:40 AM, Harald =
Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" ta=
rget=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">
First of all, let me say that I&#39;m very happy to see a plan that the peo=
ple who have been voicing the strongest opinions in this debate are willing=
 to stand behind.<br>
I believe, extremely strongly, that it is better to move forward on one pat=
h, no matter what it is (as long as it&#39;s not obviously broken), than to=
 continue to debate. This topic has long been needing a decision.<br>
<br>
Thus, I stand behind the technology proposed, and have no wish to change it=
. The purpose of this review is to help make sure it&#39;s fully described,=
 and that all parties understand it the same way.<br>
<br>
I have ignored language-only issues at this time.<br>
<br>
Introduction, paragraph 2:<br>
<br>
=A0 =A0The standard way of encoding this information in SDP is to have each=
<br>
=A0 =A0RTP flow (i.e., SSRC) appear on its own m-line.<br>
<br>
This is not strictly true, since the proposal continues to use each m-line =
for up to two flows, one in each direction, and some of the . Suggested rep=
hrase:<br>
<br>
=A0 One common way of encoding this information uses each M-line to<br>
=A0 describe one media source in each direction.<br>
<br>
&quot;media source&quot; is as defined in draft-westerlund-avtcore-<u></u>m=
ultiplex-architecture-03 section 2.1. The term is used elsewhere in the doc=
ument, and seems to have this meaning. &quot;One common&quot; is a reflecti=
on of my belief that practice has not been uniform in this area.<br>

<br>
3.1, Bundle-only M-lines<br>
<br>
This mechanism requires sending M-lines with port zero as part of a bundle =
in an SDP OFFER.<br>
It needs to note that this specification modifies RFC 5888 section 9.2 (the=
 grouping framework) by allowing such M-lines to be part of a group in an o=
ffer (but I do not think it needs to be allowed in an answer).<br>
<br>
4.1 Simple example<br>
<br>
&quot;It also shows unique payload across the audio and video m=3Dlines for=
 the Answerer that does not support BUNDLE semantics.&quot; - this doesn&#3=
9;t make sense to me; unique payloads are needed for answerers that support=
 BUNDLE semantics. Copy/paste error?<br>

<br>
Would be consistent if it said &quot;for the Answerer that supports BUNDLE =
semantics&quot;.<br>
<br>
4.4 Multiple Video with Simulcast<br>
<br>
If it&#39;s intended that draft-westerlund-avtcore-rtp-<u></u>simulcast is =
used, the ssrc-group lines should probably say SCS rather than SIMULCAST.<b=
r>
<br></blockquote><div><br></div><div><br></div><div>Regarding the Simulcast=
 ssrc-group based signaling, the following draft explains its usage:=A0<a h=
ref=3D"http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-0=
0">http://tools.ietf.org/html/draft-nandakumar-mmusic-rtcweb-grouping-00</a=
>.</div>
<div><br></div><div>The idea is very similar to the Unified Plan in that , =
ssrc-group is used to signal the ssrcs for a given MediaStreamTrack as part=
 of Simulcast. It also talks about using receiver selection of the Simulcas=
t sources offered by the Sender via=A0<a href=3D"http://tools.ietf.org/html=
/draft-lennox-mmusic-sdp-source-selection-05" style=3D"font-size:1em">draft=
-lennox-mmusic-sdp-source-selection-05</a>.</div>
<div><br></div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
And now for some stuff that isn&#39;t in the document, but might want to be=
:<br>
<br>
More examples<br>
<br>
There should probably be one example where the offer contains 1 audio and 2=
 video streams, and the answer contains 1 audio and 1 video stream (fewer s=
treams in response than in request); I believe that this answer would then =
contain &quot;a=3Drecvonly&quot; and no &quot;a=3Dmsid&quot; attribute on t=
he 3rd m-line, but it&#39;s nice to be sure.<br>

<br>
Interaction with a=3Dcontent<br>
<br>
Previously, we had discussed having a &quot;content&quot; attribute on Medi=
aStreamTracks that would make a track link to an m-line with the same &quot=
;a=3Dcontent&quot; attribute. If this is still expected to work, it might b=
e worthy of inclusion here.<br>

<br>
I think that&#39;s all for this round of review. Thanks for getting it out!=
<br>
<br>
=A0 =A0 =A0 =A0 =A0 =A0 Harald<br>
<br>
<br>
<br>
<br>
______________________________<u></u>_________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/mmusic</a><br>
</blockquote></div><br>

--089e0122f0fe0a162004e1d0975a--

From Christian.Groves@nteczone.com  Thu Jul 18 18:08:47 2013
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Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Notes from Orlando human language draft discussion
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Hello Randall,

I think you're mixing protocol functionality with application.

 From a protocol perspective I think that all you need to say is 
something like:

Clients that have a language preference related to a media stream/s 
shall set the 'humintlang-send' and/or 'humintlang-recv' attribute on 
the applicable media stream at session establishment. Receivers of the 
'humanintlang-*" attributes may then apply session handling based on the 
received language/s and according to local policy. For example: Call 
centers and PSAPs may route the session to an operator that is capable 
of communicating in the preferred language/s.

I think this describes the protocol behaviour without requiring 
behaviour depending on user type.

Regards, Christian

On 19/07/2013 2:14 AM, Randall Gellens wrote:
> Hi Christian,
>
> How about this text:
>
>    Clients acting on behalf of end users are expected to set one or both
>    'humintlang-send' and 'humintlang-recv' attributes on each media
>    stream primarily intended for human communication in an offer when
>    placing an outgoing session, but either ignore or take into
>    consideration the attributes when receiving incoming calls, based on
>    local configuration and capabilities.  Systems acting on behalf of
>    call centers and PSAPs are expected to take into account the values
>    when processing inbound calls.
>
>
> At 12:11 PM +1000 7/15/13, Christian Groves wrote:
>
>>  Hello Randall and Gunnar,
>>
>>  I think Randall's clarification would be good to have in the draft. 
>> I think that when a caller support the language it is sent for every 
>> call. Whether the language is used by a receiver depends on the 
>> receiver's local capability.
>>
>>
>>  Regards, Christian
>>
>>
>>  On 14/07/2013 4:04 AM, Gunnar Hellstrom wrote:
>>>  On 2013-07-13 10:07, Randall Gellens wrote:
>>>>  At 2:49 PM +1000 7/10/13, Christian Groves wrote:
>>>>
>>>>>   I wouldn't have thought that it would be use case specific... 
>>>>> i.e. i'd only include my language when I called an emergency 
>>>>> service. I would think an endpoint would be configured with the 
>>>>> information and then provide it whenever I made a call. Is that 
>>>>> the intent?
>>>>
>>>>  I expect so, but then, I also didn't expect it to be used much in 
>>>> the person-to-person case since usually language is implied by 
>>>> context when calling an individual.  I expect it to be used when a 
>>>> person is calling an call center (corporate or emergency).  To put 
>>>> it another way, I expect a personal client to always set language 
>>>> on outbound calls, but in most cases to ignore it on inbound 
>>>> calls.  I'd expect a corporate call center, as well as emergency 
>>>> call handling facilities, to pay attention.  I should make this 
>>>> more clear in the draft.
>>>>
>>>  I expect the setting to be retrieved from a personal profile 
>>> (likely but not necessarily stored in connection with other user 
>>> information and account information) and used in every call.
>>>
>>>  /Gunnar
>>>  _______________________________________________
>>>  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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Jonathan said:=20

"I don't understand how single-session multi-source transmission will work =
without either a) signaling "a=3Dssrc-group DDP" decoding dependencies=2C o=
r b) making various implementation-specific assumptions about the structure=
 of the SVC streams."
=20
[BA] In terms of SDP choices for expressing layering within a single m line=
=2C you can either go the a=3Dssrc-group:DDP route=2C or use distinct paylo=
ad types for each layer and RFC 5583 a=3Dgroup:DDP (this is what "Unified P=
lan" Section 4.7 appears to advocate). =20

I'm not sure how "MST within a single session" could be expressed in SDP wi=
thout BUNDLE.  Was the idea to use multiple m lines and group them together=
 but use the same port?=20
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>Jonathan said: <br><br><div><div=
>"I don't understand how single-session multi-source transmission will work=
 without either a) signaling "a=3Dssrc-group DDP" decoding dependencies=2C =
or b) making various implementation-specific assumptions about the structur=
e of the SVC streams."<br>&nbsp=3B<br>[BA] In terms of SDP choices for expr=
essing layering within a single m line=2C you can either go the a=3Dssrc-gr=
oup:DDP route=2C or use distinct payload types for each layer and RFC 5583 =
a=3Dgroup:DDP (this is what "Unified Plan" Section 4.7 appears to advocate)=
.&nbsp=3B <br><br>I'm not sure how "MST within a single session" could be e=
xpressed in SDP without BUNDLE.&nbsp=3B Was the idea to use multiple m line=
s and group them together but use the same port? <br></div></div> 		 	   		=
  </div></body>
</html>=

--_d703b482-bd81-4fe3-ade3-83c8931f1022_--

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Subject: [MMUSIC] Some comments on draft-ivov-mmusic-multiple-sources
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Overall=2C I like this draft quite a bit.  Some thoughts:=20

Section 1
   Yet=2C Offer/Answer implementations in SIP=0A=
   applications have most often used them as an envelope for a maximum=0A=
   of two RTP streams (SSRCs) at a time: one in each direction.  The=0A=
   most common reason for this has been the fact these applications=0A=
   could not meaningfully render multiple SSRCs simultaneously.

[BA] While the above is probably true of legacy O/A implementations=2C vide=
o=20
implementations today frequently can and do render multiple SSRCs simultane=
ously. =20
In particular=2C implementations supporting more than "point-to-point" call=
s=2C often
go beyond "video switching" to rendering of multiple streams=2C all declare=
d on
the same m-line.=20

In particular=2C where there are multiple layers as well as multiple source=
s going
into a mixer=2C you may see multiple SSRCs coming out of the mixer.  This
could happen because a single SSRC is used for all the layers from a given =
source
(but distinct SSRCs are used for different sources)=2C or because you have =
each
layer from a given source using a distinct SSRC.=20

Section 2

   In cases where endpoints need to be able to detect this from the SDP=0A=
   Offer/Answer they could use the "max-send-ssrc" and "max-recv-ssrc"=0A=
   attributes defined in [MAX-SSRC].  It has to be noted however that=0A=
   this mechanism is still a work in progress and as such it would still=0A=
   need to be extended to provide ways of distinguishing between=0A=
   independent flows and complementary ones such as layered FEC and RTX.

[BA] Keep in mind that if you don't use a single SSRC per source=2C then th=
e semantics
of max-send-ssrc and max-recv-ssrc become ambiguous.  So you need to be abl=
e to=20
express the maximum send and receive sources as well as the maximum sent an=
d
received layers.=20

Section 5

   Another approach could be a combination of RTCP and RTP header=0A=
   extensions [RFC5285] in a way similar to the one employed by the=0A=
   Rapid Synchronisation of RTP Flows [RFC6051].  While such a mechanism=0A=
   is not currently defined by the IETF=2C specifying it could be=0A=
   relatively straightforward:

[BA] Not sure about the "straightforward" part but there are advantages to =
using
RTP and RTCP extensions.  RTP extensions allow RTP parsing to be bypassed w=
hich
is convenient if SRTP is used (since otherwise unless SSRCs are pre-declare=
d
you might have to decrypt and parse the RTP packet to figure out what layer
it corresponds to).  Today there are commonly used RTP configurations that =
can't
easily be expressed in SDP (e.g. sending individual codec layers with disti=
nct
SSRCs=2C but within the same RTP session) but that an RTP extension could h=
elp
with.  The argument on the other side is that SDP O/A is reliable=2C wherea=
s
an RTP extension or RTCP SDES would not be.  However=2C if the endgame is W=
ebRTC
without SDP=2C then I think we need to look for RTP/RTCP solutions for thes=
e issues=2C
rather than requiring an SDP dependency.=20



 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>Overall=2C I like this draft qui=
te a bit.&nbsp=3B Some thoughts: <br><br>Section 1<br><pre>   Yet=2C Offer/=
Answer implementations in SIP=0A=
   applications have most often used them as an envelope for a maximum=0A=
   of two RTP streams (SSRCs) at a time: one in each direction.  The=0A=
   most common reason for this has been the fact these applications=0A=
   could not meaningfully render multiple SSRCs simultaneously.<br><br>[BA]=
 While the above is probably true of legacy O/A implementations=2C video <b=
r>implementations today frequently can and do render multiple SSRCs simulta=
neously.  <br>In particular=2C implementations supporting more than "point-=
to-point" calls=2C often<br>go beyond "video switching" to rendering of mul=
tiple streams=2C all declared on<br>the same m-line. <br><br>In particular=
=2C where there are multiple layers as well as multiple sources going<br>in=
to a mixer=2C you may see multiple SSRCs coming out of the mixer.  This<br>=
could happen because a single SSRC is used for all the layers from a given =
source<br>(but distinct SSRCs are used for different sources)=2C or because=
 you have each<br>layer from a given source using a distinct SSRC. <br><br>=
Section 2<br><br>   In cases where endpoints need to be able to detect this=
 from the SDP=0A=
   Offer/Answer they could use the "max-send-ssrc" and "max-recv-ssrc"=0A=
   attributes defined in [<a href=3D"http://tools.ietf.org/html/draft-ivov-=
mmusic-multiple-sources-00#ref-MAX-SSRC">MAX-SSRC</a>].  It has to be noted=
 however that=0A=
   this mechanism is still a work in progress and as such it would still=0A=
   need to be extended to provide ways of distinguishing between=0A=
   independent flows and complementary ones such as layered FEC and RTX.<br=
><br>[BA] Keep in mind that if you don't use a single SSRC per source=2C th=
en the semantics<br>of max-send-ssrc and max-recv-ssrc become ambiguous.  S=
o you need to be able to <br>express the maximum send and receive sources a=
s well as the maximum sent and<br>received layers. <br><br>Section 5<br><br=
>   Another approach could be a combination of RTCP and RTP header=0A=
   extensions [<a href=3D"http://tools.ietf.org/html/rfc5285" title=3D"&quo=
t=3BA General Mechanism for RTP Header Extensions&quot=3B">RFC5285</a>] in =
a way similar to the one employed by the=0A=
   Rapid Synchronisation of RTP Flows [<a href=3D"http://tools.ietf.org/htm=
l/rfc6051" title=3D"&quot=3BRapid Synchronisation of RTP Flows&quot=3B">RFC=
6051</a>].  While such a mechanism=0A=
   is not currently defined by the IETF=2C specifying it could be=0A=
   relatively straightforward:<br><br>[BA] Not sure about the "straightforw=
ard" part but there are advantages to using<br>RTP and RTCP extensions.  RT=
P extensions allow RTP parsing to be bypassed which<br>is convenient if SRT=
P is used (since otherwise unless SSRCs are pre-declared<br>you might have =
to decrypt and parse the RTP packet to figure out what layer<br>it correspo=
nds to).  Today there are commonly used RTP configurations that can't<br>ea=
sily be expressed in SDP (e.g. sending individual codec layers with distinc=
t<br>SSRCs=2C but within the same RTP session) but that an RTP extension co=
uld help<br>with.  The argument on the other side is that SDP O/A is reliab=
le=2C whereas<br>an RTP extension or RTCP SDES would not be.  However=2C if=
 the endgame is WebRTC<br>without SDP=2C then I think we need to look for R=
TP/RTCP solutions for these issues=2C<br>rather than requiring an SDP depen=
dency. <br><br><br></pre><br> 		 	   		  </div></body>
</html>=

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Subject: Re: [MMUSIC] Unified Plan for SDP Handling
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Jonathan said:=20
> > The app-id idea is useful for multiple reasons.  One reason similar RTP=
 extensions are used today is to avoid having to parse the payload to figur=
e out what the stream is (e.g. looking for parameter sets or SEI NAL units)=
. =20
>=20
> That sounds like an interesting idea=2C but unless I'm misunderstanding y=
ou=2C that sounds very different from what app-id (or the correlator=2C in =
Unified) is doing.  Those are for identifying streams=2C whereas this sound=
s more like identifying the sub-structure of a stream (i.e. which packets c=
ontain parameter sets or SEIs).

[BA] My understanding of the problem solved by deployed RTP extensions expr=
essing layering is similar to=20
that in:
http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext=20

That is=2C these extensions are not used to flag packets containing paramet=
er sets or SEIs.  Rather the
desire is to allow the MANE to avoid having to decrypt the SRTP packet to f=
igure out what layer it
corresponds to before deciding whether to drop or forward it.  In a circums=
tance where the SSRC
either isn't pre-declared in SDP O/A or could correspond to multiple layers=
=2C that makes some sense.=20
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>Jonathan said: <br><div>&gt=3B &=
gt=3B The app-id idea is useful for multiple reasons.  One reason similar R=
TP extensions are used today is to avoid having to parse the payload to fig=
ure out what the stream is (e.g. looking for parameter sets or SEI NAL unit=
s).  <br>&gt=3B <br>&gt=3B That sounds like an interesting idea=2C but unle=
ss I'm misunderstanding you=2C that sounds very different from what app-id =
(or the correlator=2C in Unified) is doing.  Those are for identifying stre=
ams=2C whereas this sounds more like identifying the sub-structure of a str=
eam (i.e. which packets contain parameter sets or SEIs).<br><br>[BA] My und=
erstanding of the problem solved by deployed RTP extensions expressing laye=
ring is similar to <br>that in:<br><a href=3D"http://tools.ietf.org/html/dr=
aft-fineberg-avtext-temporal-layer-ext" target=3D"_blank">http://tools.ietf=
.org/html/draft-fineberg-avtext-temporal-layer-ext</a> <br><br>That is=2C t=
hese extensions are not used to flag packets containing parameter sets or S=
EIs.&nbsp=3B Rather the<br>desire is to allow the MANE to avoid having to d=
ecrypt the SRTP packet to figure out what layer it<br>corresponds to before=
 deciding whether to drop or forward it.&nbsp=3B In a circumstance where th=
e SSRC<br>either isn't pre-declared in SDP O/A or could correspond to multi=
ple layers=2C that makes some sense. <br></div> 		 	   		  </div></body>
</html>=

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On Wed, Jul 17, 2013 at 4:23 PM, Adam Roach <adam@nostrum.com> wrote:

> On 7/17/13 09:08, Stefan H=C3=A5kansson LK wrote:
>
>> I have a couple of specific questions/comments, coming from a very
>> WebRTC centric (sorry for that) perspective:
>>
>> * Simulcast: In the current API model, MediaStreamTrack is about the
>> lowest you can get to. And the same source (camera, microphone) can
>> source several MediaStreamTrack - and each of those can use individual
>> constraints for width, height, framerate, etc. So in a WebRTC context,
>> there seem to me we would not need the simulcast solution you have
>> included. The different resolutions can be sent as different
>> MediaStreamTrack's - meaning they would be separate m-lines.
>>
>
> That's what we meant to convey with this (section 2, bullet 1): "This doe=
s
> not preclude 'application level' simulcasting; i.e., the creation of
> multiple media stream tracks from a single source."
>
> It's true that we don't broach the issue of how to indicate the need for
> simulcast via a WebRTC javascript API. I think this is something that wou=
ld
> be future work, probably in the W3C. I'm going to leave any further
> explanation to Justin, as most of the simulcast ideas in the document cam=
e
> from him.


Consider the parallel with layered coding - in that case, you have multiple
dependent encodings of the track, but it's doubtful you want the recipients
to end up with multiple MediaStreamTracks and display them all in their UI.

This mechanism is basically a way to say that I have a single
MediaStreamTrack coming in, and while there might be multiple independent
or dependent encodings of it on the wire, the recipient should only create
and display a single MediaStreamTrack on its side.

>
>
>  * Direction attribute: The current WebRTC API is "sender" oriented: the
>> application can select MediaStream(Track)'s to send. Would it for WebRTC
>> make sense to have all m-lines being unidirectional (send/recvonly)?
>>
>
> I understand how this makes things cleaner in general. The issue I have i=
s
> that it interacts poorly with legacy uses.
>
>
>    (If
>> they should be sendrecv we also need to specify when a return track
>> should reuse an existing m-line, and when a new m-line should be created=
.)
>>
>>
> I agree that this could stand to be spelled out more clearly. I do not
> think that doing so will prove particularly controversial.
>
>
> /a
> ______________________________**_________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/**listinfo/mmusic<https://www.ietf.org/mailm=
an/listinfo/mmusic>
>

--047d7ba97b94b912b404e1d52085
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jul 17, 2013 at 4:23 PM, Adam Roach <span dir=3D"ltr">&lt;<=
a href=3D"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&g=
t;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On 7/17/13 09:08, Stefan H=
=C3=A5kansson LK wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I have a couple of specific questions/comments, coming from a very<br>
WebRTC centric (sorry for that) perspective:<br>
<br>
* Simulcast: In the current API model, MediaStreamTrack is about the<br>
lowest you can get to. And the same source (camera, microphone) can<br>
source several MediaStreamTrack - and each of those can use individual<br>
constraints for width, height, framerate, etc. So in a WebRTC context,<br>
there seem to me we would not need the simulcast solution you have<br>
included. The different resolutions can be sent as different<br>
MediaStreamTrack&#39;s - meaning they would be separate m-lines.<br>
</blockquote>
<br></div>
That&#39;s what we meant to convey with this (section 2, bullet 1): &quot;T=
his does not preclude &#39;application level&#39; simulcasting; i.e., the c=
reation of multiple media stream tracks from a single source.&quot;<br>


<br>
It&#39;s true that we don&#39;t broach the issue of how to indicate the nee=
d for simulcast via a WebRTC javascript API. I think this is something that=
 would be future work, probably in the W3C. I&#39;m going to leave any furt=
her explanation to Justin, as most of the simulcast ideas in the document c=
ame from him.</blockquote>

<div><br></div><div>Consider the parallel with layered coding - in that cas=
e, you have multiple dependent encodings of the track, but it&#39;s doubtfu=
l you want the recipients to end up with multiple MediaStreamTracks and dis=
play them all in their UI.</div>

<div><br></div><div>This mechanism is basically a way to say that I have a =
single MediaStreamTrack coming in, and while there might be multiple indepe=
ndent or dependent encodings of it on the wire, the recipient should only c=
reate and display a single MediaStreamTrack on its side.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
* Direction attribute: The current WebRTC API is &quot;sender&quot; oriente=
d: the<br>
application can select MediaStream(Track)&#39;s to send. Would it for WebRT=
C<br>
make sense to have all m-lines being unidirectional (send/recvonly)?<br>
</blockquote>
<br></div>
I understand how this makes things cleaner in general. The issue I have is =
that it interacts poorly with legacy uses.<div class=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
=C2=A0 (If<br>
they should be sendrecv we also need to specify when a return track<br>
should reuse an existing m-line, and when a new m-line should be created.)<=
br>
<br>
</blockquote>
<br></div>
I agree that this could stand to be spelled out more clearly. I do not thin=
k that doing so will prove particularly controversial.<div class=3D"HOEnZb"=
><div class=3D"h5"><br>
<br>
/a<br>
______________________________<u></u>_________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/mmusic</a><br>
</div></div></blockquote></div><br></div></div>

--047d7ba97b94b912b404e1d52085--

From gunnar.hellstrom@omnitor.se  Thu Jul 18 21:45:53 2013
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Date: Fri, 19 Jul 2013 06:45:34 +0200
From: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
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Subject: Re: [MMUSIC] Notes from Orlando human language draft discussion
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On 2013-07-19 03:08, Christian Groves wrote:
> Hello Randall,
>
> I think you're mixing protocol functionality with application.
>
> From a protocol perspective I think that all you need to say is 
> something like:
>
> Clients that have a language preference related to a media stream/s 
> shall set the 'humintlang-send' and/or 'humintlang-recv' attribute on 
> the applicable media stream at session establishment. Receivers of the 
> 'humanintlang-*" attributes may then apply session handling based on 
> the received language/s and according to local policy. For example: 
> Call centers and PSAPs may route the session to an operator that is 
> capable of communicating in the preferred language/s.
>
> I think this describes the protocol behaviour without requiring 
> behaviour depending on user type.
Yes, this looks better.
With this wording, I see a risk that the answering part does not realize 
that they shall also send the attributes. So, I suggest to add "both 
when offering and answering" in the first sentence.
With that, and a couple of other small changes, the paragraph could be:

"Clients that have a language preference related to media stream/s shall 
set the 'humintlang-send' and/or 'humintlang-recv' attribute on the 
applicable media stream/s at session establishment both when offering 
and answering. Receivers of the 'humanintlang-*" attributes may apply 
session handling based on the received language preferences and 
according to local policy. For example: Call centers and PSAPs may route 
the session to an operator that is capable of communicating in the 
preferred language/s ."


/Gunnar




>
> Regards, Christian
>
> On 19/07/2013 2:14 AM, Randall Gellens wrote:
>> Hi Christian,
>>
>> How about this text:
>>
>>    Clients acting on behalf of end users are expected to set one or both
>>    'humintlang-send' and 'humintlang-recv' attributes on each media
>>    stream primarily intended for human communication in an offer when
>>    placing an outgoing session, but either ignore or take into
>>    consideration the attributes when receiving incoming calls, based on
>>    local configuration and capabilities.  Systems acting on behalf of
>>    call centers and PSAPs are expected to take into account the values
>>    when processing inbound calls.
>>
>>
>> At 12:11 PM +1000 7/15/13, Christian Groves wrote:
>>
>>>  Hello Randall and Gunnar,
>>>
>>>  I think Randall's clarification would be good to have in the draft. 
>>> I think that when a caller support the language it is sent for every 
>>> call. Whether the language is used by a receiver depends on the 
>>> receiver's local capability.
>>>
>>>
>>>  Regards, Christian
>>>
>>>
>>>  On 14/07/2013 4:04 AM, Gunnar Hellstrom wrote:
>>>>  On 2013-07-13 10:07, Randall Gellens wrote:
>>>>>  At 2:49 PM +1000 7/10/13, Christian Groves wrote:
>>>>>
>>>>>>   I wouldn't have thought that it would be use case specific... 
>>>>>> i.e. i'd only include my language when I called an emergency 
>>>>>> service. I would think an endpoint would be configured with the 
>>>>>> information and then provide it whenever I made a call. Is that 
>>>>>> the intent?
>>>>>
>>>>>  I expect so, but then, I also didn't expect it to be used much in 
>>>>> the person-to-person case since usually language is implied by 
>>>>> context when calling an individual.  I expect it to be used when a 
>>>>> person is calling an call center (corporate or emergency).  To put 
>>>>> it another way, I expect a personal client to always set language 
>>>>> on outbound calls, but in most cases to ignore it on inbound 
>>>>> calls.  I'd expect a corporate call center, as well as emergency 
>>>>> call handling facilities, to pay attention.  I should make this 
>>>>> more clear in the draft.
>>>>>
>>>>  I expect the setting to be retrieved from a personal profile 
>>>> (likely but not necessarily stored in connection with other user 
>>>> information and account information) and used in every call.
>>>>
>>>>  /Gunnar
>>>>  _______________________________________________
>>>>  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
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Martin Thomson'" <martin.thomson@gmail.com>, "'Adam Roach'" <adam@nostrum.com>
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Cc: 'Jonathan Lennox' <jonathan@vidyo.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] Unified Plan for SDP Handling - multiple media sources
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> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Martin Thomson
> Sent: 18 July, 2013 5:21 AM
> To: Adam Roach
> Cc: Jonathan Lennox; mmusic@ietf.org
> Subject: Re: [MMUSIC] Unified Plan for SDP Handling - multiple media
> sources
> 
> On 17 July 2013 14:44, Adam Roach <adam@nostrum.com> wrote:
> > On 7/17/13 16:40, Jonathan Lennox wrote:
> >>
> >> That said, what I still don't see is, if you have the correlator, why
> >> you then actually need to send the a=ssrc in the SDP at all? The
> >> correlator should be sufficient for disambiguation, and it lets the
> >> sender switch the ssrc it's sending for an m-line without needing a
fresh
> offer/answer.
> >
> >
> > That might be worth exploring. I honestly haven't given it a whole lot
> > of thought. Perhaps Justin or Martin have a more well-formed opinion
> > on the topic?
> 
> I did consider this, but there are several reasons that I prefer to have
> signaling.  The main one being that it is the only guaranteed way to get
this
> information through.  In the happy cases, the RTP header is enough, but if
> you are losing packets at session startup (a real possibility in some
> configurations), or RTP header extensions are being dropped, or you need
to
> communicate about an in progress stream and can't guarantee that you can
> cause the sender to re-send a few packets with the header extension, or
...
> 
> Basically, there are too many ways for the RTP extension to break.
> Not enough that it isn't hugely valuable for dealing with the race, but it
can't
> be the only mechanism that you rely on.
[Roni Even]  One of the reasons we have also SDES message with the ID is
because the header extension is optional and can be removed on the path. 


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


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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-01.txt - absolute preferences
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We also have the need for absolute preferences to handle and decide on.

On 2013-07-15 00:14, Gunnar Hellstrom wrote:
> 3. Absolute preference is still needed.
> I described earlier the need for an absolute preference indication. But I did not provide any coding proposal.
> I suggest to add something at the end of the language tag. Regardless if it gets before or after the * if it is there.
> Can a "+" be used to indicate a strongly preferred language and modality, no extra specification a managed language and modality
> and a "-" a non-preferred, but possible language and modality.

The background was in an earlier mail:
There is also a need for absolute preference. Or relative order of 
preference between media.

It is easy to use a sign language example fort that situation.

A person prefers to use British sign language in both directions but is 
capable with some reluctance to use English text. That is natural. Text 
is always slower and less expressive than sign language or spoken language.

A call goes to a call center that declared spoken English and English 
text in both directions.
The result can be a direct connection using English text, or invocation 
of a sign relay service translating between British sign language and 
spoken English.

An ability to declare that the British sign language is preferred over 
the English text would result in the invocation of the sign relay 
service and make both sides happy with a call with good flow.

Without the relative preference between media, the call will likely 
result in the direct text communication that will work but with less 
satisfaction.

This can be solved either by an absolute preference indication  : 
British sing language *preferred*  , English text *managed*
Or by some order of preference between the media;  video *preferred* 
text *managed*

I think an absolute preference indication is easiest to handle. Media 
preference will collide with relative preference between languages in 
each medium.

I suggest that that is introduced with some suitable coding.


It seems best t enter this in a paragraph in 7.1. I also propose that 
7.1 is subdivided in subsections to make it easier to navigate.

This addition would fit as a paragraph after this existing one:

"

  In an offer, the 'humintlang-send' values constitute a list in
    preference order (first is most preferred) of the languages the
    offerer wishes to send, and the 'humintlang-recv' values constitute a
    list in preference order of the languages the offerer wishes to
    receive.  In cases where the user wishes to use one media for sending
    and another for receiving (such as a speech-impaired user who wishes
    to send using text and receive using audio), one of the two will be
    unset.  In other cases, both SHOULD have the same values in the same
    order.  The two SHOULD NOT be set to languages which are difficult to
    match together (e.g., specifying a desire to send audio in Hungarian
    and receive audio in Portuguese will make it difficult to
    successfully complete the call)."

I suggest to add this paragraph:

"
In order to express order of preference between different media, an absolute
preference indication MAY be inserted in the attribute, after the language tag.
An exclamation mark "!" is used to indicate a strongly preferred language and modality,
no addition is used to indicate a managed language and modality, and a hash sign "#"
is used to indicate a non-preferred but possible fall-back language and modality.
An example can be a person who prefers to use British sign language in both directions but
is capable with some reluctance to use English text, calling a call center that has
competence for spoken English and English text in both directions. The result can be
a direct connection using English text, or invocation of a sign relay service translating
between British sign language and spoken English. The preference indication on British
sign language guides the call centre in this case to invoke a relay service to get most
satisfaction in the call, while in another case, a relay service for the discovered
gap between the preferred languages might not be available, and then the fall-back to
English text is a viable alternative.  "



Introduction of this coding with "!" , nothing, and "#" creates a need for a syntax chapter,
where the allowed combinations of the language tag, these markers and the "*" for connect
anyway are described more formally.



Regards,
Gunnar




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Date: Fri, 19 Jul 2013 07:49:37 -0700
To: Justin Uberti <juberti@google.com>
Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Unified Plan for SDP Handling
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This is weird - I see your reply but i don't see Stephan's email=20

On Jul 18, 2013, at 8:43 PM, Justin Uberti <juberti@google.com> wrote:

>=20
>=20
>=20
> On Wed, Jul 17, 2013 at 4:23 PM, Adam Roach <adam@nostrum.com> wrote:
>> On 7/17/13 09:08, Stefan H=C3=A5kansson LK wrote:
>>> I have a couple of specific questions/comments, coming from a very
>>> WebRTC centric (sorry for that) perspective:
>>>=20
>>> * Simulcast: In the current API model, MediaStreamTrack is about the
>>> lowest you can get to. And the same source (camera, microphone) can
>>> source several MediaStreamTrack - and each of those can use individual
>>> constraints for width, height, framerate, etc. So in a WebRTC context,
>>> there seem to me we would not need the simulcast solution you have
>>> included. The different resolutions can be sent as different
>>> MediaStreamTrack's - meaning they would be separate m-lines.
>>=20
>> That's what we meant to convey with this (section 2, bullet 1): "This doe=
s not preclude 'application level' simulcasting; i.e., the creation of multi=
ple media stream tracks from a single source."
>>=20
>> It's true that we don't broach the issue of how to indicate the need for s=
imulcast via a WebRTC javascript API. I think this is something that would b=
e future work, probably in the W3C. I'm going to leave any further explanati=
on to Justin, as most of the simulcast ideas in the document came from him.
>=20
> Consider the parallel with layered coding - in that case, you have multipl=
e dependent encodings of the track, but it's doubtful you want the recipient=
s to end up with multiple MediaStreamTracks and display them all in their UI=
.
>=20
> This mechanism is basically a way to say that I have a single MediaStreamT=
rack coming in, and while there might be multiple independent or dependent e=
ncodings of it on the wire, the recipient should only create and display a s=
ingle MediaStreamTrack on its side.
>>=20
>>=20
>>> * Direction attribute: The current WebRTC API is "sender" oriented: the
>>> application can select MediaStream(Track)'s to send. Would it for WebRTC=

>>> make sense to have all m-lines being unidirectional (send/recvonly)?
>>=20
>> I understand how this makes things cleaner in general. The issue I have i=
s that it interacts poorly with legacy uses.
>>=20
>>=20
>>>   (If
>>> they should be sendrecv we also need to specify when a return track
>>> should reuse an existing m-line, and when a new m-line should be created=
.)
>>=20
>> I agree that this could stand to be spelled out more clearly. I do not th=
ink that doing so will prove particularly controversial.
>>=20
>>=20
>> /a
>> _______________________________________________
>> 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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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div>This is weird - I see your reply but i=
 don't see Stephan's email&nbsp;</div><div><br>On Jul 18, 2013, at 8:43 PM, J=
ustin Uberti &lt;<a href=3D"mailto:juberti@google.com">juberti@google.com</a=
>&gt; wrote:<br><br></div><blockquote type=3D"cite"><div><div dir=3D"ltr"><b=
r><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed, Jul 1=
7, 2013 at 4:23 PM, Adam Roach <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@=
nostrum.com" target=3D"_blank">adam@nostrum.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex"><div class=3D"im">On 7/17/13 09:08, Stefan H=C3=
=A5kansson LK wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">
I have a couple of specific questions/comments, coming from a very<br>
WebRTC centric (sorry for that) perspective:<br>
<br>
* Simulcast: In the current API model, MediaStreamTrack is about the<br>
lowest you can get to. And the same source (camera, microphone) can<br>
source several MediaStreamTrack - and each of those can use individual<br>
constraints for width, height, framerate, etc. So in a WebRTC context,<br>
there seem to me we would not need the simulcast solution you have<br>
included. The different resolutions can be sent as different<br>
MediaStreamTrack's - meaning they would be separate m-lines.<br>
</blockquote>
<br></div>
That's what we meant to convey with this (section 2, bullet 1): "This does n=
ot preclude 'application level' simulcasting; i.e., the creation of multiple=
 media stream tracks from a single source."<br>


<br>
It's true that we don't broach the issue of how to indicate the need for sim=
ulcast via a WebRTC javascript API. I think this is something that would be f=
uture work, probably in the W3C. I'm going to leave any further explanation t=
o Justin, as most of the simulcast ideas in the document came from him.</blo=
ckquote>

<div><br></div><div>Consider the parallel with layered coding - in that case=
, you have multiple dependent encodings of the track, but it's doubtful you w=
ant the recipients to end up with multiple MediaStreamTracks and display the=
m all in their UI.</div>

<div><br></div><div>This mechanism is basically a way to say that I have a s=
ingle MediaStreamTrack coming in, and while there might be multiple independ=
ent or dependent encodings of it on the wire, the recipient should only crea=
te and display a single MediaStreamTrack on its side.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex"><div class=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">
* Direction attribute: The current WebRTC API is "sender" oriented: the<br>
application can select MediaStream(Track)'s to send. Would it for WebRTC<br>=

make sense to have all m-lines being unidirectional (send/recvonly)?<br>
</blockquote>
<br></div>
I understand how this makes things cleaner in general. The issue I have is t=
hat it interacts poorly with legacy uses.<div class=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">
&nbsp; (If<br>
they should be sendrecv we also need to specify when a return track<br>
should reuse an existing m-line, and when a new m-line should be created.)<b=
r>
<br>
</blockquote>
<br></div>
I agree that this could stand to be spelled out more clearly. I do not think=
 that doing so will prove particularly controversial.<div class=3D"HOEnZb"><=
div class=3D"h5"><br>
<br>
/a<br>
______________________________<u></u>_________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br>=

<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">h=
ttps://www.ietf.org/mailman/<u></u>listinfo/mmusic</a><br>
</div></div></blockquote></div><br></div></div>
</div></blockquote><blockquote type=3D"cite"><div><span>____________________=
___________________________</span><br><span>mmusic mailing list</span><br><s=
pan><a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a></span><br><span><=
a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org=
/mailman/listinfo/mmusic</a></span><br></div></blockquote></body></html>=

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From: Cullen Jennings <fluffy@iii.ca>
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To: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Unified Plan for SDP Handling
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I like this - it looks like a reasonable way forward=20

On Jul 15, 2013, at 12:04 PM, Adam Roach <adam@nostrum.com> wrote:

> [Cross-posting to RTCWEB; follow-ups to MMUSIC, please]
>=20
> After significant work, Justin, Martin and I have managed to produce a com=
promise plan that provides a high degree of interoperability with existing d=
evices (and future non-WebRTC devices) while not being excessively onerous f=
or WebRTC implementations or applications that use them. It's been a tricky b=
alancing act, but I think we've found a good mix between the two that can fo=
rm a solid basis for the working group to move forward.
>=20
> Rather than summarize the key points of the document in this email, I dire=
ct interested parties to section 2 of the document, which summarizes the key=
 aspects of the plan in eight relatively concise bullet points.
>=20
> I apologize for the late publication date of this document -- there's actu=
ally been a lot more work put into coming up with a unified draft than I ori=
ginally anticipated, and the production of this document took at least two w=
eeks longer than I expected it to.
>=20
> Note that this document is intended to be a plan for the work to be done i=
n this area, and not a specification in itself. The intention is that its co=
ntents are used as the basis for work in several other drafts -- some new, s=
ome not -- that form the corpus of work necessary for RTCWEB (and potentiall=
y CLUE) to move forward. Except in rare cases, the document does not attempt=
 to explicitly call out venues or documents for such work, as we (or, at the=
 very least, I) anticipate guidance from the various working group chairs to=
 assist in such decisions.
>=20
> Comments prior to Berlin would be very helpful, although this will clearly=
 be a point of significant discussion at the face-to-face meeting.
>=20
> Document link:
>=20
> http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt
>=20
> /a
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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Date: Fri, 19 Jul 2013 10:54:26 -0400
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On Jul 18, 2013, at 10:02 PM, Bernard Aboba <bernard_aboba@hotmail.com<mail=
to:bernard_aboba@hotmail.com>> wrote:

Jonathan said:

"I don't understand how single-session multi-source transmission will work =
without either a) signaling "a=3Dssrc-group DDP" decoding dependencies, or =
b) making various implementation-specific assumptions about the structure o=
f the SVC streams."

[BA] In terms of SDP choices for expressing layering within a single m line=
, you can either go the a=3Dssrc-group:DDP route, or use distinct payload t=
ypes for each layer and RFC 5583 a=3Dgroup:DDP (this is what "Unified Plan"=
 Section 4.7 appears to advocate).

I'm not sure how "MST within a single session" could be expressed in SDP wi=
thout BUNDLE.  Was the idea to use multiple m lines and group them together=
 but use the same port?

I think the authors of RFC 6190 didn't see a use case for MST within a sing=
le RTP session -- all of the interesting MST use cases they saw involved se=
parate transport flows for the separate layers, and if you're sending all t=
he layers on a single transport flow, SST is significantly simpler.

What requirement are you trying to achieve that's met by MST in a single se=
ssion, but not by SST?

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



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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://13/"></head><body style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space=
; "><br><div><div>On Jul 18, 2013, at 10:02 PM, Bernard Aboba &lt;<a href=
=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt; wro=
te:</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">Jonathan said:<span class=3D"Apple-converted-space">&nbsp=
;</span><br><br><div>"I don't understand how single-session multi-source tr=
ansmission will work without either a) signaling "a=3Dssrc-group DDP" decod=
ing dependencies, or b) making various implementation-specific assumptions =
about the structure of the SVC streams."<br>&nbsp;<br>[BA] In terms of SDP =
choices for expressing layering within a single m line, you can either go t=
he a=3Dssrc-group:DDP route, or use distinct payload types for each layer a=
nd RFC 5583 a=3Dgroup:DDP (this is what "Unified Plan" Section 4.7 appears =
to advocate).&nbsp;<span class=3D"Apple-converted-space">&nbsp;</span><br><=
br>I'm not sure how "MST within a single session" could be expressed in SDP=
 without BUNDLE.&nbsp; Was the idea to use multiple m lines and group them =
together but use the same port?<span class=3D"Apple-converted-space">&nbsp;=
</span><br></div></div></div></blockquote><br></div><div>I think the author=
s of RFC 6190 didn't see a use case for MST within a single RTP session -- =
all of the interesting MST use cases they saw involved separate transport f=
lows for the separate layers, and if you're sending all the layers on a sin=
gle transport flow, SST is significantly simpler.</div><div><br></div><div>=
What requirement are you trying to achieve that's met by MST in a single se=
ssion, but not by SST?</div><br><div apple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; color:=
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rmal; font-weight: normal; letter-spacing: normal; line-height: normal; orp=
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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>=

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From jonathan@vidyo.com  Fri Jul 19 07:59:04 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
Date: Fri, 19 Jul 2013 10:58:41 -0400
Thread-Topic: [MMUSIC] Unified Plan for SDP Handling
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On Jul 18, 2013, at 10:52 PM, Bernard Aboba <bernard_aboba@hotmail.com<mail=
to:bernard_aboba@hotmail.com>> wrote:

Jonathan said:
> > The app-id idea is useful for multiple reasons. One reason similar RTP =
extensions are used today is to avoid having to parse the payload to figure=
 out what the stream is (e.g. looking for parameter sets or SEI NAL units).
>
> That sounds like an interesting idea, but unless I'm misunderstanding you=
, that sounds very different from what app-id (or the correlator, in Unifie=
d) is doing. Those are for identifying streams, whereas this sounds more li=
ke identifying the sub-structure of a stream (i.e. which packets contain pa=
rameter sets or SEIs).

[BA] My understanding of the problem solved by deployed RTP extensions expr=
essing layering is similar to
that in:
http://tools.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext

That is, these extensions are not used to flag packets containing parameter=
 sets or SEIs.  Rather the
desire is to allow the MANE to avoid having to decrypt the SRTP packet to f=
igure out what layer it
corresponds to before deciding whether to drop or forward it.  In a circums=
tance where the SSRC
either isn't pre-declared in SDP O/A or could correspond to multiple layers=
, that makes some sense.

I agree what you're describing is similar to the Fineberg draft, and a comm=
on mechanism (codec-independent, to the extent possible) would likely be us=
eful.

I don't think it's very similar to app-id, though -- unless you're assuming=
 MST here too, and you're trying to identify the component MST streams of a=
 layered encoding?

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



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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://15/"></head><body style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space=
; "><br><div><div>On Jul 18, 2013, at 10:52 PM, Bernard Aboba &lt;<a href=
=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt; wro=
te:</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">Jonathan said:<span class=3D"Apple-converted-space">&nbsp=
;</span><br><div>&gt; &gt; The app-id idea is useful for multiple reasons. =
One reason similar RTP extensions are used today is to avoid having to pars=
e the payload to figure out what the stream is (e.g. looking for parameter =
sets or SEI NAL units).<span class=3D"Apple-converted-space">&nbsp;</span><=
br>&gt;<span class=3D"Apple-converted-space">&nbsp;</span><br>&gt; That sou=
nds like an interesting idea, but unless I'm misunderstanding you, that sou=
nds very different from what app-id (or the correlator, in Unified) is doin=
g. Those are for identifying streams, whereas this sounds more like identif=
ying the sub-structure of a stream (i.e. which packets contain parameter se=
ts or SEIs).<br><br>[BA] My understanding of the problem solved by deployed=
 RTP extensions expressing layering is similar to<span class=3D"Apple-conve=
rted-space">&nbsp;</span><br>that in:<br><a href=3D"http://tools.ietf.org/h=
tml/draft-fineberg-avtext-temporal-layer-ext" target=3D"_blank">http://tool=
s.ietf.org/html/draft-fineberg-avtext-temporal-layer-ext</a><span class=3D"=
Apple-converted-space">&nbsp;</span><br><br>That is, these extensions are n=
ot used to flag packets containing parameter sets or SEIs.&nbsp; Rather the=
<br>desire is to allow the MANE to avoid having to decrypt the SRTP packet =
to figure out what layer it<br>corresponds to before deciding whether to dr=
op or forward it.&nbsp; In a circumstance where the SSRC<br>either isn't pr=
e-declared in SDP O/A or could correspond to multiple layers, that makes so=
me sense.<span class=3D"Apple-converted-space">&nbsp;</span><br></div></div=
></div></blockquote><br></div><div>I agree what you're describing is simila=
r to the Fineberg draft, and a common mechanism (codec-independent, to the =
extent possible) would likely be useful.</div><div><br></div><div>I don't t=
hink it's very similar to app-id, though -- unless you're assuming MST here=
 too, and you're trying to identify the component MST streams of a layered =
encoding?</div><br><div apple-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=
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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>
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Subject: Re: [MMUSIC] [rtcweb] Unified Plan for SDP Handling
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Please forgive me for coming a bit late to the discussion on this, but what
types of "legacy devices" are being targeted?  At least from what I have
seen, most legacy equipment doesn't support ICE, definitely does not
support DTLS-SRTP, most do not support SDES-SRTP, none bundling, none
rtcp-mux, etc.

I believe that there might be some sort of legacy interop with disabling of
bundling, rtcp-mux possibly as well, and supporting SDES or non encrypted
sessions.  Otherwise, I don't know what legacy devices means in the context.

Is there a place where a definition of a "legacy device" (as mentioned in
your draft) is given in the webrtc specs?

Matthew Fredrickson


On Mon, Jul 15, 2013 at 2:04 PM, Adam Roach <adam@nostrum.com> wrote:

> [Cross-posting to RTCWEB; follow-ups to MMUSIC, please]
>
> After significant work, Justin, Martin and I have managed to produce a
> compromise plan that provides a high degree of interoperability with
> existing devices (and future non-WebRTC devices) while not being
> excessively onerous for WebRTC implementations or applications that use
> them. It's been a tricky balancing act, but I think we've found a good mix
> between the two that can form a solid basis for the working group to move
> forward.
>
> Rather than summarize the key points of the document in this email, I
> direct interested parties to section 2 of the document, which summarizes
> the key aspects of the plan in eight relatively concise bullet points.
>
> I apologize for the late publication date of this document -- there's
> actually been a lot more work put into coming up with a unified draft than
> I originally anticipated, and the production of this document took at least
> two weeks longer than I expected it to.
>
> Note that this document is intended to be a plan for the work to be done
> in this area, and not a specification in itself. The intention is that its
> contents are used as the basis for work in several other drafts -- some
> new, some not -- that form the corpus of work necessary for RTCWEB (and
> potentially CLUE) to move forward. Except in rare cases, the document does
> not attempt to explicitly call out venues or documents for such work, as we
> (or, at the very least, I) anticipate guidance from the various working
> group chairs to assist in such decisions.
>
> Comments prior to Berlin would be very helpful, although this will clearly
> be a point of significant discussion at the face-to-face meeting.
>
> Document link:
>
> http://www.ietf.org/id/draft-**roach-mmusic-unified-plan-00.**txt<http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt>
>
> /a
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

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<div dir=3D"ltr">Please forgive me for coming a bit late to the discussion =
on this, but what types of &quot;legacy devices&quot; are being targeted? =
=A0At least from what I have seen, most legacy equipment doesn&#39;t suppor=
t ICE, definitely does not support DTLS-SRTP, most do not support SDES-SRTP=
, none bundling, none rtcp-mux, etc.<div>
<br></div><div>I believe that there might be some sort of legacy interop wi=
th disabling of bundling, rtcp-mux possibly as well, and supporting SDES or=
 non encrypted sessions. =A0Otherwise, I don&#39;t know what legacy devices=
 means in the context.</div>
<div><br></div><div>Is there a place where a definition of a &quot;legacy d=
evice&quot; (as mentioned in your draft) is given in the webrtc specs?</div=
><div><br></div><div>Matthew Fredrickson</div></div><div class=3D"gmail_ext=
ra">
<br><br><div class=3D"gmail_quote">On Mon, Jul 15, 2013 at 2:04 PM, Adam Ro=
ach <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nostrum.com" target=3D"_bl=
ank">adam@nostrum.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x">
[Cross-posting to RTCWEB; follow-ups to MMUSIC, please]<br>
<br>
After significant work, Justin, Martin and I have managed to produce a comp=
romise plan that provides a high degree of interoperability with existing d=
evices (and future non-WebRTC devices) while not being excessively onerous =
for WebRTC implementations or applications that use them. It&#39;s been a t=
ricky balancing act, but I think we&#39;ve found a good mix between the two=
 that can form a solid basis for the working group to move forward.<br>

<br>
Rather than summarize the key points of the document in this email, I direc=
t interested parties to section 2 of the document, which summarizes the key=
 aspects of the plan in eight relatively concise bullet points.<br>
<br>
I apologize for the late publication date of this document -- there&#39;s a=
ctually been a lot more work put into coming up with a unified draft than I=
 originally anticipated, and the production of this document took at least =
two weeks longer than I expected it to.<br>

<br>
Note that this document is intended to be a plan for the work to be done in=
 this area, and not a specification in itself. The intention is that its co=
ntents are used as the basis for work in several other drafts -- some new, =
some not -- that form the corpus of work necessary for RTCWEB (and potentia=
lly CLUE) to move forward. Except in rare cases, the document does not atte=
mpt to explicitly call out venues or documents for such work, as we (or, at=
 the very least, I) anticipate guidance from the various working group chai=
rs to assist in such decisions.<br>

<br>
Comments prior to Berlin would be very helpful, although this will clearly =
be a point of significant discussion at the face-to-face meeting.<br>
<br>
Document link:<br>
<br>
<a href=3D"http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt" t=
arget=3D"_blank">http://www.ietf.org/id/draft-<u></u>roach-mmusic-unified-p=
lan-00.<u></u>txt</a><br>
<br>
/a<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote></div><br></div>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [MMUSIC] [rtcweb] Unified Plan for SDP Handling - Glare
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Subject: Re: [MMUSIC] [rtcweb] Unified Plan for SDP Handling - Glare
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Hi,

I do realize that the draft does not represent the complete glare solution.

Paul has raised some issues/questions, and I also have some. I'll leave the=
 SDP low-level ones for now, and only raise a few high-level ones.

Q1: How essential part of the unified plan is having *A* working "partial o=
/a" solution? TECHNICALLY, the other parts of the unified plan do not depen=
d on having a "partial o/a" solution, but I would like to know whether peop=
le (specially the draft authors) would be willing to move ahead with unifie=
d plan even if we would not manage to specify a working "partial o/a" solut=
ion.

Q2a: When sending a partial offer, the draft says that only the m=3D line e=
xplicitly signaled is affected. Note, however, that you will still signal i=
nformation (e.g. the SDP version value, the group:BUNDLE attribute etc) whi=
ch affect the whole SDP, and if you have multiple outstanding partial offer=
s (for different m=3D lines) that can cause issues.

Q2b: In addition, when using BUNDLE, modifying an m=3D line might implicitl=
y modify other m=3D lines, and there are rules (also indicated in the draft=
) mandating identical SDP attribute values etc. So, if you have multiple ou=
tstanding partial offers, you may end up with all kind of different "BUNDLE=
 collisions".

Q3: If I send a partial offer, for a given m=3D line, can the answer only c=
ontain that given m=3D line? What if the partial offer causes other m=3D li=
nes to be modified?


Regards,

Christer





L=E4hett=E4j=E4: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] P=
uolesta Adam Roach
L=E4hetetty: 17. hein=E4kuuta 2013 20:08
Vastaanottaja: Paul Kyzivat
Kopio: mmusic@ietf.org
Aihe: Re: [MMUSIC] [rtcweb] Unified Plan for SDP Handling

On 7/17/13 11:38, Paul Kyzivat wrote:
Adam,

While I see there can be some value to extending O/A to allow for smaller c=
ritical sections, I fear that it is going to be very messy to get "right". =
Some messy things:

- Glare between a partial O/A and a full O/A

...is glare. Resolve as normal. For SIP, that means timers.


- Is a partial O/A permitted in a sip reINVITE? Can you have
  an offerless reINVITE where the response contains a partial offer?
  Or can these only be done in UPDATE?

No, no, and no. In SIP, the protocol machinery would not allow INVITE or UP=
DATE to complete concurrently in opposite directions for any amount of over=
lap. We aren't proposing a change to SIP state machines here, only to the o=
ffer/answer model. This necessarily means that we need to carry partial off=
ers and partial answers in something else -- probably INFO.


- In SIP, processing of concurrent reINVITEs/UPDATEs containing
  partial SDP updates. This seems to already be a

You left this unfinished, but I *think* it was going to make the same point=
 I did in my response above.


- How to assign version numbers to the full SDP as a result
  of partial updates

Section 3.4.1: "... the SDP version is incremented by one..."



- I'm not sure your algorithm for resolving glare when adding
  streams works in all cases. (More below on this.)

These (and ones I haven't thought of) *may* all be resolvable. But this cer=
tainly complicates the already complex O/A process - maybe to the point tha=
t many get it wrong. (I don't want to write rfc6337bis!)

My concern with the glare resolution algorithm is that for it to work both =
sides must realize that glare occurred. And I think there are common sequen=
ces when one side won't realize it.

I'm not saying I didn't miss it, but I did work through as many permutation=
s as I could think of, and none of them are left out by the algorithm descr=
ibed in section 3.4.1.

That said, I did my analysis in the context of WebRTC, and assumed a reliab=
le, in-order delivery channel for messages. I recognize that we don't have =
that for SIP, and that messages in the same direction can be reordered. I a=
dmit that this does raise some interesting scenarios that may require addit=
ional work to resolve.


I'm reluctant to buy into a scheme that depends on this without knowing it =
it will be implementable.

I understand your reluctance, and don't want to gloss over it. This is but =
one plank in the overall plan, and it represents what we think is the best =
(but not only) solution to this specific requirement. If the issues prove i=
ntractable, we could fall back to an alternate scheme, such as the designat=
ed-offerer I describe in draft-roach-rtcweb-glareless-add-00, or the token-=
passing scheme described by Ted Hardie (cf. http://www.ietf.org/mail-archiv=
e/web/rtcweb/current/msg07361.html).

/a

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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"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></=
o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I do reali=
ze that the draft does not represent the complete glare solution.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Paul has r=
aised some issues/questions, and I also have some. I&#8217;ll leave the SDP=
 low-level ones for now, and only raise a few high-level ones.<o:p></o:p></=
span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:11.0pt;fo=
nt-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Q1</spa=
n></b><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Cali=
bri&quot;,&quot;sans-serif&quot;;color:#1F497D">: How essential part of the=
 unified
 plan is having *<b>A</b>* working &#8220;partial o/a&#8221; solution? TECH=
NICALLY, the other parts of the unified plan do not depend on having a &#82=
20;partial o/a&#8221; solution, but I would like to know whether people (sp=
ecially the draft authors) would be willing to move ahead
 with unified plan even if we would not manage to specify a working &#8220;=
partial o/a&#8221; solution.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:11.0pt;fo=
nt-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Q2a</sp=
an></b><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,&quot;sans-serif&quot;;color:#1F497D">: When sending a partial o=
ffer, the
 draft says that only the m=3D line explicitly signaled is affected. Note, =
however, that you will still signal information (e.g. the SDP version value=
, the group:BUNDLE attribute etc) which affect the whole SDP, and if you ha=
ve multiple outstanding partial offers
 (for different m=3D lines) that can cause issues.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:11.0pt;fo=
nt-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Q2b</sp=
an></b><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Cal=
ibri&quot;,&quot;sans-serif&quot;;color:#1F497D">: In addition, when using =
BUNDLE,
 modifying an m=3D line might implicitly modify other m=3D lines, and there=
 are rules (also indicated in the draft) mandating identical SDP attribute =
values etc. So, if you have multiple outstanding partial offers, you may en=
d up with all kind of different &#8220;BUNDLE
 collisions&#8221;.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:11.0pt;fo=
nt-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Q3</spa=
n></b><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Cali=
bri&quot;,&quot;sans-serif&quot;;color:#1F497D">: If I send a partial offer=
, for a
 given m=3D line, can the answer only contain that given m=3D line? What if=
 the partial offer causes other m=3D lines to be modified?<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:10.0pt;fo=
nt-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">L=E4h=
ett=E4j=E4:</span></b><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-f=
amily:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext"> mmusic-b=
ounces@ietf.org
 [mailto:mmus</span><span style=3D"font-size:10.0pt;font-family:&quot;Tahom=
a&quot;,&quot;sans-serif&quot;;color:windowtext">ic-bounces@ietf.org]
<b>Puolesta </b>Adam Roach<br>
<b>L=E4hetetty:</b> 17. hein=E4kuuta 2013 20:08<br>
<b>Vastaanottaja:</b> Paul Kyzivat<br>
<b>Kopio:</b> mmusic@ietf.org<br>
<b>Aihe:</b> Re: [MMUSIC] [rtcweb] Unified Plan for SDP Handling<o:p></o:p>=
</span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On 7/17/13 11:38, Paul Kyzivat wrote:<o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<p class=3D"MsoNormal">Adam, <br>
<br>
While I see there can be some value to extending O/A to allow for smaller c=
ritical sections, I fear that it is going to be very messy to get &quot;rig=
ht&quot;. Some messy things:
<br>
<br>
- Glare between a partial O/A and a full O/A<o:p></o:p></p>
</blockquote>
<p class=3D"MsoNormal"><br>
...is glare. Resolve as normal. For SIP, that means timers.<br>
<br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal">- Is a partial O/A permitted in a sip reINVITE? Can =
you have
<br>
&nbsp; an offerless reINVITE where the response contains a partial offer? <=
br>
&nbsp; Or can these only be done in UPDATE? <o:p></o:p></p>
<p class=3D"MsoNormal"><br>
No, no, and no. In SIP, the protocol machinery would not allow INVITE or UP=
DATE to complete concurrently in opposite directions for any amount of over=
lap. We aren't proposing a change to SIP state machines here, only to the o=
ffer/answer model. This necessarily
 means that we need to carry partial offers and partial answers in somethin=
g else -- probably INFO.<br>
<br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal">- In SIP, processing of concurrent reINVITEs/UPDATEs=
 containing
<br>
&nbsp; partial SDP updates. This seems to already be a <o:p></o:p></p>
<p class=3D"MsoNormal"><br>
You left this unfinished, but I *think* it was going to make the same point=
 I did in my response above.<br>
<br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal">- How to assign version numbers to the full SDP as a=
 result <br>
&nbsp; of partial updates <o:p></o:p></p>
<p class=3D"MsoNormal"><br>
Section 3.4.1: &quot;... the SDP version is incremented by one...&quot;<br>
<br>
<br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal">- I'm not sure your algorithm for resolving glare wh=
en adding
<br>
&nbsp; streams works in all cases. (More below on this.) <br>
<br>
These (and ones I haven't thought of) *may* all be resolvable. But this cer=
tainly complicates the already complex O/A process - maybe to the point tha=
t many get it wrong. (I don't want to write rfc6337bis!)
<br>
<br>
My concern with the glare resolution algorithm is that for it to work both =
sides must realize that glare occurred. And I think there are common sequen=
ces when one side won't realize it.
<o:p></o:p></p>
<p class=3D"MsoNormal"><br>
I'm not saying I didn't miss it, but I did work through as many permutation=
s as I could think of, and none of them are left out by the algorithm descr=
ibed in section 3.4.1.<br>
<br>
That said, I did my analysis in the context of WebRTC, and assumed a reliab=
le, in-order delivery channel for messages. I recognize that we don't have =
that for SIP, and that messages in the same direction can be reordered. I a=
dmit that this does raise some interesting
 scenarios that may require additional work to resolve.<br>
<br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal">I'm reluctant to buy into a scheme that depends on t=
his without knowing it it will be implementable.
<o:p></o:p></p>
<p class=3D"MsoNormal"><br>
I understand your reluctance, and don't want to gloss over it. This is but =
one plank in the overall plan, and it represents what we think is the best =
(but not only) solution to this specific requirement. If the issues prove i=
ntractable, we could fall back to
 an alternate scheme, such as the designated-offerer I describe in draft-ro=
ach-rtcweb-glareless-add-00, or the token-passing scheme described by Ted H=
ardie (cf.
<a href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg07361.htm=
l">http://www.ietf.org/mail-archive/web/rtcweb/current/msg07361.html</a>).<=
br>
<br>
/a<o:p></o:p></p>
</div>
</body>
</html>

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From emil@sip-communicator.org  Sat Jul 20 06:04:42 2013
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Date: Sat, 20 Jul 2013 15:04:32 +0200
From: Emil Ivov <emcho@jitsi.org>
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Subject: Re: [MMUSIC] Some comments on draft-ivov-mmusic-multiple-sources
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Hey Bernard,

On 19.07.13, 04:28, Bernard Aboba wrote:
> Overall, I like this draft quite a bit.  Some thoughts:
>
> Section 1
>
>     Yet, Offer/Answer implementations in SIP
>     applications have most often used them as an envelope for a maximum
>     of two RTP streams (SSRCs) at a time: one in each direction.  The
>     most common reason for this has been the fact these applications
>     could not meaningfully render multiple SSRCs simultaneously.
>
> [BA] While the above is probably true of legacy O/A implementations, video
> implementations today frequently can and do render multiple SSRCs simultaneously.
> In particular, implementations supporting more than "point-to-point" calls, often
> go beyond "video switching" to rendering of multiple streams, all declared on
> the same m-line.
>
> In particular, where there are multiple layers as well as multiple sources going
> into a mixer, you may see multiple SSRCs coming out of the mixer.  This
> could happen because a single SSRC is used for all the layers from a given source
> (but distinct SSRCs are used for different sources), or because you have each
> layer from a given source using a distinct SSRC.

I agree that this is where things seem to be headed. The text was mostly 
referring to "legacy" SIP devices. Most of which are probably unable to 
handle video at all. Many of the SIP hardphones that do handle video 
simply don't have the user interface that would allow showing a second 
flow. From what I've seen such devices often simply ignore the SSRC 
field which leads to weird results with multiple video flows.

> Section 2
>
>     In cases where endpoints need to be able to detect this from the SDP
>     Offer/Answer they could use the "max-send-ssrc" and "max-recv-ssrc"
>     attributes defined in [MAX-SSRC  <http://tools.ietf.org/html/draft-ivov-mmusic-multiple-sources-00#ref-MAX-SSRC>].  It has to be noted however that
>     this mechanism is still a work in progress and as such it would still
>     need to be extended to provide ways of distinguishing between
>     independent flows and complementary ones such as layered FEC and RTX.
>
> [BA] Keep in mind that if you don't use a single SSRC per source, then the semantics
> of max-send-ssrc and max-recv-ssrc become ambiguous.  So you need to be able to
> express the maximum send and receive sources as well as the maximum sent and
> received layers.

Completely agreed. That's what the text above is actually trying to say. 
MAX-SSRC may need further syntax/semantics or at least text that would 
properly address the difference between multiple independent streams 
(e.g. a video conf call) from multiple related streams (e.g. layers or 
FEC ).

> Section 5
>
>     Another approach could be a combination of RTCP and RTP header
>     extensions [RFC5285  <http://tools.ietf.org/html/rfc5285>] in a way similar to the one employed by the
>     Rapid Synchronisation of RTP Flows [RFC6051  <http://tools.ietf.org/html/rfc6051>].  While such a mechanism
>     is not currently defined by the IETF, specifying it could be
>     relatively straightforward:
>
> [BA] Not sure about the "straightforward" part but there are advantages to using
> RTP and RTCP extensions.  RTP extensions allow RTP parsing to be bypassed which
> is convenient if SRTP is used (since otherwise unless SSRCs are pre-declared
> you might have to decrypt and parse the RTP packet to figure out what layer
> it corresponds to).  Today there are commonly used RTP configurations that can't
> easily be expressed in SDP (e.g. sending individual codec layers with distinct
> SSRCs, but within the same RTP session) but that an RTP extension could help
> with.  The argument on the other side is that SDP O/A is reliable, whereas
> an RTP extension or RTCP SDES would not be.  However, if the endgame is WebRTC
> without SDP, then I think we need to look for RTP/RTCP solutions for these issues,
> rather than requiring an SDP dependency.

Agreed.

Emil

-- 
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [MMUSIC] Unified Plan for SDP Handling
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On 7/19/13 5:43 AM, Justin Uberti wrote:=0A=
>=0A=
>=0A=
>=0A=
> On Wed, Jul 17, 2013 at 4:23 PM, Adam Roach <adam@nostrum.com=0A=
> <mailto:adam@nostrum.com>> wrote:=0A=
>=0A=
>     On 7/17/13 09:08, Stefan H=E5kansson LK wrote:=0A=
>=0A=
>         I have a couple of specific questions/comments, coming from a ver=
y=0A=
>         WebRTC centric (sorry for that) perspective:=0A=
>=0A=
>         * Simulcast: In the current API model, MediaStreamTrack is about =
the=0A=
>         lowest you can get to. And the same source (camera, microphone) c=
an=0A=
>         source several MediaStreamTrack - and each of those can use=0A=
>         individual=0A=
>         constraints for width, height, framerate, etc. So in a WebRTC=0A=
>         context,=0A=
>         there seem to me we would not need the simulcast solution you hav=
e=0A=
>         included. The different resolutions can be sent as different=0A=
>         MediaStreamTrack's - meaning they would be separate m-lines.=0A=
>=0A=
>=0A=
>     That's what we meant to convey with this (section 2, bullet 1):=0A=
>     "This does not preclude 'application level' simulcasting; i.e., the=
=0A=
>     creation of multiple media stream tracks from a single source."=0A=
>=0A=
>     It's true that we don't broach the issue of how to indicate the need=
=0A=
>     for simulcast via a WebRTC javascript API. I think this is something=
=0A=
>     that would be future work, probably in the W3C. I'm going to leave=0A=
>     any further explanation to Justin, as most of the simulcast ideas in=
=0A=
>     the document came from him.=0A=
>=0A=
>=0A=
> Consider the parallel with layered coding - in that case, you have=0A=
> multiple dependent encodings of the track, but it's doubtful you want=0A=
> the recipients to end up with multiple MediaStreamTracks and display=0A=
> them all in their UI.=0A=
=0A=
I think it is not a good parallel. With layered encoding you need the =0A=
base layer to have any use of any enhancement layer. With simulcast of =0A=
the other hand every stream is a complete representation and can be used =
=0A=
on its own.=0A=
=0A=
>=0A=
> This mechanism is basically a way to say that I have a single=0A=
> MediaStreamTrack coming in, and while there might be multiple=0A=
> independent or dependent encodings of it on the wire, the recipient=0A=
> should only create and display a single MediaStreamTrack on its side.=0A=
=0A=
That's fine. What I was saying was that in a WebRTC context we don't =0A=
really need this functionality for simulcast, and using it would require =
=0A=
a new set of API functionality. But I understand Adam's response as that =
=0A=
simulcast can be solved using 'application level' simulcasting, so there =
=0A=
is really no issue.=0A=
=0A=
>=0A=
>=0A=
>=0A=
>         * Direction attribute: The current WebRTC API is "sender"=0A=
>         oriented: the=0A=
>         application can select MediaStream(Track)'s to send. Would it=0A=
>         for WebRTC=0A=
>         make sense to have all m-lines being unidirectional (send/recvonl=
y)?=0A=
>=0A=
>=0A=
>     I understand how this makes things cleaner in general. The issue I=0A=
>     have is that it interacts poorly with legacy uses.=0A=
>=0A=
>=0A=
>            (If=0A=
>         they should be sendrecv we also need to specify when a return tra=
ck=0A=
>         should reuse an existing m-line, and when a new m-line should be=
=0A=
>         created.)=0A=
>=0A=
>=0A=
>     I agree that this could stand to be spelled out more clearly. I do=0A=
>     not think that doing so will prove particularly controversial.=0A=
>=0A=
>=0A=
>     /a=0A=
>     _________________________________________________=0A=
>     mmusic mailing list=0A=
>     mmusic@ietf.org <mailto:mmusic@ietf.org>=0A=
>     https://www.ietf.org/mailman/__listinfo/mmusic=0A=
>     <https://www.ietf.org/mailman/listinfo/mmusic>=0A=
>=0A=
>=0A=
=0A=

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Emil Ivov <emcho@jitsi.org>, "Stach, Thomas" <thomas.stach@siemens-enterprise.com>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
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Date: Sun, 21 Jul 2013 17:49:32 +0000
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hi,

As I've been on vacation, I tried to go through this thread in one sweep. I=
 may have missed/misunderstood some parts, and my apologies if my comments =
below have already been dealt with.

First, regarding 180, no matter if we use PRACK or not, SIP requires the SD=
P to be identical in all 180 responses. That means you can't add new candid=
ates in subsequent 180 responses (eventhough vanilla ICE uses 180s without =
PRACK, the SDP is identical in all 180s, AFAIR).

Second, I get a little confused when we talk about the UAS sending INFO bef=
ore it knows that the dialog is established, and that we would define some =
new handling rules when an error response is received. Now, IF we want the =
UAC to inform the UAS that the dialog has been established, we don't need P=
RACK for that. The UAC could send an INFO when it receives the 180, and whe=
n the UAS receives the INFO it knows the dialog has been established. I hav=
en't thought so much about it, so I am not suggesting such mechanism at thi=
s point, I am only saying that it would be a possible solution from a SIP p=
rotocol perspective :)

Regards,

Christer


From emil@sip-communicator.org  Sun Jul 21 11:57:21 2013
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hey Christer,

On 21.07.13, 19:49, Christer Holmberg wrote:
> Hi,
>
> As I've been on vacation,

Hope you had a good one!

> I tried to go through this thread in one
> sweep. I may have missed/misunderstood some parts, and my apologies
> if my comments below have already been dealt with.
>
> First, regarding 180, no matter if we use PRACK or not, SIP requires
> the SDP to be identical in all 180 responses. That means you can't
> add new candidates in subsequent 180 responses (eventhough vanilla
> ICE uses 180s without PRACK, the SDP is identical in all 180s,
> AFAIR).

Good point!

> Second, I get a little confused when we talk about the UAS sending
> INFO before it knows that the dialog is established, and that we
> would define some new handling rules when an error response is
> received. Now, IF we want the UAC to inform the UAS that the dialog
> has been established, we don't need PRACK for that. The UAC could
> send an INFO when it receives the 180, and when the UAS receives the
> INFO it knows the dialog has been established. I haven't thought so
> much about it, so I am not suggesting such mechanism at this point, I
> am only saying that it would be a possible solution from a SIP
> protocol perspective :)

This could indeed work! It would also make a lot of sense in cases where 
both agents are doing full trickle.

Of course it does mean that, especially in half trickle scenarios, we 
basically have one full end-to-end signalling RTT (180 + INFO) during 
which trickling will be completely blocked. This could take a while.

Still, if we don't come up with a better alternative, the worst case 
(i.e. half trickle + slow signalling) shouldn't be much worse than 
vanilla ICE.

Emil

-- 
https://jitsi.org

From pkyzivat@alum.mit.edu  Sun Jul 21 14:41:05 2013
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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How is this better than PRACK?

	Thanks,
	Paul

On 7/21/13 2:56 PM, Emil Ivov wrote:
> Hey Christer,
>
> On 21.07.13, 19:49, Christer Holmberg wrote:
>> Hi,
>>
>> As I've been on vacation,
>
> Hope you had a good one!
>
>> I tried to go through this thread in one
>> sweep. I may have missed/misunderstood some parts, and my apologies
>> if my comments below have already been dealt with.
>>
>> First, regarding 180, no matter if we use PRACK or not, SIP requires
>> the SDP to be identical in all 180 responses. That means you can't
>> add new candidates in subsequent 180 responses (eventhough vanilla
>> ICE uses 180s without PRACK, the SDP is identical in all 180s,
>> AFAIR).
>
> Good point!
>
>> Second, I get a little confused when we talk about the UAS sending
>> INFO before it knows that the dialog is established, and that we
>> would define some new handling rules when an error response is
>> received. Now, IF we want the UAC to inform the UAS that the dialog
>> has been established, we don't need PRACK for that. The UAC could
>> send an INFO when it receives the 180, and when the UAS receives the
>> INFO it knows the dialog has been established. I haven't thought so
>> much about it, so I am not suggesting such mechanism at this point, I
>> am only saying that it would be a possible solution from a SIP
>> protocol perspective :)
>
> This could indeed work! It would also make a lot of sense in cases where
> both agents are doing full trickle.
>
> Of course it does mean that, especially in half trickle scenarios, we
> basically have one full end-to-end signalling RTT (180 + INFO) during
> which trickling will be completely blocked. This could take a while.
>
> Still, if we don't come up with a better alternative, the worst case
> (i.e. half trickle + slow signalling) shouldn't be much worse than
> vanilla ICE.
>
> Emil
>


From emil@sip-communicator.org  Sun Jul 21 15:06:58 2013
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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On 21.07.13, 23:40, Paul Kyzivat wrote:
> How is this better than PRACK?

Well, you don't have to worry about supporting offers and answers in the 
PRACK (which could indeed be a pain).

Emil

>
> 	Thanks,
> 	Paul
>
> On 7/21/13 2:56 PM, Emil Ivov wrote:
>> Hey Christer,
>>
>> On 21.07.13, 19:49, Christer Holmberg wrote:
>>> Hi,
>>>
>>> As I've been on vacation,
>>
>> Hope you had a good one!
>>
>>> I tried to go through this thread in one
>>> sweep. I may have missed/misunderstood some parts, and my apologies
>>> if my comments below have already been dealt with.
>>>
>>> First, regarding 180, no matter if we use PRACK or not, SIP requires
>>> the SDP to be identical in all 180 responses. That means you can't
>>> add new candidates in subsequent 180 responses (eventhough vanilla
>>> ICE uses 180s without PRACK, the SDP is identical in all 180s,
>>> AFAIR).
>>
>> Good point!
>>
>>> Second, I get a little confused when we talk about the UAS sending
>>> INFO before it knows that the dialog is established, and that we
>>> would define some new handling rules when an error response is
>>> received. Now, IF we want the UAC to inform the UAS that the dialog
>>> has been established, we don't need PRACK for that. The UAC could
>>> send an INFO when it receives the 180, and when the UAS receives the
>>> INFO it knows the dialog has been established. I haven't thought so
>>> much about it, so I am not suggesting such mechanism at this point, I
>>> am only saying that it would be a possible solution from a SIP
>>> protocol perspective :)
>>
>> This could indeed work! It would also make a lot of sense in cases where
>> both agents are doing full trickle.
>>
>> Of course it does mean that, especially in half trickle scenarios, we
>> basically have one full end-to-end signalling RTT (180 + INFO) during
>> which trickling will be completely blocked. This could take a while.
>>
>> Still, if we don't come up with a better alternative, the worst case
>> (i.e. half trickle + slow signalling) shouldn't be much worse than
>> vanilla ICE.
>>
>> Emil
>>
>
>

-- 
https://jitsi.org

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From: "Avasarala, Ranjit (NSN - IN/Bangalore)" <ranjit.avasarala@nsn.com>
To: "ext Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>, "aurelien.sollaud@orange.com" <aurelien.sollaud@orange.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
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Subject: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
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Hi
I agree of not making these parameters mandatory.=20

Regards
Ranjit



-----Original Message-----
From: ext Muthu Arul Mozhi Perumal (mperumal) [mailto:mperumal@cisco.com]=20
Sent: Monday, July 15, 2013 7:59 PM
To: aurelien.sollaud@orange.com; Avasarala, Ranjit (NSN - IN/Bangalore); mm=
usic@ietf.org
Subject: RE: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt

Agree with Aurelien. In addition, the draft explicitly specifies the behavi=
or for the default case:

   When the offer or answer has G723 and the annexa parameter is absent,
   the offerer or answerer knows that it has implied the default
   "annexa=3Dyes".  This is because the annexa attribute is part of the
   original registration of audio/G723 [RFC4856].  All implementations
   that support G723 understand the annexa attribute.  Hence, this case
   MUST be considered as if the offer or answer has G723 with
   annexa=3Dyes.

This I think should suffice to make sure implementations don't ignore it. S=
o, making these optional parameters mandatory optional isn't needed (and pr=
obably isn't right).

Muthu

|-----Original Message-----
|From: aurelien.sollaud@orange.com [mailto:aurelien.sollaud@orange.com]
|Sent: Monday, July 15, 2013 2:48 PM
|To: Avasarala, Ranjit (NSN - IN/Bangalore); Muthu Arul Mozhi Perumal (mper=
umal); mmusic@ietf.org
|Subject: RE: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
|
|Hi
|
|A lot of media subtypes have optional parameters, and as long as a default=
 value is defined, there is
|no issue.
|The problem here is not that annexa/annexb parameters are optional, but th=
at the offer/answer
|procedure was not defined for these media subtypes.
|
|-----Message d'origine-----
|De=A0: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] De la part=
 de Avasarala, Ranjit (NSN -
|IN/Bangalore)
|Envoy=E9=A0: lundi 15 juillet 2013 09:08
|=C0=A0: ext Muthu Arul Mozhi Perumal (mperumal); mmusic@ietf.org
|Objet=A0: Re: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
|
|Hi Muthu
|
|I have following comments
|1. RFC 4856 defines "annexa" and "annexb" as optional parameters. So I agr=
ee there could be
|misinpterpretations.
|2. The proposed draft should make these parameters as mandatory, otherwise=
 the implementations would
|continue to ignore the parameters.
|3. Update RFC 4856 moving "annexa" and "annexb" parameters from "optional"=
 to "required".
|
|Regards
|Ranjit
|
|
|
|-----Original Message-----
|From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf O=
f ext Muthu Arul Mozhi
|Perumal (mperumal)
|Sent: Monday, July 15, 2013 10:21 AM
|To: mmusic@ietf.org
|Subject: [MMUSIC] FW: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
|
|This updated version addresses all comments received during WGLC.
|
|- Muthu, Partha
|
|-----Original Message-----
|From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org]=
 On Behalf Of internet-
|drafts@ietf.org
|Sent: Monday, July 15, 2013 10:13 AM
|To: i-d-announce@ietf.org
|Cc: mmusic@ietf.org
|Subject: I-D Action: draft-ietf-mmusic-sdp-g723-g729-04.txt
|
|A New Internet-Draft is available from the on-line Internet-Drafts directo=
ries.
| This draft is a work item of the Multiparty Multimedia Session Control Wo=
rking Group of the IETF.
|
|	Title           : Offer/Answer Considerations for G723 Annex A and G729 A=
nnex B
|	Author(s)       : Muthu Arul Mozhi Perumal
|                          Parthasarathi Ravindran
|	Filename        : draft-ietf-mmusic-sdp-g723-g729-04.txt
|	Pages           : 8
|	Date            : 2013-07-14
|
|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 Session Description
|   protocol(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-g729-04
|
|A diff from the previous version is available at:
|http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-g723-g729-04
|
|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.i=
etf.org/ietf/1shadow-
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From christer.holmberg@ericsson.com  Sun Jul 21 23:24:28 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, Emil Ivov <emcho@jitsi.org>
Thread-Topic: VS: [MMUSIC] Trickle ICE for SIP Questions
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hi,

>How is this better than PRACK?

I didn't say it's better :)

Regards,

Christer



On 7/21/13 2:56 PM, Emil Ivov wrote:
> Hey Christer,
>
> On 21.07.13, 19:49, Christer Holmberg wrote:
>> Hi,
>>
>> As I've been on vacation,
>
> Hope you had a good one!
>
>> I tried to go through this thread in one sweep. I may have=20
>> missed/misunderstood some parts, and my apologies if my comments=20
>> below have already been dealt with.
>>
>> First, regarding 180, no matter if we use PRACK or not, SIP requires=20
>> the SDP to be identical in all 180 responses. That means you can't=20
>> add new candidates in subsequent 180 responses (eventhough vanilla=20
>> ICE uses 180s without PRACK, the SDP is identical in all 180s,=20
>> AFAIR).
>
> Good point!
>
>> Second, I get a little confused when we talk about the UAS sending=20
>> INFO before it knows that the dialog is established, and that we=20
>> would define some new handling rules when an error response is=20
>> received. Now, IF we want the UAC to inform the UAS that the dialog=20
>> has been established, we don't need PRACK for that. The UAC could=20
>> send an INFO when it receives the 180, and when the UAS receives the=20
>> INFO it knows the dialog has been established. I haven't thought so=20
>> much about it, so I am not suggesting such mechanism at this point, I=20
>> am only saying that it would be a possible solution from a SIP=20
>> protocol perspective :)
>
> This could indeed work! It would also make a lot of sense in cases=20
> where both agents are doing full trickle.
>
> Of course it does mean that, especially in half trickle scenarios, we=20
> basically have one full end-to-end signalling RTT (180 + INFO) during=20
> which trickling will be completely blocked. This could take a while.
>
> Still, if we don't come up with a better alternative, the worst case=20
> (i.e. half trickle + slow signalling) shouldn't be much worse than=20
> vanilla ICE.
>
> Emil
>


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Emil Ivov <emcho@jitsi.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: VS: [MMUSIC] Trickle ICE for SIP Questions
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Date: Mon, 22 Jul 2013 06:33:18 +0000
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hi,

>> How is this better than PRACK?
>
> Well, you don't have to worry about supporting offers and answers in the =
PRACK (which could indeed be a pain).

In my years of "PRACK experience", PRACK itself (for the purpose of acknowl=
edging reliable 18x responses) has never been a problem. But, I agree there=
 could be issues when it is used for O/A - especially if one wants to rejec=
t an offer. In my experience, however, offers in PRACK are normally used e.=
g. to change pre-condition state, and/or to remove codecs, so the likelihoo=
d of rejection based on the SDP content should be rather small.

There are also implementations that don't accept offers in PRACK to begin w=
ith, but I think that goes under "bad implementation" :)

Having said that, not specific to trickle ICE, if O/A is a show stopper for=
 the community to use PRACK, maybe that is something we need to look into o=
n a more general level.

Regards,

Christer




> On 7/21/13 2:56 PM, Emil Ivov wrote:
>> Hey Christer,
>>
>> On 21.07.13, 19:49, Christer Holmberg wrote:
>>> Hi,
>>>
>>> As I've been on vacation,
>>
>> Hope you had a good one!
>>
>>> I tried to go through this thread in one sweep. I may have=20
>>> missed/misunderstood some parts, and my apologies if my comments=20
>>> below have already been dealt with.
>>>
>>> First, regarding 180, no matter if we use PRACK or not, SIP requires=20
>>> the SDP to be identical in all 180 responses. That means you can't=20
>>> add new candidates in subsequent 180 responses (eventhough vanilla=20
>>> ICE uses 180s without PRACK, the SDP is identical in all 180s,=20
>>> AFAIR).
>>
>> Good point!
>>
>>> Second, I get a little confused when we talk about the UAS sending=20
>>> INFO before it knows that the dialog is established, and that we=20
>>> would define some new handling rules when an error response is=20
>>> received. Now, IF we want the UAC to inform the UAS that the dialog=20
>>> has been established, we don't need PRACK for that. The UAC could=20
>>> send an INFO when it receives the 180, and when the UAS receives the=20
>>> INFO it knows the dialog has been established. I haven't thought so=20
>>> much about it, so I am not suggesting such mechanism at this point,=20
>>> I am only saying that it would be a possible solution from a SIP=20
>>> protocol perspective :)
>>
>> This could indeed work! It would also make a lot of sense in cases=20
>> where both agents are doing full trickle.
>>
>> Of course it does mean that, especially in half trickle scenarios, we=20
>> basically have one full end-to-end signalling RTT (180 + INFO) during=20
>> which trickling will be completely blocked. This could take a while.
>>
>> Still, if we don't come up with a better alternative, the worst case=20
>> (i.e. half trickle + slow signalling) shouldn't be much worse than=20
>> vanilla ICE.
>>
>> Emil
>>
>
>

--
https://jitsi.org

From emil@sip-communicator.org  Mon Jul 22 03:44:47 2013
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Date: Mon, 22 Jul 2013 12:40:36 +0200
From: Emil Ivov <emcho@jitsi.org>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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On 22.07.13, 08:33, Christer Holmberg wrote:
> Hi,
>
>>> How is this better than PRACK?
>>
>> Well, you don't have to worry about supporting offers and answers in the PRACK (which could indeed be a pain).
>
> In my years of "PRACK experience", PRACK itself (for the purpose of acknowledging reliable 18x responses) has never been a problem. But, I agree there could be issues when it is used for O/A - especially if one wants to reject an offer. In my experience, however, offers in PRACK are normally used e.g. to change pre-condition state, and/or to remove codecs, so the likelihood of rejection based on the SDP content should be rather small.
>
> There are also implementations that don't accept offers in PRACK to begin with, but I think that goes under "bad implementation" :)
>
> Having said that, not specific to trickle ICE, if O/A is a show stopper for the community to use PRACK,

Well, speaking as an implementor, I can completely see how many would 
feel uneasy about adding this to working implementations. It basically 
means adding some additional complexity with no actual benefit in return.

> maybe that is something we need to look into on a more general level.

You mean something like PRACK-lite that doesn't have offer answer? This 
sounds like something that could be relatively straightforward to 
specify. It would basically have to mimic PRACK but remove the part that 
talks about the additional O/A possibilities.

Trickle ICE for SIP could then have a dependency on it ...

Emil
>
> Regards,
>
> Christer
>
>
>
>
>> On 7/21/13 2:56 PM, Emil Ivov wrote:
>>> Hey Christer,
>>>
>>> On 21.07.13, 19:49, Christer Holmberg wrote:
>>>> Hi,
>>>>
>>>> As I've been on vacation,
>>>
>>> Hope you had a good one!
>>>
>>>> I tried to go through this thread in one sweep. I may have
>>>> missed/misunderstood some parts, and my apologies if my comments
>>>> below have already been dealt with.
>>>>
>>>> First, regarding 180, no matter if we use PRACK or not, SIP requires
>>>> the SDP to be identical in all 180 responses. That means you can't
>>>> add new candidates in subsequent 180 responses (eventhough vanilla
>>>> ICE uses 180s without PRACK, the SDP is identical in all 180s,
>>>> AFAIR).
>>>
>>> Good point!
>>>
>>>> Second, I get a little confused when we talk about the UAS sending
>>>> INFO before it knows that the dialog is established, and that we
>>>> would define some new handling rules when an error response is
>>>> received. Now, IF we want the UAC to inform the UAS that the dialog
>>>> has been established, we don't need PRACK for that. The UAC could
>>>> send an INFO when it receives the 180, and when the UAS receives the
>>>> INFO it knows the dialog has been established. I haven't thought so
>>>> much about it, so I am not suggesting such mechanism at this point,
>>>> I am only saying that it would be a possible solution from a SIP
>>>> protocol perspective :)
>>>
>>> This could indeed work! It would also make a lot of sense in cases
>>> where both agents are doing full trickle.
>>>
>>> Of course it does mean that, especially in half trickle scenarios, we
>>> basically have one full end-to-end signalling RTT (180 + INFO) during
>>> which trickling will be completely blocked. This could take a while.
>>>
>>> Still, if we don't come up with a better alternative, the worst case
>>> (i.e. half trickle + slow signalling) shouldn't be much worse than
>>> vanilla ICE.
>>>
>>> Emil
>>>
>>
>>
>
> --
> https://jitsi.org
>

-- 
https://jitsi.org

From christer.holmberg@ericsson.com  Mon Jul 22 04:11:56 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: VS: VS: [MMUSIC] Trickle ICE for SIP Questions
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hi,

>>>> How is this better than PRACK?
>>>
>>> Well, you don't have to worry about supporting offers and answers in th=
e PRACK (which could indeed be a pain).
>>
>> In my years of "PRACK experience", PRACK itself (for the purpose of ackn=
owledging reliable 18x responses) has never been a=20
>> problem. But, I agree there could be issues when it is used for O/A - es=
pecially if one wants to reject an offer. In my experience, however,=20
>> offers in PRACK are normally used e.g. to change pre-condition state, an=
d/or to remove codecs, so the likelihood of rejection based on the SDP cont=
ent should be rather small.
>>
>> There are also implementations that don't accept offers in PRACK to=20
>> begin with, but I think that goes under "bad implementation" :)
>>
>> Having said that, not specific to trickle ICE, if O/A is a show=20
>> stopper for the community to use PRACK,
>
> Well, speaking as an implementor, I can completely see how many would fee=
l uneasy about adding this to working implementations. It basically means a=
dding some additional complexity with no actual benefit in return.
>
>> maybe that is something we need to look into on a more general level.
>
> You mean something like PRACK-lite that doesn't have offer answer? This s=
ounds like something that could be relatively straightforward to specify. I=
t would basically=20
> have to mimic PRACK but remove the part that talks about the additional O=
/A possibilities.

It could perhaps even be a request without a response, similar to ACK, whic=
h would make it even more "lite". When the UAS receives it, it simply cease=
s the transmission of the 18x.

Regards,

Christer

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Subject: Re: [MMUSIC] [rtcweb] Unified Plan for SDP Handling - Glare
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On 7/20/13 04:51, Christer Holmberg wrote:
> I do realize that the draft does not represent the complete glare 
> solution.

Right, and I think it would be premature to dig into those details for 
their own sake at the moment.  So, before we go into a discussion of 
what I believe are minutiea, we need to establish whether you're asking 
because it seems like an interesting problem to work on, or if you think 
that the approach has intractable flaws that would prevent developing it 
to completion. (To be clear, even though it's not phrased as a question, 
I would expect you to indicate which of those is the case, Christer).

In the former case, I'd like to defer the discussion around what I 
believe are addressable open issues in the high-level sketch, and 
instead focus on whether the overall "partial o/a" approach satisfies 
the needs of those parties for whom glare reduction is an important goal.

> *Q1*: How essential part of the unified plan is having **A** working 
> "partial o/a" solution? TECHNICALLY, the other parts of the unified 
> plan do not depend on having a "partial o/a" solution, but I would 
> like to know whether people (specially the draft authors) would be 
> willing to move ahead with unified plan even if we would not manage to 
> specify a working "partial o/a" solution.
>

Let's back up -- you're conflating mechanism with requirement here. The 
requirement is no glare in certain use cases (adding/removing streams). 
The proposed mechanism is partial o/a. The requirement is important to 
certain key participants in the working group, so I would characterize 
having a mechanism to address it as essential.

Keeping that in mind, "having a working 'partial o/a' solution," to 
re-use your phrasing, is utterly nonessential, since it is only one way 
to skin this cat. However, like I just said, having a working story for 
glare *is* essential. We've seen others (as I explained to my response 
to Paul, earlier), and may be able to come up with yet more.

> *Q2a*: When sending a partial offer, the draft says that only the m= 
> line explicitly signaled is affected. Note, however, that you will 
> still signal information (e.g. the SDP version value, the group:BUNDLE 
> attribute etc) which affect the whole SDP, and if you have multiple 
> outstanding partial offers (for different m= lines) that can cause issues.
>
> *Q2b*: In addition, when using BUNDLE, modifying an m= line might 
> implicitly modify other m= lines, and there are rules (also indicated 
> in the draft) mandating identical SDP attribute values etc. So, if you 
> have multiple outstanding partial offers, you may end up with all kind 
> of different "BUNDLE collisions".
>
> *Q3*: If I send a partial offer, for a given m= line, can the answer 
> only contain that given m= line? What if the partial offer causes 
> other m= lines to be modified?
>

These three points are, as far as I can discern, the natural engineers' 
curiosity that leads us to want to start working on fleshing out a 
solution as soon as we understand its skeleton. I don't think it's time 
for that. I have answers for these questions, but I'm 100% certain that 
providing them would start a highly distracting, if interesting, 
conversation that would delay our nailing down the other aspects of the 
unified plan.

/a

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    <div class="moz-cite-prefix">On 7/20/13 04:51, Christer Holmberg
      wrote:<br>
    </div>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C3D918D@ESESSMB209.ericsson.se"
      type="cite">
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      <div class="WordSection1"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
          lang="EN-US">I do realize that the draft does not represent
          the complete glare solution.
        </span></div>
    </blockquote>
    <br>
    Right, and I think it would be premature to dig into those details
    for their own sake at the moment.&nbsp; So, before we go into a
    discussion of what I believe are minutiea, we need to establish
    whether you're asking because it seems like an interesting problem
    to work on, or if you think that the approach has intractable flaws
    that would prevent developing it to completion. (To be clear, even
    though it's not phrased as a question, I would expect you to
    indicate which of those is the case, Christer).<br>
    <br>
    In the former case, I'd like to defer the discussion around what I
    believe are addressable open issues in the high-level sketch, and
    instead focus on whether the overall "partial o/a" approach
    satisfies the needs of those parties for whom glare reduction is an
    important goal.<br>
    <br>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C3D918D@ESESSMB209.ericsson.se"
      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"
            lang="EN-US"><o:p>&nbsp;</o:p></span><b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">Q1</span></b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">: How essential part of the unified plan is
            having *<b>A</b>* working &#8220;partial o/a&#8221; solution?
            TECHNICALLY, the other parts of the unified plan do not
            depend on having a &#8220;partial o/a&#8221; solution, but I would like
            to know whether people (specially the draft authors) would
            be willing to move ahead with unified plan even if we would
            not manage to specify a working &#8220;partial o/a&#8221; solution.</span></p>
      </div>
    </blockquote>
    <br>
    Let's back up -- you're conflating mechanism with requirement here.
    The requirement is no glare in certain use cases (adding/removing
    streams). The proposed mechanism is partial o/a. The requirement is
    important to certain key participants in the working group, so I
    would characterize having a mechanism to address it as essential.<br>
    <br>
    Keeping that in mind, "having a working 'partial o/a' solution," to
    re-use your phrasing, is utterly nonessential, since it is only one
    way to skin this cat. However, like I just said, having a working
    story for glare *is* essential. We've seen others (as I explained to
    my response to Paul, earlier), and may be able to come up with yet
    more.<br>
    <br>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C3D918D@ESESSMB209.ericsson.se"
      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"
            lang="EN-US"><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"
            lang="EN-US"><o:p></o:p></span><b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">Q2a</span></b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">: When sending a partial offer, the draft says
            that only the m= line explicitly signaled is affected. Note,
            however, that you will still signal information (e.g. the
            SDP version value, the group:BUNDLE attribute etc) which
            affect the whole SDP, and if you have multiple outstanding
            partial offers (for different m= lines) that can cause
            issues.<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"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">Q2b</span></b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">: In addition, when using BUNDLE, modifying an
            m= line might implicitly modify other m= lines, and there
            are rules (also indicated in the draft) mandating identical
            SDP attribute values etc. So, if you have multiple
            outstanding partial offers, you may end up with all kind of
            different &#8220;BUNDLE collisions&#8221;.<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"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
              lang="EN-US">Q3</span></b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">: If I send a partial offer, for a given m=
            line, can the answer only contain that given m= line? What
            if the partial offer causes other m= lines to be modified?<o:p></o:p></span></p>
        <span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
          lang="EN-US"><o:p> </o:p></span></div>
    </blockquote>
    <br>
    These three points are, as far as I can discern, the natural
    engineers' curiosity that leads us to want to start working on
    fleshing out a solution as soon as we understand its skeleton. I
    don't think it's time for that. I have answers for these questions,
    but I'm 100% certain that providing them would start a highly
    distracting, if interesting, conversation that would delay our
    nailing down the other aspects of the unified plan.<br>
    <br>
    /a<br>
  </body>
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From adam@nostrum.com  Mon Jul 22 15:08:34 2013
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Subject: Re: [MMUSIC] Unified Plan for SDP Handling - multiple media sources
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On 7/16/13 23:24, Roni Even wrote:
> Hi,
>>> 4. in section 3.2.1.1 "as well as a 16-bit identifier (expressed as an
>>> integer) used for correlation;" I think that it must be a byte string
>>> or token not integer
>> Do you have some rationale for your assertion?
>>
>>
> [Roni Even] This is the syntax of the header extension and
> extensionattributes in RFC5285. It is a byte string and not an integer
>
>

Forgive the imprecision in the draft. What we mean to say is something 
along the lines of "a 16-bit identifier, expressed as an integer in the 
SDP, encoded in the RTP header as a two-byte byte string, in network 
byte order."

I apologize for the confusion. I thought this much would be obvious 
without needing to spell it out explicitly.

/a

From christer.holmberg@ericsson.com  Mon Jul 22 21:53:01 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>
Thread-Topic: VS: [MMUSIC] [rtcweb] Unified Plan for SDP Handling - Glare
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] [rtcweb] Unified Plan for SDP Handling - Glare
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Hi,

>> I do realize that the draft does not represent the complete glare soluti=
on.=20
>
> Right, and I think it would be premature to dig into those details for th=
eir own sake at the moment.=A0 So, before we go into a discussion of what I=
 believe are minutiea, we=20
> need to establish whether you're asking because it seems like an interest=
ing problem to work on, or if you think that the approach has intractable f=
laws that would prevent=20
> developing it to completion. (To be clear, even though it's not phrased a=
s a question, I would expect you to indicate which of those is the case, Ch=
rister).

I have always thought that one of the downsides with SDP O/A is the fact th=
at you always need to include the information for all streams when you send=
 an Offer or Answer. Now, when you have 1 or 2 streams that is not a big is=
sue. But, in the rtcweb context people have been talking about a much bigge=
r number of streams, so it may become an issue.

So, I definitely think it seems like an interesting problem to work on.

I do think that there are issues with 'partial O/A'. But, as we both have s=
aid, this is not the moment to go into detailed discussions about the mecha=
nism :)

> In the former case, I'd like to defer the discussion around what I believ=
e are addressable open issues in the high-level sketch, and instead focus o=
n=20
> whether the overall "partial o/a" approach satisfies the needs of those p=
arties for whom glare reduction is an important goal.
>
>> Q1: How essential part of the unified plan is having *A* working "partia=
l o/a" solution? TECHNICALLY, the other parts of the unified plan do not de=
pend=20
>> on having a "partial o/a" solution, but I would like to know whether peo=
ple (specially the draft authors) would be willing to move ahead with unifi=
ed plan
>> even if we would not manage to specify a working "partial o/a" solution.
>
> Let's back up -- you're conflating mechanism with requirement here. The r=
equirement is no glare in certain use cases (adding/removing streams).

The draft also talks about modifying streams (section 3.4.2). Are there spe=
cific modification use cases where no glare is important, or is it about be=
ing able to modify streams in general and avoid glare?

> The proposed mechanism is partial o/a.

Correct.

> The requirement is important to certain key participants in the working g=
roup, so I would characterize having a mechanism to address it as essential=
.

Ok.

> Keeping that in mind, "having a working 'partial o/a' solution," to re-us=
e your phrasing, is utterly nonessential, since it is only one way to skin =
this cat. However, like
> I just said, having a working story for glare *is* essential. We've seen =
others (as I explained to my response to Paul, earlier), and may be able to=
 come up with yet more.

So, moving ahead with unified plan basically means adopting the requirement=
 to solve glare, but we may also look at other mechanisms?

Thanks for the clarification!

Regards,

Christer


From adam@nostrum.com  Mon Jul 22 22:21:18 2013
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] [rtcweb] Unified Plan for SDP Handling - Glare
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On 7/22/13 23:52, Christer Holmberg wrote:

> The draft also talks about modifying streams (section 3.4.2). Are there specific modification use cases where no glare is important, or is it about being able to modify streams in general and avoid glare?

I was a matter of realizing that our design allows, with very little 
additional effort, significant reduction in the probability of glare for 
stream modification in general. As this seemed like very low hanging 
fruit, it seems to be sensible to include it as part of the design.

So, to answer you question directly: it's not driven by specific use 
cases; it's just general glare probability reduction.


> So, moving ahead with unified plan basically means adopting the 
> requirement to solve glare, but we may also look at other mechanisms?

It is my understanding that making glare impossible or nearly impossible 
for the add/remove stream case is an important requirement for any plan 
(a, b, unified, or anything else). At the moment, the proposal in the 
unified plan document is the best that the authors can come up with. I 
think we're all open to the prospect of alternate designs that achieve 
the same goal, although I would hope we don't spend too much time 
exploring the solution space.

/a

From roman@telurix.com  Tue Jul 23 09:22:21 2013
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Cc: rtcweb@ietf.org, mmusic@ietf.org, tim panton <thp@westhawk.co.uk>
Subject: [MMUSIC] Unified plan IPR
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--f46d0442808ee7904104e2303130
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Changing the title and the mailing list to the more appropriate.

The situation would be a bit clearer if patent holders were to provide the
licensing policy regarding this IPR release. Given that Ericsson is
actively involved in this working group, I think it would be reasonable to
ask them for this.

Regards,
_____________
Roman Shpount


On Tue, Jul 23, 2013 at 11:51 AM, Adam Roach <adam@nostrum.com> wrote:

>  On 7/23/13 09:20, tim panton wrote:
>
> On 23 Jul 2013, at 14:00, Stefan H=E5kansson LK <stefan.lk.hakansson@eric=
sson.com> <stefan.lk.hakansson@ericsson.com> wrote:
>
>
>  On 7/23/13 5:03 AM, Justin Uberti wrote:
>
>  With the compromise reached in the Unified Plan document,
>
>  Presumably http://www.ietf.org/id/draft-roach-mmusic-unified-plan-00.txt
>
>  I'll draw folks attention to this IPR claim
> https://datatracker.ietf.org/ipr/2141/
>
> What does that mean in practice?
>
>
>
> In practice, it has very little effect.
>
> For those of you unfamiliar with IETF IPR policy, it is documented in RFC
> 3979:
>
> http://www.ietf.org/rfc/rfc3979.txt
>
> The disclosure Tim points to is made pursuant to section 6.1.3 of that
> document. During the course of developing the unified plan, the
> applications that are mentioned in that disclosure were brought to my
> attention.
>
> The reason it has little effect in practice is that the independent claim=
s
> would appear to cover every plan proposal to date (plan a, plan b, unifie=
d
> plan, and even all of the "no plan" variations). Consequently, it does no=
t
> benefit any one approach over the others.
>
> /a
>

--f46d0442808ee7904104e2303130
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Changing the title and the mailing list to the more appropriate.<br><br>The=
 situation would be a bit clearer if patent holders were to provide the lic=
ensing policy regarding this IPR release.=A0Given that Ericsson is actively=
 involved in this working group, I think it would be reasonable to ask them=
 for this.<br>
<br>Regards,<br>
<div>_____________<br>Roman Shpount</div>
<br><br><div class=3D"gmail_quote">On Tue, Jul 23, 2013 at 11:51 AM, Adam R=
oach <span dir=3D"ltr">&lt;<a href=3D"mailto:adam@nostrum.com" target=3D"_b=
lank">adam@nostrum.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">


 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div><div>
    <div>On 7/23/13 09:20, tim panton wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <pre>On 23 Jul 2013, at 14:00, Stefan H=E5kansson LK <a href=3D"mailt=
o:stefan.lk.hakansson@ericsson.com" target=3D"_blank">&lt;stefan.lk.hakanss=
on@ericsson.com&gt;</a> wrote:

</pre>
      <blockquote type=3D"cite">
        <pre>On 7/23/13 5:03 AM, Justin Uberti wrote:
</pre>
        <blockquote type=3D"cite">
          <pre>With the compromise reached in the Unified Plan document,
</pre>
        </blockquote>
        <pre>Presumably <a href=3D"http://www.ietf.org/id/draft-roach-mmusi=
c-unified-plan-00.txt" target=3D"_blank">http://www.ietf.org/id/draft-roach=
-mmusic-unified-plan-00.txt</a>
</pre>
      </blockquote>
      <pre>I&#39;ll draw folks attention to this IPR claim

<a href=3D"https://datatracker.ietf.org/ipr/2141/" target=3D"_blank">https:=
//datatracker.ietf.org/ipr/2141/</a>

What does that mean in practice?

</pre>
    </blockquote>
    <br></div></div>
    In practice, it has very little effect.<br>
    <br>
    For those of you unfamiliar with IETF IPR policy, it is documented
    in RFC 3979:<br>
    <br>
   =20
    <a href=3D"http://www.ietf.org/rfc/rfc3979.txt" target=3D"_blank">http:=
//www.ietf.org/rfc/rfc3979.txt</a><br>
    <br>
    The disclosure Tim points to is made pursuant to section 6.1.3 of
    that document. During the course of developing the unified plan, the
    applications that are mentioned in that disclosure were brought to
    my attention.<br>
    <br>
    The reason it has little effect in practice is that the independent
    claims would appear to cover every plan proposal to date (plan a,
    plan b, unified plan, and even all of the &quot;no plan&quot; variation=
s).
    Consequently, it does not benefit any one approach over the others.<spa=
n><font color=3D"#888888"><br>
    <br>
    /a<br>
  </font></span></div>

</blockquote></div><br>

--f46d0442808ee7904104e2303130--

From adam@nostrum.com  Tue Jul 23 09:52:00 2013
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Cc: rtcweb@ietf.org, mmusic@ietf.org, tim panton <thp@westhawk.co.uk>
Subject: Re: [MMUSIC] Unified plan IPR
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On 7/23/13 11:22, Roman Shpount wrote:
> The situation would be a bit clearer if patent holders were to provide 
> the licensing policy regarding this IPR release. Given that Ericsson 
> is actively involved in this working group, I think it would be 
> reasonable to ask them for this.

If process has been properly followed, the IPR holder has already been 
notified by the IETF executive director. See 
http://www.ietf.org/rfc/rfc4879.txt (section 1 paragraph 1)

I doubt agitating for action on these mailing lists is likely to produce 
useful results.

/a

From emil@sip-communicator.org  Tue Jul 23 10:06:26 2013
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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On 22.07.13, 13:11, Christer Holmberg wrote:
> Hi,
>
>>>>> How is this better than PRACK?
>>>>
>>>> Well, you don't have to worry about supporting offers and answers in the PRACK (which could indeed be a pain).
>>>
>>> In my years of "PRACK experience", PRACK itself (for the purpose of acknowledging reliable 18x responses) has never been a
>>> problem. But, I agree there could be issues when it is used for O/A - especially if one wants to reject an offer. In my experience, however,
>>> offers in PRACK are normally used e.g. to change pre-condition state, and/or to remove codecs, so the likelihood of rejection based on the SDP content should be rather small.
>>>
>>> There are also implementations that don't accept offers in PRACK to
>>> begin with, but I think that goes under "bad implementation" :)
>>>
>>> Having said that, not specific to trickle ICE, if O/A is a show
>>> stopper for the community to use PRACK,
>>
>> Well, speaking as an implementor, I can completely see how many would feel uneasy about adding this to working implementations. It basically means adding some additional complexity with no actual benefit in return.
>>
>>> maybe that is something we need to look into on a more general level.
>>
>> You mean something like PRACK-lite that doesn't have offer answer? This sounds like something that could be relatively straightforward to specify. It would basically
>> have to mimic PRACK but remove the part that talks about the additional O/A possibilities.
>
> It could perhaps even be a request without a response, similar to ACK, which would make it even more "lite". When the UAS receives it, it simply ceases the transmission of the 18x.

Hmm ... this sounds like something that would make a lot of proxies and 
SIP stacks unhappy.

I actually think we could pull this off with simple PRACKs. Only, rather 
than using the "100rel" tag we go for something like "100rel-lite" which 
could be defined to indicate that a UA supports PRACK but only for 
reliability (and PRACK bodies are not allowed).

Emil


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From pkyzivat@alum.mit.edu  Tue Jul 23 11:00:16 2013
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Subject: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-01
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I just reviewed this new version. I have a number of comments:

* Section 5:

You don't go into the details of the O/A rules. But I infer you assume 
that the answer will be a subset of the offer, with the priorities 
possibly changed. This is a common strategy with SDP O/A, but it is 
limiting because it doesn't allow the answerer to indicate capabilities 
it has that weren't mentioned by the offerer. Knowing that could be 
useful, especially if a mismatch is going to result in recruiting a 
relay service.

I suggest that answer allow the answerer to indicate all of its 
capabilities and preferences, and at the same time indicate choices 
made. (If a choice was made.)

* Section 7.1:

Instead of humanintlang-send and humanintlang-recv, how about just 
giving the lang tag and a send/recv/sendrecv parameter with it? This 
would be more concise when sendrecv can be used.

Suggest using a parameter (q) to indicate relative preference for 
languages, rather than ordering. This can show degree of preference, and 
can show when multiple languages have the same preference.

Example:

a=humanintlang:EN;sendrecv;q=1,ES;recv;q=.5

To accommodate my comment about section 5, perhaps the answer can "mark" 
the alternative(s) that have been chosen for use, if any. (I'm not sure 
I like this, but for now, lets assume a "*" suffix on the direction in 
the answer.) E.g., assuming the offer above, then an answer like:

a=humanintlang:EN;sendrecv*;q=1,FR:recv;q=.5

Also in this section, the following:

    Note that while signed language tags are used with a video stream to
    indicate sign language, a spoken language tag for a video stream in
    parallel with an audio stream with the same spoken language tag
    indicates a request for a supplemental video stream to see the
    speaker.

might be ambiguous in the presence of many streams. It might be good to 
define a grouping framework (RFC 5888) usage to group an audio stream 
with a video stream that contains the speaker(s) of the audio.

Also in this section, the following:

    Clients acting on behalf of end users are expected to set one or both
    'humintlang-send' and 'humintlang-recv' attributes on each media
    stream in an offer when placing an outgoing session but ignore the
    attributes when receiving incoming calls.  Systems acting on behalf
    of call centers and PSAPs are expected to take into account the
    values when processing inbound calls.

Are you intending to exclude clients taking a more active role? (E.g., 
Isn't it possible that a client receiving an incoming call in a language 
it doesn't support might bridge in a relay service?)

Should a client that can't do anything useful with the language info 
still indicate its language preferences in the answer? I would think 
this would still be helpful. It might allow the caller, or a middlebox, 
to compensate in some way. What I suggest above allows that.

* Section 7.2 (Advisory vs Required):

Rather than "*" suffix, would be cleaner to have a separate indicator. 
And it seems like this should not be on a particular language tag. If 
you speak both Spanish and English, you may want the call to succeed if 
the callee can do one or the other, and otherwise fail. A tag per 
language doesn't do that.

Need more discussion for how to denote this.

	Thanks,
	Paul


From christer.holmberg@ericsson.com  Tue Jul 23 12:23:58 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hi,

You are right that sending a request, without a response, would cause probl=
ems.

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Emil Ivov [emcho@jitsi.org]
To: Christer Holmberg [christer.holmberg@ericsson.com]
CC: Paul Kyzivat [pkyzivat@alum.mit.edu]; Stach, Thomas [thomas.stach@sieme=
ns-enterprise.com]; mmusic@ietf.org [mmusic@ietf.org]
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions


On 22.07.13, 13:11, Christer Holmberg wrote:
> Hi,
>
>>>>> How is this better than PRACK?
>>>>
>>>> Well, you don't have to worry about supporting offers and answers in t=
he PRACK (which could indeed be a pain).
>>>
>>> In my years of "PRACK experience", PRACK itself (for the purpose of ack=
nowledging reliable 18x responses) has never been a
>>> problem. But, I agree there could be issues when it is used for O/A - e=
specially if one wants to reject an offer. In my experience, however,
>>> offers in PRACK are normally used e.g. to change pre-condition state, a=
nd/or to remove codecs, so the likelihood of rejection based on the SDP con=
tent should be rather small.
>>>
>>> There are also implementations that don't accept offers in PRACK to
>>> begin with, but I think that goes under "bad implementation" :)
>>>
>>> Having said that, not specific to trickle ICE, if O/A is a show
>>> stopper for the community to use PRACK,
>>
>> Well, speaking as an implementor, I can completely see how many would fe=
el uneasy about adding this to working implementations. It basically means =
adding some additional complexity with no actual benefit in return.
>>
>>> maybe that is something we need to look into on a more general level.
>>
>> You mean something like PRACK-lite that doesn't have offer answer? This =
sounds like something that could be relatively straightforward to specify. =
It would basically
>> have to mimic PRACK but remove the part that talks about the additional =
O/A possibilities.
>
> It could perhaps even be a request without a response, similar to ACK, wh=
ich would make it even more "lite". When the UAS receives it, it simply cea=
ses the transmission of the 18x.

Hmm ... this sounds like something that would make a lot of proxies and
SIP stacks unhappy.

I actually think we could pull this off with simple PRACKs. Only, rather
than using the "100rel" tag we go for something like "100rel-lite" which
could be defined to indicate that a UA supports PRACK but only for
reliability (and PRACK bodies are not allowed).

Emil


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<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">You are right that sending a request, without a=
 response, would cause problems.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Emil Ivov [emcho@jitsi.org]<br>
<b>To:</b> Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>CC:</b> Paul Kyzivat [pkyzivat@alum.mit.edu]; Stach, Thomas [thomas.stac=
h@siemens-enterprise.com]; mmusic@ietf.org [mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] Trickle ICE for SIP Questions</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
<br>
On 22.07.13, 13:11, Christer Holmberg wrote:<br>
&gt; Hi,<br>
&gt;<br>
&gt;&gt;&gt;&gt;&gt; How is this better than PRACK?<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; Well, you don't have to worry about supporting offers and =
answers in the PRACK (which could indeed be a pain).<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; In my years of &quot;PRACK experience&quot;, PRACK itself (for=
 the purpose of acknowledging reliable 18x responses) has never been a<br>
&gt;&gt;&gt; problem. But, I agree there could be issues when it is used fo=
r O/A - especially if one wants to reject an offer. In my experience, howev=
er,<br>
&gt;&gt;&gt; offers in PRACK are normally used e.g. to change pre-condition=
 state, and/or to remove codecs, so the likelihood of rejection based on th=
e SDP content should be rather small.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; There are also implementations that don't accept offers in PRA=
CK to<br>
&gt;&gt;&gt; begin with, but I think that goes under &quot;bad implementati=
on&quot; :)<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Having said that, not specific to trickle ICE, if O/A is a sho=
w<br>
&gt;&gt;&gt; stopper for the community to use PRACK,<br>
&gt;&gt;<br>
&gt;&gt; Well, speaking as an implementor, I can completely see how many wo=
uld feel uneasy about adding this to working implementations. It basically =
means adding some additional complexity with no actual benefit in return.<b=
r>
&gt;&gt;<br>
&gt;&gt;&gt; maybe that is something we need to look into on a more general=
 level.<br>
&gt;&gt;<br>
&gt;&gt; You mean something like PRACK-lite that doesn't have offer answer?=
 This sounds like something that could be relatively straightforward to spe=
cify. It would basically<br>
&gt;&gt; have to mimic PRACK but remove the part that talks about the addit=
ional O/A possibilities.<br>
&gt;<br>
&gt; It could perhaps even be a request without a response, similar to ACK,=
 which would make it even more &quot;lite&quot;. When the UAS receives it, =
it simply ceases the transmission of the 18x.<br>
<br>
Hmm ... this sounds like something that would make a lot of proxies and <br=
>
SIP stacks unhappy.<br>
<br>
I actually think we could pull this off with simple PRACKs. Only, rather <b=
r>
than using the &quot;100rel&quot; tag we go for something like &quot;100rel=
-lite&quot; which <br>
could be defined to indicate that a UA supports PRACK but only for <br>
reliability (and PRACK bodies are not allowed).<br>
<br>
Emil<br>
<br>
<br>
-- <br>
<a href=3D"https://jitsi.org">https://jitsi.org</a><br>
</div>
</span></font>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C404166ESESSMB209erics_--

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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Adam Roach <adam@nostrum.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Unified plan IPR
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>, tim panton <thp@westhawk.co.uk>
Subject: Re: [MMUSIC] [rtcweb] Unified plan IPR
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Adam Roach:
>
> I doubt agitating for action on these mailing lists is likely to produce =
useful
> results.

This appears to be generally true, yes.

One wonders why we have them.

Matthew Kaufman


From gunnar.hellstrom@omnitor.se  Tue Jul 23 15:37:02 2013
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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-01
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On 2013-07-23 20:00, Paul Kyzivat wrote:
> I just reviewed this new version. I have a number of comments:
Good comments Paul,
>
> * Section 5:
>
> You don't go into the details of the O/A rules. But I infer you assume 
> that the answer will be a subset of the offer, with the priorities 
> possibly changed. This is a common strategy with SDP O/A, but it is 
> limiting because it doesn't allow the answerer to indicate 
> capabilities it has that weren't mentioned by the offerer. Knowing 
> that could be useful, especially if a mismatch is going to result in 
> recruiting a relay service.
>
> I suggest that answer allow the answerer to indicate all of its 
> capabilities and preferences, and at the same time indicate choices 
> made. (If a choice was made.)
<GH>Yes, very useful to indicate all.
>
> * Section 7.1:
>
> Instead of humanintlang-send and humanintlang-recv, how about just 
> giving the lang tag and a send/recv/sendrecv parameter with it? This 
> would be more concise when sendrecv can be used.
>
> Suggest using a parameter (q) to indicate relative preference for 
> languages, rather than ordering. This can show degree of preference, 
> and can show when multiple languages have the same preference.
>
<GH>Have you seen my request for an indication of level of preference 
between the different modalities? A possiblility to indicate for example 
that British Sign Language is highly preferred, but English text possible.
The q values could be used for coding of such relative level of 
preference even between modalities. a q=1 for BSL in video, and a q=0.1 
for English in text.   Right?
> Example:
>
> a=humanintlang:EN;sendrecv;q=1,ES;recv;q=.5
>
> To accommodate my comment about section 5, perhaps the answer can 
> "mark" the alternative(s) that have been chosen for use, if any. (I'm 
> not sure I like this, but for now, lets assume a "*" suffix on the 
> direction in the answer.) E.g., assuming the offer above, then an 
> answer like:
>
> a=humanintlang:EN;sendrecv*;q=1,FR:recv;q=.5
>
> Also in this section, the following:
>
>    Note that while signed language tags are used with a video stream to
>    indicate sign language, a spoken language tag for a video stream in
>    parallel with an audio stream with the same spoken language tag
>    indicates a request for a supplemental video stream to see the
>    speaker.
>
> might be ambiguous in the presence of many streams. It might be good 
> to define a grouping framework (RFC 5888) usage to group an audio 
> stream with a video stream that contains the speaker(s) of the audio.
<GH>This is both a threat and a solution. There are already other 
reasons to group media in SDP. We have a risk of ending up in a very 
complex coding in SDP to get this correctly specified in an environment 
where there are also other reasons to group m-lines. This is why I am 
still hesitating if the whole thing would not fit better as tags on the 
SIP level.
>
> Also in this section, the following:
>
>    Clients acting on behalf of end users are expected to set one or both
>    'humintlang-send' and 'humintlang-recv' attributes on each media
>    stream in an offer when placing an outgoing session but ignore the
>    attributes when receiving incoming calls.  Systems acting on behalf
>    of call centers and PSAPs are expected to take into account the
>    values when processing inbound calls.
>
> Are you intending to exclude clients taking a more active role? (E.g., 
> Isn't it possible that a client receiving an incoming call in a 
> language it doesn't support might bridge in a relay service?)
<GH>I have asked for deletion of this limiting paragraph.
>
> Should a client that can't do anything useful with the language info 
> still indicate its language preferences in the answer? I would think 
> this would still be helpful. It might allow the caller, or a 
> middlebox, to compensate in some way. What I suggest above allows that.
<GH>Yes, I prefer that view. The answering part shall answer if it can. 
middleboxes or the offering part may do the invocation of services to 
cover the gaps.
>
> * Section 7.2 (Advisory vs Required):
>
> Rather than "*" suffix, would be cleaner to have a separate indicator. 
> And it seems like this should not be on a particular language tag. If 
> you speak both Spanish and English, you may want the call to succeed 
> if the callee can do one or the other, and otherwise fail. A tag per 
> language doesn't do that.

<GH>Yes, tricky.  I am getting a feeling that we should start a draft 
describing usage in parallel to this one, so that use cases can be 
verified.

>
> Need more discussion for how to denote this.
>
>     Thanks,
>     Paul
>
Gunnar
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From pkyzivat@alum.mit.edu  Tue Jul 23 17:05:49 2013
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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-01
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On 7/23/13 6:35 PM, Gunnar Hellstrom wrote:
>
> On 2013-07-23 20:00, Paul Kyzivat wrote:
>> I just reviewed this new version. I have a number of comments:
> Good comments Paul,
>>
>> * Section 5:
>>
>> You don't go into the details of the O/A rules. But I infer you assume
>> that the answer will be a subset of the offer, with the priorities
>> possibly changed. This is a common strategy with SDP O/A, but it is
>> limiting because it doesn't allow the answerer to indicate
>> capabilities it has that weren't mentioned by the offerer. Knowing
>> that could be useful, especially if a mismatch is going to result in
>> recruiting a relay service.
>>
>> I suggest that answer allow the answerer to indicate all of its
>> capabilities and preferences, and at the same time indicate choices
>> made. (If a choice was made.)
> <GH>Yes, very useful to indicate all.
>>
>> * Section 7.1:
>>
>> Instead of humanintlang-send and humanintlang-recv, how about just
>> giving the lang tag and a send/recv/sendrecv parameter with it? This
>> would be more concise when sendrecv can be used.
>>
>> Suggest using a parameter (q) to indicate relative preference for
>> languages, rather than ordering. This can show degree of preference,
>> and can show when multiple languages have the same preference.
>>
> <GH>Have you seen my request for an indication of level of preference
> between the different modalities? A possiblility to indicate for example
> that British Sign Language is highly preferred, but English text possible.
> The q values could be used for coding of such relative level of
> preference even between modalities. a q=1 for BSL in video, and a q=0.1
> for English in text.   Right?

That would work within a single medium.

But I don't think we can use that mechanism to trade off between 
different media. For that I think some other, more complex, mechanism 
would be needed.

Or else we would need a more complex definition of which things are 
being prioritized against one another in order to use the q-values.

>> Example:
>>
>> a=humanintlang:EN;sendrecv;q=1,ES;recv;q=.5
>>
>> To accommodate my comment about section 5, perhaps the answer can
>> "mark" the alternative(s) that have been chosen for use, if any. (I'm
>> not sure I like this, but for now, lets assume a "*" suffix on the
>> direction in the answer.) E.g., assuming the offer above, then an
>> answer like:
>>
>> a=humanintlang:EN;sendrecv*;q=1,FR:recv;q=.5
>>
>> Also in this section, the following:
>>
>>    Note that while signed language tags are used with a video stream to
>>    indicate sign language, a spoken language tag for a video stream in
>>    parallel with an audio stream with the same spoken language tag
>>    indicates a request for a supplemental video stream to see the
>>    speaker.
>>
>> might be ambiguous in the presence of many streams. It might be good
>> to define a grouping framework (RFC 5888) usage to group an audio
>> stream with a video stream that contains the speaker(s) of the audio.
> <GH>This is both a threat and a solution. There are already other
> reasons to group media in SDP. We have a risk of ending up in a very
> complex coding in SDP to get this correctly specified in an environment
> where there are also other reasons to group m-lines. This is why I am
> still hesitating if the whole thing would not fit better as tags on the
> SIP level.

I notice Henning just asked about this today, in a different forum.

Perhaps some things are better done at the sip level, and others at the 
SDP level.

Its also possible that correlations between different media streams 
should be left to other mechanisms. E.g., CLUE could signal better 
correlations between audio and video.

	Thanks,
	Paul

>>
>> Also in this section, the following:
>>
>>    Clients acting on behalf of end users are expected to set one or both
>>    'humintlang-send' and 'humintlang-recv' attributes on each media
>>    stream in an offer when placing an outgoing session but ignore the
>>    attributes when receiving incoming calls.  Systems acting on behalf
>>    of call centers and PSAPs are expected to take into account the
>>    values when processing inbound calls.
>>
>> Are you intending to exclude clients taking a more active role? (E.g.,
>> Isn't it possible that a client receiving an incoming call in a
>> language it doesn't support might bridge in a relay service?)
> <GH>I have asked for deletion of this limiting paragraph.
>>
>> Should a client that can't do anything useful with the language info
>> still indicate its language preferences in the answer? I would think
>> this would still be helpful. It might allow the caller, or a
>> middlebox, to compensate in some way. What I suggest above allows that.
> <GH>Yes, I prefer that view. The answering part shall answer if it can.
> middleboxes or the offering part may do the invocation of services to
> cover the gaps.
>>
>> * Section 7.2 (Advisory vs Required):
>>
>> Rather than "*" suffix, would be cleaner to have a separate indicator.
>> And it seems like this should not be on a particular language tag. If
>> you speak both Spanish and English, you may want the call to succeed
>> if the callee can do one or the other, and otherwise fail. A tag per
>> language doesn't do that.
>
> <GH>Yes, tricky.  I am getting a feeling that we should start a draft
> describing usage in parallel to this one, so that use cases can be
> verified.
>
>>
>> Need more discussion for how to denote this.
>>
>>     Thanks,
>>     Paul
>>
> Gunnar
>> _______________________________________________
>> 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: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Emil Ivov <emcho@jitsi.org>, MMUSIC IETF WG <mmusic@ietf.org>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
Thread-Index: AQHOiB3rsesO7v/XCUiNMNrfupXiAQ==
Date: Wed, 24 Jul 2013 03:28:52 +0000
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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I prefer the 180 resend approach and all the right things are already looki=
ng at them.  I have always objected to mandating use of PRACK. Obviously I'=
m fine with things that have PRACK can use it but I want some solution for =
things that don't.=20

One small note, the SDP in the 180 can change as long as the to-tag is also=
 changed.=20



On Jul 3, 2013, at 8:13 AM, Emil Ivov <emcho@jitsi.org> wrote:

> Hey all,
>=20
> Christer, Enrico and I are preparing the next version of Trickle ICE for =
SIP. Now that discussions on BUNDLE and the plans seem to be winding down, =
we wanted to run a few questions by the working group.
>=20
> Q1: Making reliable provisional responses and PRACK mandatory. Obviously =
this would be nice to avoid, so the question is: is there a reasonable mech=
anism to achieve this (and by reasonable, we mean something that wouldn't b=
e harder than implementing support for PRACK).
>=20
> There was some discussion about this back in April and there was a sugges=
tion for implementing a 5245-style hack where the answerer basically resend=
s the 180 until it knows that it has been received.
>=20
> 5245 uses connectivity checks for this (i.e. it stops 180 retransmissions=
 when the first connectivity check is received) but we don't have that opti=
on here since the 180 may contain either none or only host candidates so th=
ere are strong chances that no binding request would be received on them.
>=20
> Thomas also suggested a second option which would be to also use INFO req=
uests with trickled candidates as an indication that 180 was received. This=
 however wouldn't work with half trickle so we are basically back to vanill=
a ICE for all (non-re) INVITEs.
>=20
> Another option would be to mandate an INFO request with "end-of-candidate=
s" in response to the 180, but that would be just the same as mandating PRA=
CK support.
>=20
> Thomas also suggested that the answerer can start sending INFOs right aft=
er it sends its answer in the 180 and then it can just resend the 180 if th=
e INFOs result in a 481 response.
>=20
> Personally I think this could potentially be made to work, but it would i=
mply a level of complexity that considerably exceeds PRACK support.
>=20
> Opinions?
>=20
> Q2: How do we send INFOs? Are they blocking or do we just send them in pa=
rallel? If the latter, then what happens when an INFO fails because it is r=
eceived out of order? Do we just tell the application to resend the candida=
tes asap?
>=20
> This also leads to the following question:
>=20
> Q3: What exactly do we send in INFOs? Just the latest batch of freshly le=
arned candidates or all candidates we've learned so far? Dale suggested tha=
t if we do this cumulatively we wouldn't need to worry about the case with =
the out-of-order INFOs from Q2 since the information gets resent anyway. A =
drawback here would obviously be that this adds more complexity for trickle=
 ICE users (WebRTC applications specifically)
>=20
> A third option would be to allow both and leave it to the application.
>=20
> Comments are most welcome!
>=20
> Emil
>=20
> --=20
> https://jitsi.org
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From gunnar.hellstrom@omnitor.se  Tue Jul 23 23:48:59 2013
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Subject: Re: [MMUSIC] draft-gellens-mmusic-negotiating-human-language-01
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On 2013-07-24 02:05, Paul Kyzivat wrote:
> On 7/23/13 6:35 PM, Gunnar Hellstrom wrote:
>>
>> On 2013-07-23 20:00, Paul Kyzivat wrote:
>>> I just reviewed this new version. I have a number of comments:
>> Good comments Paul,
>>>
>>> * Section 5:
>>>
>>> You don't go into the details of the O/A rules. But I infer you assume
>>> that the answer will be a subset of the offer, with the priorities
>>> possibly changed. This is a common strategy with SDP O/A, but it is
>>> limiting because it doesn't allow the answerer to indicate
>>> capabilities it has that weren't mentioned by the offerer. Knowing
>>> that could be useful, especially if a mismatch is going to result in
>>> recruiting a relay service.
>>>
>>> I suggest that answer allow the answerer to indicate all of its
>>> capabilities and preferences, and at the same time indicate choices
>>> made. (If a choice was made.)
>> <GH>Yes, very useful to indicate all.
>>>
>>> * Section 7.1:
>>>
>>> Instead of humanintlang-send and humanintlang-recv, how about just
>>> giving the lang tag and a send/recv/sendrecv parameter with it? This
>>> would be more concise when sendrecv can be used.
>>>
>>> Suggest using a parameter (q) to indicate relative preference for
>>> languages, rather than ordering. This can show degree of preference,
>>> and can show when multiple languages have the same preference.
>>>
>> <GH>Have you seen my request for an indication of level of preference
>> between the different modalities? A possiblility to indicate for example
>> that British Sign Language is highly preferred, but English text 
>> possible.
>> The q values could be used for coding of such relative level of
>> preference even between modalities. a q=1 for BSL in video, and a q=0.1
>> for English in text.   Right?
>
> That would work within a single medium.
>
> But I don't think we can use that mechanism to trade off between 
> different media. For that I think some other, more complex, mechanism 
> would be needed.
>
> Or else we would need a more complex definition of which things are 
> being prioritized against one another in order to use the q-values.
I proposed a simpler mechanism in a recent mail. It was to insert at the 
end of the humintlang tag a character a "!" for highly preferred, 
nothing for manageable and "#" for
a non-preferred but possible fall-back language and modality.

I think it will be very important for most users to have a chance to 
indicate this three-level preference order between different media as 
well as between different languages within one medium.

I am hesitating a bit for the type of coding I proposed, and saw your 
proposal of q-value as something cleaner. When coding, you could make 
sure that the last resort medium gets q-values under 0.4, the strongly 
preferred between 0.7 and 1 and the managed ones between 0.4 and 0.6.

You can see the earlier motivation for the preference between modalities 
in the archive at:
http://www.ietf.org/mail-archive/web/mmusic/current/msg11871.html


>
>>> Example:
>>>
>>> a=humanintlang:EN;sendrecv;q=1,ES;recv;q=.5
>>>
>>> To accommodate my comment about section 5, perhaps the answer can
>>> "mark" the alternative(s) that have been chosen for use, if any. (I'm
>>> not sure I like this, but for now, lets assume a "*" suffix on the
>>> direction in the answer.) E.g., assuming the offer above, then an
>>> answer like:
>>>
>>> a=humanintlang:EN;sendrecv*;q=1,FR:recv;q=.5
>>>
>>> Also in this section, the following:
>>>
>>>    Note that while signed language tags are used with a video stream to
>>>    indicate sign language, a spoken language tag for a video stream in
>>>    parallel with an audio stream with the same spoken language tag
>>>    indicates a request for a supplemental video stream to see the
>>>    speaker.
>>>
>>> might be ambiguous in the presence of many streams. It might be good
>>> to define a grouping framework (RFC 5888) usage to group an audio
>>> stream with a video stream that contains the speaker(s) of the audio.
>> <GH>This is both a threat and a solution. There are already other
>> reasons to group media in SDP. We have a risk of ending up in a very
>> complex coding in SDP to get this correctly specified in an environment
>> where there are also other reasons to group m-lines. This is why I am
>> still hesitating if the whole thing would not fit better as tags on the
>> SIP level.
>
> I notice Henning just asked about this today, in a different forum.
>
> Perhaps some things are better done at the sip level, and others at 
> the SDP level.
>
> Its also possible that correlations between different media streams 
> should be left to other mechanisms. E.g., CLUE could signal better 
> correlations between audio and video.
Yes, but here we have three media. video, audio and text. CLUE need to 
take text on board.

But you are right, if this mechanism is moved to SIP level, then we get 
the complex task to point out to what m-lines with the same medium do 
the humintlang-tags for that medium relate to. Simplifications need to 
be taken, so that we do not end up in specifying another CLUE.

I think that part of the background for this draft is that the mechanism 
shall be usable in 3GPP Multimedia Telephony . That might be a reason 
why it ended up specified in SDP.

Gunnar
>
>     Thanks,
>     Paul
>
>>>
>>> Also in this section, the following:
>>>
>>>    Clients acting on behalf of end users are expected to set one or 
>>> both
>>>    'humintlang-send' and 'humintlang-recv' attributes on each media
>>>    stream in an offer when placing an outgoing session but ignore the
>>>    attributes when receiving incoming calls.  Systems acting on behalf
>>>    of call centers and PSAPs are expected to take into account the
>>>    values when processing inbound calls.
>>>
>>> Are you intending to exclude clients taking a more active role? (E.g.,
>>> Isn't it possible that a client receiving an incoming call in a
>>> language it doesn't support might bridge in a relay service?)
>> <GH>I have asked for deletion of this limiting paragraph.
>>>
>>> Should a client that can't do anything useful with the language info
>>> still indicate its language preferences in the answer? I would think
>>> this would still be helpful. It might allow the caller, or a
>>> middlebox, to compensate in some way. What I suggest above allows that.
>> <GH>Yes, I prefer that view. The answering part shall answer if it can.
>> middleboxes or the offering part may do the invocation of services to
>> cover the gaps.
>>>
>>> * Section 7.2 (Advisory vs Required):
>>>
>>> Rather than "*" suffix, would be cleaner to have a separate indicator.
>>> And it seems like this should not be on a particular language tag. If
>>> you speak both Spanish and English, you may want the call to succeed
>>> if the callee can do one or the other, and otherwise fail. A tag per
>>> language doesn't do that.
>>
>> <GH>Yes, tricky.  I am getting a feeling that we should start a draft
>> describing usage in parallel to this one, so that use cases can be
>> verified.
>>
>>>
>>> Need more discussion for how to denote this.
>>>
>>>     Thanks,
>>>     Paul
>>>
>> Gunnar
>>> _______________________________________________
>>> 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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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "mmusic@ietf.org" <mmusic@ietf.org>, tim panton <thp@westhawk.co.uk>
Subject: Re: [MMUSIC] [rtcweb] Unified plan IPR
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FYI,=0A=
=0A=
I've asked internally in my company for advice on what needs to be done =0A=
(if anything) from our side. IPRs and handling of them is outside my =0A=
competence.=0A=
=0A=
Stefan=0A=
=0A=
=0A=
On 7/23/13 6:22 PM, Roman Shpount wrote:=0A=
> Changing the title and the mailing list to the more appropriate.=0A=
>=0A=
> The situation would be a bit clearer if patent holders were to provide=0A=
> the licensing policy regarding this IPR release. Given that Ericsson is=
=0A=
> actively involved in this working group, I think it would be reasonable=
=0A=
> to ask them for this.=0A=
>=0A=
> Regards,=0A=
> _____________=0A=
> Roman Shpount=0A=
>=0A=
>=0A=
> On Tue, Jul 23, 2013 at 11:51 AM, Adam Roach <adam@nostrum.com=0A=
> <mailto:adam@nostrum.com>> wrote:=0A=
>=0A=
>     On 7/23/13 09:20, tim panton wrote:=0A=
>>     On 23 Jul 2013, at 14:00, Stefan H=E5kansson LK<stefan.lk.hakansson@=
ericsson.com>  <mailto:stefan.lk.hakansson@ericsson.com>  wrote:=0A=
>>=0A=
>>>     On 7/23/13 5:03 AM, Justin Uberti wrote:=0A=
>>>>     With the compromise reached in the Unified Plan document,=0A=
>>>     Presumablyhttp://www.ietf.org/id/draft-roach-mmusic-unified-plan-00=
.txt=0A=
>>     I'll draw folks attention to this IPR claim=0A=
>>=0A=
>>     https://datatracker.ietf.org/ipr/2141/=0A=
>>=0A=
>>     What does that mean in practice?=0A=
>>=0A=
>=0A=
>     In practice, it has very little effect.=0A=
>=0A=
>     For those of you unfamiliar with IETF IPR policy, it is documented=0A=
>     in RFC 3979:=0A=
>=0A=
>     http://www.ietf.org/rfc/rfc3979.txt=0A=
>=0A=
>     The disclosure Tim points to is made pursuant to section 6.1.3 of=0A=
>     that document. During the course of developing the unified plan, the=
=0A=
>     applications that are mentioned in that disclosure were brought to=0A=
>     my attention.=0A=
>=0A=
>     The reason it has little effect in practice is that the independent=
=0A=
>     claims would appear to cover every plan proposal to date (plan a,=0A=
>     plan b, unified plan, and even all of the "no plan" variations).=0A=
>     Consequently, it does not benefit any one approach over the others.=
=0A=
>=0A=
>     /a=0A=
>=0A=
>=0A=
=0A=

From alan.b.johnston@gmail.com  Wed Jul 24 02:45:50 2013
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Date: Wed, 24 Jul 2013 05:45:48 -0400
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From: Alan Johnston <alan.b.johnston@gmail.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Cc: MMUSIC IETF WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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--f46d043c815e36bfcc04e23ec6fa
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Agree with Cullen - if there is a reasonable approach (such as
retransmitting 180) that avoids PRACK, then we should use this approach.

- Alan -


On Tue, Jul 23, 2013 at 11:28 PM, Cullen Jennings (fluffy) <fluffy@cisco.com
> wrote:

>
> I prefer the 180 resend approach and all the right things are already
> looking at them.  I have always objected to mandating use of PRACK.
> Obviously I'm fine with things that have PRACK can use it but I want some
> solution for things that don't.
>
> One small note, the SDP in the 180 can change as long as the to-tag is
> also changed.
>
>
>
> On Jul 3, 2013, at 8:13 AM, Emil Ivov <emcho@jitsi.org> wrote:
>
> > Hey all,
> >
> > Christer, Enrico and I are preparing the next version of Trickle ICE for
> SIP. Now that discussions on BUNDLE and the plans seem to be winding down,
> we wanted to run a few questions by the working group.
> >
> > Q1: Making reliable provisional responses and PRACK mandatory. Obviously
> this would be nice to avoid, so the question is: is there a reasonable
> mechanism to achieve this (and by reasonable, we mean something that
> wouldn't be harder than implementing support for PRACK).
> >
> > There was some discussion about this back in April and there was a
> suggestion for implementing a 5245-style hack where the answerer basically
> resends the 180 until it knows that it has been received.
> >
> > 5245 uses connectivity checks for this (i.e. it stops 180
> retransmissions when the first connectivity check is received) but we don't
> have that option here since the 180 may contain either none or only host
> candidates so there are strong chances that no binding request would be
> received on them.
> >
> > Thomas also suggested a second option which would be to also use INFO
> requests with trickled candidates as an indication that 180 was received.
> This however wouldn't work with half trickle so we are basically back to
> vanilla ICE for all (non-re) INVITEs.
> >
> > Another option would be to mandate an INFO request with
> "end-of-candidates" in response to the 180, but that would be just the same
> as mandating PRACK support.
> >
> > Thomas also suggested that the answerer can start sending INFOs right
> after it sends its answer in the 180 and then it can just resend the 180 if
> the INFOs result in a 481 response.
> >
> > Personally I think this could potentially be made to work, but it would
> imply a level of complexity that considerably exceeds PRACK support.
> >
> > Opinions?
> >
> > Q2: How do we send INFOs? Are they blocking or do we just send them in
> parallel? If the latter, then what happens when an INFO fails because it is
> received out of order? Do we just tell the application to resend the
> candidates asap?
> >
> > This also leads to the following question:
> >
> > Q3: What exactly do we send in INFOs? Just the latest batch of freshly
> learned candidates or all candidates we've learned so far? Dale suggested
> that if we do this cumulatively we wouldn't need to worry about the case
> with the out-of-order INFOs from Q2 since the information gets resent
> anyway. A drawback here would obviously be that this adds more complexity
> for trickle ICE users (WebRTC applications specifically)
> >
> > A third option would be to allow both and leave it to the application.
> >
> > Comments are most welcome!
> >
> > Emil
> >
> > --
> > https://jitsi.org
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www.ietf.org/mailman/listinfo/mmusic
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

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<div dir=3D"ltr">Agree with Cullen - if there is a reasonable approach (suc=
h as retransmitting 180) that avoids PRACK, then we should use this approac=
h.<div><br></div><div>- Alan -</div></div><div class=3D"gmail_extra"><br><b=
r>
<div class=3D"gmail_quote">On Tue, Jul 23, 2013 at 11:28 PM, Cullen Jenning=
s (fluffy) <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" target=
=3D"_blank">fluffy@cisco.com</a>&gt;</span> wrote:<br><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex">
<br>
I prefer the 180 resend approach and all the right things are already looki=
ng at them. =A0I have always objected to mandating use of PRACK. Obviously =
I&#39;m fine with things that have PRACK can use it but I want some solutio=
n for things that don&#39;t.<br>

<br>
One small note, the SDP in the 180 can change as long as the to-tag is also=
 changed.<br>
<br>
<br>
<br>
On Jul 3, 2013, at 8:13 AM, Emil Ivov &lt;<a href=3D"mailto:emcho@jitsi.org=
">emcho@jitsi.org</a>&gt; wrote:<br>
<br>
&gt; Hey all,<br>
&gt;<br>
&gt; Christer, Enrico and I are preparing the next version of Trickle ICE f=
or SIP. Now that discussions on BUNDLE and the plans seem to be winding dow=
n, we wanted to run a few questions by the working group.<br>
&gt;<br>
&gt; Q1: Making reliable provisional responses and PRACK mandatory. Obvious=
ly this would be nice to avoid, so the question is: is there a reasonable m=
echanism to achieve this (and by reasonable, we mean something that wouldn&=
#39;t be harder than implementing support for PRACK).<br>

&gt;<br>
&gt; There was some discussion about this back in April and there was a sug=
gestion for implementing a 5245-style hack where the answerer basically res=
ends the 180 until it knows that it has been received.<br>
&gt;<br>
&gt; 5245 uses connectivity checks for this (i.e. it stops 180 retransmissi=
ons when the first connectivity check is received) but we don&#39;t have th=
at option here since the 180 may contain either none or only host candidate=
s so there are strong chances that no binding request would be received on =
them.<br>

&gt;<br>
&gt; Thomas also suggested a second option which would be to also use INFO =
requests with trickled candidates as an indication that 180 was received. T=
his however wouldn&#39;t work with half trickle so we are basically back to=
 vanilla ICE for all (non-re) INVITEs.<br>

&gt;<br>
&gt; Another option would be to mandate an INFO request with &quot;end-of-c=
andidates&quot; in response to the 180, but that would be just the same as =
mandating PRACK support.<br>
&gt;<br>
&gt; Thomas also suggested that the answerer can start sending INFOs right =
after it sends its answer in the 180 and then it can just resend the 180 if=
 the INFOs result in a 481 response.<br>
&gt;<br>
&gt; Personally I think this could potentially be made to work, but it woul=
d imply a level of complexity that considerably exceeds PRACK support.<br>
&gt;<br>
&gt; Opinions?<br>
&gt;<br>
&gt; Q2: How do we send INFOs? Are they blocking or do we just send them in=
 parallel? If the latter, then what happens when an INFO fails because it i=
s received out of order? Do we just tell the application to resend the cand=
idates asap?<br>

&gt;<br>
&gt; This also leads to the following question:<br>
&gt;<br>
&gt; Q3: What exactly do we send in INFOs? Just the latest batch of freshly=
 learned candidates or all candidates we&#39;ve learned so far? Dale sugges=
ted that if we do this cumulatively we wouldn&#39;t need to worry about the=
 case with the out-of-order INFOs from Q2 since the information gets resent=
 anyway. A drawback here would obviously be that this adds more complexity =
for trickle ICE users (WebRTC applications specifically)<br>

&gt;<br>
&gt; A third option would be to allow both and leave it to the application.=
<br>
&gt;<br>
&gt; Comments are most welcome!<br>
&gt;<br>
&gt; Emil<br>
&gt;<br>
&gt; --<br>
&gt; <a href=3D"https://jitsi.org" target=3D"_blank">https://jitsi.org</a><=
br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; <a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
<br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
</blockquote></div><br></div>

--f46d043c815e36bfcc04e23ec6fa--

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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hey Alan, Cullen,

On 24.07.13, 11:45, Alan Johnston wrote:
> Agree with Cullen - if there is a reasonable approach (such as
> retransmitting 180) that avoids PRACK, then we should use this approach.

There is one and it comes down to requiring the remote side to send an 
INFO request as soon as it gets it. The INFO request could contain 
trickled candidates (in the case of full trickle) or just 
end-of-candidates (in the case of half-trickle).

This could work and, personally, I have no issue with it.

I believe the suggestion that Christer made was: given how the above 
basically works as a PRACK minus the O/A and given how the O/A in PRACKs 
seems to be a deal breaker for many, then we might want to generalize 
the above mechanism so that other specs can also use it.

Cheers,
Emil


> - Alan -
>
>
> On Tue, Jul 23, 2013 at 11:28 PM, Cullen Jennings (fluffy)
> <fluffy@cisco.com <mailto:fluffy@cisco.com>> wrote:
>
>
>     I prefer the 180 resend approach and all the right things are
>     already looking at them.  I have always objected to mandating use of
>     PRACK. Obviously I'm fine with things that have PRACK can use it but
>     I want some solution for things that don't.
>
>     One small note, the SDP in the 180 can change as long as the to-tag
>     is also changed.
>
>
>
>     On Jul 3, 2013, at 8:13 AM, Emil Ivov <emcho@jitsi.org
>     <mailto:emcho@jitsi.org>> wrote:
>
>      > Hey all,
>      >
>      > Christer, Enrico and I are preparing the next version of Trickle
>     ICE for SIP. Now that discussions on BUNDLE and the plans seem to be
>     winding down, we wanted to run a few questions by the working group.
>      >
>      > Q1: Making reliable provisional responses and PRACK mandatory.
>     Obviously this would be nice to avoid, so the question is: is there
>     a reasonable mechanism to achieve this (and by reasonable, we mean
>     something that wouldn't be harder than implementing support for PRACK).
>      >
>      > There was some discussion about this back in April and there was
>     a suggestion for implementing a 5245-style hack where the answerer
>     basically resends the 180 until it knows that it has been received.
>      >
>      > 5245 uses connectivity checks for this (i.e. it stops 180
>     retransmissions when the first connectivity check is received) but
>     we don't have that option here since the 180 may contain either none
>     or only host candidates so there are strong chances that no binding
>     request would be received on them.
>      >
>      > Thomas also suggested a second option which would be to also use
>     INFO requests with trickled candidates as an indication that 180 was
>     received. This however wouldn't work with half trickle so we are
>     basically back to vanilla ICE for all (non-re) INVITEs.
>      >
>      > Another option would be to mandate an INFO request with
>     "end-of-candidates" in response to the 180, but that would be just
>     the same as mandating PRACK support.
>      >
>      > Thomas also suggested that the answerer can start sending INFOs
>     right after it sends its answer in the 180 and then it can just
>     resend the 180 if the INFOs result in a 481 response.
>      >
>      > Personally I think this could potentially be made to work, but it
>     would imply a level of complexity that considerably exceeds PRACK
>     support.
>      >
>      > Opinions?
>      >
>      > Q2: How do we send INFOs? Are they blocking or do we just send
>     them in parallel? If the latter, then what happens when an INFO
>     fails because it is received out of order? Do we just tell the
>     application to resend the candidates asap?
>      >
>      > This also leads to the following question:
>      >
>      > Q3: What exactly do we send in INFOs? Just the latest batch of
>     freshly learned candidates or all candidates we've learned so far?
>     Dale suggested that if we do this cumulatively we wouldn't need to
>     worry about the case with the out-of-order INFOs from Q2 since the
>     information gets resent anyway. A drawback here would obviously be
>     that this adds more complexity for trickle ICE users (WebRTC
>     applications specifically)
>      >
>      > A third option would be to allow both and leave it to the
>     application.
>      >
>      > Comments are most welcome!
>      >
>      > Emil
>      >
>      > --
>      > https://jitsi.org
>      > _______________________________________________
>      > mmusic mailing list
>      > mmusic@ietf.org <mailto:mmusic@ietf.org>
>      > https://www.ietf.org/mailman/listinfo/mmusic
>
>     _______________________________________________
>     mmusic mailing list
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>
>

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From pkyzivat@alum.mit.edu  Wed Jul 24 07:58:44 2013
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Subject: [MMUSIC] draft-ivov-mmusic-multiple-sources-00
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I have one question about this draft:

Section 2.1 says:

    Unless an endpoint detects the corresponding max-ssrc or upper level
    protocol indicators that a remote peer can actually handle multiple
    streams within a single "m=" line, it MUST assume that such support
    is unavailable.

But the example in section 2 only uses max-ssrc for video, not audio. 
Does that mean that only one audio stream should be used in the example?

	Thanks,
	Paul

From emil@sip-communicator.org  Wed Jul 24 08:03:57 2013
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On 24.07.13, 16:57, Paul Kyzivat wrote:
> I have one question about this draft:
>
> Section 2.1 says:
>
>      Unless an endpoint detects the corresponding max-ssrc or upper level
>      protocol indicators that a remote peer can actually handle multiple
>      streams within a single "m=" line, it MUST assume that such support
>      is unavailable.
>
> But the example in section 2 only uses max-ssrc for video, not audio.
> Does that mean that only one audio stream should be used in the example?

Good question. The lack of max-ssrc for audio was not intentional but I 
think it would be legal and it would indeed mean that a maximum of one 
SSRC is allowed for that "m=" line.

Emil
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From pkyzivat@alum.mit.edu  Wed Jul 24 08:55:22 2013
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Subject: [MMUSIC] draft-roach-mmusic-unified-plan-00
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Some comments after carefully rereading this draft:

Section 3.2.1 says:

    A different problem can arise when interoperating with legacy
    equipment.  A number of circumstances can lead to the inability of a
    legacy endpoint to include SSRC information in its SDP.  For example,
    in a system that decomposes signaling and media into different
    network devices, the protocol used to communicate between the boxes
    frequently will not include SSRC information, making it impossible to
    include in the SDP.  If these devices choose to implement bundling,
    correlation of media streams to m-lines requires an alternate
    correlator.

    These cases (and possibly other similar situations) can be
    ameliorated by using information in the media stream itself as a
    correlator to the SDP offer.  If a packet arrives with an SSRC that
    is not yet associated with an m-line, we would ideally have some
    means of correlating it prior to the arrival of the answer.

    The authors reiterate and emphasize that this technique is used
    solely for the purposes of correlation of an RTP stream to an SDP
    m-line after that stream has already been demultiplexed.
    Demultiplexing of multiple streams on a single transport address
    continues to be based on SSRC values.

This suggests that it is only a problem with legacy devices, but I think 
that is not so. E.g., media can outrun the answer in all cases.

It also seems to exclude the possibility that in some cases we might 
*want* a particular SSRC to be associated with different m-lines over 
time. This is a case that actually comes up in CLUE:

We have an MCU that is providing multiple capture-encodings to a 
consumer for rendering. I think that with the unified plan CLUE would 
map a capture-encoding (conceptually similar to a media stream track) to 
an m-line, and that this would then be rendered to a particular device 
or window.

But the captures being delivered by the MCU may be "switched". For 
instance, one capture-encoding might represent the image of the loudest 
speaker, and another an image of the 2nd loudest speaker. When a 
particular speaker moves from being loudest speaker to 2nd loudest, the 
MCU will want to move the SSRC for the image of that speaker from one 
capture-encoding to another - and hence from one m-line to another.

This seems to be excluded, except by sending a new O/A every time the 
speaker switches, which isn't feasible. If this means that CLUE can't 
represent capture-encodings as m-lines, then there is no benefit to CLUE 
for the unified plan over No Plan (Emil's multiple-sources draft).

Section 3.2.2/3.2.3:

IMO the primary justification for the various multiple-m-line approaches 
is that then one can presume to first associate a packet to an m-line, 
and then transitively associate all the attributes of that m-line with 
that packet. What you describe here is a special case of that.

The unified plan has a spin on this. You preferentially use the SSRC to 
associate a packet to an m-line, and only use other properties of a 
packet if the SSRC-m-line association isn't yet known. But as mentioned 
above, that is a limitation.

Section 3.3:

This falls back to dependency on signaling specific SSRCs. I don't see 
how the other correlation mechanisms mentioned earlier can be employed 
with these features. That can mean confusion for media received before 
the answer.

	Thanks,
	Paul

From adam@nostrum.com  Wed Jul 24 09:03:34 2013
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Subject: Re: [MMUSIC] draft-roach-mmusic-unified-plan-00
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On 7/24/13 10:55, Paul Kyzivat wrote:
> This seems to be excluded, except by sending a new O/A every time the 
> speaker switches, which isn't feasible. If this means that CLUE can't 
> represent capture-encodings as m-lines, then there is no benefit to 
> CLUE for the unified plan over No Plan (Emil's multiple-sources draft). 

That's interesting. I wonder whether we should simply move to the RTP 
header extension described in section 3.2.1.1 as the primary means of 
correlation. It would seem to address the case you describe.

/a

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Date: Wed, 24 Jul 2013 18:09:45 +0200
From: Emil Ivov <emcho@jitsi.org>
Organization: Jitsi
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To: Eric Rescorla <ekr@rtfm.com>
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Cc: =?ISO-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, MMUSIC IETF WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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(redirecting to mmusic)

On 24.07.13, 16:29, Eric Rescorla wrote:
>
>
>
> On Wed, Jul 24, 2013 at 7:08 AM, I=F1aki Baz Castillo <ibc@aliax.net
> <mailto:ibc@aliax.net>> wrote:
>
>     Hi, let's suppose WebRTC 1.0 borns with "Unified Plan" built-in, an=
d
>     my browser receives a session invitation from a conference server, =
and
>     there are 8 audio participants in the conference.
>
>     So the SDP I receive has 8 "m=3Daudio" lines, maybe with different =
audio
>     codecs in each line. And following SDP O/A rules my browser will
>     generate SDP answer that will also contain 8 "m=3Daudio" lines.
>
>     Now, where it is supposed my browser will send my single audio trac=
k
>     to the conference server? What does it mean that I receive 8 "m=3Da=
udio"
>     lines while I'm just able to send a single audio track? which m=3Da=
udio
>     line should I use for sending RTP?
>
>
> One of them will be a=3Dsendrecv. The others will be a=3Dsendonly.
> The sendrecv line is the one you use to send media.

I have a few questions here but before that, could you please clarify=20
the following:

Are you saying that the above is going to be the case on the offer that=20
the callee receives? In other words, the caller "knew" that the callee=20
intended to only send one stream so it preset all directions accordingly?=


Emil



--=20
https://jitsi.org


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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Emil Ivov'" <emcho@jitsi.org>, "'Alan Johnston'" <alan.b.johnston@gmail.com>
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Cc: "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, 'MMUSIC IETF WG' <mmusic@ietf.org>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hi Emil,

As Cullen suggested in case 180 with different to-tag is used by UAS then
INFO package itself is not required right?

Thanks
Partha

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Emil Ivov
> Sent: Wednesday, July 24, 2013 3:31 PM
> To: Alan Johnston
> Cc: Cullen Jennings (fluffy); MMUSIC IETF WG
> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
> 
> Hey Alan, Cullen,
> 
> On 24.07.13, 11:45, Alan Johnston wrote:
> > Agree with Cullen - if there is a reasonable approach (such as
> > retransmitting 180) that avoids PRACK, then we should use this
> approach.
> 
> There is one and it comes down to requiring the remote side to send an
> INFO request as soon as it gets it. The INFO request could contain
> trickled candidates (in the case of full trickle) or just
> end-of-candidates (in the case of half-trickle).
> 
> This could work and, personally, I have no issue with it.
> 
> I believe the suggestion that Christer made was: given how the above
> basically works as a PRACK minus the O/A and given how the O/A in
> PRACKs
> seems to be a deal breaker for many, then we might want to generalize
> the above mechanism so that other specs can also use it.
> 
> Cheers,
> Emil
> 
> 
> > - Alan -
> >
> >
> > On Tue, Jul 23, 2013 at 11:28 PM, Cullen Jennings (fluffy)
> > <fluffy@cisco.com <mailto:fluffy@cisco.com>> wrote:
> >
> >
> >     I prefer the 180 resend approach and all the right things are
> >     already looking at them.  I have always objected to mandating use
> of
> >     PRACK. Obviously I'm fine with things that have PRACK can use it
> but
> >     I want some solution for things that don't.
> >
> >     One small note, the SDP in the 180 can change as long as the to-
> tag
> >     is also changed.
> >
> >
> >
> >     On Jul 3, 2013, at 8:13 AM, Emil Ivov <emcho@jitsi.org
> >     <mailto:emcho@jitsi.org>> wrote:
> >
> >      > Hey all,
> >      >
> >      > Christer, Enrico and I are preparing the next version of
> Trickle
> >     ICE for SIP. Now that discussions on BUNDLE and the plans seem to
> be
> >     winding down, we wanted to run a few questions by the working
> group.
> >      >
> >      > Q1: Making reliable provisional responses and PRACK mandatory.
> >     Obviously this would be nice to avoid, so the question is: is
> there
> >     a reasonable mechanism to achieve this (and by reasonable, we
> mean
> >     something that wouldn't be harder than implementing support for
> PRACK).
> >      >
> >      > There was some discussion about this back in April and there
> was
> >     a suggestion for implementing a 5245-style hack where the
> answerer
> >     basically resends the 180 until it knows that it has been
> received.
> >      >
> >      > 5245 uses connectivity checks for this (i.e. it stops 180
> >     retransmissions when the first connectivity check is received)
> but
> >     we don't have that option here since the 180 may contain either
> none
> >     or only host candidates so there are strong chances that no
> binding
> >     request would be received on them.
> >      >
> >      > Thomas also suggested a second option which would be to also
> use
> >     INFO requests with trickled candidates as an indication that 180
> was
> >     received. This however wouldn't work with half trickle so we are
> >     basically back to vanilla ICE for all (non-re) INVITEs.
> >      >
> >      > Another option would be to mandate an INFO request with
> >     "end-of-candidates" in response to the 180, but that would be
> just
> >     the same as mandating PRACK support.
> >      >
> >      > Thomas also suggested that the answerer can start sending
> INFOs
> >     right after it sends its answer in the 180 and then it can just
> >     resend the 180 if the INFOs result in a 481 response.
> >      >
> >      > Personally I think this could potentially be made to work, but
> it
> >     would imply a level of complexity that considerably exceeds PRACK
> >     support.
> >      >
> >      > Opinions?
> >      >
> >      > Q2: How do we send INFOs? Are they blocking or do we just send
> >     them in parallel? If the latter, then what happens when an INFO
> >     fails because it is received out of order? Do we just tell the
> >     application to resend the candidates asap?
> >      >
> >      > This also leads to the following question:
> >      >
> >      > Q3: What exactly do we send in INFOs? Just the latest batch of
> >     freshly learned candidates or all candidates we've learned so
> far?
> >     Dale suggested that if we do this cumulatively we wouldn't need
> to
> >     worry about the case with the out-of-order INFOs from Q2 since
> the
> >     information gets resent anyway. A drawback here would obviously
> be
> >     that this adds more complexity for trickle ICE users (WebRTC
> >     applications specifically)
> >      >
> >      > A third option would be to allow both and leave it to the
> >     application.
> >      >
> >      > Comments are most welcome!
> >      >
> >      > Emil
> >      >
> >      > --
> >      > https://jitsi.org
> >      > _______________________________________________
> >      > 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
> >
> >
> 
> --
> https://jitsi.org
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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--089e0122f0fedc706604e245e18a
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Offerer has all sendrecv , the answerer has one sendrecv (since he can send
only one audio track) and rest recvonly

so the answerer is ok to receive all the audio tracks sent by the offerer,
and he is sending only one from his side (marked as sendrecv)


On Wed, Jul 24, 2013 at 9:09 AM, Emil Ivov <emcho@jitsi.org> wrote:

> (redirecting to mmusic)
>
>
> On 24.07.13, 16:29, Eric Rescorla wrote:
>
>>
>>
>>
>> On Wed, Jul 24, 2013 at 7:08 AM, I=F1aki Baz Castillo <ibc@aliax.net
>> <mailto:ibc@aliax.net>> wrote:
>>
>>     Hi, let's suppose WebRTC 1.0 borns with "Unified Plan" built-in, and
>>     my browser receives a session invitation from a conference server, a=
nd
>>     there are 8 audio participants in the conference.
>>
>>     So the SDP I receive has 8 "m=3Daudio" lines, maybe with different a=
udio
>>     codecs in each line. And following SDP O/A rules my browser will
>>     generate SDP answer that will also contain 8 "m=3Daudio" lines.
>>
>>     Now, where it is supposed my browser will send my single audio track
>>     to the conference server? What does it mean that I receive 8 "m=3Dau=
dio"
>>     lines while I'm just able to send a single audio track? which m=3Dau=
dio
>>     line should I use for sending RTP?
>>
>>
>> One of them will be a=3Dsendrecv. The others will be a=3Dsendonly.
>> The sendrecv line is the one you use to send media.
>>
>
> I have a few questions here but before that, could you please clarify the
> following:
>
> Are you saying that the above is going to be the case on the offer that
> the callee receives? In other words, the caller "knew" that the callee
> intended to only send one stream so it preset all directions accordingly?
>
> Emil
>
>
>
> --
> https://jitsi.org
>
> ______________________________**_________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/**listinfo/mmusic<https://www.ietf.org/mailm=
an/listinfo/mmusic>
>

--089e0122f0fedc706604e245e18a
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Offerer has all sendrecv , the answerer has one sendrecv (since he can send=
 only one audio track) and rest recvonly<div><br></div><div>so the answerer=
 is ok to receive all the audio tracks sent by the offerer, and he is sendi=
ng only one from his side (marked as sendrecv)</div>
<div><br></div><div><br><div class=3D"gmail_quote">On Wed, Jul 24, 2013 at =
9:09 AM, Emil Ivov <span dir=3D"ltr">&lt;<a href=3D"mailto:emcho@jitsi.org"=
 target=3D"_blank">emcho@jitsi.org</a>&gt;</span> wrote:<br><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">
(redirecting to mmusic)<div class=3D"im"><br>
<br>
On 24.07.13, 16:29, Eric Rescorla wrote:<br>
</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex"><div class=3D"im">
<br>
<br>
<br>
On Wed, Jul 24, 2013 at 7:08 AM, I=F1aki Baz Castillo &lt;<a href=3D"mailto=
:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a><br></div><div><div clas=
s=3D"h5">
&lt;mailto:<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net=
</a>&gt;&gt; wrote:<br>
<br>
=A0 =A0 Hi, let&#39;s suppose WebRTC 1.0 borns with &quot;Unified Plan&quot=
; built-in, and<br>
=A0 =A0 my browser receives a session invitation from a conference server, =
and<br>
=A0 =A0 there are 8 audio participants in the conference.<br>
<br>
=A0 =A0 So the SDP I receive has 8 &quot;m=3Daudio&quot; lines, maybe with =
different audio<br>
=A0 =A0 codecs in each line. And following SDP O/A rules my browser will<br=
>
=A0 =A0 generate SDP answer that will also contain 8 &quot;m=3Daudio&quot; =
lines.<br>
<br>
=A0 =A0 Now, where it is supposed my browser will send my single audio trac=
k<br>
=A0 =A0 to the conference server? What does it mean that I receive 8 &quot;=
m=3Daudio&quot;<br>
=A0 =A0 lines while I&#39;m just able to send a single audio track? which m=
=3Daudio<br>
=A0 =A0 line should I use for sending RTP?<br>
<br>
<br>
One of them will be a=3Dsendrecv. The others will be a=3Dsendonly.<br>
The sendrecv line is the one you use to send media.<br>
</div></div></blockquote>
<br>
I have a few questions here but before that, could you please clarify the f=
ollowing:<br>
<br>
Are you saying that the above is going to be the case on the offer that the=
 callee receives? In other words, the caller &quot;knew&quot; that the call=
ee intended to only send one stream so it preset all directions accordingly=
?<span class=3D"HOEnZb"><font color=3D"#888888"><br>

<br>
Emil<br>
<br>
<br>
<br>
-- <br>
<a href=3D"https://jitsi.org" target=3D"_blank">https://jitsi.org</a><br>
<br>
______________________________<u></u>_________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/mmusic</a><br>
</font></span></blockquote></div><br></div>

--089e0122f0fedc706604e245e18a--

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From: "Vijaya Mandava (vimandav)" <vimandav@cisco.com>
To: Parthasarathi R <partha@parthasarathi.co.in>, "'Emil Ivov'" <emcho@jitsi.org>, "'Alan Johnston'" <alan.b.johnston@gmail.com>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, 'MMUSIC IETF WG' <mmusic@ietf.org>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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180 with different to-tag would mean call is forked.
If uac side do not support call forking, then we cannot use this 180
response to collect trickled candidates.

As Emil said, I think INFO is the right way to go here.

Thanks,
Vijaya

On 7/24/13 12:55 PM, "Parthasarathi R" <partha@parthasarathi.co.in> wrote:

>Hi Emil,
>
>As Cullen suggested in case 180 with different to-tag is used by UAS then
>INFO package itself is not required right?
>
>Thanks
>Partha
>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>> Behalf Of Emil Ivov
>> Sent: Wednesday, July 24, 2013 3:31 PM
>> To: Alan Johnston
>> Cc: Cullen Jennings (fluffy); MMUSIC IETF WG
>> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
>>=20
>> Hey Alan, Cullen,
>>=20
>> On 24.07.13, 11:45, Alan Johnston wrote:
>> > Agree with Cullen - if there is a reasonable approach (such as
>> > retransmitting 180) that avoids PRACK, then we should use this
>> approach.
>>=20
>> There is one and it comes down to requiring the remote side to send an
>> INFO request as soon as it gets it. The INFO request could contain
>> trickled candidates (in the case of full trickle) or just
>> end-of-candidates (in the case of half-trickle).
>>=20
>> This could work and, personally, I have no issue with it.
>>=20
>> I believe the suggestion that Christer made was: given how the above
>> basically works as a PRACK minus the O/A and given how the O/A in
>> PRACKs
>> seems to be a deal breaker for many, then we might want to generalize
>> the above mechanism so that other specs can also use it.
>>=20
>> Cheers,
>> Emil
>>=20
>>=20
>> > - Alan -
>> >
>> >
>> > On Tue, Jul 23, 2013 at 11:28 PM, Cullen Jennings (fluffy)
>> > <fluffy@cisco.com <mailto:fluffy@cisco.com>> wrote:
>> >
>> >
>> >     I prefer the 180 resend approach and all the right things are
>> >     already looking at them.  I have always objected to mandating use
>> of
>> >     PRACK. Obviously I'm fine with things that have PRACK can use it
>> but
>> >     I want some solution for things that don't.
>> >
>> >     One small note, the SDP in the 180 can change as long as the to-
>> tag
>> >     is also changed.
>> >
>> >
>> >
>> >     On Jul 3, 2013, at 8:13 AM, Emil Ivov <emcho@jitsi.org
>> >     <mailto:emcho@jitsi.org>> wrote:
>> >
>> >      > Hey all,
>> >      >
>> >      > Christer, Enrico and I are preparing the next version of
>> Trickle
>> >     ICE for SIP. Now that discussions on BUNDLE and the plans seem to
>> be
>> >     winding down, we wanted to run a few questions by the working
>> group.
>> >      >
>> >      > Q1: Making reliable provisional responses and PRACK mandatory.
>> >     Obviously this would be nice to avoid, so the question is: is
>> there
>> >     a reasonable mechanism to achieve this (and by reasonable, we
>> mean
>> >     something that wouldn't be harder than implementing support for
>> PRACK).
>> >      >
>> >      > There was some discussion about this back in April and there
>> was
>> >     a suggestion for implementing a 5245-style hack where the
>> answerer
>> >     basically resends the 180 until it knows that it has been
>> received.
>> >      >
>> >      > 5245 uses connectivity checks for this (i.e. it stops 180
>> >     retransmissions when the first connectivity check is received)
>> but
>> >     we don't have that option here since the 180 may contain either
>> none
>> >     or only host candidates so there are strong chances that no
>> binding
>> >     request would be received on them.
>> >      >
>> >      > Thomas also suggested a second option which would be to also
>> use
>> >     INFO requests with trickled candidates as an indication that 180
>> was
>> >     received. This however wouldn't work with half trickle so we are
>> >     basically back to vanilla ICE for all (non-re) INVITEs.
>> >      >
>> >      > Another option would be to mandate an INFO request with
>> >     "end-of-candidates" in response to the 180, but that would be
>> just
>> >     the same as mandating PRACK support.
>> >      >
>> >      > Thomas also suggested that the answerer can start sending
>> INFOs
>> >     right after it sends its answer in the 180 and then it can just
>> >     resend the 180 if the INFOs result in a 481 response.
>> >      >
>> >      > Personally I think this could potentially be made to work, but
>> it
>> >     would imply a level of complexity that considerably exceeds PRACK
>> >     support.
>> >      >
>> >      > Opinions?
>> >      >
>> >      > Q2: How do we send INFOs? Are they blocking or do we just send
>> >     them in parallel? If the latter, then what happens when an INFO
>> >     fails because it is received out of order? Do we just tell the
>> >     application to resend the candidates asap?
>> >      >
>> >      > This also leads to the following question:
>> >      >
>> >      > Q3: What exactly do we send in INFOs? Just the latest batch of
>> >     freshly learned candidates or all candidates we've learned so
>> far?
>> >     Dale suggested that if we do this cumulatively we wouldn't need
>> to
>> >     worry about the case with the out-of-order INFOs from Q2 since
>> the
>> >     information gets resent anyway. A drawback here would obviously
>> be
>> >     that this adds more complexity for trickle ICE users (WebRTC
>> >     applications specifically)
>> >      >
>> >      > A third option would be to allow both and leave it to the
>> >     application.
>> >      >
>> >      > Comments are most welcome!
>> >      >
>> >      > Emil
>> >      >
>> >      > --
>> >      > https://jitsi.org
>> >      > _______________________________________________
>> >      > 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
>> >
>> >
>>=20
>> --
>> https://jitsi.org
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www.ietf.org/mailman/listinfo/mmusic


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On Wed, Jul 24, 2013 at 9:09 AM, Emil Ivov <emcho@jitsi.org> wrote:

> (redirecting to mmusic)
>
>
> On 24.07.13, 16:29, Eric Rescorla wrote:
>
>>
>>
>>
>> On Wed, Jul 24, 2013 at 7:08 AM, I=F1aki Baz Castillo <ibc@aliax.net
>> <mailto:ibc@aliax.net>> wrote:
>>
>>     Hi, let's suppose WebRTC 1.0 borns with "Unified Plan" built-in, and
>>     my browser receives a session invitation from a conference server, a=
nd
>>     there are 8 audio participants in the conference.
>>
>>     So the SDP I receive has 8 "m=3Daudio" lines, maybe with different a=
udio
>>     codecs in each line. And following SDP O/A rules my browser will
>>     generate SDP answer that will also contain 8 "m=3Daudio" lines.
>>
>>     Now, where it is supposed my browser will send my single audio track
>>     to the conference server? What does it mean that I receive 8 "m=3Dau=
dio"
>>     lines while I'm just able to send a single audio track? which m=3Dau=
dio
>>     line should I use for sending RTP?
>>
>>
>> One of them will be a=3Dsendrecv. The others will be a=3Dsendonly.
>> The sendrecv line is the one you use to send media.
>>
>
> I have a few questions here but before that, could you please clarify the
> following:
>
> Are you saying that the above is going to be the case on the offer that
> the callee receives? In other words, the caller "knew" that the callee
> intended to only send one stream so it preset all directions accordingly?


It depends on the context obviously. But in the case of (for instance) a
Hangouts
like system if the conference bridge were the offerer, it would send a pile
of
m-lines with only one being sendrecv (or maybe even recvonly) for the
user's media and the rest being sendonly.

-Ekr

--001a11c20dfef9733004e2461e62
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Jul 24, 2013 at 9:09 AM, Emil Ivov <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:emcho@jitsi.org" target=3D"_blank">emcho@jitsi.org</a>&gt;<=
/span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">(redirecting to mmusic)<div class=3D"im"><br=
>
<br>
On 24.07.13, 16:29, Eric Rescorla wrote:<br>
</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex"><div class=3D"im">
<br>
<br>
<br>
On Wed, Jul 24, 2013 at 7:08 AM, I=F1aki Baz Castillo &lt;<a href=3D"mailto=
:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a><br></div><div><div clas=
s=3D"h5">
&lt;mailto:<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net=
</a>&gt;&gt; wrote:<br>
<br>
=A0 =A0 Hi, let&#39;s suppose WebRTC 1.0 borns with &quot;Unified Plan&quot=
; built-in, and<br>
=A0 =A0 my browser receives a session invitation from a conference server, =
and<br>
=A0 =A0 there are 8 audio participants in the conference.<br>
<br>
=A0 =A0 So the SDP I receive has 8 &quot;m=3Daudio&quot; lines, maybe with =
different audio<br>
=A0 =A0 codecs in each line. And following SDP O/A rules my browser will<br=
>
=A0 =A0 generate SDP answer that will also contain 8 &quot;m=3Daudio&quot; =
lines.<br>
<br>
=A0 =A0 Now, where it is supposed my browser will send my single audio trac=
k<br>
=A0 =A0 to the conference server? What does it mean that I receive 8 &quot;=
m=3Daudio&quot;<br>
=A0 =A0 lines while I&#39;m just able to send a single audio track? which m=
=3Daudio<br>
=A0 =A0 line should I use for sending RTP?<br>
<br>
<br>
One of them will be a=3Dsendrecv. The others will be a=3Dsendonly.<br>
The sendrecv line is the one you use to send media.<br>
</div></div></blockquote>
<br>
I have a few questions here but before that, could you please clarify the f=
ollowing:<br>
<br>
Are you saying that the above is going to be the case on the offer that the=
 callee receives? In other words, the caller &quot;knew&quot; that the call=
ee intended to only send one stream so it preset all directions accordingly=
?</blockquote>

<div><br></div><div>It depends on the context obviously. But in the case of=
 (for instance) a Hangouts</div><div>like system if the conference bridge w=
ere the offerer, it would send a pile of</div><div>m-lines with only one be=
ing sendrecv (or maybe even recvonly) for the</div>

<div>user&#39;s media and the rest being sendonly.</div><div><br></div><div=
>-Ekr</div><div><br></div></div></div></div>

--001a11c20dfef9733004e2461e62--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Vijaya Mandava (vimandav)" <vimandav@cisco.com>, Parthasarathi R <partha@parthasarathi.co.in>, 'Emil Ivov' <emcho@jitsi.org>, 'Alan Johnston' <alan.b.johnston@gmail.com>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, 'MMUSIC IETF WG' <mmusic@ietf.org>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hi,

Keep in mind that, no matter if we use PRACK, PRACK-lite, INFO, or simply r=
e-transmission, the SDP can not change from one 180 to another (for the sam=
e To tag, that is).

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Vijaya Mandava (vimandav) [vimandav@cisco.com]
To: Parthasarathi R [partha@parthasarathi.co.in]; &apos;Emil Ivov&apos; [em=
cho@jitsi.org]; &apos;Alan Johnston&apos; [alan.b.johnston@gmail.com]
CC: Cullen Jennings (fluffy) [fluffy@cisco.com]; &apos;MMUSIC IETF WG&apos;=
 [mmusic@ietf.org]
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
180 with different to-tag would mean call is forked.
If uac side do not support call forking, then we cannot use this 180
response to collect trickled candidates.

As Emil said, I think INFO is the right way to go here.

Thanks,
Vijaya

On 7/24/13 12:55 PM, "Parthasarathi R" <partha@parthasarathi.co.in> wrote:

>Hi Emil,
>
>As Cullen suggested in case 180 with different to-tag is used by UAS then
>INFO package itself is not required right?
>
>Thanks
>Partha
>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
>> Behalf Of Emil Ivov
>> Sent: Wednesday, July 24, 2013 3:31 PM
>> To: Alan Johnston
>> Cc: Cullen Jennings (fluffy); MMUSIC IETF WG
>> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
>>
>> Hey Alan, Cullen,
>>
>> On 24.07.13, 11:45, Alan Johnston wrote:
>> > Agree with Cullen - if there is a reasonable approach (such as
>> > retransmitting 180) that avoids PRACK, then we should use this
>> approach.
>>
>> There is one and it comes down to requiring the remote side to send an
>> INFO request as soon as it gets it. The INFO request could contain
>> trickled candidates (in the case of full trickle) or just
>> end-of-candidates (in the case of half-trickle).
>>
>> This could work and, personally, I have no issue with it.
>>
>> I believe the suggestion that Christer made was: given how the above
>> basically works as a PRACK minus the O/A and given how the O/A in
>> PRACKs
>> seems to be a deal breaker for many, then we might want to generalize
>> the above mechanism so that other specs can also use it.
>>
>> Cheers,
>> Emil
>>
>>
>> > - Alan -
>> >
>> >
>> > On Tue, Jul 23, 2013 at 11:28 PM, Cullen Jennings (fluffy)
>> > <fluffy@cisco.com <mailto:fluffy@cisco.com>> wrote:
>> >
>> >
>> >     I prefer the 180 resend approach and all the right things are
>> >     already looking at them.  I have always objected to mandating use
>> of
>> >     PRACK. Obviously I'm fine with things that have PRACK can use it
>> but
>> >     I want some solution for things that don't.
>> >
>> >     One small note, the SDP in the 180 can change as long as the to-
>> tag
>> >     is also changed.
>> >
>> >
>> >
>> >     On Jul 3, 2013, at 8:13 AM, Emil Ivov <emcho@jitsi.org
>> >     <mailto:emcho@jitsi.org>> wrote:
>> >
>> >      > Hey all,
>> >      >
>> >      > Christer, Enrico and I are preparing the next version of
>> Trickle
>> >     ICE for SIP. Now that discussions on BUNDLE and the plans seem to
>> be
>> >     winding down, we wanted to run a few questions by the working
>> group.
>> >      >
>> >      > Q1: Making reliable provisional responses and PRACK mandatory.
>> >     Obviously this would be nice to avoid, so the question is: is
>> there
>> >     a reasonable mechanism to achieve this (and by reasonable, we
>> mean
>> >     something that wouldn't be harder than implementing support for
>> PRACK).
>> >      >
>> >      > There was some discussion about this back in April and there
>> was
>> >     a suggestion for implementing a 5245-style hack where the
>> answerer
>> >     basically resends the 180 until it knows that it has been
>> received.
>> >      >
>> >      > 5245 uses connectivity checks for this (i.e. it stops 180
>> >     retransmissions when the first connectivity check is received)
>> but
>> >     we don't have that option here since the 180 may contain either
>> none
>> >     or only host candidates so there are strong chances that no
>> binding
>> >     request would be received on them.
>> >      >
>> >      > Thomas also suggested a second option which would be to also
>> use
>> >     INFO requests with trickled candidates as an indication that 180
>> was
>> >     received. This however wouldn't work with half trickle so we are
>> >     basically back to vanilla ICE for all (non-re) INVITEs.
>> >      >
>> >      > Another option would be to mandate an INFO request with
>> >     "end-of-candidates" in response to the 180, but that would be
>> just
>> >     the same as mandating PRACK support.
>> >      >
>> >      > Thomas also suggested that the answerer can start sending
>> INFOs
>> >     right after it sends its answer in the 180 and then it can just
>> >     resend the 180 if the INFOs result in a 481 response.
>> >      >
>> >      > Personally I think this could potentially be made to work, but
>> it
>> >     would imply a level of complexity that considerably exceeds PRACK
>> >     support.
>> >      >
>> >      > Opinions?
>> >      >
>> >      > Q2: How do we send INFOs? Are they blocking or do we just send
>> >     them in parallel? If the latter, then what happens when an INFO
>> >     fails because it is received out of order? Do we just tell the
>> >     application to resend the candidates asap?
>> >      >
>> >      > This also leads to the following question:
>> >      >
>> >      > Q3: What exactly do we send in INFOs? Just the latest batch of
>> >     freshly learned candidates or all candidates we've learned so
>> far?
>> >     Dale suggested that if we do this cumulatively we wouldn't need
>> to
>> >     worry about the case with the out-of-order INFOs from Q2 since
>> the
>> >     information gets resent anyway. A drawback here would obviously
>> be
>> >     that this adds more complexity for trickle ICE users (WebRTC
>> >     applications specifically)
>> >      >
>> >      > A third option would be to allow both and leave it to the
>> >     application.
>> >      >
>> >      > Comments are most welcome!
>> >      >
>> >      > Emil
>> >      >
>> >      > --
>> >      > https://jitsi.org
>> >      > _______________________________________________
>> >      > 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
>> >
>> >
>>
>> --
>> https://jitsi.org
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www.ietf.org/mailman/listinfo/mmusic

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

--_000_7594FB04B1934943A5C02806D1A2204B1C405F55ESESSMB209erics_
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Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
ding-left: 4pt; border-left: #800000 2px solid; } --></style>
</head>
<body>
<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">Keep in mind that, no matter if we use PRACK, P=
RACK-lite, INFO, or simply re-transmission, the SDP can not change from one=
 180 to another (for the same To tag, that is).</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Vijaya Mandava (vimandav) [vimandav@cisco.com]<br>
<b>To:</b> Parthasarathi R [partha@parthasarathi.co.in]; &amp;apos;Emil Ivo=
v&amp;apos; [emcho@jitsi.org]; &amp;apos;Alan Johnston&amp;apos; [alan.b.jo=
hnston@gmail.com]<br>
<b>CC:</b> Cullen Jennings (fluffy) [fluffy@cisco.com]; &amp;apos;MMUSIC IE=
TF WG&amp;apos; [mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] Trickle ICE for SIP Questions</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">180 with different to-tag would mean call is forke=
d.<br>
If uac side do not support call forking, then we cannot use this 180<br>
response to collect trickled candidates.<br>
<br>
As Emil said, I think INFO is the right way to go here.<br>
<br>
Thanks,<br>
Vijaya<br>
<br>
On 7/24/13 12:55 PM, &quot;Parthasarathi R&quot; &lt;partha@parthasarathi.c=
o.in&gt; wrote:<br>
<br>
&gt;Hi Emil,<br>
&gt;<br>
&gt;As Cullen suggested in case 180 with different to-tag is used by UAS th=
en<br>
&gt;INFO package itself is not required right?<br>
&gt;<br>
&gt;Thanks<br>
&gt;Partha<br>
&gt;<br>
&gt;&gt; -----Original Message-----<br>
&gt;&gt; From: mmusic-bounces@ietf.org [<a href=3D"mailto:mmusic-bounces@ie=
tf.org">mailto:mmusic-bounces@ietf.org</a>] On<br>
&gt;&gt; Behalf Of Emil Ivov<br>
&gt;&gt; Sent: Wednesday, July 24, 2013 3:31 PM<br>
&gt;&gt; To: Alan Johnston<br>
&gt;&gt; Cc: Cullen Jennings (fluffy); MMUSIC IETF WG<br>
&gt;&gt; Subject: Re: [MMUSIC] Trickle ICE for SIP Questions<br>
&gt;&gt; <br>
&gt;&gt; Hey Alan, Cullen,<br>
&gt;&gt; <br>
&gt;&gt; On 24.07.13, 11:45, Alan Johnston wrote:<br>
&gt;&gt; &gt; Agree with Cullen - if there is a reasonable approach (such a=
s<br>
&gt;&gt; &gt; retransmitting 180) that avoids PRACK, then we should use thi=
s<br>
&gt;&gt; approach.<br>
&gt;&gt; <br>
&gt;&gt; There is one and it comes down to requiring the remote side to sen=
d an<br>
&gt;&gt; INFO request as soon as it gets it. The INFO request could contain=
<br>
&gt;&gt; trickled candidates (in the case of full trickle) or just<br>
&gt;&gt; end-of-candidates (in the case of half-trickle).<br>
&gt;&gt; <br>
&gt;&gt; This could work and, personally, I have no issue with it.<br>
&gt;&gt; <br>
&gt;&gt; I believe the suggestion that Christer made was: given how the abo=
ve<br>
&gt;&gt; basically works as a PRACK minus the O/A and given how the O/A in<=
br>
&gt;&gt; PRACKs<br>
&gt;&gt; seems to be a deal breaker for many, then we might want to general=
ize<br>
&gt;&gt; the above mechanism so that other specs can also use it.<br>
&gt;&gt; <br>
&gt;&gt; Cheers,<br>
&gt;&gt; Emil<br>
&gt;&gt; <br>
&gt;&gt; <br>
&gt;&gt; &gt; - Alan -<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt; On Tue, Jul 23, 2013 at 11:28 PM, Cullen Jennings (fluffy)<br=
>
&gt;&gt; &gt; &lt;fluffy@cisco.com &lt;<a href=3D"mailto:fluffy@cisco.com">=
mailto:fluffy@cisco.com</a>&gt;&gt; wrote:<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; I prefer the 180 resend approach and =
all the right things are<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; already looking at them.&nbsp; I have=
 always objected to mandating use<br>
&gt;&gt; of<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; PRACK. Obviously I'm fine with things=
 that have PRACK can use it<br>
&gt;&gt; but<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; I want some solution for things that =
don't.<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; One small note, the SDP in the 180 ca=
n change as long as the to-<br>
&gt;&gt; tag<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; is also changed.<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; On Jul 3, 2013, at 8:13 AM, Emil Ivov=
 &lt;emcho@jitsi.org<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; &lt;<a href=3D"mailto:emcho@jitsi.org=
">mailto:emcho@jitsi.org</a>&gt;&gt; wrote:<br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Hey all,<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Christer, Enrico and I are=
 preparing the next version of<br>
&gt;&gt; Trickle<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; ICE for SIP. Now that discussions on =
BUNDLE and the plans seem to<br>
&gt;&gt; be<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; winding down, we wanted to run a few =
questions by the working<br>
&gt;&gt; group.<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Q1: Making reliable provis=
ional responses and PRACK mandatory.<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; Obviously this would be nice to avoid=
, so the question is: is<br>
&gt;&gt; there<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; a reasonable mechanism to achieve thi=
s (and by reasonable, we<br>
&gt;&gt; mean<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; something that wouldn't be harder tha=
n implementing support for<br>
&gt;&gt; PRACK).<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; There was some discussion =
about this back in April and there<br>
&gt;&gt; was<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; a suggestion for implementing a 5245-=
style hack where the<br>
&gt;&gt; answerer<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; basically resends the 180 until it kn=
ows that it has been<br>
&gt;&gt; received.<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; 5245 uses connectivity che=
cks for this (i.e. it stops 180<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; retransmissions when the first connec=
tivity check is received)<br>
&gt;&gt; but<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; we don't have that option here since =
the 180 may contain either<br>
&gt;&gt; none<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; or only host candidates so there are =
strong chances that no<br>
&gt;&gt; binding<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; request would be received on them.<br=
>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Thomas also suggested a se=
cond option which would be to also<br>
&gt;&gt; use<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; INFO requests with trickled candidate=
s as an indication that 180<br>
&gt;&gt; was<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; received. This however wouldn't work =
with half trickle so we are<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; basically back to vanilla ICE for all=
 (non-re) INVITEs.<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Another option would be to=
 mandate an INFO request with<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; &quot;end-of-candidates&quot; in resp=
onse to the 180, but that would be<br>
&gt;&gt; just<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; the same as mandating PRACK support.<=
br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Thomas also suggested that=
 the answerer can start sending<br>
&gt;&gt; INFOs<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; right after it sends its answer in th=
e 180 and then it can just<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; resend the 180 if the INFOs result in=
 a 481 response.<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Personally I think this co=
uld potentially be made to work, but<br>
&gt;&gt; it<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; would imply a level of complexity tha=
t considerably exceeds PRACK<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; support.<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Opinions?<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Q2: How do we send INFOs? =
Are they blocking or do we just send<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; them in parallel? If the latter, then=
 what happens when an INFO<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; fails because it is received out of o=
rder? Do we just tell the<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; application to resend the candidates =
asap?<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; This also leads to the fol=
lowing question:<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Q3: What exactly do we sen=
d in INFOs? Just the latest batch of<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; freshly learned candidates or all can=
didates we've learned so<br>
&gt;&gt; far?<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; Dale suggested that if we do this cum=
ulatively we wouldn't need<br>
&gt;&gt; to<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; worry about the case with the out-of-=
order INFOs from Q2 since<br>
&gt;&gt; the<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; information gets resent anyway. A dra=
wback here would obviously<br>
&gt;&gt; be<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; that this adds more complexity for tr=
ickle ICE users (WebRTC<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; applications specifically)<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; A third option would be to=
 allow both and leave it to the<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; application.<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Comments are most welcome!=
<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Emil<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; --<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; <a href=3D"https://jitsi.o=
rg">https://jitsi.org</a><br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; __________________________=
_____________________<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; mmusic mailing list<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; mmusic@ietf.org &lt;<a hre=
f=3D"mailto:mmusic@ietf.org">mailto:mmusic@ietf.org</a>&gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; <a href=3D"https://www.iet=
f.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic=
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&gt;&gt; &gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; _____________________________________=
__________<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; mmusic mailing list<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; mmusic@ietf.org &lt;<a href=3D"mailto=
:mmusic@ietf.org">mailto:mmusic@ietf.org</a>&gt;<br>
&gt;&gt; &gt;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"https://www.ietf.org/mailm=
an/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
&gt;&gt; &gt;<br>
&gt;&gt; &gt;<br>
&gt;&gt; <br>
&gt;&gt; --<br>
&gt;&gt; <a href=3D"https://jitsi.org">https://jitsi.org</a><br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; mmusic mailing list<br>
&gt;&gt; mmusic@ietf.org<br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://w=
ww.ietf.org/mailman/listinfo/mmusic</a><br>
&gt;<br>
&gt;_______________________________________________<br>
&gt;mmusic mailing list<br>
&gt;mmusic@ietf.org<br>
&gt;<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.ie=
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<br>
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Vijaya Mandava (vimandav)" <vimandav@cisco.com>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
Thread-Index: AQHOiKfcSVOSvmOQIEyFWeTagsf8fQ==
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Cc: Alan Johnston <alan.b.johnston@gmail.com>, MMUSIC IETF WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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On Jul 24, 2013, at 12:22 PM, Vijaya Mandava (vimandav) <vimandav@cisco.com=
> wrote:

> 180 with different to-tag would mean call is forked.
> If uac side do not support call forking, then we cannot use this 180
> response to collect trickled candidates.

If it is a UAC, it supports this so I don't see the problem.


From christer.holmberg@ericsson.com  Wed Jul 24 13:34:48 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, "Vijaya Mandava (vimandav)" <vimandav@cisco.com>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hi,

Wouldn't the UAC need to know that the 180s come from the same UAS?

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Cullen Jennings (fluffy) [fluffy@cisco.com]
To: Vijaya Mandava (vimandav) [vimandav@cisco.com]
CC: Alan Johnston [alan.b.johnston@gmail.com]; MMUSIC IETF WG [mmusic@ietf.=
org]
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions

On Jul 24, 2013, at 12:22 PM, Vijaya Mandava (vimandav) <vimandav@cisco.com=
> wrote:

> 180 with different to-tag would mean call is forked.
> If uac side do not support call forking, then we cannot use this 180
> response to collect trickled candidates.

If it is a UAC, it supports this so I don't see the problem.

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<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">Wouldn't the UAC need to know that the 180s com=
e from the same UAS?</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
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<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Cullen Jennings (fluffy) [fluffy@cisco.com]<br>
<b>To:</b> Vijaya Mandava (vimandav) [vimandav@cisco.com]<br>
<b>CC:</b> Alan Johnston [alan.b.johnston@gmail.com]; MMUSIC IETF WG [mmusi=
c@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] Trickle ICE for SIP Questions</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
On Jul 24, 2013, at 12:22 PM, Vijaya Mandava (vimandav) &lt;vimandav@cisco.=
com&gt; wrote:<br>
<br>
&gt; 180 with different to-tag would mean call is forked.<br>
&gt; If uac side do not support call forking, then we cannot use this 180<b=
r>
&gt; response to collect trickled candidates.<br>
<br>
If it is a UAC, it supports this so I don't see the problem.<br>
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From emil@sip-communicator.org  Wed Jul 24 13:49:23 2013
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Date: Wed, 24 Jul 2013 22:49:12 +0200
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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On 24.07.13, 21:56, Cullen Jennings (fluffy) wrote:
>
> On Jul 24, 2013, at 12:22 PM, Vijaya Mandava (vimandav) <vimandav@cisco.com> wrote:
>
>> 180 with different to-tag would mean call is forked.
>> If uac side do not support call forking, then we cannot use this 180
>> response to collect trickled candidates.
>
> If it is a UAC, it supports this so I don't see the problem.

One problem is that the UAC could support it to the extent of making it 
visible to the user:

   "Your contact is being alerted on these N locations"

And of course there's also the fact that we would be creating several 
dialogs per call, that we'd then need to clean and garbage collect.

Why would we want to go there?

Emil

-- 
https://jitsi.org

From christer.holmberg@ericsson.com  Wed Jul 24 13:58:02 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Emil Ivov <emcho@jitsi.org>, "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
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There is a very nice, easy and lightweight mechanism for cleaning: 199 :)

Regards,

Christer


Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Emil Ivov [emcho@jitsi.org]
To: Cullen Jennings (fluffy) [fluffy@cisco.com]
CC: Alan Johnston [alan.b.johnston@gmail.com]; MMUSIC IETF WG [mmusic@ietf.=
org]
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions


On 24.07.13, 21:56, Cullen Jennings (fluffy) wrote:
>
> On Jul 24, 2013, at 12:22 PM, Vijaya Mandava (vimandav) <vimandav@cisco.c=
om> wrote:
>
>> 180 with different to-tag would mean call is forked.
>> If uac side do not support call forking, then we cannot use this 180
>> response to collect trickled candidates.
>
> If it is a UAC, it supports this so I don't see the problem.

One problem is that the UAC could support it to the extent of making it
visible to the user:

   "Your contact is being alerted on these N locations"

And of course there's also the fact that we would be creating several
dialogs per call, that we'd then need to clean and garbage collect.

Why would we want to go there?

Emil

--
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<html>
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<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
ding-left: 4pt; border-left: #800000 2px solid; } --></style>
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<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">There is a very nice, easy and lightweight&nbsp;mecha=
nism for cleaning: 199 :)</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Emil Ivov [emcho@jitsi.org]<br>
<b>To:</b> Cullen Jennings (fluffy) [fluffy@cisco.com]<br>
<b>CC:</b> Alan Johnston [alan.b.johnston@gmail.com]; MMUSIC IETF WG [mmusi=
c@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] Trickle ICE for SIP Questions</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
<br>
On 24.07.13, 21:56, Cullen Jennings (fluffy) wrote:<br>
&gt;<br>
&gt; On Jul 24, 2013, at 12:22 PM, Vijaya Mandava (vimandav) &lt;vimandav@c=
isco.com&gt; wrote:<br>
&gt;<br>
&gt;&gt; 180 with different to-tag would mean call is forked.<br>
&gt;&gt; If uac side do not support call forking, then we cannot use this 1=
80<br>
&gt;&gt; response to collect trickled candidates.<br>
&gt;<br>
&gt; If it is a UAC, it supports this so I don't see the problem.<br>
<br>
One problem is that the UAC could support it to the extent of making it <br=
>
visible to the user:<br>
<br>
&nbsp;&nbsp; &quot;Your contact is being alerted on these N locations&quot;=
<br>
<br>
And of course there's also the fact that we would be creating several <br>
dialogs per call, that we'd then need to clean and garbage collect.<br>
<br>
Why would we want to go there?<br>
<br>
Emil<br>
<br>
-- <br>
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From pkyzivat@alum.mit.edu  Wed Jul 24 14:01:36 2013
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Subject: [MMUSIC] draft-ietf-mmusic-sdp-bundle-negotiation-03
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Christer,

Here are some points I encountered while reviewing this version in prep 
for the meeting:

I trust that the requirement for identical c= values may be met by 
putting one c= at session level. Might be good to say that somewhere. It 
ought to be obvious, but some people make strange assumptions when 
things aren't explicit.

Open Issue #1: AFAIK grouping semantics requires the answer to include 
the same, or a subset, of the mids in the offer. So I think the only 
question here is whether it can be a subset. IMO the answer to that is 
YES, it can be a subset. Some may be dropped due to having been refused 
(port zero), and others just because there is a desire to have those 
particular ones unbundled.

Open Issue #2: I don't understand the issue. Why is this any different? 
Just follow the rules for establishing a new bundle.

Open Issue #3: Why not? Of course will need to supply a unique addr/port 
for each m-line, but that is nothing special. I also see no reason why 
you can't retain the bundle while assigning unique addr/port to each 
m-line, thus restarting the bundle negotiation. While I don't know why 
you might want to do that I see no reason to forbid it. (One reason to 
do that might be a 3pcc transfer.)

Section 6.3 should mention the possibility of accepting the bundle but 
omitting some of the m-lines from it.

	Thanks,
	Paul

From emil@sip-communicator.org  Wed Jul 24 14:17:49 2013
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Message-ID: <51F04473.5020401@jitsi.org>
Date: Wed, 24 Jul 2013 23:17:39 +0200
From: Emil Ivov <emcho@jitsi.org>
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To: Eric Rescorla <ekr@rtfm.com>
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Cc: =?ISO-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, MMUSIC IETF WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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On 24.07.13, 20:31, Eric Rescorla wrote:
>
>
>
> On Wed, Jul 24, 2013 at 9:09 AM, Emil Ivov <emcho@jitsi.org
> <mailto:emcho@jitsi.org>> wrote:
>
>     (redirecting to mmusic)
>
>
>     On 24.07.13, 16:29, Eric Rescorla wrote:
>
>
>
>
>         On Wed, Jul 24, 2013 at 7:08 AM, I=F1aki Baz Castillo
>         <ibc@aliax.net <mailto:ibc@aliax.net>
>         <mailto:ibc@aliax.net <mailto:ibc@aliax.net>>> wrote:
>
>              Hi, let's suppose WebRTC 1.0 borns with "Unified Plan"
>         built-in, and
>              my browser receives a session invitation from a conference=

>         server, and
>              there are 8 audio participants in the conference.
>
>              So the SDP I receive has 8 "m=3Daudio" lines, maybe with
>         different audio
>              codecs in each line. And following SDP O/A rules my browse=
r
>         will
>              generate SDP answer that will also contain 8 "m=3Daudio" l=
ines.
>
>              Now, where it is supposed my browser will send my single
>         audio track
>              to the conference server? What does it mean that I receive=

>         8 "m=3Daudio"
>              lines while I'm just able to send a single audio track?
>         which m=3Daudio
>              line should I use for sending RTP?
>
>
>         One of them will be a=3Dsendrecv. The others will be a=3Dsendon=
ly.
>         The sendrecv line is the one you use to send media.
>
>
>     I have a few questions here but before that, could you please
>     clarify the following:
>
>     Are you saying that the above is going to be the case on the offer
>     that the callee receives? In other words, the caller "knew" that th=
e
>     callee intended to only send one stream so it preset all directions=

>     accordingly?
>
>
> It depends on the context obviously. But in the case of (for instance) =
a
> Hangouts
> like system if the conference bridge were the offerer, it would send a
> pile of
> m-lines with only one being sendrecv (or maybe even recvonly) for the
> user's media and the rest being sendonly.

Sorry, I am still not getting it. How does it know exactly how many "m=3D=
"=20
lines to place in a=3Dsendrecv? How does it know if the callee is plannin=
g=20
on sending a webcam, or two, or webcam+slides? Or maybe 8 other streams=20
because the callee turns out to be another MCU?

When you say that this depends on the context, do you mean that it has=20
to be hardcoded/pre-provisioned or handled by app-specific signalling?

Emil


--=20
https://jitsi.org


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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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On 7/24/13 16:17, Emil Ivov wrote:
>
>
> On 24.07.13, 20:31, Eric Rescorla wrote:
>>
>> It depends on the context obviously. But in the case of (for instance) a
>> Hangouts
>> like system if the conference bridge were the offerer, it would send a
>> pile of
>> m-lines with only one being sendrecv (or maybe even recvonly) for the
>> user's media and the rest being sendonly.
>
> Sorry, I am still not getting it. How does it know exactly how many 
> "m=" lines to place in a=sendrecv?

I think you need to re-read and meditate on section 5.1 of RFC 3264. 
This stuff is actually pretty well defined, and I'm not sure having an 
on-list tutorial is ideal.

Your follow-up questions are likely answered in section 6.1 of that same 
document.

/a

From christer.holmberg@ericsson.com  Wed Jul 24 14:43:45 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, IETF MMUSIC WG <mmusic@ietf.org>
Thread-Topic: draft-ietf-mmusic-sdp-bundle-negotiation-03
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Subject: Re: [MMUSIC] draft-ietf-mmusic-sdp-bundle-negotiation-03
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Hi Paul,

I will look at your e-mail later, but I just want to make sure it is based =
on version -04, as the subject refers to -03 :)

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Paul Kyzivat [pkyzivat@alum.mit.edu]
To: IETF MMUSIC WG [mmusic@ietf.org]; Christer Holmberg [christer.holmberg@=
ericsson.com]
Subject: draft-ietf-mmusic-sdp-bundle-negotiation-03
Christer,

Here are some points I encountered while reviewing this version in prep
for the meeting:

I trust that the requirement for identical c=3D values may be met by
putting one c=3D at session level. Might be good to say that somewhere. It
ought to be obvious, but some people make strange assumptions when
things aren't explicit.

Open Issue #1: AFAIK grouping semantics requires the answer to include
the same, or a subset, of the mids in the offer. So I think the only
question here is whether it can be a subset. IMO the answer to that is
YES, it can be a subset. Some may be dropped due to having been refused
(port zero), and others just because there is a desire to have those
particular ones unbundled.

Open Issue #2: I don't understand the issue. Why is this any different?
Just follow the rules for establishing a new bundle.

Open Issue #3: Why not? Of course will need to supply a unique addr/port
for each m-line, but that is nothing special. I also see no reason why
you can't retain the bundle while assigning unique addr/port to each
m-line, thus restarting the bundle negotiation. While I don't know why
you might want to do that I see no reason to forbid it. (One reason to
do that might be a 3pcc transfer.)

Section 6.3 should mention the possibility of accepting the bundle but
omitting some of the m-lines from it.

        Thanks,
        Paul

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1">
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
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</head>
<body>
<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi Paul,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">I will look at your e-mail later, but I just wa=
nt to make sure it is based on version -04, as the subject refers to -03 :)=
</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Paul Kyzivat [pkyzivat@alum.mit.edu]<br>
<b>To:</b> IETF MMUSIC WG [mmusic@ietf.org]; Christer Holmberg [christer.ho=
lmberg@ericsson.com]<br>
<b>Subject:</b> draft-ietf-mmusic-sdp-bundle-negotiation-03</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">Christer,<br>
<br>
Here are some points I encountered while reviewing this version in prep <br=
>
for the meeting:<br>
<br>
I trust that the requirement for identical c=3D values may be met by <br>
putting one c=3D at session level. Might be good to say that somewhere. It =
<br>
ought to be obvious, but some people make strange assumptions when <br>
things aren't explicit.<br>
<br>
Open Issue #1: AFAIK grouping semantics requires the answer to include <br>
the same, or a subset, of the mids in the offer. So I think the only <br>
question here is whether it can be a subset. IMO the answer to that is <br>
YES, it can be a subset. Some may be dropped due to having been refused <br=
>
(port zero), and others just because there is a desire to have those <br>
particular ones unbundled.<br>
<br>
Open Issue #2: I don't understand the issue. Why is this any different? <br=
>
Just follow the rules for establishing a new bundle.<br>
<br>
Open Issue #3: Why not? Of course will need to supply a unique addr/port <b=
r>
for each m-line, but that is nothing special. I also see no reason why <br>
you can't retain the bundle while assigning unique addr/port to each <br>
m-line, thus restarting the bundle negotiation. While I don't know why <br>
you might want to do that I see no reason to forbid it. (One reason to <br>
do that might be a 3pcc transfer.)<br>
<br>
Section 6.3 should mention the possibility of accepting the bundle but <br>
omitting some of the m-lines from it.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
</div>
</span></font>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C408670ESESSMB209erics_--

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Date: Wed, 24 Jul 2013 17:48:34 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [MMUSIC] draft-ietf-mmusic-sdp-bundle-negotiation-03
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On 7/24/13 5:43 PM, Christer Holmberg wrote:
> Hi Paul,
> I will look at your e-mail later, but I just want to make sure it is
> based on version -04, as the subject refers to -03 :)

Sorry. Yes, I reviewed -04 and just gave the wrong number.
I was looking at a diff, and probably copied the number from the wrong pane.

	Sorry,
	Paul

> Regards,
> Christer
>
>
> Sent from */Windows/* using *TouchDown* (www.nitrodesk.com)
>
> -----Original Message-----
> *From:* Paul Kyzivat [pkyzivat@alum.mit.edu]
> *To:* IETF MMUSIC WG [mmusic@ietf.org]; Christer Holmberg
> [christer.holmberg@ericsson.com]
> *Subject:* draft-ietf-mmusic-sdp-bundle-negotiation-03
> Christer,
>
> Here are some points I encountered while reviewing this version in prep
> for the meeting:
>
> I trust that the requirement for identical c= values may be met by
> putting one c= at session level. Might be good to say that somewhere. It
> ought to be obvious, but some people make strange assumptions when
> things aren't explicit.
>
> Open Issue #1: AFAIK grouping semantics requires the answer to include
> the same, or a subset, of the mids in the offer. So I think the only
> question here is whether it can be a subset. IMO the answer to that is
> YES, it can be a subset. Some may be dropped due to having been refused
> (port zero), and others just because there is a desire to have those
> particular ones unbundled.
>
> Open Issue #2: I don't understand the issue. Why is this any different?
> Just follow the rules for establishing a new bundle.
>
> Open Issue #3: Why not? Of course will need to supply a unique addr/port
> for each m-line, but that is nothing special. I also see no reason why
> you can't retain the bundle while assigning unique addr/port to each
> m-line, thus restarting the bundle negotiation. While I don't know why
> you might want to do that I see no reason to forbid it. (One reason to
> do that might be a 3pcc transfer.)
>
> Section 6.3 should mention the possibility of accepting the bundle but
> omitting some of the m-lines from it.
>
>          Thanks,
>          Paul


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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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On 24.07.13, 23:27, Adam Roach wrote:
> On 7/24/13 16:17, Emil Ivov wrote:
>>
>>
>> On 24.07.13, 20:31, Eric Rescorla wrote:
>>>
>>> It depends on the context obviously. But in the case of (for instance=
) a
>>> Hangouts
>>> like system if the conference bridge were the offerer, it would send =
a
>>> pile of
>>> m-lines with only one being sendrecv (or maybe even recvonly) for the=

>>> user's media and the rest being sendonly.
>>
>> Sorry, I am still not getting it. How does it know exactly how many
>> "m=3D" lines to place in a=3Dsendrecv?
>
> I think you need to re-read and meditate on section 5.1 of RFC 3264.
> This stuff is actually pretty well defined, and I'm not sure having an
> on-list tutorial is ideal.
>
> Your follow-up questions are likely answered in section 6.1 of that sam=
e
> document.

Sorry Adam, I must be having my dumb-day today, but I don't see how 3264 =

is even related to what I am asking.

The original query here was about a bridge that invites an endpoint into =

a conference using "unified". The bridge knows that it will be streaming =

8 flows (e.g. from 8 participants) so it creates 8 "m=3D" lines.

I believe I=F1aki's question was: if the callee only has one stream to=20
send, how does it pick an "m=3D" line to associate it with (e.g. for the =

purposes of codec selection).

Eric's answer was: the Offer from the bridge would have marked *one*,=20
and only one of the "m=3D" lines as "a=3Dsendrecv", so that's the line yo=
u use.

My follow up question was then: how did the bridge know in advance, that =

the callee only had one media source and not any other number.

This is not a loaded question. I am trying to figure out what the best=20
practices with "unified" are.

Again, I might be missing something, so if you feel this would be=20
perfectly obvious to everyone else, then please ignore.

Emil

--=20
https://jitsi.org


From adam@nostrum.com  Wed Jul 24 15:22:02 2013
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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On 7/24/13 17:01, Emil Ivov wrote:
>
> Sorry Adam, I must be having my dumb-day today, but I don't see how 
> 3264 is even related to what I am asking.

Did you even look at the sections that I specifically cited? They 
discuss precisely how you determine how to set 
sendrecv/sendonly/recvonly/inactive attributes.

In the example EKR gave, the bridge is only asking for one video stream 
because that's what makes sense for *it*. It's not guessing how many 
cameras you might have.

/a

From emil@sip-communicator.org  Wed Jul 24 15:58:59 2013
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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On 25.07.13, 00:21, Adam Roach wrote:
> On 7/24/13 17:01, Emil Ivov wrote:
>>
>> Sorry Adam, I must be having my dumb-day today, but I don't see how
>> 3264 is even related to what I am asking.
>
> Did you even look at the sections that I specifically cited?

I did and they were delightful! Thanks for recommending!

> They
> discuss precisely how you determine how to set
> sendrecv/sendonly/recvonly/inactive attributes.

I don't think this is accurate. At least not in the context of the=20
current scenario. What these sections talk about is how to set the=20
direction attribute once you *know* what direction you'd like to have=20
for your streams.

It does not talk about how you get to know what you want. Obviously this =

would be quite out of scope there.

> In the example EKR gave, the bridge is only asking for one video stream=

> because that's what makes sense for *it*. It's not guessing how many
> cameras you might have.

So you are confirming that the number of media sources, that the bridge=20
expected to get from the callee, was hard-coded or pre-provisioned. Right=
?

If so, then that's cool and I off to my next question:

How would you handle the case where such pre-provisioning is impossible=20
and a callee could respond with any number of sources.

To simplify let's get back to I=F1aki's scenario of 8 "m=3D" lines in the=
=20
offer, but lets also assume that different callees can answer with=20
anything between 1 and 8 sources.

In this case, do you set all your "m=3D" lines to "a=3Dsendrecv"?

If so, we are back to I=F1aki's original question: how does a=20
single-source callee know which if these "m=3D" lines to pick for its own=
=20
stream?

I suppose one possible answer would be: it just takes any "m=3D" line=20
that matches its needs and sends stuff there. It also sets everything=20
else to "a=3Drecvonly" in the answer. Is this what you'd suggest?

Or maybe the callee sends an answer where it sets everything to=20
"a=3Drecvonly" and then re-offers with a ninth "a=3Dsendonly" "m=3D" line=
?

Or does the callee use app-specific signalling to tell the bridge that=20
it has N sources so that the bridge would offer with N "a=3Dsendrecv" "m=3D=
"=20
lines and 8-N "a=3Dsendonly" "m=3D" lines?

Or are all three of the above legal?

Emil

--=20
https://jitsi.org


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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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On 7/24/13 17:58, Emil Ivov wrote:
> So you are confirming that the number of media sources, that the 
> bridge expected to get from the callee, was hard-coded or 
> pre-provisioned. Right?

It's part of the application. Sure. "Hard-coded" is a reasonable way to 
think about it.

> To simplify let's get back to Iñaki's scenario of 8 "m=" lines in the 
> offer, but lets also assume that different callees can answer with 
> anything between 1 and 8 sources.
>
> In this case, do you set all your "m=" lines to "a=sendrecv"?

If the bridge wanted to deal with receiving that many media streams, sure.

> If so, we are back to Iñaki's original question: how does a 
> single-source callee know which if these "m=" lines to pick for its 
> own stream?
>
> I suppose one possible answer would be: it just takes any "m=" line 
> that matches its needs and sends stuff there. It also sets everything 
> else to "a=recvonly" in the answer. Is this what you'd suggest?
>
> Or maybe the callee sends an answer where it sets everything to 
> "a=recvonly" and then re-offers with a ninth "a=sendonly" "m=" line?
>
> Or does the callee use app-specific signalling to tell the bridge that 
> it has N sources so that the bridge would offer with N "a=sendrecv" 
> "m=" lines and 8-N "a=sendonly" "m=" lines?
>
> Or are all three of the above legal?

Yes, the first one is what I would recommend.

The second one is certainly valid, but why would you do that? It delays 
the callee-to-caller media unnecessarily.

The third one is fine, too, assuming you want to bind yourself to that 
kind of signaling (it may complicate interdomain federation, if that 
ever becomes important to you).

But this is all pretty easy to derive from RFC 3264 section 6.1, really.

/a

From randy@qti.qualcomm.com  Wed Jul 24 17:38:33 2013
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Lots of good discussion, especially on the issues of complexity vs 
completeness, preferences (none, 3-level, q-value), and back to if 
SDP is the right choice.

I'm thinking another face-to-face discussion, such as we did in 
Orlando, would be helpful in moving towards resolution.  I don't 
think the 15 minutes on the agenda at mmusic will be sufficient for 
this, so I've set a Doodle poll on a time in Berlin.  Please use the 
link below to indicate when you can make it.

http://doodle.com/u6vcppxw3tuncbcd

-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly selected tag: ---------------
The most savage controversies are those about matters as to which there
is no good evidence either way.
    --Bertrand Russell

From partha@parthasarathi.co.in  Thu Jul 25 08:48:54 2013
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Emil Ivov'" <emcho@jitsi.org>, "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>
References: <1CDFD781608D924094E43F573C351961BDE2DC@xmb-rcd-x13.cisco.com> <C5E08FE080ACFD4DAE31E4BDBF944EB11360D0D2@xmb-aln-x02.cisco.com> <51F03DC8.70908@jitsi.org>
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hi Emil,

In the mentioned situation, different To-tag with same contact header for
trickled SDP, So there will be no "N" locations. 

My reading of the draft is that this draft is not required in case UPDATE
(RFC 3311) method is supported. UPDATE with SDP will update trickle
candidate within 18x. 

The aim of the draft is to support Trickle ICE for SIP user agent which does
not support RFC 3311 for last 11 years. AFAIK, different "to tag" is the
simpler way to solve the problem in UAS side. I have concern w.r.t INFO
mechanism as it adds one method for handling SDP and complicates the
existing SDP O/A handling (RFC 6337).

Including Christer in this mail thread as he replied in the parallel mail
thread.

Thanks
Partha

> -----Original Message-----
> From: Emil Ivov [mailto:emcho@jitsi.org]
> Sent: Thursday, July 25, 2013 2:19 AM
> To: Cullen Jennings (fluffy)
> Cc: Vijaya Mandava (vimandav); Parthasarathi R; Alan Johnston; MMUSIC
> IETF WG
> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
> 
> 
> 
> On 24.07.13, 21:56, Cullen Jennings (fluffy) wrote:
> >
> > On Jul 24, 2013, at 12:22 PM, Vijaya Mandava (vimandav)
> <vimandav@cisco.com> wrote:
> >
> >> 180 with different to-tag would mean call is forked.
> >> If uac side do not support call forking, then we cannot use this 180
> >> response to collect trickled candidates.
> >
> > If it is a UAC, it supports this so I don't see the problem.
> 
> One problem is that the UAC could support it to the extent of making it
> visible to the user:
> 
>    "Your contact is being alerted on these N locations"
> 
> And of course there's also the fact that we would be creating several
> dialogs per call, that we'd then need to clean and garbage collect.
> 
> Why would we want to go there?
> 
> Emil
> 
> --
> https://jitsi.org


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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Cullen Jennings \(fluffy\)'" <fluffy@cisco.com>, "'Vijaya Mandava \(vimandav\)'" <vimandav@cisco.com>
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Hi Christer,

 

The only way to identify whether it is same from UAS is to look into contact
header/IP address of the SIP response. In this situation, contact header and
IP address of SIP response is same all for "to-tag".

 

My concern is w.r.t INFO with SDP handling. I have written in the different
thread.

 

Thanks

Partha

 

From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of
Christer Holmberg
Sent: Thursday, July 25, 2013 2:05 AM
To: Cullen Jennings (fluffy); Vijaya Mandava (vimandav)
Cc: Alan Johnston; MMUSIC IETF WG
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions

 

Hi,

 

Wouldn't the UAC need to know that the 180s come from the same UAS?

 

Regards,

 

Christer

 



Sent from Windows using TouchDown (www.nitrodesk.com)


-----Original Message-----
From: Cullen Jennings (fluffy) [fluffy@cisco.com]
To: Vijaya Mandava (vimandav) [vimandav@cisco.com]
CC: Alan Johnston [alan.b.johnston@gmail.com]; MMUSIC IETF WG
[mmusic@ietf.org]
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions


On Jul 24, 2013, at 12:22 PM, Vijaya Mandava (vimandav) <vimandav@cisco.com>
wrote:

> 180 with different to-tag would mean call is forked.
> If uac side do not support call forking, then we cannot use this 180
> response to collect trickled candidates.

If it is a UAC, it supports this so I don't see the problem.

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


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<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Christer,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=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:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>The only way to identify whether it is same from UAS is =
to look
into contact header/IP address of the SIP response. In this situation, =
contact
header and IP address of SIP response is same all for =
&#8220;to-tag&#8221;.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=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:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>My concern is w.r.t INFO with SDP handling. I have =
written in
the different thread.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=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:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Partha<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

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0in 4.0pt'>

<div>

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0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] <b>On Behalf Of =
</b>Christer
Holmberg<br>
<b>Sent:</b> Thursday, July 25, 2013 2:05 AM<br>
<b>To:</b> Cullen Jennings (fluffy); Vijaya Mandava (vimandav)<br>
<b>Cc:</b> Alan Johnston; MMUSIC IETF WG<br>
<b>Subject:</b> Re: [MMUSIC] Trickle ICE for SIP =
Questions<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Hi,<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Wouldn't the UAC need to know that the 180s come from the =
same
UAS?<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Regards,<o:p></o:p></span></p>

</div>

<div>

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style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Christer<o:p></o:p></span></p>

</div>

<div>

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color:black'>&nbsp;<o:p></o:p></span></p>

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<div>

<p class=3DMsoNormal><span =
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color:black'><br>
<br>
Sent from <b><i>Windows</i></b> using </span><b><span =
style=3D'font-size:11.0pt;
font-family:"Calibri","sans-serif";color:blue'>TouchDown</span></b><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:black'=
>
(www.nitrodesk.com)<o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'><br>
-----Original Message-----<br>
<b>From:</b> Cullen Jennings (fluffy) [fluffy@cisco.com]<br>
<b>To:</b> Vijaya Mandava (vimandav) [vimandav@cisco.com]<br>
<b>CC:</b> Alan Johnston [alan.b.johnston@gmail.com]; MMUSIC IETF WG
[mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] Trickle ICE for SIP =
Questions<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt'><br>
On Jul 24, 2013, at 12:22 PM, Vijaya Mandava (vimandav)
&lt;vimandav@cisco.com&gt; wrote:<br>
<br>
&gt; 180 with different to-tag would mean call is forked.<br>
&gt; If uac side do not support call forking, then we cannot use this =
180<br>
&gt; response to collect trickled candidates.<br>
<br>
If it is a UAC, it supports this so I don't see the problem.<br>
<br>
_______________________________________________<br>
mmusic mailing list<br>
mmusic@ietf.org<br>
<a =
href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.or=
g/mailman/listinfo/mmusic</a><o:p></o:p></span></p>

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From emil@sip-communicator.org  Thu Jul 25 09:38:28 2013
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Date: Thu, 25 Jul 2013 18:38:16 +0200
From: Emil Ivov <emcho@jitsi.org>
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hey Partha,

On 25.07.13, 17:48, Parthasarathi R wrote:
> Hi Emil,
>
> In the mentioned situation, different To-tag with same contact header for
> trickled SDP, So there will be no "N" locations.

Having the same contact header does not automatically suggest that this 
is somehow a fake fork (unless I am missing the text that says this in 
3261). You could potentially have this if the fork is done by a B2BUA or 
a PSTN gateway for example.

> My reading of the draft is that this draft is not required in case UPDATE
> (RFC 3311) method is supported. UPDATE with SDP will update trickle
> candidate within 18x.

Use of UPDATE has already been discussed and we pretty much agreed not 
to go that way.

A mail from Alan and the preceding discussion:
http://www.ietf.org/mail-archive/web/mmusic/current/msg10660.html

A follow up with a nice summary from Dale:
http://www.ietf.org/mail-archive/web/mmusic/current/msg10678.html

> The aim of the draft is to support Trickle ICE for SIP user agent which does
> not support RFC 3311 for last 11 years. AFAIK, different "to tag" is the
> simpler way to solve the problem in UAS side. I have concern w.r.t INFO
> mechanism as it adds one method for handling SDP and complicates the
> existing SDP O/A handling (RFC 6337).

That's the thing, trickle ICE is not part of Offer/Answer. It's a 
separate channel for agent-to-agent communication. Use of SDP is just a 
convenient coincidence.

> Including Christer in this mail thread as he replied in the parallel mail
> thread.

I believe he is on the list I don't think that's an issue :)

Cheers,
Emil
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: Emil Ivov [mailto:emcho@jitsi.org]
>> Sent: Thursday, July 25, 2013 2:19 AM
>> To: Cullen Jennings (fluffy)
>> Cc: Vijaya Mandava (vimandav); Parthasarathi R; Alan Johnston; MMUSIC
>> IETF WG
>> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
>>
>>
>>
>> On 24.07.13, 21:56, Cullen Jennings (fluffy) wrote:
>>>
>>> On Jul 24, 2013, at 12:22 PM, Vijaya Mandava (vimandav)
>> <vimandav@cisco.com> wrote:
>>>
>>>> 180 with different to-tag would mean call is forked.
>>>> If uac side do not support call forking, then we cannot use this 180
>>>> response to collect trickled candidates.
>>>
>>> If it is a UAC, it supports this so I don't see the problem.
>>
>> One problem is that the UAC could support it to the extent of making it
>> visible to the user:
>>
>>     "Your contact is being alerted on these N locations"
>>
>> And of course there's also the fact that we would be creating several
>> dialogs per call, that we'd then need to clean and garbage collect.
>>
>> Why would we want to go there?
>>
>> Emil
>>
>> --
>> https://jitsi.org
>
>

-- 
https://jitsi.org

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Subject: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
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(I was rereviewing this in prep for the meeting)

Section 4 says that trickled candidates are transported as INFO payloads 
containing SDP formatted according to ivov-mmusic-trickle-ice. But 
AFAICT that draft doesn't specify this. Also the example in this section 
shows an info payload that contains only a subset of SDP, and doesn't 
conform to mime type application/sdp.

Section 5.6 again references the trickle-ice draft (section 4) for 
definition of the body of the info message, but that section of that 
draft doesn't contain any such definition.

Aside from that, it occurs to me that trickle-ice, and for that matter 
ice in sip even without trickle, is a good candidate for preconditions. 
The use of preconditions could eliminate the need for half trickle. This 
would of course require a new precondition type, but that is doable.

By using preconditions, you can establish the dialog without alerting 
the callee. Then the alert can be deferred until at least one workable 
path has been detected.

(But I suppose the hate for preconditions is so great that this is not 
to be considered.)

	Thanks,
	Paul

From christer.holmberg@ericsson.com  Thu Jul 25 11:36:48 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, IETF MMUSIC WG <mmusic@ietf.org>, Emil Ivov <emcho@jitsi.org>
Thread-Topic: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
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Date: Thu, 25 Jul 2013 18:36:41 +0000
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Subject: Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
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Hi Paul,

Are you sure a new type is needed? The existing types are rather generic, a=
ren't they?

Also, if people don't like the precondition mechanism, another option is to=
 simply set the streams to inactive until enough candidates are collected, =
and specify that trickle ICE UASs don't alert the user until the direction =
attribute is changed to a non-inactive value. E.g. 3GPP has done that for m=
any years.

Regards,

Christer


Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Paul Kyzivat [pkyzivat@alum.mit.edu]
To: IETF MMUSIC WG [mmusic@ietf.org]; Emil Ivov [emcho@jitsi.org]
Subject: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
(I was rereviewing this in prep for the meeting)

Section 4 says that trickled candidates are transported as INFO payloads
containing SDP formatted according to ivov-mmusic-trickle-ice. But
AFAICT that draft doesn't specify this. Also the example in this section
shows an info payload that contains only a subset of SDP, and doesn't
conform to mime type application/sdp.

Section 5.6 again references the trickle-ice draft (section 4) for
definition of the body of the info message, but that section of that
draft doesn't contain any such definition.

Aside from that, it occurs to me that trickle-ice, and for that matter
ice in sip even without trickle, is a good candidate for preconditions.
The use of preconditions could eliminate the need for half trickle. This
would of course require a new precondition type, but that is doable.

By using preconditions, you can establish the dialog without alerting
the callee. Then the alert can be deferred until at least one workable
path has been detected.

(But I suppose the hate for preconditions is so great that this is not
to be considered.)

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

--_000_7594FB04B1934943A5C02806D1A2204B1C408F15ESESSMB209erics_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
ding-left: 4pt; border-left: #800000 2px solid; } --></style>
</head>
<body>
<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi Paul,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">Are you sure a new type is needed? The existing=
 types are rather generic, aren't they?</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Also, if people don't like the precondition mec=
hanism, another option is to simply set the streams to inactive until enoug=
h candidates are collected, and specify that trickle ICE UASs don't alert t=
he user until the direction attribute
 is changed to a non-inactive value. E.g. 3GPP has done that for many years=
.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Paul Kyzivat [pkyzivat@alum.mit.edu]<br>
<b>To:</b> IETF MMUSIC WG [mmusic@ietf.org]; Emil Ivov [emcho@jitsi.org]<br=
>
<b>Subject:</b> [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">(I was rereviewing this in prep for the meeting)<b=
r>
<br>
Section 4 says that trickled candidates are transported as INFO payloads <b=
r>
containing SDP formatted according to ivov-mmusic-trickle-ice. But <br>
AFAICT that draft doesn't specify this. Also the example in this section <b=
r>
shows an info payload that contains only a subset of SDP, and doesn't <br>
conform to mime type application/sdp.<br>
<br>
Section 5.6 again references the trickle-ice draft (section 4) for <br>
definition of the body of the info message, but that section of that <br>
draft doesn't contain any such definition.<br>
<br>
Aside from that, it occurs to me that trickle-ice, and for that matter <br>
ice in sip even without trickle, is a good candidate for preconditions. <br=
>
The use of preconditions could eliminate the need for half trickle. This <b=
r>
would of course require a new precondition type, but that is doable.<br>
<br>
By using preconditions, you can establish the dialog without alerting <br>
the callee. Then the alert can be deferred until at least one workable <br>
path has been detected.<br>
<br>
(But I suppose the hate for preconditions is so great that this is not <br>
to be considered.)<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
_______________________________________________<br>
mmusic mailing list<br>
mmusic@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.o=
rg/mailman/listinfo/mmusic</a><br>
</div>
</span></font>
</body>
</html>

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From pkyzivat@alum.mit.edu  Thu Jul 25 11:55:37 2013
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Subject: Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
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On 7/25/13 2:36 PM, Christer Holmberg wrote:
> Hi Paul,
> Are you sure a new type is needed? The existing types are rather
> generic, aren't they?

I'm not sure, but I thought they were more specific than that.
Maybe there is some type that is sufficiently general.

> Also, if people don't like the precondition mechanism, another option is
> to simply set the streams to inactive until enough candidates are
> collected, and specify that trickle ICE UASs don't alert the user until
> the direction attribute is changed to a non-inactive value. E.g. 3GPP
> has done that for many years.

I hadn't heard of specifying a change in alerting behavior based on 
inactive. That seems weird and improper to me.

	Thanks,
	Paul

> Regards,
> Christer
>
> Sent from */Windows/* using *TouchDown* (www.nitrodesk.com)
>
> -----Original Message-----
> *From:* Paul Kyzivat [pkyzivat@alum.mit.edu]
> *To:* IETF MMUSIC WG [mmusic@ietf.org]; Emil Ivov [emcho@jitsi.org]
> *Subject:* [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
> (I was rereviewing this in prep for the meeting)
>
> Section 4 says that trickled candidates are transported as INFO payloads
> containing SDP formatted according to ivov-mmusic-trickle-ice. But
> AFAICT that draft doesn't specify this. Also the example in this section
> shows an info payload that contains only a subset of SDP, and doesn't
> conform to mime type application/sdp.
>
> Section 5.6 again references the trickle-ice draft (section 4) for
> definition of the body of the info message, but that section of that
> draft doesn't contain any such definition.
>
> Aside from that, it occurs to me that trickle-ice, and for that matter
> ice in sip even without trickle, is a good candidate for preconditions.
> The use of preconditions could eliminate the need for half trickle. This
> would of course require a new precondition type, but that is doable.
>
> By using preconditions, you can establish the dialog without alerting
> the callee. Then the alert can be deferred until at least one workable
> path has been detected.
>
> (But I suppose the hate for preconditions is so great that this is not
> to be considered.)
>
>          Thanks,
>          Paul
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From emil@sip-communicator.org  Thu Jul 25 11:57:32 2013
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Subject: Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
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Hey folks,

On 25.07.13, 20:36, Christer Holmberg wrote:
> Hi Paul,
> Are you sure a new type is needed? The existing types are rather
> generic, aren't they?

Back in Boston we decided to do something like this:

http://tools.ietf.org/html/draft-ivov-dispatch-sdpfrag-00

More below:

> On 25.07.13, 19:09, Paul Kyzivat wrote:
>
> Section 4 says that trickled candidates are transported as INFO payloads
> containing SDP formatted according to ivov-mmusic-trickle-ice. But
> AFAICT that draft doesn't specify this.

Well, it just borrows the format from 5245 and says that this is how you 
represent trickled candidates.

> Also the example in this section
> shows an info payload that contains only a subset of SDP, and doesn't
> conform to mime type application/sdp.

Yup, should be application/sdpfrag.

> Section 5.6 again references the trickle-ice draft (section 4) for
> definition of the body of the info message, but that section of that
> draft doesn't contain any such definition.

Will need to fix.

> Aside from that, it occurs to me that trickle-ice, and for that matter
> ice in sip even without trickle, is a good candidate for preconditions.
> The use of preconditions could eliminate the need for half trickle. This
> would of course require a new precondition type, but that is doable.

Hmmm ... that's a good point!

> By using preconditions, you can establish the dialog without alerting
> the callee. Then the alert can be deferred until at least one workable
> path has been detected.
>
> (But I suppose the hate for preconditions is so great that this is not
> to be considered.)

Well, it doesn't have to be mandatory ...

Emil

>
>          Thanks,
>          Paul
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

-- 
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
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Hi Paul,

I don't thin 3261 specifies when a user must be alerted, does it? ;)

Anyway, it was just one example of how people have achieved "delayed user a=
lerting" without using the precondition mechanism.

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Paul Kyzivat [pkyzivat@alum.mit.edu]
To: Christer Holmberg [christer.holmberg@ericsson.com]
CC: IETF MMUSIC WG [mmusic@ietf.org]; Emil Ivov [emcho@jitsi.org]
Subject: Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
On 7/25/13 2:36 PM, Christer Holmberg wrote:
> Hi Paul,
> Are you sure a new type is needed? The existing types are rather
> generic, aren't they?

I'm not sure, but I thought they were more specific than that.
Maybe there is some type that is sufficiently general.

> Also, if people don't like the precondition mechanism, another option is
> to simply set the streams to inactive until enough candidates are
> collected, and specify that trickle ICE UASs don't alert the user until
> the direction attribute is changed to a non-inactive value. E.g. 3GPP
> has done that for many years.

I hadn't heard of specifying a change in alerting behavior based on
inactive. That seems weird and improper to me.

        Thanks,
        Paul

> Regards,
> Christer
>
> Sent from */Windows/* using *TouchDown* (www.nitrodesk.com<http://www.nit=
rodesk.com>)
>
> -----Original Message-----
> *From:* Paul Kyzivat [pkyzivat@alum.mit.edu]
> *To:* IETF MMUSIC WG [mmusic@ietf.org]; Emil Ivov [emcho@jitsi.org]
> *Subject:* [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
> (I was rereviewing this in prep for the meeting)
>
> Section 4 says that trickled candidates are transported as INFO payloads
> containing SDP formatted according to ivov-mmusic-trickle-ice. But
> AFAICT that draft doesn't specify this. Also the example in this section
> shows an info payload that contains only a subset of SDP, and doesn't
> conform to mime type application/sdp.
>
> Section 5.6 again references the trickle-ice draft (section 4) for
> definition of the body of the info message, but that section of that
> draft doesn't contain any such definition.
>
> Aside from that, it occurs to me that trickle-ice, and for that matter
> ice in sip even without trickle, is a good candidate for preconditions.
> The use of preconditions could eliminate the need for half trickle. This
> would of course require a new precondition type, but that is doable.
>
> By using preconditions, you can establish the dialog without alerting
> the callee. Then the alert can be deferred until at least one workable
> path has been detected.
>
> (But I suppose the hate for preconditions is so great that this is not
> to be considered.)
>
>          Thanks,
>          Paul
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


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<div><br>
Hi Paul,</div>
<div>&nbsp;</div>
<div>I don't thin 3261 specifies when a user must be alerted, does it? ;)</=
div>
<div>&nbsp;</div>
<div>Anyway, it was just&nbsp;one example of how people have achieved &quot=
;delayed user alerting&quot; without using the precondition mechanism.</div=
>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div><br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Paul Kyzivat [pkyzivat@alum.mit.edu]<br>
<b>To:</b> Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>CC:</b> IETF MMUSIC WG [mmusic@ietf.org]; Emil Ivov [emcho@jitsi.org]<br=
>
<b>Subject:</b> Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00</span></d=
iv>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">On 7/25/13 2:36 PM, Christer Holmberg wrote:<br>
&gt; Hi Paul,<br>
&gt; Are you sure a new type is needed? The existing types are rather<br>
&gt; generic, aren't they?<br>
<br>
I'm not sure, but I thought they were more specific than that.<br>
Maybe there is some type that is sufficiently general.<br>
<br>
&gt; Also, if people don't like the precondition mechanism, another option =
is<br>
&gt; to simply set the streams to inactive until enough candidates are<br>
&gt; collected, and specify that trickle ICE UASs don't alert the user unti=
l<br>
&gt; the direction attribute is changed to a non-inactive value. E.g. 3GPP<=
br>
&gt; has done that for many years.<br>
<br>
I hadn't heard of specifying a change in alerting behavior based on <br>
inactive. That seems weird and improper to me.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
<br>
&gt; Regards,<br>
&gt; Christer<br>
&gt;<br>
&gt; Sent from */Windows/* using *TouchDown* (<a href=3D"http://www.nitrode=
sk.com">www.nitrodesk.com</a>)<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; *From:* Paul Kyzivat [pkyzivat@alum.mit.edu]<br>
&gt; *To:* IETF MMUSIC WG [mmusic@ietf.org]; Emil Ivov [emcho@jitsi.org]<br=
>
&gt; *Subject:* [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00<br>
&gt; (I was rereviewing this in prep for the meeting)<br>
&gt;<br>
&gt; Section 4 says that trickled candidates are transported as INFO payloa=
ds<br>
&gt; containing SDP formatted according to ivov-mmusic-trickle-ice. But<br>
&gt; AFAICT that draft doesn't specify this. Also the example in this secti=
on<br>
&gt; shows an info payload that contains only a subset of SDP, and doesn't<=
br>
&gt; conform to mime type application/sdp.<br>
&gt;<br>
&gt; Section 5.6 again references the trickle-ice draft (section 4) for<br>
&gt; definition of the body of the info message, but that section of that<b=
r>
&gt; draft doesn't contain any such definition.<br>
&gt;<br>
&gt; Aside from that, it occurs to me that trickle-ice, and for that matter=
<br>
&gt; ice in sip even without trickle, is a good candidate for preconditions=
.<br>
&gt; The use of preconditions could eliminate the need for half trickle. Th=
is<br>
&gt; would of course require a new precondition type, but that is doable.<b=
r>
&gt;<br>
&gt; By using preconditions, you can establish the dialog without alerting<=
br>
&gt; the callee. Then the alert can be deferred until at least one workable=
<br>
&gt; path has been detected.<br>
&gt;<br>
&gt; (But I suppose the hate for preconditions is so great that this is not=
<br>
&gt; to be considered.)<br>
&gt;<br>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; mmusic@ietf.org<br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.i=
etf.org/mailman/listinfo/mmusic</a><br>
<br>
</div>
</span></font>
</body>
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
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Hi Emil,

I don't think Paul was talking about a new MIME types, but the type values =
used for the precondition mechanism. Paul?

Regards,

Christer

Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Emil Ivov [emcho@jitsi.org]
To: Christer Holmberg [christer.holmberg@ericsson.com]
CC: Paul Kyzivat [pkyzivat@alum.mit.edu]; IETF MMUSIC WG [mmusic@ietf.org]
Subject: Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
Hey folks,

On 25.07.13, 20:36, Christer Holmberg wrote:
> Hi Paul,
> Are you sure a new type is needed? The existing types are rather
> generic, aren't they?

Back in Boston we decided to do something like this:

http://tools.ietf.org/html/draft-ivov-dispatch-sdpfrag-00

More below:

> On 25.07.13, 19:09, Paul Kyzivat wrote:
>
> Section 4 says that trickled candidates are transported as INFO payloads
> containing SDP formatted according to ivov-mmusic-trickle-ice. But
> AFAICT that draft doesn't specify this.

Well, it just borrows the format from 5245 and says that this is how you
represent trickled candidates.

> Also the example in this section
> shows an info payload that contains only a subset of SDP, and doesn't
> conform to mime type application/sdp.

Yup, should be application/sdpfrag.

> Section 5.6 again references the trickle-ice draft (section 4) for
> definition of the body of the info message, but that section of that
> draft doesn't contain any such definition.

Will need to fix.

> Aside from that, it occurs to me that trickle-ice, and for that matter
> ice in sip even without trickle, is a good candidate for preconditions.
> The use of preconditions could eliminate the need for half trickle. This
> would of course require a new precondition type, but that is doable.

Hmmm ... that's a good point!

> By using preconditions, you can establish the dialog without alerting
> the callee. Then the alert can be deferred until at least one workable
> path has been detected.
>
> (But I suppose the hate for preconditions is so great that this is not
> to be considered.)

Well, it doesn't have to be mandatory ...

Emil

>
>          Thanks,
>          Paul
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div><br>
Hi Emil,</div>
<div>&nbsp;</div>
<div>I don't think Paul was talking about a new MIME types, but the type va=
lues used for the precondition mechanism. Paul?</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<div><br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Emil Ivov [emcho@jitsi.org]<br>
<b>To:</b> Christer Holmberg [christer.holmberg@ericsson.com]<br>
<b>CC:</b> Paul Kyzivat [pkyzivat@alum.mit.edu]; IETF MMUSIC WG [mmusic@iet=
f.org]<br>
<b>Subject:</b> Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00</span></d=
iv>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">Hey folks,<br>
<br>
On 25.07.13, 20:36, Christer Holmberg wrote:<br>
&gt; Hi Paul,<br>
&gt; Are you sure a new type is needed? The existing types are rather<br>
&gt; generic, aren't they?<br>
<br>
Back in Boston we decided to do something like this:<br>
<br>
<a href=3D"http://tools.ietf.org/html/draft-ivov-dispatch-sdpfrag-00">http:=
//tools.ietf.org/html/draft-ivov-dispatch-sdpfrag-00</a><br>
<br>
More below:<br>
<br>
&gt; On 25.07.13, 19:09, Paul Kyzivat wrote:<br>
&gt;<br>
&gt; Section 4 says that trickled candidates are transported as INFO payloa=
ds<br>
&gt; containing SDP formatted according to ivov-mmusic-trickle-ice. But<br>
&gt; AFAICT that draft doesn't specify this.<br>
<br>
Well, it just borrows the format from 5245 and says that this is how you <b=
r>
represent trickled candidates.<br>
<br>
&gt; Also the example in this section<br>
&gt; shows an info payload that contains only a subset of SDP, and doesn't<=
br>
&gt; conform to mime type application/sdp.<br>
<br>
Yup, should be application/sdpfrag.<br>
<br>
&gt; Section 5.6 again references the trickle-ice draft (section 4) for<br>
&gt; definition of the body of the info message, but that section of that<b=
r>
&gt; draft doesn't contain any such definition.<br>
<br>
Will need to fix.<br>
<br>
&gt; Aside from that, it occurs to me that trickle-ice, and for that matter=
<br>
&gt; ice in sip even without trickle, is a good candidate for preconditions=
.<br>
&gt; The use of preconditions could eliminate the need for half trickle. Th=
is<br>
&gt; would of course require a new precondition type, but that is doable.<b=
r>
<br>
Hmmm ... that's a good point!<br>
<br>
&gt; By using preconditions, you can establish the dialog without alerting<=
br>
&gt; the callee. Then the alert can be deferred until at least one workable=
<br>
&gt; path has been detected.<br>
&gt;<br>
&gt; (But I suppose the hate for preconditions is so great that this is not=
<br>
&gt; to be considered.)<br>
<br>
Well, it doesn't have to be mandatory ...<br>
<br>
Emil<br>
<br>
&gt;<br>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; mmusic@ietf.org<br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.i=
etf.org/mailman/listinfo/mmusic</a><br>
<br>
-- <br>
<a href=3D"https://jitsi.org">https://jitsi.org</a><br>
</div>
</span></font>
</body>
</html>

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From pkyzivat@alum.mit.edu  Thu Jul 25 13:05:07 2013
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Cc: IETF MMUSIC WG <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
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On 7/25/13 2:57 PM, Emil Ivov wrote:
> Hey folks,
>
> On 25.07.13, 20:36, Christer Holmberg wrote:
>> Hi Paul,
>> Are you sure a new type is needed? The existing types are rather
>> generic, aren't they?
>
> Back in Boston we decided to do something like this:
>
> http://tools.ietf.org/html/draft-ivov-dispatch-sdpfrag-00

I thought Christer was talking about my suggestion that preconditions 
would require a new precondition type. Nothing to do with the mime type.

> More below:
>
>> On 25.07.13, 19:09, Paul Kyzivat wrote:
>>
>> Section 4 says that trickled candidates are transported as INFO payloads
>> containing SDP formatted according to ivov-mmusic-trickle-ice. But
>> AFAICT that draft doesn't specify this.
>
> Well, it just borrows the format from 5245 and says that this is how you
> represent trickled candidates.

But all of that assumes a full SDP body. And it seems you don't.
Your draft somewhere has a mention that the trickled stuff must be keyed 
to a media section via mid, but nothing formally written about it.

>> Also the example in this section
>> shows an info payload that contains only a subset of SDP, and doesn't
>> conform to mime type application/sdp.
>
> Yup, should be application/sdpfrag.

Well, AFAIK there is currently no such mime type. So its necessary to 
define it.

And while you could define a generic one, you have some specific 
requirements about what this needs to contain, so you might want to 
define application/trickle-ice-sdp with all the requisite rules.

>> Section 5.6 again references the trickle-ice draft (section 4) for
>> definition of the body of the info message, but that section of that
>> draft doesn't contain any such definition.
>
> Will need to fix.
>
>> Aside from that, it occurs to me that trickle-ice, and for that matter
>> ice in sip even without trickle, is a good candidate for preconditions.
>> The use of preconditions could eliminate the need for half trickle. This
>> would of course require a new precondition type, but that is doable.
>
> Hmmm ... that's a good point!
>
>> By using preconditions, you can establish the dialog without alerting
>> the callee. Then the alert can be deferred until at least one workable
>> path has been detected.
>>
>> (But I suppose the hate for preconditions is so great that this is not
>> to be considered.)
>
> Well, it doesn't have to be mandatory ...
>
> Emil
>
>>
>>          Thanks,
>>          Paul
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www.ietf.org/mailman/listinfo/mmusic
>


From pkyzivat@alum.mit.edu  Thu Jul 25 13:07:47 2013
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Subject: Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
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On 7/25/13 3:08 PM, Christer Holmberg wrote:
>
> Hi Paul,
> I don't thin 3261 specifies when a user must be alerted, does it? ;)

No, I don't think so.

I didn't say it was an invalid thing to do, just improper and weird. :-)

IOW, it doesn't smell right to me.

	Thanks,
	Paul

> Anyway, it was just one example of how people have achieved "delayed
> user alerting" without using the precondition mechanism.
> Regards,
> Christer
>
> Sent from */Windows/* using *TouchDown* (www.nitrodesk.com)
>
> -----Original Message-----
> *From:* Paul Kyzivat [pkyzivat@alum.mit.edu]
> *To:* Christer Holmberg [christer.holmberg@ericsson.com]
> *CC:* IETF MMUSIC WG [mmusic@ietf.org]; Emil Ivov [emcho@jitsi.org]
> *Subject:* Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
> On 7/25/13 2:36 PM, Christer Holmberg wrote:
>> Hi Paul,
>> Are you sure a new type is needed? The existing types are rather
>> generic, aren't they?
>
> I'm not sure, but I thought they were more specific than that.
> Maybe there is some type that is sufficiently general.
>
>> Also, if people don't like the precondition mechanism, another option is
>> to simply set the streams to inactive until enough candidates are
>> collected, and specify that trickle ICE UASs don't alert the user until
>> the direction attribute is changed to a non-inactive value. E.g. 3GPP
>> has done that for many years.
>
> I hadn't heard of specifying a change in alerting behavior based on
> inactive. That seems weird and improper to me.
>
>          Thanks,
>          Paul
>
>> Regards,
>> Christer
>>
>> Sent from */Windows/* using *TouchDown* (www.nitrodesk.com <http://www.nitrodesk.com>)
>>
>> -----Original Message-----
>> *From:* Paul Kyzivat [pkyzivat@alum.mit.edu]
>> *To:* IETF MMUSIC WG [mmusic@ietf.org]; Emil Ivov [emcho@jitsi.org]
>> *Subject:* [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
>> (I was rereviewing this in prep for the meeting)
>>
>> Section 4 says that trickled candidates are transported as INFO payloads
>> containing SDP formatted according to ivov-mmusic-trickle-ice. But
>> AFAICT that draft doesn't specify this. Also the example in this section
>> shows an info payload that contains only a subset of SDP, and doesn't
>> conform to mime type application/sdp.
>>
>> Section 5.6 again references the trickle-ice draft (section 4) for
>> definition of the body of the info message, but that section of that
>> draft doesn't contain any such definition.
>>
>> Aside from that, it occurs to me that trickle-ice, and for that matter
>> ice in sip even without trickle, is a good candidate for preconditions.
>> The use of preconditions could eliminate the need for half trickle. This
>> would of course require a new precondition type, but that is doable.
>>
>> By using preconditions, you can establish the dialog without alerting
>> the callee. Then the alert can be deferred until at least one workable
>> path has been detected.
>>
>> (But I suppose the hate for preconditions is so great that this is not
>> to be considered.)
>>
>>          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 emil@sip-communicator.org  Thu Jul 25 13:42:36 2013
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Subject: Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
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Hey Paul,

On 25.07.13, 22:04, Paul Kyzivat wrote:
> On 7/25/13 2:57 PM, Emil Ivov wrote:
>> Hey folks,
>>
>> On 25.07.13, 20:36, Christer Holmberg wrote:
>>> Hi Paul,
>>> Are you sure a new type is needed? The existing types are rather
>>> generic, aren't they?
>>
>> Back in Boston we decided to do something like this:
>>
>> http://tools.ietf.org/html/draft-ivov-dispatch-sdpfrag-00
>
> I thought Christer was talking about my suggestion that preconditions
> would require a new precondition type. Nothing to do with the mime type.

Sorry, the above was meant to go underneath your comment about how this: 
"doesn't conform to mime type application/sdp."

But I messed up the quoting.

>> More below:
>>
>>> On 25.07.13, 19:09, Paul Kyzivat wrote:
>>>
>>> Section 4 says that trickled candidates are transported as INFO payloads
>>> containing SDP formatted according to ivov-mmusic-trickle-ice. But
>>> AFAICT that draft doesn't specify this.
>>
>> Well, it just borrows the format from 5245 and says that this is how you
>> represent trickled candidates.
>
> But all of that assumes a full SDP body.

I don't see how the syntax of an "a=candidate" line depends on the full 
SDP body, so I don't see the problem with just pointing there. Again, I 
mean this only about syntax.

> And it seems you don't.
> Your draft somewhere has a mention that the trickled stuff must be keyed
> to a media section via mid, but nothing formally written about it.

Yes agreed. This will have to be expanded in trickle-ice-sip.
>
>>> Also the example in this section
>>> shows an info payload that contains only a subset of SDP, and doesn't
>>> conform to mime type application/sdp.
>>
>> Yup, should be application/sdpfrag.
>
> Well, AFAIK there is currently no such mime type. So its necessary to
> define it.

This is the point of the draft I pasted above:
http://tools.ietf.org/html/draft-ivov-dispatch-sdpfrag-00

Of course, it does need more work (by the way, help there is more than 
welcome so if you want to join just let me know).

> And while you could define a generic one, you have some specific
> requirements about what this needs to contain, so you might want to
> define application/trickle-ice-sdp with all the requisite rules.

The original idea (which I believe came from Adam) was that this could 
be avoided in a way similar to how application/sipfrag does it. I 
basically specifies syntax (as a SIP subset) and leaves semantics to 
specs that are using it, like for example 3515

Do you see any reason why this couldn't work here?

Emil

>>> Section 5.6 again references the trickle-ice draft (section 4) for
>>> definition of the body of the info message, but that section of that
>>> draft doesn't contain any such definition.
>>
>> Will need to fix.
>>
>>> Aside from that, it occurs to me that trickle-ice, and for that matter
>>> ice in sip even without trickle, is a good candidate for preconditions.
>>> The use of preconditions could eliminate the need for half trickle. This
>>> would of course require a new precondition type, but that is doable.
>>
>> Hmmm ... that's a good point!
>>
>>> By using preconditions, you can establish the dialog without alerting
>>> the callee. Then the alert can be deferred until at least one workable
>>> path has been detected.
>>>
>>> (But I suppose the hate for preconditions is so great that this is not
>>> to be considered.)
>>
>> Well, it doesn't have to be mandatory ...
>>
>> Emil
>>
>>>
>>>           Thanks,
>>>           Paul
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>
> .
>

-- 
https://jitsi.org

From pkyzivat@alum.mit.edu  Thu Jul 25 14:46:16 2013
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Date: Thu, 25 Jul 2013 17:46:02 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: IETF MMUSIC WG <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
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On 7/25/13 4:42 PM, Emil Ivov wrote:

>>>> On 25.07.13, 19:09, Paul Kyzivat wrote:
>>>>
>>>> Section 4 says that trickled candidates are transported as INFO
>>>> payloads
>>>> containing SDP formatted according to ivov-mmusic-trickle-ice. But
>>>> AFAICT that draft doesn't specify this.
>>>
>>> Well, it just borrows the format from 5245 and says that this is how you
>>> represent trickled candidates.
>>
>> But all of that assumes a full SDP body.
>
> I don't see how the syntax of an "a=candidate" line depends on the full
> SDP body, so I don't see the problem with just pointing there. Again, I
> mean this only about syntax.

My point is that 5245 never suggests that ICE will be represented in SDP 
in anything but a full SDP body.

If you said that you were proposing to send subsets of SDP as described 
in 5245, that would make some sense. (As long as you specify how to 
select the subset.)

>>>> Also the example in this section
>>>> shows an info payload that contains only a subset of SDP, and doesn't
>>>> conform to mime type application/sdp.
>>>
>>> Yup, should be application/sdpfrag.
>>
>> Well, AFAIK there is currently no such mime type. So its necessary to
>> define it.
>
> This is the point of the draft I pasted above:
> http://tools.ietf.org/html/draft-ivov-dispatch-sdpfrag-00
>
> Of course, it does need more work (by the way, help there is more than
> welcome so if you want to join just let me know).

Oh, duh. That slipped right by me. But of course it is incomplete.

I could help, once I understand what it needs to say. The first decision is:

>> And while you could define a generic one, you have some specific
>> requirements about what this needs to contain, so you might want to
>> define application/trickle-ice-sdp with all the requisite rules.
>
> The original idea (which I believe came from Adam) was that this could
> be avoided in a way similar to how application/sipfrag does it. I
> basically specifies syntax (as a SIP subset) and leaves semantics to
> specs that are using it, like for example 3515
>
> Do you see any reason why this couldn't work here?

Its a choice. In the case of sipfrag, IIRC it was defined for use with 
3515. In that it isn't crucial what is in it. So it works there.

In your case what goes in it is more critical. It can still be defined 
this way. But it will take some care to get it right.

The main thing is that it be done somewhere!

Can an INFO trickle ICE candidates for more than one m-line?
If so, it gets messy. Consider a full SDP:

...
m=...
a=mid:1
a=candidate:...
...
m=...
a=candidate:...
a=mid:2
a=candidate:...

(AFAIK the above is entirely legal.)

Now consider a sipfrag for trickle from the above:

a=mid:1
a=candidate:...
a=candidate:...
a=mid:2
a=candidate:...

Now, which candidates go with which m-line? So, I think your sdpfrag 
needs special constraints that don't apply to ICE in full SDP.


	Thanks,
	Paul

> Emil
>
>>>> Section 5.6 again references the trickle-ice draft (section 4) for
>>>> definition of the body of the info message, but that section of that
>>>> draft doesn't contain any such definition.
>>>
>>> Will need to fix.
>>>
>>>> Aside from that, it occurs to me that trickle-ice, and for that matter
>>>> ice in sip even without trickle, is a good candidate for preconditions.
>>>> The use of preconditions could eliminate the need for half trickle.
>>>> This
>>>> would of course require a new precondition type, but that is doable.
>>>
>>> Hmmm ... that's a good point!
>>>
>>>> By using preconditions, you can establish the dialog without alerting
>>>> the callee. Then the alert can be deferred until at least one workable
>>>> path has been detected.
>>>>
>>>> (But I suppose the hate for preconditions is so great that this is not
>>>> to be considered.)
>>>
>>> Well, it doesn't have to be mandatory ...
>>>
>>> Emil
>>>
>>>>
>>>>           Thanks,
>>>>           Paul
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> .
>>
>


From emil@sip-communicator.org  Thu Jul 25 15:32:40 2013
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Date: Fri, 26 Jul 2013 00:32:31 +0200
From: Emil Ivov <emcho@jitsi.org>
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Cc: IETF MMUSIC WG <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
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Hey Paul,

On 25.07.13, 23:46, Paul Kyzivat wrote:
> On 7/25/13 4:42 PM, Emil Ivov wrote:
>
>>>>> On 25.07.13, 19:09, Paul Kyzivat wrote:
>>>>>
>>>>> Section 4 says that trickled candidates are transported as INFO
>>>>> payloads
>>>>> containing SDP formatted according to ivov-mmusic-trickle-ice. But
>>>>> AFAICT that draft doesn't specify this.
>>>>
>>>> Well, it just borrows the format from 5245 and says that this is how you
>>>> represent trickled candidates.
>>>
>>> But all of that assumes a full SDP body.
>>
>> I don't see how the syntax of an "a=candidate" line depends on the full
>> SDP body, so I don't see the problem with just pointing there. Again, I
>> mean this only about syntax.
>
> My point is that 5245 never suggests that ICE will be represented in SDP
> in anything but a full SDP body.
>
> If you said that you were proposing to send subsets of SDP as described
> in 5245, that would make some sense.

The section that deals with this is 9.1

http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00#section-9.1

The intention is exactly what you describe above.

> (As long as you specify how to
> select the subset.)

The subset is an "a=candidate" line. I don't think select is the right 
word here though. You just encode an ICE candidate, as you would encode 
it for an SDP offer or answer.

>>>>> Also the example in this section
>>>>> shows an info payload that contains only a subset of SDP, and doesn't
>>>>> conform to mime type application/sdp.
>>>>
>>>> Yup, should be application/sdpfrag.
>>>
>>> Well, AFAIK there is currently no such mime type. So its necessary to
>>> define it.
>>
>> This is the point of the draft I pasted above:
>> http://tools.ietf.org/html/draft-ivov-dispatch-sdpfrag-00
>>
>> Of course, it does need more work (by the way, help there is more than
>> welcome so if you want to join just let me know).
>
> Oh, duh. That slipped right by me. But of course it is incomplete.
>
> I could help, once I understand what it needs to say.

Great!

> The first decision is:

Did you miss something here?

>>> And while you could define a generic one, you have some specific
>>> requirements about what this needs to contain, so you might want to
>>> define application/trickle-ice-sdp with all the requisite rules.
>>
>> The original idea (which I believe came from Adam) was that this could
>> be avoided in a way similar to how application/sipfrag does it. I
>> basically specifies syntax (as a SIP subset) and leaves semantics to
>> specs that are using it, like for example 3515
>>
>> Do you see any reason why this couldn't work here?
>
> Its a choice. In the case of sipfrag, IIRC it was defined for use with
> 3515. In that it isn't crucial what is in it. So it works there.
>
> In your case what goes in it is more critical. It can still be defined
> this way. But it will take some care to get it right.
>
> The main thing is that it be done somewhere!

Agreed!

> Can an INFO trickle ICE candidates for more than one m-line?

Yes.

> If so, it gets messy. Consider a full SDP:
>
> ...
> m=...
> a=mid:1
> a=candidate:...
> ...
> m=...
> a=candidate:...
> a=mid:2
> a=candidate:...
>
> (AFAIK the above is entirely legal.)
>
> Now consider a sipfrag for trickle from the above:
>
> a=mid:1
> a=candidate:...
> a=candidate:...
> a=mid:2
> a=candidate:...
>
> Now, which candidates go with which m-line? So, I think your sdpfrag
> needs special constraints that don't apply to ICE in full SDP.

I think the problem with the above is that you are trying to present it 
as an extract from some pre-existing offer or answer, while it isn't.

When you are trickling you find yourself in a situation where you have 
one or more newly learned candidates. You want to send those to the 
remote agent.

To do this you group them by stream (as in "m=" line). For each group 
you start by creating an "a=mid:x" attribute that matches the one from 
the "m=" line that created the stream. You then create and append 
"a=candidate" lines for every candidate in that group.

That's it.

The receiving peer does the opposite: it parses the "a=mid:x" line, it 
finds the stream object that it refers to, then parses all the 
candidates and adds them there.

Does this make sense?

Emil


>
> 	Thanks,
> 	Paul
>
>> Emil
>>
>>>>> Section 5.6 again references the trickle-ice draft (section 4) for
>>>>> definition of the body of the info message, but that section of that
>>>>> draft doesn't contain any such definition.
>>>>
>>>> Will need to fix.
>>>>
>>>>> Aside from that, it occurs to me that trickle-ice, and for that matter
>>>>> ice in sip even without trickle, is a good candidate for preconditions.
>>>>> The use of preconditions could eliminate the need for half trickle.
>>>>> This
>>>>> would of course require a new precondition type, but that is doable.
>>>>
>>>> Hmmm ... that's a good point!
>>>>
>>>>> By using preconditions, you can establish the dialog without alerting
>>>>> the callee. Then the alert can be deferred until at least one workable
>>>>> path has been detected.
>>>>>
>>>>> (But I suppose the hate for preconditions is so great that this is not
>>>>> to be considered.)
>>>>
>>>> Well, it doesn't have to be mandatory ...
>>>>
>>>> Emil
>>>>
>>>>>
>>>>>            Thanks,
>>>>>            Paul
>>>>> _______________________________________________
>>>>> mmusic mailing list
>>>>> mmusic@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>
>>>
>>> .
>>>
>>
>
>

-- 
https://jitsi.org

From pkyzivat@alum.mit.edu  Thu Jul 25 16:26:35 2013
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: IETF MMUSIC WG <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
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Emil,

On 7/25/13 6:32 PM, Emil Ivov wrote:
> Hey Paul,
>
> On 25.07.13, 23:46, Paul Kyzivat wrote:
>> On 7/25/13 4:42 PM, Emil Ivov wrote:
>>
>>>>>> On 25.07.13, 19:09, Paul Kyzivat wrote:
>>>>>>
>>>>>> Section 4 says that trickled candidates are transported as INFO
>>>>>> payloads
>>>>>> containing SDP formatted according to ivov-mmusic-trickle-ice. But
>>>>>> AFAICT that draft doesn't specify this.
>>>>>
>>>>> Well, it just borrows the format from 5245 and says that this is
>>>>> how you
>>>>> represent trickled candidates.
>>>>
>>>> But all of that assumes a full SDP body.
>>>
>>> I don't see how the syntax of an "a=candidate" line depends on the full
>>> SDP body, so I don't see the problem with just pointing there. Again, I
>>> mean this only about syntax.
>>
>> My point is that 5245 never suggests that ICE will be represented in SDP
>> in anything but a full SDP body.
>>
>> If you said that you were proposing to send subsets of SDP as described
>> in 5245, that would make some sense.
>
> The section that deals with this is 9.1
>
> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00#section-9.1
>
> The intention is exactly what you describe above.

Maybe you intended that. But I don't find sufficient words in the doc to 
determine what you mean. The words that seem to pertain to this are:

    Given that such lines do not provide a relationship between the
    candidate and the m line that it relates to, signalling protocols
    using trickle ICE MUST establish that relation themselves.  This can
    be done in one of two ways: using the m-line index, or an MID
    [RFC3388].  The m-line index is a zero-based index, referring to the
    Nth m-line in the SDP.  The MID uses "media stream identification",
    as defined in [RFC3388], to identify the m-line.

So I'm supposed to infer that you are proposing to use a subset of SDP 
from that???

Also, what does "signalling protocols using trickle ICE MUST establish 
that relation themselves" *mean*? Does it mean that you intend the INFO 
message containing the trickle ice info in its body to specify the 
m-line somewhere other than in the same body???

I would expect that the body you send would include something to do 
this. If you use mid, then the syntax to do so already exists, sort of. 
If you use m-line index, then it doesn't.

>> (As long as you specify how to
>> select the subset.)
>
> The subset is an "a=candidate" line. I don't think select is the right
> word here though. You just encode an ICE candidate, as you would encode
> it for an SDP offer or answer.
>
>>>>>> Also the example in this section
>>>>>> shows an info payload that contains only a subset of SDP, and doesn't
>>>>>> conform to mime type application/sdp.
>>>>>
>>>>> Yup, should be application/sdpfrag.
>>>>
>>>> Well, AFAIK there is currently no such mime type. So its necessary to
>>>> define it.
>>>
>>> This is the point of the draft I pasted above:
>>> http://tools.ietf.org/html/draft-ivov-dispatch-sdpfrag-00
>>>
>>> Of course, it does need more work (by the way, help there is more than
>>> welcome so if you want to join just let me know).
>>
>> Oh, duh. That slipped right by me. But of course it is incomplete.
>>
>> I could help, once I understand what it needs to say.
>
> Great!
>
>> The first decision is:
>
> Did you miss something here?

I forget. It doesn't matter.

>>>> And while you could define a generic one, you have some specific
>>>> requirements about what this needs to contain, so you might want to
>>>> define application/trickle-ice-sdp with all the requisite rules.
>>>
>>> The original idea (which I believe came from Adam) was that this could
>>> be avoided in a way similar to how application/sipfrag does it. I
>>> basically specifies syntax (as a SIP subset) and leaves semantics to
>>> specs that are using it, like for example 3515
>>>
>>> Do you see any reason why this couldn't work here?
>>
>> Its a choice. In the case of sipfrag, IIRC it was defined for use with
>> 3515. In that it isn't crucial what is in it. So it works there.
>>
>> In your case what goes in it is more critical. It can still be defined
>> this way. But it will take some care to get it right.
>>
>> The main thing is that it be done somewhere!
>
> Agreed!
>
>> Can an INFO trickle ICE candidates for more than one m-line?
>
> Yes.
>
>> If so, it gets messy. Consider a full SDP:
>>
>> ...
>> m=...
>> a=mid:1
>> a=candidate:...
>> ...
>> m=...
>> a=candidate:...
>> a=mid:2
>> a=candidate:...
>>
>> (AFAIK the above is entirely legal.)
>>
>> Now consider a sipfrag for trickle from the above:
>>
>> a=mid:1
>> a=candidate:...
>> a=candidate:...
>> a=mid:2
>> a=candidate:...
>>
>> Now, which candidates go with which m-line? So, I think your sdpfrag
>> needs special constraints that don't apply to ICE in full SDP.
>
> I think the problem with the above is that you are trying to present it
> as an extract from some pre-existing offer or answer, while it isn't.
>
> When you are trickling you find yourself in a situation where you have
> one or more newly learned candidates. You want to send those to the
> remote agent.
>
> To do this you group them by stream (as in "m=" line). For each group
> you start by creating an "a=mid:x" attribute that matches the one from
> the "m=" line that created the stream. You then create and append
> "a=candidate" lines for every candidate in that group.
>
> That's it.

I thought that is what you had in mind.

But you have said you want to use application/sdpfrag, which means it 
should be viewed as a subset of a full SDP. And that is especially so if 
you intend to reference m-lines in an SDP, but don't want to include 
them. And you haven't defined any new ordering rules for lines other 
than what exist for SDP. And SDP allows the ordering I show above, which 
doesn't work for you.

I'll also mention that I think you need to explicitly reference the SDP 
offer or answer that this trickle updates. Probably by replicating the 
o-line from that SDP.

So there are some significant syntax rules that need to be followed 
here. I would expect something like:

- must start with o-line of the SDP it is updating
- has one or more "media sections"
- each media section begins with an a:mid line, referencing
   a mid defined in the referenced SDP
- each a:mid line is followed by one or more a:candidate lines
...

If going that way, then I think it is better to define a mime type that 
consists of exactly this syntax, with some extensibility hooks.

	Thanks,
	Paul

> The receiving peer does the opposite: it parses the "a=mid:x" line, it
> finds the stream object that it refers to, then parses all the
> candidates and adds them there.
>
> Does this make sense?
>
> Emil
>
>
>>
>>     Thanks,
>>     Paul
>>
>>> Emil
>>>
>>>>>> Section 5.6 again references the trickle-ice draft (section 4) for
>>>>>> definition of the body of the info message, but that section of that
>>>>>> draft doesn't contain any such definition.
>>>>>
>>>>> Will need to fix.
>>>>>
>>>>>> Aside from that, it occurs to me that trickle-ice, and for that
>>>>>> matter
>>>>>> ice in sip even without trickle, is a good candidate for
>>>>>> preconditions.
>>>>>> The use of preconditions could eliminate the need for half trickle.
>>>>>> This
>>>>>> would of course require a new precondition type, but that is doable.
>>>>>
>>>>> Hmmm ... that's a good point!
>>>>>
>>>>>> By using preconditions, you can establish the dialog without alerting
>>>>>> the callee. Then the alert can be deferred until at least one
>>>>>> workable
>>>>>> path has been detected.
>>>>>>
>>>>>> (But I suppose the hate for preconditions is so great that this is
>>>>>> not
>>>>>> to be considered.)
>>>>>
>>>>> Well, it doesn't have to be mandatory ...
>>>>>
>>>>> Emil
>>>>>
>>>>>>
>>>>>>            Thanks,
>>>>>>            Paul
>>>>>> _______________________________________________
>>>>>> mmusic mailing list
>>>>>> mmusic@ietf.org
>>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>>
>>>>
>>>> .
>>>>
>>>
>>
>>
>


From pkyzivat@alum.mit.edu  Thu Jul 25 16:33:06 2013
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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On 7/23/13 1:06 PM, Emil Ivov wrote:

> I actually think we could pull this off with simple PRACKs. Only, rather
> than using the "100rel" tag we go for something like "100rel-lite" which
> could be defined to indicate that a UA supports PRACK but only for
> reliability (and PRACK bodies are not allowed).

This is crazy!

Every new option presents backward compatibility issues, implementation 
issues, ...

It doesn't make sense to define a new option that is a subset of a long 
time existing option. This is *not* rocket science. Just tell people 
they need to support 100rel as a precondition for trickle ICE. If they 
find trickle ice worthwhile they can implement it.

I'd be amazed if there are any SBCs that don't support PRACK. (But 
Hadriel will fill us in about that.)

	Thanks,
	Paul


From fluffy@cisco.com  Thu Jul 25 16:37:34 2013
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Emil Ivov <emcho@jitsi.org>
Thread-Topic: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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On Jul 24, 2013, at 4:58 PM, Emil Ivov <emcho@jitsi.org> wrote:

> If so, we are back to I=F1aki's original question: how does a single-sour=
ce callee know which if these "m=3D" lines to pick for its own stream?

If they don't have a=3Dcontent line (see http://tools.ietf.org/html/rfc4796=
) , you probably don't know which one to use and just use the first. If the=
y no have the a=3Dcontent slide, then you likely have the information you n=
eed to make the right decision by matching up to the one that has the value=
 for a=3Dcontent as the video you want to send. Note this would work better=
 if the registry also had front, left and right.=20





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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Adam Roach <adam@nostrum.com>
Thread-Topic: [MMUSIC] draft-roach-mmusic-unified-plan-00
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Cc: IETF MMUSIC WG <mmusic@ietf.org>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [MMUSIC] draft-roach-mmusic-unified-plan-00
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On Jul 24, 2013, at 10:03 AM, Adam Roach <adam@nostrum.com> wrote:

> On 7/24/13 10:55, Paul Kyzivat wrote:
>> This seems to be excluded, except by sending a new O/A every time the sp=
eaker switches, which isn't feasible. If this means that CLUE can't represe=
nt capture-encodings as m-lines, then there is no benefit to CLUE for the u=
nified plan over No Plan (Emil's multiple-sources draft).=20
>=20
> That's interesting. I wonder whether we should simply move to the RTP hea=
der extension described in section 3.2.1.1 as the primary means of correlat=
ion. It would seem to address the case you describe.
>=20
> /a

SSRC can collide too so they might be wrong but the  header extension does =
not have that problem. It seems to me the algorithm is roughly

if you have  header extension, use that, if not then if you know the SSRC u=
se that, if not then if you have unique PTs, use that. Perhaps there are ot=
her things to use too but roughly those get used in that order.=20




From pkyzivat@alum.mit.edu  Thu Jul 25 17:19:46 2013
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Subject: Re: [MMUSIC] draft-roach-mmusic-unified-plan-00
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On 7/25/13 7:41 PM, Cullen Jennings (fluffy) wrote:
>
>
> On Jul 24, 2013, at 10:03 AM, Adam Roach <adam@nostrum.com> wrote:
>
>> On 7/24/13 10:55, Paul Kyzivat wrote:
>>> This seems to be excluded, except by sending a new O/A every time the speaker switches, which isn't feasible. If this means that CLUE can't represent capture-encodings as m-lines, then there is no benefit to CLUE for the unified plan over No Plan (Emil's multiple-sources draft).
>>
>> That's interesting. I wonder whether we should simply move to the RTP header extension described in section 3.2.1.1 as the primary means of correlation. It would seem to address the case you describe.
>>
>> /a
>
> SSRC can collide too so they might be wrong but the  header extension does not have that problem. It seems to me the algorithm is roughly
>
> if you have  header extension, use that, if not then if you know the SSRC use that, if not then if you have unique PTs, use that. Perhaps there are other things to use too but roughly those get used in that order.

I commented on this before. You don't need to have an ordering.

If the combination of things you have is unique, use that. If not, then 
you have a problem.

	Thanks,
	Paul


From emil@sip-communicator.org  Thu Jul 25 17:24:48 2013
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Cc: IETF MMUSIC WG <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] draft-ivov-mmusic-trickle-ice-sip-00
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On 26.07.13, 01:26, Paul Kyzivat wrote:
> Emil,
>
> On 7/25/13 6:32 PM, Emil Ivov wrote:
>> Hey Paul,
>>
>> On 25.07.13, 23:46, Paul Kyzivat wrote:
>>> On 7/25/13 4:42 PM, Emil Ivov wrote:
>>>
>>>>>>> On 25.07.13, 19:09, Paul Kyzivat wrote:
>>>>>>>
>>>>>>> Section 4 says that trickled candidates are transported as INFO
>>>>>>> payloads
>>>>>>> containing SDP formatted according to ivov-mmusic-trickle-ice. But
>>>>>>> AFAICT that draft doesn't specify this.
>>>>>>
>>>>>> Well, it just borrows the format from 5245 and says that this is
>>>>>> how you
>>>>>> represent trickled candidates.
>>>>>
>>>>> But all of that assumes a full SDP body.
>>>>
>>>> I don't see how the syntax of an "a=candidate" line depends on the full
>>>> SDP body, so I don't see the problem with just pointing there. Again, I
>>>> mean this only about syntax.
>>>
>>> My point is that 5245 never suggests that ICE will be represented in SDP
>>> in anything but a full SDP body.
>>>
>>> If you said that you were proposing to send subsets of SDP as described
>>> in 5245, that would make some sense.
>>
>> The section that deals with this is 9.1
>>
>> http://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-00#section-9.1
>>
>> The intention is exactly what you describe above.
>
> Maybe you intended that. But I don't find sufficient words in the doc to
> determine what you mean. The words that seem to pertain to this are:
>
>      Given that such lines do not provide a relationship between the
>      candidate and the m line that it relates to, signalling protocols
>      using trickle ICE MUST establish that relation themselves.  This can
>      be done in one of two ways: using the m-line index, or an MID
>      [RFC3388].  The m-line index is a zero-based index, referring to the
>      Nth m-line in the SDP.  The MID uses "media stream identification",
>      as defined in [RFC3388], to identify the m-line.
>
> So I'm supposed to infer that you are proposing to use a subset of SDP
> from that???

No, you are not. This is the generic trickle ICE draft and it says: when 
using trickle with a specific protocol, make sure you match candidates 
to m= lines through use of MIDs.

It can't say anything more here because XMPP would encode the MID value 
  one way (i.e. using content names attributes), SIP another (using 
a=mid) and WebRTC apps in yet another (likely json).

> Also, what does "signalling protocols using trickle ICE MUST establish
> that relation themselves" *mean*?

I hope the above answers your question.

> Does it mean that you intend the INFO
> message containing the trickle ice info in its body to specify the
> m-line somewhere other than in the same body???

I don't understand this part. Didn't my previous mail clarify this?

Here's an example:

          INFO sip:alice@example.com SIP/2.0
          ...
          Info-Package: trickle-ice
          Content-type: application/sdp
          Content-Disposition: Info-Package
          Content-length: ...

          a=mid:1
          a=candidate:1 1 UDP 1658497328 192.168.100.33 5000 typ host
          a=candidate:2 1 UDP 1658497328 96.1.2.3 5000 typ srflx
          a=mid:2
          a=candidate:2 1 UDP 1658497328 96.1.2.3 5002 typ srflx
          a=end-of-candidates

That's the entire body and there are no m= lines in it.

> I would expect that the body you send would include something to do
> this. If you use mid, then the syntax to do so already exists, sort of.
> If you use m-line index, then it doesn't.

We kind of decided not to use "m=" line index in one of the previous 
meetings so we are only going for MID. The draft hasn't been updated 
since so it still refers to "m=" line index in a few locations but they 
will go.

>>> (As long as you specify how to
>>> select the subset.)
>>
>> The subset is an "a=candidate" line. I don't think select is the right
>> word here though. You just encode an ICE candidate, as you would encode
>> it for an SDP offer or answer.
>>
>>>>>>> Also the example in this section
>>>>>>> shows an info payload that contains only a subset of SDP, and doesn't
>>>>>>> conform to mime type application/sdp.
>>>>>>
>>>>>> Yup, should be application/sdpfrag.
>>>>>
>>>>> Well, AFAIK there is currently no such mime type. So its necessary to
>>>>> define it.
>>>>
>>>> This is the point of the draft I pasted above:
>>>> http://tools.ietf.org/html/draft-ivov-dispatch-sdpfrag-00
>>>>
>>>> Of course, it does need more work (by the way, help there is more than
>>>> welcome so if you want to join just let me know).
>>>
>>> Oh, duh. That slipped right by me. But of course it is incomplete.
>>>
>>> I could help, once I understand what it needs to say.
>>
>> Great!
>>
>>> The first decision is:
>>
>> Did you miss something here?
>
> I forget. It doesn't matter.
>
>>>>> And while you could define a generic one, you have some specific
>>>>> requirements about what this needs to contain, so you might want to
>>>>> define application/trickle-ice-sdp with all the requisite rules.
>>>>
>>>> The original idea (which I believe came from Adam) was that this could
>>>> be avoided in a way similar to how application/sipfrag does it. I
>>>> basically specifies syntax (as a SIP subset) and leaves semantics to
>>>> specs that are using it, like for example 3515
>>>>
>>>> Do you see any reason why this couldn't work here?
>>>
>>> Its a choice. In the case of sipfrag, IIRC it was defined for use with
>>> 3515. In that it isn't crucial what is in it. So it works there.
>>>
>>> In your case what goes in it is more critical. It can still be defined
>>> this way. But it will take some care to get it right.
>>>
>>> The main thing is that it be done somewhere!
>>
>> Agreed!
>>
>>> Can an INFO trickle ICE candidates for more than one m-line?
>>
>> Yes.
>>
>>> If so, it gets messy. Consider a full SDP:
>>>
>>> ...
>>> m=...
>>> a=mid:1
>>> a=candidate:...
>>> ...
>>> m=...
>>> a=candidate:...
>>> a=mid:2
>>> a=candidate:...
>>>
>>> (AFAIK the above is entirely legal.)
>>>
>>> Now consider a sipfrag for trickle from the above:
>>>
>>> a=mid:1
>>> a=candidate:...
>>> a=candidate:...
>>> a=mid:2
>>> a=candidate:...
>>>
>>> Now, which candidates go with which m-line? So, I think your sdpfrag
>>> needs special constraints that don't apply to ICE in full SDP.
>>
>> I think the problem with the above is that you are trying to present it
>> as an extract from some pre-existing offer or answer, while it isn't.
>>
>> When you are trickling you find yourself in a situation where you have
>> one or more newly learned candidates. You want to send those to the
>> remote agent.
>>
>> To do this you group them by stream (as in "m=" line). For each group
>> you start by creating an "a=mid:x" attribute that matches the one from
>> the "m=" line that created the stream. You then create and append
>> "a=candidate" lines for every candidate in that group.
>>
>> That's it.
>
> I thought that is what you had in mind.
>
> But you have said you want to use application/sdpfrag, which means it
> should be viewed as a subset of a full SDP.

Doesn't this apply to the INFO example that I pasted above? Why could it 
not be viewed as a subset of *some* valid full SDP?

> And that is especially so if
> you intend to reference m-lines in an SDP

No, there's no such intention any more.

> , but don't want to include
> them. And you haven't defined any new ordering rules for lines other
> than what exist for SDP. And SDP allows the ordering I show above, which
> doesn't work for you.

I agree that there's no such definition in the draft. I did provide 
example rules in a couple of paragraphs above though. Do you see a 
problem using this logic as basis for defining the ordering in 
trickle-ice-sip?

> I'll also mention that I think you need to explicitly reference the SDP
> offer or answer that this trickle updates. Probably by replicating the
> o-line from that SDP.

That's a good point. We do need this for SIP. I'll make sure we add it.

> So there are some significant syntax rules that need to be followed
> here. I would expect something like:
>
> - must start with o-line of the SDP it is updating
> - has one or more "media sections"
> - each media section begins with an a:mid line, referencing
>     a mid defined in the referenced SDP
> - each a:mid line is followed by one or more a:candidate lines
> ...

Sounds great to me!

> If going that way, then I think it is better to define a mime type that
> consists of exactly this syntax, with some extensibility hooks.

Are you proposing this because you think this MIME type could be reused 
by someone else?

Personally I don't really mind.

However, if we leave sdpfrag with the current intention (i.e. any SDP 
line or combination of lines work) and then define the specifics in 
trickle-ice-sip, then we have a MIME type for the a=candidate lines that 
come out of WebRTC browsers, should anyone every need this.

Cheers,
Emil


>
> 	Thanks,
> 	Paul
>
>> The receiving peer does the opposite: it parses the "a=mid:x" line, it
>> finds the stream object that it refers to, then parses all the
>> candidates and adds them there.
>>
>> Does this make sense?
>>
>> Emil
>>
>>
>>>
>>>      Thanks,
>>>      Paul
>>>
>>>> Emil
>>>>
>>>>>>> Section 5.6 again references the trickle-ice draft (section 4) for
>>>>>>> definition of the body of the info message, but that section of that
>>>>>>> draft doesn't contain any such definition.
>>>>>>
>>>>>> Will need to fix.
>>>>>>
>>>>>>> Aside from that, it occurs to me that trickle-ice, and for that
>>>>>>> matter
>>>>>>> ice in sip even without trickle, is a good candidate for
>>>>>>> preconditions.
>>>>>>> The use of preconditions could eliminate the need for half trickle.
>>>>>>> This
>>>>>>> would of course require a new precondition type, but that is doable.
>>>>>>
>>>>>> Hmmm ... that's a good point!
>>>>>>
>>>>>>> By using preconditions, you can establish the dialog without alerting
>>>>>>> the callee. Then the alert can be deferred until at least one
>>>>>>> workable
>>>>>>> path has been detected.
>>>>>>>
>>>>>>> (But I suppose the hate for preconditions is so great that this is
>>>>>>> not
>>>>>>> to be considered.)
>>>>>>
>>>>>> Well, it doesn't have to be mandatory ...
>>>>>>
>>>>>> Emil
>>>>>>
>>>>>>>
>>>>>>>             Thanks,
>>>>>>>             Paul
>>>>>>> _______________________________________________
>>>>>>> mmusic mailing list
>>>>>>> mmusic@ietf.org
>>>>>>> https://www.ietf.org/mailman/listinfo/mmusic
>>>>>>
>>>>>
>>>>> .
>>>>>
>>>>
>>>
>>>
>>
>
>

-- 
https://jitsi.org

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, Emil Ivov <emcho@jitsi.org>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hi,

I am not sure we would need to define anything new - we could simply say th=
at Trickle ICE UAs don't send Offer/Answers in PRACKSs :)

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Paul Kyzivat [pkyzivat@alum.mit.edu]
To: Emil Ivov [emcho@jitsi.org]
CC: Christer Holmberg [christer.holmberg@ericsson.com]; Stach, Thomas [thom=
as.stach@siemens-enterprise.com]; mmusic@ietf.org [mmusic@ietf.org]
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
On 7/23/13 1:06 PM, Emil Ivov wrote:

> I actually think we could pull this off with simple PRACKs. Only, rather
> than using the "100rel" tag we go for something like "100rel-lite" which
> could be defined to indicate that a UA supports PRACK but only for
> reliability (and PRACK bodies are not allowed).

This is crazy!

Every new option presents backward compatibility issues, implementation
issues, ...

It doesn't make sense to define a new option that is a subset of a long
time existing option. This is *not* rocket science. Just tell people
they need to support 100rel as a precondition for trickle ICE. If they
find trickle ice worthwhile they can implement it.

I'd be amazed if there are any SBCs that don't support PRACK. (But
Hadriel will fill us in about that.)

        Thanks,
        Paul


--_000_7594FB04B1934943A5C02806D1A2204B1C40A57FESESSMB209erics_
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1">
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
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<body>
<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">I am not sure we would need to define anything =
new - we could simply say that Trickle ICE UAs don't send Offer/Answers in =
PRACKSs :)</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Paul Kyzivat [pkyzivat@alum.mit.edu]<br>
<b>To:</b> Emil Ivov [emcho@jitsi.org]<br>
<b>CC:</b> Christer Holmberg [christer.holmberg@ericsson.com]; Stach, Thoma=
s [thomas.stach@siemens-enterprise.com]; mmusic@ietf.org [mmusic@ietf.org]<=
br>
<b>Subject:</b> Re: [MMUSIC] Trickle ICE for SIP Questions</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">On 7/23/13 1:06 PM, Emil Ivov wrote:<br>
<br>
&gt; I actually think we could pull this off with simple PRACKs. Only, rath=
er<br>
&gt; than using the &quot;100rel&quot; tag we go for something like &quot;1=
00rel-lite&quot; which<br>
&gt; could be defined to indicate that a UA supports PRACK but only for<br>
&gt; reliability (and PRACK bodies are not allowed).<br>
<br>
This is crazy!<br>
<br>
Every new option presents backward compatibility issues, implementation <br=
>
issues, ...<br>
<br>
It doesn't make sense to define a new option that is a subset of a long <br=
>
time existing option. This is *not* rocket science. Just tell people <br>
they need to support 100rel as a precondition for trickle ICE. If they <br>
find trickle ice worthwhile they can implement it.<br>
<br>
I'd be amazed if there are any SBCs that don't support PRACK. (But <br>
Hadriel will fill us in about that.)<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
<br>
</div>
</span></font>
</body>
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [MMUSIC] draft-roach-mmusic-unified-plan-00
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On Jul 25, 2013, at 6:19 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> On 7/25/13 7:41 PM, Cullen Jennings (fluffy) wrote:
>>=20
>>=20
>> On Jul 24, 2013, at 10:03 AM, Adam Roach <adam@nostrum.com> wrote:
>>=20
>>> On 7/24/13 10:55, Paul Kyzivat wrote:
>>>> This seems to be excluded, except by sending a new O/A every time the =
speaker switches, which isn't feasible. If this means that CLUE can't repre=
sent capture-encodings as m-lines, then there is no benefit to CLUE for the=
 unified plan over No Plan (Emil's multiple-sources draft).
>>>=20
>>> That's interesting. I wonder whether we should simply move to the RTP h=
eader extension described in section 3.2.1.1 as the primary means of correl=
ation. It would seem to address the case you describe.
>>>=20
>>> /a
>>=20
>> SSRC can collide too so they might be wrong but the  header extension do=
es not have that problem. It seems to me the algorithm is roughly
>>=20
>> if you have  header extension, use that, if not then if you know the SSR=
C use that, if not then if you have unique PTs, use that. Perhaps there are=
 other things to use too but roughly those get used in that order.
>=20
> I commented on this before. You don't need to have an ordering.
>=20
> If the combination of things you have is unique, use that. If not, then y=
ou have a problem.
>=20

That would be fine with me too.=20



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From: "Vijaya Mandava (vimandav)" <vimandav@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Emil Ivov <emcho@jitsi.org>
Thread-Topic: [MMUSIC] Trickle ICE for SIP Questions
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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To add more to this -
PRACK can be enabled or disabled. Trickle Ice UA's would not be affected by=
 this, Since we rely on INFOs to send trickled candidates.

Thanks,
Vijaya

From: Christer Holmberg <christer.holmberg@ericsson.com<mailto:christer.hol=
mberg@ericsson.com>>
Date: Friday, July 26, 2013 4:29 AM
To: Paul Kyzivat <pkyzivat@alum.mit.edu<mailto:pkyzivat@alum.mit.edu>>, Emi=
l Ivov <emcho@jitsi.org<mailto:emcho@jitsi.org>>
Cc: "mmusic_ietf.org" <mmusic@ietf.org<mailto:mmusic@ietf.org>>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions

I am not sure we would need to define anything new - we could simply say th=
at Trickle ICE UAs don't send Offer/Answers in PRACKSs :)


--_000_1CDFD781608D924094E43F573C3519610230DAF3xmbrcdx13ciscoc_
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Content-ID: <8E796032090E00428C291F8F8830495A@emea.cisco.com>
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif; ">
<div>To add more to this -</div>
<div>PRACK can be enabled or disabled. Trickle Ice UA's would not be affect=
ed by this, Since we rely on INFOs to send trickled candidates.</div>
<div><br>
</div>
<div>Thanks,</div>
<div>Vijaya</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.com</=
a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Friday, July 26, 2013 4:29 AM=
<br>
<span style=3D"font-weight:bold">To: </span>Paul Kyzivat &lt;<a href=3D"mai=
lto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a>&gt;, Emil Ivov &lt;<a =
href=3D"mailto:emcho@jitsi.org">emcho@jitsi.org</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>&quot;mmusic_ietf.org&quot; &lt=
;<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [MMUSIC] Trickle ICE f=
or SIP Questions<br>
</div>
<div><br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, Arial, Helvetica, =
sans-serif; font-size: 15px; font-style: normal; font-variant: normal; font=
-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; t=
ext-align: -webkit-auto; text-indent: 0px; text-transform: none; white-spac=
e: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -w=
ebkit-text-stroke-width: 0px; ">
<font face=3D"Calibri">I am not sure we would need to define anything new -=
 we could simply say that Trickle ICE UAs don't send Offer/Answers in PRACK=
Ss :)</font></div>
<br class=3D"Apple-interchange-newline">
</span>
</body>
</html>

--_000_1CDFD781608D924094E43F573C3519610230DAF3xmbrcdx13ciscoc_--

From partha@parthasarathi.co.in  Fri Jul 26 11:03:02 2013
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hi Emil,

Please read inline.

Thanks
Partha

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Emil Ivov
> Sent: Thursday, July 25, 2013 10:08 PM
> To: 'MMUSIC IETF WG'
> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
> 
> Hey Partha,
> 
> On 25.07.13, 17:48, Parthasarathi R wrote:
> > Hi Emil,
> >
> > In the mentioned situation, different To-tag with same contact header
> for
> > trickled SDP, So there will be no "N" locations.
> 
> Having the same contact header does not automatically suggest that this
> is somehow a fake fork (unless I am missing the text that says this in
> 3261). You could potentially have this if the fork is done by a B2BUA
> or
> a PSTN gateway for example.
> 
<Partha>The discussion is not about fake fork. The contact indicates
location of the UAS. In case of PSTN & B2BUA as UAS, PSTN & B2BUA contact
SIP URI is the location for UAC. From the UAC perspective, there is no "N"
location in UAS due to fake forking as you expect.</Partha>

> > My reading of the draft is that this draft is not required in case
> UPDATE
> > (RFC 3311) method is supported. UPDATE with SDP will update trickle
> > candidate within 18x.
> 
> Use of UPDATE has already been discussed and we pretty much agreed not
> to go that way.
> 
> A mail from Alan and the preceding discussion:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg10660.html
> 
> A follow up with a nice summary from Dale:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg10678.html

<Partha> Sorry for the delay discussion. I'm still seeing few aspects  
Missing from the discussion. I'll write the separate mail thread for this
</Partha> 

> 
> > The aim of the draft is to support Trickle ICE for SIP user agent
> which does
> > not support RFC 3311 for last 11 years. AFAIK, different "to tag" is
> the
> > simpler way to solve the problem in UAS side. I have concern w.r.t
> INFO
> > mechanism as it adds one method for handling SDP and complicates the
> > existing SDP O/A handling (RFC 6337).
> 
> That's the thing, trickle ICE is not part of Offer/Answer. It's a
> separate channel for agent-to-agent communication. Use of SDP is just a
> convenient coincidence.
> 
> > Including Christer in this mail thread as he replied in the parallel
> mail
> > thread.
> 
> I believe he is on the list I don't think that's an issue :)
> 
> Cheers,
> Emil
> >
> > Thanks
> > Partha
> >
> >> -----Original Message-----
> >> From: Emil Ivov [mailto:emcho@jitsi.org]
> >> Sent: Thursday, July 25, 2013 2:19 AM
> >> To: Cullen Jennings (fluffy)
> >> Cc: Vijaya Mandava (vimandav); Parthasarathi R; Alan Johnston;
> MMUSIC
> >> IETF WG
> >> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
> >>
> >>
> >>
> >> On 24.07.13, 21:56, Cullen Jennings (fluffy) wrote:
> >>>
> >>> On Jul 24, 2013, at 12:22 PM, Vijaya Mandava (vimandav)
> >> <vimandav@cisco.com> wrote:
> >>>
> >>>> 180 with different to-tag would mean call is forked.
> >>>> If uac side do not support call forking, then we cannot use this
> 180
> >>>> response to collect trickled candidates.
> >>>
> >>> If it is a UAC, it supports this so I don't see the problem.
> >>
> >> One problem is that the UAC could support it to the extent of making
> it
> >> visible to the user:
> >>
> >>     "Your contact is being alerted on these N locations"
> >>
> >> And of course there's also the fact that we would be creating
> several
> >> dialogs per call, that we'd then need to clean and garbage collect.
> >>
> >> Why would we want to go there?
> >>
> >> Emil
> >>
> >> --
> >> https://jitsi.org
> >
> >
> 
> --
> https://jitsi.org
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From christer.holmberg@ericsson.com  Fri Jul 26 11:44:02 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Vijaya Mandava (vimandav)" <vimandav@cisco.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>, Emil Ivov <emcho@jitsi.org>
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hi,

Wasn't this discussion started due to the fact that the UAS cannot send INF=
O until it knows that a dialog has been established?

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Vijaya Mandava (vimandav) [vimandav@cisco.com]
To: Christer Holmberg [christer.holmberg@ericsson.com]; Paul Kyzivat [pkyzi=
vat@alum.mit.edu]; Emil Ivov [emcho@jitsi.org]
CC: mmusic_ietf.org [mmusic@ietf.org]
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
To add more to this -
PRACK can be enabled or disabled. Trickle Ice UA's would not be affected by=
 this, Since we rely on INFOs to send trickled candidates.

Thanks,
Vijaya

From: Christer Holmberg <christer.holmberg@ericsson.com<mailto:christer.hol=
mberg@ericsson.com>>
Date: Friday, July 26, 2013 4:29 AM
To: Paul Kyzivat <pkyzivat@alum.mit.edu<mailto:pkyzivat@alum.mit.edu>>, Emi=
l Ivov <emcho@jitsi.org<mailto:emcho@jitsi.org>>
Cc: "mmusic_ietf.org" <mmusic@ietf.org<mailto:mmusic@ietf.org>>
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions

I am not sure we would need to define anything new - we could simply say th=
at Trickle ICE UAs don't send Offer/Answers in PRACKSs :)


--_000_7594FB04B1934943A5C02806D1A2204B1C40AC1AESESSMB209erics_
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<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
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</head>
<body style=3D"word-wrap:break-word; color:rgb(0,0,0); font-size:14px; font=
-family:Calibri,sans-serif">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi,</span></div>
<div>&nbsp;</div>
<div>Wasn't this discussion started due to the fact that the UAS cannot sen=
d INFO until it knows that a dialog has been established?</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Vijaya Mandava (vimandav) [vimandav@cisco.com]<br>
<b>To:</b> Christer Holmberg [christer.holmberg@ericsson.com]; Paul Kyzivat=
 [pkyzivat@alum.mit.edu]; Emil Ivov [emcho@jitsi.org]<br>
<b>CC:</b> mmusic_ietf.org [mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] Trickle ICE for SIP Questions</span>
<div>
<div>To add more to this -</div>
<div>PRACK can be enabled or disabled. Trickle Ice UA's would not be affect=
ed by this, Since we rely on INFOs to send trickled candidates.</div>
<div><br>
</div>
<div>Thanks,</div>
<div>Vijaya</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; border-bottom:medium none; border-left:medium none; padding-bottom:0i=
n; padding-left:0in; padding-right:0in; border-top:#b5c4df 1pt solid; borde=
r-right:medium none; padding-top:3pt">
<span style=3D"font-weight:bold">From: </span>Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.com</=
a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Friday, July 26, 2013 4:29 AM=
<br>
<span style=3D"font-weight:bold">To: </span>Paul Kyzivat &lt;<a href=3D"mai=
lto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a>&gt;, Emil Ivov &lt;<a =
href=3D"mailto:emcho@jitsi.org">emcho@jitsi.org</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>&quot;mmusic_ietf.org&quot; &lt=
;<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [MMUSIC] Trickle ICE f=
or SIP Questions<br>
</div>
<div><br>
</div>
<div style=3D"color:rgb(0,0,0); font-family:Calibri,Arial,Helvetica,sans-se=
rif; font-size:15px; font-style:normal; font-variant:normal; font-weight:no=
rmal; letter-spacing:normal; line-height:normal; orphans:2; text-indent:0px=
; text-transform:none; white-space:normal; widows:2; word-spacing:0px">
<font face=3D"Calibri">I am not sure we would need to define anything new -=
 we could simply say that Trickle ICE UAs don't send Offer/Answers in PRACK=
Ss :)</font></div>
<br class=3D"Apple-interchange-newline">
</span></div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C40AC1AESESSMB209erics_--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Parthasarathi R <partha@parthasarathi.co.in>, 'Emil Ivov' <emcho@jitsi.org>, 'MMUSIC IETF WG' <mmusic@ietf.org>
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Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
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Hi,

Two different UASs could have the same Contact value, if they e.g. are loca=
ted in private networks.

Of course, you could use the instance-id, but then we would need to require=
 support of that extension...

Regards,

Christer


Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Parthasarathi R [partha@parthasarathi.co.in]
To: &apos;Emil Ivov&apos; [emcho@jitsi.org]; &apos;MMUSIC IETF WG&apos; [mm=
usic@ietf.org]
Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
Hi Emil,

Please read inline.

Thanks
Partha

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Emil Ivov
> Sent: Thursday, July 25, 2013 10:08 PM
> To: 'MMUSIC IETF WG'
> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
>
> Hey Partha,
>
> On 25.07.13, 17:48, Parthasarathi R wrote:
> > Hi Emil,
> >
> > In the mentioned situation, different To-tag with same contact header
> for
> > trickled SDP, So there will be no "N" locations.
>
> Having the same contact header does not automatically suggest that this
> is somehow a fake fork (unless I am missing the text that says this in
> 3261). You could potentially have this if the fork is done by a B2BUA
> or
> a PSTN gateway for example.
>
<Partha>The discussion is not about fake fork. The contact indicates
location of the UAS. In case of PSTN & B2BUA as UAS, PSTN & B2BUA contact
SIP URI is the location for UAC. From the UAC perspective, there is no "N"
location in UAS due to fake forking as you expect.</Partha>

> > My reading of the draft is that this draft is not required in case
> UPDATE
> > (RFC 3311) method is supported. UPDATE with SDP will update trickle
> > candidate within 18x.
>
> Use of UPDATE has already been discussed and we pretty much agreed not
> to go that way.
>
> A mail from Alan and the preceding discussion:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg10660.html
>
> A follow up with a nice summary from Dale:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg10678.html

<Partha> Sorry for the delay discussion. I'm still seeing few aspects
Missing from the discussion. I'll write the separate mail thread for this
</Partha>

>
> > The aim of the draft is to support Trickle ICE for SIP user agent
> which does
> > not support RFC 3311 for last 11 years. AFAIK, different "to tag" is
> the
> > simpler way to solve the problem in UAS side. I have concern w.r.t
> INFO
> > mechanism as it adds one method for handling SDP and complicates the
> > existing SDP O/A handling (RFC 6337).
>
> That's the thing, trickle ICE is not part of Offer/Answer. It's a
> separate channel for agent-to-agent communication. Use of SDP is just a
> convenient coincidence.
>
> > Including Christer in this mail thread as he replied in the parallel
> mail
> > thread.
>
> I believe he is on the list I don't think that's an issue :)
>
> Cheers,
> Emil
> >
> > Thanks
> > Partha
> >
> >> -----Original Message-----
> >> From: Emil Ivov [mailto:emcho@jitsi.org]
> >> Sent: Thursday, July 25, 2013 2:19 AM
> >> To: Cullen Jennings (fluffy)
> >> Cc: Vijaya Mandava (vimandav); Parthasarathi R; Alan Johnston;
> MMUSIC
> >> IETF WG
> >> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
> >>
> >>
> >>
> >> On 24.07.13, 21:56, Cullen Jennings (fluffy) wrote:
> >>>
> >>> On Jul 24, 2013, at 12:22 PM, Vijaya Mandava (vimandav)
> >> <vimandav@cisco.com> wrote:
> >>>
> >>>> 180 with different to-tag would mean call is forked.
> >>>> If uac side do not support call forking, then we cannot use this
> 180
> >>>> response to collect trickled candidates.
> >>>
> >>> If it is a UAC, it supports this so I don't see the problem.
> >>
> >> One problem is that the UAC could support it to the extent of making
> it
> >> visible to the user:
> >>
> >>     "Your contact is being alerted on these N locations"
> >>
> >> And of course there's also the fact that we would be creating
> several
> >> dialogs per call, that we'd then need to clean and garbage collect.
> >>
> >> Why would we want to go there?
> >>
> >> Emil
> >>
> >> --
> >> https://jitsi.org
> >
> >
>
> --
> https://jitsi.org
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

_______________________________________________
mmusic mailing list
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 font-size:11pt">
<div><br>
Hi,</div>
<div>&nbsp;</div>
<div>Two different UASs could have the same Contact value, if they e.g. are=
 located in private networks.</div>
<div>&nbsp;</div>
<div>Of course, you could use the instance-id, but then we would need to re=
quire support of that extension...</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<div>&nbsp;</div>
<div><br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Parthasarathi R [partha@parthasarathi.co.in]<br>
<b>To:</b> &amp;apos;Emil Ivov&amp;apos; [emcho@jitsi.org]; &amp;apos;MMUSI=
C IETF WG&amp;apos; [mmusic@ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] Trickle ICE for SIP Questions</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">Hi Emil,<br>
<br>
Please read inline.<br>
<br>
Thanks<br>
Partha<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: mmusic-bounces@ietf.org [<a href=3D"mailto:mmusic-bounces@ietf.o=
rg">mailto:mmusic-bounces@ietf.org</a>] On<br>
&gt; Behalf Of Emil Ivov<br>
&gt; Sent: Thursday, July 25, 2013 10:08 PM<br>
&gt; To: 'MMUSIC IETF WG'<br>
&gt; Subject: Re: [MMUSIC] Trickle ICE for SIP Questions<br>
&gt; <br>
&gt; Hey Partha,<br>
&gt; <br>
&gt; On 25.07.13, 17:48, Parthasarathi R wrote:<br>
&gt; &gt; Hi Emil,<br>
&gt; &gt;<br>
&gt; &gt; In the mentioned situation, different To-tag with same contact he=
ader<br>
&gt; for<br>
&gt; &gt; trickled SDP, So there will be no &quot;N&quot; locations.<br>
&gt; <br>
&gt; Having the same contact header does not automatically suggest that thi=
s<br>
&gt; is somehow a fake fork (unless I am missing the text that says this in=
<br>
&gt; 3261). You could potentially have this if the fork is done by a B2BUA<=
br>
&gt; or<br>
&gt; a PSTN gateway for example.<br>
&gt; <br>
&lt;Partha&gt;The discussion is not about fake fork. The contact indicates<=
br>
location of the UAS. In case of PSTN &amp; B2BUA as UAS, PSTN &amp; B2BUA c=
ontact<br>
SIP URI is the location for UAC. From the UAC perspective, there is no &quo=
t;N&quot;<br>
location in UAS due to fake forking as you expect.&lt;/Partha&gt;<br>
<br>
&gt; &gt; My reading of the draft is that this draft is not required in cas=
e<br>
&gt; UPDATE<br>
&gt; &gt; (RFC 3311) method is supported. UPDATE with SDP will update trick=
le<br>
&gt; &gt; candidate within 18x.<br>
&gt; <br>
&gt; Use of UPDATE has already been discussed and we pretty much agreed not=
<br>
&gt; to go that way.<br>
&gt; <br>
&gt; A mail from Alan and the preceding discussion:<br>
&gt; <a href=3D"http://www.ietf.org/mail-archive/web/mmusic/current/msg1066=
0.html">http://www.ietf.org/mail-archive/web/mmusic/current/msg10660.html</=
a><br>
&gt; <br>
&gt; A follow up with a nice summary from Dale:<br>
&gt; <a href=3D"http://www.ietf.org/mail-archive/web/mmusic/current/msg1067=
8.html">http://www.ietf.org/mail-archive/web/mmusic/current/msg10678.html</=
a><br>
<br>
&lt;Partha&gt; Sorry for the delay discussion. I'm still seeing few aspects=
&nbsp; <br>
Missing from the discussion. I'll write the separate mail thread for this<b=
r>
&lt;/Partha&gt; <br>
<br>
&gt; <br>
&gt; &gt; The aim of the draft is to support Trickle ICE for SIP user agent=
<br>
&gt; which does<br>
&gt; &gt; not support RFC 3311 for last 11 years. AFAIK, different &quot;to=
 tag&quot; is<br>
&gt; the<br>
&gt; &gt; simpler way to solve the problem in UAS side. I have concern w.r.=
t<br>
&gt; INFO<br>
&gt; &gt; mechanism as it adds one method for handling SDP and complicates =
the<br>
&gt; &gt; existing SDP O/A handling (RFC 6337).<br>
&gt; <br>
&gt; That's the thing, trickle ICE is not part of Offer/Answer. It's a<br>
&gt; separate channel for agent-to-agent communication. Use of SDP is just =
a<br>
&gt; convenient coincidence.<br>
&gt; <br>
&gt; &gt; Including Christer in this mail thread as he replied in the paral=
lel<br>
&gt; mail<br>
&gt; &gt; thread.<br>
&gt; <br>
&gt; I believe he is on the list I don't think that's an issue :)<br>
&gt; <br>
&gt; Cheers,<br>
&gt; Emil<br>
&gt; &gt;<br>
&gt; &gt; Thanks<br>
&gt; &gt; Partha<br>
&gt; &gt;<br>
&gt; &gt;&gt; -----Original Message-----<br>
&gt; &gt;&gt; From: Emil Ivov [<a href=3D"mailto:emcho@jitsi.org">mailto:em=
cho@jitsi.org</a>]<br>
&gt; &gt;&gt; Sent: Thursday, July 25, 2013 2:19 AM<br>
&gt; &gt;&gt; To: Cullen Jennings (fluffy)<br>
&gt; &gt;&gt; Cc: Vijaya Mandava (vimandav); Parthasarathi R; Alan Johnston=
;<br>
&gt; MMUSIC<br>
&gt; &gt;&gt; IETF WG<br>
&gt; &gt;&gt; Subject: Re: [MMUSIC] Trickle ICE for SIP Questions<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; On 24.07.13, 21:56, Cullen Jennings (fluffy) wrote:<br>
&gt; &gt;&gt;&gt;<br>
&gt; &gt;&gt;&gt; On Jul 24, 2013, at 12:22 PM, Vijaya Mandava (vimandav)<b=
r>
&gt; &gt;&gt; &lt;vimandav@cisco.com&gt; wrote:<br>
&gt; &gt;&gt;&gt;<br>
&gt; &gt;&gt;&gt;&gt; 180 with different to-tag would mean call is forked.<=
br>
&gt; &gt;&gt;&gt;&gt; If uac side do not support call forking, then we cann=
ot use this<br>
&gt; 180<br>
&gt; &gt;&gt;&gt;&gt; response to collect trickled candidates.<br>
&gt; &gt;&gt;&gt;<br>
&gt; &gt;&gt;&gt; If it is a UAC, it supports this so I don't see the probl=
em.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; One problem is that the UAC could support it to the extent of=
 making<br>
&gt; it<br>
&gt; &gt;&gt; visible to the user:<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp; &quot;Your contact is being alerted o=
n these N locations&quot;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; And of course there's also the fact that we would be creating=
<br>
&gt; several<br>
&gt; &gt;&gt; dialogs per call, that we'd then need to clean and garbage co=
llect.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Why would we want to go there?<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Emil<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; --<br>
&gt; &gt;&gt; <a href=3D"https://jitsi.org">https://jitsi.org</a><br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; <br>
&gt; --<br>
&gt; <a href=3D"https://jitsi.org">https://jitsi.org</a><br>
&gt; _______________________________________________<br>
&gt; mmusic mailing list<br>
&gt; mmusic@ietf.org<br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.i=
etf.org/mailman/listinfo/mmusic</a><br>
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--_000_7594FB04B1934943A5C02806D1A2204B1C40ACA8ESESSMB209erics_--

From ari.keranen@ericsson.com  Sat Jul 27 01:58:42 2013
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From: =?ISO-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>
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Cc: Flemming Andreasen <fandreas@cisco.com>
Subject: [MMUSIC] Note takers for MMUSIC at IETF 87
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Hi all,

We are looking for two note takers and a jabber scribe for the MMUSIC 
sessions at the Berlin meeting. If you are attending and would be able 
to take notes, please contact the chairs.


Thanks,
Ari & Flemming

From harald@alvestrand.no  Sat Jul 27 08:49:42 2013
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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On 07/24/2013 11:27 PM, Adam Roach wrote:
> On 7/24/13 16:17, Emil Ivov wrote:
>>
>>
>> On 24.07.13, 20:31, Eric Rescorla wrote:
>>>
>>> It depends on the context obviously. But in the case of (for
>>> instance) a
>>> Hangouts
>>> like system if the conference bridge were the offerer, it would send a
>>> pile of
>>> m-lines with only one being sendrecv (or maybe even recvonly) for the
>>> user's media and the rest being sendonly.
>>
>> Sorry, I am still not getting it. How does it know exactly how many
>> "m=" lines to place in a=sendrecv?
>
> I think you need to re-read and meditate on section 5.1 of RFC 3264.
> This stuff is actually pretty well defined, and I'm not sure having an
> on-list tutorial is ideal.
>
> Your follow-up questions are likely answered in section 6.1 of that
> same document.

Adam, I couldn't find anything in those sections that addressed the
question either, so either I'm as uneducated in RFC 3264 as Eric, or
we're talking about 2 different questions.

I perceive the question to be a variant of the scenario that's
explicitly outlined in the -unified- draft in section 3.5:

3.5.  Negotiation of Stream Ordinality

   Within advanced applications, circumstances can easily arise in which
   the party creating the offer does not know ahead of time the number
   of streams the remote party will desire.  For example, in a meet-me
   videoconference application that sends a separate stream for each
   participant, a client creating an offer to send to the conference
   focus does not necessarily know how many video streams to indicate in
   its SDP.  Although this can be potentially be solved in an
   application-specific way (e.g., by always offering the maximum number
   of streams known to be supported by the application), this is not
   always desirable or even possible.

   To address this situation, a three-way handshake can be employed.

In addition to the situation outlined there, if an offerer places an
m-line in "sendonly" or "inactive" mode, it is making it unavailable for
the answerer, and I believe that a consistent answer is the same as in
section 3.5:

A three-way handshake can be employed.

In fact, I think it MUST be employed.


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


From emil@sip-communicator.org  Sat Jul 27 09:16:51 2013
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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On 25.07.13, 02:00, Adam Roach wrote:
> On 7/24/13 17:58, Emil Ivov wrote:
>> So you are confirming that the number of media sources, that the
>> bridge expected to get from the callee, was hard-coded or
>> pre-provisioned. Right?
>
> It's part of the application. Sure. "Hard-coded" is a reasonable way to=

> think about it.

A follow up question here then: How strong of an indication is the=20
direction? Let's say you do offer me a session with eight streams, seven =

of which are a=3Dsendonly and only one is a=3Dsendrecv. Does this mean th=
at=20
I should never consider sending you more than one stream?

In other words, once we complete the first offer/answer, can I send you=20
a new offer with additional a=3Dsendrecv "m=3D" lines? Following 3264 wou=
ld=20
mean that I could ... but if I did, I would essentially be ignore your=20
wish to only receive a single stream.

Another question, when rendering the a=3Dsendonly lines in the above=20
8-stream offer, should the callee's GUI label them as being "on hold"?=20
Given the original scenario, I don't think we'd want that, but how does=20
the UA know this? Do we extend 3264 to say that a call is either=20
entirely on hold or it is not?

One more: it is entirely possible that I missed this but I couldn't find =

anything in 3264 that says whether or not a callee can, at some point,=20
change change the direction of an a=3Dsendonly line to a=3Dsendrecv. Do w=
e=20
need to say a word about this?

>> To simplify let's get back to I=F1aki's scenario of 8 "m=3D" lines in =
the
>> offer, but lets also assume that different callees can answer with
>> anything between 1 and 8 sources.
>>
>> In this case, do you set all your "m=3D" lines to "a=3Dsendrecv"?
>
> If the bridge wanted to deal with receiving that many media streams, su=
re.
>
>> If so, we are back to I=F1aki's original question: how does a
>> single-source callee know which if these "m=3D" lines to pick for its
>> own stream?
>>
>> I suppose one possible answer would be: it just takes any "m=3D" line
>> that matches its needs and sends stuff there. It also sets everything
>> else to "a=3Drecvonly" in the answer. Is this what you'd suggest?
>>
>> Or maybe the callee sends an answer where it sets everything to
>> "a=3Drecvonly" and then re-offers with a ninth "a=3Dsendonly" "m=3D" l=
ine?
>>
>> Or does the callee use app-specific signalling to tell the bridge that=

>> it has N sources so that the bridge would offer with N "a=3Dsendrecv"
>> "m=3D" lines and 8-N "a=3Dsendonly" "m=3D" lines?
>>
>> Or are all three of the above legal?
>
> Yes, the first one is what I would recommend.

OK

> The second one is certainly valid, but why would you do that? It delays=

> the callee-to-caller media unnecessarily.

I am just trying to figure out the exact semantics of the 3/4-way=20
handshake. So far the way I get it is that I express as much of my=20
intentions as I could in my answer. Then, if I need to say something=20
more, and only then, I send another offer. Is this correct? Or is it=20
more specific than that and I should only follow up with an update offer =

in case I'd like to add streams?

> The third one is fine, too, assuming you want to bind yourself to that
> kind of signaling (it may complicate interdomain federation, if that
> ever becomes important to you).

Well I do care about federation by I worry more about it when we start=20
mentioning hard-coded assumptions in MCUs. An application-layer protocol =

could always be properly specified at some point (and some already are).

> But this is all pretty easy to derive from RFC 3264 section 6.1, really=
=2E

I don't really think it is that obvious (otherwise we wouldn't really=20
need this document, would we)

I think the issue with directions is that in 3264 they only have an "m=3D=
"=20
line scope. What you and Eric have implied is that now they actually=20
mean something for the entire session.

Emil

--=20
https://jitsi.org


From christer.holmberg@ericsson.com  Sat Jul 27 12:58:28 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Emil Ivov <emcho@jitsi.org>, Adam Roach <adam@nostrum.com>
Thread-Topic: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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Cc: =?iso-8859-1?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>, MMUSIC IETF WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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Hi Emil,

I think you questions are not specific to the unified plan, but to Offer/An=
swer in general.

First, you can always offer additional m- lines. The receiver may accept th=
em, or it may not, but just because you previously received an Offer with 8=
 m- lines doesn't automatically means the Offerer can't handle more.

Second, it is allowed to send an Offer, where you suggest to change a sendo=
nly m- line to sendrecv. Again, the receiver may accept it, or it may not.

Regards,

Christer



Sent from Windows using TouchDown (www.nitrodesk.com)

-----Original Message-----
From: Emil Ivov [emcho@jitsi.org]
To: Adam Roach [adam@nostrum.com]
CC: I=C3=B1aki Baz Castillo [ibc@aliax.net]; MMUSIC IETF WG [mmusic@ietf.or=
g]
Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
On 25.07.13, 02:00, Adam Roach wrote:
> On 7/24/13 17:58, Emil Ivov wrote:
>> So you are confirming that the number of media sources, that the
>> bridge expected to get from the callee, was hard-coded or
>> pre-provisioned. Right?
>
> It's part of the application. Sure. "Hard-coded" is a reasonable way to
> think about it.

A follow up question here then: How strong of an indication is the
direction? Let's say you do offer me a session with eight streams, seven
of which are a=3Dsendonly and only one is a=3Dsendrecv. Does this mean that
I should never consider sending you more than one stream?

In other words, once we complete the first offer/answer, can I send you
a new offer with additional a=3Dsendrecv "m=3D" lines? Following 3264 would
mean that I could ... but if I did, I would essentially be ignore your
wish to only receive a single stream.

Another question, when rendering the a=3Dsendonly lines in the above
8-stream offer, should the callee's GUI label them as being "on hold"?
Given the original scenario, I don't think we'd want that, but how does
the UA know this? Do we extend 3264 to say that a call is either
entirely on hold or it is not?

One more: it is entirely possible that I missed this but I couldn't find
anything in 3264 that says whether or not a callee can, at some point,
change change the direction of an a=3Dsendonly line to a=3Dsendrecv. Do we
need to say a word about this?

>> To simplify let's get back to I=F1aki's scenario of 8 "m=3D" lines in th=
e
>> offer, but lets also assume that different callees can answer with
>> anything between 1 and 8 sources.
>>
>> In this case, do you set all your "m=3D" lines to "a=3Dsendrecv"?
>
> If the bridge wanted to deal with receiving that many media streams, sure=
.
>
>> If so, we are back to I=F1aki's original question: how does a
>> single-source callee know which if these "m=3D" lines to pick for its
>> own stream?
>>
>> I suppose one possible answer would be: it just takes any "m=3D" line
>> that matches its needs and sends stuff there. It also sets everything
>> else to "a=3Drecvonly" in the answer. Is this what you'd suggest?
>>
>> Or maybe the callee sends an answer where it sets everything to
>> "a=3Drecvonly" and then re-offers with a ninth "a=3Dsendonly" "m=3D" lin=
e?
>>
>> Or does the callee use app-specific signalling to tell the bridge that
>> it has N sources so that the bridge would offer with N "a=3Dsendrecv"
>> "m=3D" lines and 8-N "a=3Dsendonly" "m=3D" lines?
>>
>> Or are all three of the above legal?
>
> Yes, the first one is what I would recommend.

OK

> The second one is certainly valid, but why would you do that? It delays
> the callee-to-caller media unnecessarily.

I am just trying to figure out the exact semantics of the 3/4-way
handshake. So far the way I get it is that I express as much of my
intentions as I could in my answer. Then, if I need to say something
more, and only then, I send another offer. Is this correct? Or is it
more specific than that and I should only follow up with an update offer
in case I'd like to add streams?

> The third one is fine, too, assuming you want to bind yourself to that
> kind of signaling (it may complicate interdomain federation, if that
> ever becomes important to you).

Well I do care about federation by I worry more about it when we start
mentioning hard-coded assumptions in MCUs. An application-layer protocol
could always be properly specified at some point (and some already are).

> But this is all pretty easy to derive from RFC 3264 section 6.1, really.

I don't really think it is that obvious (otherwise we wouldn't really
need this document, would we)

I think the issue with directions is that in 3264 they only have an "m=3D"
line scope. What you and Eric have implied is that now they actually
mean something for the entire session.

Emil

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<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
ding-left: 4pt; border-left: #800000 2px solid; } --></style>
</head>
<body>
<div style=3D"font-family:Calibri,Arial,Helvetica,sans-serif; font-size:11p=
t; color:black">
<div><span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-s=
erif; font-size:11pt">Hi Emil,</span></div>
<div>&nbsp;</div>
<div><font face=3D"Calibri">I think you questions are not specific to the u=
nified plan, but to Offer/Answer in general.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">First, you can always offer additional m- lines=
. The receiver may accept them, or it may not, but just because you previou=
sly received an Offer with 8 m- lines doesn't automatically means the Offer=
er can't handle more.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Second, it is allowed to send an Offer, where y=
ou suggest to change a sendonly m- line to sendrecv. Again, the receiver ma=
y accept it, or it may not.</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Regards,</font></div>
<div><font face=3D"Calibri"></font>&nbsp;</div>
<div><font face=3D"Calibri">Christer</font></div>
<span style=3D"color:black; font-family:Calibri,Arial,Helvetica,sans-serif;=
 font-size:11pt">
<div>&nbsp;</div>
<div><br>
<br>
Sent from <b><i>Windows</i></b> using <b style=3D"color:blue">TouchDown</b>=
 (www.nitrodesk.com)</div>
</span>
<div></div>
<br>
<span style=3D"color:black">-----Original Message-----<br>
<b>From:</b> Emil Ivov [emcho@jitsi.org]<br>
<b>To:</b> Adam Roach [adam@nostrum.com]<br>
<b>CC:</b> I=C3=B1aki Baz Castillo [ibc@aliax.net]; MMUSIC IETF WG [mmusic@=
ietf.org]<br>
<b>Subject:</b> Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lin=
es</span></div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">On 25.07.13, 02:00, Adam Roach wrote:<br>
&gt; On 7/24/13 17:58, Emil Ivov wrote:<br>
&gt;&gt; So you are confirming that the number of media sources, that the<b=
r>
&gt;&gt; bridge expected to get from the callee, was hard-coded or<br>
&gt;&gt; pre-provisioned. Right?<br>
&gt;<br>
&gt; It's part of the application. Sure. &quot;Hard-coded&quot; is a reason=
able way to<br>
&gt; think about it.<br>
<br>
A follow up question here then: How strong of an indication is the <br>
direction? Let's say you do offer me a session with eight streams, seven <b=
r>
of which are a=3Dsendonly and only one is a=3Dsendrecv. Does this mean that=
 <br>
I should never consider sending you more than one stream?<br>
<br>
In other words, once we complete the first offer/answer, can I send you <br=
>
a new offer with additional a=3Dsendrecv &quot;m=3D&quot; lines? Following =
3264 would <br>
mean that I could ... but if I did, I would essentially be ignore your <br>
wish to only receive a single stream.<br>
<br>
Another question, when rendering the a=3Dsendonly lines in the above <br>
8-stream offer, should the callee's GUI label them as being &quot;on hold&q=
uot;? <br>
Given the original scenario, I don't think we'd want that, but how does <br=
>
the UA know this? Do we extend 3264 to say that a call is either <br>
entirely on hold or it is not?<br>
<br>
One more: it is entirely possible that I missed this but I couldn't find <b=
r>
anything in 3264 that says whether or not a callee can, at some point, <br>
change change the direction of an a=3Dsendonly line to a=3Dsendrecv. Do we =
<br>
need to say a word about this?<br>
<br>
&gt;&gt; To simplify let's get back to I=F1aki's scenario of 8 &quot;m=3D&q=
uot; lines in the<br>
&gt;&gt; offer, but lets also assume that different callees can answer with=
<br>
&gt;&gt; anything between 1 and 8 sources.<br>
&gt;&gt;<br>
&gt;&gt; In this case, do you set all your &quot;m=3D&quot; lines to &quot;=
a=3Dsendrecv&quot;?<br>
&gt;<br>
&gt; If the bridge wanted to deal with receiving that many media streams, s=
ure.<br>
&gt;<br>
&gt;&gt; If so, we are back to I=F1aki's original question: how does a<br>
&gt;&gt; single-source callee know which if these &quot;m=3D&quot; lines to=
 pick for its<br>
&gt;&gt; own stream?<br>
&gt;&gt;<br>
&gt;&gt; I suppose one possible answer would be: it just takes any &quot;m=
=3D&quot; line<br>
&gt;&gt; that matches its needs and sends stuff there. It also sets everyth=
ing<br>
&gt;&gt; else to &quot;a=3Drecvonly&quot; in the answer. Is this what you'd=
 suggest?<br>
&gt;&gt;<br>
&gt;&gt; Or maybe the callee sends an answer where it sets everything to<br=
>
&gt;&gt; &quot;a=3Drecvonly&quot; and then re-offers with a ninth &quot;a=
=3Dsendonly&quot; &quot;m=3D&quot; line?<br>
&gt;&gt;<br>
&gt;&gt; Or does the callee use app-specific signalling to tell the bridge =
that<br>
&gt;&gt; it has N sources so that the bridge would offer with N &quot;a=3Ds=
endrecv&quot;<br>
&gt;&gt; &quot;m=3D&quot; lines and 8-N &quot;a=3Dsendonly&quot; &quot;m=3D=
&quot; lines?<br>
&gt;&gt;<br>
&gt;&gt; Or are all three of the above legal?<br>
&gt;<br>
&gt; Yes, the first one is what I would recommend.<br>
<br>
OK<br>
<br>
&gt; The second one is certainly valid, but why would you do that? It delay=
s<br>
&gt; the callee-to-caller media unnecessarily.<br>
<br>
I am just trying to figure out the exact semantics of the 3/4-way <br>
handshake. So far the way I get it is that I express as much of my <br>
intentions as I could in my answer. Then, if I need to say something <br>
more, and only then, I send another offer. Is this correct? Or is it <br>
more specific than that and I should only follow up with an update offer <b=
r>
in case I'd like to add streams?<br>
<br>
&gt; The third one is fine, too, assuming you want to bind yourself to that=
<br>
&gt; kind of signaling (it may complicate interdomain federation, if that<b=
r>
&gt; ever becomes important to you).<br>
<br>
Well I do care about federation by I worry more about it when we start <br>
mentioning hard-coded assumptions in MCUs. An application-layer protocol <b=
r>
could always be properly specified at some point (and some already are).<br=
>
<br>
&gt; But this is all pretty easy to derive from RFC 3264 section 6.1, reall=
y.<br>
<br>
I don't really think it is that obvious (otherwise we wouldn't really <br>
need this document, would we)<br>
<br>
I think the issue with directions is that in 3264 they only have an &quot;m=
=3D&quot; <br>
line scope. What you and Eric have implied is that now they actually <br>
mean something for the entire session.<br>
<br>
Emil<br>
<br>
-- <br>
<a href=3D"https://jitsi.org">https://jitsi.org</a><br>
<br>
_______________________________________________<br>
mmusic mailing list<br>
mmusic@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.o=
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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On 7/25/13 2:00 AM, Adam Roach wrote:
> On 7/24/13 17:58, Emil Ivov wrote:
>> So you are confirming that the number of media sources, that the
>> bridge expected to get from the callee, was hard-coded or
>> pre-provisioned. Right?
>
> It's part of the application. Sure. "Hard-coded" is a reasonable way to
> think about it.
>
>> To simplify let's get back to Iñaki's scenario of 8 "m=" lines in the
>> offer, but lets also assume that different callees can answer with
>> anything between 1 and 8 sources.
>>
>> In this case, do you set all your "m=" lines to "a=sendrecv"?
>
> If the bridge wanted to deal with receiving that many media streams, sure.
>
>> If so, we are back to Iñaki's original question: how does a
>> single-source callee know which if these "m=" lines to pick for its
>> own stream?
>>
>> I suppose one possible answer would be: it just takes any "m=" line
>> that matches its needs and sends stuff there. It also sets everything
>> else to "a=recvonly" in the answer. Is this what you'd suggest?
>>
>> Or maybe the callee sends an answer where it sets everything to
>> "a=recvonly" and then re-offers with a ninth "a=sendonly" "m=" line?
>>
>> Or does the callee use app-specific signalling to tell the bridge that
>> it has N sources so that the bridge would offer with N "a=sendrecv"
>> "m=" lines and 8-N "a=sendonly" "m=" lines?
>>
>> Or are all three of the above legal?
>
> Yes, the first one is what I would recommend.

There are potential problems with this approach. Having answered this 
and chosen the m-line for bidirectional use all is well for the moment. 
But then, one side may make another offer, proposing to change something 
about that m-line. At that point, its unclear whether the change is 
intended for the stream it is sending, the one it is receiving, or both.

For instance, some devices can really only use one codec, so the new 
offer might be to lock down to the first codec that was chosen. But it 
may be that the desire by the other side was to use a different codec 
for each direction.

I don't have an answer here. It is certainly tempting to use an m-line 
bidirectionally, at least in absence of bundle. With bundle there is a 
lot less motivation to do that.

	Thanks,
	Paul

> The second one is certainly valid, but why would you do that? It delays
> the callee-to-caller media unnecessarily.
>
> The third one is fine, too, assuming you want to bind yourself to that
> kind of signaling (it may complicate interdomain federation, if that
> ever becomes important to you).
>
> But this is all pretty easy to derive from RFC 3264 section 6.1, really.
>
> /a
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From emil@sip-communicator.org  Sun Jul 28 04:04:36 2013
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Date: Sun, 28 Jul 2013 13:04:21 +0200
From: Emil Ivov <emcho@jitsi.org>
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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On 27.07.13, 21:58, Christer Holmberg wrote:
> Hi Emil,
> I think you questions are not specific to the unified plan, but to
> Offer/Answer in general.

Well that's the thing, they *are* about unified plan. Looking for=20
answers to these questions in 3264 would be the same as looking for=20
traffic regulations and ordinances in the manual of your car.

The answers you can get from 3264 are that pretty much anything and=20
everything is possible. Some of that ambiguity would have to be removed. =

Ideally we'd do this in the spec but if we don't, then implementations=20
would do it implicitly.

> First, you can always offer additional m- lines. The receiver may accep=
t
> them, or it may not, but just because you previously received an Offer
> with 8 m- lines doesn't automatically means the Offerer can't handle mo=
re.
> Second, it is allowed to send an Offer, where you suggest to change a
> sendonly m- line to sendrecv. Again, the receiver may accept it, or it

This is a great example of how 3264 is not enough.

Let's say for example that you want to add a new stream by switching an=20
a=3Dsendonly "m=3D" line to a=3Dsendrecv. However, I consider this line t=
o be=20
"on hold" and I don't want you thinking that you can unhold it, so I=20
reject it. At the same time I am totally fine with you adding a new one=20
for your stream.

Inversely, I might have sent you an offer with 8 "m=3D" lines as an=20
indication that this is the maximum number of streams that I can deal=20
with (this is actually labelled as bad practice in unified but I didn't=20
find an alternative). In this case, I might be OK with you changing my=20
directions to enable new streams but I would reject new "m=3D" lines=20
because I don't want to have any more streams than the 8 already in the=20
SDP.

In both of the above cases there are different ways of achieving the=20
same thing and each time one would work, while others wouldn't. Yet it=20
would be hard for the remote party to pick the right one.


Paul just made a case for using different sets of "m=3D" lines for either=
=20
direction. This is totally valid from a 3264 perspective, but without=20
further information, an implementation would have to go through a number =

of trials and errors until it figures out that this is how you expect=20
things to happen.

Emil



> may not.
> Regards,
> Christer
>
>
> Sent from */Windows/* using *TouchDown* (www.nitrodesk.com)
>
> -----Original Message-----
> *From:* Emil Ivov [emcho@jitsi.org]
> *To:* Adam Roach [adam@nostrum.com]
> *CC:* I=C3=B1aki Baz Castillo [ibc@aliax.net]; MMUSIC IETF WG [mmusic@i=
etf.org]
> *Subject:* Re: [MMUSIC] [rtcweb] About unified Plan and multiple m line=
s
> On 25.07.13, 02:00, Adam Roach wrote:
>> On 7/24/13 17:58, Emil Ivov wrote:
>>> So you are confirming that the number of media sources, that the
>>> bridge expected to get from the callee, was hard-coded or
>>> pre-provisioned. Right?
>>
>> It's part of the application. Sure. "Hard-coded" is a reasonable way t=
o
>> think about it.
>
> A follow up question here then: How strong of an indication is the
> direction? Let's say you do offer me a session with eight streams, seve=
n
> of which are a=3Dsendonly and only one is a=3Dsendrecv. Does this mean =
that
> I should never consider sending you more than one stream?
>
> In other words, once we complete the first offer/answer, can I send you=

> a new offer with additional a=3Dsendrecv "m=3D" lines? Following 3264 w=
ould
> mean that I could ... but if I did, I would essentially be ignore your
> wish to only receive a single stream.
>
> Another question, when rendering the a=3Dsendonly lines in the above
> 8-stream offer, should the callee's GUI label them as being "on hold"?
> Given the original scenario, I don't think we'd want that, but how does=

> the UA know this? Do we extend 3264 to say that a call is either
> entirely on hold or it is not?
>
> One more: it is entirely possible that I missed this but I couldn't fin=
d
> anything in 3264 that says whether or not a callee can, at some point,
> change change the direction of an a=3Dsendonly line to a=3Dsendrecv. Do=
 we
> need to say a word about this?
>
>>> To simplify let's get back to I=F1aki's scenario of 8 "m=3D" lines in=
 the
>>> offer, but lets also assume that different callees can answer with
>>> anything between 1 and 8 sources.
>>>
>>> In this case, do you set all your "m=3D" lines to "a=3Dsendrecv"?
>>
>> If the bridge wanted to deal with receiving that many media streams, s=
ure.
>>
>>> If so, we are back to I=F1aki's original question: how does a
>>> single-source callee know which if these "m=3D" lines to pick for its=

>>> own stream?
>>>
>>> I suppose one possible answer would be: it just takes any "m=3D" line=

>>> that matches its needs and sends stuff there. It also sets everything=

>>> else to "a=3Drecvonly" in the answer. Is this what you'd suggest?
>>>
>>> Or maybe the callee sends an answer where it sets everything to
>>> "a=3Drecvonly" and then re-offers with a ninth "a=3Dsendonly" "m=3D" =
line?
>>>
>>> Or does the callee use app-specific signalling to tell the bridge tha=
t
>>> it has N sources so that the bridge would offer with N "a=3Dsendrecv"=

>>> "m=3D" lines and 8-N "a=3Dsendonly" "m=3D" lines?
>>>
>>> Or are all three of the above legal?
>>
>> Yes, the first one is what I would recommend.
>
> OK
>
>> The second one is certainly valid, but why would you do that? It delay=
s
>> the callee-to-caller media unnecessarily.
>
> I am just trying to figure out the exact semantics of the 3/4-way
> handshake. So far the way I get it is that I express as much of my
> intentions as I could in my answer. Then, if I need to say something
> more, and only then, I send another offer. Is this correct? Or is it
> more specific than that and I should only follow up with an update offe=
r
> in case I'd like to add streams?
>
>> The third one is fine, too, assuming you want to bind yourself to that=

>> kind of signaling (it may complicate interdomain federation, if that
>> ever becomes important to you).
>
> Well I do care about federation by I worry more about it when we start
> mentioning hard-coded assumptions in MCUs. An application-layer protoco=
l
> could always be properly specified at some point (and some already are)=
=2E
>
>> But this is all pretty easy to derive from RFC 3264 section 6.1, reall=
y.
>
> I don't really think it is that obvious (otherwise we wouldn't really
> need this document, would we)
>
> I think the issue with directions is that in 3264 they only have an "m=3D=
"
> line scope. What you and Eric have implied is that now they actually
> mean something for the entire session.
>
> Emil
>
> --
> https://jitsi.org
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

--=20
https://jitsi.org


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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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On 7/28/13 1:04 PM, Emil Ivov wrote:

> Paul just made a case for using different sets of "m=" lines for either
> direction. This is totally valid from a 3264 perspective, but without
> further information, an implementation would have to go through a number
> of trials and errors until it figures out that this is how you expect
> things to happen.

I don't claim to know what is *right* here. I commented on some 
difficulties of using m-lines bidirectionally.

ISTM that 3264 is predicated on sendrecv describing a bi-directional 
usage, rather than two uni-directional usages. This can be inferred from 
the use of a single set of matched attributes for the two directions - 
no ability to say "use codec X in one direction and codec Y in the other 
direction."

But I am not aware of any clear guidelines about when you logically have 
a single bidirectional usage vs. two unidirectional usages.

I think we are lacking a conceptual model of the distinction.

In certain cases history provides a precedent. In particular, a basic 
audio call. In that case we can how common usage links the two 
directions: when I want to "put the call on hold" I want to stop 
receiving the stream from the far end and I either want to stop sending, 
or else I will send some sort of MoH stuff. These actions are viewed as 
coupled, so using a single m-line works out.

I suppose, if I had one sendonly and one recvonly audio m-line, then I 
could figure out what to to with the SDP. But would the other side 
understand?

But when you start adding streams - even audio/video - then those common 
usages provide little guidance for what to do. Does hold do make the 
same changes to both audio and video? Or are they managed separately.

And once I have more than one audio or video in each direction, it 
becomes quite unclear.

We have been living in blissful ignorance. Once we start seriously 
thinking about multiple audio or video m-lines we are forced to actually 
think about this stuff, make some decisons, and write them down.

	Thanks,
	Paul


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... chiming in after vacation

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Emil Ivov
> Sent: Wednesday, 24 July, 2013 12:01
> To: Alan Johnston
> Cc: Cullen Jennings (fluffy); MMUSIC IETF WG
> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
>=20
> Hey Alan, Cullen,
>=20
> On 24.07.13, 11:45, Alan Johnston wrote:
> > Agree with Cullen - if there is a reasonable approach (such as
> > retransmitting 180) that avoids PRACK, then we should use this
> approach.
>=20
> There is one and it comes down to requiring the remote side to send an
> INFO request as soon as it gets it. The INFO request could contain
> trickled candidates (in the case of full trickle) or just
> end-of-candidates (in the case of half-trickle).
>=20
> This could work and, personally, I have no issue with it.
[TS] I think this also works for me.
>=20
> I believe the suggestion that Christer made was: given how the above
> basically works as a PRACK minus the O/A and given how the O/A in
> PRACKs
> seems to be a deal breaker for many,=20
[TS] It is not only the O/A in PRACK, but also the arbitrary number of furt=
her O/As in UPDATEs before the 200OK to the initial INVITE.
> then we might want to generalize
> the above mechanism so that other specs can also use it.
[TS] Yes, I think that would be good.=20
Paul also seemed to propose to do something generic with my INFO/481/ re-se=
nd 18x proposal.=20
http://www.ietf.org/mail-archive/web/mmusic/current/msg11754.html=20
I liked that as well, but as said above I could also live with acknowledgin=
g the 18x with an immediate INFO from the UAC.
Regards=20
Thomas
>=20
> Cheers,
> Emil
>=20
>=20
> > - Alan -
> >
> >
> > On Tue, Jul 23, 2013 at 11:28 PM, Cullen Jennings (fluffy)
> > <fluffy@cisco.com <mailto:fluffy@cisco.com>> wrote:
> >
> >
> >     I prefer the 180 resend approach and all the right things are
> >     already looking at them.  I have always objected to mandating use
> of
> >     PRACK. Obviously I'm fine with things that have PRACK can use it
> but
> >     I want some solution for things that don't.
> >
> >     One small note, the SDP in the 180 can change as long as the to-
> tag
> >     is also changed.
> >
> >
> >
> >     On Jul 3, 2013, at 8:13 AM, Emil Ivov <emcho@jitsi.org
> >     <mailto:emcho@jitsi.org>> wrote:
> >
> >      > Hey all,
> >      >
> >      > Christer, Enrico and I are preparing the next version of
> Trickle
> >     ICE for SIP. Now that discussions on BUNDLE and the plans seem to
> be
> >     winding down, we wanted to run a few questions by the working
> group.
> >      >
> >      > Q1: Making reliable provisional responses and PRACK mandatory.
> >     Obviously this would be nice to avoid, so the question is: is
> there
> >     a reasonable mechanism to achieve this (and by reasonable, we
> mean
> >     something that wouldn't be harder than implementing support for
> PRACK).
> >      >
> >      > There was some discussion about this back in April and there
> was
> >     a suggestion for implementing a 5245-style hack where the
> answerer
> >     basically resends the 180 until it knows that it has been
> received.
> >      >
> >      > 5245 uses connectivity checks for this (i.e. it stops 180
> >     retransmissions when the first connectivity check is received)
> but
> >     we don't have that option here since the 180 may contain either
> none
> >     or only host candidates so there are strong chances that no
> binding
> >     request would be received on them.
> >      >
> >      > Thomas also suggested a second option which would be to also
> use
> >     INFO requests with trickled candidates as an indication that 180
> was
> >     received. This however wouldn't work with half trickle so we are
> >     basically back to vanilla ICE for all (non-re) INVITEs.
> >      >
> >      > Another option would be to mandate an INFO request with
> >     "end-of-candidates" in response to the 180, but that would be
> just
> >     the same as mandating PRACK support.
> >      >
> >      > Thomas also suggested that the answerer can start sending
> INFOs
> >     right after it sends its answer in the 180 and then it can just
> >     resend the 180 if the INFOs result in a 481 response.
> >      >
> >      > Personally I think this could potentially be made to work, but
> it
> >     would imply a level of complexity that considerably exceeds PRACK
> >     support.
> >      >
> >      > Opinions?
> >      >
> >      > Q2: How do we send INFOs? Are they blocking or do we just send
> >     them in parallel? If the latter, then what happens when an INFO
> >     fails because it is received out of order? Do we just tell the
> >     application to resend the candidates asap?
> >      >
> >      > This also leads to the following question:
> >      >
> >      > Q3: What exactly do we send in INFOs? Just the latest batch of
> >     freshly learned candidates or all candidates we've learned so
> far?
> >     Dale suggested that if we do this cumulatively we wouldn't need
> to
> >     worry about the case with the out-of-order INFOs from Q2 since
> the
> >     information gets resent anyway. A drawback here would obviously
> be
> >     that this adds more complexity for trickle ICE users (WebRTC
> >     applications specifically)
> >      >
> >      > A third option would be to allow both and leave it to the
> >     application.
> >      >
> >      > Comments are most welcome!
> >      >
> >      > Emil
> >      >
> >      > --
> >      > https://jitsi.org
> >      > _______________________________________________
> >      > 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
> >
> >
>=20
> --
> https://jitsi.org
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

From partha@parthasarathi.co.in  Sun Jul 28 10:57:31 2013
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Emil Ivov'" <emcho@jitsi.org>, "'MMUSIC IETF WG'" <mmusic@ietf.org>
References: <1CDFD781608D924094E43F573C351961BDE2DC@xmb-rcd-x13.cisco.com>	<C5E08FE080ACFD4DAE31E4BDBF944EB11360D0D2@xmb-aln-x02.cisco.com>	<51F03DC8.70908@jitsi.org> <00d401ce894e$70aa81c0$51ff8540$@co.in> <51F15478.3060104@jitsi.org>
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Date: Sun, 28 Jul 2013 23:27:16 +0530
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Cc: 'Alan Johnston' <alan.b.johnston@gmail.com>, "'Dale R. Worley'" <worley@ariadne.com>
Subject: [MMUSIC] UPDATE vs INFO for SIP Trickle ICE [was RE: Trickle ICE for SIP Questions]
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Hi Emil & all,

I had gone through UPDATE vs INFO argument for SIP trickle ICE. As Trickle
ICE is candidate sharing only and also shall happen incrementally, so the
INFO based back channel mechanism was preferred. I have concern with SDP
back channel mechanism after the first O/A is complete as there are more
possibity for race conditions between INFO from UAS and UPDATE from UAC.
UPDATE from UAC is sent across for multiple reasons and  I'll list update
few of the scenarios which come to my mind immediately :

1) Bundle
2) ICE restart
3) Commit for the single codec after negotiation

INFO for trickle ICE leads to confusion due to race conditions between
UPDATE from UAC and un-ordered INFO from UAS. 

Also, it is not clear to me how media IP address as part back channel as it
is part of SDP O/A. In case SDP is designed to carry to media codec
negotiation & media IP address separately (like Jingle/H.245), back channel
mechanism works well 

Thanks
Partha 
 

> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On
> Behalf Of Emil Ivov
> Sent: Thursday, July 25, 2013 10:08 PM
> To: 'MMUSIC IETF WG'
> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
> 
> Hey Partha,
> 
> On 25.07.13, 17:48, Parthasarathi R wrote:
> > Hi Emil,
> >
> > In the mentioned situation, different To-tag with same contact header
> for
> > trickled SDP, So there will be no "N" locations.
> 
> Having the same contact header does not automatically suggest that this
> is somehow a fake fork (unless I am missing the text that says this in
> 3261). You could potentially have this if the fork is done by a B2BUA
> or
> a PSTN gateway for example.
> 
> > My reading of the draft is that this draft is not required in case
> UPDATE
> > (RFC 3311) method is supported. UPDATE with SDP will update trickle
> > candidate within 18x.
> 
> Use of UPDATE has already been discussed and we pretty much agreed not
> to go that way.
> 
> A mail from Alan and the preceding discussion:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg10660.html
> 
> A follow up with a nice summary from Dale:
> http://www.ietf.org/mail-archive/web/mmusic/current/msg10678.html
> 
> > The aim of the draft is to support Trickle ICE for SIP user agent
> which does
> > not support RFC 3311 for last 11 years. AFAIK, different "to tag" is
> the
> > simpler way to solve the problem in UAS side. I have concern w.r.t
> INFO
> > mechanism as it adds one method for handling SDP and complicates the
> > existing SDP O/A handling (RFC 6337).
> 
> That's the thing, trickle ICE is not part of Offer/Answer. It's a
> separate channel for agent-to-agent communication. Use of SDP is just a
> convenient coincidence.
> 
> > Including Christer in this mail thread as he replied in the parallel
> mail
> > thread.
> 
> I believe he is on the list I don't think that's an issue :)
> 
> Cheers,
> Emil
> >
> > Thanks
> > Partha
> >
> >> -----Original Message-----
> >> From: Emil Ivov [mailto:emcho@jitsi.org]
> >> Sent: Thursday, July 25, 2013 2:19 AM
> >> To: Cullen Jennings (fluffy)
> >> Cc: Vijaya Mandava (vimandav); Parthasarathi R; Alan Johnston;
> MMUSIC
> >> IETF WG
> >> Subject: Re: [MMUSIC] Trickle ICE for SIP Questions
> >>
> >>
> >>
> >> On 24.07.13, 21:56, Cullen Jennings (fluffy) wrote:
> >>>
> >>> On Jul 24, 2013, at 12:22 PM, Vijaya Mandava (vimandav)
> >> <vimandav@cisco.com> wrote:
> >>>
> >>>> 180 with different to-tag would mean call is forked.
> >>>> If uac side do not support call forking, then we cannot use this
> 180
> >>>> response to collect trickled candidates.
> >>>
> >>> If it is a UAC, it supports this so I don't see the problem.
> >>
> >> One problem is that the UAC could support it to the extent of making
> it
> >> visible to the user:
> >>
> >>     "Your contact is being alerted on these N locations"
> >>
> >> And of course there's also the fact that we would be creating
> several
> >> dialogs per call, that we'd then need to clean and garbage collect.
> >>
> >> Why would we want to go there?
> >>
> >> Emil
> >>
> >> --
> >> https://jitsi.org
> >
> >
> 
> --
> https://jitsi.org
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From ari.keranen@ericsson.com  Mon Jul 29 00:53:51 2013
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Cc: Flemming Andreasen <fandreas@cisco.com>
Subject: [MMUSIC] New Liaison Statement, "LS on Missing IANA Registration Information on SDP Attributes in RFC 2326"
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MMUSIC,

We have received a liaison statement from 3GPP regarding missing SDP 
attributes at the IANA registry (see the e-mail below for details).

As the statement says:
> SA4 has identified that IANA registrations of three SDP attributes defined in RFC 2326, namely â€˜controlâ€™, â€˜rangeâ€™ and â€˜etagâ€™, are missing

As a solution, the RTSP 2.0 draft already registers 'control' and 
'range' for RFC 2326 and RFC 2326bis 
(http://tools.ietf.org/html/draft-ietf-mmusic-rfc2326bis-34#section-22.15). 
However, in RFC2326bis 'etag' is renamed into 'mtag'.

Hence, it seems that we should also register 'etag' but for that 
reference RTSP 1.0 (RFC 2326) only.

Please review the liaison statement and our proposed solution and send 
your comments to us and to the list.


Thanks,
Ari & Flemming
(MMUSIC co-chairs)

On 7/18/13 6:12 PM, Liaison Statement Management Tool wrote:
> Title: LS on Missing IANA Registration Information on SDP Attributes in RFC 2326
> Submission Date: 2013-07-16
> URL of the IETF Web page: http://datatracker.ietf.org/liaison/1267/
> Please reply by 2013-09-17
> From: 3GPP TSG SA WG4 (Susanna Kooistra <3GPPLiaison@etsi.org>)
> To: Multiparty Multimedia Session Control (Flemming Andreasen <fandreas@cisco.com>, Ari Keranen <ari.keranen@ericsson.com>)
> Cc: Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>,Richard Barnes <rlb@ipv.sx>,Multiparty Multimedia Session Control Discussion List <mmusic-request@ietf.org>
> Reponse Contact: Ozgur Oyman <ozgur.oyman@intel.com>
> Technical Contact:
> Purpose: For action
>
> Body:
> Attachments:
>
>      LS on Missing IANA Registration Information on SDP Attributes in RFC 2326
>      https://datatracker.ietf.org/documents/LIAISON/liaison-2013-07-16-3gpp-tsg-sa-wg4-mmusic-ls-on-missing-iana-registration-information-on-sdp-attributes-in-rfc-2326-attachment-1.doc
>


From ibc@aliax.net  Mon Jul 29 02:33:16 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 29 Jul 2013 11:32:05 +0200
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To: Emil Ivov <emcho@jitsi.org>
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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2013/7/28 Emil Ivov <emcho@jitsi.org>:
> This is a great example of how 3264 is not enough.
>
> Let's say for example that you want to add a new stream by switching an
> a=3Dsendonly "m=3D" line to a=3Dsendrecv. However, I consider this line t=
o be "on
> hold" and I don't want you thinking that you can unhold it, so I reject i=
t.
> At the same time I am totally fine with you adding a new one for your
> stream.
>
> Inversely, I might have sent you an offer with 8 "m=3D" lines as an indic=
ation
> that this is the maximum number of streams that I can deal with (this is
> actually labelled as bad practice in unified but I didn't find an
> alternative). In this case, I might be OK with you changing my directions=
 to
> enable new streams but I would reject new "m=3D" lines because I don't wa=
nt to
> have any more streams than the 8 already in the SDP.
>
> In both of the above cases there are different ways of achieving the same
> thing and each time one would work, while others wouldn't. Yet it would b=
e
> hard for the remote party to pick the right one.
>
>
> Paul just made a case for using different sets of "m=3D" lines for either
> direction. This is totally valid from a 3264 perspective, but without
> further information, an implementation would have to go through a number =
of
> trials and errors until it figures out that this is how you expect things=
 to
> happen.


The problem of SDP is not just the format or the O/A mandate, but the
fact that it was designed for legacy communications (i.e. single
telephone call) in which it is 100% obvious that both parties would
send audio to the other. It sometimes also works well when adding a
video stream (not so well when disabling and re-enabling it again).

So SDP assumes that a communication uses the same UDP IP:port tuple
for sending audio and receiving its corresponding audio, and thus, SDP
represents them in a single m=3Dline. Note however that in WebRTC we
have multiplexed RTP.


So now we have a powerfull scenario in which a conference server wants
to send the client (browser) 8 audio tracks (from 8 participants) and
the webcam video of 3 of them, and the client just wants to send its
own audio and video track. This should be EASY:

- The conference server offers 8 audio tracks + 3 video tracks.
- The browser offers 1 audio track + 1 video track.

But not... we are in SDP, things cannot be cool and modern. Instead:

- The conference server offers 8 audio tracks + 3 video tracks.
- The browser *answers* with 8 audio tracks + 3 videos tracks.
  (but most of them "a=3Drecvonly").

This is a hack. The 8 m=3Daudio lines offered by the server belong to 8
participants (from Alice, Bob, Carol...). Why should the browser
*choose* one of those m=3Daudio lines for sending its own audio track?
how can this stuff make sense in WebRTC where we have multiplexed RTP
flows? Why should the browser set the "Alice's audio track" for
sending and receiving and all the others for just receiving?


And *worse*:

- The conference server offers 8 audio tracks from 8 clients without webcam=
.
- The browser *wants* to offer 1 audio track +  1 video track (it does
have webcam).

FAIL !!! You cannot do that with SDP! The browser MUST reply with
exactly 8 m=3Daudio lines and ZERO m=3Dvideo lines, so if the browser
wants to add video it must re-negotiate later by adding a video track
(a new SDP O/A round trip).


So, leaving aside the SDP format and its O/A mandate, the fact is that
SDP semantics (even with Plan-Unified) is bad for modern
communications in which receiving a RTP flow does not mean that we
also send a RTP flow (and vice versa).





--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 29 Jul 2013 13:28:22 +0200
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Cc: MMUSIC IETF WG <mmusic@ietf.org>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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On Mon, Jul 29, 2013 at 11:32 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrot=
e:

>
> - The conference server offers 8 audio tracks from 8 clients without
> webcam.
> - The browser *wants* to offer 1 audio track +  1 video track (it does
> have webcam).
>
> FAIL !!! You cannot do that with SDP! The browser MUST reply with
> exactly 8 m=3Daudio lines and ZERO m=3Dvideo lines, so if the browser
> wants to add video it must re-negotiate later by adding a video track
> (a new SDP O/A round trip).
>

Obviously the conference server could be foolish in this way, but in
the type of scenario you suggest, the conference server would most
likely offer 8 sendonly audio tracks (for sending audio to the client)
and then one recvonly audio track and one recvonly video track
to elicit the client's a/v session, which will work reasonably well.

-Ekr

>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Jul 29, 2013 at 11:32 AM, I=F1aki Baz Castillo <span dir=3D=
"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net<=
/a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
- The conference server offers 8 audio tracks from 8 clients without webcam=
.<br>
- The browser *wants* to offer 1 audio track + =A01 video track (it does<br=
>
have webcam).<br>
<br>
FAIL !!! You cannot do that with SDP! The browser MUST reply with<br>
exactly 8 m=3Daudio lines and ZERO m=3Dvideo lines, so if the browser<br>
wants to add video it must re-negotiate later by adding a video track<br>
(a new SDP O/A round trip).<br></blockquote><div><br></div><div>Obviously t=
he conference server could be foolish in this way, but in</div><div>the typ=
e of scenario you suggest, the conference server would most</div><div>

likely offer 8 sendonly audio tracks (for sending audio to the client)</div=
><div>and then one recvonly audio track and one recvonly video track</div><=
div>to elicit the client&#39;s a/v session, which will work reasonably well=
.</div>

<div><br></div><div>-Ekr</div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br></blockquo=
te></div></div></div>

--001a11c20dfea4fac704e2a4ccbc--

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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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--
I=C3=B1aki Baz Castillo
<ibc@aliax.net>
El 29/07/2013 13:29, "Eric Rescorla" <ekr@rtfm.com> escribi=C3=B3:
>
>
>
>
> On Mon, Jul 29, 2013 at 11:32 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net>
wrote:
>>
>>
>> - The conference server offers 8 audio tracks from 8 clients without
webcam.
>> - The browser *wants* to offer 1 audio track +  1 video track (it does
>> have webcam).
>>
>> FAIL !!! You cannot do that with SDP! The browser MUST reply with
>> exactly 8 m=3Daudio lines and ZERO m=3Dvideo lines, so if the browser
>> wants to add video it must re-negotiate later by adding a video track
>> (a new SDP O/A round trip).
>
>
> Obviously the conference server could be foolish in this way, but in
> the type of scenario you suggest, the conference server would most
> likely offer 8 sendonly audio tracks (for sending audio to the client)
> and then one recvonly audio track and one recvonly video track
> to elicit the client's a/v session, which will work reasonably well.

And what about if the browser/device has two webcams (letf and right) and
wants to send both? How can the server determine it? Or should the device
does re-negotiation?

I am pretty sure SDP semantics is bad for modern RTC communications like
those we expect to take place in WebRTC.

--0015175df19ee5669904e2a55c54
Content-Type: text/html; charset=UTF-8
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<p dir=3D"ltr"></p>
<p dir=3D"ltr">--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
El 29/07/2013 13:29, &quot;Eric Rescorla&quot; &lt;<a href=3D"mailto:ekr@rt=
fm.com">ekr@rtfm.com</a>&gt; escribi=C3=B3:<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On Mon, Jul 29, 2013 at 11:32 AM, I=C3=B1aki Baz Castillo &lt;<a href=
=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; - The conference server offers 8 audio tracks from 8 clients witho=
ut webcam.<br>
&gt;&gt; - The browser *wants* to offer 1 audio track + =C2=A01 video track=
 (it does<br>
&gt;&gt; have webcam).<br>
&gt;&gt;<br>
&gt;&gt; FAIL !!! You cannot do that with SDP! The browser MUST reply with<=
br>
&gt;&gt; exactly 8 m=3Daudio lines and ZERO m=3Dvideo lines, so if the brow=
ser<br>
&gt;&gt; wants to add video it must re-negotiate later by adding a video tr=
ack<br>
&gt;&gt; (a new SDP O/A round trip).<br>
&gt;<br>
&gt;<br>
&gt; Obviously the conference server could be foolish in this way, but in<b=
r>
&gt; the type of scenario you suggest, the conference server would most<br>
&gt; likely offer 8 sendonly audio tracks (for sending audio to the client)=
<br>
&gt; and then one recvonly audio track and one recvonly video track<br>
&gt; to elicit the client&#39;s a/v session, which will work reasonably wel=
l.<br></p>
<p dir=3D"ltr">And what about if the browser/device has two webcams (letf a=
nd right) and wants to send both? How can the server determine it? Or shoul=
d the device does re-negotiation?</p>
<p dir=3D"ltr">I am pretty sure SDP semantics is bad for modern RTC communi=
cations like those we expect to take place in WebRTC.<br>
</p>

--0015175df19ee5669904e2a55c54--

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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 29 Jul 2013 14:45:05 +0200
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: MMUSIC IETF WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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2013/7/29 Christer Holmberg <christer.holmberg@ericsson.com>:
> No matter what protocol you use, how can the server know what the browser
> =E2=80=9Cwants to do=E2=80=9D before the server has even contacted the br=
owser???

Exactly !!!

That is the point Christer. The server (the offerer in this case) MAY
NOT know what the browser/client wants to offer (unless there is some
kind of announcement via previous registration or there is a hardcoded
scenario which well-known/proprietary devices).

So, if we assume that the "session offerer" (the server in this case)
MAY NOT be able to know which kind of media and/or how many tracks the
browser would like to send, why do we mandate a media signaling
protocol (SDP) in which the offerer MUST tell the browser/client how
many tracks (and which kind of media) it can send??

SDP is the problem here.

A good design would be someone in which the offerer (server) tells the
browser "I will send you 8 audio tracks and 3 videos tracks" and the
browser would reply "OK, I will send you 1 audio track and 2 video
tracks". Or a more complete one:

- The offerer tells the browser "I will send you 8 audio tracks and 3
videos tracks".

- The browser confirms all the audio tracks but tells the server to
disable 2 video tracks".

- The browser tells the server "I will send you 1 audio track and 2
video tracks".

- The server accepts receiving all of them.


This is: an unidirectional-track-negotiation media protocol. But that
is not what SDP is, so here we are.


Regards.




--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>
Thread-Topic: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 29 Jul 2013 16:32:10 +0200
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Cc: MMUSIC IETF WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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2013/7/29 Christer Holmberg <christer.holmberg@ericsson.com>:
>> Or a more complete one:
>>
>> - The offerer tells the browser "I will send you 8 audio tracks and 3 vi=
deos tracks".
>>
>> - The browser confirms all the audio tracks but tells the server to disa=
ble 2 video tracks".
>>
>> - The browser tells the server "I will send you 1 audio track and 2 vide=
o tracks".
>>
>> - The server accepts receiving all of them.
>>
>> This is: an unidirectional-track-negotiation media protocol. But that is=
 not what SDP is, so here we are.
>
> You CAN do this with SDP, using uni-directional streams: the server offer=
 8+3 sendonly streams, and then the browser offers 1-3 sendonly streams.
>
> (I DO agree it is not how SDP has traditionally been used)

And sure you agree that WebRTC browsers and devices will NOT allow
that because, regardless how "flexible" SDP is, the fact is that
devices do not allow all that flexibility.

So my concerns (and Emil's ones) remain alive. But, for simplifying, I
will summarize them in a single question:

- Alice, using her browser, can enter in an *active* audio/video
conference in which there are already 8 participants.

- The browser sends the SDP offer to the conference server.

- Should the browser *know* that there are 8 participants (so it
should add 8 m=3Daudio and 8 m=3Dvideo lines)?

- Or should it just add a single audio and video m lines? If so, could
the server reply a SDP answer with 9 audio lines and 9 video lines?
really? I am 100% sure that browsers will not understand it.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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On 7/29/13 4:32 PM, IÃ±aki Baz Castillo wrote:
> 2013/7/29 Christer Holmberg <christer.holmberg@ericsson.com>:
>>> Or a more complete one:
>>>
>>> - The offerer tells the browser "I will send you 8 audio tracks and 3 videos tracks".
>>>
>>> - The browser confirms all the audio tracks but tells the server to disable 2 video tracks".
>>>
>>> - The browser tells the server "I will send you 1 audio track and 2 video tracks".
>>>
>>> - The server accepts receiving all of them.
>>>
>>> This is: an unidirectional-track-negotiation media protocol. But that is not what SDP is, so here we are.
>>
>> You CAN do this with SDP, using uni-directional streams: the server offer 8+3 sendonly streams, and then the browser offers 1-3 sendonly streams.
>>
>> (I DO agree it is not how SDP has traditionally been used)
>
> And sure you agree that WebRTC browsers and devices will NOT allow
> that because, regardless how "flexible" SDP is, the fact is that
> devices do not allow all that flexibility.
>
> So my concerns (and Emil's ones) remain alive. But, for simplifying, I
> will summarize them in a single question:
>
> - Alice, using her browser, can enter in an *active* audio/video
> conference in which there are already 8 participants.
>
> - The browser sends the SDP offer to the conference server.
>
> - Should the browser *know* that there are 8 participants (so it
> should add 8 m=audio and 8 m=video lines)?

Probably not. Maybe in some cases it can. (Because the web server told it.)

> - Or should it just add a single audio and video m lines?

This is certainly the straightforward thing to do.

> If so, could
> the server reply a SDP answer with 9 audio lines and 9 video lines?
> really? I am 100% sure that browsers will not understand it.

No, this is not valid SDP. The best that can be done is for the server 
to answer the two m-lines. And then to send another offer that adds the 
other m-lines.

This is not ideal. But migrating away from SDP for sip has been tried 
and given up as infeasible.

Conceivably something different can be done for rtcweb if it wants to 
diverge from SDP and/or O/A. But if so, then the gatewaying to sip will 
be harder.

Perhaps we should reconstitute SDPng for rtcweb. :-)

	Thanks,
	Paul


From christer.holmberg@ericsson.com  Mon Jul 29 08:03:52 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
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Thread-Topic: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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2013/7/29 Christer Holmberg <christer.holmberg@ericsson.com>:
>> So my concerns (and Emil's ones) remain alive. But, for simplifying, I w=
ill summarize them in a single question:
>>
>> - Alice, using her browser, can enter in an *active* audio/video confere=
nce in which there are already 8 participants.
>>
>> - The browser sends the SDP offer to the conference server.
>>
>> - Should the browser *know* that there are 8 participants (so it should =
add 8 m=3Daudio and 8 m=3Dvideo lines)?
>>
>> - Or should it just add a single audio and video m lines? If so, could t=
he server reply a SDP answer with 9 audio lines and 9 video lines?
>> really? I am 100% sure that browsers will not understand it.
>
> The browser can, in the first Offer, only include 1+1 m- lines. Then, the=
 server can inform the browser about the number of participants,

Via custom signaling, right? so we have a media session signaling
mandate (SDP O/A) which is not suitable for properly negotiating my
very basic scenario.


> and the browser can create a second offer accordingly, e.g. with 8+8 m- l=
ines.
>
> (...or, the number of participants might be provided to the browser befor=
e it sends the first offer.)

Again via custom signaling.



> I think one problem is that people often are so "fixated" in doing everyt=
hing in a single O/A transaction. During the session establishment, it does=
n't really matter if you have to send one or two more Offers :)

May be that si good enough for SIP, but may be not for other (future)
RTC protocols.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Thread-Topic: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 29 Jul 2013 19:01:03 +0200
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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2013/7/29 Christer Holmberg <christer.holmberg@ericsson.com>:
>>> The browser can, in the first Offer, only include 1+1 m- lines. Then,
>>> the server can inform the browser about the number of participants,
>>
>> Via custom signaling, right? so we have a media session signaling mandat=
e (SDP O/A) which is not suitable for properly negotiating my very basic sc=
enario.
>
> SDP O/A is used to negotiate the media channels - not to exchange generic=
 capabilities (e.g. number of participants in the conference).

I was thinking in the "standard video conference server" (purchased by
so many web providers) Cullen was talking about in other mail thread.

Anyhow I don't know why the browser must know the exact number of
participants in a conference before making its SDP offer and calling
to the conference server. It just does not make sense. Nobody will
comment it?



- I'm Alice calling to the conference server. This is my audio and
video track (2 m lines).

- OK Alice, I'm the server. Take those 8 audio tracks and 3 video
tracks from other participants already in the conference (so 11 m
lines).

Why something like this is not possible with SDP? Well, because SDP
was designed assuming bidirectional tracks (single telephony audio
track in which obviously both sides talk) and thus SDP is not good for
WebRTC in which we expect servers providing many tracks to clients.





--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Date: Mon, 29 Jul 2013 22:53:46 -0700
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Subject: Re: [MMUSIC] SDP for conferencing scenario?
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> - Web application with a SIP over WebSocket client running in the web.
>=20
> - The web user is provided with a conference SIP URI in which there
> are *already* 8 participants (5 of them emitting audio and video and 3
> just emitting audio).
>=20
> - The user calls=2C from his webphone=2C to the given URI to join the con=
ference.
>=20
> Let's imagine that the JS app knows the number of participant in the conf=
erence.
> Let's imagine my browser have mic and webcam.
>=20
> QUESTION:
>=20
> How can my browser join the conference without requiring SDP
> renegotiation from the server and=2C at the same time=2C being able to
> send audio/video and receive audio/video from others (different tracks
> / m=3Dlines)?

I=F1aki --
=20
Is the term "conference" used here in the same way as in RFC 4353/4575/5239=
?  Or are you envisaging a set of peer-to-peer conversations? =20
 		 	   		  =

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--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'><br>&gt=3B - Web application wit=
h a SIP over WebSocket client running in the web.<br>&gt=3B <br>&gt=3B - Th=
e web user is provided with a conference SIP URI in which there<br>&gt=3B a=
re *already* 8 participants (5 of them emitting audio and video and 3<br>&g=
t=3B just emitting audio).<br>&gt=3B <br>&gt=3B - The user calls=2C from hi=
s webphone=2C to the given URI to join the conference.<br>&gt=3B <br>&gt=3B=
 Let's imagine that the JS app knows the number of participant in the confe=
rence.<br>&gt=3B Let's imagine my browser have mic and webcam.<br>&gt=3B <b=
r>&gt=3B QUESTION:<br>&gt=3B <br>&gt=3B How can my browser join the confere=
nce without requiring SDP<br>&gt=3B renegotiation from the server and=2C at=
 the same time=2C being able to<br>&gt=3B send audio/video and receive audi=
o/video from others (different tracks<br>&gt=3B / m=3Dlines)?<br><BR>I=F1ak=
i --<BR>&nbsp=3B<BR>Is the term "conference" used here in the same way as i=
n RFC 4353/4575/5239?&nbsp=3B Or are you envisaging a set of peer-to-peer c=
onversations?&nbsp=3B <BR> 		 	   		  </div></body>
</html>=

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From ibc@aliax.net  Tue Jul 30 00:40:53 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 30 Jul 2013 09:40:24 +0200
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Subject: Re: [MMUSIC] SDP for conferencing scenario?
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2013/7/30 Bernard Aboba <bernard_aboba@hotmail.com>:
>> - Web application with a SIP over WebSocket client running in the web.
>>
>> - The web user is provided with a conference SIP URI in which there
>> are *already* 8 participants (5 of them emitting audio and video and 3
>> just emitting audio).
>>
>> - The user calls, from his webphone, to the given URI to join the
>> conference.
>>
>> Let's imagine that the JS app knows the number of participant in the
>> conference.
>> Let's imagine my browser have mic and webcam.
>>
>> QUESTION:
>>
>> How can my browser join the conference without requiring SDP
>> renegotiation from the server and, at the same time, being able to
>> send audio/video and receive audio/video from others (different tracks
>> / m=3Dlines)?
>

> I=C3=B1aki
>
> Is the term "conference" used here in the same way as in RFC 4353/4575/52=
39?
> Or are you envisaging a set of peer-to-peer conversations?

Not exaclty. It may be but it is not required. Let's imagine I don't
use SIP but any other sig protocol (but still can be SIP). I just want
to call from my browser to a conference server in which there is
already an active conference with N participants. I want to be able to
offer my audio/video tracks and the server to accept them by also
telling me its N audio/video tracks. Without SDP renegotiation.

Is it possible? No, it is not, due to SDP semantics.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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> I just want to call from my browser to a conference server in which there=
 is
> already an active conference with N participants. I want to be able to
> offer my audio/video tracks and the server to accept them by also
> telling me its N audio/video tracks. Without SDP renegotiation.

[BA]  Putting the SDP API issues aside for the moment=2C  I can imagine an =
on-the-wire SDP offer/answer exchange which could address your scenario.  F=
or example=2C one could envisage a "Plan B"-ish approach where we have an S=
DP offer with two m lines (one for audio=2C one for video) and a max-recv-s=
ource attribute=2C and an answer also with two m-lines.  In a "mixer" topol=
ogy where the SSRCs are known=2C a=3Dssrc attributes could be included.  Fo=
r an "RTP translator" topology as described in "No Plan" the SSRCs might no=
t be known=2C so they couldn't be included.=20
=20
Note that even if you don't want SDP in-the-API=2C you'd still need to addr=
ess the RTP issues=2C such as which RTP topologies are to be supported.   S=
ince interoperability among video conferencing systems is less than univers=
al=2C these issues are non-trivial.=20
=20
=20
=20

=20
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>&gt=3B I just want to call from =
my browser to a conference server in which there is<br>&gt=3B already an ac=
tive conference with N participants. I want to be able to<br>&gt=3B offer m=
y audio/video tracks and the server to accept them by also<br>&gt=3B tellin=
g me its N audio/video tracks. Without SDP renegotiation.<br><BR>[BA]&nbsp=
=3B Putting the SDP API issues aside for the moment=2C&nbsp=3B I can imagin=
e an on-the-wire SDP offer/answer exchange which could address your scenari=
o.&nbsp=3B&nbsp=3BFor example=2C one&nbsp=3Bcould envisage a "Plan B"-ish&n=
bsp=3Bapproach where we have an&nbsp=3BSDP offer with two m lines (one for =
audio=2C one for video) and a max-recv-source attribute=2C and an answer al=
so with two m-lines.&nbsp=3B In a "mixer" topology where the SSRCs are know=
n=2C a=3Dssrc attributes could be included.&nbsp=3B For an "RTP translator"=
 topology as described in "No Plan" the SSRCs might not be known=2C so they=
 couldn't be included. <BR>&nbsp=3B<BR>Note that even if you don't want SDP=
 in-the-API=2C you'd still need to address the RTP issues=2C such as which =
RTP topologies are to be supported.&nbsp=3B&nbsp=3B Since interoperability =
among video conferencing systems is less than universal=2C these issues are=
 non-trivial. <BR>&nbsp=3B<BR>&nbsp=3B<BR>&nbsp=3B<BR><br>&nbsp=3B<BR> 		 	=
   		  </div></body>
</html>=

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From harald@alvestrand.no  Tue Jul 30 01:59:14 2013
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Subject: Re: [MMUSIC] [rtcweb] About unified Plan and multiple m lines
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Inaki, this is abuse of mailing lists.

Raising the issue of "how do I do this with the current proposal made to
MMUSIC" (aka "unified plan") is reasonable. You have been answered;
there are multiple ways that are valid SDP, and it is reasonable to have
a discussion on which ones to recommand, and if any of them should be
recommended against.

Your screed about how the fact that there's work to be done here
"proves" that SDP is unsuitable for RTCWEB belongs in the WEBRTC mailing
list.

On 07/29/2013 07:01 PM, IÃ±aki Baz Castillo wrote:
> 2013/7/29 Christer Holmberg <christer.holmberg@ericsson.com>:
>>>> The browser can, in the first Offer, only include 1+1 m- lines. Then,
>>>> the server can inform the browser about the number of participants,
>>> Via custom signaling, right? so we have a media session signaling mandate (SDP O/A) which is not suitable for properly negotiating my very basic scenario.
>> SDP O/A is used to negotiate the media channels - not to exchange generic capabilities (e.g. number of participants in the conference).
> I was thinking in the "standard video conference server" (purchased by
> so many web providers) Cullen was talking about in other mail thread.
>
> Anyhow I don't know why the browser must know the exact number of
> participants in a conference before making its SDP offer and calling
> to the conference server. It just does not make sense. Nobody will
> comment it?
>
>
>
> - I'm Alice calling to the conference server. This is my audio and
> video track (2 m lines).
>
> - OK Alice, I'm the server. Take those 8 audio tracks and 3 video
> tracks from other participants already in the conference (so 11 m
> lines).
>
> Why something like this is not possible with SDP? Well, because SDP
> was designed assuming bidirectional tracks (single telephony audio
> track in which obviously both sides talk) and thus SDP is not good for
> WebRTC in which we expect servers providing many tracks to clients.
>
>
>
>
>


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 30 Jul 2013 11:27:19 +0200
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Subject: Re: [MMUSIC] SDP for conferencing scenario?
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2013/7/30 Bernard Aboba <bernard_aboba@hotmail.com>:
> Putting the SDP API issues aside for the moment,  I can imagine an on-the=
-wire SDP offer/answer exchange which could address your scenario.

Of course, this is a SDP problem/limitation, a SDP problem which
constrains the design of WebRTC applications.




--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From bernard_aboba@hotmail.com  Tue Jul 30 04:30:56 2013
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From: Bernard Aboba <bernard_aboba@hotmail.com>
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Subject: [MMUSIC] Question on Section 4.7 of Unified Plan
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I have a question on the SDP in Section 4.7 of Unified Plan.   Looking at t=
he SDP=2C I wasn't sure if this was attempting to illustrate a combination =
of simulcast and layering=2C or just layering.  Also=2C it wasn't clear to =
me which sources corresponded to which layers. =20

   m=3Dvideo 0 RTP/SAVPF 96 97 100
   a=3Dmsid:ma tb
   a=3Dextmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 60725
   b=3DAS:2500
   a=3Dmid:m1
   a=3Drtpmap:96 H264/90000
   a=3Dfmtp:96 max-fr=3D30=3Bmax-fs=3D8040
[BA] Is this intended to be a simulcast or layered stream?  If the former=
=2C should it be H264-SVC?=20
   a=3Drtpmap:97 H264/90000
   a=3Dfmtp:97 max-fr=3D15=3Bmax-fs=3D1200
   a=3Drtpmap:100 H264-SVC/90000
   a=3Dfmtp:100 max-fr=3D30=3Bmax-fs=3D8040
   a=3Ddepend:100 lay m1:96=2C97=3B
   a=3Dssrc:48970
   a=3Dssrc:90898
   a=3Dssrc:66997
[BA] What is the intended mapping between layers and sources?=20
   a=3Dsendrecv
   a=3Drtcp-mux
   a=3Dbundle-only

 		 	   		  =

--_fdab3d2f-0c36-44b6-bd01-67d70156f6c4_
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<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'>I have&nbsp=3Ba question on the =
SDP in Section 4.7 of Unified Plan.&nbsp=3B&nbsp=3B&nbsp=3BLooking at the S=
DP=2C I wasn't sure if this was attempting to illustrate a combination of s=
imulcast and layering=2C or just layering.&nbsp=3B Also=2C it wasn't clear =
to me which sources corresponded to which layers.&nbsp=3B <BR><br>&nbsp=3B&=
nbsp=3B m=3Dvideo 0 RTP/SAVPF 96 97 100<br>&nbsp=3B&nbsp=3B a=3Dmsid:ma tb<=
br>&nbsp=3B&nbsp=3B a=3Dextmap:1 urn:ietf:params:rtp-hdrext:stream-correlat=
or 60725<br>&nbsp=3B&nbsp=3B b=3DAS:2500<br>&nbsp=3B&nbsp=3B a=3Dmid:m1<br>=
&nbsp=3B&nbsp=3B a=3Drtpmap:96 H264/90000<br>&nbsp=3B&nbsp=3B a=3Dfmtp:96 m=
ax-fr=3D30=3Bmax-fs=3D8040<BR>[BA] Is this&nbsp=3Bintended to be&nbsp=3Ba s=
imulcast or layered stream?&nbsp=3B&nbsp=3BIf the former=2C should it be H2=
64-SVC? <br>&nbsp=3B&nbsp=3B a=3Drtpmap:97 H264/90000<br>&nbsp=3B&nbsp=3B a=
=3Dfmtp:97 max-fr=3D15=3Bmax-fs=3D1200<br>&nbsp=3B&nbsp=3B a=3Drtpmap:100 H=
264-SVC/90000<br>&nbsp=3B&nbsp=3B a=3Dfmtp:100 max-fr=3D30=3Bmax-fs=3D8040<=
br>&nbsp=3B&nbsp=3B a=3Ddepend:100 lay m1:96=2C97=3B<br>&nbsp=3B&nbsp=3B a=
=3Dssrc:48970<br>&nbsp=3B&nbsp=3B a=3Dssrc:90898<br>&nbsp=3B&nbsp=3B a=3Dss=
rc:66997<BR>[BA]&nbsp=3BWhat is the intended mapping between layers and sou=
rces? <br>&nbsp=3B&nbsp=3B a=3Dsendrecv<br>&nbsp=3B&nbsp=3B a=3Drtcp-mux<br=
>&nbsp=3B&nbsp=3B a=3Dbundle-only<br><BR> 		 	   		  </div></body>
</html>=

--_fdab3d2f-0c36-44b6-bd01-67d70156f6c4_--

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To: "mmusic@ietf.org" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "dispatch@ietf.org" <dispatch@ietf.org>
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To: Bernard Aboba <bernard_aboba@hotmail.com>
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Subject: Re: [MMUSIC] Question on Section 4.7 of Unified Plan
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Bernard, I think there are a few typos in this one as it got cut and =
pasted around. I suspect it should be something more like below =85


On Jul 30, 2013, at 1:30 PM, Bernard Aboba <bernard_aboba@hotmail.com> =
wrote:

> I have a question on the SDP in Section 4.7 of Unified Plan.   Looking =
at the SDP, I wasn't sure if this was attempting to illustrate a =
combination of simulcast and layering, or just layering.  Also, it =
wasn't clear to me which sources corresponded to which layers. =20
>=20
>    m=3Dvideo 0 RTP/SAVPF 96 97 100
>    a=3Dmsid:ma tb
>    a=3Dextmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 60725
>    b=3DAS:2500
>    a=3Dmid:m1
>    a=3Drtpmap:96 H264/90000


>    a=3Dfmtp:96 max-fr=3D30;max-fs=3D8040
> [BA] Is this intended to be a simulcast or layered stream?  If the =
former, should it be H264-SVC?=20
>    a=3Drtpmap:97 H264/90000

Yes, seems to me that should be H264-SVC


>    a=3Dfmtp:97 max-fr=3D15;max-fs=3D1200
>    a=3Drtpmap:100 H264-SVC/90000
>    a=3Dfmtp:100 max-fr=3D30;max-fs=3D8040
>    a=3Ddepend:100 lay m1:96,97;

and I suspect the above line should be "a=3Ddepend:100 lay m1:96"


>    a=3Dssrc:48970
>    a=3Dssrc:90898
>    a=3Dssrc:66997
> [BA] What is the intended mapping between layers and sources?=20
>    a=3Dsendrecv
>    a=3Drtcp-mux
>    a=3Dbundle-only
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic


From martin.thomson@gmail.com  Wed Jul 31 00:22:15 2013
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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http://tools.ietf.org/html/rfc5245#section-4.1.2.1

After talking to Jonathan, several things have become apparent.

The prioritization calculation described in Section 4.1.2.1 of RFC
5245 is not mandatory, it's SHOULD only.  This means that the guidance
regarding prioritization of IPv6 over IPv4 is not applied as strictly
and blindly as implied by the happy eyeballs draft.  Implementations
can choose a different prioritization.  (I was under the
misapprehension that only part of the priority was signaled, but
that's clearly not the case.)

That implies to me that the happy-eyeballs draft is, as best, only
worth pursuing as an information addendum that provides guidance on
non-stupid selection of priorities for dual-stack hosts.

However, given the existence of an RFC5245bis effort, I am going to
suggest that the guidance provided instead be folded into a revised
Section 4.1.2.1.

Another problem that can be addressed at the same time is a removal of
all the normative language that surrounds this optional feature.  The
current 4.1.2.1 has lots of "MUST" statements that are both
unnecessary and confusing.

--Martin

From bernard_aboba@hotmail.com  Wed Jul 31 00:31:48 2013
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From: Bernard Aboba <bernard_aboba@hotmail.com>
To: Martin Thomson <martin.thomson@gmail.com>, "mmusic@ietf.org" <mmusic@ietf.org>
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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I agree that it doesn't make sense to pursue the happy-eyeballs draft and t=
hat folding fixes into RFC 5245bis is the best approach.=20
=20
A number of ICE implementations do not follow the priority guidance in RFC =
5245 Section 4.1.2.1 because it doesn't work=2C and also because guidelines=
 such as DoD 2008 UCR Change 3 require that implementations be able to prio=
ritize IPv4 over IPv6.   So in practice=2C the advice in RFC 5245 Section 4=
.1.2.1 has long been overtaken by events.=20
=20
> http://tools.ietf.org/html/rfc5245#section-4.1.2.1
>=20
> After talking to Jonathan=2C several things have become apparent.
>=20
> The prioritization calculation described in Section 4.1.2.1 of RFC
> 5245 is not mandatory=2C it's SHOULD only.  This means that the guidance
> regarding prioritization of IPv6 over IPv4 is not applied as strictly
> and blindly as implied by the happy eyeballs draft.  Implementations
> can choose a different prioritization.  (I was under the
> misapprehension that only part of the priority was signaled=2C but
> that's clearly not the case.)
>=20
> That implies to me that the happy-eyeballs draft is=2C as best=2C only
> worth pursuing as an information addendum that provides guidance on
> non-stupid selection of priorities for dual-stack hosts.
>=20
> However=2C given the existence of an RFC5245bis effort=2C I am going to
> suggest that the guidance provided instead be folded into a revised
> Section 4.1.2.1.
>=20
> Another problem that can be addressed at the same time is a removal of
> all the normative language that surrounds this optional feature.  The
> current 4.1.2.1 has lots of "MUST" statements that are both
> unnecessary and confusing.
>=20
> --Martin
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
 		 	   		  =

--_d3c1abc0-a21c-46e5-bba4-086d4c9efe8a_
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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 12pt=3B
font-family:Calibri
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--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>I agree that it doesn't make sen=
se to pursue the happy-eyeballs draft and that folding fixes into RFC 5245b=
is is the best approach.&nbsp=3B<BR>&nbsp=3B<BR>A number of ICE implementat=
ions&nbsp=3Bdo not follow the&nbsp=3Bpriority guidance in RFC&nbsp=3B5245 S=
ection 4.1.2.1 because it doesn't&nbsp=3Bwork=2C and also because guideline=
s such as DoD 2008 UCR Change 3 require that implementations be able to pri=
oritize IPv4 over IPv6.&nbsp=3B&nbsp=3B So in practice=2C the advice in RFC=
 5245 Section 4.1.2.1 has&nbsp=3Blong been overtaken by events. <br>&nbsp=
=3B<BR><div>&gt=3B http://tools.ietf.org/html/rfc5245#section-4.1.2.1<br>&g=
t=3B <br>&gt=3B After talking to Jonathan=2C several things have become app=
arent.<br>&gt=3B <br>&gt=3B The prioritization calculation described in Sec=
tion 4.1.2.1 of RFC<br>&gt=3B 5245 is not mandatory=2C it's SHOULD only.  T=
his means that the guidance<br>&gt=3B regarding prioritization of IPv6 over=
 IPv4 is not applied as strictly<br>&gt=3B and blindly as implied by the ha=
ppy eyeballs draft.  Implementations<br>&gt=3B can choose a different prior=
itization.  (I was under the<br>&gt=3B misapprehension that only part of th=
e priority was signaled=2C but<br>&gt=3B that's clearly not the case.)<br>&=
gt=3B <br>&gt=3B That implies to me that the happy-eyeballs draft is=2C as =
best=2C only<br>&gt=3B worth pursuing as an information addendum that provi=
des guidance on<br>&gt=3B non-stupid selection of priorities for dual-stack=
 hosts.<br>&gt=3B <br>&gt=3B However=2C given the existence of an RFC5245bi=
s effort=2C I am going to<br>&gt=3B suggest that the guidance provided inst=
ead be folded into a revised<br>&gt=3B Section 4.1.2.1.<br>&gt=3B <br>&gt=
=3B Another problem that can be addressed at the same time is a removal of<=
br>&gt=3B all the normative language that surrounds this optional feature. =
 The<br>&gt=3B current 4.1.2.1 has lots of "MUST" statements that are both<=
br>&gt=3B unnecessary and confusing.<br>&gt=3B <br>&gt=3B --Martin<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>=

--_d3c1abc0-a21c-46e5-bba4-086d4c9efe8a_--

From ari.keranen@ericsson.com  Wed Jul 31 02:23:41 2013
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Cc: "Tirumaleswar Reddy \(tireddy\)" <tireddy@cisco.com>
Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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Hi,

If we can do this as updated recommendations on how to prioritize the 
candidates, then I think it makes a lot of sense to integrate this into 
5245bis. If it gets much more complicated than this, probably separate 
document makes more sense.

Tiru mentioned that simply tweaking the priorities may not work in all 
cases, so perhaps we should have a look at those cases and see if we can 
make them work.


Cheers,
Ari (as individual)

On 7/31/13 9:31 AM, Bernard Aboba wrote:
> I agree that it doesn't make sense to pursue the happy-eyeballs draft
> and that folding fixes into RFC 5245bis is the best approach.
>
> A number of ICE implementations do not follow the priority guidance in
> RFC 5245 Section 4.1.2.1 because it doesn't work, and also because
> guidelines such as DoD 2008 UCR Change 3 require that implementations be
> able to prioritize IPv4 over IPv6.   So in practice, the advice in RFC
> 5245 Section 4.1.2.1 has long been overtaken by events.
>
>  > http://tools.ietf.org/html/rfc5245#section-4.1.2.1
>  >
>  > After talking to Jonathan, several things have become apparent.
>  >
>  > The prioritization calculation described in Section 4.1.2.1 of RFC
>  > 5245 is not mandatory, it's SHOULD only. This means that the guidance
>  > regarding prioritization of IPv6 over IPv4 is not applied as strictly
>  > and blindly as implied by the happy eyeballs draft. Implementations
>  > can choose a different prioritization. (I was under the
>  > misapprehension that only part of the priority was signaled, but
>  > that's clearly not the case.)
>  >
>  > That implies to me that the happy-eyeballs draft is, as best, only
>  > worth pursuing as an information addendum that provides guidance on
>  > non-stupid selection of priorities for dual-stack hosts.
>  >
>  > However, given the existence of an RFC5245bis effort, I am going to
>  > suggest that the guidance provided instead be folded into a revised
>  > Section 4.1.2.1.
>  >
>  > Another problem that can be addressed at the same time is a removal of
>  > all the normative language that surrounds this optional feature. The
>  > current 4.1.2.1 has lots of "MUST" statements that are both
>  > unnecessary and confusing.
>  >
>  > --Martin
>  > _______________________________________________
>  > mmusic mailing list
>  > mmusic@ietf.org
>  > https://www.ietf.org/mailman/listinfo/mmusic
>
>
> _______________________________________________
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From martin.thomson@gmail.com  Wed Jul 31 02:25:46 2013
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On 31 July 2013 11:19, Ari Ker=C3=A4nen <ari.keranen@ericsson.com> wrote:
> Tiru mentioned that simply tweaking the priorities may not work in all
> cases, so perhaps we should have a look at those cases and see if we can
> make them work.

I don't believe that to be a problem.  Obviously, unless both peers
are doing the same sort of prioritization, you can get some silly
results, but that's always going to be a hazard.

I don't believe that we need any additional recommendations.

From dwing@cisco.com  Wed Jul 31 04:19:42 2013
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On Jul 31, 2013, at 11:25 AM, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> On 31 July 2013 11:19, Ari Ker=E4nen <ari.keranen@ericsson.com> wrote:
>> Tiru mentioned that simply tweaking the priorities may not work in =
all
>> cases, so perhaps we should have a look at those cases and see if we =
can
>> make them work.
>=20
> I don't believe that to be a problem.  Obviously, unless both peers
> are doing the same sort of prioritization, you can get some silly
> results, but that's always going to be a hazard.
>=20
> I don't believe that we need any additional recommendations.

I agree the happy eyeballs recommendation (or, "non-stupid selection of =
priorities for dual-stack hosts") would best be folded into RFC5245bis.

However, about a year ago Jonathan Lennox had expressed some concern =
that if the two peers have different ordering, their suicide packets =
(necessary to create mappings in their own respective NATs) would have =
to suffer an additional retry.  I guess that is okay, and why you said =
it isn't a problem.  I'm fine with that, if that's what you meant.  ICE =
has its SHOULD for the ordering and chooses one end to be authoritative =
for the ordering to avoid the problem of each peer using a different =
order.  If we're confident that guidance in ICE is over-stated and =
unnecessary, that certainly makes things easier that the two ICE peers =
can choose their own ordering willy-nilly.

-d



From jonathan@vidyo.com  Wed Jul 31 04:51:18 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: "mmusic@ietf.org" <mmusic@ietf.org>
Date: Wed, 31 Jul 2013 07:51:03 -0400
Thread-Topic: Proposed definition for "same codec configuration"
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Subject: [MMUSIC] Proposed definition for "same codec configuration"
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As I mentioned at the mic today, my straw-man proposed definition for "same=
 codec configuration" is:

Two codec configurations are the same if the PT has the same value for the =
"rtpmap" and "fmtp" SDP parameters.


Known issues with this definition:

Some codecs (notably H.264's sprop-* parameters, and all of Opus's paramete=
rs) have fmtp parameters describe behavior of a sender, or else are non-nor=
mative indications about the modes that would be most useful but aren't act=
ually necessary to configure a decoder.  So it's overly restrictive to insi=
st that fmtp be identical between all the lines.

Getting this to work would, I think, require analysis in the style of draft=
-nandakumar-mmusic-sdp-mux-attributes for every fmtp parameter of every cod=
ec anyone cares about.

The other issue is whether there are other SDP parameters (presumably ones =
that are specified in terms of PT) that also need to be considered to be pa=
rt of the "codec configuration" and needs to be aligned.  Imageattr was pro=
posed as one possibility.
--
Jonathan Lennox
jonathan@vidyo.com



From martin.thomson@gmail.com  Wed Jul 31 04:51:38 2013
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On 31 July 2013 13:19, Dan Wing <dwing@cisco.com> wrote:
> I agree the happy eyeballs recommendation (or, "non-stupid selection of p=
riorities for dual-stack hosts") would best be folded into RFC5245bis.

Great.

> However, about a year ago Jonathan Lennox had expressed some concern that=
 if the two peers have different ordering, their suicide packets (necessary=
 to create mappings in their own respective NATs) would have to suffer an a=
dditional retry.  I guess that is okay, and why you said it isn't a problem=
.  I'm fine with that, if that's what you meant.  ICE has its SHOULD for th=
e ordering and chooses one end to be authoritative for the ordering to avoi=
d the problem of each peer using a different order.  If we're confident tha=
t guidance in ICE is over-stated and unnecessary, that certainly makes thin=
gs easier that the two ICE peers can choose their own ordering willy-nilly.

I don't believe that, except for the pathological case where there is
a collision in priority values, this results in peers generating a
different priority order for candidate pairs.  Even in those
pathological cases, the difference in times that arises from this
(i.e., the 20ms pacing timer) is probably still *significantly*
smaller than the delay that arises from the fact that the signaling
information is complete at the answerer well before the signaling
arrives at the offerer.

From martin.thomson@gmail.com  Wed Jul 31 05:02:34 2013
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [MMUSIC] Proposed definition for "same codec configuration"
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On 31 July 2013 13:51, Jonathan Lennox <jonathan@vidyo.com> wrote:
> As I mentioned at the mic today, my straw-man proposed definition for "sa=
me codec configuration" is:
>
> Two codec configurations are the same if the PT has the same value for th=
e "rtpmap" and "fmtp" SDP parameters.

This needs to also include the major type that is included in the m=3D
line (i.e., "audio" or "video").


> Known issues with this definition:
>
> Some codecs (notably H.264's sprop-* parameters, and all of Opus's parame=
ters) have fmtp parameters describe behavior of a sender, or else are non-n=
ormative indications about the modes that would be most useful but aren't a=
ctually necessary to configure a decoder.  So it's overly restrictive to in=
sist that fmtp be identical between all the lines.

I don't believe that the utility provided by allowing the fmtp to be
split in this way outweighs the costs.  It's far simpler if this is
just defined as above.

> Getting this to work would, I think, require analysis in the style of dra=
ft-nandakumar-mmusic-sdp-mux-attributes for every fmtp parameter of every c=
odec anyone cares about.

Exactly.  Let's not do that work.  Just doing the work would be hard.
Gaining consensus on it might also cause it to take some significant
amount of time.  If it's super-important, the rfc-to-be for bundle can
be updated.

> The other issue is whether there are other SDP parameters (presumably one=
s that are specified in terms of PT) that also need to be considered to be =
part of the "codec configuration" and needs to be aligned.  Imageattr was p=
roposed as one possibility.

I don't believe imageattr to be relevant in this case.  If it were
relevant, it would only be because the codec in question didn't signal
resolution in-band, but in that case, fmtp line should contain the
resolution in that case.

From martin.thomson@gmail.com  Wed Jul 31 05:16:41 2013
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: [MMUSIC] unified plan glareless atomicity improvement
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The draft currently doesn't say much about how the glareless
negotiation features relate to a=sendrecv/sendonly/etc...  I think
that using recvonly and sendonly could reduce the probability of
encountering glare.

It's a detail.  And obviously, there are tradeoffs here.  By choosing
to use sendonly in an offer, you foreclose the ability of the answerer
to use that m-line for their streams.  So you would need to generate
extra recvonly m-lines in order to enable the 1-RTT negotiation.
There may be other implications that then stem from using extra
m-lines, especially in legacy interop cases, e.g. where BUNDLE
negotiation fails, or you can't access SSRCs etc..., each of which
have various negative consequences.

I think that the only guidance needed is something like:
"Implementations MAY choose to exclusively use m-lines with a=sendonly
or a=recvonly in order to further reduce the probably of encountering
glare."

From jonathan@vidyo.com  Wed Jul 31 05:36:48 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Martin Thomson <martin.thomson@gmail.com>
Date: Wed, 31 Jul 2013 08:33:40 -0400
Thread-Topic: [MMUSIC] Proposed definition for "same codec configuration"
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Subject: Re: [MMUSIC] Proposed definition for "same codec configuration"
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On Jul 31, 2013, at 2:01 PM, Martin Thomson <martin.thomson@gmail.com> wrot=
e:

> On 31 July 2013 13:51, Jonathan Lennox <jonathan@vidyo.com> wrote:
>> As I mentioned at the mic today, my straw-man proposed definition for "s=
ame codec configuration" is:
>>=20
>> Two codec configurations are the same if the PT has the same value for t=
he "rtpmap" and "fmtp" SDP parameters.
>=20
> This needs to also include the major type that is included in the m=3D
> line (i.e., "audio" or "video").

Right.


>> Known issues with this definition:
>>=20
>> Some codecs (notably H.264's sprop-* parameters, and all of Opus's param=
eters) have fmtp parameters describe behavior of a sender, or else are non-=
normative indications about the modes that would be most useful but aren't =
actually necessary to configure a decoder.  So it's overly restrictive to i=
nsist that fmtp be identical between all the lines.
>=20
> I don't believe that the utility provided by allowing the fmtp to be
> split in this way outweighs the costs.  It's far simpler if this is
> just defined as above.
>=20
>> Getting this to work would, I think, require analysis in the style of dr=
aft-nandakumar-mmusic-sdp-mux-attributes for every fmtp parameter of every =
codec anyone cares about.
>=20
> Exactly.  Let's not do that work.  Just doing the work would be hard.
> Gaining consensus on it might also cause it to take some significant
> amount of time.  If it's super-important, the rfc-to-be for bundle can
> be updated.

I'm inclined to agree, unless someone wants to do the work.  (I don't want =
to do the work.)  But then, I didn't want to do the work for BUNDLE either,=
 and people decided it was important enough that they went ahead and did it=
.


>> The other issue is whether there are other SDP parameters (presumably on=
es that are specified in terms of PT) that also need to be considered to be=
 part of the "codec configuration" and needs to be aligned.  Imageattr was =
proposed as one possibility.
>=20
> I don't believe imageattr to be relevant in this case.  If it were
> relevant, it would only be because the codec in question didn't signal
> resolution in-band, but in that case, fmtp line should contain the
> resolution in that case.


I'm inclined to agree here too.

To a large extent, this overlaps with the work being done in draft-nandakum=
ar-mmusic-sdp-mux-attributes.  It really doesn't matter if we say that para=
meters have to align per PT because "codec configuration" says so, or wheth=
er parameters have to align per PT because the mux-attributes draft says so=
.

--
Jonathan Lennox
jonathan@vidyo.com



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To: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On Jul 31, 2013, at 1:51 PM, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> On 31 July 2013 13:19, Dan Wing <dwing@cisco.com> wrote:
>> I agree the happy eyeballs recommendation (or, "non-stupid selection =
of priorities for dual-stack hosts") would best be folded into =
RFC5245bis.
>=20
> Great.
>=20
>> However, about a year ago Jonathan Lennox had expressed some concern =
that if the two peers have different ordering, their suicide packets =
(necessary to create mappings in their own respective NATs) would have =
to suffer an additional retry.  I guess that is okay, and why you said =
it isn't a problem.  I'm fine with that, if that's what you meant.  ICE =
has its SHOULD for the ordering and chooses one end to be authoritative =
for the ordering to avoid the problem of each peer using a different =
order.  If we're confident that guidance in ICE is over-stated and =
unnecessary, that certainly makes things easier that the two ICE peers =
can choose their own ordering willy-nilly.
>=20
> I don't believe that, except for the pathological case where there is
> a collision in priority values, this results in peers generating a
> different priority order for candidate pairs.  Even in those
> pathological cases, the difference in times that arises from this
> (i.e., the 20ms pacing timer) is probably still *significantly*
> smaller than the delay that arises from the fact that the signaling
> information is complete at the answerer well before the signaling
> arrives at the offerer.

I agree, but I would like a careful analysis before we declare =
ICE-CONTROLLED/ICE-CONTROLLING was an over-design of ICE, which is what =
we're saying if both peers are allowed to choose their own priority =
order.  And whatever the result of that analysis it should become part =
of rfc5245bis.

-d


From jonathan@vidyo.com  Wed Jul 31 05:44:15 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Dan Wing <dwing@cisco.com>
Date: Wed, 31 Jul 2013 08:44:05 -0400
Thread-Topic: [MMUSIC] RFC 5245 Section 4.1.2.1,	ice-happy-eyeballs and ICE bis
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On Jul 31, 2013, at 1:19 PM, Dan Wing <dwing@cisco.com> wrote:

> I agree the happy eyeballs recommendation (or, "non-stupid selection of p=
riorities for dual-stack hosts") would best be folded into RFC5245bis.
>=20
> However, about a year ago Jonathan Lennox had expressed some concern that=
 if the two peers have different ordering, their suicide packets (necessary=
 to create mappings in their own respective NATs) would have to suffer an a=
dditional retry.  I guess that is okay, and why you said it isn't a problem=
.  I'm fine with that, if that's what you meant.  ICE has its SHOULD for th=
e ordering and chooses one end to be authoritative for the ordering to avoi=
d the problem of each peer using a different order.  If we're confident tha=
t guidance in ICE is over-stated and unnecessary, that certainly makes thin=
gs easier that the two ICE peers can choose their own ordering willy-nilly.

If the two peers have different ordering for their *check lists*, then yes,=
 you get a problem.

However, if the two peers are both following the check list ordering algori=
thm of section 5.7.2 of RFC 5245 -- based on the signaled candidate priorit=
ies -- I think it doesn't matter whether they're both using the section 4.1=
.2.1 algorithm to calculate their *candidate* priorities.

This is confusing because section 4.1.2 claims there *will* be a problem.  =
I can't figure out why this would be true, though.  I'm inclined to suspect=
 this is a mistake in RFC 5245.

--
Jonathan Lennox
jonathan@vidyo.com



From dwing@cisco.com  Wed Jul 31 05:56:06 2013
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From: Dan Wing <dwing@cisco.com>
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To: Jonathan Lennox <jonathan@vidyo.com>
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On Jul 31, 2013, at 2:44 PM, Jonathan Lennox <jonathan@vidyo.com> wrote:

>=20
> On Jul 31, 2013, at 1:19 PM, Dan Wing <dwing@cisco.com> wrote:
>=20
>> I agree the happy eyeballs recommendation (or, "non-stupid selection =
of priorities for dual-stack hosts") would best be folded into =
RFC5245bis.
>>=20
>> However, about a year ago Jonathan Lennox had expressed some concern =
that if the two peers have different ordering, their suicide packets =
(necessary to create mappings in their own respective NATs) would have =
to suffer an additional retry.  I guess that is okay, and why you said =
it isn't a problem.  I'm fine with that, if that's what you meant.  ICE =
has its SHOULD for the ordering and chooses one end to be authoritative =
for the ordering to avoid the problem of each peer using a different =
order.  If we're confident that guidance in ICE is over-stated and =
unnecessary, that certainly makes things easier that the two ICE peers =
can choose their own ordering willy-nilly.
>=20
> If the two peers have different ordering for their *check lists*, then =
yes, you get a problem.
>=20
> However, if the two peers are both following the check list ordering =
algorithm of section 5.7.2 of RFC 5245 -- based on the signaled =
candidate priorities -- I think it doesn't matter whether they're both =
using the section 4.1.2.1 algorithm to calculate their *candidate* =
priorities.
>=20
> This is confusing because section 4.1.2 claims there *will* be a =
problem.  I can't figure out why this would be true, though.  I'm =
inclined to suspect this is a mistake in RFC 5245.

Works for me.

-d



From martin.thomson@gmail.com  Wed Jul 31 05:57:38 2013
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On 31 July 2013 14:37, Dan Wing <dwing@cisco.com> wrote:
> I agree, but I would like a careful analysis before we declare ICE-CONTRO=
LLED/ICE-CONTROLLING was an over-design of ICE, which is what we're saying =
if both peers are allowed to choose their own priority order.  And whatever=
 the result of that analysis it should become part of rfc5245bis.

This has absolutely no impact on ICE-CONTROLLED/ICE-CONTROLLING roles
or any of the other mandatory features of ICE.

In ICE, both peers select a priority order for their own candidates.
Check lists are ordered based on the mandatory formula described in
Section 5.7.2.  http://tools.ietf.org/html/rfc5245#section-5.7.2

ICE-CONTROLLED/ICE-CONTROLLING *is* over-designed, but that's never
been brought up until you mentioned it.

From jonathan@vidyo.com  Wed Jul 31 06:02:49 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Dan Wing <dwing@cisco.com>
Date: Wed, 31 Jul 2013 09:02:16 -0400
Thread-Topic: [MMUSIC] RFC 5245 Section 4.1.2.1,	ice-happy-eyeballs and ICE bis
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On Jul 31, 2013, at 2:37 PM, Dan Wing <dwing@cisco.com> wrote:
>=20
> I agree, but I would like a careful analysis before we declare ICE-CONTRO=
LLED/ICE-CONTROLLING was an over-design of ICE, which is what we're saying =
if both peers are allowed to choose their own priority order.  And whatever=
 the result of that analysis it should become part of rfc5245bis.

I know you sent this before my latest comment, but just to get it on the re=
cord -- ICE-CONTROLLED/ICE-CONTROLLING affects the section 5.7.2 algorithm,=
 and so is important for the two sides to agree on a check list ordering.  =
It doesn't affect the section 4.1.2.1 algorithm.

--
Jonathan Lennox
jonathan@vidyo.com



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Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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I am not sure if this proposal answer this topic .. I had a similar
proposal with uni-directional handling w.r.t glare ...

http://tools.ietf.org/id/draft-nandakumar-rtcweb-glare-handling-00.txt

Cheers
./S
PS: Its been a while i read the above draft myself .. So please forgive me
, if this is totally irrelevant :)



On Wed, Jul 31, 2013 at 2:16 PM, Martin Thomson <martin.thomson@gmail.com>wrote:

> The draft currently doesn't say much about how the glareless
> negotiation features relate to a=sendrecv/sendonly/etc...  I think
> that using recvonly and sendonly could reduce the probability of
> encountering glare.
>
> It's a detail.  And obviously, there are tradeoffs here.  By choosing
> to use sendonly in an offer, you foreclose the ability of the answerer
> to use that m-line for their streams.  So you would need to generate
> extra recvonly m-lines in order to enable the 1-RTT negotiation.
> There may be other implications that then stem from using extra
> m-lines, especially in legacy interop cases, e.g. where BUNDLE
> negotiation fails, or you can't access SSRCs etc..., each of which
> have various negative consequences.
>
> I think that the only guidance needed is something like:
> "Implementations MAY choose to exclusively use m-lines with a=sendonly
> or a=recvonly in order to further reduce the probably of encountering
> glare."
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

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I am not sure if this proposal answer this topic .. I had a similar proposa=
l with uni-directional handling w.r.t glare ...=A0<div><br></div><div><a hr=
ef=3D"http://tools.ietf.org/id/draft-nandakumar-rtcweb-glare-handling-00.tx=
t">http://tools.ietf.org/id/draft-nandakumar-rtcweb-glare-handling-00.txt</=
a></div>
<div><br></div><div>Cheers</div><div>./S</div><div>PS: Its been a while i r=
ead the above draft myself .. So please forgive me , if this is totally irr=
elevant :)</div><div><br></div><div><br></div><div><br><div class=3D"gmail_=
quote">
On Wed, Jul 31, 2013 at 2:16 PM, Martin Thomson <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gma=
il.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
The draft currently doesn&#39;t say much about how the glareless<br>
negotiation features relate to a=3Dsendrecv/sendonly/etc... =A0I think<br>
that using recvonly and sendonly could reduce the probability of<br>
encountering glare.<br>
<br>
It&#39;s a detail. =A0And obviously, there are tradeoffs here. =A0By choosi=
ng<br>
to use sendonly in an offer, you foreclose the ability of the answerer<br>
to use that m-line for their streams. =A0So you would need to generate<br>
extra recvonly m-lines in order to enable the 1-RTT negotiation.<br>
There may be other implications that then stem from using extra<br>
m-lines, especially in legacy interop cases, e.g. where BUNDLE<br>
negotiation fails, or you can&#39;t access SSRCs etc..., each of which<br>
have various negative consequences.<br>
<br>
I think that the only guidance needed is something like:<br>
&quot;Implementations MAY choose to exclusively use m-lines with a=3Dsendon=
ly<br>
or a=3Drecvonly in order to further reduce the probably of encountering<br>
glare.&quot;<br>
_______________________________________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
</blockquote></div><br></div>

--047d7bf10aeab0b91b04e2cecabc--

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Date: Wed, 31 Jul 2013 15:40:12 +0200
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From: Suhas Nandakumar <suhasietf@gmail.com>
To: Martin Thomson <martin.thomson@gmail.com>
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Cc: Jonathan Lennox <jonathan@vidyo.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Subject: Re: [MMUSIC] Proposed definition for "same codec configuration"
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Agreed !!!

On Wed, Jul 31, 2013 at 2:01 PM, Martin Thomson <martin.thomson@gmail.com>wrote:

> On 31 July 2013 13:51, Jonathan Lennox <jonathan@vidyo.com> wrote:
> > As I mentioned at the mic today, my straw-man proposed definition for
> "same codec configuration" is:
> >
> > Two codec configurations are the same if the PT has the same value for
> the "rtpmap" and "fmtp" SDP parameters.
>
> This needs to also include the major type that is included in the m=
> line (i.e., "audio" or "video").
>
>
> > Known issues with this definition:
> >
> > Some codecs (notably H.264's sprop-* parameters, and all of Opus's
> parameters) have fmtp parameters describe behavior of a sender, or else are
> non-normative indications about the modes that would be most useful but
> aren't actually necessary to configure a decoder.  So it's overly
> restrictive to insist that fmtp be identical between all the lines.
>
> I don't believe that the utility provided by allowing the fmtp to be
> split in this way outweighs the costs.  It's far simpler if this is
> just defined as above.
>
> > Getting this to work would, I think, require analysis in the style of
> draft-nandakumar-mmusic-sdp-mux-attributes for every fmtp parameter of
> every codec anyone cares about.
>
> Exactly.  Let's not do that work.  Just doing the work would be hard.
> Gaining consensus on it might also cause it to take some significant
> amount of time.  If it's super-important, the rfc-to-be for bundle can
> be updated.
>
> > The other issue is whether there are other SDP parameters (presumably
> ones that are specified in terms of PT) that also need to be considered to
> be part of the "codec configuration" and needs to be aligned.  Imageattr
> was proposed as one possibility.
>
> I don't believe imageattr to be relevant in this case.  If it were
> relevant, it would only be because the codec in question didn't signal
> resolution in-band, but in that case, fmtp line should contain the
> resolution in that case.
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>

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Agreed !!!<br><br><div class=3D"gmail_quote">On Wed, Jul 31, 2013 at 2:01 P=
M, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gm=
ail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt;</span> wrote:<b=
r><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">
<div class=3D"im">On 31 July 2013 13:51, Jonathan Lennox &lt;<a href=3D"mai=
lto:jonathan@vidyo.com">jonathan@vidyo.com</a>&gt; wrote:<br>
&gt; As I mentioned at the mic today, my straw-man proposed definition for =
&quot;same codec configuration&quot; is:<br>
&gt;<br>
&gt; Two codec configurations are the same if the PT has the same value for=
 the &quot;rtpmap&quot; and &quot;fmtp&quot; SDP parameters.<br>
<br>
</div>This needs to also include the major type that is included in the m=
=3D<br>
line (i.e., &quot;audio&quot; or &quot;video&quot;).<br>
<div class=3D"im"><br>
<br>
&gt; Known issues with this definition:<br>
&gt;<br>
&gt; Some codecs (notably H.264&#39;s sprop-* parameters, and all of Opus&#=
39;s parameters) have fmtp parameters describe behavior of a sender, or els=
e are non-normative indications about the modes that would be most useful b=
ut aren&#39;t actually necessary to configure a decoder. =A0So it&#39;s ove=
rly restrictive to insist that fmtp be identical between all the lines.<br>

<br>
</div>I don&#39;t believe that the utility provided by allowing the fmtp to=
 be<br>
split in this way outweighs the costs. =A0It&#39;s far simpler if this is<b=
r>
just defined as above.<br>
<div class=3D"im"><br>
&gt; Getting this to work would, I think, require analysis in the style of =
draft-nandakumar-mmusic-sdp-mux-attributes for every fmtp parameter of ever=
y codec anyone cares about.<br>
<br>
</div>Exactly. =A0Let&#39;s not do that work. =A0Just doing the work would =
be hard.<br>
Gaining consensus on it might also cause it to take some significant<br>
amount of time. =A0If it&#39;s super-important, the rfc-to-be for bundle ca=
n<br>
be updated.<br>
<div class=3D"im"><br>
&gt; The other issue is whether there are other SDP parameters (presumably =
ones that are specified in terms of PT) that also need to be considered to =
be part of the &quot;codec configuration&quot; and needs to be aligned. =A0=
Imageattr was proposed as one possibility.<br>

<br>
</div>I don&#39;t believe imageattr to be relevant in this case. =A0If it w=
ere<br>
relevant, it would only be because the codec in question didn&#39;t signal<=
br>
resolution in-band, but in that case, fmtp line should contain the<br>
resolution in that case.<br>
<div class=3D"HOEnZb"><div class=3D"h5">___________________________________=
____________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/mmusic</a><br>
</div></div></blockquote></div><br>

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To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [MMUSIC] unified plan glareless atomicity improvement
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Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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On 2013-07-31 14:16, Martin Thomson wrote:=0A=
> The draft currently doesn't say much about how the glareless=0A=
> negotiation features relate to a=3Dsendrecv/sendonly/etc...  I think=0A=
> that using recvonly and sendonly could reduce the probability of=0A=
> encountering glare.=0A=
>=0A=
> It's a detail.  And obviously, there are tradeoffs here.  By choosing=0A=
> to use sendonly in an offer, you foreclose the ability of the answerer=0A=
> to use that m-line for their streams.  So you would need to generate=0A=
> extra recvonly m-lines in order to enable the 1-RTT negotiation.=0A=
> There may be other implications that then stem from using extra=0A=
> m-lines, especially in legacy interop cases, e.g. where BUNDLE=0A=
> negotiation fails, or you can't access SSRCs etc..., each of which=0A=
> have various negative consequences.=0A=
>=0A=
> I think that the only guidance needed is something like:=0A=
> "Implementations MAY choose to exclusively use m-lines with a=3Dsendonly=
=0A=
> or a=3Drecvonly in order to further reduce the probably of encountering=
=0A=
> glare."=0A=
=0A=
This proposal makes a lot of sense to me. Stefan=0A=
=0A=
> _______________________________________________=0A=
> mmusic mailing list=0A=
> mmusic@ietf.org=0A=
> https://www.ietf.org/mailman/listinfo/mmusic=0A=
>=0A=
=0A=

From martin.thomson@gmail.com  Wed Jul 31 06:45:29 2013
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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On 31 July 2013 15:34, Suhas Nandakumar <suhasietf@gmail.com> wrote:
> I am not sure if this proposal answer this topic .. I had a similar proposal
> with uni-directional handling w.r.t glare ...
>
> http://tools.ietf.org/id/draft-nandakumar-rtcweb-glare-handling-00.txt

I think that in many respects, the unified approach to glare handling
is similar to your glare draft.  However, I believe that the unified
plan approach doesn't mandate the crazy m-line reordering shenanigans
you propose there :)  Any unified plan implementation would no longer
require a strictly ordered set of m-lines.

From suhasietf@gmail.com  Wed Jul 31 06:46:44 2013
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From: Suhas Nandakumar <suhasietf@gmail.com>
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Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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--047d7bb70e129aabc404e2cef4ed
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100% Agreed on the martial arts issue. :-)

Cheers
Suhas

On Wed, Jul 31, 2013 at 3:45 PM, Martin Thomson <martin.thomson@gmail.com>wrote:

> On 31 July 2013 15:34, Suhas Nandakumar <suhasietf@gmail.com> wrote:
> > I am not sure if this proposal answer this topic .. I had a similar
> proposal
> > with uni-directional handling w.r.t glare ...
> >
> > http://tools.ietf.org/id/draft-nandakumar-rtcweb-glare-handling-00.txt
>
> I think that in many respects, the unified approach to glare handling
> is similar to your glare draft.  However, I believe that the unified
> plan approach doesn't mandate the crazy m-line reordering shenanigans
> you propose there :)  Any unified plan implementation would no longer
> require a strictly ordered set of m-lines.
>

--047d7bb70e129aabc404e2cef4ed
Content-Type: text/html; charset=ISO-8859-1
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100% Agreed on the martial arts issue. :-) =A0<div><br></div><div>Cheers</d=
iv><div>Suhas</div><div><br><div class=3D"gmail_quote">On Wed, Jul 31, 2013=
 at 3:45 PM, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.=
thomson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt;</span=
> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On 31 July 2013 15:34, Suh=
as Nandakumar &lt;<a href=3D"mailto:suhasietf@gmail.com">suhasietf@gmail.co=
m</a>&gt; wrote:<br>

&gt; I am not sure if this proposal answer this topic .. I had a similar pr=
oposal<br>
&gt; with uni-directional handling w.r.t glare ...<br>
&gt;<br>
&gt; <a href=3D"http://tools.ietf.org/id/draft-nandakumar-rtcweb-glare-hand=
ling-00.txt" target=3D"_blank">http://tools.ietf.org/id/draft-nandakumar-rt=
cweb-glare-handling-00.txt</a><br>
<br>
</div>I think that in many respects, the unified approach to glare handling=
<br>
is similar to your glare draft. =A0However, I believe that the unified<br>
plan approach doesn&#39;t mandate the crazy m-line reordering shenanigans<b=
r>
you propose there :) =A0Any unified plan implementation would no longer<br>
require a strictly ordered set of m-lines.<br>
</blockquote></div><br></div>

--047d7bb70e129aabc404e2cef4ed--

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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: =?iso-8859-1?Q?Ari_Ker=E4nen?= <ari.keranen@ericsson.com>, "mmusic@ietf.org" <mmusic@ietf.org>
Thread-Topic: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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> -----Original Message-----
> From: Ari Ker=E4nen [mailto:ari.keranen@ericsson.com]
> Sent: Wednesday, July 31, 2013 2:50 PM
> To: mmusic@ietf.org
> Cc: Bernard Aboba; Martin Thomson; Tirumaleswar Reddy (tireddy)
> Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and IC=
E bis
>=20
> Hi,
>=20
> If we can do this as updated recommendations on how to prioritize the
> candidates, then I think it makes a lot of sense to integrate this into
> 5245bis. If it gets much more complicated than this, probably separate
> document makes more sense.
>=20
> Tiru mentioned that simply tweaking the priorities may not work in all
> cases, so perhaps we should have a look at those cases and see if we can
> make them work.

Hi Ari,

Using the formula in RFC 5245

   priority =3D (2^24)*(type preference) +
              (2^8)*(local preference) +
              (2^0)*(256 - component ID)

Type preference of=20
  Host candidates > Server-reflexive candidates > Relayed candidates

(For example host candidate is 126, server-reflexive 105)

that causes the problem that when candidate pair priority is calculated (pr=
iority =3D 2^32*MIN(G,D) + 2*MAX(G,D) + (G>D?1:0))
The candidate pair IPv4 (host, server-reflexive) will only be tested only a=
fter all IPv6 host candidate-pairs, IPv4 host candidate pairs.

Even if policy table RFC 6724 is changed so that IPv4 addresses have higher=
 precedence that does not solve the problem, It will still result in IPv4 (=
host, server-reflexive) to be tested only after all IPv4 host candidate pai=
rs, IPv6 host candidate-pairs.

Hence during the WG session, I had explained that by tweaking the candidate=
 priority will not solve the problem, but the candidate pair priority has t=
o be adjusted after offer/exchange such that after NNN number of preferred =
IPv6 candidate pairs the highest-priority IPv4 candidate-pair is tested and=
 the above adjustment continued until all candidate pairs of the preferred =
address family are exhausted.

This will gave a chance for candidate pair with server-reflexive candidates=
 to be tested earlier, which is not possible just by increasing the local c=
andidate priority.

Best Regards,
--Tiru.

>=20
>=20
> Cheers,
> Ari (as individual)
>=20
> On 7/31/13 9:31 AM, Bernard Aboba wrote:
> > I agree that it doesn't make sense to pursue the happy-eyeballs draft
> > and that folding fixes into RFC 5245bis is the best approach.
> >
> > A number of ICE implementations do not follow the priority guidance in
> > RFC 5245 Section 4.1.2.1 because it doesn't work, and also because
> > guidelines such as DoD 2008 UCR Change 3 require that implementations b=
e
> > able to prioritize IPv4 over IPv6.   So in practice, the advice in RFC
> > 5245 Section 4.1.2.1 has long been overtaken by events.
> >
> >  > http://tools.ietf.org/html/rfc5245#section-4.1.2.1
> >  >
> >  > After talking to Jonathan, several things have become apparent.
> >  >
> >  > The prioritization calculation described in Section 4.1.2.1 of RFC
> >  > 5245 is not mandatory, it's SHOULD only. This means that the guidanc=
e
> >  > regarding prioritization of IPv6 over IPv4 is not applied as strictl=
y
> >  > and blindly as implied by the happy eyeballs draft. Implementations
> >  > can choose a different prioritization. (I was under the
> >  > misapprehension that only part of the priority was signaled, but
> >  > that's clearly not the case.)
> >  >
> >  > That implies to me that the happy-eyeballs draft is, as best, only
> >  > worth pursuing as an information addendum that provides guidance on
> >  > non-stupid selection of priorities for dual-stack hosts.
> >  >
> >  > However, given the existence of an RFC5245bis effort, I am going to
> >  > suggest that the guidance provided instead be folded into a revised
> >  > Section 4.1.2.1.
> >  >
> >  > Another problem that can be addressed at the same time is a removal =
of
> >  > all the normative language that surrounds this optional feature. The
> >  > current 4.1.2.1 has lots of "MUST" statements that are both
> >  > unnecessary and confusing.
> >  >
> >  > --Martin
> >  > _______________________________________________
> >  > 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 martin.thomson@gmail.com  Wed Jul 31 07:34:50 2013
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On 31 July 2013 16:13, Tirumaleswar Reddy (tireddy) <tireddy@cisco.com> wrote:
> Using the formula in RFC 5245
>
>    priority = (2^24)*(type preference) +
>               (2^8)*(local preference) +
>               (2^0)*(256 - component ID)

This formula is just a recommendation (a SHOULD, to be fair).  So the
rest of your thesis isn't really valid.

From jonathan@vidyo.com  Wed Jul 31 07:45:35 2013
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From: Jonathan Lennox <jonathan@vidyo.com>
To: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
Date: Wed, 31 Jul 2013 10:45:08 -0400
Thread-Topic: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On Jul 31, 2013, at 4:13 PM, "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.=
com> wrote:
>=20
> Using the formula in RFC 5245
>=20
>   priority =3D (2^24)*(type preference) +
>              (2^8)*(local preference) +
>              (2^0)*(256 - component ID)
>=20
> Type preference of=20
>  Host candidates > Server-reflexive candidates > Relayed candidates
>=20
> (For example host candidate is 126, server-reflexive 105)
>=20
> that causes the problem that when candidate pair priority is calculated (=
priority =3D 2^32*MIN(G,D) + 2*MAX(G,D) + (G>D?1:0))
> The candidate pair IPv4 (host, server-reflexive) will only be tested only=
 after all IPv6 host candidate-pairs, IPv4 host candidate pairs.
>=20
> Even if policy table RFC 6724 is changed so that IPv4 addresses have high=
er precedence that does not solve the problem, It will still result in IPv4=
 (host, server-reflexive) to be tested only after all IPv4 host candidate p=
airs, IPv6 host candidate-pairs.
>=20
> Hence during the WG session, I had explained that by tweaking the candida=
te priority will not solve the problem, but the candidate pair priority has=
 to be adjusted after offer/exchange such that after NNN number of preferre=
d IPv6 candidate pairs the highest-priority IPv4 candidate-pair is tested a=
nd the above adjustment continued until all candidate pairs of the preferre=
d address family are exhausted.

The first formula you quote is the one that RFC 5245 defines as recommended=
, not mandatory.  What we're suggesting be changed is not the "local prefer=
ence" field, but actually the calculated 32-bit "priority" field for the ca=
ndidate.  You can set your priorities such that some server-reflexive candi=
dates have higher priority than some host candidates, and the calculation o=
f the pair's 64-bit priority will still work.

We need to be pretty careful when defining a new formula for how these prio=
rities are set, though.


> This will gave a chance for candidate pair with server-reflexive candidat=
es to be tested earlier, which is not possible just by increasing the local=
 candidate priority.
>=20
> Best Regards,
> --Tiru.
>=20
>>=20
>>=20
>> Cheers,
>> Ari (as individual)
>>=20
>> On 7/31/13 9:31 AM, Bernard Aboba wrote:
>>> I agree that it doesn't make sense to pursue the happy-eyeballs draft
>>> and that folding fixes into RFC 5245bis is the best approach.
>>>=20
>>> A number of ICE implementations do not follow the priority guidance in
>>> RFC 5245 Section 4.1.2.1 because it doesn't work, and also because
>>> guidelines such as DoD 2008 UCR Change 3 require that implementations b=
e
>>> able to prioritize IPv4 over IPv6.   So in practice, the advice in RFC
>>> 5245 Section 4.1.2.1 has long been overtaken by events.
>>>=20
>>>> http://tools.ietf.org/html/rfc5245#section-4.1.2.1
>>>>=20
>>>> After talking to Jonathan, several things have become apparent.
>>>>=20
>>>> The prioritization calculation described in Section 4.1.2.1 of RFC
>>>> 5245 is not mandatory, it's SHOULD only. This means that the guidance
>>>> regarding prioritization of IPv6 over IPv4 is not applied as strictly
>>>> and blindly as implied by the happy eyeballs draft. Implementations
>>>> can choose a different prioritization. (I was under the
>>>> misapprehension that only part of the priority was signaled, but
>>>> that's clearly not the case.)
>>>>=20
>>>> That implies to me that the happy-eyeballs draft is, as best, only
>>>> worth pursuing as an information addendum that provides guidance on
>>>> non-stupid selection of priorities for dual-stack hosts.
>>>>=20
>>>> However, given the existence of an RFC5245bis effort, I am going to
>>>> suggest that the guidance provided instead be folded into a revised
>>>> Section 4.1.2.1.
>>>>=20
>>>> Another problem that can be addressed at the same time is a removal of
>>>> all the normative language that surrounds this optional feature. The
>>>> current 4.1.2.1 has lots of "MUST" statements that are both
>>>> unnecessary and confusing.
>>>>=20
>>>> --Martin
>>>> _______________________________________________
>>>> 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
>=20

--
Jonathan Lennox
jonathan@vidyo.com



From ari.keranen@ericsson.com  Wed Jul 31 08:56:46 2013
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Cc: "mmusic@ietf.org" <mmusic@ietf.org>, "Tirumaleswar Reddy \(tireddy\)" <tireddy@cisco.com>
Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On 7/31/13 4:45 PM, Jonathan Lennox wrote:
>
> On Jul 31, 2013, at 4:13 PM, "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com> wrote:
>>
>> Using the formula in RFC 5245
>>
>>    priority = (2^24)*(type preference) +
>>               (2^8)*(local preference) +
>>               (2^0)*(256 - component ID)
>>
>> Type preference of
>>   Host candidates > Server-reflexive candidates > Relayed candidates
>>
>> (For example host candidate is 126, server-reflexive 105)
>>
>> that causes the problem that when candidate pair priority is calculated (priority = 2^32*MIN(G,D) + 2*MAX(G,D) + (G>D?1:0))
>> The candidate pair IPv4 (host, server-reflexive) will only be tested only after all IPv6 host candidate-pairs, IPv4 host candidate pairs.
>>
>> Even if policy table RFC 6724 is changed so that IPv4 addresses have higher precedence that does not solve the problem, It will still result in IPv4 (host, server-reflexive) to be tested only after all IPv4 host candidate pairs, IPv6 host candidate-pairs.
>>
>> Hence during the WG session, I had explained that by tweaking the candidate priority will not solve the problem, but the candidate pair priority has to be adjusted after offer/exchange such that after NNN number of preferred IPv6 candidate pairs the highest-priority IPv4 candidate-pair is tested and the above adjustment continued until all candidate pairs of the preferred address family are exhausted.
>
> The first formula you quote is the one that RFC 5245 defines as recommended, not mandatory.  What we're suggesting be changed is not the "local preference" field, but actually the calculated 32-bit "priority" field for the candidate.  You can set your priorities such that some server-reflexive candidates have higher priority than some host candidates, and the calculation of the pair's 64-bit priority will still work.
>
> We need to be pretty careful when defining a new formula for how these priorities are set, though.

If it's only about updating that formula, this indeed looks like a good 
way forward.

Just need to think how to get all we want to express (and what do we 
actually want to express) in a priority formula that isn't overly 
complex. Expressing things like "some IPv4 host-srvrflx should be 
prioritized over (some) IPv6 host-host but v4 host-host below (some) v6 
host-host" etc may get a bit messy.


Cheers,
Ari

From harald@alvestrand.no  Wed Jul 31 09:02:32 2013
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On 07/31/2013 01:51 PM, Jonathan Lennox wrote:
> As I mentioned at the mic today, my straw-man proposed definition for "same codec configuration" is:
>
> Two codec configurations are the same if the PT has the same value for the "rtpmap" and "fmtp" SDP parameters.
>
>
> Known issues with this definition:
>
> Some codecs (notably H.264's sprop-* parameters, and all of Opus's parameters) have fmtp parameters describe behavior of a sender, or else are non-normative indications about the modes that would be most useful but aren't actually necessary to configure a decoder.  So it's overly restrictive to insist that fmtp be identical between all the lines.
>
> Getting this to work would, I think, require analysis in the style of draft-nandakumar-mmusic-sdp-mux-attributes for every fmtp parameter of every codec anyone cares about.
>
> The other issue is whether there are other SDP parameters (presumably ones that are specified in terms of PT) that also need to be considered to be part of the "codec configuration" and needs to be aligned.  Imageattr was proposed as one possibility.
>
With regard to imageattr, draft-nandakumar classifies it as NORMAL,
which I think means that it only applies to flows that are mapped to the
M-line in which the attribute occurs.

So in the SDP fragment

m=video 1234 RTP/AVPF 97
a=ssrc:9876
a=rtpmap:97 VP8
a=imageattr:97 [x=1280,y=720]

m=video 2345 RTP/AVPF 97
a=ssrc:5432
a=rtpmap:97 VP8
a=imageattr:97 [x=640,y=360]

I think it is clear which flow each imageattr is intended to point to.

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Date: Wed, 31 Jul 2013 18:16:21 +0200
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From: Martin Thomson <martin.thomson@gmail.com>
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Cc: Jonathan Lennox <jonathan@vidyo.com>, "mmusic@ietf.org" <mmusic@ietf.org>, "Tirumaleswar Reddy \(tireddy\)" <tireddy@cisco.com>
Subject: Re: [MMUSIC] RFC 5245 Section 4.1.2.1, ice-happy-eyeballs and ICE bis
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On 31 July 2013 17:56, Ari Ker=C3=A4nen <ari.keranen@ericsson.com> wrote:
> overly complex

Yes.  Given that this is a very small part of a spec that already fits
that description, we need to be careful.

From pkyzivat@alum.mit.edu  Wed Jul 31 09:44:32 2013
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Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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I don't understand how sendonly and recvonly reduce glare.
If I offer a sendonly m-line, it still must be answered. And 
subsequently an offer for that m-line can come from either side, 
possibly producing glare.

ISTM that you should offer a sendrecv m-line iff you intend a 
bidirectional flow, where the two sides are related, will have common 
properties and lifetimes.

There is still a problem if both sides want a bidirectional flow, but 
aren't agreed on who should initiate it. In that case they both might 
offer and glare. And there there remains the question of whether they 
recognize this as glare, rather than thinking they need both flows.

	Thanks,
	Paul

On 7/31/13 2:16 PM, Martin Thomson wrote:
> The draft currently doesn't say much about how the glareless
> negotiation features relate to a=sendrecv/sendonly/etc...  I think
> that using recvonly and sendonly could reduce the probability of
> encountering glare.
>
> It's a detail.  And obviously, there are tradeoffs here.  By choosing
> to use sendonly in an offer, you foreclose the ability of the answerer
> to use that m-line for their streams.  So you would need to generate
> extra recvonly m-lines in order to enable the 1-RTT negotiation.
> There may be other implications that then stem from using extra
> m-lines, especially in legacy interop cases, e.g. where BUNDLE
> negotiation fails, or you can't access SSRCs etc..., each of which
> have various negative consequences.
>
> I think that the only guidance needed is something like:
> "Implementations MAY choose to exclusively use m-lines with a=sendonly
> or a=recvonly in order to further reduce the probably of encountering
> glare."
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>


From pkyzivat@alum.mit.edu  Wed Jul 31 10:08:22 2013
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Subject: Re: [MMUSIC] Proposed definition for "same codec configuration"
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On 7/31/13 1:51 PM, Jonathan Lennox wrote:
> As I mentioned at the mic today, my straw-man proposed definition for "same codec configuration" is:
>
> Two codec configurations are the same if the PT has the same value for the "rtpmap" and "fmtp" SDP parameters.
>
>
> Known issues with this definition:
>
> Some codecs (notably H.264's sprop-* parameters, and all of Opus's parameters) have fmtp parameters describe behavior of a sender, or else are non-normative indications about the modes that would be most useful but aren't actually necessary to configure a decoder.  So it's overly restrictive to insist that fmtp be identical between all the lines.
>
> Getting this to work would, I think, require analysis in the style of draft-nandakumar-mmusic-sdp-mux-attributes for every fmtp parameter of every codec anyone cares about.
>
> The other issue is whether there are other SDP parameters (presumably ones that are specified in terms of PT) that also need to be considered to be part of the "codec configuration" and needs to be aligned.  Imageattr was proposed as one possibility.

I have a couple of questions:

When configuring encoder/decoder parameters, regardless of what sort of 
line does it:

AFAIK there is separate encoder/decoder state for each SSRC, right?

So these parameters are just templates that are used for however many 
encoders &/ decoders are set up for that m-line. (Each with a different 
SSRC.)

If you are mapping packets to m-lines by SSRC, then having different 
values per-m-line works.

Now suppose you were mapping packets to m-lines via appid rather than 
SSRC. And suppose initially all packets having appid X have SSID 1111, 
and that is the first appearance of 1111. Then presumably the attributes 
of the m-line corresponding to appid X will determine how the decoder 
are initialized.

Then SSID 1111 is reassigned to appid Y, and is mapped to a different 
m-line. That m-line has different decoder parameters. Should the decoder 
for 1111 be reconfigured?

	Thanks,
	Paul


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Subject: Re: [MMUSIC] unified plan glareless atomicity improvement
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On 31 July 2013 18:42, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> I don't understand how sendonly and recvonly reduce glare.

It just reduces the probability of a change collision by further
increasing the atomicity of the overall fragmented document.  For a
bi-directional line in a=sendrecv, there is more *stuff*.

There really isn't such a thing as a bidirectional flow once you are
talking about BUNGLE.
