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From: <simon.becot@orange.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: New draft submitted draft-copeland-rtcweb-p2p-idp-auth-00
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Date: Mon, 3 Oct 2016 13:17:22 +0000
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Subject: [rtcweb] New draft submitted draft-copeland-rtcweb-p2p-idp-auth-00
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Hi,
I would like to inform you that a new draft has been submitted on identity =
requirements for the WebRTC communication: https://www.ietf.org/id/draft-co=
peland-rtcweb-p2p-idp-auth-00.txt .
This draft has been written by a group of people from Orange, Deutsche Tele=
kom and Institut Mines Telecom, involved in reThink collaborative project (=
https://rethink-project.eu/ ). It studies the relationships of WebRTC commu=
nication users with their web Calling Services (CS) and their Identity Prov=
iders (IdPs), in order to identify important requirements for IdP based pee=
r-to-peer authentication.
We would be very pleased to present this to the next IETF meeting.

Simon B=E9cot,
Orange Labs

___________________________________________________________________________=
______________________________________________

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osoft-com:office:office" xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
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<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal">I would like to inform you that a new draft has been=
 submitted on identity requirements for the WebRTC communication:
<a href=3D"https://www.ietf.org/id/draft-copeland-rtcweb-p2p-idp-auth-00.tx=
t">https://www.ietf.org/id/draft-copeland-rtcweb-p2p-idp-auth-00.txt</a> .<=
o:p></o:p></p>
<p class=3D"MsoNormal">This draft has been written by a group of people fro=
m Orange, Deutsche Telekom and Institut Mines Telecom, involved in reThink =
collaborative project (<a href=3D"https://rethink-project.eu/">https://reth=
ink-project.eu/</a> ). It studies the
 relationships of WebRTC communication users with their web Calling Service=
s (CS) and their Identity Providers (IdPs), in order to identify important =
requirements for IdP based peer-to-peer authentication.<o:p></o:p></p>
<p class=3D"MsoNormal">We would be very pleased to present this to the next=
 IETF meeting.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Simon B=E9cot, <o:p></o:p></p>
<p class=3D"MsoNormal">Orange Labs<o:p></o:p></p>
</div>
<PRE>______________________________________________________________________=
___________________________________________________

Ce message et ses pieces jointes peuvent contenir des informations confiden=
tielles ou privilegiees et ne doivent donc
pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu=
 ce message par erreur, veuillez le signaler
a l'expediteur et le detruire ainsi que les pieces jointes. Les messages el=
ectroniques 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 inf=
ormation 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 dele=
te 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.
</PRE></body>
</html>

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From nobody Tue Oct  4 00:36:08 2016
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To: Michael Welzl <michawe@ifi.uio.no>
References: <20160804135822.15952.16587.idtracker@ietfa.amsl.com> <AEBFDB5A-17DB-450D-8779-FDDCC2782C2D@ifi.uio.no> <d9f43cf8-652c-d03c-339a-b7295a3f2d30@alvestrand.no> <247D0AC1-B726-4C65-868F-66C920C4A55A@ifi.uio.no> <8995aea4-7016-bc93-e58c-c3a4d778c91d@alvestrand.no> <B1E206C6-DBE7-441B-B0B6-C5270A15E71E@ifi.uio.no>
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] =?utf-8?q?Mirja_K=C3=BChlewind=27s_No_Objection_on_draft?= =?utf-8?q?-ietf-rtcweb-transports-15=3A_=28with_COMMENT=29?=
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After listening to the (silent) list and considering the matter further,
I'm adding the following paragraph at the end of the "local
prioritization" section:

   Note that this specification doesn't dictate when disparate streams
   are to be "congestion controlled under the same congestion control
   regime".  The issue of coupling congestion controllers is explored
further in
   [I-D.ietf-rmcat-coupled-cc].

and adding that as an informational reference.


Den 30. sep. 2016 14:28, skrev Michael Welzl:
> 
>> On 30. sep. 2016, at 12.59, Harald Alvestrand <harald@alvestrand.no> wrote:
>>
>> Den 30. sep. 2016 12:31, skrev Michael Welzl:
>>>
>>>> On 30. sep. 2016, at 11.41, Harald Alvestrand <harald@alvestrand.no> wrote:
>>>>
>>>> Wanting to ask ....
>>>>
>>>> do you have commitment from some of the browser vendors that they intend
>>>> to add -coupled to their congestion control code?
>>>
>>> No committment, but I did have some hope that you guys at Google would be interested?
>>> Stefan was interested in coupling the data channel with the media channel, which is probably a straightforward way of adapting our coupling algorithm.
>>>
>>> We’re going to work on this code ourselves; we already have preliminary code on coupling + shared bottleneck detection in Chromium, but this is work in progress that was interrupted for a year due to a contract that let us focus on coupling TCP for a while (which also works very well, BTW; we’ll provide an update at ICCRG in Seoul). So we’re (well: Safiqul is) just about to get back to it.
>>>
>>>
>>>> That would go a long way towards allaying my fear that this would be
>>>> another RFC 6919 "MUST (BUT WE KNOW YOU WON'T)" requirement.
>>>
>>> I think what I’m asking for is a far cry from a MUST.
>>> It’s merely about putting a reference there such that implementers see both possibilities (coupling and/or dscp), not only one (dscp).
>>>
>>
>> The DSCP section would be the wrong section for it anyway.
>>
>> References to -coupled should be in the "local prioritization" section
>> (4.1).
> 
> I agree. I didn’t mean for this to be in the same section.
> 
> 
>> My worry about -coupled in the previous round is that it's a *specific*
>> instance of coupled congestion controllers (and experimental-track).
>>
>> We don't yet know if it's *the* way.
> 
> Understood - but:
> 
> 1) draft-ietf-rmcat-coupled-cc doesn’t say that the algorithm in it is *the* way. I believe it correctly acknowledges that you could e.g. also have a single congestion control instance for everything ("congestion manager” style coupling). Both implementations have their pro’s and con’s - our algorithm is easier in some ways, e.g. for turning on and off (which is useful when coupling flows to multiple destinations).
> 
> 
> 2) as I said, we have lots of indications that this algorithm works for *many* controllers with only minimal changes.
> 
> Some details:
> We successfully used it with NADA and GCC ( http://heim.ifi.uio.no/michawe/research/publications/noms2016.pdf ), but also TFRC and RAP  ( http://heim.ifi.uio.no/michawe/research/publications/csws14.pdf ), LEDBAT ( slide 31: http://heim.ifi.uio.no/michawe/research/publications/caia2014.pdf ), also LEDBAT combined with another congestion control (don’t remember which, maybe RAP), and most recently TCP. TCP is perhaps an extreme case because it has so many states. If you ignore all this and just blindly apply our algorithm, you get pretty much the behavior of E-TCP, which isn’t bad ( http://www.isi.edu/div7/publication_files/effects_of_ensemble.pdf ). We have improved on it by correctly incorporating the states - this makes things better, but note that all of this is better than simply leaving flows as they are and let them compete against each other, in particular when the goal is to minimize latency.
> 
> 
> I’d find it pretty strange if this isn’t enough reason to justify a mention of this possibility and a reference.
> 
> 
>> That said, I personally wouldn't worry too much about mentioning it, but
>> it's a post-IESG change without IESG buy-in, so I'll want more
>> authorization to do that.
> 
> Sure
> 
> cheers,
> Michael
> 


From nobody Tue Oct  4 01:42:32 2016
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Real-Time Communication in WEB-browsers of the IETF.

        Title           : Transports for WebRTC
        Author          : Harald Alvestrand
	Filename        : draft-ietf-rtcweb-transports-16.txt
	Pages           : 20
	Date            : 2016-10-04

Abstract:
   This document describes the data transport protocols used by WebRTC,
   including the protocols used for interaction with intermediate boxes
   such as firewalls, relays and NAT boxes.


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

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

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


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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From nobody Tue Oct  4 01:48:05 2016
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I believe this version closes all post-IESG open issues on -transports.

Thank you for your consideration.

-------- Videresendt melding --------
Emne: New Version Notification for draft-ietf-rtcweb-transports-16.txt
Dato: Tue, 04 Oct 2016 01:42:30 -0700
Fra: internet-drafts@ietf.org
Til: Harald Alvestrand <harald@alvestrand.no>


A new version of I-D, draft-ietf-rtcweb-transports-16.txt
has been successfully submitted by Harald Alvestrand and posted to the
IETF repository.

Name:		draft-ietf-rtcweb-transports
Revision:	16
Title:		Transports for WebRTC
Document date:	2016-10-04
Group:		rtcweb
Pages:		20
URL:
https://www.ietf.org/internet-drafts/draft-ietf-rtcweb-transports-16.txt
Status:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/
Htmlized:       https://tools.ietf.org/html/draft-ietf-rtcweb-transports-16
Diff:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-16

Abstract:
   This document describes the data transport protocols used by WebRTC,
   including the protocols used for interaction with intermediate boxes
   such as firewalls, relays and NAT boxes.




Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

The IETF Secretariat


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From: Colin Perkins <csp@csperkins.org>
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References: <CAOW+2dtN-tfmnsep8FjOFD2R2uZbUwZxHfDGBwx3N13Ue9Nb2w@mail.gmail.com> <E1B53795-4B14-408B-BFF1-A305EE45AD20@iii.ca> <66CB8C7B-56D1-4AC9-B0F8-4F1689E2CA89@vidyo.com> <E7C35776-C554-47C5-A522-F1C02629041F@iii.ca> <A52D1EFE-6A7D-418F-A77B-176E184EBAB9@vidyo.com> <07afeaa4-06fb-b6a9-b2d7-f944b471fde1@ericsson.com> <949bdcde-b24d-f2d4-e855-91424b2be30a@alvestrand.no> <4837a71e-0e54-29d0-0a96-75f3fdbc7e1b@ericsson.com> <2a67053c-bd75-cc85-eeba-e57f64667ab1@alvestrand.no>
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Subject: Re: [rtcweb] JSEP: RTP demux algorithms
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> On 26 Sep 2016, at 10:35, Harald Alvestrand <harald@alvestrand.no> =
wrote:
>=20
> Den 26. sep. 2016 11:14, skrev Magnus Westerlund:
>> Den 2016-09-26 kl. 10:48, skrev Harald Alvestrand:
>>> Den 23. sep. 2016 10:56, skrev Magnus Westerlund:
>>>> Hi,
>>>>=20
>>>> I think the use cases looks fairly covering.
>>>>=20
>>>> I want to add an alternative way of thinking on how the SDES item =
(MID,
>>>> RID) related demultiplexing part is handled. So the RID and MID in =
RTP
>>>> header extensions or in RTSP SDES packets results in these values =
being
>>>> bound to the SSRC until further notice. So packets arriving without
>>>> actual RTP header extensions can still be handled according to the =
SSRCs
>>>> associated SDES items. =46rom my perspective all the cases with MID =
and
>>>> RID can be handled on a per packet basis rules without latching the =
SSRC
>>>> to a specific receiver. Instead what is "latched" is the SDES =
values to
>>>> the SSRC. Still at the point of change there needs to take the
>>>> combination of SSRC and Seq into account to avoid miss routing =
packets.
>>>>=20
>>>> So my general handling algorithm would be this sequence:
>>>>=20
>>>> 1. Check if RTP header extensions update any SDES item for the =
SSRC.
>>>>   a. If they do, mark the Sequence number and keep the old SDES =
item as
>>>> valid prior to this extended sequence number.
>>>>=20
>>>> 2. Determine the set of relevant SDES items that applies for this
>>>> packet, i.e. use SSRC + RTP seq to look up the SDES items, e.g. MID =
and
>>>> RID.
>>>>=20
>>>> 3. Perform the demuxing algorithm
>>>=20
>>> I don't understand this as written.
>>> When an RTP packet arrives before the RTCP packet containing SDES, =
we
>>> can't latch to the content of the RTCP packet; we haven't seen it =
yet.
>>=20
>> So if neither RTP header extension nor RTCP SDES item with RID/MID =
for
>> that SSRC has been received, then you don't have that information and
>> you will have to either queue or drop that packet if the relation =
isn't
>> otherwise clear through PT or SSRC signalling in SDP.
>>=20
>>>=20
>>> You can, in theory, latch RID and MID for the SSRC to the values in =
the
>>> SDES packet you haven't seen yet, but that seems somewhat backwards.
>>>=20
>>> The problematic sequence is:
>>>=20
>>> 1) RTP packet(s) with header extensions
>>> 2) RTP packet(s) without header extensions
>>> 3) SDES RTCP packet
>>=20
>> I don't see why the above is problematic. Step one gives you the SDES
>> item for that SSRC. Thus in step 2 one knows that the SSRC has the
>> MID/RID value that you need.
>>=20
>> I don't understand why you are not binding the SDES info from the RTP
>> header extension to the SSRC.
>=20
> OK, I had missed the change of the RID header extension from being its
> own header extension to being a component of the SDES header extension =
only.
>=20
> Then it all makes sense! Thanks!

I think I agree with Magnus. RTP flows are based around SSRCs, so the =
key to demultiplexing is that RTP packets are grouped by SSRC, then that =
SSRC is mapped to some higher level role based on some combination of =
signalled SSRC, MID, RID, etc.=20

I=92d suggest that the right way to write Section 7 of JSEP is not =93how =
to route RTP packets=94, as RTP packets are always grouped by SSRC, but =
rather how to map SSRCs to higher-level roles, based on updates of the =
mapping received in SDP, RTCP SDES, and RTP header extensions.

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





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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Colin Perkins <csp@csperkins.org>, Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] JSEP: RTP demux algorithms
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Subject: Re: [rtcweb] JSEP: RTP demux algorithms
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Hi,

I still question why the low-level RTP demuxing shall be described in
JSEP, instead of e.g., BUNDLE.

Keep in mind that all BUNDLE implementations may not implement JSEP, but I
still think we want the RTP demuxing to be done in the same way?

Regards,

Christer


On 04/10/16 14:41, "rtcweb on behalf of Colin Perkins"
<rtcweb-bounces@ietf.org on behalf of csp@csperkins.org> wrote:

>
>> On 26 Sep 2016, at 10:35, Harald Alvestrand <harald@alvestrand.no>
>>wrote:
>>=20
>> Den 26. sep. 2016 11:14, skrev Magnus Westerlund:
>>> Den 2016-09-26 kl. 10:48, skrev Harald Alvestrand:
>>>> Den 23. sep. 2016 10:56, skrev Magnus Westerlund:
>>>>> Hi,
>>>>>=20
>>>>> I think the use cases looks fairly covering.
>>>>>=20
>>>>> I want to add an alternative way of thinking on how the SDES item
>>>>>(MID,
>>>>> RID) related demultiplexing part is handled. So the RID and MID in
>>>>>RTP
>>>>> header extensions or in RTSP SDES packets results in these values
>>>>>being
>>>>> bound to the SSRC until further notice. So packets arriving without
>>>>> actual RTP header extensions can still be handled according to the
>>>>>SSRCs
>>>>> associated SDES items. From my perspective all the cases with MID and
>>>>> RID can be handled on a per packet basis rules without latching the
>>>>>SSRC
>>>>> to a specific receiver. Instead what is "latched" is the SDES values
>>>>>to
>>>>> the SSRC. Still at the point of change there needs to take the
>>>>> combination of SSRC and Seq into account to avoid miss routing
>>>>>packets.
>>>>>=20
>>>>> So my general handling algorithm would be this sequence:
>>>>>=20
>>>>> 1. Check if RTP header extensions update any SDES item for the SSRC.
>>>>>   a. If they do, mark the Sequence number and keep the old SDES item
>>>>>as
>>>>> valid prior to this extended sequence number.
>>>>>=20
>>>>> 2. Determine the set of relevant SDES items that applies for this
>>>>> packet, i.e. use SSRC + RTP seq to look up the SDES items, e.g. MID
>>>>>and
>>>>> RID.
>>>>>=20
>>>>> 3. Perform the demuxing algorithm
>>>>=20
>>>> I don't understand this as written.
>>>> When an RTP packet arrives before the RTCP packet containing SDES, we
>>>> can't latch to the content of the RTCP packet; we haven't seen it yet.
>>>=20
>>> So if neither RTP header extension nor RTCP SDES item with RID/MID for
>>> that SSRC has been received, then you don't have that information and
>>> you will have to either queue or drop that packet if the relation isn't
>>> otherwise clear through PT or SSRC signalling in SDP.
>>>=20
>>>>=20
>>>> You can, in theory, latch RID and MID for the SSRC to the values in
>>>>the
>>>> SDES packet you haven't seen yet, but that seems somewhat backwards.
>>>>=20
>>>> The problematic sequence is:
>>>>=20
>>>> 1) RTP packet(s) with header extensions
>>>> 2) RTP packet(s) without header extensions
>>>> 3) SDES RTCP packet
>>>=20
>>> I don't see why the above is problematic. Step one gives you the SDES
>>> item for that SSRC. Thus in step 2 one knows that the SSRC has the
>>> MID/RID value that you need.
>>>=20
>>> I don't understand why you are not binding the SDES info from the RTP
>>> header extension to the SSRC.
>>=20
>> OK, I had missed the change of the RID header extension from being its
>> own header extension to being a component of the SDES header extension
>>only.
>>=20
>> Then it all makes sense! Thanks!
>
>I think I agree with Magnus. RTP flows are based around SSRCs, so the key
>to demultiplexing is that RTP packets are grouped by SSRC, then that SSRC
>is mapped to some higher level role based on some combination of
>signalled SSRC, MID, RID, etc.
>
>I=B9d suggest that the right way to write Section 7 of JSEP is not =B3how =
to
>route RTP packets=B2, as RTP packets are always grouped by SSRC, but rathe=
r
>how to map SSRCs to higher-level roles, based on updates of the mapping
>received in SDP, RTCP SDES, and RTP header extensions.
>
>--=20
>Colin Perkins
>https://csperkins.org/
>
>
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] JSEP: RTP demux algorithms
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> On 4 Oct 2016, at 13:00, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
> I still question why the low-level RTP demuxing shall be described in
> JSEP, instead of e.g., BUNDLE.
>=20
> Keep in mind that all BUNDLE implementations may not implement JSEP, =
but I
> still think we want the RTP demuxing to be done in the same way?

Don=E2=80=99t you need both?=20

BUNDLE would specify how to map an RTP SSRC onto something described in =
SDP, using the combination of a=3Dssrc: lines, and the information =
carried in RTCP and RTP header extensions (MID, RID, etc.). That is, it =
maps SSRC -> =E2=80=9Crole=E2=80=9D.

JSEP would specify how to map that =E2=80=9Crole=E2=80=9D onto the =
concepts visible in the Javascript API.

Colin




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





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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Colin Perkins <csp@csperkins.org>
Thread-Topic: [rtcweb] JSEP: RTP demux algorithms
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Subject: Re: [rtcweb] JSEP: RTP demux algorithms
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Hi,

>> On 4 Oct 2016, at 13:00, Christer Holmberg
>><christer.holmberg@ericsson.com> wrote:
>>=20
>> I still question why the low-level RTP demuxing shall be described in
>> JSEP, instead of e.g., BUNDLE.
>>=20
>> Keep in mind that all BUNDLE implementations may not implement JSEP,
>>but I
>> still think we want the RTP demuxing to be done in the same way?
>
>Don=B9t you need both?

Yes.

>BUNDLE would specify how to map an RTP SSRC onto something described in
>SDP, using the combination of a=3Dssrc: lines, and the information carried
>in RTCP and RTP header extensions (MID, RID, etc.). That is, it maps SSRC
>-> =B3role=B2.

Yes.

>JSEP would specify how to map that =B3role=B2 onto the concepts visible in
>the Javascript API.

Yes. But, based on the discussions it seems like JSEP would define what
you suggest should be done in BUNDLE.

Regards,

Christer



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Den 04. okt. 2016 14:00, skrev Christer Holmberg:
> Hi,
> 
> I still question why the low-level RTP demuxing shall be described in
> JSEP, instead of e.g., BUNDLE.
> 
> Keep in mind that all BUNDLE implementations may not implement JSEP, but I
> still think we want the RTP demuxing to be done in the same way?

Does BUNDLE say how to handle RID?

Seriously, please get BUNDLE out the door and don't tangle it in any
more new stuff.


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From: Christer Holmberg <christer.holmberg@ericsson.com>
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Hi,

>>I still question why the low-level RTP demuxing shall be described in
>> JSEP, instead of e.g., BUNDLE.
>>=20
>> Keep in mind that all BUNDLE implementations may not implement JSEP,
>>but I
>> still think we want the RTP demuxing to be done in the same way?
>
>Does BUNDLE say how to handle RID?

No. Cullen wanted more text, though, but I am not sure whether that
covered RID.

>Seriously, please get BUNDLE out the door and don't tangle it in any
>more new stuff.

If that was meant to me, I will pretend I didn=B9t see it=8A

Regards,

Christer


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From: Harald Alvestrand <harald@alvestrand.no>
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Den 04. okt. 2016 14:29, skrev Christer Holmberg:
> Hi,
> 
>>> I still question why the low-level RTP demuxing shall be described in
>>> JSEP, instead of e.g., BUNDLE.
>>>
>>> Keep in mind that all BUNDLE implementations may not implement JSEP,
>>> but I
>>> still think we want the RTP demuxing to be done in the same way?
>>
>> Does BUNDLE say how to handle RID?
> 
> No. Cullen wanted more text, though, but I am not sure whether that
> covered RID.
> 
>> Seriously, please get BUNDLE out the door and don't tangle it in any
>> more new stuff.
> 
> If that was meant to me, I will pretend I didn¹t see itŠ

It's actually a grave concern of mine - more addressed to the chairs of
the relevant WGs than to anyone else.

Any open document is a magnet for new functionality, and any new
functionality delays the document's publication.

The bundle document, with its associated changes in registration
procedures for SDP attributes (the mux category), needs to be seen as
stable. RFC publication is the best way to make it so.

Datatracker says we've been working on this document since 2011. It's
now at version -32, and was in state "Minor updates needed to address
WGLC comments" in May.

If we can segregate new requests for functionality into other documents,
I think that is a win.

Harald


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thanks!  i agree with this update

Sent from my iPhone

> On 4. okt. 2016, at 10.35, Harald Alvestrand <harald@alvestrand.no> wrote:=

>=20
> After listening to the (silent) list and considering the matter further,
> I'm adding the following paragraph at the end of the "local
> prioritization" section:
>=20
>   Note that this specification doesn't dictate when disparate streams
>   are to be "congestion controlled under the same congestion control
>   regime".  The issue of coupling congestion controllers is explored
> further in
>   [I-D.ietf-rmcat-coupled-cc].
>=20
> and adding that as an informational reference.
>=20
>=20
> Den 30. sep. 2016 14:28, skrev Michael Welzl:
>>=20
>>> On 30. sep. 2016, at 12.59, Harald Alvestrand <harald@alvestrand.no> wro=
te:
>>>=20
>>> Den 30. sep. 2016 12:31, skrev Michael Welzl:
>>>>=20
>>>>> On 30. sep. 2016, at 11.41, Harald Alvestrand <harald@alvestrand.no> w=
rote:
>>>>>=20
>>>>> Wanting to ask ....
>>>>>=20
>>>>> do you have commitment from some of the browser vendors that they inte=
nd
>>>>> to add -coupled to their congestion control code?
>>>>=20
>>>> No committment, but I did have some hope that you guys at Google would b=
e interested?
>>>> Stefan was interested in coupling the data channel with the media chann=
el, which is probably a straightforward way of adapting our coupling algorit=
hm.
>>>>=20
>>>> We=E2=80=99re going to work on this code ourselves; we already have pre=
liminary code on coupling + shared bottleneck detection in Chromium, but thi=
s is work in progress that was interrupted for a year due to a contract that=
 let us focus on coupling TCP for a while (which also works very well, BTW; w=
e=E2=80=99ll provide an update at ICCRG in Seoul). So we=E2=80=99re (well: S=
afiqul is) just about to get back to it.
>>>>=20
>>>>=20
>>>>> That would go a long way towards allaying my fear that this would be
>>>>> another RFC 6919 "MUST (BUT WE KNOW YOU WON'T)" requirement.
>>>>=20
>>>> I think what I=E2=80=99m asking for is a far cry from a MUST.
>>>> It=E2=80=99s merely about putting a reference there such that implement=
ers see both possibilities (coupling and/or dscp), not only one (dscp).
>>>=20
>>> The DSCP section would be the wrong section for it anyway.
>>>=20
>>> References to -coupled should be in the "local prioritization" section
>>> (4.1).
>>=20
>> I agree. I didn=E2=80=99t mean for this to be in the same section.
>>=20
>>=20
>>> My worry about -coupled in the previous round is that it's a *specific*
>>> instance of coupled congestion controllers (and experimental-track).
>>>=20
>>> We don't yet know if it's *the* way.
>>=20
>> Understood - but:
>>=20
>> 1) draft-ietf-rmcat-coupled-cc doesn=E2=80=99t say that the algorithm in i=
t is *the* way. I believe it correctly acknowledges that you could e.g. also=
 have a single congestion control instance for everything ("congestion manag=
er=E2=80=9D style coupling). Both implementations have their pro=E2=80=99s a=
nd con=E2=80=99s - our algorithm is easier in some ways, e.g. for turning on=
 and off (which is useful when coupling flows to multiple destinations).
>>=20
>>=20
>> 2) as I said, we have lots of indications that this algorithm works for *=
many* controllers with only minimal changes.
>>=20
>> Some details:
>> We successfully used it with NADA and GCC ( http://heim.ifi.uio.no/michaw=
e/research/publications/noms2016.pdf ), but also TFRC and RAP  ( http://heim=
.ifi.uio.no/michawe/research/publications/csws14.pdf ), LEDBAT ( slide 31: h=
ttp://heim.ifi.uio.no/michawe/research/publications/caia2014.pdf ), also LED=
BAT combined with another congestion control (don=E2=80=99t remember which, m=
aybe RAP), and most recently TCP. TCP is perhaps an extreme case because it h=
as so many states. If you ignore all this and just blindly apply our algorit=
hm, you get pretty much the behavior of E-TCP, which isn=E2=80=99t bad ( htt=
p://www.isi.edu/div7/publication_files/effects_of_ensemble.pdf ). We have im=
proved on it by correctly incorporating the states - this makes things bette=
r, but note that all of this is better than simply leaving flows as they are=
 and let them compete against each other, in particular when the goal is to m=
inimize latency.
>>=20
>>=20
>> I=E2=80=99d find it pretty strange if this isn=E2=80=99t enough reason to=
 justify a mention of this possibility and a reference.
>>=20
>>=20
>>> That said, I personally wouldn't worry too much about mentioning it, but=

>>> it's a post-IESG change without IESG buy-in, so I'll want more
>>> authorization to do that.
>>=20
>> Sure
>>=20
>> cheers,
>> Michael
>>=20


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Tue, 4 Oct 2016 14:58:14 -0700
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Subject: Re: [rtcweb] JSEP: RTP demux algorithms
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Harald said:

"If we can segregate new requests for functionality into other documents,
I think that is a win."

[BA] In general, I agree with Harald.  However, there is one issue that I
do think that BUNDLE needs to address, which is what happens when there are
non-unique PTs.  JSEP Section 7 says:

      "If any payload type could map to more than one RtpReceiver, map to
      the RtpReceiver whose m=3D section appears earliest in the local
description."



While this is one way of handling it, there are implementations that
have solved the problem in a different way.  For example, I believe
that ORTC Lib maps to the RtpReceiver whose m=3D section contains no
specified SSRCs.


At this point, I do not feel strongly about *which* solution is
chosen, but since we have multiple non-interoperable BUNDLE
implementations, I would like the BUNDLE draft to add a sentence
choosing one approach, which can be referenced in JSEP Section 7.


On Tue, Oct 4, 2016 at 6:05 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> Den 04. okt. 2016 14:29, skrev Christer Holmberg:
> > Hi,
> >
> >>> I still question why the low-level RTP demuxing shall be described in
> >>> JSEP, instead of e.g., BUNDLE.
> >>>
> >>> Keep in mind that all BUNDLE implementations may not implement JSEP,
> >>> but I
> >>> still think we want the RTP demuxing to be done in the same way?
> >>
> >> Does BUNDLE say how to handle RID?
> >
> > No. Cullen wanted more text, though, but I am not sure whether that
> > covered RID.
> >
> >> Seriously, please get BUNDLE out the door and don't tangle it in any
> >> more new stuff.
> >
> > If that was meant to me, I will pretend I didn=C2=B9t see it=C5=A0
>
> It's actually a grave concern of mine - more addressed to the chairs of
> the relevant WGs than to anyone else.
>
> Any open document is a magnet for new functionality, and any new
> functionality delays the document's publication.
>
> The bundle document, with its associated changes in registration
> procedures for SDP attributes (the mux category), needs to be seen as
> stable. RFC publication is the best way to make it so.
>
> Datatracker says we've been working on this document since 2011. It's
> now at version -32, and was in state "Minor updates needed to address
> WGLC comments" in May.
>
> If we can segregate new requests for functionality into other documents,
> I think that is a win.
>
> Harald
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">Harald said:=C2=A0<div><br></div><div>&quot;<span style=3D=
"font-size:12.8px">If we can segregate new requests for functionality into =
other documents,</span></div><div><span style=3D"font-size:12.8px">I think =
that is a win.</span>&quot;</div><div><br></div><div>[BA] In general, I agr=
ee with Harald.=C2=A0 However, there is one issue that I do think that BUND=
LE needs to address, which is what happens when there are non-unique PTs.=
=C2=A0 JSEP Section 7 says:=C2=A0</div><div><br></div><pre class=3D"gmail-n=
ewpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color=
:rgb(0,0,0)">      &quot;If any payload type could map to more than one Rtp=
Receiver, map to
      the RtpReceiver whose m=3D section appears earliest in the local=C2=
=A0description.&quot;</pre><pre class=3D"gmail-newpage" style=3D"font-size:=
13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre=
 class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin=
-bottom:0px;color:rgb(0,0,0)"><br></pre><pre class=3D"gmail-newpage" style=
=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">=
<span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:s=
mall;white-space:normal">While this is one way of handling it, there are im=
plementations that have solved the problem in a different way.=C2=A0 For ex=
ample, I believe that ORTC Lib maps to the RtpReceiver whose m=3D section c=
ontains no specified SSRCs.=C2=A0</span><br></pre><pre class=3D"gmail-newpa=
ge" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb=
(0,0,0)"><span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;fo=
nt-size:small;white-space:normal"><br></span></pre><pre class=3D"gmail-newp=
age" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rg=
b(0,0,0)"><span style=3D"color:rgb(34,34,34);font-family:arial,sans-serif;f=
ont-size:small;white-space:normal">At this point, I do not feel strongly ab=
out *which* solution is chosen, but since we have multiple non-interoperabl=
e BUNDLE implementations, I would like the BUNDLE draft to add a sentence c=
hoosing one approach, which can be referenced in JSEP Section 7.=C2=A0</spa=
n></pre></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On =
Tue, Oct 4, 2016 at 6:05 AM, Harald Alvestrand <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestrand.no</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"><span class=3D"">Den =
04. okt. 2016 14:29, skrev Christer Holmberg:<br>
&gt; Hi,<br>
&gt;<br>
&gt;&gt;&gt; I still question why the low-level RTP demuxing shall be descr=
ibed in<br>
&gt;&gt;&gt; JSEP, instead of e.g., BUNDLE.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Keep in mind that all BUNDLE implementations may not implement=
 JSEP,<br>
&gt;&gt;&gt; but I<br>
&gt;&gt;&gt; still think we want the RTP demuxing to be done in the same wa=
y?<br>
&gt;&gt;<br>
&gt;&gt; Does BUNDLE say how to handle RID?<br>
&gt;<br>
&gt; No. Cullen wanted more text, though, but I am not sure whether that<br=
>
&gt; covered RID.<br>
&gt;<br>
&gt;&gt; Seriously, please get BUNDLE out the door and don&#39;t tangle it =
in any<br>
&gt;&gt; more new stuff.<br>
&gt;<br>
&gt; If that was meant to me, I will pretend I didn=C2=B9t see it=C5=A0<br>
<br>
</span>It&#39;s actually a grave concern of mine - more addressed to the ch=
airs of<br>
the relevant WGs than to anyone else.<br>
<br>
Any open document is a magnet for new functionality, and any new<br>
functionality delays the document&#39;s publication.<br>
<br>
The bundle document, with its associated changes in registration<br>
procedures for SDP attributes (the mux category), needs to be seen as<br>
stable. RFC publication is the best way to make it so.<br>
<br>
Datatracker says we&#39;ve been working on this document since 2011. It&#39=
;s<br>
now at version -32, and was in state &quot;Minor updates needed to address<=
br>
WGLC comments&quot; in May.<br>
<br>
If we can segregate new requests for functionality into other documents,<br=
>
I think that is a win.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Harald<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
______________________________<wbr>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/rtcweb</a><br=
>
</div></div></blockquote></div><br></div>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Bernard Aboba <bernard.aboba@gmail.com>, Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] JSEP: RTP demux algorithms
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Subject: Re: [rtcweb] JSEP: RTP demux algorithms
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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WG,

After discussion in AVTEXT and MMUSIC regarding the inclusion of MID and 
RID as SDES items that this do exposes labels that previously only have 
existed in the signalling plane in the media plane. And especially in 
the RTP header extensions, where even if the media payload is encrypted 
the header extension is not encrypted.

The risk with this is primarily a privacy and fingerprinting risk. And 
the proposed mitgation is encryption of the RTP header extensions in 
both the bundle and avtext-rid documents.

This leads to the conclusion that for RTCWeb, we must consider to act on 
these recommendations and decide on which implementation and usage 
requirement the protection of these field should have.

My proposal is that implementation and use of RFC6904 encryption of the 
RTP header extensions are REQUIRED. For RTCP it is actually unclear if 
there is mandatory to use encrypted SRTCP. I think it should be required 
and that can be clarified in Section 5.5 of 
draft-ietf-rtcweb-security-arch.


Opinions?


Cheers

Magnus Westerlund

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


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but but!! sorry for missing this:

i said it would be fair to cite the paper that identifies the dscp blackholi=
ng problem; you said ok, if i give you the xml bit for it; i did; you still d=
on't cite it. please do.

Sent from my iPhone

> On 4. okt. 2016, at 11.47, Harald Alvestrand <harald@alvestrand.no> wrote:=

>=20
>=20
> I believe this version closes all post-IESG open issues on -transports.
>=20
> Thank you for your consideration.
>=20
> -------- Videresendt melding --------
> Emne: New Version Notification for draft-ietf-rtcweb-transports-16.txt
> Dato: Tue, 04 Oct 2016 01:42:30 -0700
> Fra: internet-drafts@ietf.org
> Til: Harald Alvestrand <harald@alvestrand.no>
>=20
>=20
> A new version of I-D, draft-ietf-rtcweb-transports-16.txt
> has been successfully submitted by Harald Alvestrand and posted to the
> IETF repository.
>=20
> Name:        draft-ietf-rtcweb-transports
> Revision:    16
> Title:        Transports for WebRTC
> Document date:    2016-10-04
> Group:        rtcweb
> Pages:        20
> URL:
> https://www.ietf.org/internet-drafts/draft-ietf-rtcweb-transports-16.txt
> Status:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/
> Htmlized:       https://tools.ietf.org/html/draft-ietf-rtcweb-transports-1=
6
> Diff:
> https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-transports-16
>=20
> Abstract:
>   This document describes the data transport protocols used by WebRTC,
>   including the protocols used for interaction with intermediate boxes
>   such as firewalls, relays and NAT boxes.
>=20
>=20
>=20
>=20
> Please note that it may take a couple of minutes from the time of submissi=
on
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> The IETF Secretariat
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Thu Oct  6 12:07:49 2016
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To: Michael Welzl <michawe@ifi.uio.no>
References: <147557055077.12851.1347339442333163275.idtracker@ietfa.amsl.com> <b471f3cb-d330-aa25-25d2-ddd945046f27@alvestrand.no> <3B4EEF3A-AF5E-4DA6-8142-BDE0B6E7E075@ifi.uio.no>
From: Harald Alvestrand <harald@alvestrand.no>
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Date: Thu, 6 Oct 2016 21:07:40 +0200
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Subject: Re: [rtcweb] Fwd: New Version Notification for draft-ietf-rtcweb-transports-16.txt
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On 10/06/2016 04:09 PM, Michael Welzl wrote:
> but but!! sorry for missing this:
>
> i said it would be fair to cite the paper that identifies the dscp blackholing problem; you said ok, if i give you the xml bit for it; i did; you still don't cite it. please do.

Oops. I didn't file an issue for it, which I should have done.
So I forgot.

>
> Sent from my iPhone
>
>> On 4. okt. 2016, at 11.47, Harald Alvestrand <harald@alvestrand.no> wrote:
>>
>>
>> I believe this version closes all post-IESG open issues on -transports.
>>
>> Thank you for your consideration.
>>
>> -------- Videresendt melding --------
>> Emne: New Version Notification for draft-ietf-rtcweb-transports-16.txt
>> Dato: Tue, 04 Oct 2016 01:42:30 -0700
>> Fra: internet-drafts@ietf.org
>> Til: Harald Alvestrand <harald@alvestrand.no>
>>
>>
>> A new version of I-D, draft-ietf-rtcweb-transports-16.txt
>> has been successfully submitted by Harald Alvestrand and posted to the
>> IETF repository.
>>
>> Name:        draft-ietf-rtcweb-transports
>> Revision:    16
>> Title:        Transports for WebRTC
>> Document date:    2016-10-04
>> Group:        rtcweb
>> Pages:        20
>> URL:
>> https://www.ietf.org/internet-drafts/draft-ietf-rtcweb-transports-16.txt
>> Status:
>> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/
>> Htmlized:       https://tools.ietf.org/html/draft-ietf-rtcweb-transports-16
>> Diff:
>> https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-16
>>
>> Abstract:
>>   This document describes the data transport protocols used by WebRTC,
>>   including the protocols used for interaction with intermediate boxes
>>   such as firewalls, relays and NAT boxes.
>>
>>
>>
>>
>> Please note that it may take a couple of minutes from the time of submission
>> until the htmlized version and diff are available at tools.ietf.org.
>>
>> The IETF Secretariat
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


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From: Justin Uberti <juberti@google.com>
Date: Fri, 7 Oct 2016 10:11:38 -0700
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Subject: [rtcweb] Clarification on JSEP addTrack algorithm change
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--089e0122f5480db9cb053e4982f4
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At IETF 96, we discussed a modification
<https://docs.google.com/presentation/d/1Ay80ztS2eMtnKWcUA7Q91uai9ICQTgrfMbZsjquQpN8/edit#slide=id.g155b90bad5_0_101>
to how addTrack would match new MSTs to m= sections (details at the link).

At the time, I proposed the following text:
*If the PeerConnection is in the have-remote-offer state, the track will be
attached to the first compatible transceiver that was created by
setRemoteDescription and does not have a local track. Otherwise, a new
transceiver will be created.*

The intent here was to only perform automatic track matching on m= sections
that were added by a newly received offer/reoffer (and not on existing
a=recvonly m= sections). However, Peter Thatcher pointed out that this text
was somewhat ambiguous and would allow an existing a=recvonly m= section to
be matched during a reoffer.

To address this, Peter and I have agreed on the following revised text:
*If the PeerConnection is in the have-remote-offer state, the track will be
attached to the first compatible transceiver that was created by the most
recent setRemoteDescription and does not have a local track. Otherwise, a
new transceiver will be created.*

This is being tracked in JSEP issue 288
<https://github.com/rtcweb-wg/jsep/issues/288>.

--089e0122f5480db9cb053e4982f4
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<div dir=3D"ltr">At IETF 96, we discussed a <a href=3D"https://docs.google.=
com/presentation/d/1Ay80ztS2eMtnKWcUA7Q91uai9ICQTgrfMbZsjquQpN8/edit#slide=
=3Did.g155b90bad5_0_101">modification</a> to how addTrack would match new M=
STs to m=3D sections (details at the link).<div><br></div><div>At the time,=
 I proposed the following text:</div><div><div><i>If the PeerConnection is =
in the have-remote-offer state, the track will be attached to the first com=
patible transceiver that was created by setRemoteDescription and does not h=
ave a local track. Otherwise, a new transceiver will be created.</i></div><=
/div><div><i><br></i></div><div>The intent here was to only perform automat=
ic track matching on m=3D sections that were added by a newly received offe=
r/reoffer (and not on existing a=3Drecvonly m=3D sections). However, Peter =
Thatcher pointed out that this text was somewhat ambiguous and would allow =
an existing a=3Drecvonly m=3D section to be matched during a reoffer.</div>=
<div><br></div><div>To address this, Peter and I have agreed on the followi=
ng revised text:</div><div><i>If the PeerConnection is in the have-remote-o=
ffer state, the track will be attached to the first compatible transceiver =
that was created by<b> the most recent</b> setRemoteDescription and does no=
t have a local track. Otherwise, a new transceiver will be created.</i><br>=
</div><div><i><br></i></div><div>This is being tracked in JSEP=C2=A0<a href=
=3D"https://github.com/rtcweb-wg/jsep/issues/288">issue 288</a>.</div><div>=
<br></div><div><br></div></div>

--089e0122f5480db9cb053e4982f4--


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From: Cullen Jennings <fluffy@iii.ca>
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Cc: RTCWeb IETF <rtcweb@ietf.org>
Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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How are these a significant fingerprinting problem ?


> On Oct 6, 2016, at 7:55 AM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:
>=20
> WG,
>=20
> After discussion in AVTEXT and MMUSIC regarding the inclusion of MID =
and RID as SDES items that this do exposes labels that previously only =
have existed in the signalling plane in the media plane. And especially =
in the RTP header extensions, where even if the media payload is =
encrypted the header extension is not encrypted.
>=20
> The risk with this is primarily a privacy and fingerprinting risk. And =
the proposed mitgation is encryption of the RTP header extensions in =
both the bundle and avtext-rid documents.
>=20
> This leads to the conclusion that for RTCWeb, we must consider to act =
on these recommendations and decide on which implementation and usage =
requirement the protection of these field should have.
>=20
> My proposal is that implementation and use of RFC6904 encryption of =
the RTP header extensions are REQUIRED. For RTCP it is actually unclear =
if there is mandatory to use encrypted SRTCP. I think it should be =
required and that can be clarified in Section 5.5 of =
draft-ietf-rtcweb-security-arch.
>=20
>=20
> Opinions?
>=20
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Eric Rescorla <ekr@rtfm.com>
Date: Fri, 7 Oct 2016 10:52:54 -0700
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Subject: Re: [rtcweb] Clarification on JSEP addTrack algorithm change
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This is fine with me, but I think we should modify the *text* to make clear
what the intent is.

Perhaps something like this:
"For example, if a PeerConnection receives an offer with an m= section that
is send-recv
a call to addTrack() prior to calling createAnswer() can attach a track to
that transceiver.
However, after createAnswer() and setLocalDescription() have been called,
any call
to addTrack() will create a new transceiver even if the new track would
otherwise be compatible"



On Fri, Oct 7, 2016 at 10:11 AM, Justin Uberti <juberti@google.com> wrote:

> At IETF 96, we discussed a modification
> <https://docs.google.com/presentation/d/1Ay80ztS2eMtnKWcUA7Q91uai9ICQTgrfMbZsjquQpN8/edit#slide=id.g155b90bad5_0_101>
> to how addTrack would match new MSTs to m= sections (details at the link).
>
> At the time, I proposed the following text:
> *If the PeerConnection is in the have-remote-offer state, the track will
> be attached to the first compatible transceiver that was created by
> setRemoteDescription and does not have a local track. Otherwise, a new
> transceiver will be created.*
>
> The intent here was to only perform automatic track matching on m=
> sections that were added by a newly received offer/reoffer (and not on
> existing a=recvonly m= sections). However, Peter Thatcher pointed out that
> this text was somewhat ambiguous and would allow an existing a=recvonly m=
> section to be matched during a reoffer.
>
> To address this, Peter and I have agreed on the following revised text:
> *If the PeerConnection is in the have-remote-offer state, the track will
> be attached to the first compatible transceiver that was created by the
> most recent setRemoteDescription and does not have a local track.
> Otherwise, a new transceiver will be created.*
>
> This is being tracked in JSEP issue 288
> <https://github.com/rtcweb-wg/jsep/issues/288>.
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">This is fine with me, but I think we should modify the *te=
xt* to make clear what the intent is.<div><br></div><div>Perhaps something =
like this:</div><div>&quot;For example, if a PeerConnection receives an off=
er with an m=3D section that is send-recv</div><div>a call to addTrack() pr=
ior to calling createAnswer() can attach a track to that transceiver.</div>=
<div>However, after createAnswer() and setLocalDescription() have been call=
ed, any call</div><div>to addTrack() will create a new transceiver even if =
the new track would otherwise be compatible&quot;</div><div><br></div><div>=
<br></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">O=
n Fri, Oct 7, 2016 at 10:11 AM, Justin Uberti <span dir=3D"ltr">&lt;<a href=
=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</a>&gt;=
</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .=
8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">At IETF 9=
6, we discussed a <a href=3D"https://docs.google.com/presentation/d/1Ay80zt=
S2eMtnKWcUA7Q91uai9ICQTgrfMbZsjquQpN8/edit#slide=3Did.g155b90bad5_0_101" ta=
rget=3D"_blank">modification</a> to how addTrack would match new MSTs to m=
=3D sections (details at the link).<div><br></div><div>At the time, I propo=
sed the following text:</div><div><div><i>If the PeerConnection is in the h=
ave-remote-offer state, the track will be attached to the first compatible =
transceiver that was created by setRemoteDescription and does not have a lo=
cal track. Otherwise, a new transceiver will be created.</i></div></div><di=
v><i><br></i></div><div>The intent here was to only perform automatic track=
 matching on m=3D sections that were added by a newly received offer/reoffe=
r (and not on existing a=3Drecvonly m=3D sections). However, Peter Thatcher=
 pointed out that this text was somewhat ambiguous and would allow an exist=
ing a=3Drecvonly m=3D section to be matched during a reoffer.</div><div><br=
></div><div>To address this, Peter and I have agreed on the following revis=
ed text:</div><div><i>If the PeerConnection is in the have-remote-offer sta=
te, the track will be attached to the first compatible transceiver that was=
 created by<b> the most recent</b> setRemoteDescription and does not have a=
 local track. Otherwise, a new transceiver will be created.</i><br></div><d=
iv><i><br></i></div><div>This is being tracked in JSEP=C2=A0<a href=3D"http=
s://github.com/rtcweb-wg/jsep/issues/288" target=3D"_blank">issue 288</a>.<=
/div><div><br></div><div><br></div></div>
<br>______________________________<wbr>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/rtcweb</a><br=
>
<br></blockquote></div><br></div>

--001a114926a0d48637053e4a1639--


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From: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [rtcweb] [MMUSIC] Review of draft-ietf-mmusic-sdp-bundle-negotiation-32 - Magnus' comments - MID security
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> On Oct 7, 2016, at 1:41 AM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:
>=20
> Den 2016-10-06 kl. 23:59, skrev Cullen Jennings:
>>=20
>> I'm not getting the problem here and mandating 6904 is not an easy
>> thing to do.
>=20
> So, lets be clear on what I have proposed. For bundle in general it is =
RECOMMENDED to encrypt MIDs in RTP header extensions. I have also =
proposed that for RTCWeb implementation and use should be mandated. I =
was not clear in what applies when an WebRTC endpoint talks to a legacy =
endpoint what is expected here. I guess this is the thorny issue.
>=20
>>=20
>> I'm not getting what the issue is here. If mid are random, or are
>> just a count of n'th m-line in the sdp, what is the problem with
>> exposing them to people are getting the media?
>=20
> The issue is that these identifiers are going from having only been =
visiable in the signalling messages to be visible also on the RTP media =
plane. And without RFC6904 even if one uses SRTP, these values are =
exposed. This have several different effects as I see it. So if the =
generation algorithm are not strictly defined, the implementation gets =
possible to fingerprint. In addition the device gets to be fingerprinted =
based on the number of streams offered and their identifiers. Thus, it =
is not only what is actually used that is exposed, but what was =
originally offered, i.e. the full offer leaks into unencrypted domain on =
the media plane. Note, this later can't be dealt with any mid id =
creation rules, it will still leak. Thus devices that have more exotic =
configurations when it comes to media streams, i.e. number of cameras =
etc this can still be fingerprinted by passive attacks from third =
parties.
>=20
> It is the passive attack from third parties that has me worried here. =
A peer in the signalling has massively more powerful fingerprinting, but =
that requires one to be in the signalling context, not passively observe =
traffic that goes past.
>=20

I'm still not getting the fingerprint problem. How is this different =
than PT and SSRC. Yes, I agree the MID and RID - which were always =
intended to go in the RTP packet need to be generated not to have =
sensitive information in them.=20


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From: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] Clarification on JSEP addTrack algorithm change
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Upon reflection I think this is clear enough

On Fri, Oct 7, 2016 at 10:52 AM, Eric Rescorla <ekr@rtfm.com> wrote:

> This is fine with me, but I think we should modify the *text* to make
> clear what the intent is.
>
> Perhaps something like this:
> "For example, if a PeerConnection receives an offer with an m= section
> that is send-recv
> a call to addTrack() prior to calling createAnswer() can attach a track to
> that transceiver.
> However, after createAnswer() and setLocalDescription() have been called,
> any call
> to addTrack() will create a new transceiver even if the new track would
> otherwise be compatible"
>
>
>
> On Fri, Oct 7, 2016 at 10:11 AM, Justin Uberti <juberti@google.com> wrote:
>
>> At IETF 96, we discussed a modification
>> <https://docs.google.com/presentation/d/1Ay80ztS2eMtnKWcUA7Q91uai9ICQTgrfMbZsjquQpN8/edit#slide=id.g155b90bad5_0_101>
>> to how addTrack would match new MSTs to m= sections (details at the link).
>>
>> At the time, I proposed the following text:
>> *If the PeerConnection is in the have-remote-offer state, the track will
>> be attached to the first compatible transceiver that was created by
>> setRemoteDescription and does not have a local track. Otherwise, a new
>> transceiver will be created.*
>>
>> The intent here was to only perform automatic track matching on m=
>> sections that were added by a newly received offer/reoffer (and not on
>> existing a=recvonly m= sections). However, Peter Thatcher pointed out that
>> this text was somewhat ambiguous and would allow an existing a=recvonly m=
>> section to be matched during a reoffer.
>>
>> To address this, Peter and I have agreed on the following revised text:
>> *If the PeerConnection is in the have-remote-offer state, the track will
>> be attached to the first compatible transceiver that was created by the
>> most recent setRemoteDescription and does not have a local track.
>> Otherwise, a new transceiver will be created.*
>>
>> This is being tracked in JSEP issue 288
>> <https://github.com/rtcweb-wg/jsep/issues/288>.
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>

--001a1142f87a1ce10e053e4aef0c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Upon reflection I think this is clear enough</div><div cla=
ss=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Oct 7, 2016 at 10=
:52 AM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com"=
 target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div dir=3D"ltr">This is fine with me, but I think we should =
modify the *text* to make clear what the intent is.<div><br></div><div>Perh=
aps something like this:</div><div>&quot;For example, if a PeerConnection r=
eceives an offer with an m=3D section that is send-recv</div><div>a call to=
 addTrack() prior to calling createAnswer() can attach a track to that tran=
sceiver.</div><div>However, after createAnswer() and setLocalDescription() =
have been called, any call</div><div>to addTrack() will create a new transc=
eiver even if the new track would otherwise be compatible&quot;</div><div><=
br></div><div><br></div></div><div class=3D"gmail_extra"><br><div class=3D"=
gmail_quote"><div><div class=3D"h5">On Fri, Oct 7, 2016 at 10:11 AM, Justin=
 Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=
=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<br></div></div><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex"><div><div class=3D"h5"><div dir=3D"ltr">At IETF 96,=
 we discussed a <a href=3D"https://docs.google.com/presentation/d/1Ay80ztS2=
eMtnKWcUA7Q91uai9ICQTgrfMbZsjquQpN8/edit#slide=3Did.g155b90bad5_0_101" targ=
et=3D"_blank">modification</a> to how addTrack would match new MSTs to m=3D=
 sections (details at the link).<div><br></div><div>At the time, I proposed=
 the following text:</div><div><div><i>If the PeerConnection is in the have=
-remote-offer state, the track will be attached to the first compatible tra=
nsceiver that was created by setRemoteDescription and does not have a local=
 track. Otherwise, a new transceiver will be created.</i></div></div><div><=
i><br></i></div><div>The intent here was to only perform automatic track ma=
tching on m=3D sections that were added by a newly received offer/reoffer (=
and not on existing a=3Drecvonly m=3D sections). However, Peter Thatcher po=
inted out that this text was somewhat ambiguous and would allow an existing=
 a=3Drecvonly m=3D section to be matched during a reoffer.</div><div><br></=
div><div>To address this, Peter and I have agreed on the following revised =
text:</div><div><i>If the PeerConnection is in the have-remote-offer state,=
 the track will be attached to the first compatible transceiver that was cr=
eated by<b> the most recent</b> setRemoteDescription and does not have a lo=
cal track. Otherwise, a new transceiver will be created.</i><br></div><div>=
<i><br></i></div><div>This is being tracked in JSEP=C2=A0<a href=3D"https:/=
/github.com/rtcweb-wg/jsep/issues/288" target=3D"_blank">issue 288</a>.</di=
v><div><br></div><div><br></div></div>
<br></div></div>______________________________<wbr>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/rtcweb</a><br=
>
<br></blockquote></div><br></div>
</blockquote></div><br></div>

--001a1142f87a1ce10e053e4aef0c--


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To: Cullen Jennings <fluffy@iii.ca>
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Cc: RTCWeb IETF <rtcweb@ietf.org>
Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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I don't see how snooping the MID and RID would provide info that could not b=
e obtained in other ways. For example, an observer can tell audio from video=
 traffic just by looking at packet sizes. Similarly, simulcast streams will o=
riginate from different SSRCs so no need to snoop the RID to figure out that=
 there are multiple streams being sent (or even which ones are related since=
 traffic will be correlated).

> On Oct 7, 2016, at 10:41, Cullen Jennings <fluffy@iii.ca> wrote:
>=20
>=20
> How are these a significant fingerprinting problem ?
>=20
>=20
>> On Oct 6, 2016, at 7:55 AM, Magnus Westerlund <magnus.westerlund@ericsson=
.com> wrote:
>>=20
>> WG,
>>=20
>> After discussion in AVTEXT and MMUSIC regarding the inclusion of MID and R=
ID as SDES items that this do exposes labels that previously only have exist=
ed in the signalling plane in the media plane. And especially in the RTP hea=
der extensions, where even if the media payload is encrypted the header exte=
nsion is not encrypted.
>>=20
>> The risk with this is primarily a privacy and fingerprinting risk. And th=
e proposed mitgation is encryption of the RTP header extensions in both the b=
undle and avtext-rid documents.
>>=20
>> This leads to the conclusion that for RTCWeb, we must consider to act on t=
hese recommendations and decide on which implementation and usage requiremen=
t the protection of these field should have.
>>=20
>> My proposal is that implementation and use of RFC6904 encryption of the R=
TP header extensions are REQUIRED. For RTCP it is actually unclear if there i=
s mandatory to use encrypted SRTCP. I think it should be required and that c=
an be clarified in Section 5.5 of draft-ietf-rtcweb-security-arch.
>>=20
>>=20
>> Opinions?
>>=20
>>=20
>> Cheers
>>=20
>> Magnus Westerlund
>>=20
>> ----------------------------------------------------------------------
>> Services, Media and Network features, Ericsson Research EAB/TXM
>> ----------------------------------------------------------------------
>> Ericsson AB                 | Phone  +46 10 7148287
>> F=C3=A4r=C3=B6gatan 6                 | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Bernard Aboba <bernard.aboba@gmail.com>
Thread-Topic: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
Thread-Index: AQHSIRSvyLgtO7YE+0OQTKhN6ScPbaCeosTH
Date: Sat, 8 Oct 2016 14:44:25 +0000
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References: <e536bad2-08b1-4f77-8c75-6bc3b639c398@ericsson.com> <B6ECFC24-F28E-4E35-9437-B7DACB41EF69@iii.ca>, <DD1447CA-29F2-44FF-B08F-3CC0814C9748@gmail.com>
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Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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MID and RID are similar to PT in this regard. They are all arbitrary values=
 that only have semantic meaning when combined with a media description in =
external signaling. Any genuine concern over privacy or fingerprinting issu=
es with MID and RID should first consider PT. I'm unaware of any concerns e=
xpressed over unencrypted PT.

Note that it is possible to use encrypted SRTCP to convey MID and RID witho=
ut including them in SRTP packets. One proposed version of the RTP demux ru=
les has explicitly noted this approach of using RTCP to latch MID/RID to an=
 SSRC even if RTP does not contain those header extensions for that SSRC.=20

Mandating encrypted SRTCP seems more palatable than mandating encrypted RTP=
 header extensions. But I don't think a good case has been made yet for eit=
her, especially considering PT is always unencrypted.=20

Mo

On Oct 7, 2016, at 11:33 PM, Bernard Aboba <bernard.aboba@gmail.com> wrote:

I don't see how snooping the MID and RID would provide info that could not =
be obtained in other ways. For example, an observer can tell audio from vid=
eo traffic just by looking at packet sizes. Similarly, simulcast streams wi=
ll originate from different SSRCs so no need to snoop the RID to figure out=
 that there are multiple streams being sent (or even which ones are related=
 since traffic will be correlated).

> On Oct 7, 2016, at 10:41, Cullen Jennings <fluffy@iii.ca> wrote:
>=20
>=20
> How are these a significant fingerprinting problem ?
>=20
>=20
>> On Oct 6, 2016, at 7:55 AM, Magnus Westerlund <magnus.westerlund@ericsso=
n.com> wrote:
>>=20
>> WG,
>>=20
>> After discussion in AVTEXT and MMUSIC regarding the inclusion of MID and=
 RID as SDES items that this do exposes labels that previously only have ex=
isted in the signalling plane in the media plane. And especially in the RTP=
 header extensions, where even if the media payload is encrypted the header=
 extension is not encrypted.
>>=20
>> The risk with this is primarily a privacy and fingerprinting risk. And t=
he proposed mitgation is encryption of the RTP header extensions in both th=
e bundle and avtext-rid documents.
>>=20
>> This leads to the conclusion that for RTCWeb, we must consider to act on=
 these recommendations and decide on which implementation and usage require=
ment the protection of these field should have.
>>=20
>> My proposal is that implementation and use of RFC6904 encryption of the =
RTP header extensions are REQUIRED. For RTCP it is actually unclear if ther=
e is mandatory to use encrypted SRTCP. I think it should be required and th=
at can be clarified in Section 5.5 of draft-ietf-rtcweb-security-arch.
>>=20
>>=20
>> Opinions?
>>=20
>>=20
>> Cheers
>>=20
>> Magnus Westerlund
>>=20
>> ----------------------------------------------------------------------
>> Services, Media and Network features, Ericsson Research EAB/TXM
>> ----------------------------------------------------------------------
>> Ericsson AB                 | Phone  +46 10 7148287
>> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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


From nobody Sun Oct  9 03:24:27 2016
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>, Bernard Aboba <bernard.aboba@gmail.com>
Thread-Topic: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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Hi,

On the MMUSIC list Jonathan L informed the community about the following te=
xt in RFC 7904 (SDES header extensions):

   "In RTP sessions where any type of confidentiality protection is
   enabled for RTCP, the SDES item header extensions MUST also be
   protected."

So, *IF* we assume that "any type of confidentiality protection" is enabled=
 for RTCP, I guess the answer is pretty clear, or?

Regards,

Christer


-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Mo Zanaty (mzana=
ty)
Sent: 08 October 2016 17:44
To: Bernard Aboba <bernard.aboba@gmail.com>
Cc: RTCWeb IETF <rtcweb@ietf.org>
Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID=
 and RID SDES items

MID and RID are similar to PT in this regard. They are all arbitrary values=
 that only have semantic meaning when combined with a media description in =
external signaling. Any genuine concern over privacy or fingerprinting issu=
es with MID and RID should first consider PT. I'm unaware of any concerns e=
xpressed over unencrypted PT.

Note that it is possible to use encrypted SRTCP to convey MID and RID witho=
ut including them in SRTP packets. One proposed version of the RTP demux ru=
les has explicitly noted this approach of using RTCP to latch MID/RID to an=
 SSRC even if RTP does not contain those header extensions for that SSRC.=20

Mandating encrypted SRTCP seems more palatable than mandating encrypted RTP=
 header extensions. But I don't think a good case has been made yet for eit=
her, especially considering PT is always unencrypted.=20

Mo

On Oct 7, 2016, at 11:33 PM, Bernard Aboba <bernard.aboba@gmail.com> wrote:

I don't see how snooping the MID and RID would provide info that could not =
be obtained in other ways. For example, an observer can tell audio from vid=
eo traffic just by looking at packet sizes. Similarly, simulcast streams wi=
ll originate from different SSRCs so no need to snoop the RID to figure out=
 that there are multiple streams being sent (or even which ones are related=
 since traffic will be correlated).

> On Oct 7, 2016, at 10:41, Cullen Jennings <fluffy@iii.ca> wrote:
>=20
>=20
> How are these a significant fingerprinting problem ?
>=20
>=20
>> On Oct 6, 2016, at 7:55 AM, Magnus Westerlund <magnus.westerlund@ericsso=
n.com> wrote:
>>=20
>> WG,
>>=20
>> After discussion in AVTEXT and MMUSIC regarding the inclusion of MID and=
 RID as SDES items that this do exposes labels that previously only have ex=
isted in the signalling plane in the media plane. And especially in the RTP=
 header extensions, where even if the media payload is encrypted the header=
 extension is not encrypted.
>>=20
>> The risk with this is primarily a privacy and fingerprinting risk. And t=
he proposed mitgation is encryption of the RTP header extensions in both th=
e bundle and avtext-rid documents.
>>=20
>> This leads to the conclusion that for RTCWeb, we must consider to act on=
 these recommendations and decide on which implementation and usage require=
ment the protection of these field should have.
>>=20
>> My proposal is that implementation and use of RFC6904 encryption of the =
RTP header extensions are REQUIRED. For RTCP it is actually unclear if ther=
e is mandatory to use encrypted SRTCP. I think it should be required and th=
at can be clarified in Section 5.5 of draft-ietf-rtcweb-security-arch.
>>=20
>>=20
>> Opinions?
>>=20
>>=20
>> Cheers
>>=20
>> Magnus Westerlund
>>=20
>> ---------------------------------------------------------------------
>> - Services, Media and Network features, Ericsson Research EAB/TXM
>> ----------------------------------------------------------------------
>> Ericsson AB                 | Phone  +46 10 7148287
>> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>> ---------------------------------------------------------------------
>> -
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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

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


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To: Christer Holmberg <christer.holmberg@ericsson.com>, "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>, Bernard Aboba <bernard.aboba@gmail.com>
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Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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Den 2016-10-09 kl. 12:24, skrev Christer Holmberg:
> Hi,
>
> On the MMUSIC list Jonathan L informed the community about the following text in RFC 7904 (SDES header extensions):
>
>    "In RTP sessions where any type of confidentiality protection is
>    enabled for RTCP, the SDES item header extensions MUST also be
>    protected."
>
> So, *IF* we assume that "any type of confidentiality protection" is enabled for RTCP, I guess the answer is pretty clear, or?
>

Yes, from my perspective this is settled. RFC 7904 forces encryption of 
the MID value in an WebRTC context. If the RTCWeb WG do not agree with 
this, then it needs to work on changing RFC 7904. I do not support 
chaning that RFC's conclusion that information that is protected in RTCP 
needs to be protected also in RTP headers.

When it comes to the information leakage and the fingerprinting 
possibilities of SSRCs, PTs etc, it is present and getting worse in 
multi-stream systems.

So when you had a single media stream per endpoint and that used a 
random SSRC and one PT value during the life-time of the RTP session, 
there is very little information leakage here. The codec possibly leaks 
through PT, but the packet length, at least for audio leaks that 
information anyway.

However, when one starts to have a endpoint that uses multiple media 
streams, and certain configurations of RTX and FEC, then the publicly 
visable fingerprinting surface from RTP streams goes up. Now both 
implementation choices as well as device hardware configurations starts 
to show in the unencrypted RTP headers. So MID and RID is one additional 
parameter that increases the potential for fingerprinting here.

So what is the reason given that we protect the RTP payload and the RTCP 
packets with encryption, to not also protected the RTP header extensions?

Cheers

Magnus Westerlund

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


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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I think this is huge complexity that is not currently supported and not nee=
ded. How about we generate the RID such that the RID for the n'th m line is=
 n. Or generate it as a random number. Do you see any problem with this?

I think this removes all the fingerprinting issues. I'm fine with advice th=
at say how to generate the RID in a way that is privacy sensitive but I'm n=
ot a fan of making RID require support for RTP header encryption.=20



> On Oct 6, 2016, at 7:55 AM, Magnus Westerlund <magnus.westerlund@ericsson=
.com> wrote:
>=20
> WG,
>=20
> After discussion in AVTEXT and MMUSIC regarding the inclusion of MID and =
RID as SDES items that this do exposes labels that previously only have exi=
sted in the signalling plane in the media plane. And especially in the RTP =
header extensions, where even if the media payload is encrypted the header =
extension is not encrypted.
>=20
> The risk with this is primarily a privacy and fingerprinting risk. And th=
e proposed mitgation is encryption of the RTP header extensions in both the=
 bundle and avtext-rid documents.
>=20
> This leads to the conclusion that for RTCWeb, we must consider to act on =
these recommendations and decide on which implementation and usage requirem=
ent the protection of these field should have.
>=20
> My proposal is that implementation and use of RFC6904 encryption of the R=
TP header extensions are REQUIRED. For RTCP it is actually unclear if there=
 is mandatory to use encrypted SRTCP. I think it should be required and tha=
t can be clarified in Section 5.5 of draft-ietf-rtcweb-security-arch.
>=20
>=20
> Opinions?
>=20
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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> On Oct 10, 2016, at 3:05 AM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:
>=20
> Den 2016-10-09 kl. 12:24, skrev Christer Holmberg:
>> Hi,
>>=20
>> On the MMUSIC list Jonathan L informed the community about the =
following text in RFC 7904 (SDES header extensions):
>>=20
>>   "In RTP sessions where any type of confidentiality protection is
>>   enabled for RTCP, the SDES item header extensions MUST also be
>>   protected."
>>=20

Just one note - I think you mean RFC 7941 here.=20


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Cullen Jennings <fluffy@iii.ca>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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>> On the MMUSIC list Jonathan L informed the community about the following=
 text in >> RFC 7904 (SDES header extensions):
>>=20
>>   "In RTP sessions where any type of confidentiality protection is
>>   enabled for RTCP, the SDES item header extensions MUST also be
>>   protected."
>=20
>
> Just one note - I think you mean RFC 7941 here.=20

Correct :)

Regards,

Christer


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To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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Den 2016-10-13 kl. 20:12, skrev Cullen Jennings (fluffy):
>
> I think this is huge complexity that is not currently supported and
> not needed. How about we generate the RID such that the RID for the
> n'th m line is n. Or generate it as a random number. Do you see any
> problem with this?
>
> I think this removes all the fingerprinting issues. I'm fine with
> advice that say how to generate the RID in a way that is privacy
> sensitive but I'm not a fan of making RID require support for RTP
> header encryption.
>

So, let disregard the fingerprinting issue for now. It is clear that I 
am in the rough and that this whole issue would benefit from a broad 
review on what we leak when using ICE and RTP/RTCP.

What I don't understand here is that people have argued and agreed that 
we should encrypt the RTP and RTCP packets per default. Then when we 
comes to RTP header extensions we are no longer ready to provide that 
protection per default by ensuring RFC6904 implementation support. So 
RID and MID may not be the most sensitive fields if it has well 
constructed values. Then we have the audio level extensions where Client 
to mixer is RECOMMENDED to be implemented and REQUIRES RFC6904.

So what is the issue, is it that you currently lack RFC6904 
implementations, and thus this require more implementation work, or are 
there additional reasons for wanting to expose this information?

To conclude, I personally think default usage of RFC6904 on header 
extensions would be a good thing. I currently see no reasons for 
changing the requirements that RFC 7941 puts on WebRTC, forcing the 
usage of RFC6904 encryption. If you want to something else, please 
provide a proposal for what that alternative would be.

Cheers

Magnus Westerlund


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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Date: Fri, 14 Oct 2016 13:08:05 +0000
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Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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> On Oct 14, 2016, at 2:26 AM, Magnus Westerlund <magnus.westerlund@ericsso=
n.com> wrote:
>=20
> Den 2016-10-13 kl. 20:12, skrev Cullen Jennings (fluffy):
>>=20
>> I think this is huge complexity that is not currently supported and
>> not needed. How about we generate the RID such that the RID for the
>> n'th m line is n. Or generate it as a random number. Do you see any
>> problem with this?
>>=20
>> I think this removes all the fingerprinting issues. I'm fine with
>> advice that say how to generate the RID in a way that is privacy
>> sensitive but I'm not a fan of making RID require support for RTP
>> header encryption.
>>=20
>=20
> So, let disregard the fingerprinting issue for now. It is clear that I am=
 in the rough and that this whole issue would benefit from a broad review o=
n what we leak when using ICE and RTP/RTCP.

That broad review was largely done as part of PERC.=20

>=20
> What I don't understand here is that people have argued and agreed that w=
e should encrypt the RTP and RTCP packets per default. Then when we comes t=
o RTP header extensions we are no longer ready to provide that protection p=
er default by ensuring RFC6904 implementation support. So RID and MID may n=
ot be the most sensitive fields if it has well constructed values. Then we =
have the audio level extensions where Client to mixer is RECOMMENDED to be =
implemented and REQUIRES RFC6904.

So the audio levels are recommended in WebRTC but this MID stuff is a very =
generic mechanism that will like be broadly used to deal with RTP lack of e=
xtensibility of size PT space. Lot of stuff does not implement audio levels=
 and some stuff that implements it does not encrypt it. The 6904 mechanism =
is not exactly elegant, not easy to implement, and doubles the size the hea=
ders which is not great for audio. I think it is fair to say we might do it=
 differently if doing it over again.=20

>=20
> So what is the issue, is it that you currently lack RFC6904 implementatio=
ns, and thus this require more implementation work, or are there additional=
 reasons for wanting to expose this information?

Yes this adds needless security complexity that people don't see the value =
in thus it becomes difficult to convince implementors to do it.

>=20
> To conclude, I personally think default usage of RFC6904 on header extens=
ions would be a good thing. I currently see no reasons for changing the req=
uirements that RFC 7941 puts on WebRTC, forcing the usage of RFC6904 encryp=
tion. If you want to something else, please provide a proposal for what tha=
t alternative would be.

Originally MID was not a SDES items, I'd propose just moving it back to a n=
ormal RTP Header. In looking at trying to use this in deployments, it's har=
d to see what value transporting it in RTCP as well as RTP provided.  When =
that was originally proposed, I don't think anyone realized that mean encry=
pting it and did not really care if it was normal or SDES as it was about t=
he same work. But if SDES means it needs to be encrypted, then I think we n=
eed to reconsider that.=20


>=20
> Cheers
>=20
> Magnus Westerlund
>=20


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From: Jonathan Lennox <jonathan@vidyo.com>
To: Cullen Jennings <fluffy@cisco.com>
Thread-Topic: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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Den 2016-10-14 kl. 15:08, skrev Cullen Jennings (fluffy):
>
>> On Oct 14, 2016, at 2:26 AM, Magnus Westerlund
>> <magnus.westerlund@ericsson.com> wrote:
>>
>> Den 2016-10-13 kl. 20:12, skrev Cullen Jennings (fluffy):
>>>
>>> I think this is huge complexity that is not currently supported
>>> and not needed. How about we generate the RID such that the RID
>>> for the n'th m line is n. Or generate it as a random number. Do
>>> you see any problem with this?
>>>
>>> I think this removes all the fingerprinting issues. I'm fine
>>> with advice that say how to generate the RID in a way that is
>>> privacy sensitive but I'm not a fan of making RID require support
>>> for RTP header encryption.
>>>
>>
>> So, let disregard the fingerprinting issue for now. It is clear
>> that I am in the rough and that this whole issue would benefit from
>> a broad review on what we leak when using ICE and RTP/RTCP.
>
> That broad review was largely done as part of PERC.
>

I do not agree that review happened in that context. I can't remember 
that we discussed fingerprinting at all. We quickly put any effort of 
privacy in regards to that the communication occurs out of scope. All 
the focus in PERC is to my understanding on maintaining confidentiality 
of the media content as well as preventing third party injection, 
modifications (other than sub selection of content)  and replay attacks.



>>
>> What I don't understand here is that people have argued and agreed
>> that we should encrypt the RTP and RTCP packets per default. Then
>> when we comes to RTP header extensions we are no longer ready to
>> provide that protection per default by ensuring RFC6904
>> implementation support. So RID and MID may not be the most
>> sensitive fields if it has well constructed values. Then we have
>> the audio level extensions where Client to mixer is RECOMMENDED to
>> be implemented and REQUIRES RFC6904.
>
> So the audio levels are recommended in WebRTC but this MID stuff is a
> very generic mechanism that will like be broadly used to deal with
> RTP lack of extensibility of size PT space. Lot of stuff does not
> implement audio levels and some stuff that implements it does not
> encrypt it. The 6904 mechanism is not exactly elegant, not easy to
> implement, and doubles the size the headers which is not great for
> audio. I think it is fair to say we might do it differently if doing
> it over again.
>

I think Jonathan had an excellent reply on this.

>>
>> So what is the issue, is it that you currently lack RFC6904
>> implementations, and thus this require more implementation work, or
>> are there additional reasons for wanting to expose this
>> information?
>
> Yes this adds needless security complexity that people don't see the
> value in thus it becomes difficult to convince implementors to do
> it.
>
>>
>> To conclude, I personally think default usage of RFC6904 on header
>> extensions would be a good thing. I currently see no reasons for
>> changing the requirements that RFC 7941 puts on WebRTC, forcing the
>> usage of RFC6904 encryption. If you want to something else, please
>> provide a proposal for what that alternative would be.
>
> Originally MID was not a SDES items, I'd propose just moving it back
> to a normal RTP Header. In looking at trying to use this in
> deployments, it's hard to see what value transporting it in RTCP as
> well as RTP provided.  When that was originally proposed, I don't
> think anyone realized that mean encrypting it and did not really care
> if it was normal or SDES as it was about the same work. But if SDES
> means it needs to be encrypted, then I think we need to reconsider
> that.
>

So, I think moving away from SDES item is a bad idea. As that would 
create additional implementation burden, rather than having the same 
processing for a number of SDES items of similar type. RID, MID and 
CNAME all need similar processing for initial startup handling. So 
moving RID out, while still having the other two just creates an 
additional cases to implement.

When it comes to RTCP inclusion of these SDES items I know of two reasons.

Reason one, is that one can actually use RTCP to send these for an SSRC 
prior to having RTP data to send for it. That enables one to have the 
association in place prior to actually having to handle media data.

Reason two is the saftey net RTCP provides. It enables a receiver to 
detect if its state is out of sync. Yes, this could be realized using 
periodic repeats in the RTP header extension also. However, that doesn't 
work if the SSRC currently don't have any data to send.

Considering the state that draft-ietf-avtext-rid is in. Deciding to 
change it to not be an SDES item is going be an equal if not greater 
hassle to actually pull it back to the WG to include a beef up of the 
security consdieration with an motivation why the rules in RFC7941 do 
not apply to it. Assuming that is what the consensus ends up being.

Cheers

Magnus Westerlund

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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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Date: Tue, 18 Oct 2016 13:28:50 +0000
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Cc: RTCWeb IETF <rtcweb@ietf.org>
Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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Hi,

draft-ietf-avtext-rid also talks about encryption, and references RFC
7201. Any input from the authors of that document? Adam? Suhas? Peter?


Regards,

Christer



On 17/10/16 11:46, "rtcweb on behalf of Magnus Westerlund"
<rtcweb-bounces@ietf.org on behalf of magnus.westerlund@ericsson.com>
wrote:

>Den 2016-10-14 kl. 15:08, skrev Cullen Jennings (fluffy):
>>
>>> On Oct 14, 2016, at 2:26 AM, Magnus Westerlund
>>> <magnus.westerlund@ericsson.com> wrote:
>>>
>>> Den 2016-10-13 kl. 20:12, skrev Cullen Jennings (fluffy):
>>>>
>>>> I think this is huge complexity that is not currently supported
>>>> and not needed. How about we generate the RID such that the RID
>>>> for the n'th m line is n. Or generate it as a random number. Do
>>>> you see any problem with this?
>>>>
>>>> I think this removes all the fingerprinting issues. I'm fine
>>>> with advice that say how to generate the RID in a way that is
>>>> privacy sensitive but I'm not a fan of making RID require support
>>>> for RTP header encryption.
>>>>
>>>
>>> So, let disregard the fingerprinting issue for now. It is clear
>>> that I am in the rough and that this whole issue would benefit from
>>> a broad review on what we leak when using ICE and RTP/RTCP.
>>
>> That broad review was largely done as part of PERC.
>>
>
>I do not agree that review happened in that context. I can't remember
>that we discussed fingerprinting at all. We quickly put any effort of
>privacy in regards to that the communication occurs out of scope. All
>the focus in PERC is to my understanding on maintaining confidentiality
>of the media content as well as preventing third party injection,
>modifications (other than sub selection of content)  and replay attacks.
>
>
>
>>>
>>> What I don't understand here is that people have argued and agreed
>>> that we should encrypt the RTP and RTCP packets per default. Then
>>> when we comes to RTP header extensions we are no longer ready to
>>> provide that protection per default by ensuring RFC6904
>>> implementation support. So RID and MID may not be the most
>>> sensitive fields if it has well constructed values. Then we have
>>> the audio level extensions where Client to mixer is RECOMMENDED to
>>> be implemented and REQUIRES RFC6904.
>>
>> So the audio levels are recommended in WebRTC but this MID stuff is a
>> very generic mechanism that will like be broadly used to deal with
>> RTP lack of extensibility of size PT space. Lot of stuff does not
>> implement audio levels and some stuff that implements it does not
>> encrypt it. The 6904 mechanism is not exactly elegant, not easy to
>> implement, and doubles the size the headers which is not great for
>> audio. I think it is fair to say we might do it differently if doing
>> it over again.
>>
>
>I think Jonathan had an excellent reply on this.
>
>>>
>>> So what is the issue, is it that you currently lack RFC6904
>>> implementations, and thus this require more implementation work, or
>>> are there additional reasons for wanting to expose this
>>> information?
>>
>> Yes this adds needless security complexity that people don't see the
>> value in thus it becomes difficult to convince implementors to do
>> it.
>>
>>>
>>> To conclude, I personally think default usage of RFC6904 on header
>>> extensions would be a good thing. I currently see no reasons for
>>> changing the requirements that RFC 7941 puts on WebRTC, forcing the
>>> usage of RFC6904 encryption. If you want to something else, please
>>> provide a proposal for what that alternative would be.
>>
>> Originally MID was not a SDES items, I'd propose just moving it back
>> to a normal RTP Header. In looking at trying to use this in
>> deployments, it's hard to see what value transporting it in RTCP as
>> well as RTP provided.  When that was originally proposed, I don't
>> think anyone realized that mean encrypting it and did not really care
>> if it was normal or SDES as it was about the same work. But if SDES
>> means it needs to be encrypted, then I think we need to reconsider
>> that.
>>
>
>So, I think moving away from SDES item is a bad idea. As that would
>create additional implementation burden, rather than having the same
>processing for a number of SDES items of similar type. RID, MID and
>CNAME all need similar processing for initial startup handling. So
>moving RID out, while still having the other two just creates an
>additional cases to implement.
>
>When it comes to RTCP inclusion of these SDES items I know of two reasons.
>
>Reason one, is that one can actually use RTCP to send these for an SSRC
>prior to having RTP data to send for it. That enables one to have the
>association in place prior to actually having to handle media data.
>
>Reason two is the saftey net RTCP provides. It enables a receiver to
>detect if its state is out of sync. Yes, this could be realized using
>periodic repeats in the RTP header extension also. However, that doesn't
>work if the SSRC currently don't have any data to send.
>
>Considering the state that draft-ietf-avtext-rid is in. Deciding to
>change it to not be an SDES item is going be an equal if not greater
>hassle to actually pull it back to the WG to include a beef up of the
>security consdieration with an motivation why the rules in RFC7941 do
>not apply to it. Assuming that is what the consensus ends up being.
>
>Cheers
>
>Magnus Westerlund
>
>----------------------------------------------------------------------
>Services, Media and Network features, Ericsson Research EAB/TXM
>----------------------------------------------------------------------
>Ericsson AB                 | Phone  +46 10 7148287
>F=E4r=F6gatan 6                 | Mobile +46 73 0949079
>SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>----------------------------------------------------------------------
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb


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Howdy,

The chairs are currently putting together the draft agenda for Seoul.  We
expect the lion's share will go to JSEP, as we hope to have a WG last call
issued before the meeting.  We have also received an agenda request for
adraft-copeland-rtcweb-p2p-idp-auth; if you have other agenda items to
request time for, please let us or the list know.

thanks,

Ted

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<div dir=3D"ltr"><div><div><div>Howdy,<br><br></div>The chairs are currentl=
y putting together the draft agenda for Seoul.=C2=A0 We expect the lion&#39=
;s share will go to JSEP, as we hope to have a WG last call issued before t=
he meeting.=C2=A0 We have also received an agenda request for adraft-copela=
nd-rtcweb-p2p-idp-auth; if you have other agenda items to request time for,=
 please let us or the list know.<br><br></div>thanks,<br><br></div>Ted<br><=
/div>

--001a113cabd860fa72053f262edb--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Ted Hardie <ted.ietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Agenda requests for IETF 97
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Archived-At: <https://mailarchive.ietf.org/arch/msg/rtcweb/PDhi76UVdZqG1FpT06DBzfAWmBM>
Subject: Re: [rtcweb] Agenda requests for IETF 97
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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Cc: RTCWeb IETF <rtcweb@ietf.org>
Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID and RID SDES items
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Hi,

I'd like some guidance on how to move forward and try to solve this issue. =
I really don't want us to switch to another mechanism at this point.

Regards,

Christer

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlun=
d
Sent: 17 October 2016 11:47
To: Cullen Jennings (fluffy) <fluffy@cisco.com>
Cc: RTCWeb IETF <rtcweb@ietf.org>
Subject: Re: [rtcweb] Mandating encryption of RTP header extensions for MID=
 and RID SDES items

Den 2016-10-14 kl. 15:08, skrev Cullen Jennings (fluffy):
>
>> On Oct 14, 2016, at 2:26 AM, Magnus Westerlund=20
>> <magnus.westerlund@ericsson.com> wrote:
>>
>> Den 2016-10-13 kl. 20:12, skrev Cullen Jennings (fluffy):
>>>
>>> I think this is huge complexity that is not currently supported and=20
>>> not needed. How about we generate the RID such that the RID for the=20
>>> n'th m line is n. Or generate it as a random number. Do you see any=20
>>> problem with this?
>>>
>>> I think this removes all the fingerprinting issues. I'm fine with=20
>>> advice that say how to generate the RID in a way that is privacy=20
>>> sensitive but I'm not a fan of making RID require support for RTP=20
>>> header encryption.
>>>
>>
>> So, let disregard the fingerprinting issue for now. It is clear that=20
>> I am in the rough and that this whole issue would benefit from a=20
>> broad review on what we leak when using ICE and RTP/RTCP.
>
> That broad review was largely done as part of PERC.
>

I do not agree that review happened in that context. I can't remember that =
we discussed fingerprinting at all. We quickly put any effort of privacy in=
 regards to that the communication occurs out of scope. All the focus in PE=
RC is to my understanding on maintaining confidentiality of the media conte=
nt as well as preventing third party injection, modifications (other than s=
ub selection of content)  and replay attacks.



>>
>> What I don't understand here is that people have argued and agreed=20
>> that we should encrypt the RTP and RTCP packets per default. Then=20
>> when we comes to RTP header extensions we are no longer ready to=20
>> provide that protection per default by ensuring RFC6904=20
>> implementation support. So RID and MID may not be the most sensitive=20
>> fields if it has well constructed values. Then we have the audio=20
>> level extensions where Client to mixer is RECOMMENDED to be=20
>> implemented and REQUIRES RFC6904.
>
> So the audio levels are recommended in WebRTC but this MID stuff is a=20
> very generic mechanism that will like be broadly used to deal with RTP=20
> lack of extensibility of size PT space. Lot of stuff does not=20
> implement audio levels and some stuff that implements it does not=20
> encrypt it. The 6904 mechanism is not exactly elegant, not easy to=20
> implement, and doubles the size the headers which is not great for=20
> audio. I think it is fair to say we might do it differently if doing=20
> it over again.
>

I think Jonathan had an excellent reply on this.

>>
>> So what is the issue, is it that you currently lack RFC6904=20
>> implementations, and thus this require more implementation work, or=20
>> are there additional reasons for wanting to expose this information?
>
> Yes this adds needless security complexity that people don't see the=20
> value in thus it becomes difficult to convince implementors to do it.
>
>>
>> To conclude, I personally think default usage of RFC6904 on header=20
>> extensions would be a good thing. I currently see no reasons for=20
>> changing the requirements that RFC 7941 puts on WebRTC, forcing the=20
>> usage of RFC6904 encryption. If you want to something else, please=20
>> provide a proposal for what that alternative would be.
>
> Originally MID was not a SDES items, I'd propose just moving it back=20
> to a normal RTP Header. In looking at trying to use this in=20
> deployments, it's hard to see what value transporting it in RTCP as=20
> well as RTP provided.  When that was originally proposed, I don't=20
> think anyone realized that mean encrypting it and did not really care=20
> if it was normal or SDES as it was about the same work. But if SDES=20
> means it needs to be encrypted, then I think we need to reconsider=20
> that.
>

So, I think moving away from SDES item is a bad idea. As that would create =
additional implementation burden, rather than having the same processing fo=
r a number of SDES items of similar type. RID, MID and CNAME all need simil=
ar processing for initial startup handling. So moving RID out, while still =
having the other two just creates an additional cases to implement.

When it comes to RTCP inclusion of these SDES items I know of two reasons.

Reason one, is that one can actually use RTCP to send these for an SSRC pri=
or to having RTP data to send for it. That enables one to have the associat=
ion in place prior to actually having to handle media data.

Reason two is the saftey net RTCP provides. It enables a receiver to detect=
 if its state is out of sync. Yes, this could be realized using periodic re=
peats in the RTP header extension also. However, that doesn't work if the S=
SRC currently don't have any data to send.

Considering the state that draft-ietf-avtext-rid is in. Deciding to change =
it to not be an SDES item is going be an equal if not greater hassle to act=
ually pull it back to the WG to include a beef up of the security consdiera=
tion with an motivation why the rules in RFC7941 do not apply to it. Assumi=
ng that is what the consensus ends up being.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
F=E4r=F6gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

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


From nobody Fri Oct 21 11:03:36 2016
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Real-Time Communication in WEB-browsers of the IETF.

        Title           : Javascript Session Establishment Protocol
        Authors         : Justin Uberti
                          Cullen Jennings
                          Eric Rescorla
	Filename        : draft-ietf-rtcweb-jsep-17.txt
	Pages           : 94
	Date            : 2016-10-21

Abstract:
   This document describes the mechanisms for allowing a Javascript
   application to control the signaling plane of a multimedia session
   via the interface specified in the W3C RTCPeerConnection API, and
   discusses how this relates to existing signaling protocols.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-17

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-jsep-17


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


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Subject: [rtcweb] Working Group Last Call: JSEP
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--001a1134e308e71835053f6504b3
Content-Type: text/plain; charset=UTF-8

The chairs would like to start a working group last call on
draft-ietf-rtcweb-jset-17 to end on November 9th, 2016, 17:00 KST.

Please review thoroughly, as working through the last comments will be the
major effort of our working group meeting in Seoul.

>From a process perspective, we will be working through the comments as
issues against the github repo.  In order to accommodate folks who do not
use github, the chairs will enter a new issue for any issue raised on the
mailing list.  That means that if you are raising a new issue, please send
it to the list with a new subject line so that the chairs catch it.

If you prefer, you may also enter the issues at
https://github.com/rtcweb-wg/jsep.  If you do so, please send a copy of the
issue to the mailing list so that conversation and resolution occurs here.

This has been a long road, but this a very major milestone.  Thanks to the
authors for their work so far.

regards,

Ted, Cullen, Sean


A New Internet-Draft is available from the on-line Internet-Drafts
directories.
This draft is a work item of the Real-Time Communication in WEB-browsers of
the IETF.

        Title           : Javascript Session Establishment Protocol
        Authors         : Justin Uberti
                          Cullen Jennings
                          Eric Rescorla
        Filename        : draft-ietf-rtcweb-jsep-17.txt
        Pages           : 94
        Date            : 2016-10-21

Abstract:
   This document describes the mechanisms for allowing a Javascript
   application to control the signaling plane of a multimedia session
   via the interface specified in the W3C RTCPeerConnection API, and
   discusses how this relates to existing signaling protocols.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-17

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-jsep-17


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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

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

--001a1134e308e71835053f6504b3
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div><div>The chairs would like to start a working group l=
ast call on draft-ietf-rtcweb-jset-17 to end on November 9th, 2016, 17:00 K=
ST.<br>=C2=A0 <br>Please review thoroughly, as working through the last com=
ments will be the major effort of our working group meeting in Seoul.<br><b=
r></div>From a process perspective, we will be working through the comments=
 as issues against the github repo.=C2=A0 In order to accommodate folks who=
 do not use github, the chairs will enter a new issue for any issue raised =
on the mailing list.=C2=A0 That means that if you are raising a new issue, =
please send it to the list with a new subject line so that the chairs catch=
 it.<br><br></div>If you prefer, you may also enter the issues at <a href=
=3D"https://github.com/rtcweb-wg/jsep">https://github.com/rtcweb-wg/jsep</a=
>.=C2=A0 If you do so, please send a copy of the issue to the mailing list =
so that conversation and resolution occurs here.<br><div><div><div><div cla=
ss=3D"gmail_quote"><br></div><div class=3D"gmail_quote">This has been a lon=
g road, but this a very major milestone.=C2=A0 Thanks to the authors for th=
eir work so far.<br><br></div><div class=3D"gmail_quote">regards,<br><br></=
div><div class=3D"gmail_quote">Ted, Cullen, Sean<br></div><div class=3D"gma=
il_quote"><br><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
This draft is a work item of the Real-Time Communication in WEB-browsers of=
 the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 Javascript Session Establishment Protocol<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Authors=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: Just=
in Uberti<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Cullen Jennings<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Eric Rescorla<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-jsep-17.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 94<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2016-10-21<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document describes the mechanisms for allowing a Javascri=
pt<br>
=C2=A0 =C2=A0application to control the signaling plane of a multimedia ses=
sion<br>
=C2=A0 =C2=A0via the interface specified in the W3C RTCPeerConnection API, =
and<br>
=C2=A0 =C2=A0discusses how this relates to existing signaling protocols.<br=
>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/" rel=3D=
"noreferrer" target=3D"_blank">https://datatracker.ietf.org/<wbr>doc/draft-=
ietf-rtcweb-jsep/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-17" rel=3D"no=
referrer" target=3D"_blank">https://tools.ietf.org/html/<wbr>draft-ietf-rtc=
web-jsep-17</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-jsep-17" r=
el=3D"noreferrer" target=3D"_blank">https://www.ietf.org/rfcdiff?<wbr>url2=
=3Ddraft-ietf-rtcweb-jsep-17</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" rel=3D"noreferrer" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" rel=3D"noreferrer" target=
=3D"_blank">ftp://ftp.ietf.org/internet-<wbr>drafts/</a><br>
<br>
______________________________<wbr>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/rtcweb</a><br=
>
</div><br></div></div></div></div>

--001a1134e308e71835053f6504b3--


From nobody Fri Oct 21 16:26:00 2016
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From: "\"IETF Secretariat\"" <agenda@ietf.org>
To: <ted.ietf@gmail.com>, <rtcweb-chairs@ietf.org>
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Cc: rtcweb@ietf.org
Subject: [rtcweb] rtcweb - Requested sessions have been scheduled for IETF 97
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Dear Ted Hardie,

The session(s) that you have requested have been scheduled.
Below is the scheduled session information followed by
the original request. 

rtcweb Session 1 (2:00:00)
    Monday, Afternoon Session II 1550-1750
    Room Name: Grand Ballroom 3 size: 175
    ---------------------------------------------
    rtcweb Session 2 (1:00:00)
    Wednesday, Afternoon Session II 1520-1620
    Room Name: Studio 4 size: 100
    ---------------------------------------------
    


Request Information:


---------------------------------------------------------
Working Group Name: Real-Time Communication in WEB-browsers
Area Name: Applications and Real-Time Area
Session Requester: Ted Hardie

Number of Sessions: 2
Length of Session(s):  2 Hours, 1 Hour
Number of Attendees: 100
Conflicts to Avoid: 
 First Priority: tls rmcat aqm avtcore avtext clue dispatch httpbis stir acme mmusic payload sipcore perc
 Second Priority: dprive tcpinc straw tsvwg tsvarea ace uta netvc capport
 Third Priority: insipid irtfopen ianaplan dane opsawg


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


From nobody Sun Oct 23 19:15:35 2016
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Subject: Re: [rtcweb] JSEP: Associate answer with offer
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--Apple-Mail=_E3E96035-83E4-40DB-A40E-F509BB65D52C
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One way to deal with this is, the JS app doing the signalling needs to =
take care of being able to deal with it.=20


> On Jul 26, 2016, at 3:35 AM, Christer Holmberg =
<christer.holmberg@ericsson.com <mailto:christer.holmberg@ericsson.com>> =
wrote:
>=20
> Hi,
> =20
> Assume the following:
> =20
> 1)      Alice calls createOffer(), and sends the generated offer =
(offer X) towards Bob.
> 2)      Alice calls createOffer() again, and sends the generated offer =
(offer Y) towards Bob.
> 3)      Alice receives an answer from Bob.
> =20
> How does Alice know whether the answer is associated with offer X (in =
which case I assume Alice should discard it) or with offer Y?
> =20
> Since JSEP allows multiple outstanding offers (per section 3.2), there =
needs to be a mechanism for associating an answer with a specific offer.
> =20
> Regards,
> =20
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb =
<https://www.ietf.org/mailman/listinfo/rtcweb>

--Apple-Mail=_E3E96035-83E4-40DB-A40E-F509BB65D52C
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"><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-line-break: after-white-space;" =
class=3D""><div class=3D""><br class=3D""></div>One way to deal with =
this is, the JS app doing the signalling needs to take care of being =
able to deal with it.&nbsp;<div class=3D""><br class=3D""></div><div =
class=3D""><br class=3D""><div class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D"">On Jul 26, 2016, at 3:35 AM, Christer =
Holmberg &lt;<a href=3D"mailto:christer.holmberg@ericsson.com" =
class=3D"">christer.holmberg@ericsson.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><div =
class=3D"WordSection1" style=3D"page: WordSection1; font-family: =
Helvetica; font-size: 14px; font-style: normal; font-variant-caps: =
normal; font-weight: normal; letter-spacing: normal; orphans: auto; =
text-align: start; text-indent: 0px; text-transform: none; white-space: =
normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: =
0px;"><div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 11pt; =
font-family: Calibri, sans-serif;" class=3D"">Hi,<o:p =
class=3D""></o:p></div><div style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 11pt; font-family: Calibri, sans-serif;" class=3D""><o:p =
class=3D"">&nbsp;</o:p></div><div style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 11pt; font-family: Calibri, sans-serif;" class=3D"">Assume =
the following:<o:p class=3D""></o:p></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif;" =
class=3D""><o:p class=3D"">&nbsp;</o:p></div><div style=3D"margin: 0cm =
0cm 0.0001pt 36pt; font-size: 11pt; font-family: Calibri, sans-serif; =
text-indent: -18pt;" class=3D""><span class=3D"">1)<span =
style=3D"font-style: normal; font-variant-caps: normal; font-weight: =
normal; font-size: 7pt; line-height: normal; font-family: 'Times New =
Roman';" class=3D"">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span =
class=3D"Apple-converted-space">&nbsp;</span></span></span>Alice calls =
createOffer(), and sends the generated offer (offer X) towards Bob.<o:p =
class=3D""></o:p></div><div style=3D"margin: 0cm 0cm 0.0001pt 36pt; =
font-size: 11pt; font-family: Calibri, sans-serif; text-indent: -18pt;" =
class=3D""><span class=3D"">2)<span style=3D"font-style: normal; =
font-variant-caps: normal; font-weight: normal; font-size: 7pt; =
line-height: normal; font-family: 'Times New Roman';" =
class=3D"">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span =
class=3D"Apple-converted-space">&nbsp;</span></span></span>Alice calls =
createOffer() again, and sends the generated offer (offer Y) towards =
Bob.<o:p class=3D""></o:p></div><div style=3D"margin: 0cm 0cm 0.0001pt =
36pt; font-size: 11pt; font-family: Calibri, sans-serif; text-indent: =
-18pt;" class=3D""><span class=3D"">3)<span style=3D"font-style: normal; =
font-variant-caps: normal; font-weight: normal; font-size: 7pt; =
line-height: normal; font-family: 'Times New Roman';" =
class=3D"">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<span =
class=3D"Apple-converted-space">&nbsp;</span></span></span>Alice =
receives an answer from Bob.<o:p class=3D""></o:p></div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 11pt; font-family: =
Calibri, sans-serif;" class=3D""><o:p class=3D"">&nbsp;</o:p></div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 11pt; font-family: =
Calibri, sans-serif;" class=3D"">How does Alice know whether the answer =
is associated with offer X (in which case I assume Alice should discard =
it) or with offer Y?<o:p class=3D""></o:p></div><div style=3D"margin: =
0cm 0cm 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif;" =
class=3D""><o:p class=3D"">&nbsp;</o:p></div><div style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif;" =
class=3D"">Since JSEP allows multiple outstanding offers (per section =
3.2), there needs to be a mechanism for associating an answer with a =
specific offer.<o:p class=3D""></o:p></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif;" =
class=3D""><o:p class=3D"">&nbsp;</o:p></div><div style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif;" =
class=3D"">Regards,<o:p class=3D""></o:p></div><div style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif;" =
class=3D""><o:p class=3D"">&nbsp;</o:p></div><div style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif;" =
class=3D"">Christer<o:p class=3D""></o:p></div></div><span =
style=3D"font-family: Helvetica; font-size: 14px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; float: none; display: inline =
!important;" =
class=3D"">_______________________________________________</span><br =
style=3D"font-family: Helvetica; font-size: 14px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px;" class=3D""><span style=3D"font-family: =
Helvetica; font-size: 14px; font-style: normal; font-variant-caps: =
normal; font-weight: normal; letter-spacing: normal; orphans: auto; =
text-align: start; text-indent: 0px; text-transform: none; white-space: =
normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; =
float: none; display: inline !important;" class=3D"">rtcweb mailing =
list</span><br style=3D"font-family: Helvetica; font-size: 14px; =
font-style: normal; font-variant-caps: normal; font-weight: normal; =
letter-spacing: normal; orphans: auto; text-align: start; text-indent: =
0px; text-transform: none; white-space: normal; widows: auto; =
word-spacing: 0px; -webkit-text-stroke-width: 0px;" class=3D""><a =
href=3D"mailto:rtcweb@ietf.org" style=3D"color: rgb(149, 79, 114); =
text-decoration: underline; font-family: Helvetica; font-size: 14px; =
font-style: normal; font-variant-caps: normal; font-weight: normal; =
letter-spacing: normal; orphans: auto; text-align: start; text-indent: =
0px; text-transform: none; white-space: normal; widows: auto; =
word-spacing: 0px; -webkit-text-stroke-width: 0px;" =
class=3D"">rtcweb@ietf.org</a><br style=3D"font-family: Helvetica; =
font-size: 14px; font-style: normal; font-variant-caps: normal; =
font-weight: normal; letter-spacing: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; white-space: normal; =
widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;" =
class=3D""><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
style=3D"color: rgb(149, 79, 114); text-decoration: underline; =
font-family: Helvetica; font-size: 14px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px;" =
class=3D"">https://www.ietf.org/mailman/listinfo/rtcweb</a></div></blockqu=
ote></div><br class=3D""></div></body></html>=

--Apple-Mail=_E3E96035-83E4-40DB-A40E-F509BB65D52C--


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From: Taylor Brandstetter <deadbeef@google.com>
Date: Mon, 24 Oct 2016 12:12:29 -0700
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Subject: Re: [rtcweb] JSEP: Associate answer with offer
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I don't think JSEP section 3.2 allows multiple outstanding offers. To quote:

   "In [RFC3264], the constraint at the signaling level is that only one
   offer can be outstanding for a given session, but at the media stack
   level, a new offer can be generated at any point.  For example, when
   using SIP for signaling, if one offer is sent, then cancelled using a
   SIP CANCEL, another offer can be generated even though no answer was
   received for the first offer."

RFC 3264 states that an agent MUST be prepared to receive media after
sending an offer (even before receiving an answer). So allowing multiple
outstanding offers would severely complicate this. My assumption was that
there is only one outstanding offer, which is the one last applied via
setLocalDescription.

On Sun, Oct 23, 2016 at 7:15 PM, Cullen Jennings <fluffy@cisco.com> wrote:

>
> One way to deal with this is, the JS app doing the signalling needs to
> take care of being able to deal with it.
>
>
> On Jul 26, 2016, at 3:35 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
> Hi,
>
> Assume the following:
>
> 1)      Alice calls createOffer(), and sends the generated offer (offer
> X) towards Bob.
> 2)      Alice calls createOffer() again, and sends the generated offer
> (offer Y) towards Bob.
> 3)      Alice receives an answer from Bob.
>
> How does Alice know whether the answer is associated with offer X (in
> which case I assume Alice should discard it) or with offer Y?
>
> Since JSEP allows multiple outstanding offers (per section 3.2), there
> needs to be a mechanism for associating an answer with a specific offer.
>
> Regards,
>
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">I=C2=A0don&#39;t think JSEP section 3.2 allows multiple ou=
tstanding offers. To quote:<br><br>=C2=A0 =C2=A0&quot;In [RFC3264], the con=
straint at the signaling level is that only one<br>=C2=A0 =C2=A0offer can b=
e outstanding for a given session, but at the media stack<br>=C2=A0 =C2=A0l=
evel, a new offer can be generated at any point.=C2=A0 For example, when<br=
>=C2=A0 =C2=A0using SIP for signaling, if one offer is sent, then cancelled=
 using a<br>=C2=A0 =C2=A0SIP CANCEL, another offer can be generated even th=
ough no answer was<br>=C2=A0 =C2=A0received for the first offer.&quot;<br><=
br>RFC 3264 states that an agent MUST be prepared to receive media after se=
nding an offer (even before receiving an answer). So allowing multiple outs=
tanding offers would severely complicate this. My assumption was that there=
 is only one outstanding offer, which is the one last applied via setLocalD=
escription.=C2=A0<div class=3D"gmail_extra"><div class=3D"gmail_quote"><br>=
</div><div class=3D"gmail_quote">On Sun, Oct 23, 2016 at 7:15 PM, Cullen Je=
nnings <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"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex"><div style=3D"word-wrap:break-word"><div><br></div>One way to deal wi=
th this is, the JS app doing the signalling needs to take care of being abl=
e to deal with it.=C2=A0<div><br></div><div><br><div><blockquote type=3D"ci=
te"><div><div class=3D"h5"><div>On Jul 26, 2016, at 3:35 AM, Christer Holmb=
erg &lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank"=
>christer.holmberg@ericsson.<wbr>com</a>&gt; wrote:</div><br class=3D"m_285=
7520053865078467Apple-interchange-newline"></div></div><div><div><div class=
=3D"h5"><div class=3D"m_2857520053865078467WordSection1" style=3D"font-fami=
ly:Helvetica;font-size:14px;font-style:normal;font-variant-caps:normal;font=
-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-=
transform:none;white-space:normal;word-spacing:0px"><div style=3D"margin:0c=
m 0cm 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">Hi,<u></u><u>=
</u></div><div style=3D"margin:0cm 0cm 0.0001pt;font-size:11pt;font-family:=
Calibri,sans-serif"><u></u>=C2=A0<u></u></div><div style=3D"margin:0cm 0cm =
0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">Assume the followin=
g:<u></u><u></u></div><div style=3D"margin:0cm 0cm 0.0001pt;font-size:11pt;=
font-family:Calibri,sans-serif"><u></u>=C2=A0<u></u></div><div style=3D"mar=
gin:0cm 0cm 0.0001pt 36pt;font-size:11pt;font-family:Calibri,sans-serif"><s=
pan>1)<span style=3D"font-style:normal;font-variant-caps:normal;font-weight=
:normal;font-size:7pt;line-height:normal;font-family:&#39;Times New Roman&#=
39;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0<span class=3D"m_2857520053865078467Appl=
e-converted-space">=C2=A0</span></span></span>Alice calls createOffer(), an=
d sends the generated offer (offer X) towards Bob.<u></u><u></u></div><div =
style=3D"margin:0cm 0cm 0.0001pt 36pt;font-size:11pt;font-family:Calibri,sa=
ns-serif"><span>2)<span style=3D"font-style:normal;font-variant-caps:normal=
;font-weight:normal;font-size:7pt;line-height:normal;font-family:&#39;Times=
 New Roman&#39;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0<span class=3D"m_28575200538=
65078467Apple-converted-space">=C2=A0</span></span></span>Alice calls creat=
eOffer() again, and sends the generated offer (offer Y) towards Bob.<u></u>=
<u></u></div><div style=3D"margin:0cm 0cm 0.0001pt 36pt;font-size:11pt;font=
-family:Calibri,sans-serif"><span>3)<span style=3D"font-style:normal;font-v=
ariant-caps:normal;font-weight:normal;font-size:7pt;line-height:normal;font=
-family:&#39;Times New Roman&#39;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0<span clas=
s=3D"m_2857520053865078467Apple-converted-space">=C2=A0</span></span></span=
>Alice receives an answer from Bob.<u></u><u></u></div><div style=3D"margin=
:0cm 0cm 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif"><u></u>=C2=
=A0<u></u></div><div style=3D"margin:0cm 0cm 0.0001pt;font-size:11pt;font-f=
amily:Calibri,sans-serif">How does Alice know whether the answer is associa=
ted with offer X (in which case I assume Alice should discard it) or with o=
ffer Y?<u></u><u></u></div><div style=3D"margin:0cm 0cm 0.0001pt;font-size:=
11pt;font-family:Calibri,sans-serif"><u></u>=C2=A0<u></u></div><div style=
=3D"margin:0cm 0cm 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">=
Since JSEP allows multiple outstanding offers (per section 3.2), there need=
s to be a mechanism for associating an answer with a specific offer.<u></u>=
<u></u></div><div style=3D"margin:0cm 0cm 0.0001pt;font-size:11pt;font-fami=
ly:Calibri,sans-serif"><u></u>=C2=A0<u></u></div><div style=3D"margin:0cm 0=
cm 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">Regards,<u></u><=
u></u></div><div style=3D"margin:0cm 0cm 0.0001pt;font-size:11pt;font-famil=
y:Calibri,sans-serif"><u></u>=C2=A0<u></u></div><div style=3D"margin:0cm 0c=
m 0.0001pt;font-size:11pt;font-family:Calibri,sans-serif">Christer<u></u><u=
></u></div></div></div></div><span style=3D"font-family:Helvetica;font-size=
:14px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-=
spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-s=
pace:normal;word-spacing:0px;float:none;display:inline!important">_________=
_____________________<wbr>_________________</span><br style=3D"font-family:=
Helvetica;font-size:14px;font-style:normal;font-variant-caps:normal;font-we=
ight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-tra=
nsform:none;white-space:normal;word-spacing:0px"><span style=3D"font-family=
:Helvetica;font-size:14px;font-style:normal;font-variant-caps:normal;font-w=
eight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-tr=
ansform:none;white-space:normal;word-spacing:0px;float:none;display:inline!=
important">rtcweb mailing list</span><br style=3D"font-family:Helvetica;fon=
t-size:14px;font-style:normal;font-variant-caps:normal;font-weight:normal;l=
etter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;w=
hite-space:normal;word-spacing:0px"><a href=3D"mailto:rtcweb@ietf.org" styl=
e=3D"color:rgb(149,79,114);text-decoration:underline;font-family:Helvetica;=
font-size:14px;font-style:normal;font-variant-caps:normal;font-weight:norma=
l;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:non=
e;white-space:normal;word-spacing:0px" target=3D"_blank">rtcweb@ietf.org</a=
><br style=3D"font-family:Helvetica;font-size:14px;font-style:normal;font-v=
ariant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:star=
t;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px">=
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" style=3D"color:rgb=
(149,79,114);text-decoration:underline;font-family:Helvetica;font-size:14px=
;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spaci=
ng:normal;text-align:start;text-indent:0px;text-transform:none;white-space:=
normal;word-spacing:0px" target=3D"_blank">https://www.ietf.org/mailman/<wb=
r>listinfo/rtcweb</a></div></blockquote></div><br></div></div><br>_________=
_____________________<wbr>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/rtcweb</a><br=
>
<br></blockquote></div><br></div></div>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Taylor Brandstetter <deadbeef@google.com>, Cullen Jennings <fluffy@cisco.com>
Thread-Topic: [rtcweb] JSEP: Associate answer with offer
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From: "Aijun Wang" <wangaijun@tsinghua.org.cn>
To: <rtcweb@ietf.org>
Date: Tue, 25 Oct 2016 09:52:04 +0800
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Subject: [rtcweb] OARS solution for webrtc application
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Hi, all respectable RTCWEB experts:

 

Here we propose one draft that can cover peer to peer communication scenario
under various complex network environment(Cone/Symmetry NAT, IPv4&IPv6
Coexistence etc.) with the unified solution. 

https://tools.ietf.org/html/draft-wang-rtcweb-oars-00

 

This document proposes a new relay-based NAT traversal architecture called
OARS which could simplify the data communication process
    between two hosts that locates behind some non-BEHAVE compliant [RFC4787
<https://tools.ietf.org/html/rfc4787> ] [RFC5382
<https://tools.ietf.org/html/rfc5382> ] NAT devices.  The key mechanism in
OARS is the
    newly defined "Couple" operation (using STUN [RFC5389
<https://tools.ietf.org/html/rfc5389> ] message format) which allows the
Relay servers to be easily incorporated
    into existing CGN/CDN nodes which are already deployed within the
network in a distributed manner.

 

One brief introduction can be accessed at
https://www.ietf.org/proceedings/96/slides/slides-96-tram-1.pdf, wish to
hear your valuable comments on this topic.

 

 

Best Regards.

 

Aijun Wang

China Telecom Beijing Research Institute

Network R&D and Operation Support Department

 

 


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	{size:612.0pt 792.0pt;
	margin:72.0pt 90.0pt 72.0pt 90.0pt;}
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	{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DZH-CN link=3Dblue =
vlink=3Dpurple style=3D'text-justify-trim:punctuation'><div =
class=3DWordSection1><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt'>Hi, all respectable RTCWEB =
experts:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'font-size:12.0pt'>Here we =
propose one draft that can cover peer to peer communication scenario =
under various complex network environment(Cone/Symmetry NAT, =
IPv4&amp;IPv6 Coexistence etc.) with the unified solution. =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US><a =
href=3D"https://tools.ietf.org/html/draft-wang-rtcweb-oars-00">https://to=
ols.ietf.org/html/draft-wang-rtcweb-oars-00</a><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><pre =
style=3D'page-break-before:always'><span lang=3DEN>This document =
proposes a new relay-based NAT traversal architecture called OARS which =
could simplify the data communication =
process<o:p></o:p></span></pre><pre =
style=3D'page-break-before:always'><span lang=3DEN>&nbsp;&nbsp;&nbsp; =
between two hosts that locates behind some non-BEHAVE compliant [<a =
href=3D"https://tools.ietf.org/html/rfc4787" title=3D"&quot;Network =
Address Translation (NAT) Behavioral Requirements for Unicast =
UDP&quot;">RFC4787</a>] [<a href=3D"https://tools.ietf.org/html/rfc5382" =
title=3D"&quot;NAT Behavioral Requirements for TCP&quot;">RFC5382</a>] =
NAT devices.&nbsp; The key mechanism in OARS is =
the<o:p></o:p></span></pre><pre style=3D'page-break-before:always'><span =
lang=3DEN>&nbsp;&nbsp;&nbsp; newly defined &quot;Couple&quot; operation =
(using STUN [<a href=3D"https://tools.ietf.org/html/rfc5389" =
title=3D"&quot;Session Traversal Utilities for NAT =
(STUN)&quot;">RFC5389</a>] message format) which allows the Relay =
servers to be easily incorporated<o:p></o:p></span></pre><pre =
style=3D'page-break-before:always'><span lang=3DEN>&nbsp;&nbsp;&nbsp; =
into existing CGN/CDN nodes which are already deployed within the =
network in a distributed manner.</span><span =
lang=3DEN-US><o:p></o:p></span></pre><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:12.0pt'>One brief introduction can be =
accessed at <a =
href=3D"https://www.ietf.org/proceedings/96/slides/slides-96-tram-1.pdf">=
https://www.ietf.org/proceedings/96/slides/slides-96-tram-1.pdf</a>, =
wish to hear your valuable comments on this =
topic.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>Best Regards.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>Aijun Wang<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>China Telecom Beijing Research =
Institute<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>Network R&amp;D and Operation Support =
Department<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p></div></body></html>
------=_NextPart_000_0155_01D22EA5.70B1AEA0--


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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [rtcweb] Working Group Last Call: JSEP
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Hi,

This is a well written draft. I think Section 6 needs work, along the =
lines of the recent discussion and updates to BUNDLE, but the rest of =
the document seems to be in good shape.

Section 6 says:

   Note: The following algorithm does not yet have WG consensus but is
   included here as something concrete for the working group to discuss.

I don=E2=80=99t think what=E2=80=99s written here is right yet. The RTP =
specification has rules for forming RTP packets into RTP streams based =
on their SSRC, and for associating RTCP packets with RTP streams. =
Similarly, BUNDLE, as of -35, has rules for mapping RTP streams to m=3D =
sections based on the MID. This section seems to be trying to replicate =
these, rather than building on them.=20

I think this section should leave the mapping from RTP/RTCP packets to =
RTP streams to RTP, and the mapping from RTP streams to m=3D sections to =
SDP (and BUNDLE, if used). It should talk instead about how RTP streams =
associated with m=3D sections are mapped onto RtpSenders and =
RtpReceivers. This may sound like just a terminology difference, but I =
think it=E2=80=99s important to get the layering and division of =
responsibilities right, if the protocol is to be maintainable and =
extensible.


I also had some minor nits on the rest of the document:

- Section 5.1.1, 1st paragraph: The list of mandatory to implement =
standards is =E2=80=9Cderived from the requirements outlined in =
[I-D.ietf-rtcweb-rtp-usage]=E2=80=9D. Should the draft say somewhere =
that =E2=80=9CRTP media transport MUST be implemented according to =
[I-D.ietf-rtcweb-rtp-usage]=E2=80=9D?=20

- Section 5.1.1: It=E2=80=99s easy to skim-read R-5 this as =E2=80=9CMUST=E2=
=80=9D, since all the other requirements are MUST. It might be more =
visually distinct if written as  =E2=80=9CSHALL NOT=E2=80=9D rather than =
=E2=80=9CMUST NOT=E2=80=9D.

- Section 5.1.1: R-16 should be removed.

- Section 5.2.1, top of page 38: The header extension is in Section 14 =
of BUNDLE, not Section 11 (also second bullet point on page 39).

- Section 5.8, 3rd bullet on page 60: I suggest changing =E2=80=9CIf the =
MID header extension is supported, prepare to demux RTP data intended =
for this media section=E2=80=9C to =E2=80=9C=E2=80=A6prepare to demux =
RTP streams intended=E2=80=A6=E2=80=9D, to match the recent discussion =
around terminology in BUNDLE.

Colin




> On 21 Oct 2016, at 20:25, Ted Hardie <ted.ietf@gmail.com> wrote:
>=20
> The chairs would like to start a working group last call on =
draft-ietf-rtcweb-jset-17 to end on November 9th, 2016, 17:00 KST.
>  =20
> Please review thoroughly, as working through the last comments will be =
the major effort of our working group meeting in Seoul.
>=20
> =46rom a process perspective, we will be working through the comments =
as issues against the github repo.  In order to accommodate folks who do =
not use github, the chairs will enter a new issue for any issue raised =
on the mailing list.  That means that if you are raising a new issue, =
please send it to the list with a new subject line so that the chairs =
catch it.
>=20
> If you prefer, you may also enter the issues at =
https://github.com/rtcweb-wg/jsep <https://github.com/rtcweb-wg/jsep>.  =
If you do so, please send a copy of the issue to the mailing list so =
that conversation and resolution occurs here.
>=20
> This has been a long road, but this a very major milestone.  Thanks to =
the authors for their work so far.
>=20
> regards,
>=20
> Ted, Cullen, Sean
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the Real-Time Communication in =
WEB-browsers of the IETF.
>=20
>         Title           : Javascript Session Establishment Protocol
>         Authors         : Justin Uberti
>                           Cullen Jennings
>                           Eric Rescorla
>         Filename        : draft-ietf-rtcweb-jsep-17.txt
>         Pages           : 94
>         Date            : 2016-10-21
>=20
> Abstract:
>    This document describes the mechanisms for allowing a Javascript
>    application to control the signaling plane of a multimedia session
>    via the interface specified in the W3C RTCPeerConnection API, and
>    discusses how this relates to existing signaling protocols.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/ =
<https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/>
>=20
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-17 =
<https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-17>
>=20
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-jsep-17 =
<https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-jsep-17>
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org =
<http://tools.ietf.org/>.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/ =
<ftp://ftp.ietf.org/internet-drafts/>
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb =
<https://www.ietf.org/mailman/listinfo/rtcweb>
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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





--Apple-Mail=_F4D33ECE-2B42-4A49-A04B-50FA1132FCB1
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=utf-8

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D"">Hi,<div class=3D""><br class=3D""></div><div class=3D"">This =
is a well written draft. I think Section 6 needs work, along the lines =
of the recent discussion and updates to BUNDLE, but the rest of the =
document seems to be in good shape.</div><div class=3D""><br =
class=3D""></div><div class=3D"">Section 6 says:</div><div class=3D""><br =
class=3D""></div><div class=3D""><div class=3D"">&nbsp; &nbsp;Note: The =
following algorithm does not yet have WG consensus but is</div><div =
class=3D"">&nbsp; &nbsp;included here as something concrete for the =
working group to discuss.</div></div><div class=3D""><br =
class=3D""></div><div class=3D"">I don=E2=80=99t think what=E2=80=99s =
written here is right yet. The RTP specification has rules for forming =
RTP packets into RTP streams based on their SSRC, and for associating =
RTCP packets with RTP streams. Similarly, BUNDLE, as of -35, has rules =
for mapping RTP streams to m=3D sections based on the MID. This section =
seems to be trying to replicate these, rather than building on =
them.&nbsp;</div><div class=3D""><br class=3D""></div><div class=3D"">I =
think this section should leave the mapping from RTP/RTCP packets to RTP =
streams to RTP, and the mapping from RTP streams to m=3D sections to SDP =
(and BUNDLE, if used). It should talk instead about how RTP streams =
associated with m=3D sections are mapped onto RtpSenders and =
RtpReceivers. This may sound like just a terminology difference, but I =
think it=E2=80=99s important to get the layering and division of =
responsibilities right, if the protocol is to be maintainable and =
extensible.</div><div class=3D""><br class=3D""></div><div class=3D""><br =
class=3D""></div><div class=3D"">I also had some minor nits on the rest =
of the document:</div><div class=3D""><br class=3D""></div><div =
class=3D""><div class=3D"">- Section 5.1.1, 1st paragraph: The list of =
mandatory to implement standards is =E2=80=9Cderived from the =
requirements outlined in [I-D.ietf-rtcweb-rtp-usage]=E2=80=9D. Should =
the draft say somewhere that =E2=80=9CRTP media transport MUST be =
implemented according to [I-D.ietf-rtcweb-rtp-usage]=E2=80=9D?&nbsp;</div>=
<div class=3D""><br class=3D""></div><div class=3D"">- Section 5.1.1: =
It=E2=80=99s easy to skim-read R-5 this as =E2=80=9CMUST=E2=80=9D, since =
all the other requirements are MUST. It might be more visually distinct =
if written as &nbsp;=E2=80=9CSHALL NOT=E2=80=9D rather than =E2=80=9CMUST =
NOT=E2=80=9D.</div><div class=3D""><br class=3D""></div><div class=3D"">- =
Section 5.1.1: R-16 should be removed.</div><div class=3D""><br =
class=3D""></div><div class=3D"">- Section 5.2.1, top of page 38: The =
header extension is in Section 14 of BUNDLE, not Section 11 (also second =
bullet point on page 39).</div><div class=3D""><br class=3D""></div><div =
class=3D"">- Section 5.8, 3rd bullet on page 60: I suggest changing =
=E2=80=9CIf the MID header extension is supported, prepare to demux RTP =
data intended for this media section=E2=80=9C to =E2=80=9C=E2=80=A6prepare=
 to demux RTP streams intended=E2=80=A6=E2=80=9D, to match the recent =
discussion around terminology in BUNDLE.</div></div><div class=3D""><br =
class=3D""></div><div class=3D"">Colin</div><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""><div><blockquote =
type=3D"cite" class=3D""><div class=3D"">On 21 Oct 2016, at 20:25, Ted =
Hardie &lt;<a href=3D"mailto:ted.ietf@gmail.com" =
class=3D"">ted.ietf@gmail.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><div dir=3D"ltr" =
class=3D""><div class=3D""><div class=3D"">The chairs would like to =
start a working group last call on draft-ietf-rtcweb-jset-17 to end on =
November 9th, 2016, 17:00 KST.<br class=3D"">&nbsp; <br class=3D"">Please =
review thoroughly, as working through the last comments will be the =
major effort of our working group meeting in Seoul.<br class=3D""><br =
class=3D""></div>=46rom a process perspective, we will be working =
through the comments as issues against the github repo.&nbsp; In order =
to accommodate folks who do not use github, the chairs will enter a new =
issue for any issue raised on the mailing list.&nbsp; That means that if =
you are raising a new issue, please send it to the list with a new =
subject line so that the chairs catch it.<br class=3D""><br =
class=3D""></div>If you prefer, you may also enter the issues at <a =
href=3D"https://github.com/rtcweb-wg/jsep" =
class=3D"">https://github.com/rtcweb-wg/jsep</a>.&nbsp; If you do so, =
please send a copy of the issue to the mailing list so that conversation =
and resolution occurs here.<br class=3D""><div class=3D""><div =
class=3D""><div class=3D""><div class=3D"gmail_quote"><br =
class=3D""></div><div class=3D"gmail_quote">This has been a long road, =
but this a very major milestone.&nbsp; Thanks to the authors for their =
work so far.<br class=3D""><br class=3D""></div><div =
class=3D"gmail_quote">regards,<br class=3D""><br class=3D""></div><div =
class=3D"gmail_quote">Ted, Cullen, Sean<br class=3D""></div><div =
class=3D"gmail_quote"><br class=3D""><br class=3D"">
A New Internet-Draft is available from the on-line Internet-Drafts =
directories.<br class=3D"">
This draft is a work item of the Real-Time Communication in WEB-browsers =
of the IETF.<br class=3D"">
<br class=3D"">
&nbsp; &nbsp; &nbsp; &nbsp; Title&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;: Javascript Session Establishment Protocol<br class=3D"">
&nbsp; &nbsp; &nbsp; &nbsp; Authors&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;: =
Justin Uberti<br class=3D"">
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; Cullen Jennings<br class=3D"">
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; Eric Rescorla<br class=3D"">
&nbsp; &nbsp; &nbsp; &nbsp; Filename&nbsp; &nbsp; &nbsp; &nbsp; : =
draft-ietf-rtcweb-jsep-17.txt<br class=3D"">
&nbsp; &nbsp; &nbsp; &nbsp; Pages&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;: 94<br class=3D"">
&nbsp; &nbsp; &nbsp; &nbsp; Date&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; : 2016-10-21<br class=3D"">
<br class=3D"">
Abstract:<br class=3D"">
&nbsp; &nbsp;This document describes the mechanisms for allowing a =
Javascript<br class=3D"">
&nbsp; &nbsp;application to control the signaling plane of a multimedia =
session<br class=3D"">
&nbsp; &nbsp;via the interface specified in the W3C RTCPeerConnection =
API, and<br class=3D"">
&nbsp; &nbsp;discusses how this relates to existing signaling =
protocols.<br class=3D"">
<br class=3D"">
<br class=3D"">
The IETF datatracker status page for this draft is:<br class=3D"">
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/" =
rel=3D"noreferrer" target=3D"_blank" =
class=3D"">https://datatracker.ietf.org/<wbr =
class=3D"">doc/draft-ietf-rtcweb-jsep/</a><br class=3D"">
<br class=3D"">
There's also a htmlized version available at:<br class=3D"">
<a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-17" =
rel=3D"noreferrer" target=3D"_blank" =
class=3D"">https://tools.ietf.org/html/<wbr =
class=3D"">draft-ietf-rtcweb-jsep-17</a><br class=3D"">
<br class=3D"">
A diff from the previous version is available at:<br class=3D"">
<a href=3D"https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-jsep-17" =
rel=3D"noreferrer" target=3D"_blank" =
class=3D"">https://www.ietf.org/rfcdiff?<wbr =
class=3D"">url2=3Ddraft-ietf-rtcweb-jsep-17</a><br class=3D"">
<br class=3D"">
<br class=3D"">
Please note that it may take a couple of minutes from the time of =
submission<br class=3D"">
until the htmlized version and diff are available at <a =
href=3D"http://tools.ietf.org/" rel=3D"noreferrer" target=3D"_blank" =
class=3D"">tools.ietf.org</a>.<br class=3D"">
<br class=3D"">
Internet-Drafts are also available by anonymous FTP at:<br class=3D"">
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" rel=3D"noreferrer" =
target=3D"_blank" class=3D"">ftp://ftp.ietf.org/internet-<wbr =
class=3D"">drafts/</a><br class=3D"">
<br class=3D"">
______________________________<wbr class=3D"">_________________<br =
class=3D"">
rtcweb mailing list<br class=3D"">
<a href=3D"mailto:rtcweb@ietf.org" class=3D"">rtcweb@ietf.org</a><br =
class=3D"">
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
rel=3D"noreferrer" target=3D"_blank" =
class=3D"">https://www.ietf.org/mailman/<wbr =
class=3D"">listinfo/rtcweb</a><br class=3D"">
</div><br class=3D""></div></div></div></div>
_______________________________________________<br class=3D"">rtcweb =
mailing list<br class=3D""><a href=3D"mailto:rtcweb@ietf.org" =
class=3D"">rtcweb@ietf.org</a><br =
class=3D"">https://www.ietf.org/mailman/listinfo/rtcweb<br =
class=3D""></div></blockquote></div><br class=3D""><div class=3D"">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
font-variant-ligatures: normal; font-variant-position: normal; =
font-variant-numeric: normal; font-variant-alternates: normal; =
font-variant-east-asian: normal; line-height: normal; border-spacing: =
0px;"><span class=3D"Apple-style-span" style=3D"font-size: 9px;"><div =
class=3D""><br class=3D"Apple-interchange-newline"><br =
class=3D"khtml-block-placeholder"></div><div class=3D"">--&nbsp;</div><div=
 class=3D""></div><div class=3D"">Colin Perkins</div><div class=3D""><a =
href=3D"https://csperkins.org/" =
class=3D"">https://csperkins.org/</a></div><div class=3D""><br =
class=3D""></div></span></span><br class=3D"Apple-interchange-newline"><br=
 class=3D"Apple-interchange-newline">
</div>
<br class=3D""></div></body></html>=

--Apple-Mail=_F4D33ECE-2B42-4A49-A04B-50FA1132FCB1--


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Tue, 25 Oct 2016 11:03:27 -0700
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To: Colin Perkins <csp@csperkins.org>
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Working Group Last Call: JSEP
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Colin said:

"I don=E2=80=99t think what=E2=80=99s written here is right yet. The RTP sp=
ecification has
rules for forming RTP packets into RTP streams based on their SSRC, and for
associating RTCP packets with RTP streams. Similarly, BUNDLE, as of -35,
has rules for mapping RTP streams to m=3D sections based on the MID. This
section seems to be trying to replicate these, rather than building on
them."

I think this section should leave the mapping from RTP/RTCP packets to RTP
streams to RTP, and the mapping from RTP streams to m=3D sections to SDP (a=
nd
BUNDLE, if used). It should talk instead about how RTP streams associated
with m=3D sections are mapped onto RtpSenders and RtpReceivers. This may
sound like just a terminology difference, but I think it=E2=80=99s importan=
t to get
the layering and division of responsibilities right, if the protocol is to
be maintainable and extensible."

[BA] The advice given in the JSEP document is more detailed than what is
provided in BUNDLE and also covers situations (overlapping payload types)
that BUNDLE mandates packet drops for.  So I don't believe that the
material is entirely overlapping.

On Tue, Oct 25, 2016 at 10:42 AM, Colin Perkins <csp@csperkins.org> wrote:

> Hi,
>
> This is a well written draft. I think Section 6 needs work, along the
> lines of the recent discussion and updates to BUNDLE, but the rest of the
> document seems to be in good shape.
>
> Section 6 says:
>
>    Note: The following algorithm does not yet have WG consensus but is
>    included here as something concrete for the working group to discuss.
>
> I don=E2=80=99t think what=E2=80=99s written here is right yet. The RTP s=
pecification has
> rules for forming RTP packets into RTP streams based on their SSRC, and f=
or
> associating RTCP packets with RTP streams. Similarly, BUNDLE, as of -35,
> has rules for mapping RTP streams to m=3D sections based on the MID. This
> section seems to be trying to replicate these, rather than building on
> them.
>
> I think this section should leave the mapping from RTP/RTCP packets to RT=
P
> streams to RTP, and the mapping from RTP streams to m=3D sections to SDP =
(and
> BUNDLE, if used). It should talk instead about how RTP streams associated
> with m=3D sections are mapped onto RtpSenders and RtpReceivers. This may
> sound like just a terminology difference, but I think it=E2=80=99s import=
ant to get
> the layering and division of responsibilities right, if the protocol is t=
o
> be maintainable and extensible.
>
>
> I also had some minor nits on the rest of the document:
>
> - Section 5.1.1, 1st paragraph: The list of mandatory to implement
> standards is =E2=80=9Cderived from the requirements outlined in
> [I-D.ietf-rtcweb-rtp-usage]=E2=80=9D. Should the draft say somewhere that=
 =E2=80=9CRTP
> media transport MUST be implemented according to
> [I-D.ietf-rtcweb-rtp-usage]=E2=80=9D?
>
> - Section 5.1.1: It=E2=80=99s easy to skim-read R-5 this as =E2=80=9CMUST=
=E2=80=9D, since all the
> other requirements are MUST. It might be more visually distinct if writte=
n
> as  =E2=80=9CSHALL NOT=E2=80=9D rather than =E2=80=9CMUST NOT=E2=80=9D.
>
> - Section 5.1.1: R-16 should be removed.
>
> - Section 5.2.1, top of page 38: The header extension is in Section 14 of
> BUNDLE, not Section 11 (also second bullet point on page 39).
>
> - Section 5.8, 3rd bullet on page 60: I suggest changing =E2=80=9CIf the =
MID
> header extension is supported, prepare to demux RTP data intended for thi=
s
> media section=E2=80=9C to =E2=80=9C=E2=80=A6prepare to demux RTP streams =
intended=E2=80=A6=E2=80=9D, to match the
> recent discussion around terminology in BUNDLE.
>
> Colin
>
>
>
>
> On 21 Oct 2016, at 20:25, Ted Hardie <ted.ietf@gmail.com> wrote:
>
> The chairs would like to start a working group last call on
> draft-ietf-rtcweb-jset-17 to end on November 9th, 2016, 17:00 KST.
>
> Please review thoroughly, as working through the last comments will be th=
e
> major effort of our working group meeting in Seoul.
>
> From a process perspective, we will be working through the comments as
> issues against the github repo.  In order to accommodate folks who do not
> use github, the chairs will enter a new issue for any issue raised on the
> mailing list.  That means that if you are raising a new issue, please sen=
d
> it to the list with a new subject line so that the chairs catch it.
>
> If you prefer, you may also enter the issues at
> https://github.com/rtcweb-wg/jsep.  If you do so, please send a copy of
> the issue to the mailing list so that conversation and resolution occurs
> here.
>
> This has been a long road, but this a very major milestone.  Thanks to th=
e
> authors for their work so far.
>
> regards,
>
> Ted, Cullen, Sean
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> This draft is a work item of the Real-Time Communication in WEB-browsers
> of the IETF.
>
>         Title           : Javascript Session Establishment Protocol
>         Authors         : Justin Uberti
>                           Cullen Jennings
>                           Eric Rescorla
>         Filename        : draft-ietf-rtcweb-jsep-17.txt
>         Pages           : 94
>         Date            : 2016-10-21
>
> Abstract:
>    This document describes the mechanisms for allowing a Javascript
>    application to control the signaling plane of a multimedia session
>    via the interface specified in the W3C RTCPeerConnection API, and
>    discusses how this relates to existing signaling protocols.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/
>
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-17
>
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-jsep-17
>
>
> Please note that it may take a couple of minutes from the time of
> submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
> --
> Colin Perkins
> https://csperkins.org/
>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--94eb2c0b70fe861ad2053fb454aa
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Colin said:=C2=A0<div><br></div><div>&quot;<span style=3D"=
font-size:12.8px">I don=E2=80=99t think what=E2=80=99s written here is righ=
t yet. The RTP specification has rules for forming RTP packets into RTP str=
eams based on their SSRC, and for associating RTCP packets with RTP streams=
. Similarly, BUNDLE, as of -35, has rules for mapping RTP streams to m=3D s=
ections based on the MID. This section seems to be trying to replicate thes=
e, rather than building on them.&quot;</span></div><div style=3D"font-size:=
12.8px"><br></div><div><span style=3D"font-size:12.8px">I think this sectio=
n should leave the mapping from RTP/RTCP packets to RTP streams to RTP, and=
 the mapping from RTP streams to m=3D sections to SDP (and BUNDLE, if used)=
. It should talk instead about how RTP streams associated with m=3D section=
s are mapped onto RtpSenders and RtpReceivers. This may sound like just a t=
erminology difference, but I think it=E2=80=99s important to get the layeri=
ng and division of responsibilities right, if the protocol is to be maintai=
nable and extensible.</span>&quot;</div><div><br></div><div><span style=3D"=
font-size:12.8px">[BA] The advice given in the JSEP document is more detail=
ed than what is provided in BUNDLE and also covers situations (overlapping =
payload types) that BUNDLE mandates packet drops for.=C2=A0 So I don&#39;t =
believe that the material is entirely overlapping. =C2=A0</span></div></div=
><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tue, Oct 25, =
2016 at 10:42 AM, Colin Perkins <span dir=3D"ltr">&lt;<a href=3D"mailto:csp=
@csperkins.org" target=3D"_blank">csp@csperkins.org</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 style=3D"word-wrap:break-word">Hi,<di=
v><br></div><div>This is a well written draft. I think Section 6 needs work=
, along the lines of the recent discussion and updates to BUNDLE, but the r=
est of the document seems to be in good shape.</div><div><br></div><div>Sec=
tion 6 says:</div><div><br></div><div><div>=C2=A0 =C2=A0Note: The following=
 algorithm does not yet have WG consensus but is</div><div>=C2=A0 =C2=A0inc=
luded here as something concrete for the working group to discuss.</div></d=
iv><div><br></div><div>I don=E2=80=99t think what=E2=80=99s written here is=
 right yet. The RTP specification has rules for forming RTP packets into RT=
P streams based on their SSRC, and for associating RTCP packets with RTP st=
reams. Similarly, BUNDLE, as of -35, has rules for mapping RTP streams to m=
=3D sections based on the MID. This section seems to be trying to replicate=
 these, rather than building on them.=C2=A0</div><div><br></div><div>I thin=
k this section should leave the mapping from RTP/RTCP packets to RTP stream=
s to RTP, and the mapping from RTP streams to m=3D sections to SDP (and BUN=
DLE, if used). It should talk instead about how RTP streams associated with=
 m=3D sections are mapped onto RtpSenders and RtpReceivers. This may sound =
like just a terminology difference, but I think it=E2=80=99s important to g=
et the layering and division of responsibilities right, if the protocol is =
to be maintainable and extensible.</div><div><br></div><div><br></div><div>=
I also had some minor nits on the rest of the document:</div><div><br></div=
><div><div>- Section 5.1.1, 1st paragraph: The list of mandatory to impleme=
nt standards is =E2=80=9Cderived from the requirements outlined in [I-D.iet=
f-rtcweb-rtp-usage]=E2=80=9D. Should the draft say somewhere that =E2=80=9C=
RTP media transport MUST be implemented according to [I-D.ietf-rtcweb-rtp-u=
sage]=E2=80=9D?=C2=A0</div><div><br></div><div>- Section 5.1.1: It=E2=80=99=
s easy to skim-read R-5 this as =E2=80=9CMUST=E2=80=9D, since all the other=
 requirements are MUST. It might be more visually distinct if written as =
=C2=A0=E2=80=9CSHALL NOT=E2=80=9D rather than =E2=80=9CMUST NOT=E2=80=9D.</=
div><div><br></div><div>- Section 5.1.1: R-16 should be removed.</div><div>=
<br></div><div>- Section 5.2.1, top of page 38: The header extension is in =
Section 14 of BUNDLE, not Section 11 (also second bullet point on page 39).=
</div><div><br></div><div>- Section 5.8, 3rd bullet on page 60: I suggest c=
hanging =E2=80=9CIf the MID header extension is supported, prepare to demux=
 RTP data intended for this media section=E2=80=9C to =E2=80=9C=E2=80=A6pre=
pare to demux RTP streams intended=E2=80=A6=E2=80=9D, to match the recent d=
iscussion around terminology in BUNDLE.</div></div><div><br></div><div>Coli=
n</div><div><br></div><div><br></div><div><br></div><div><div><div class=3D=
"h5"><br><div><blockquote type=3D"cite"><div>On 21 Oct 2016, at 20:25, Ted =
Hardie &lt;<a href=3D"mailto:ted.ietf@gmail.com" target=3D"_blank">ted.ietf=
@gmail.com</a>&gt; wrote:</div><br class=3D"m_7891120878399878764Apple-inte=
rchange-newline"><div><div dir=3D"ltr"><div><div>The chairs would like to s=
tart a working group last call on draft-ietf-rtcweb-jset-17 to end on Novem=
ber 9th, 2016, 17:00 KST.<br>=C2=A0 <br>Please review thoroughly, as workin=
g through the last comments will be the major effort of our working group m=
eeting in Seoul.<br><br></div>From a process perspective, we will be workin=
g through the comments as issues against the github repo.=C2=A0 In order to=
 accommodate folks who do not use github, the chairs will enter a new issue=
 for any issue raised on the mailing list.=C2=A0 That means that if you are=
 raising a new issue, please send it to the list with a new subject line so=
 that the chairs catch it.<br><br></div>If you prefer, you may also enter t=
he issues at <a href=3D"https://github.com/rtcweb-wg/jsep" target=3D"_blank=
">https://github.com/rtcweb-wg/<wbr>jsep</a>.=C2=A0 If you do so, please se=
nd a copy of the issue to the mailing list so that conversation and resolut=
ion occurs here.<br><div><div><div><div class=3D"gmail_quote"><br></div><di=
v class=3D"gmail_quote">This has been a long road, but this a very major mi=
lestone.=C2=A0 Thanks to the authors for their work so far.<br><br></div><d=
iv class=3D"gmail_quote">regards,<br><br></div><div class=3D"gmail_quote">T=
ed, Cullen, Sean<br></div><div class=3D"gmail_quote"><br><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
This draft is a work item of the Real-Time Communication in WEB-browsers of=
 the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 Javascript Session Establishment Protocol<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Authors=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: Just=
in Uberti<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Cullen Jennings<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Eric Rescorla<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-jsep-17.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 94<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2016-10-21<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document describes the mechanisms for allowing a Javascri=
pt<br>
=C2=A0 =C2=A0application to control the signaling plane of a multimedia ses=
sion<br>
=C2=A0 =C2=A0via the interface specified in the W3C RTCPeerConnection API, =
and<br>
=C2=A0 =C2=A0discusses how this relates to existing signaling protocols.<br=
>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/" rel=3D=
"noreferrer" target=3D"_blank">https://datatracker.ietf.org/d<wbr>oc/draft-=
ietf-rtcweb-jsep/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-17" rel=3D"no=
referrer" target=3D"_blank">https://tools.ietf.org/html/dr<wbr>aft-ietf-rtc=
web-jsep-17</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-jsep-17" r=
el=3D"noreferrer" target=3D"_blank">https://www.ietf.org/rfcdiff?u<wbr>rl2=
=3Ddraft-ietf-rtcweb-jsep-17</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org/" rel=3D"noreferrer" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" rel=3D"noreferrer" target=
=3D"_blank">ftp://ftp.ietf.org/internet-dr<wbr>afts/</a><br>
<br>
______________________________<wbr>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/rtcweb</a><br=
>
</div><br></div></div></div></div>
______________________________<wbr>_________________<br>rtcweb mailing list=
<br><a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a=
><br><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/<wbr>listinfo/rtcweb</a><br></div></block=
quote></div><br></div></div><span class=3D"HOEnZb"><font color=3D"#888888">=
<div>
<span class=3D"m_7891120878399878764Apple-style-span" style=3D"border-colla=
pse:separate;font-variant-ligatures:normal;font-variant-numeric:normal;font=
-variant-alternates:normal;font-variant-east-asian:normal;line-height:norma=
l;border-spacing:0px"><span class=3D"m_7891120878399878764Apple-style-span"=
 style=3D"font-size:9px"><div><br class=3D"m_7891120878399878764Apple-inter=
change-newline"><br class=3D"m_7891120878399878764khtml-block-placeholder">=
</div><div>--=C2=A0</div><div></div><div>Colin Perkins</div><div><a href=3D=
"https://csperkins.org/" target=3D"_blank">https://csperkins.org/</a></div>=
<div><br></div></span></span><br class=3D"m_7891120878399878764Apple-interc=
hange-newline"><br class=3D"m_7891120878399878764Apple-interchange-newline"=
>
</div>
<br></font></span></div></div><br>______________________________<wbr>______=
___________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinfo/rtcweb</a><br=
>
<br></blockquote></div><br></div>

--94eb2c0b70fe861ad2053fb454aa--


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From: Ted Hardie <ted.ietf@gmail.com>
Date: Tue, 25 Oct 2016 11:11:42 -0700
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Working Group Last Call: JSEP
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--001a1141ad8a98d5ab053fb4723d
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Hi Colin,

On Tue, Oct 25, 2016 at 10:42 AM, Colin Perkins <csp@csperkins.org> wrote:

> Hi,
>
> This is a well written draft. I think Section 6 needs work, along the
> lines of the recent discussion and updates to BUNDLE, but the rest of the
> document seems to be in good shape.
>
> Section 6 says:
>
>    Note: The following algorithm does not yet have WG consensus but is
>    included here as something concrete for the working group to discuss.
>
> I don=E2=80=99t think what=E2=80=99s written here is right yet. The RTP s=
pecification has
> rules for forming RTP packets into RTP streams based on their SSRC, and f=
or
> associating RTCP packets with RTP streams. Similarly, BUNDLE, as of -35,
> has rules for mapping RTP streams to m=3D sections based on the MID. This
> section seems to be trying to replicate these, rather than building on
> them.
>
> I think this section should leave the mapping from RTP/RTCP packets to RT=
P
> streams to RTP, and the mapping from RTP streams to m=3D sections to SDP =
(and
> BUNDLE, if used). It should talk instead about how RTP streams associated
> with m=3D sections are mapped onto RtpSenders and RtpReceivers. This may
> sound like just a terminology difference, but I think it=E2=80=99s import=
ant to get
> the layering and division of responsibilities right, if the protocol is t=
o
> be maintainable and extensible.
>
>
I've entered the above as an issue in the tracker:
https://github.com/rtcweb-wg/jsep/issues/364 .

Further commentary on the list, please.



>
> I also had some minor nits on the rest of the document:
>
> - Section 5.1.1, 1st paragraph: The list of mandatory to implement
> standards is =E2=80=9Cderived from the requirements outlined in
> [I-D.ietf-rtcweb-rtp-usage]=E2=80=9D. Should the draft say somewhere that=
 =E2=80=9CRTP
> media transport MUST be implemented according to
> [I-D.ietf-rtcweb-rtp-usage]=E2=80=9D?
>
> - Section 5.1.1: It=E2=80=99s easy to skim-read R-5 this as =E2=80=9CMUST=
=E2=80=9D, since all the
> other requirements are MUST. It might be more visually distinct if writte=
n
> as  =E2=80=9CSHALL NOT=E2=80=9D rather than =E2=80=9CMUST NOT=E2=80=9D.
>
> - Section 5.1.1: R-16 should be removed.
>
> - Section 5.2.1, top of page 38: The header extension is in Section 14 of
> BUNDLE, not Section 11 (also second bullet point on page 39).
>
> - Section 5.8, 3rd bullet on page 60: I suggest changing =E2=80=9CIf the =
MID
> header extension is supported, prepare to demux RTP data intended for thi=
s
> media section=E2=80=9C to =E2=80=9C=E2=80=A6prepare to demux RTP streams =
intended=E2=80=A6=E2=80=9D, to match the
> recent discussion around terminology in BUNDLE.
>
>
I've entered these as a single editorial issue, at the moment,
https://github.com/rtcweb-wg/jsep/issues/365 .  If any need further
discussion, we can break the relevant pieces out.

Thanks for your review!

Ted



> Colin
>
>
>
>
> On 21 Oct 2016, at 20:25, Ted Hardie <ted.ietf@gmail.com> wrote:
>
> The chairs would like to start a working group last call on
> draft-ietf-rtcweb-jset-17 to end on November 9th, 2016, 17:00 KST.
>
> Please review thoroughly, as working through the last comments will be th=
e
> major effort of our working group meeting in Seoul.
>
> From a process perspective, we will be working through the comments as
> issues against the github repo.  In order to accommodate folks who do not
> use github, the chairs will enter a new issue for any issue raised on the
> mailing list.  That means that if you are raising a new issue, please sen=
d
> it to the list with a new subject line so that the chairs catch it.
>
> If you prefer, you may also enter the issues at
> https://github.com/rtcweb-wg/jsep.  If you do so, please send a copy of
> the issue to the mailing list so that conversation and resolution occurs
> here.
>
> This has been a long road, but this a very major milestone.  Thanks to th=
e
> authors for their work so far.
>
> regards,
>
> Ted, Cullen, Sean
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> This draft is a work item of the Real-Time Communication in WEB-browsers
> of the IETF.
>
>         Title           : Javascript Session Establishment Protocol
>         Authors         : Justin Uberti
>                           Cullen Jennings
>                           Eric Rescorla
>         Filename        : draft-ietf-rtcweb-jsep-17.txt
>         Pages           : 94
>         Date            : 2016-10-21
>
> Abstract:
>    This document describes the mechanisms for allowing a Javascript
>    application to control the signaling plane of a multimedia session
>    via the interface specified in the W3C RTCPeerConnection API, and
>    discusses how this relates to existing signaling protocols.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/
>
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-17
>
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-jsep-17
>
>
> Please note that it may take a couple of minutes from the time of
> submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
> --
> Colin Perkins
> https://csperkins.org/
>
>
>
>
>

--001a1141ad8a98d5ab053fb4723d
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi Colin,<br><div><div class=3D"gmail_extra"><br><div clas=
s=3D"gmail_quote">On Tue, Oct 25, 2016 at 10:42 AM, Colin Perkins <span dir=
=3D"ltr">&lt;<a href=3D"mailto:csp@csperkins.org" target=3D"_blank">csp@csp=
erkins.org</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding=
-left:1ex"><div style=3D"overflow-wrap: break-word;">Hi,<div><br></div><div=
>This is a well written draft. I think Section 6 needs work, along the line=
s of the recent discussion and updates to BUNDLE, but the rest of the docum=
ent seems to be in good shape.</div><div><br></div><div>Section 6 says:</di=
v><div><br></div><div><div>=C2=A0 =C2=A0Note: The following algorithm does =
not yet have WG consensus but is</div><div>=C2=A0 =C2=A0included here as so=
mething concrete for the working group to discuss.</div></div><div><br></di=
v><div>I don=E2=80=99t think what=E2=80=99s written here is right yet. The =
RTP specification has rules for forming RTP packets into RTP streams based =
on their SSRC, and for associating RTCP packets with RTP streams. Similarly=
, BUNDLE, as of -35, has rules for mapping RTP streams to m=3D sections bas=
ed on the MID. This section seems to be trying to replicate these, rather t=
han building on them.=C2=A0</div><div><br></div><div>I think this section s=
hould leave the mapping from RTP/RTCP packets to RTP streams to RTP, and th=
e mapping from RTP streams to m=3D sections to SDP (and BUNDLE, if used). I=
t should talk instead about how RTP streams associated with m=3D sections a=
re mapped onto RtpSenders and RtpReceivers. This may sound like just a term=
inology difference, but I think it=E2=80=99s important to get the layering =
and division of responsibilities right, if the protocol is to be maintainab=
le and extensible.</div><div><br></div></div></blockquote><div><br></div><d=
iv>I&#39;ve entered the above as an issue in the tracker:=C2=A0 <a href=3D"=
https://github.com/rtcweb-wg/jsep/issues/364">https://github.com/rtcweb-wg/=
jsep/issues/364</a> .<br><br></div><div>Further commentary on the list, ple=
ase.<br><br>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0=
px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><=
div style=3D"overflow-wrap: break-word;"><div></div><div><br></div><div>I a=
lso had some minor nits on the rest of the document:</div><div><br></div><d=
iv><div>- Section 5.1.1, 1st paragraph: The list of mandatory to implement =
standards is =E2=80=9Cderived from the requirements outlined in [I-D.ietf-r=
tcweb-rtp-usage]=E2=80=9D. Should the draft say somewhere that =E2=80=9CRTP=
 media transport MUST be implemented according to [I-D.ietf-rtcweb-rtp-usag=
e]=E2=80=9D?=C2=A0</div><div><br></div><div>- Section 5.1.1: It=E2=80=99s e=
asy to skim-read R-5 this as =E2=80=9CMUST=E2=80=9D, since all the other re=
quirements are MUST. It might be more visually distinct if written as =C2=
=A0=E2=80=9CSHALL NOT=E2=80=9D rather than =E2=80=9CMUST NOT=E2=80=9D.</div=
><div><br></div><div>- Section 5.1.1: R-16 should be removed.</div><div><br=
></div><div>- Section 5.2.1, top of page 38: The header extension is in Sec=
tion 14 of BUNDLE, not Section 11 (also second bullet point on page 39).</d=
iv><div><br></div><div>- Section 5.8, 3rd bullet on page 60: I suggest chan=
ging =E2=80=9CIf the MID header extension is supported, prepare to demux RT=
P data intended for this media section=E2=80=9C to =E2=80=9C=E2=80=A6prepar=
e to demux RTP streams intended=E2=80=A6=E2=80=9D, to match the recent disc=
ussion around terminology in BUNDLE.</div></div><div><br></div></div></bloc=
kquote><div><br></div><div>I&#39;ve entered these as a single editorial iss=
ue, at the moment, <a href=3D"https://github.com/rtcweb-wg/jsep/issues/365"=
>https://github.com/rtcweb-wg/jsep/issues/365</a> .=C2=A0 If any need furth=
er discussion, we can break the relevant pieces out.<br><br></div><div>Than=
ks for your review!<br><br></div><div>Ted<br></div><div><br>=C2=A0</div><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-lef=
t:1px solid rgb(204,204,204);padding-left:1ex"><div style=3D"overflow-wrap:=
 break-word;"><div></div><div>Colin</div><div><br></div><div><br></div><div=
><br></div><div><div><div class=3D"gmail-h5"><br><div><blockquote type=3D"c=
ite"><div>On 21 Oct 2016, at 20:25, Ted Hardie &lt;<a href=3D"mailto:ted.ie=
tf@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt; wrote:</div><br =
class=3D"gmail-m_-5384951257281185921Apple-interchange-newline"><div><div d=
ir=3D"ltr"><div><div>The chairs would like to start a working group last ca=
ll on draft-ietf-rtcweb-jset-17 to end on November 9th, 2016, 17:00 KST.<br=
>=C2=A0 <br>Please review thoroughly, as working through the last comments =
will be the major effort of our working group meeting in Seoul.<br><br></di=
v>From a process perspective, we will be working through the comments as is=
sues against the github repo.=C2=A0 In order to accommodate folks who do no=
t use github, the chairs will enter a new issue for any issue raised on the=
 mailing list.=C2=A0 That means that if you are raising a new issue, please=
 send it to the list with a new subject line so that the chairs catch it.<b=
r><br></div>If you prefer, you may also enter the issues at <a href=3D"http=
s://github.com/rtcweb-wg/jsep" target=3D"_blank">https://github.com/rtcweb-=
wg/<wbr>jsep</a>.=C2=A0 If you do so, please send a copy of the issue to th=
e mailing list so that conversation and resolution occurs here.<br><div><di=
v><div><div class=3D"gmail_quote"><br></div><div class=3D"gmail_quote">This=
 has been a long road, but this a very major milestone.=C2=A0 Thanks to the=
 authors for their work so far.<br><br></div><div class=3D"gmail_quote">reg=
ards,<br><br></div><div class=3D"gmail_quote">Ted, Cullen, Sean<br></div><d=
iv class=3D"gmail_quote"><br><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
This draft is a work item of the Real-Time Communication in WEB-browsers of=
 the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 Javascript Session Establishment Protocol<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Authors=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: Just=
in Uberti<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Cullen Jennings<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Eric Rescorla<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-jsep-17.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 94<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2016-10-21<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document describes the mechanisms for allowing a Javascri=
pt<br>
=C2=A0 =C2=A0application to control the signaling plane of a multimedia ses=
sion<br>
=C2=A0 =C2=A0via the interface specified in the W3C RTCPeerConnection API, =
and<br>
=C2=A0 =C2=A0discusses how this relates to existing signaling protocols.<br=
>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/" rel=3D=
"noreferrer" target=3D"_blank">https://datatracker.ietf.org/d<wbr>oc/draft-=
ietf-rtcweb-jsep/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-17" rel=3D"no=
referrer" target=3D"_blank">https://tools.ietf.org/html/dr<wbr>aft-ietf-rtc=
web-jsep-17</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-jsep-17" r=
el=3D"noreferrer" target=3D"_blank">https://www.ietf.org/rfcdiff?u<wbr>rl2=
=3Ddraft-ietf-rtcweb-jsep-17</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org/" rel=3D"noreferrer" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" rel=3D"noreferrer" target=
=3D"_blank">ftp://ftp.ietf.org/internet-dr<wbr>afts/</a><br>
<br>
______________________________<wbr>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/rtcweb</a><br=
>
</div><br></div></div></div></div>
______________________________<wbr>_________________<br>rtcweb mailing list=
<br><a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a=
><br><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/<wbr>listinfo/rtcweb</a><br></div></block=
quote></div><br></div></div><span class=3D"gmail-HOEnZb"><font color=3D"#88=
8888"><div>
<span class=3D"gmail-m_-5384951257281185921Apple-style-span" style=3D"borde=
r-collapse:separate;font-variant-ligatures:normal;font-variant-numeric:norm=
al;font-variant-alternates:normal;font-variant-east-asian:normal;line-heigh=
t:normal;border-spacing:0px"><span class=3D"gmail-m_-5384951257281185921App=
le-style-span" style=3D"font-size:9px"><div><br class=3D"gmail-m_-538495125=
7281185921Apple-interchange-newline"><br class=3D"gmail-m_-5384951257281185=
921khtml-block-placeholder"></div><div>--=C2=A0</div><div></div><div>Colin =
Perkins</div><div><a href=3D"https://csperkins.org/" target=3D"_blank">http=
s://csperkins.org/</a></div><div><br></div></span></span><br class=3D"gmail=
-m_-5384951257281185921Apple-interchange-newline"><br class=3D"gmail-m_-538=
4951257281185921Apple-interchange-newline">
</div>
<br></font></span></div></div></blockquote></div><br></div></div></div>

--001a1141ad8a98d5ab053fb4723d--


From nobody Wed Oct 26 08:24:38 2016
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-transports-17.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Real-Time Communication in WEB-browsers of the IETF.

        Title           : Transports for WebRTC
        Author          : Harald Alvestrand
	Filename        : draft-ietf-rtcweb-transports-17.txt
	Pages           : 20
	Date            : 2016-10-26

Abstract:
   This document describes the data transport protocols used by WebRTC,
   including the protocols used for interaction with intermediate boxes
   such as firewalls, relays and NAT boxes.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-transports-17

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-17


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


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--001a114485f436b9fd053fce2f71
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Some comments.

      Construct a table mapping payload type to RtpReceiver for each
      RtpReceiver configured to receive from this transport and for each
      payload type that RtpReceiver is configured to receive.  The
      payload types of a given RtpReceiver are found in the m= section
      corresponding to that RtpReceiver in the local description.  If
      any payload type could map to more than one RtpReceiver, map to
      the RtpReceiver whose m= section appears earliest in the local
      description.


[BA] In the case where a payload type can map to  more than one
RtpReceiver, what if the earliest m-line is "send-only" or "inactive"?
 Wouldn't you want to map to the  RtpReceiver whose m-line is
"recv-only" or "send-recv" and is earliest?


      If the packet's payload type is in the payload type table, update
      the the incoming SSRC mapping table to include an entry that maps
      the packet's SSRC to the RtpReceiver for that payload type.  In
      addition, deliver the packet to the associated RtpReceiver and
      stop.


[BA] I believe that PT-based SSRC latching breaks two of Cullen's
proposed use cases (see:
https://mailarchive.ietf.org/arch/msg/rtcweb/cvcpy-Rnj4ntL66haNnKHy2_Yvw
):


Codec change, case 1: The PTs are all unique, and no SSRCs were signaled in
SDP  There is no RID, MID, FEC or RTX. Sender switches codecs, and the PT
changes (but SSRC stays the same).  The packet goes to the receiver that
matches the new PT.

Codec change, case 2:  The PTs are all unique and SSRCs are signaled in
SDP.  There is no RID, MID, FEC or RTX. Sender switches codecs and the PT
changes (but SSRC stays the same).  SSRC does not match any signaled ones.
The packet goes to the receiver that matches the new PT.

[BA] For the RTCP routing, it is more clear to use the RFC 3550 terms
for the various SSRC fields.  For exchange, change:

      "If the packet is of type SR, and the sender SSRC for the packet is
      found in the incoming SSRC table, deliver a copy of the packet to
      the RtpReceiver associated with that SSRC.  In addition, for each
      report block in the report whose SSRC is found in the outgoing
      SSRC table, deliver a copy of the RTCP packet to the RtpSender
      associated with that SSRC."

To:


      "If the packet is of type SR, and the "SSRC of sender" for the packet is
      found in the incoming SSRC table, deliver a copy of the packet to
      the RtpReceiver associated with that SSRC.  In addition, for each
      report block in the report whose "SSRC of source" is found in the outgoing
      SSRC table, deliver a copy of the RTCP packet to the RtpSender
      associated with that SSRC."



[BA] Change:


      "If the packet is of type RR, for each report block in the packet
      whose SSRC is found in the outgoing SSRC table, deliver a copy of
      the RTCP packet to the RtpSender associated with that SSRC."


To:


      "If the packet is of type RR, for each report block in the packet
      whose "SSRC of source" is found in the outgoing SSRC table,
deliver a copy of
      the RTCP packet to the RtpSender associated with that SSRC."



[BA] The RTCP SDES packet has no "SSRC of packet sender" field, so change:


      "If the packet is of type SDES, and the sender SSRC for the packet
      is found in the incoming SSRC table, deliver the packet to the
      RtpReceiver associated with that SSRC.  In addition, for each
      chunk in the packet that contains a MID that is in the table
      mapping MID to RtpReceiver, update the incoming SSRC mapping table
      to include an entry that maps the SSRC for that chunk to the
      RtpReceiver associated with that MID.  (This case can occur when
      RTCP for a source is received before any RTP packets.)"


To:


      "If the packet is of type SDES, for each chunk containing

      an "SSRC/CSRC" field found in the incoming SSRC table,

      deliver the packet to the RtpReceiver associated with that SSRC.

      In addition, for each chunk in the packet that contains a MID
that is in the table
      mapping MID to RtpReceiver, update the incoming SSRC mapping table
      to include an entry that maps the SSRC for that chunk to the
      RtpReceiver associated with that MID. (This case can occur when
      RTCP for a source is received before any RTP packets.)"


[BA] Change:



      "If the packet is of type BYE, for each SSRC indicated in the
      packet that is found in the incoming SSRC table, deliver a copy of
      the packet to the RtpReceiver associated with that SSRC."


To:



      "If the packet is of type BYE, for each "SSRC/CSRC" indicated in the
      packet that is found in the incoming SSRC table, deliver a copy of
      the packet to the RtpReceiver associated with that SSRC."


[BA] Change:


      "If the packet is of type RTPFB or PSFB, as defined in [RFC4585
<https://tools.ietf.org/html/rfc4585>],
      and the media source SSRC for the packet is found in the outgoing
      SSRC table, deliver the packet to the RtpSender associated with
      that SSRC."


To:


      "If the packet is of type RTPFB or PSFB, as defined in [RFC4585
<https://tools.ietf.org/html/rfc4585>],
      and the "SSRC of media source" for the packet is found in the outgoing
      SSRC table, deliver the packet to the RtpSender associated with
      that SSRC."


[BA] Since FIR feedback messages can contain multiple FCI fields, I think you

need to add the following text (see RFC 5104 Section 4.3.1.1):


      "In addition, for FIR feedback messages, as defined in [RFC5104],

      for each "SSRC" indicated in an FCI entry, deliver a copy of the

      packet to the RtpSender associated with that SSRC.

--001a114485f436b9fd053fce2f71
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Some comments.<div><br></div><div><pre class=3D"gmail-newp=
age" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rg=
b(0,0,0)">      Construct a table mapping payload type to RtpReceiver for e=
ach
      RtpReceiver configured to receive from this transport and for each
      payload type that RtpReceiver is configured to receive.  The
      payload types of a given RtpReceiver are found in the m=3D section
      corresponding to that RtpReceiver in the local description.  If
      any payload type could map to more than one RtpReceiver, map to
      the RtpReceiver whose m=3D section appears earliest in the local
      description.</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.=
3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre cl=
ass=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bo=
ttom:0px;color:rgb(0,0,0)"><span style=3D"color:rgb(34,34,34);font-family:a=
rial,sans-serif;font-size:small;white-space:normal">[BA] In the case where =
a payload type can map to =C2=A0more than one RtpReceiver, w</span><span st=
yle=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;whi=
te-space:normal">hat if the earliest m-line is &quot;send-only&quot; or &qu=
ot;inactive&quot;?=C2=A0 Wouldn&#39;t you want to map to the =C2=A0RtpRecei=
ver whose m-line is &quot;recv-only&quot; or &quot;send-recv&quot; and is e=
arliest?=C2=A0</span></pre><pre class=3D"gmail-newpage" style=3D"font-size:=
13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span style=3D=
"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;white-spa=
ce:normal"><br></span></pre><pre class=3D"gmail-newpage" style=3D"margin-to=
p:0px;margin-bottom:0px"><pre class=3D"gmail-newpage" style=3D"color:rgb(0,=
0,0);font-size:13.3333px;margin-top:0px;margin-bottom:0px">      If the pac=
ket&#39;s payload type is in the payload type table, update
      the the incoming SSRC mapping table to include an entry that maps
      the packet&#39;s SSRC to the RtpReceiver for that payload type.  In
      addition, deliver the packet to the associated RtpReceiver and
      stop.</pre><pre class=3D"gmail-newpage" style=3D"color:rgb(0,0,0);fon=
t-size:13.3333px;margin-top:0px;margin-bottom:0px"><br></pre><pre class=3D"=
gmail-newpage" style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"ari=
al, sans-serif"><span style=3D"white-space:normal">[BA] I believe that PT-b=
ased SSRC latching breaks two of Cullen&#39;s proposed use cases (see:=C2=
=A0<a href=3D"https://mailarchive.ietf.org/arch/msg/rtcweb/cvcpy-Rnj4ntL66h=
aNnKHy2_Yvw">https://mailarchive.ietf.org/arch/msg/rtcweb/cvcpy-Rnj4ntL66ha=
NnKHy2_Yvw</a> ):=C2=A0</span></font></pre><pre class=3D"gmail-newpage" sty=
le=3D"margin-top:0px;margin-bottom:0px"><font face=3D"arial, sans-serif"><s=
pan style=3D"white-space:normal"><br></span></font></pre><pre class=3D"gmai=
l-newpage" style=3D"margin-top:0px;margin-bottom:0px"><pre class=3D"gmail-w=
ordwrap" style=3D"box-sizing:border-box;overflow:auto;font-family:menlo,mon=
aco,consolas,&quot;courier new&quot;,monospace;font-size:13px;padding:0px;m=
argin-top:0px;margin-bottom:10px;line-height:1.42857;word-break:normal;word=
-wrap:normal;color:rgb(51,51,51);border:0px none black;border-radius:4px;wh=
ite-space:pre-wrap">Codec change, case 1: The PTs are all unique, and no SS=
RCs were signaled in
SDP  There is no RID, MID, FEC or RTX. Sender switches codecs, and the PT
changes (but SSRC stays the same).  The packet goes to the receiver that
matches the new PT.</pre><pre class=3D"gmail-wordwrap" style=3D"box-sizing:=
border-box;overflow:auto;font-family:menlo,monaco,consolas,&quot;courier ne=
w&quot;,monospace;font-size:13px;padding:0px;margin-top:0px;margin-bottom:1=
0px;line-height:1.42857;word-break:normal;word-wrap:normal;color:rgb(51,51,=
51);border:0px none black;border-radius:4px;white-space:pre-wrap"><pre clas=
s=3D"gmail-wordwrap" style=3D"box-sizing:border-box;overflow:auto;font-fami=
ly:menlo,monaco,consolas,&quot;courier new&quot;,monospace;padding:0px;marg=
in-top:0px;margin-bottom:10px;line-height:1.42857;word-break:normal;word-wr=
ap:normal;border:0px none black;border-radius:4px;white-space:pre-wrap">Cod=
ec change, case 2:  The PTs are all unique and SSRCs are signaled in
SDP.  There is no RID, MID, FEC or RTX. Sender switches codecs and the PT
changes (but SSRC stays the same).  SSRC does not match any signaled ones.
The packet goes to the receiver that matches the new PT.</pre><pre class=3D=
"gmail-wordwrap" style=3D"box-sizing:border-box;overflow:auto;font-family:m=
enlo,monaco,consolas,&quot;courier new&quot;,monospace;padding:0px;margin-t=
op:0px;margin-bottom:10px;line-height:1.42857;word-break:normal;word-wrap:n=
ormal;border:0px none black;border-radius:4px;white-space:pre-wrap"><pre cl=
ass=3D"gmail-newpage" style=3D"color:rgb(34,34,34);font-size:small;margin-t=
op:0px;margin-bottom:0px"><font face=3D"arial, sans-serif"><span style=3D"w=
hite-space:normal">[BA] For the RTCP routing, it is more clear to use the R=
FC 3550 terms for the various SSRC fields.=C2=A0 For exchange, change:</spa=
n></font></pre></pre><pre class=3D"gmail-wordwrap" style=3D"box-sizing:bord=
er-box;overflow:auto;font-family:menlo,monaco,consolas,&quot;courier new&qu=
ot;,monospace;padding:0px;margin-top:0px;margin-bottom:10px;line-height:1.4=
2857;word-break:normal;word-wrap:normal;border:0px none black;border-radius=
:4px;white-space:pre-wrap"><pre class=3D"gmail-newpage" style=3D"font-size:=
13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">      &quot;If=
 the packet is of type SR, and the sender SSRC for the packet is
      found in the incoming SSRC table, deliver a copy of the packet to
      the RtpReceiver associated with that SSRC.  In addition, for each
      report block in the report whose SSRC is found in the outgoing
      SSRC table, deliver a copy of the RTCP packet to the RtpSender
      associated with that SSRC.&quot;</pre></pre></pre></pre><pre class=3D=
"gmail-newpage" style=3D"margin-top:0px;margin-bottom:0px"><font face=3D"ar=
ial, sans-serif"><span style=3D"white-space:normal">To:</span></font></pre>=
<pre class=3D"gmail-newpage" style=3D"margin-top:0px;margin-bottom:0px"><fo=
nt face=3D"arial, sans-serif"><span style=3D"white-space:normal"><br></span=
></font></pre><pre class=3D"gmail-newpage" style=3D"color:rgb(0,0,0);font-s=
ize:13.3333px;margin-top:0px;margin-bottom:0px"><pre class=3D"gmail-newpage=
" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">      &quo=
t;If the packet is of type SR, and the &quot;SSRC of sender&quot; for the p=
acket is
      found in the incoming SSRC table, deliver a copy of the packet to
      the RtpReceiver associated with that SSRC.  In addition, for each
      report block in the report whose &quot;SSRC of source&quot; is found =
in the outgoing
      SSRC table, deliver a copy of the RTCP packet to the RtpSender
      associated with that SSRC.&quot;</pre><pre class=3D"gmail-newpage" st=
yle=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px"><br></pre><pre=
 class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin=
-bottom:0px"><br></pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3=
333px;margin-top:0px;margin-bottom:0px"><span style=3D"color:rgb(34,34,34);=
font-family:arial,sans-serif;font-size:small;white-space:normal">[BA] Chang=
e:</span><br></pre></pre></pre><pre class=3D"gmail-newpage" style=3D"font-s=
ize:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span styl=
e=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;white=
-space:normal"><br></span></pre><pre class=3D"gmail-newpage" style=3D"font-=
size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><pre clas=
s=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bott=
om:0px">      &quot;If the packet is of type RR, for each report block in t=
he packet
      whose SSRC is found in the outgoing SSRC table, deliver a copy of
      the RTCP packet to the RtpSender associated with that SSRC.&quot;</pr=
e></pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-to=
p:0px;margin-bottom:0px;color:rgb(0,0,0)"><span style=3D"color:rgb(34,34,34=
);font-family:arial,sans-serif;font-size:small;white-space:normal"><br></sp=
an></pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-t=
op:0px;margin-bottom:0px;color:rgb(0,0,0)"><pre class=3D"gmail-newpage" sty=
le=3D"color:rgb(34,34,34);font-size:small;margin-top:0px;margin-bottom:0px"=
><font face=3D"arial, sans-serif"><span style=3D"white-space:normal">To:=C2=
=A0</span></font></pre><pre class=3D"gmail-newpage" style=3D"color:rgb(34,3=
4,34);font-size:small;margin-top:0px;margin-bottom:0px"><font face=3D"arial=
, sans-serif"><span style=3D"white-space:normal"><br></span></font></pre><p=
re class=3D"gmail-newpage" style=3D"color:rgb(34,34,34);font-size:small;mar=
gin-top:0px;margin-bottom:0px"><pre class=3D"gmail-newpage" style=3D"color:=
rgb(0,0,0);font-size:13.3333px;margin-top:0px;margin-bottom:0px">      &quo=
t;If the packet is of type RR, for each report block in the packet
      whose <span style=3D"font-size:13.3333px;font-family:arial,sans-serif=
">&quot;SSRC of source&quot;</span><span style=3D"font-size:13.3333px;font-=
family:arial,sans-serif"> is found in the outgoing SSRC table, deliver a co=
py of</span><br>      the RTCP packet to the RtpSender associated with that=
 SSRC.&quot;</pre><pre class=3D"gmail-newpage" style=3D"color:rgb(0,0,0);fo=
nt-size:13.3333px;margin-top:0px;margin-bottom:0px"><br></pre><pre class=3D=
"gmail-newpage" style=3D"color:rgb(0,0,0);font-size:13.3333px;margin-top:0p=
x;margin-bottom:0px"><br></pre><pre class=3D"gmail-newpage" style=3D"color:=
rgb(0,0,0);font-size:13.3333px;margin-top:0px;margin-bottom:0px"><span styl=
e=3D"color:rgb(34,34,34);font-family:arial,sans-serif;font-size:small;white=
-space:normal">[BA] The RTCP SDES packet has no &quot;SSRC of packet sender=
&quot; field, so change:=C2=A0</span><br></pre><pre class=3D"gmail-newpage"=
 style=3D"color:rgb(0,0,0);font-size:13.3333px;margin-top:0px;margin-bottom=
:0px"><br></pre><pre class=3D"gmail-newpage" style=3D"color:rgb(0,0,0);font=
-size:13.3333px;margin-top:0px;margin-bottom:0px"><pre class=3D"gmail-newpa=
ge" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">      &q=
uot;If the packet is of type SDES, and the sender SSRC for the packet
      is found in the incoming SSRC table, deliver the packet to the
      RtpReceiver associated with that SSRC.  In addition, for each
      chunk in the packet that contains a MID that is in the table
      mapping MID to RtpReceiver, update the incoming SSRC mapping table
      to include an entry that maps the SSRC for that chunk to the
      RtpReceiver associated with that MID.  (This case can occur when
      RTCP for a source is received before any RTP packets.)&quot;</pre><pr=
e class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margi=
n-bottom:0px"><br></pre><pre class=3D"gmail-newpage" style=3D"font-size:13.=
3333px;margin-top:0px;margin-bottom:0px">To:</pre><pre class=3D"gmail-newpa=
ge" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px"><br></pr=
e><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;=
margin-bottom:0px">      &quot;If the packet is of type SDES, for each chun=
k containing</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px=
;margin-top:0px;margin-bottom:0px">      an &quot;SSRC/CSRC&quot; field fou=
nd in the incoming SSRC table,</pre><pre class=3D"gmail-newpage" style=3D"f=
ont-size:13.3333px;margin-top:0px;margin-bottom:0px">      deliver the pack=
et to the RtpReceiver associated with that SSRC.</pre><pre class=3D"gmail-n=
ewpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">    =
  In addition, for each chunk in the packet that contains a MID that is in =
the table
      mapping MID to RtpReceiver, update the incoming SSRC mapping table
      to include an entry that maps the SSRC for that chunk to the
      RtpReceiver associated with that MID. (This case can occur when
      RTCP for a source is received before any RTP packets.)&quot; </pre><p=
re class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;marg=
in-bottom:0px"><br></pre><pre class=3D"gmail-newpage" style=3D"font-size:13=
.3333px;margin-top:0px;margin-bottom:0px">[BA] Change: </pre><pre class=3D"=
gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0p=
x"><br></pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;marg=
in-top:0px;margin-bottom:0px"><br></pre><pre class=3D"gmail-newpage" style=
=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px"><pre class=3D"gma=
il-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">=
      &quot;If the packet is of type BYE, for each SSRC indicated in the
      packet that is found in the incoming SSRC table, deliver a copy of
      the packet to the RtpReceiver associated with that SSRC.&quot;</pre><=
/pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0=
px;margin-bottom:0px"><br></pre><pre class=3D"gmail-newpage" style=3D"font-=
size:13.3333px;margin-top:0px;margin-bottom:0px"><span style=3D"font-size:1=
3.3333px">To:</span></pre><pre class=3D"gmail-newpage" style=3D"font-size:1=
3.3333px;margin-top:0px;margin-bottom:0px"><span style=3D"font-size:13.3333=
px;font-family:arial,sans-serif"><br></span></pre><pre class=3D"gmail-newpa=
ge" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px"><span st=
yle=3D"font-size:13.3333px;font-family:arial,sans-serif"> </span><br></pre>=
<pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;ma=
rgin-bottom:0px">      &quot;If the packet is of type BYE, for each &quot;S=
SRC/CSRC&quot; indicated in the
      packet that is found in the incoming SSRC table, deliver a copy of
      the packet to the RtpReceiver associated with that SSRC.&quot;</pre><=
pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;mar=
gin-bottom:0px"><br></pre><pre class=3D"gmail-newpage" style=3D"font-size:1=
3.3333px;margin-top:0px;margin-bottom:0px">[BA] Change: </pre><pre class=3D=
"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0=
px"><br></pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;mar=
gin-top:0px;margin-bottom:0px"><pre class=3D"gmail-newpage" style=3D"font-s=
ize:13.3333px;margin-top:0px;margin-bottom:0px">      &quot;If the packet i=
s of type RTPFB or PSFB, as defined in [<a href=3D"https://tools.ietf.org/h=
tml/rfc4585" title=3D"&quot;Extended RTP Profile for Real-time Transport Co=
ntrol Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>],
      and the media source SSRC for the packet is found in the outgoing
      SSRC table, deliver the packet to the RtpSender associated with
      that SSRC.&quot;</pre><pre class=3D"gmail-newpage" style=3D"font-size=
:13.3333px;margin-top:0px;margin-bottom:0px"><br></pre><pre class=3D"gmail-=
newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">To:=
</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:=
0px;margin-bottom:0px"><br></pre><pre class=3D"gmail-newpage" style=3D"font=
-size:13.3333px;margin-top:0px;margin-bottom:0px">      &quot;If the packet=
 is of type RTPFB or PSFB, as defined in [<a href=3D"https://tools.ietf.org=
/html/rfc4585" title=3D"&quot;Extended RTP Profile for Real-time Transport =
Control Protocol (RTCP)-Based Feedback (RTP/AVPF)&quot;">RFC4585</a>],
      and the &quot;SSRC of media source&quot; for the packet is found in t=
he outgoing
      SSRC table, deliver the packet to the RtpSender associated with
      that SSRC.&quot;</pre><pre class=3D"gmail-newpage" style=3D"font-size=
:13.3333px;margin-top:0px;margin-bottom:0px"><br></pre><pre class=3D"gmail-=
newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">[BA=
] Since FIR feedback messages can contain multiple FCI fields, I think you<=
/pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0=
px;margin-bottom:0px">need to add the following text <span style=3D"font-si=
ze:13.3333px;font-family:arial,sans-serif">(see RFC 5104 Section 4.3.1.1): =
</span></pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;marg=
in-top:0px;margin-bottom:0px"><br></pre><pre class=3D"gmail-newpage" style=
=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">      &quot;In ad=
dition, for FIR feedback messages, as defined in [RFC5104],</pre><pre class=
=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-botto=
m:0px">      for each &quot;SSRC&quot; indicated in an FCI entry, deliver a=
 copy of the</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px=
;margin-top:0px;margin-bottom:0px">      packet to the RtpSender associated=
 with that SSRC.</pre></pre><pre class=3D"gmail-newpage" style=3D"font-size=
:13.3333px;margin-top:0px;margin-bottom:0px">  </pre></pre></pre></pre></di=
v></div>

--001a114485f436b9fd053fce2f71--


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Subject: [rtcweb] CyberSec2017 Call for Papers and Special Session
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You are invited to contribute, participate and/or submit your research 
paper.

The Fifth International Conference on Cyber Security, Cyber Welfare and 
Digital Forensic (CyberSec2017)

St. Mary's University, Addis Ababa, Ethiopia
April 22-24, 2017
http://sdiwc.net/conferences/6th-international-cyber-security-cyber-welfare-digital-forensic/

Paper Submission Deadline: March 1, 2017


==========================
  Call for Special Session
==========================

The conference also welcomes researchers to organize special session in 
any one of the tracks in cyber security and digital forensic with 
regards to the conference topic.

Mechanics:

Special Sessions may address one or more tracks, but they should be 
organized under a unified theme.  Each special session should attract 6 
registered papers. These papers will be part of the Conference 
Proceedings, and they will be published in the proceedings of the 
conference and will be sent to at least 10 indexing places.

Submission:

Send your session title and summary as detailed in the conference 
website.


From nobody Thu Oct 27 06:43:43 2016
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Cc: draft-ietf-rtcweb-transports@ietf.org, rtcweb-chairs@ietf.org, rtcweb@ietf.org, The IESG <iesg@ietf.org>, rfc-editor@rfc-editor.org
Subject: [rtcweb] Protocol Action: 'Transports for WebRTC' to Proposed Standard (draft-ietf-rtcweb-transports-17.txt)
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The IESG has approved the following document:
- 'Transports for WebRTC'
  (draft-ietf-rtcweb-transports-17.txt) as Proposed Standard

This document is the product of the Real-Time Communication in
WEB-browsers Working Group.

The IESG contact persons are Alexey Melnikov, Ben Campbell and Alissa
Cooper.

A URL of this Internet Draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/





Technical Summary

  This document describes the data transport protocols used by WebRTC,
  including the protocols used for interaction with intermediate boxes
  such as firewalls, relays and NAT boxes.

Working Group Summary

  Nothing particularly controversial in this spec. It is largely just a 
  pointer to other specifications saying if you do WebRTC, then you need 
  to what RFC X,Y, and Z says. 

  Based on the WGLC comments, the chairs suggest adding an RFC Ed note 
  to include an informative reference to draft-ietf-rtcweb-return 
  however there is not WG consensus to make this a MUST or SHOULD 
  implement. (Cleary nothing forbids implementing it so it is already an 
  MAY implement)

Document Quality

  Much of this is implemented in chrome and firefox which share a 
  significant amount of code with respect to this so some parts of it 
  has 2 implementations, some parts 1, and some none. 

Personnel

  Cullen Jennings is the document shepherd. Alissa Cooper is the 
  responsible AD.


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Thu, 27 Oct 2016 09:48:33 -0700
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Subject: [rtcweb] JSEP Section 3.7:Simulcast
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--94eb2c07bb88520f9a053fdb84f8
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   JSEP currently does not provide an API to configure receipt of
   simulcast.  This means that if simulcast is offered by the remote
   endpoint, the answer generated by a JSEP endpoint will not indicate
   support for receipt of simulcast, and as such the remote endpoint
   will only send a single encoding per m= section.

[BA] In the description of addTransceiver, WebRTC 1.0 Section 5.1 states:

However whensetRemoteDescription is called with a corresponding remote
description that is able to send multiple RTP encodings as defined in
[JSEP <https://www.w3.org/TR/webrtc/#bib-JSEP>], the RTCRtpReceiver
<https://www.w3.org/TR/webrtc/#idl-def-rtcrtpreceiver> may receive
multiple RTP encodings and the parameters retrieved via the
transceiver's receiver.getParameters() will reflect the encodings
negotiated.

This appears to indicate that a JSEP endpoint may indicate support for
receipt of simulcast in an answer.

   In addition, when
   the JSEP endpoint is the answerer, the permitted encodings for the
   RTPSender must be consistent with the offer, but this information is
   currently not surfaced through any API.

[BA] Not sure what this is trying to say. Once setRemoteDescription is

called on the JSEP endpoint, won't it be possible to retrieve the sender

settings via sender.getParameters() and the receiver settings via

receiver.getParameters()?

   This means that established
   simulcast streams will continue to work through a received re-offer,
   but setting up initial simulcast by way of a received offer requires
   out-of-band signaling or SDP inspection.  Future versions of this
   specification may add additional APIs to provide this control.

[BA] Does this mean that a JSEP endpoint cannot respond to an offer to receive

simulcast with an Answer indicating that it will send simulcast? Note sure why

this would be the case, particularly if addTransceiver was called prior to

setRemoteDescription indicating the ability to send simulcast.

--94eb2c07bb88520f9a053fdb84f8
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<div dir=3D"ltr"><pre style=3D"color:rgb(0,0,0)">   JSEP currently does not=
 provide an API to configure receipt of
   simulcast.  This means that if simulcast is offered by the remote
   endpoint, the answer generated by a JSEP endpoint will not indicate
   support for receipt of simulcast, and as such the remote endpoint
   will only send a single encoding per m=3D section.  </pre><pre style=3D"=
color:rgb(0,0,0)">[BA] In the description of addTransceiver, WebRTC 1.0 Sec=
tion 5.1 states: </pre><pre style=3D"color:rgb(0,0,0)">However when<code st=
yle=3D"color:orange;font-size:0.9em;font-family:menlo,consolas,&quot;dejavu=
 sans mono&quot;,monaco,monospace;white-space:normal">setRemoteDescription<=
/code><span style=3D"font-family:sans-serif;font-size:medium;white-space:no=
rmal">=C2=A0is called with a corresponding remote description that is able =
to send multiple RTP encodings as defined in [</span><cite style=3D"font-fa=
mily:sans-serif;font-size:medium;white-space:normal"><a class=3D"gmail-bibr=
ef" href=3D"https://www.w3.org/TR/webrtc/#bib-JSEP" style=3D"text-decoratio=
n:none;font-style:normal;color:rgb(3,69,117);border-bottom:1px solid rgb(18=
7,187,187);padding:0px 1px">JSEP</a></cite><span style=3D"font-family:sans-=
serif;font-size:medium;white-space:normal">], the=C2=A0</span><code style=
=3D"color:orange;font-size:0.9em;font-family:menlo,consolas,&quot;dejavu sa=
ns mono&quot;,monaco,monospace;white-space:normal"><a href=3D"https://www.w=
3.org/TR/webrtc/#idl-def-rtcrtpreceiver" class=3D"gmail-internalDFN" style=
=3D"color:rgb(3,69,117);border-bottom:1px solid rgb(187,187,187);text-decor=
ation:none;padding:0px 1px"><code style=3D"color:orange;font-size:14.4px;fo=
nt-family:menlo,consolas,&quot;dejavu sans mono&quot;,monaco,monospace">RTC=
RtpReceiver</code></a>=C2=A0</code><span style=3D"font-family:sans-serif;fo=
nt-size:medium;white-space:normal">may receive multiple RTP encodings and t=
he parameters retrieved via the transceiver&#39;s=C2=A0</span><code style=
=3D"color:orange;font-size:0.9em;font-family:menlo,consolas,&quot;dejavu sa=
ns mono&quot;,monaco,monospace;white-space:normal">receiver.getParameters()=
</code><span style=3D"font-family:sans-serif;font-size:medium;white-space:n=
ormal">=C2=A0will reflect the encodings negotiated.</span><br></pre><pre st=
yle=3D"color:rgb(0,0,0)"><span style=3D"font-family:sans-serif;font-size:me=
dium;white-space:normal">This appears to indicate that a JSEP endpoint may =
indicate support for receipt of simulcast in an answer.=C2=A0</span></pre><=
pre style=3D"color:rgb(0,0,0)">   In addition, when
   the JSEP endpoint is the answerer, the permitted encodings for the
   RTPSender must be consistent with the offer, but this information is
   currently not surfaced through any API.  </pre><pre style=3D"color:rgb(0=
,0,0)">[BA] Not sure what this is trying to say. Once setRemoteDescription =
is</pre><pre style=3D"color:rgb(0,0,0)">called on the JSEP endpoint, won&#3=
9;t it be possible to retrieve the sender </pre><pre style=3D"color:rgb(0,0=
,0)">settings via sender.getParameters() and the receiver settings via </pr=
e><pre style=3D"color:rgb(0,0,0)">receiver.getParameters()? </pre><pre styl=
e=3D"color:rgb(0,0,0)">   This means that established
   simulcast streams will continue to work through a received re-offer,
   but setting up initial simulcast by way of a received offer requires
   out-of-band signaling or SDP inspection.  Future versions of this
   specification may add additional APIs to provide this control.</pre><pre=
 style=3D"color:rgb(0,0,0)">[BA] Does this mean that a JSEP endpoint cannot=
 respond to an offer to receive</pre><pre style=3D"color:rgb(0,0,0)">simulc=
ast with an Answer indicating that it will send simulcast? Note sure why</p=
re><pre style=3D"color:rgb(0,0,0)">this would be the case, particularly if =
addTransceiver was called prior to </pre><pre style=3D"color:rgb(0,0,0)">se=
tRemoteDescription indicating the ability to send simulcast.</pre></div>

--94eb2c07bb88520f9a053fdb84f8--


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Subject: [rtcweb] JSEP Section 4.2.2
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*4.2.2.  stopped

   The stopped method returns "true" if the transceiver has been
   stopped, either by a call to stopTransceiver or by applying an answer*
   that govern *rejects* the creation and processing of offers
*associated m= section,* and answers.
*"false" otherwise.*


*[BA] This should be rewritten as follows: *

*    The stopped attribute has the value "true if the RtpTransceiver
has been stopped,*

*    either by a call to transceiver.stop() or by applying an answer that*

*    rejects the associated m= section and "false" otherwise.*

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<div dir=3D"ltr"><pre style=3D"color:rgb(0,0,0)"><strong><font color=3D"gre=
en">4.2.2.  stopped

   The stopped method returns &quot;true&quot; if the transceiver has been
   stopped, either by a call to stopTransceiver or by applying an answer</f=
ont></strong>
   that <strike><font color=3D"red">govern</font></strike> <strong><font co=
lor=3D"green">rejects</font></strong> the <strike><font color=3D"red">creat=
ion and processing of offers</font></strike> <strong><font color=3D"green">=
associated m=3D section,</font></strong> and <strike><font color=3D"red">an=
swers.</font></strike> <strong><font color=3D"green">&quot;false&quot; othe=
rwise.<br></font></strong></pre><pre style=3D"color:rgb(0,0,0)"><strong><fo=
nt color=3D"green"><br></font></strong></pre><pre style=3D"color:rgb(0,0,0)=
"><strong><font color=3D"green">[BA] This should be rewritten as follows: <=
/font></strong></pre><pre style=3D"color:rgb(0,0,0)"><strong><font color=3D=
"green">    The stopped attribute has the value &quot;true if the RtpTransc=
eiver has been stopped,</font></strong></pre><pre style=3D"color:rgb(0,0,0)=
"><strong><font color=3D"green">    either by a call to transceiver.stop() =
or by applying an answer that</font></strong></pre><pre style=3D"color:rgb(=
0,0,0)"><strong><font color=3D"green">    rejects the associated m=3D secti=
on and &quot;false&quot; otherwise.</font></strong></pre><pre style=3D"colo=
r:rgb(0,0,0)"><br></pre></div>

--001a114485f41e11f0053fdba16a--


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From: Sean Turner <sean@sn3rd.com>
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Subject: Re: [rtcweb] Protocol Action: 'Transports for WebRTC' to Proposed Standard (draft-ietf-rtcweb-transports-17.txt)
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Congrats! Another draft has progressed down the road:
https://giphy.com/gifs/car-reactiongifs-mrw-T4YX44tlqOKaI/fullscreen

spt

> On Oct 27, 2016, at 09:43, The IESG <iesg-secretary@ietf.org> wrote:
> 
> The IESG has approved the following document:
> - 'Transports for WebRTC'
>  (draft-ietf-rtcweb-transports-17.txt) as Proposed Standard
> 
> This document is the product of the Real-Time Communication in
> WEB-browsers Working Group.
> 
> The IESG contact persons are Alexey Melnikov, Ben Campbell and Alissa
> Cooper.
> 
> A URL of this Internet Draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/
> 
> 
> 
> 
> 
> Technical Summary
> 
>  This document describes the data transport protocols used by WebRTC,
>  including the protocols used for interaction with intermediate boxes
>  such as firewalls, relays and NAT boxes.
> 
> Working Group Summary
> 
>  Nothing particularly controversial in this spec. It is largely just a 
>  pointer to other specifications saying if you do WebRTC, then you need 
>  to what RFC X,Y, and Z says. 
> 
>  Based on the WGLC comments, the chairs suggest adding an RFC Ed note 
>  to include an informative reference to draft-ietf-rtcweb-return 
>  however there is not WG consensus to make this a MUST or SHOULD 
>  implement. (Cleary nothing forbids implementing it so it is already an 
>  MAY implement)
> 
> Document Quality
> 
>  Much of this is implemented in chrome and firefox which share a 
>  significant amount of code with respect to this so some parts of it 
>  has 2 implementations, some parts 1, and some none. 
> 
> Personnel
> 
>  Cullen Jennings is the document shepherd. Alissa Cooper is the 
>  responsible AD.


From nobody Fri Oct 28 23:48:57 2016
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To: Sean Turner <sean@sn3rd.com>, draft-ietf-rtcweb-transports@ietf.org
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On 10/28/2016 02:59 AM, Sean Turner wrote:
> Congrats! Another draft has progressed down the road:
> https://giphy.com/gifs/car-reactiongifs-mrw-T4YX44tlqOKaI/fullscreen

Hooray!

It is now part of RFC Editor's cluster 238, a giant snarl of dependencies:

https://www.rfc-editor.org/cluster_info.php?cid=C238

An outstanding item here is the -security drafts; do we have any updated
status on those?

>
> spt
>
>> On Oct 27, 2016, at 09:43, The IESG <iesg-secretary@ietf.org> wrote:
>>
>> The IESG has approved the following document:
>> - 'Transports for WebRTC'
>>  (draft-ietf-rtcweb-transports-17.txt) as Proposed Standard
>>
>> This document is the product of the Real-Time Communication in
>> WEB-browsers Working Group.
>>
>> The IESG contact persons are Alexey Melnikov, Ben Campbell and Alissa
>> Cooper.
>>
>> A URL of this Internet Draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/
>>
>>
>>
>>
>>
>> Technical Summary
>>
>>  This document describes the data transport protocols used by WebRTC,
>>  including the protocols used for interaction with intermediate boxes
>>  such as firewalls, relays and NAT boxes.
>>
>> Working Group Summary
>>
>>  Nothing particularly controversial in this spec. It is largely just a 
>>  pointer to other specifications saying if you do WebRTC, then you need 
>>  to what RFC X,Y, and Z says. 
>>
>>  Based on the WGLC comments, the chairs suggest adding an RFC Ed note 
>>  to include an informative reference to draft-ietf-rtcweb-return 
>>  however there is not WG consensus to make this a MUST or SHOULD 
>>  implement. (Cleary nothing forbids implementing it so it is already an 
>>  MAY implement)
>>
>> Document Quality
>>
>>  Much of this is implemented in chrome and firefox which share a 
>>  significant amount of code with respect to this so some parts of it 
>>  has 2 implementations, some parts 1, and some none. 
>>
>> Personnel
>>
>>  Cullen Jennings is the document shepherd. Alissa Cooper is the 
>>  responsible AD.


-- 
Surveillance is pervasive. Go Dark.


From nobody Mon Oct 31 16:49:54 2016
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Real-Time Communication in WEB-browsers of the IETF.

        Title           : WebRTC Forward Error Correction Requirements
        Author          : Justin Uberti
	Filename        : draft-ietf-rtcweb-fec-04.txt
	Pages           : 10
	Date            : 2016-10-31

Abstract:
   This document provides information and requirements for how Forward
   Error Correction (FEC) should be used by WebRTC applications.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-fec-04

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-fec-04


Please note that it may take a couple of minutes from the time of submission
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From nobody Mon Oct 31 16:58:02 2016
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Real-Time Communication in WEB-browsers of the IETF.

        Title           : WebRTC IP Address Handling Requirements
        Authors         : Justin Uberti
                          Guo-wei Shieh
	Filename        : draft-ietf-rtcweb-ip-handling-02.txt
	Pages           : 8
	Date            : 2016-10-31

Abstract:
   This document provides best practices for how IP addresses should be
   handled by WebRTC applications.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-ip-handling/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-02

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