
From nobody Mon May  2 13:58:59 2016
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To: rtcweb@ietf.org
From: Randell Jesup <randell-ietf@jesup.org>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-ip-handling-01.txt
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On 4/26/2016 4:11 AM, Philipp Hancke wrote:
> I stumbled upon a (now deleted) stackoverflow posting yesterday in 
> which obfuscated code "used webrtc" to hack the users router.
> Basically webrtc was used to get the local ip address from which the 
> routers ip is inferred. Then a default admin account is tried.
>
> It turns out the idea is not knew, I heard from some former coworkers 
> that they used this in penetration testing before. But it has now 
> reached stackoverflow which means there will soon be rumors and all 
> kinds of "webrtc allows hacking your router" stories.

Definitely not new, nor (as you state) requiring webrtc in any manner.

> The principal problem here is default passwords. WebRTC makes it 
> slightly easier by removing the need to run a series of http requests 
> to find out the routers ip. Should this be mentioned as an example in 
> #1 of the problem statement?

My only concern is not wanting to publish a blueprint for attacks (even 
if known to some).  Perhaps irrelevant.

-- 
Randell Jesup -- rjesup a t mozilla d o t com
Please please please don't email randell-ietf@jesup.org!  Way too much spam


From nobody Tue May  3 19:40:46 2016
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Subject: [rtcweb] Ben Campbell's Yes on draft-ietf-rtcweb-alpn-03: (with COMMENT)
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Ben Campbell has entered the following ballot position for
draft-ietf-rtcweb-alpn-03: Yes

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introductory paragraph, however.)


Please refer to https://www.ietf.org/iesg/statement/discuss-criteria.html
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The document, along with other ballot positions, can be found here:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-alpn/



----------------------------------------------------------------------
COMMENT:
----------------------------------------------------------------------

Should I-D.ietf-rtcweb-security-arch be a normative reference, due to the
citation in section 4?



From nobody Tue May  3 20:23:42 2016
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From: Martin Thomson <martin.thomson@gmail.com>
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On 4 May 2016 at 12:40, Ben Campbell <ben@nostrum.com> wrote:
> Should I-D.ietf-rtcweb-security-arch be a normative reference, due to the
> citation in section 4?

That was intended largely as a "such as", would be clearer if it were
changed to: Peer authentication, such as that provided by Section X of
[I-D.ietf...arch], ...


From nobody Tue May  3 20:28:52 2016
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From: "Ben Campbell" <ben@nostrum.com>
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On 3 May 2016, at 22:23, Martin Thomson wrote:

> On 4 May 2016 at 12:40, Ben Campbell <ben@nostrum.com> wrote:
>> Should I-D.ietf-rtcweb-security-arch be a normative reference, due to 
>> the
>> citation in section 4?
>
> That was intended largely as a "such as", would be clearer if it were
> changed to: Peer authentication, such as that provided by Section X of
> [I-D.ietf...arch], ...

Possibly--but do you expect this to be a situation where a webrtc peer 
can pick from any number of available peer authentication mechanisms in 
addition to what is in security-arch, or is this pretty much it?


From nobody Tue May  3 20:42:48 2016
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From: Martin Thomson <martin.thomson@gmail.com>
To: Ben Campbell <ben@nostrum.com>
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Subject: Re: [rtcweb] Ben Campbell's Yes on draft-ietf-rtcweb-alpn-03: (with COMMENT)
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On 4 May 2016 at 13:28, Ben Campbell <ben@nostrum.com> wrote:
> On 3 May 2016, at 22:23, Martin Thomson wrote:
>
>> On 4 May 2016 at 12:40, Ben Campbell <ben@nostrum.com> wrote:
>>>
>>> Should I-D.ietf-rtcweb-security-arch be a normative reference, due to the
>>> citation in section 4?
>>
>>
>> That was intended largely as a "such as", would be clearer if it were
>> changed to: Peer authentication, such as that provided by Section X of
>> [I-D.ietf...arch], ...
>
>
> Possibly--but do you expect this to be a situation where a webrtc peer can
> pick from any number of available peer authentication mechanisms in addition
> to what is in security-arch, or is this pretty much it?

That's the right question.  It's the latter, so let's just elevate
this to normative.  It might be that this blocks publication, but
think that the world can do without another RFC for a little longer.


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Stephen Farrell has entered the following ballot position for
draft-ietf-rtcweb-alpn-03: No Objection

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----------------------------------------------------------------------
COMMENT:
----------------------------------------------------------------------



- I suspect the term "confidential" as used here will turn out
to mislead or confuse some folks. The meaning is clear if one
reads the draft, but of course many people will just read some
stackexchange answer. It's probably too late to try change
that unless someone has a good term beginning with "c" to use
for c-werbrtc. The potential for confusion I think will be
that the other label might be assumed to not use a good
confidentiality mechanism on the wire, so folks might get
concerned that e.g. their DataChannel stuff can be read by a
middlebox.  (I just mention this in case the concern is
either new or has been bubbling up in the WG, feel entirely
free to ignore me if you want.)

- I forget how the screen sharing issue for WebRTC was
resolved. In any case, do the handling of screen sharing and
c-webrtc interact? Do you need to explain that there's some
non-browser "access" (origination really) of media on the
screen-sharer's machine?

- "clever arrangement of mirrors" - that is a nice way to
explain the futility of DRM :-)



From nobody Wed May  4 07:36:52 2016
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On 4 May 2016 at 17:32, Stephen Farrell <stephen.farrell@cs.tcd.ie> wrote:
> - I suspect the term "confidential" as used here will turn out
> to mislead or confuse some folks. The meaning is clear if one
> reads the draft, but of course many people will just read some
> stackexchange answer. It's probably too late to try change
> that unless someone has a good term beginning with "c" to use
> for c-werbrtc. The potential for confusion I think will be
> that the other label might be assumed to not use a good
> confidentiality mechanism on the wire, so folks might get
> concerned that e.g. their DataChannel stuff can be read by a
> middlebox.  (I just mention this in case the concern is
> either new or has been bubbling up in the WG, feel entirely
> free to ignore me if you want.)

Alissa raised the same point in her review.  I believe that the
conclusion was what we have.  I agree that as a very selective
confidentiality, it's potentially misleading.  In practice, this is a
feature that will require really good understanding of how this all
fits together, so I'm not concerned about readership misunderstanding.

> - I forget how the screen sharing issue for WebRTC was
> resolved. In any case, do the handling of screen sharing and
> c-webrtc interact? Do you need to explain that there's some
> non-browser "access" (origination really) of media on the
> screen-sharer's machine?

Screen sharing looks the same as other media in this regard.  Of
course, screen sharing is still basically inaccessible in browsers due
to the huge problems it engenders.


From nobody Wed May  4 08:29:27 2016
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From: "Black, David" <david.black@emc.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Greetings,

I write as the shepherd for the WebRTC QoS draft (draft-ietf-tsvwg-rtcweb-q=
os).
IETF Last Call on this draft uncovered an issue that appears to also requir=
e
changes in the Web RTC transports draft (draft-ietf-rtcweb-transports).

The Web RTC draft  specifies different Diffserv Codepoints (DSCPs) for use =
with
Interactive vs. Non-Interactive Video (with or without audio in both cases)=
:
https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-5

The IETF LC issue is: How does an implementer determine whether video is
interactive vs. non-interactive?

The answer (unexpected to at least me) is that this can't be done for curre=
nt
WebRTC - all video has to be treated as interactive because there's no API
support to distinguish interactive video from non-interactive video.  That
doesn't appear to be stated anywhere obvious, and a TSVWG draft seems
like the wrong place to do so.=20

Hence, this message contains initial proposed changes to both the WebRTC
Transports draft and WebRTC QoS draft to resolve this issue without removin=
g
any of the DSCPs currently specified for video in the WebRTC QoS draft.

We should resolve this within the next few days, as the WebRTC QoS draft is
(finally!) on the IESG telechat agenda for May 19.

--- WebRTC Transports draft - proposed changes

In Section 4 Media Prioritization, after the following existing text:

   For media, a "media flow", which can be an "audio flow" or a "video
   flow", is what [RFC7656] calls a "media source", which results in a
   "source RTP stream" and one or more "redundancy RTP streams".  This
   specification does not describe prioritization between the RTP
   streams that come from a single "media source".

Add a new paragraph:

   All media flows in WebRTC are assumed to be interactive; there is
   no browser API support for non-interactive media, aside from sending
   non-interactive media content via the data channel.

This ought to cite a W3C reference for the WebRTC API - the following
appears plausible:

   [W3C.WD-webrtc-20120209]
              Bergkvist, A., Burnett, D., Jennings, C., and A.
              Narayanan, "WebRTC 1.0: Real-time Communication Between
              Browsers", World Wide Web Consortium WD WD-webrtc-
              20120209, February 2012,
              <http://www.w3.org/TR/2012/WD-webrtc-20120209>.
  =20
-- WebRTC QoS draft - proposed changes

In Section 5 DSCP Mappings, after the following existing text:

   The browser SHOULD first select the flow type of the flow.  Within
   the flow type, the relative importance of the flow SHOULD be used to
   select the appropriate DSCP value.

Add a new paragraph:

   The current WebRTC browser API does not support non-interactive
   video; all video is assumed to be interactive [I-D.ietf-rtcweb-transport=
s].
   Browsers MUST NOT use the DSCP mappings for Non-Interactive Video
   in Table 1.  Non-browser implementations of WebRTC MAY use the
   DSCP mappings for Non-Interactive Video in Table 1 for video that is
   known to not be interactive, e.g., all video in a video playback applica=
tion
   based on a custom implementation of the WebRTC protocols would not
   be interactive.

------

Comments are welcome - as noted, above,  we should resolve this in
the next few days in order to avoid pulling the WebRTC QoS draft off
of the May 19 IESG telechat agenda. Please reply to both the TSVWG and
RTCWEB lists, as we need  a coherent solution across both drafts.

I want to thank Colin Perkins for suggesting the video playback example.

Thanks, --David (as WebRTC QoS draft shepherd)


From nobody Wed May  4 12:00:51 2016
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Kathleen Moriarty has entered the following ballot position for
draft-ietf-rtcweb-alpn-03: No Objection

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----------------------------------------------------------------------
COMMENT:
----------------------------------------------------------------------

I agree with Stephen's comments on the word confidentiality, but can't
think of an alternate word.  I think text describing how this is limited
would be helpful in the introduction.  The clearest (at least to me)
description of what is meant by confidentiality doesn't appear until the
security considerations section.



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To: "Black, David" <david.black@emc.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
References: <CE03DB3D7B45C245BCA0D243277949362E993518@MX104CL02.corp.emc.com>
From: Brian E Carpenter <brian.e.carpenter@gmail.com>
Organization: University of Auckland
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Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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I fear I have an up-level question: what is the definition of "interactive"?

draft-ietf-tsvwg-rtcweb-qos doesn't define it, presumably because it was
assumed to be obvious, but thinking about David's suggestions, I decided
that it isn't so obvious.

https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-4 says:

   The following are the inputs provided by the WebRTC application to
   the browser:

   o  Flow Type: The browser provides this input as it knows if the flow
      is audio, interactive video with or without audio, non-interactive
      video with or without audio, or data.

Firstly, I'm puzzled by the apparent contradiction between the first
sentence (in which the app provides input to the browser) and the second
sentence (in which to the browser provides input to something else).

Then, is it clear how the browser "knows" this? If David's statement below
about the API is correct, the browser actually cannot "provide this input"
as stated in the above quotation. Or I am confused.

Finally, why is audio not also subdivided into interactive and
non-interactive? As far as I can see, both are logically possible.

Regards
   Brian

On 05/05/2016 03:26, Black, David wrote:
> Greetings,
> 
> I write as the shepherd for the WebRTC QoS draft (draft-ietf-tsvwg-rtcweb-qos).
> IETF Last Call on this draft uncovered an issue that appears to also require
> changes in the Web RTC transports draft (draft-ietf-rtcweb-transports).
> 
> The Web RTC draft  specifies different Diffserv Codepoints (DSCPs) for use with
> Interactive vs. Non-Interactive Video (with or without audio in both cases):
> https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-5
> 
> The IETF LC issue is: How does an implementer determine whether video is
> interactive vs. non-interactive?
> 
> The answer (unexpected to at least me) is that this can't be done for current
> WebRTC - all video has to be treated as interactive because there's no API
> support to distinguish interactive video from non-interactive video.  That
> doesn't appear to be stated anywhere obvious, and a TSVWG draft seems
> like the wrong place to do so. 
> 
> Hence, this message contains initial proposed changes to both the WebRTC
> Transports draft and WebRTC QoS draft to resolve this issue without removing
> any of the DSCPs currently specified for video in the WebRTC QoS draft.
> 
> We should resolve this within the next few days, as the WebRTC QoS draft is
> (finally!) on the IESG telechat agenda for May 19.
> 
> --- WebRTC Transports draft - proposed changes
> 
> In Section 4 Media Prioritization, after the following existing text:
> 
>    For media, a "media flow", which can be an "audio flow" or a "video
>    flow", is what [RFC7656] calls a "media source", which results in a
>    "source RTP stream" and one or more "redundancy RTP streams".  This
>    specification does not describe prioritization between the RTP
>    streams that come from a single "media source".
> 
> Add a new paragraph:
> 
>    All media flows in WebRTC are assumed to be interactive; there is
>    no browser API support for non-interactive media, aside from sending
>    non-interactive media content via the data channel.
> 
> This ought to cite a W3C reference for the WebRTC API - the following
> appears plausible:
> 
>    [W3C.WD-webrtc-20120209]
>               Bergkvist, A., Burnett, D., Jennings, C., and A.
>               Narayanan, "WebRTC 1.0: Real-time Communication Between
>               Browsers", World Wide Web Consortium WD WD-webrtc-
>               20120209, February 2012,
>               <http://www.w3.org/TR/2012/WD-webrtc-20120209>.
>    
> -- WebRTC QoS draft - proposed changes
> 
> In Section 5 DSCP Mappings, after the following existing text:
> 
>    The browser SHOULD first select the flow type of the flow.  Within
>    the flow type, the relative importance of the flow SHOULD be used to
>    select the appropriate DSCP value.
> 
> Add a new paragraph:
> 
>    The current WebRTC browser API does not support non-interactive
>    video; all video is assumed to be interactive [I-D.ietf-rtcweb-transports].
>    Browsers MUST NOT use the DSCP mappings for Non-Interactive Video
>    in Table 1.  Non-browser implementations of WebRTC MAY use the
>    DSCP mappings for Non-Interactive Video in Table 1 for video that is
>    known to not be interactive, e.g., all video in a video playback application
>    based on a custom implementation of the WebRTC protocols would not
>    be interactive.
> 
> ------
> 
> Comments are welcome - as noted, above,  we should resolve this in
> the next few days in order to avoid pulling the WebRTC QoS draft off
> of the May 19 IESG telechat agenda. Please reply to both the TSVWG and
> RTCWEB lists, as we need  a coherent solution across both drafts.
> 
> I want to thank Colin Perkins for suggesting the video playback example.
> 
> Thanks, --David (as WebRTC QoS draft shepherd)
> 
> 


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Wed, 4 May 2016 16:01:41 -0700
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Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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--94eb2c11bee8a0430c05320c36a4
Content-Type: text/plain; charset=UTF-8

Brian Carpenter said:

"Then, is it clear how the browser "knows" this?"

[BA] The application explicitly provides information to the browser by
setting the RTCRtpEncodingParameters.priority attribute via the
sender.setParameters() API:

dictionary RTCRtpEncodingParameters {             unsigned long
ssrc <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-ssrc>;
            RTCRtpRtxParameters rtx
<http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-rtx>;
          RTCRtpFecParameters fec
<http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-fec>;
          boolean             active
<http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-active>;
            RTCPriorityType     priority
<http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-priority>;
            unsigned long       maxBitrate
<http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-maxBitrate>;
            unsigned long       maxFramerate
<http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-maxFramerate>;
            DOMString           rid
<http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-rid>;
          double              scaleResolutionDownBy
<http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-scaleResolutionDownBy>
= 1.0;
};


The Priority and QoS Model is described in the latest WebRTC 1.0 Editor's
draft (http://w3c.github.io/webrtc-pc/) Section 4.10:

4.10 Priority and QoS Model

Many applications have multiple media flows of the same data type and often
some of the flows are more important than others. WebRTC uses the priority
and Quality of Service (QoS) framework described in [RTCWEB-TRANSPORT] and [
TSVWG-RTCWEB-QOS] to provide priority and DSCP marking for packets that
will help provide QoS in some networking environments. The priority setting
can be used to indicate the relative priority of various flows. The
priority API allows the JavaScript applications to tell the browser whether
a particular media flow is high, medium, low or of very low importance to
the application by setting the priorityproperty of
RTCRtpEncodingParameters objects
to one of the following values.
4.10.1 RTCPriorityType Enum

enum RTCPriorityType {
    "very-low <http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.very-low>",
    "low <http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.low>",
    "medium <http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.medium>",
    "high <http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.high>"
};


On Wed, May 4, 2016 at 1:28 PM, Brian E Carpenter <
brian.e.carpenter@gmail.com> wrote:

> I fear I have an up-level question: what is the definition of
> "interactive"?
>
> draft-ietf-tsvwg-rtcweb-qos doesn't define it, presumably because it was
> assumed to be obvious, but thinking about David's suggestions, I decided
> that it isn't so obvious.
>
> https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-4 says:
>
>    The following are the inputs provided by the WebRTC application to
>    the browser:
>
>    o  Flow Type: The browser provides this input as it knows if the flow
>       is audio, interactive video with or without audio, non-interactive
>       video with or without audio, or data.
>
> Firstly, I'm puzzled by the apparent contradiction between the first
> sentence (in which the app provides input to the browser) and the second
> sentence (in which to the browser provides input to something else).
>
> Then, is it clear how the browser "knows" this? If David's statement below
> about the API is correct, the browser actually cannot "provide this input"
> as stated in the above quotation. Or I am confused.
>
> Finally, why is audio not also subdivided into interactive and
> non-interactive? As far as I can see, both are logically possible.
>
> Regards
>    Brian
>
> On 05/05/2016 03:26, Black, David wrote:
> > Greetings,
> >
> > I write as the shepherd for the WebRTC QoS draft
> (draft-ietf-tsvwg-rtcweb-qos).
> > IETF Last Call on this draft uncovered an issue that appears to also
> require
> > changes in the Web RTC transports draft (draft-ietf-rtcweb-transports).
> >
> > The Web RTC draft  specifies different Diffserv Codepoints (DSCPs) for
> use with
> > Interactive vs. Non-Interactive Video (with or without audio in both
> cases):
> > https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-5
> >
> > The IETF LC issue is: How does an implementer determine whether video is
> > interactive vs. non-interactive?
> >
> > The answer (unexpected to at least me) is that this can't be done for
> current
> > WebRTC - all video has to be treated as interactive because there's no
> API
> > support to distinguish interactive video from non-interactive video.
> That
> > doesn't appear to be stated anywhere obvious, and a TSVWG draft seems
> > like the wrong place to do so.
> >
> > Hence, this message contains initial proposed changes to both the WebRTC
> > Transports draft and WebRTC QoS draft to resolve this issue without
> removing
> > any of the DSCPs currently specified for video in the WebRTC QoS draft.
> >
> > We should resolve this within the next few days, as the WebRTC QoS draft
> is
> > (finally!) on the IESG telechat agenda for May 19.
> >
> > --- WebRTC Transports draft - proposed changes
> >
> > In Section 4 Media Prioritization, after the following existing text:
> >
> >    For media, a "media flow", which can be an "audio flow" or a "video
> >    flow", is what [RFC7656] calls a "media source", which results in a
> >    "source RTP stream" and one or more "redundancy RTP streams".  This
> >    specification does not describe prioritization between the RTP
> >    streams that come from a single "media source".
> >
> > Add a new paragraph:
> >
> >    All media flows in WebRTC are assumed to be interactive; there is
> >    no browser API support for non-interactive media, aside from sending
> >    non-interactive media content via the data channel.
> >
> > This ought to cite a W3C reference for the WebRTC API - the following
> > appears plausible:
> >
> >    [W3C.WD-webrtc-20120209]
> >               Bergkvist, A., Burnett, D., Jennings, C., and A.
> >               Narayanan, "WebRTC 1.0: Real-time Communication Between
> >               Browsers", World Wide Web Consortium WD WD-webrtc-
> >               20120209, February 2012,
> >               <http://www.w3.org/TR/2012/WD-webrtc-20120209>.
> >
> > -- WebRTC QoS draft - proposed changes
> >
> > In Section 5 DSCP Mappings, after the following existing text:
> >
> >    The browser SHOULD first select the flow type of the flow.  Within
> >    the flow type, the relative importance of the flow SHOULD be used to
> >    select the appropriate DSCP value.
> >
> > Add a new paragraph:
> >
> >    The current WebRTC browser API does not support non-interactive
> >    video; all video is assumed to be interactive
> [I-D.ietf-rtcweb-transports].
> >    Browsers MUST NOT use the DSCP mappings for Non-Interactive Video
> >    in Table 1.  Non-browser implementations of WebRTC MAY use the
> >    DSCP mappings for Non-Interactive Video in Table 1 for video that is
> >    known to not be interactive, e.g., all video in a video playback
> application
> >    based on a custom implementation of the WebRTC protocols would not
> >    be interactive.
> >
> > ------
> >
> > Comments are welcome - as noted, above,  we should resolve this in
> > the next few days in order to avoid pulling the WebRTC QoS draft off
> > of the May 19 IESG telechat agenda. Please reply to both the TSVWG and
> > RTCWEB lists, as we need  a coherent solution across both drafts.
> >
> > I want to thank Colin Perkins for suggesting the video playback example.
> >
> > Thanks, --David (as WebRTC QoS draft shepherd)
> >
> >
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--94eb2c11bee8a0430c05320c36a4
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Brian Carpenter said:=C2=A0<div><br></div><div>&quot;<span=
 style=3D"font-size:12.8px">Then, is it clear how the browser &quot;knows&q=
uot; this?</span>&quot;</div><div><br></div><div>[BA] The application expli=
citly provides information to the browser by setting the RTCRtpEncodingPara=
meters.priority attribute via the sender.setParameters() API:=C2=A0</div><d=
iv><br></div><div><pre class=3D"" style=3D"margin-left:2em;white-space:pre-=
wrap;font-size:0.9em;font-family:Menlo,Consolas,&#39;DejaVu Sans Mono&#39;,=
Monaco,monospace;margin-top:1em;margin-bottom:1em;overflow:auto;border-top-=
width:1px;border-top-style:solid;border-top-color:rgb(144,184,222);border-b=
ottom-width:1px;border-bottom-style:solid;border-bottom-color:rgb(144,184,2=
22);padding:1em;line-height:17.28px;color:rgb(0,0,0)"><span class=3D"" id=
=3D"idl-def-RTCRtpEncodingParameters">dictionary <span class=3D"" style=3D"=
font-weight:bold;color:rgb(0,90,156)">RTCRtpEncodingParameters</span> {
<span class=3D"">             <span class=3D"" style=3D"color:rgb(0,90,156)=
">unsigned long</span>       <span class=3D"" style=3D"color:rgb(255,69,0)"=
><a href=3D"http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-s=
src" style=3D"color:rgb(3,69,117);border-bottom-width:1px;border-bottom-sty=
le:solid;border-bottom-color:rgb(187,187,187);text-decoration:none;padding:=
0px 1px;margin-top:0px;margin-bottom:0px">ssrc</a></span>;</span>
<span class=3D"">             <span class=3D"" style=3D"color:rgb(0,90,156)=
"><code style=3D"font-size:14.4px;font-family:Menlo,Consolas,&#39;DejaVu Sa=
ns Mono&#39;,Monaco,monospace;color:rgb(200,53,0)">RTCRtpRtxParameters</cod=
e></span> <span class=3D"" style=3D"color:rgb(255,69,0)"><a href=3D"http://=
w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-rtx" style=3D"color:=
rgb(3,69,117);border-bottom-width:1px;border-bottom-style:solid;border-bott=
om-color:rgb(187,187,187);text-decoration:none;padding:0px 1px;margin-top:0=
px;margin-bottom:0px">rtx</a></span>;</span>
<span class=3D"">             <span class=3D"" style=3D"color:rgb(0,90,156)=
"><code style=3D"font-size:14.4px;font-family:Menlo,Consolas,&#39;DejaVu Sa=
ns Mono&#39;,Monaco,monospace;color:rgb(200,53,0)">RTCRtpFecParameters</cod=
e></span> <span class=3D"" style=3D"color:rgb(255,69,0)"><a href=3D"http://=
w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-fec" style=3D"color:=
rgb(3,69,117);border-bottom-width:1px;border-bottom-style:solid;border-bott=
om-color:rgb(187,187,187);text-decoration:none;padding:0px 1px;margin-top:0=
px;margin-bottom:0px">fec</a></span>;</span>
<span class=3D"">             <span class=3D"" style=3D"color:rgb(0,90,156)=
">boolean</span>             <span class=3D"" style=3D"color:rgb(255,69,0)"=
><a href=3D"http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-a=
ctive" style=3D"color:rgb(3,69,117);border-bottom-width:1px;border-bottom-s=
tyle:solid;border-bottom-color:rgb(187,187,187);text-decoration:none;paddin=
g:0px 1px;margin-top:0px;margin-bottom:0px">active</a></span>;</span>
<span class=3D"">             <span class=3D"" style=3D"color:rgb(0,90,156)=
"><code style=3D"font-size:14.4px;font-family:Menlo,Consolas,&#39;DejaVu Sa=
ns Mono&#39;,Monaco,monospace;color:rgb(200,53,0)">RTCPriorityType</code></=
span>     <span class=3D"" style=3D"color:rgb(255,69,0)"><a href=3D"http://=
w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-priority" style=3D"c=
olor:rgb(3,69,117);border-bottom-width:1px;border-bottom-style:solid;border=
-bottom-color:rgb(187,187,187);text-decoration:none;padding:0px 1px;margin-=
top:0px;margin-bottom:0px">priority</a></span>;</span>
<span class=3D"">             <span class=3D"" style=3D"color:rgb(0,90,156)=
">unsigned long</span>       <span class=3D"" style=3D"color:rgb(255,69,0)"=
><a href=3D"http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-m=
axBitrate" style=3D"color:rgb(3,69,117);border-bottom-width:1px;border-bott=
om-style:solid;border-bottom-color:rgb(187,187,187);text-decoration:none;pa=
dding:0px 1px;margin-top:0px;margin-bottom:0px">maxBitrate</a></span>;</spa=
n>
<span class=3D"">             <span class=3D"" style=3D"color:rgb(0,90,156)=
">unsigned long</span>       <span class=3D"" style=3D"color:rgb(255,69,0)"=
><a href=3D"http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-m=
axFramerate" style=3D"color:rgb(3,69,117);border-bottom-width:1px;border-bo=
ttom-style:solid;border-bottom-color:rgb(187,187,187);text-decoration:none;=
padding:0px 1px;margin-top:0px;margin-bottom:0px">maxFramerate</a></span>;<=
/span>
<span class=3D"">             <span class=3D"" style=3D"color:rgb(0,90,156)=
">DOMString</span>           <span class=3D"" style=3D"color:rgb(255,69,0)"=
><a href=3D"http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-r=
id" style=3D"color:rgb(3,69,117);border-bottom-width:1px;border-bottom-styl=
e:solid;border-bottom-color:rgb(187,187,187);text-decoration:none;padding:0=
px 1px;margin-top:0px;margin-bottom:0px">rid</a></span>;</span>
<span class=3D"">             <span class=3D"" style=3D"color:rgb(0,90,156)=
">double</span>              <span class=3D"" style=3D"color:rgb(255,69,0)"=
><a href=3D"http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-s=
caleResolutionDownBy" style=3D"color:rgb(3,69,117);border-bottom-width:1px;=
border-bottom-style:solid;border-bottom-color:rgb(187,187,187);text-decorat=
ion:none;padding:0px 1px;margin-top:0px;margin-bottom:0px">scaleResolutionD=
ownBy</a></span> =3D <span class=3D"">1.0</span>;</span>
};</span></pre></div><div><br></div><div>The Priority and QoS Model is desc=
ribed in the latest WebRTC 1.0 Editor&#39;s draft (<a href=3D"http://w3c.gi=
thub.io/webrtc-pc/" target=3D"_blank">http://w3c.github.io/webrtc-pc/</a>) =
Section 4.10:=C2=A0</div><div><div><br></div><div><span style=3D"color:rgb(=
0,90,156);font-family:sans-serif;font-size:19.2px;line-height:1.2">4.10=C2=
=A0</span><span style=3D"color:rgb(0,90,156);font-family:sans-serif;font-si=
ze:19.2px;line-height:1.2">Priority and QoS Model</span><br></div><div><p s=
tyle=3D"margin:1em 0px;color:rgb(0,0,0);font-family:sans-serif;font-size:me=
dium;line-height:24px">Many applications have multiple media flows of the s=
ame data type and often some of the flows are more important than others. W=
ebRTC uses the priority and Quality of Service (QoS) framework described in=
 [<cite>RTCWEB-TRANSPORT</cite>] and [<cite>TSVWG-RTCWEB-QOS</cite>] to pro=
vide priority and DSCP marking for packets that will help provide QoS in so=
me networking environments. The priority setting can be used to indicate th=
e relative priority of various flows. The priority API allows the JavaScrip=
t applications to tell the browser whether a particular media flow is high,=
 medium, low or of very low importance to the application by setting the=C2=
=A0<code style=3D"font-size:0.9em;font-family:Menlo,Consolas,&#39;DejaVu Sa=
ns Mono&#39;,Monaco,monospace;color:rgb(200,53,0)">priority</code>property =
of=C2=A0<code style=3D"font-size:0.9em;font-family:Menlo,Consolas,&#39;Deja=
Vu Sans Mono&#39;,Monaco,monospace;color:rgb(200,53,0)"><code style=3D"font=
-size:14.4px;font-family:Menlo,Consolas,&#39;DejaVu Sans Mono&#39;,Monaco,m=
onospace">RTCRtpEncodingParameters</code></code>=C2=A0objects to one of the=
 following values.</p><h4 style=3D"font-stretch:normal;font-size:16px;font-=
family:inherit;line-height:1.2;margin-top:3rem">4.10.1=C2=A0RTCPriorityType=
 Enum</h4></div><div><br></div><div><pre style=3D"margin-left:2em;white-spa=
ce:pre-wrap;font-size:0.9em;font-family:Menlo,Consolas,&#39;DejaVu Sans Mon=
o&#39;,Monaco,monospace;margin-top:1em;margin-bottom:1em;overflow:auto;bord=
er-top-width:1px;border-top-style:solid;border-top-color:rgb(144,184,222);b=
order-bottom-width:1px;border-bottom-style:solid;border-bottom-color:rgb(14=
4,184,222);padding:1em;line-height:17.28px;color:rgb(0,0,0)">enum <span sty=
le=3D"font-weight:bold;color:rgb(0,90,156)">RTCPriorityType</span> {
    &quot;<a href=3D"http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityTyp=
e.very-low" target=3D"_blank" style=3D"color:rgb(3,69,117);border-bottom-wi=
dth:1px;border-bottom-style:solid;border-bottom-color:rgb(187,187,187);text=
-decoration:none;padding:0px 1px;margin-top:0px;margin-bottom:0px">very-low=
</a>&quot;,
    &quot;<a href=3D"http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityTyp=
e.low" target=3D"_blank" style=3D"color:rgb(3,69,117);border-bottom-width:1=
px;border-bottom-style:solid;border-bottom-color:rgb(187,187,187);text-deco=
ration:none;padding:0px 1px;margin-top:0px;margin-bottom:0px">low</a>&quot;=
,
    &quot;<a href=3D"http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityTyp=
e.medium" target=3D"_blank" style=3D"color:rgb(3,69,117);border-bottom-widt=
h:1px;border-bottom-style:solid;border-bottom-color:rgb(187,187,187);text-d=
ecoration:none;padding:0px 1px;margin-top:0px;margin-bottom:0px">medium</a>=
&quot;,
    &quot;<a href=3D"http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityTyp=
e.high" target=3D"_blank" style=3D"color:rgb(3,69,117);border-bottom-width:=
1px;border-bottom-style:solid;border-bottom-color:rgb(187,187,187);text-dec=
oration:none;padding:0px 1px;margin-top:0px;margin-bottom:0px">high</a>&quo=
t;
};</pre></div></div></div><div class=3D"gmail_extra"><br><div class=3D"gmai=
l_quote">On Wed, May 4, 2016 at 1:28 PM, Brian E Carpenter <span dir=3D"ltr=
">&lt;<a href=3D"mailto:brian.e.carpenter@gmail.com" target=3D"_blank">bria=
n.e.carpenter@gmail.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">I fear I have an up-level question: what is the definition of &quot;in=
teractive&quot;?<br>
<br>
draft-ietf-tsvwg-rtcweb-qos doesn&#39;t define it, presumably because it wa=
s<br>
assumed to be obvious, but thinking about David&#39;s suggestions, I decide=
d<br>
that it isn&#39;t so obvious.<br>
<br>
<a href=3D"https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#secti=
on-4" rel=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/html/draf=
t-ietf-tsvwg-rtcweb-qos-15#section-4</a> says:<br>
<br>
=C2=A0 =C2=A0The following are the inputs provided by the WebRTC applicatio=
n to<br>
=C2=A0 =C2=A0the browser:<br>
<br>
=C2=A0 =C2=A0o=C2=A0 Flow Type: The browser provides this input as it knows=
 if the flow<br>
=C2=A0 =C2=A0 =C2=A0 is audio, interactive video with or without audio, non=
-interactive<br>
=C2=A0 =C2=A0 =C2=A0 video with or without audio, or data.<br>
<br>
Firstly, I&#39;m puzzled by the apparent contradiction between the first<br=
>
sentence (in which the app provides input to the browser) and the second<br=
>
sentence (in which to the browser provides input to something else).<br>
<br>
Then, is it clear how the browser &quot;knows&quot; this? If David&#39;s st=
atement below<br>
about the API is correct, the browser actually cannot &quot;provide this in=
put&quot;<br>
as stated in the above quotation. Or I am confused.<br>
<br>
Finally, why is audio not also subdivided into interactive and<br>
non-interactive? As far as I can see, both are logically possible.<br>
<br>
Regards<br>
<span class=3D"HOEnZb"><font color=3D"#888888">=C2=A0 =C2=A0Brian<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
On 05/05/2016 03:26, Black, David wrote:<br>
&gt; Greetings,<br>
&gt;<br>
&gt; I write as the shepherd for the WebRTC QoS draft (draft-ietf-tsvwg-rtc=
web-qos).<br>
&gt; IETF Last Call on this draft uncovered an issue that appears to also r=
equire<br>
&gt; changes in the Web RTC transports draft (draft-ietf-rtcweb-transports)=
.<br>
&gt;<br>
&gt; The Web RTC draft=C2=A0 specifies different Diffserv Codepoints (DSCPs=
) for use with<br>
&gt; Interactive vs. Non-Interactive Video (with or without audio in both c=
ases):<br>
&gt; <a href=3D"https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#=
section-5" rel=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/html=
/draft-ietf-tsvwg-rtcweb-qos-15#section-5</a><br>
&gt;<br>
&gt; The IETF LC issue is: How does an implementer determine whether video =
is<br>
&gt; interactive vs. non-interactive?<br>
&gt;<br>
&gt; The answer (unexpected to at least me) is that this can&#39;t be done =
for current<br>
&gt; WebRTC - all video has to be treated as interactive because there&#39;=
s no API<br>
&gt; support to distinguish interactive video from non-interactive video.=
=C2=A0 That<br>
&gt; doesn&#39;t appear to be stated anywhere obvious, and a TSVWG draft se=
ems<br>
&gt; like the wrong place to do so.<br>
&gt;<br>
&gt; Hence, this message contains initial proposed changes to both the WebR=
TC<br>
&gt; Transports draft and WebRTC QoS draft to resolve this issue without re=
moving<br>
&gt; any of the DSCPs currently specified for video in the WebRTC QoS draft=
.<br>
&gt;<br>
&gt; We should resolve this within the next few days, as the WebRTC QoS dra=
ft is<br>
&gt; (finally!) on the IESG telechat agenda for May 19.<br>
&gt;<br>
&gt; --- WebRTC Transports draft - proposed changes<br>
&gt;<br>
&gt; In Section 4 Media Prioritization, after the following existing text:<=
br>
&gt;<br>
&gt;=C2=A0 =C2=A0 For media, a &quot;media flow&quot;, which can be an &quo=
t;audio flow&quot; or a &quot;video<br>
&gt;=C2=A0 =C2=A0 flow&quot;, is what [RFC7656] calls a &quot;media source&=
quot;, which results in a<br>
&gt;=C2=A0 =C2=A0 &quot;source RTP stream&quot; and one or more &quot;redun=
dancy RTP streams&quot;.=C2=A0 This<br>
&gt;=C2=A0 =C2=A0 specification does not describe prioritization between th=
e RTP<br>
&gt;=C2=A0 =C2=A0 streams that come from a single &quot;media source&quot;.=
<br>
&gt;<br>
&gt; Add a new paragraph:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 All media flows in WebRTC are assumed to be interactive; =
there is<br>
&gt;=C2=A0 =C2=A0 no browser API support for non-interactive media, aside f=
rom sending<br>
&gt;=C2=A0 =C2=A0 non-interactive media content via the data channel.<br>
&gt;<br>
&gt; This ought to cite a W3C reference for the WebRTC API - the following<=
br>
&gt; appears plausible:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 [W3C.WD-webrtc-20120209]<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Bergkvist, A., B=
urnett, D., Jennings, C., and A.<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Narayanan, &quot=
;WebRTC 1.0: Real-time Communication Between<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Browsers&quot;, =
World Wide Web Consortium WD WD-webrtc-<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A020120209, Februa=
ry 2012,<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;<a href=3D"h=
ttp://www.w3.org/TR/2012/WD-webrtc-20120209" rel=3D"noreferrer" target=3D"_=
blank">http://www.w3.org/TR/2012/WD-webrtc-20120209</a>&gt;.<br>
&gt;<br>
&gt; -- WebRTC QoS draft - proposed changes<br>
&gt;<br>
&gt; In Section 5 DSCP Mappings, after the following existing text:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 The browser SHOULD first select the flow type of the flow=
.=C2=A0 Within<br>
&gt;=C2=A0 =C2=A0 the flow type, the relative importance of the flow SHOULD=
 be used to<br>
&gt;=C2=A0 =C2=A0 select the appropriate DSCP value.<br>
&gt;<br>
&gt; Add a new paragraph:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 The current WebRTC browser API does not support non-inter=
active<br>
&gt;=C2=A0 =C2=A0 video; all video is assumed to be interactive [I-D.ietf-r=
tcweb-transports].<br>
&gt;=C2=A0 =C2=A0 Browsers MUST NOT use the DSCP mappings for Non-Interacti=
ve Video<br>
&gt;=C2=A0 =C2=A0 in Table 1.=C2=A0 Non-browser implementations of WebRTC M=
AY use the<br>
&gt;=C2=A0 =C2=A0 DSCP mappings for Non-Interactive Video in Table 1 for vi=
deo that is<br>
&gt;=C2=A0 =C2=A0 known to not be interactive, e.g., all video in a video p=
layback application<br>
&gt;=C2=A0 =C2=A0 based on a custom implementation of the WebRTC protocols =
would not<br>
&gt;=C2=A0 =C2=A0 be interactive.<br>
&gt;<br>
&gt; ------<br>
&gt;<br>
&gt; Comments are welcome - as noted, above,=C2=A0 we should resolve this i=
n<br>
&gt; the next few days in order to avoid pulling the WebRTC QoS draft off<b=
r>
&gt; of the May 19 IESG telechat agenda. Please reply to both the TSVWG and=
<br>
&gt; RTCWEB lists, as we need=C2=A0 a coherent solution across both drafts.=
<br>
&gt;<br>
&gt; I want to thank Colin Perkins for suggesting the video playback exampl=
e.<br>
&gt;<br>
&gt; Thanks, --David (as WebRTC QoS draft shepherd)<br>
&gt;<br>
&gt;<br>
<br>
_______________________________________________<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/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--94eb2c11bee8a0430c05320c36a4--


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To: Bernard Aboba <bernard.aboba@gmail.com>
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From: Brian E Carpenter <brian.e.carpenter@gmail.com>
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Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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Thanks Bernard.

All the same, I think the phrasing I quoted at
https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-4
is problematic; it implies that there's a Boolean property "interactive"
which clearly is not supported in the encoding parameters.

Regards
   Brian

On 05/05/2016 11:01, Bernard Aboba wrote:
> Brian Carpenter said: 
> 
> "Then, is it clear how the browser "knows" this?"
> 
> [BA] The application explicitly provides information to the browser by setting the RTCRtpEncodingParameters.priority attribute
> via the sender.setParameters() API: 
> 
> dictionary RTCRtpEncodingParameters { unsigned long ssrc <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-ssrc>;
> |RTCRtpRtxParameters| rtx <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-rtx>; |RTCRtpFecParameters| fec
> <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-fec>; boolean active
> <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-active>; |RTCPriorityType| priority
> <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-priority>; unsigned long maxBitrate
> <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-maxBitrate>; unsigned long maxFramerate
> <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-maxFramerate>; DOMString rid
> <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-rid>; double scaleResolutionDownBy
> <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-scaleResolutionDownBy> = 1.0; };
> 
> 
> The Priority and QoS Model is described in the latest WebRTC 1.0 Editor's draft (http://w3c.github.io/webrtc-pc/) Section 4.10: 
> 
> 4.10 Priority and QoS Model
> 
> Many applications have multiple media flows of the same data type and often some of the flows are more important than others.
> WebRTC uses the priority and Quality of Service (QoS) framework described in [RTCWEB-TRANSPORT] and [TSVWG-RTCWEB-QOS] to
> provide priority and DSCP marking for packets that will help provide QoS in some networking environments. The priority setting
> can be used to indicate the relative priority of various flows. The priority API allows the JavaScript applications to tell the
> browser whether a particular media flow is high, medium, low or of very low importance to the application by setting
> the |priority|property of ||RTCRtpEncodingParameters|| objects to one of the following values.
> 
> 
>         4.10.1 RTCPriorityType Enum
> 
> 
> enum RTCPriorityType {
>     "very-low <http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.very-low>",
>     "low <http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.low>",
>     "medium <http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.medium>",
>     "high <http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.high>"
> };
> 
> 
> On Wed, May 4, 2016 at 1:28 PM, Brian E Carpenter <brian.e.carpenter@gmail.com <mailto:brian.e.carpenter@gmail.com>> wrote:
> 
>     I fear I have an up-level question: what is the definition of "interactive"?
> 
>     draft-ietf-tsvwg-rtcweb-qos doesn't define it, presumably because it was
>     assumed to be obvious, but thinking about David's suggestions, I decided
>     that it isn't so obvious.
> 
>     https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-4 says:
> 
>        The following are the inputs provided by the WebRTC application to
>        the browser:
> 
>        o  Flow Type: The browser provides this input as it knows if the flow
>           is audio, interactive video with or without audio, non-interactive
>           video with or without audio, or data.
> 
>     Firstly, I'm puzzled by the apparent contradiction between the first
>     sentence (in which the app provides input to the browser) and the second
>     sentence (in which to the browser provides input to something else).
> 
>     Then, is it clear how the browser "knows" this? If David's statement below
>     about the API is correct, the browser actually cannot "provide this input"
>     as stated in the above quotation. Or I am confused.
> 
>     Finally, why is audio not also subdivided into interactive and
>     non-interactive? As far as I can see, both are logically possible.
> 
>     Regards
>        Brian
> 
>     On 05/05/2016 03:26, Black, David wrote:
>     > Greetings,
>     >
>     > I write as the shepherd for the WebRTC QoS draft (draft-ietf-tsvwg-rtcweb-qos).
>     > IETF Last Call on this draft uncovered an issue that appears to also require
>     > changes in the Web RTC transports draft (draft-ietf-rtcweb-transports).
>     >
>     > The Web RTC draft  specifies different Diffserv Codepoints (DSCPs) for use with
>     > Interactive vs. Non-Interactive Video (with or without audio in both cases):
>     > https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-5
>     >
>     > The IETF LC issue is: How does an implementer determine whether video is
>     > interactive vs. non-interactive?
>     >
>     > The answer (unexpected to at least me) is that this can't be done for current
>     > WebRTC - all video has to be treated as interactive because there's no API
>     > support to distinguish interactive video from non-interactive video.  That
>     > doesn't appear to be stated anywhere obvious, and a TSVWG draft seems
>     > like the wrong place to do so.
>     >
>     > Hence, this message contains initial proposed changes to both the WebRTC
>     > Transports draft and WebRTC QoS draft to resolve this issue without removing
>     > any of the DSCPs currently specified for video in the WebRTC QoS draft.
>     >
>     > We should resolve this within the next few days, as the WebRTC QoS draft is
>     > (finally!) on the IESG telechat agenda for May 19.
>     >
>     > --- WebRTC Transports draft - proposed changes
>     >
>     > In Section 4 Media Prioritization, after the following existing text:
>     >
>     >    For media, a "media flow", which can be an "audio flow" or a "video
>     >    flow", is what [RFC7656] calls a "media source", which results in a
>     >    "source RTP stream" and one or more "redundancy RTP streams".  This
>     >    specification does not describe prioritization between the RTP
>     >    streams that come from a single "media source".
>     >
>     > Add a new paragraph:
>     >
>     >    All media flows in WebRTC are assumed to be interactive; there is
>     >    no browser API support for non-interactive media, aside from sending
>     >    non-interactive media content via the data channel.
>     >
>     > This ought to cite a W3C reference for the WebRTC API - the following
>     > appears plausible:
>     >
>     >    [W3C.WD-webrtc-20120209]
>     >               Bergkvist, A., Burnett, D., Jennings, C., and A.
>     >               Narayanan, "WebRTC 1.0: Real-time Communication Between
>     >               Browsers", World Wide Web Consortium WD WD-webrtc-
>     >               20120209, February 2012,
>     >               <http://www.w3.org/TR/2012/WD-webrtc-20120209>.
>     >
>     > -- WebRTC QoS draft - proposed changes
>     >
>     > In Section 5 DSCP Mappings, after the following existing text:
>     >
>     >    The browser SHOULD first select the flow type of the flow.  Within
>     >    the flow type, the relative importance of the flow SHOULD be used to
>     >    select the appropriate DSCP value.
>     >
>     > Add a new paragraph:
>     >
>     >    The current WebRTC browser API does not support non-interactive
>     >    video; all video is assumed to be interactive [I-D.ietf-rtcweb-transports].
>     >    Browsers MUST NOT use the DSCP mappings for Non-Interactive Video
>     >    in Table 1.  Non-browser implementations of WebRTC MAY use the
>     >    DSCP mappings for Non-Interactive Video in Table 1 for video that is
>     >    known to not be interactive, e.g., all video in a video playback application
>     >    based on a custom implementation of the WebRTC protocols would not
>     >    be interactive.
>     >
>     > ------
>     >
>     > Comments are welcome - as noted, above,  we should resolve this in
>     > the next few days in order to avoid pulling the WebRTC QoS draft off
>     > of the May 19 IESG telechat agenda. Please reply to both the TSVWG and
>     > RTCWEB lists, as we need  a coherent solution across both drafts.
>     >
>     > I want to thank Colin Perkins for suggesting the video playback example.
>     >
>     > Thanks, --David (as WebRTC QoS draft shepherd)
>     >
>     >
> 
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Wed, 4 May 2016 18:41:10 -0700
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Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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Brian said:

"All the same, I think the phrasing I quoted at
https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-4
is problematic; it implies that there's a Boolean property "interactive"
which clearly is not supported in the encoding parameters."

[BA] Overall, I'm not clear that this thread is asking the right questions,
or making correct statements about the existing API.  To my mind, the real
question is whether a "non-interactive" marking/priority would deliver
anything useful to application developers, not whether the API could
support it.

The encoding parameters can be used to allow an application to specify a
priority attribute.  If a "non-interactive" marking and associated priority
attribute were to be defined, the current API could allow applications to
specify that.  So citing a 4-year old version of the WebRTC 1.0
specification as justification for the lack of API functionality doesn't
make sense to me.   Since the browser just does what it is told to do by
the application, whether the browser "knows" whether the media is
interactive or not also does not matter.

The real question is whether a  "non-interactive" marking/priority setting
would be useful to application developers.

Uni-directional communications (e.g. "sendonly", "recvonly") is supported
within WebRTC.  Would those kind of flows benefit from a "non-interactive"
marking? In most cases, I suspect that in general it would not be useful (a
possible exception might be use of SVC codecs protecting only the base
layer with FEC/RTX).

It is conceivable that an application might want to attach a track to a
sender that isn't obtained from getUserMedia() (see:
http://w3c.github.io/mediacapture-main/) or getDisplayMedia() (see
https://w3c.github.io/mediacapture-screen-share/).  Again, the question is
whether a "non-interactive" marking/priority would be helpful in those
cases.  I suspect the answer to that is also "no" for similar reasons.

On Wed, May 4, 2016 at 5:36 PM, Brian E Carpenter <
brian.e.carpenter@gmail.com> wrote:

> Thanks Bernard.
>
> All the same, I think the phrasing I quoted at
> https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-4
> is problematic; it implies that there's a Boolean property "interactive"
> which clearly is not supported in the encoding parameters.
>
> Regards
>    Brian
>
> On 05/05/2016 11:01, Bernard Aboba wrote:
> > Brian Carpenter said:
> >
> > "Then, is it clear how the browser "knows" this?"
> >
> > [BA] The application explicitly provides information to the browser by
> setting the RTCRtpEncodingParameters.priority attribute
> > via the sender.setParameters() API:
> >
> > dictionary RTCRtpEncodingParameters { unsigned long ssrc <
> http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-ssrc>;
> > |RTCRtpRtxParameters| rtx <
> http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-rtx>;
> |RTCRtpFecParameters| fec
> > <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-fec>;
> boolean active
> > <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-active>;
> |RTCPriorityType| priority
> > <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-priority>;
> unsigned long maxBitrate
> > <
> http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-maxBitrate>;
> unsigned long maxFramerate
> > <
> http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-maxFramerate>;
> DOMString rid
> > <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-rid>;
> double scaleResolutionDownBy
> > <
> http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-scaleResolutionDownBy>
> = 1.0; };
> >
> >
> > The Priority and QoS Model is described in the latest WebRTC 1.0
> Editor's draft (http://w3c.github.io/webrtc-pc/) Section 4.10:
> >
> > 4.10 Priority and QoS Model
> >
> > Many applications have multiple media flows of the same data type and
> often some of the flows are more important than others.
> > WebRTC uses the priority and Quality of Service (QoS) framework
> described in [RTCWEB-TRANSPORT] and [TSVWG-RTCWEB-QOS] to
> > provide priority and DSCP marking for packets that will help provide QoS
> in some networking environments. The priority setting
> > can be used to indicate the relative priority of various flows. The
> priority API allows the JavaScript applications to tell the
> > browser whether a particular media flow is high, medium, low or of very
> low importance to the application by setting
> > the |priority|property of ||RTCRtpEncodingParameters|| objects to one of
> the following values.
> >
> >
> >         4.10.1 RTCPriorityType Enum
> >
> >
> > enum RTCPriorityType {
> >     "very-low <
> http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.very-low>",
> >     "low <http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.low>",
> >     "medium <
> http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.medium>",
> >     "high <http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.high
> >"
> > };
> >
> >
> > On Wed, May 4, 2016 at 1:28 PM, Brian E Carpenter <
> brian.e.carpenter@gmail.com <mailto:brian.e.carpenter@gmail.com>> wrote:
> >
> >     I fear I have an up-level question: what is the definition of
> "interactive"?
> >
> >     draft-ietf-tsvwg-rtcweb-qos doesn't define it, presumably because it
> was
> >     assumed to be obvious, but thinking about David's suggestions, I
> decided
> >     that it isn't so obvious.
> >
> >     https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-4
> says:
> >
> >        The following are the inputs provided by the WebRTC application to
> >        the browser:
> >
> >        o  Flow Type: The browser provides this input as it knows if the
> flow
> >           is audio, interactive video with or without audio,
> non-interactive
> >           video with or without audio, or data.
> >
> >     Firstly, I'm puzzled by the apparent contradiction between the first
> >     sentence (in which the app provides input to the browser) and the
> second
> >     sentence (in which to the browser provides input to something else).
> >
> >     Then, is it clear how the browser "knows" this? If David's statement
> below
> >     about the API is correct, the browser actually cannot "provide this
> input"
> >     as stated in the above quotation. Or I am confused.
> >
> >     Finally, why is audio not also subdivided into interactive and
> >     non-interactive? As far as I can see, both are logically possible.
> >
> >     Regards
> >        Brian
> >
> >     On 05/05/2016 03:26, Black, David wrote:
> >     > Greetings,
> >     >
> >     > I write as the shepherd for the WebRTC QoS draft
> (draft-ietf-tsvwg-rtcweb-qos).
> >     > IETF Last Call on this draft uncovered an issue that appears to
> also require
> >     > changes in the Web RTC transports draft
> (draft-ietf-rtcweb-transports).
> >     >
> >     > The Web RTC draft  specifies different Diffserv Codepoints (DSCPs)
> for use with
> >     > Interactive vs. Non-Interactive Video (with or without audio in
> both cases):
> >     >
> https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-5
> >     >
> >     > The IETF LC issue is: How does an implementer determine whether
> video is
> >     > interactive vs. non-interactive?
> >     >
> >     > The answer (unexpected to at least me) is that this can't be done
> for current
> >     > WebRTC - all video has to be treated as interactive because
> there's no API
> >     > support to distinguish interactive video from non-interactive
> video.  That
> >     > doesn't appear to be stated anywhere obvious, and a TSVWG draft
> seems
> >     > like the wrong place to do so.
> >     >
> >     > Hence, this message contains initial proposed changes to both the
> WebRTC
> >     > Transports draft and WebRTC QoS draft to resolve this issue
> without removing
> >     > any of the DSCPs currently specified for video in the WebRTC QoS
> draft.
> >     >
> >     > We should resolve this within the next few days, as the WebRTC QoS
> draft is
> >     > (finally!) on the IESG telechat agenda for May 19.
> >     >
> >     > --- WebRTC Transports draft - proposed changes
> >     >
> >     > In Section 4 Media Prioritization, after the following existing
> text:
> >     >
> >     >    For media, a "media flow", which can be an "audio flow" or a
> "video
> >     >    flow", is what [RFC7656] calls a "media source", which results
> in a
> >     >    "source RTP stream" and one or more "redundancy RTP streams".
> This
> >     >    specification does not describe prioritization between the RTP
> >     >    streams that come from a single "media source".
> >     >
> >     > Add a new paragraph:
> >     >
> >     >    All media flows in WebRTC are assumed to be interactive; there
> is
> >     >    no browser API support for non-interactive media, aside from
> sending
> >     >    non-interactive media content via the data channel.
> >     >
> >     > This ought to cite a W3C reference for the WebRTC API - the
> following
> >     > appears plausible:
> >     >
> >     >    [W3C.WD-webrtc-20120209]
> >     >               Bergkvist, A., Burnett, D., Jennings, C., and A.
> >     >               Narayanan, "WebRTC 1.0: Real-time Communication
> Between
> >     >               Browsers", World Wide Web Consortium WD WD-webrtc-
> >     >               20120209, February 2012,
> >     >               <http://www.w3.org/TR/2012/WD-webrtc-20120209>.
> >     >
> >     > -- WebRTC QoS draft - proposed changes
> >     >
> >     > In Section 5 DSCP Mappings, after the following existing text:
> >     >
> >     >    The browser SHOULD first select the flow type of the flow.
> Within
> >     >    the flow type, the relative importance of the flow SHOULD be
> used to
> >     >    select the appropriate DSCP value.
> >     >
> >     > Add a new paragraph:
> >     >
> >     >    The current WebRTC browser API does not support non-interactive
> >     >    video; all video is assumed to be interactive
> [I-D.ietf-rtcweb-transports].
> >     >    Browsers MUST NOT use the DSCP mappings for Non-Interactive
> Video
> >     >    in Table 1.  Non-browser implementations of WebRTC MAY use the
> >     >    DSCP mappings for Non-Interactive Video in Table 1 for video
> that is
> >     >    known to not be interactive, e.g., all video in a video
> playback application
> >     >    based on a custom implementation of the WebRTC protocols would
> not
> >     >    be interactive.
> >     >
> >     > ------
> >     >
> >     > Comments are welcome - as noted, above,  we should resolve this in
> >     > the next few days in order to avoid pulling the WebRTC QoS draft
> off
> >     > of the May 19 IESG telechat agenda. Please reply to both the TSVWG
> and
> >     > RTCWEB lists, as we need  a coherent solution across both drafts.
> >     >
> >     > I want to thank Colin Perkins for suggesting the video playback
> example.
> >     >
> >     > Thanks, --David (as WebRTC QoS draft shepherd)
> >     >
> >     >
> >
> >     _______________________________________________
> >     rtcweb mailing list
> >     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> >     https://www.ietf.org/mailman/listinfo/rtcweb
> >
> >
>

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<div dir=3D"ltr">Brian said:=C2=A0<div><br></div><div>&quot;<span style=3D"=
font-size:12.8px">All the same, I think the phrasing I quoted at</span></di=
v><a href=3D"https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#sec=
tion-4" rel=3D"noreferrer" target=3D"_blank" style=3D"font-size:12.8px">htt=
ps://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-4</a><br st=
yle=3D"font-size:12.8px"><span style=3D"font-size:12.8px">is problematic; i=
t implies that there&#39;s a Boolean property &quot;interactive&quot;</span=
><br style=3D"font-size:12.8px"><div><span style=3D"font-size:12.8px">which=
 clearly is not supported in the encoding parameters.</span>&quot;</div><di=
v><br></div><div>[BA] Overall, I&#39;m not clear that this thread is asking=
 the right questions, or making correct statements about the existing API.=
=C2=A0 To my mind, the real question is whether a &quot;non-interactive&quo=
t; marking/priority would deliver anything useful to application developers=
, not whether the API could support it.=C2=A0</div><div><br></div><div>The =
encoding parameters can be used to allow an application to specify a priori=
ty attribute.=C2=A0 If a &quot;non-interactive&quot; marking and associated=
 priority attribute were to be defined, the current API could allow applica=
tions to specify that.=C2=A0 So citing a 4-year old version of the WebRTC 1=
.0 specification as justification for the lack of API functionality doesn&#=
39;t make sense to me. =C2=A0 Since the browser just does what it is told t=
o do by the application, whether the browser &quot;knows&quot; whether the =
media is interactive or not also does not matter.=C2=A0</div><div><br></div=
><div>The real question is whether a =C2=A0&quot;non-interactive&quot; mark=
ing/priority setting would be useful to application developers.=C2=A0</div>=
<div><br></div><div>Uni-directional communications (e.g. &quot;sendonly&quo=
t;, &quot;recvonly&quot;) is supported within WebRTC.=C2=A0 Would those kin=
d of flows benefit from a &quot;non-interactive&quot; marking? In most case=
s, I suspect that in general it would not be useful (a possible exception m=
ight be use of SVC codecs protecting only the base layer with FEC/RTX). =C2=
=A0</div><div><br></div><div>It is conceivable that an application might wa=
nt to attach a track to a sender that isn&#39;t obtained from getUserMedia(=
) (see: <a href=3D"http://w3c.github.io/mediacapture-main/">http://w3c.gith=
ub.io/mediacapture-main/</a>) or getDisplayMedia() (see=C2=A0<a href=3D"htt=
ps://w3c.github.io/mediacapture-screen-share/">https://w3c.github.io/mediac=
apture-screen-share/</a>).=C2=A0 Again, the question is whether a &quot;non=
-interactive&quot; marking/priority would be helpful in those cases.=C2=A0 =
I suspect the answer to that is also &quot;no&quot; for similar reasons.=C2=
=A0</div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On=
 Wed, May 4, 2016 at 5:36 PM, Brian E Carpenter <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:brian.e.carpenter@gmail.com" target=3D"_blank">brian.e.carpent=
er@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Thanks=
 Bernard.<br>
<br>
All the same, I think the phrasing I quoted at<br>
<a href=3D"https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#secti=
on-4" rel=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/html/draf=
t-ietf-tsvwg-rtcweb-qos-15#section-4</a><br>
is problematic; it implies that there&#39;s a Boolean property &quot;intera=
ctive&quot;<br>
which clearly is not supported in the encoding parameters.<br>
<br>
Regards<br>
=C2=A0 =C2=A0Brian<br>
<span class=3D""><br>
On 05/05/2016 11:01, Bernard Aboba wrote:<br>
&gt; Brian Carpenter said:<br>
&gt;<br>
&gt; &quot;Then, is it clear how the browser &quot;knows&quot; this?&quot;<=
br>
&gt;<br>
&gt; [BA] The application explicitly provides information to the browser by=
 setting the RTCRtpEncodingParameters.priority attribute<br>
&gt; via the sender.setParameters() API:<br>
&gt;<br>
</span>&gt; dictionary RTCRtpEncodingParameters { unsigned long ssrc &lt;<a=
 href=3D"http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-ssrc=
" rel=3D"noreferrer" target=3D"_blank">http://w3c.github.io/webrtc-pc/#widl=
-RTCRtpEncodingParameters-ssrc</a>&gt;;<br>
&gt; |RTCRtpRtxParameters| rtx &lt;<a href=3D"http://w3c.github.io/webrtc-p=
c/#widl-RTCRtpEncodingParameters-rtx" rel=3D"noreferrer" target=3D"_blank">=
http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-rtx</a>&gt;; =
|RTCRtpFecParameters| fec<br>
&gt; &lt;<a href=3D"http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingPara=
meters-fec" rel=3D"noreferrer" target=3D"_blank">http://w3c.github.io/webrt=
c-pc/#widl-RTCRtpEncodingParameters-fec</a>&gt;; boolean active<br>
&gt; &lt;<a href=3D"http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingPara=
meters-active" rel=3D"noreferrer" target=3D"_blank">http://w3c.github.io/we=
brtc-pc/#widl-RTCRtpEncodingParameters-active</a>&gt;; |RTCPriorityType| pr=
iority<br>
&gt; &lt;<a href=3D"http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingPara=
meters-priority" rel=3D"noreferrer" target=3D"_blank">http://w3c.github.io/=
webrtc-pc/#widl-RTCRtpEncodingParameters-priority</a>&gt;; unsigned long ma=
xBitrate<br>
&gt; &lt;<a href=3D"http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingPara=
meters-maxBitrate" rel=3D"noreferrer" target=3D"_blank">http://w3c.github.i=
o/webrtc-pc/#widl-RTCRtpEncodingParameters-maxBitrate</a>&gt;; unsigned lon=
g maxFramerate<br>
&gt; &lt;<a href=3D"http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingPara=
meters-maxFramerate" rel=3D"noreferrer" target=3D"_blank">http://w3c.github=
.io/webrtc-pc/#widl-RTCRtpEncodingParameters-maxFramerate</a>&gt;; DOMStrin=
g rid<br>
&gt; &lt;<a href=3D"http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingPara=
meters-rid" rel=3D"noreferrer" target=3D"_blank">http://w3c.github.io/webrt=
c-pc/#widl-RTCRtpEncodingParameters-rid</a>&gt;; double scaleResolutionDown=
By<br>
&gt; &lt;<a href=3D"http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingPara=
meters-scaleResolutionDownBy" rel=3D"noreferrer" target=3D"_blank">http://w=
3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-scaleResolutionDownBy=
</a>&gt; =3D 1.0; };<br>
<span class=3D"">&gt;<br>
&gt;<br>
&gt; The Priority and QoS Model is described in the latest WebRTC 1.0 Edito=
r&#39;s draft (<a href=3D"http://w3c.github.io/webrtc-pc/" rel=3D"noreferre=
r" target=3D"_blank">http://w3c.github.io/webrtc-pc/</a>) Section 4.10:<br>
&gt;<br>
&gt; 4.10 Priority and QoS Model<br>
&gt;<br>
&gt; Many applications have multiple media flows of the same data type and =
often some of the flows are more important than others.<br>
&gt; WebRTC uses the priority and Quality of Service (QoS) framework descri=
bed in [RTCWEB-TRANSPORT] and [TSVWG-RTCWEB-QOS] to<br>
&gt; provide priority and DSCP marking for packets that will help provide Q=
oS in some networking environments. The priority setting<br>
&gt; can be used to indicate the relative priority of various flows. The pr=
iority API allows the JavaScript applications to tell the<br>
&gt; browser whether a particular media flow is high, medium, low or of ver=
y low importance to the application by setting<br>
</span>&gt; the |priority|property of ||RTCRtpEncodingParameters|| objects =
to one of the following values.<br>
<span class=3D"">&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A04.10.1 RTCPriorityType Enum<br>
&gt;<br>
&gt;<br>
&gt; enum RTCPriorityType {<br>
</span>&gt;=C2=A0 =C2=A0 =C2=A0&quot;very-low &lt;<a href=3D"http://w3c.git=
hub.io/webrtc-pc/#idl-def-RTCPriorityType.very-low" rel=3D"noreferrer" targ=
et=3D"_blank">http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.very-=
low</a>&gt;&quot;,<br>
&gt;=C2=A0 =C2=A0 =C2=A0&quot;low &lt;<a href=3D"http://w3c.github.io/webrt=
c-pc/#idl-def-RTCPriorityType.low" rel=3D"noreferrer" target=3D"_blank">htt=
p://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.low</a>&gt;&quot;,<br>
&gt;=C2=A0 =C2=A0 =C2=A0&quot;medium &lt;<a href=3D"http://w3c.github.io/we=
brtc-pc/#idl-def-RTCPriorityType.medium" rel=3D"noreferrer" target=3D"_blan=
k">http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.medium</a>&gt;&q=
uot;,<br>
&gt;=C2=A0 =C2=A0 =C2=A0&quot;high &lt;<a href=3D"http://w3c.github.io/webr=
tc-pc/#idl-def-RTCPriorityType.high" rel=3D"noreferrer" target=3D"_blank">h=
ttp://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.high</a>&gt;&quot;<b=
r>
&gt; };<br>
<div><div class=3D"h5">&gt;<br>
&gt;<br>
&gt; On Wed, May 4, 2016 at 1:28 PM, Brian E Carpenter &lt;<a href=3D"mailt=
o:brian.e.carpenter@gmail.com">brian.e.carpenter@gmail.com</a> &lt;mailto:<=
a href=3D"mailto:brian.e.carpenter@gmail.com">brian.e.carpenter@gmail.com</=
a>&gt;&gt; wrote:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0I fear I have an up-level question: what is the def=
inition of &quot;interactive&quot;?<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0draft-ietf-tsvwg-rtcweb-qos doesn&#39;t define it, =
presumably because it was<br>
&gt;=C2=A0 =C2=A0 =C2=A0assumed to be obvious, but thinking about David&#39=
;s suggestions, I decided<br>
&gt;=C2=A0 =C2=A0 =C2=A0that it isn&#39;t so obvious.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0<a href=3D"https://tools.ietf.org/html/draft-ietf-t=
svwg-rtcweb-qos-15#section-4" rel=3D"noreferrer" target=3D"_blank">https://=
tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-4</a> says:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 The following are the inputs provided by th=
e WebRTC application to<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 the browser:<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 o=C2=A0 Flow Type: The browser provides thi=
s input as it knows if the flow<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0is audio, interactive video wi=
th or without audio, non-interactive<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0video with or without audio, o=
r data.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0Firstly, I&#39;m puzzled by the apparent contradict=
ion between the first<br>
&gt;=C2=A0 =C2=A0 =C2=A0sentence (in which the app provides input to the br=
owser) and the second<br>
&gt;=C2=A0 =C2=A0 =C2=A0sentence (in which to the browser provides input to=
 something else).<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0Then, is it clear how the browser &quot;knows&quot;=
 this? If David&#39;s statement below<br>
&gt;=C2=A0 =C2=A0 =C2=A0about the API is correct, the browser actually cann=
ot &quot;provide this input&quot;<br>
&gt;=C2=A0 =C2=A0 =C2=A0as stated in the above quotation. Or I am confused.=
<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0Finally, why is audio not also subdivided into inte=
ractive and<br>
&gt;=C2=A0 =C2=A0 =C2=A0non-interactive? As far as I can see, both are logi=
cally possible.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0Regards<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 Brian<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0On 05/05/2016 03:26, Black, David wrote:<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; Greetings,<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; I write as the shepherd for the WebRTC QoS dra=
ft (draft-ietf-tsvwg-rtcweb-qos).<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; IETF Last Call on this draft uncovered an issu=
e that appears to also require<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; changes in the Web RTC transports draft (draft=
-ietf-rtcweb-transports).<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; The Web RTC draft=C2=A0 specifies different Di=
ffserv Codepoints (DSCPs) for use with<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; Interactive vs. Non-Interactive Video (with or=
 without audio in both cases):<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; <a href=3D"https://tools.ietf.org/html/draft-i=
etf-tsvwg-rtcweb-qos-15#section-5" rel=3D"noreferrer" target=3D"_blank">htt=
ps://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-5</a><br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; The IETF LC issue is: How does an implementer =
determine whether video is<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; interactive vs. non-interactive?<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; The answer (unexpected to at least me) is that=
 this can&#39;t be done for current<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; WebRTC - all video has to be treated as intera=
ctive because there&#39;s no API<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; support to distinguish interactive video from =
non-interactive video.=C2=A0 That<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; doesn&#39;t appear to be stated anywhere obvio=
us, and a TSVWG draft seems<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; like the wrong place to do so.<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; Hence, this message contains initial proposed =
changes to both the WebRTC<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; Transports draft and WebRTC QoS draft to resol=
ve this issue without removing<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; any of the DSCPs currently specified for video=
 in the WebRTC QoS draft.<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; We should resolve this within the next few day=
s, as the WebRTC QoS draft is<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; (finally!) on the IESG telechat agenda for May=
 19.<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; --- WebRTC Transports draft - proposed changes=
<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; In Section 4 Media Prioritization, after the f=
ollowing existing text:<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 For media, a &quot;media flow&quo=
t;, which can be an &quot;audio flow&quot; or a &quot;video<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 flow&quot;, is what [RFC7656] cal=
ls a &quot;media source&quot;, which results in a<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 &quot;source RTP stream&quot; and=
 one or more &quot;redundancy RTP streams&quot;.=C2=A0 This<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 specification does not describe p=
rioritization between the RTP<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 streams that come from a single &=
quot;media source&quot;.<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; Add a new paragraph:<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 All media flows in WebRTC are ass=
umed to be interactive; there is<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 no browser API support for non-in=
teractive media, aside from sending<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 non-interactive media content via=
 the data channel.<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; This ought to cite a W3C reference for the Web=
RTC API - the following<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; appears plausible:<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 [W3C.WD-webrtc-20120209]<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0Bergkvist, A., Burnett, D., Jennings, C., and A.<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0Narayanan, &quot;WebRTC 1.0: Real-time Communication Between<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0Browsers&quot;, World Wide Web Consortium WD WD-webrtc-<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A020120209, February 2012,<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0&lt;<a href=3D"http://www.w3.org/TR/2012/WD-webrtc-20120209" rel=
=3D"noreferrer" target=3D"_blank">http://www.w3.org/TR/2012/WD-webrtc-20120=
209</a>&gt;.<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; -- WebRTC QoS draft - proposed changes<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; In Section 5 DSCP Mappings, after the followin=
g existing text:<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 The browser SHOULD first select t=
he flow type of the flow.=C2=A0 Within<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 the flow type, the relative impor=
tance of the flow SHOULD be used to<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 select the appropriate DSCP value=
.<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; Add a new paragraph:<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 The current WebRTC browser API do=
es not support non-interactive<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 video; all video is assumed to be=
 interactive [I-D.ietf-rtcweb-transports].<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 Browsers MUST NOT use the DSCP ma=
ppings for Non-Interactive Video<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 in Table 1.=C2=A0 Non-browser imp=
lementations of WebRTC MAY use the<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 DSCP mappings for Non-Interactive=
 Video in Table 1 for video that is<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 known to not be interactive, e.g.=
, all video in a video playback application<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 based on a custom implementation =
of the WebRTC protocols would not<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;=C2=A0 =C2=A0 be interactive.<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; ------<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; Comments are welcome - as noted, above,=C2=A0 =
we should resolve this in<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; the next few days in order to avoid pulling th=
e WebRTC QoS draft off<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; of the May 19 IESG telechat agenda. Please rep=
ly to both the TSVWG and<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; RTCWEB lists, as we need=C2=A0 a coherent solu=
tion across both drafts.<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; I want to thank Colin Perkins for suggesting t=
he video playback example.<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt; Thanks, --David (as WebRTC QoS draft shepherd)=
<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0&gt;<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0_______________________________________________<br>
&gt;=C2=A0 =C2=A0 =C2=A0rtcweb mailing list<br>
</div></div>&gt;=C2=A0 =C2=A0 =C2=A0<a href=3D"mailto:rtcweb@ietf.org">rtcw=
eb@ietf.org</a> &lt;mailto:<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.o=
rg</a>&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0<a href=3D"https://www.ietf.org/mailman/listinfo/rt=
cweb" rel=3D"noreferrer" target=3D"_blank">https://www.ietf.org/mailman/lis=
tinfo/rtcweb</a><br>
&gt;<br>
&gt;<br>
</blockquote></div><br></div>

--94eb2c12541800d71f05320e7161--


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To: Bernard Aboba <bernard.aboba@gmail.com>
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From: Brian E Carpenter <brian.e.carpenter@gmail.com>
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Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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> To my mind, the real
> question is whether a "non-interactive" marking/priority would deliver
> anything useful to application developers, 

Absolutely. As I understand things, a media stream in an interactive
session has one requirement that is definitely more stringent than
for a non-interactive session: lower latency. So the question is how
to ensure that the most suitable DSCP is chosen to achieve the required
latency. But that takes us back to
https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-5
where by some magic the code needs to know the flow type.

This is why trying to compress the subtleties of QoS into a "priority"
parameter is bound to fail. I don't understand how
https://tools.ietf.org/html/draft-ietf-rtcweb-transports-12#section-4.2
is going to work without extending the API.

Regards
   Brian


On 05/05/2016 13:41, Bernard Aboba wrote:
> Brian said:
> 
> "All the same, I think the phrasing I quoted at
> https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-4
> is problematic; it implies that there's a Boolean property "interactive"
> which clearly is not supported in the encoding parameters."
> 
> [BA] Overall, I'm not clear that this thread is asking the right questions,
> or making correct statements about the existing API.  To my mind, the real
> question is whether a "non-interactive" marking/priority would deliver
> anything useful to application developers, not whether the API could
> support it.
> 
> The encoding parameters can be used to allow an application to specify a
> priority attribute.  If a "non-interactive" marking and associated priority
> attribute were to be defined, the current API could allow applications to
> specify that.  So citing a 4-year old version of the WebRTC 1.0
> specification as justification for the lack of API functionality doesn't
> make sense to me.   Since the browser just does what it is told to do by
> the application, whether the browser "knows" whether the media is
> interactive or not also does not matter.
> 
> The real question is whether a  "non-interactive" marking/priority setting
> would be useful to application developers.
> 
> Uni-directional communications (e.g. "sendonly", "recvonly") is supported
> within WebRTC.  Would those kind of flows benefit from a "non-interactive"
> marking? In most cases, I suspect that in general it would not be useful (a
> possible exception might be use of SVC codecs protecting only the base
> layer with FEC/RTX).
> 
> It is conceivable that an application might want to attach a track to a
> sender that isn't obtained from getUserMedia() (see:
> http://w3c.github.io/mediacapture-main/) or getDisplayMedia() (see
> https://w3c.github.io/mediacapture-screen-share/).  Again, the question is
> whether a "non-interactive" marking/priority would be helpful in those
> cases.  I suspect the answer to that is also "no" for similar reasons.
> 
> On Wed, May 4, 2016 at 5:36 PM, Brian E Carpenter <
> brian.e.carpenter@gmail.com> wrote:
> 
>> Thanks Bernard.
>>
>> All the same, I think the phrasing I quoted at
>> https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-4
>> is problematic; it implies that there's a Boolean property "interactive"
>> which clearly is not supported in the encoding parameters.
>>
>> Regards
>>    Brian
>>
>> On 05/05/2016 11:01, Bernard Aboba wrote:
>>> Brian Carpenter said:
>>>
>>> "Then, is it clear how the browser "knows" this?"
>>>
>>> [BA] The application explicitly provides information to the browser by
>> setting the RTCRtpEncodingParameters.priority attribute
>>> via the sender.setParameters() API:
>>>
>>> dictionary RTCRtpEncodingParameters { unsigned long ssrc <
>> http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-ssrc>;
>>> |RTCRtpRtxParameters| rtx <
>> http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-rtx>;
>> |RTCRtpFecParameters| fec
>>> <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-fec>;
>> boolean active
>>> <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-active>;
>> |RTCPriorityType| priority
>>> <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-priority>;
>> unsigned long maxBitrate
>>> <
>> http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-maxBitrate>;
>> unsigned long maxFramerate
>>> <
>> http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-maxFramerate>;
>> DOMString rid
>>> <http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-rid>;
>> double scaleResolutionDownBy
>>> <
>> http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-scaleResolutionDownBy>
>> = 1.0; };
>>>
>>>
>>> The Priority and QoS Model is described in the latest WebRTC 1.0
>> Editor's draft (http://w3c.github.io/webrtc-pc/) Section 4.10:
>>>
>>> 4.10 Priority and QoS Model
>>>
>>> Many applications have multiple media flows of the same data type and
>> often some of the flows are more important than others.
>>> WebRTC uses the priority and Quality of Service (QoS) framework
>> described in [RTCWEB-TRANSPORT] and [TSVWG-RTCWEB-QOS] to
>>> provide priority and DSCP marking for packets that will help provide QoS
>> in some networking environments. The priority setting
>>> can be used to indicate the relative priority of various flows. The
>> priority API allows the JavaScript applications to tell the
>>> browser whether a particular media flow is high, medium, low or of very
>> low importance to the application by setting
>>> the |priority|property of ||RTCRtpEncodingParameters|| objects to one of
>> the following values.
>>>
>>>
>>>         4.10.1 RTCPriorityType Enum
>>>
>>>
>>> enum RTCPriorityType {
>>>     "very-low <
>> http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.very-low>",
>>>     "low <http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.low>",
>>>     "medium <
>> http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.medium>",
>>>     "high <http://w3c.github.io/webrtc-pc/#idl-def-RTCPriorityType.high
>>> "
>>> };
>>>
>>>
>>> On Wed, May 4, 2016 at 1:28 PM, Brian E Carpenter <
>> brian.e.carpenter@gmail.com <mailto:brian.e.carpenter@gmail.com>> wrote:
>>>
>>>     I fear I have an up-level question: what is the definition of
>> "interactive"?
>>>
>>>     draft-ietf-tsvwg-rtcweb-qos doesn't define it, presumably because it
>> was
>>>     assumed to be obvious, but thinking about David's suggestions, I
>> decided
>>>     that it isn't so obvious.
>>>
>>>     https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-4
>> says:
>>>
>>>        The following are the inputs provided by the WebRTC application to
>>>        the browser:
>>>
>>>        o  Flow Type: The browser provides this input as it knows if the
>> flow
>>>           is audio, interactive video with or without audio,
>> non-interactive
>>>           video with or without audio, or data.
>>>
>>>     Firstly, I'm puzzled by the apparent contradiction between the first
>>>     sentence (in which the app provides input to the browser) and the
>> second
>>>     sentence (in which to the browser provides input to something else).
>>>
>>>     Then, is it clear how the browser "knows" this? If David's statement
>> below
>>>     about the API is correct, the browser actually cannot "provide this
>> input"
>>>     as stated in the above quotation. Or I am confused.
>>>
>>>     Finally, why is audio not also subdivided into interactive and
>>>     non-interactive? As far as I can see, both are logically possible.
>>>
>>>     Regards
>>>        Brian
>>>
>>>     On 05/05/2016 03:26, Black, David wrote:
>>>     > Greetings,
>>>     >
>>>     > I write as the shepherd for the WebRTC QoS draft
>> (draft-ietf-tsvwg-rtcweb-qos).
>>>     > IETF Last Call on this draft uncovered an issue that appears to
>> also require
>>>     > changes in the Web RTC transports draft
>> (draft-ietf-rtcweb-transports).
>>>     >
>>>     > The Web RTC draft  specifies different Diffserv Codepoints (DSCPs)
>> for use with
>>>     > Interactive vs. Non-Interactive Video (with or without audio in
>> both cases):
>>>     >
>> https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-5
>>>     >
>>>     > The IETF LC issue is: How does an implementer determine whether
>> video is
>>>     > interactive vs. non-interactive?
>>>     >
>>>     > The answer (unexpected to at least me) is that this can't be done
>> for current
>>>     > WebRTC - all video has to be treated as interactive because
>> there's no API
>>>     > support to distinguish interactive video from non-interactive
>> video.  That
>>>     > doesn't appear to be stated anywhere obvious, and a TSVWG draft
>> seems
>>>     > like the wrong place to do so.
>>>     >
>>>     > Hence, this message contains initial proposed changes to both the
>> WebRTC
>>>     > Transports draft and WebRTC QoS draft to resolve this issue
>> without removing
>>>     > any of the DSCPs currently specified for video in the WebRTC QoS
>> draft.
>>>     >
>>>     > We should resolve this within the next few days, as the WebRTC QoS
>> draft is
>>>     > (finally!) on the IESG telechat agenda for May 19.
>>>     >
>>>     > --- WebRTC Transports draft - proposed changes
>>>     >
>>>     > In Section 4 Media Prioritization, after the following existing
>> text:
>>>     >
>>>     >    For media, a "media flow", which can be an "audio flow" or a
>> "video
>>>     >    flow", is what [RFC7656] calls a "media source", which results
>> in a
>>>     >    "source RTP stream" and one or more "redundancy RTP streams".
>> This
>>>     >    specification does not describe prioritization between the RTP
>>>     >    streams that come from a single "media source".
>>>     >
>>>     > Add a new paragraph:
>>>     >
>>>     >    All media flows in WebRTC are assumed to be interactive; there
>> is
>>>     >    no browser API support for non-interactive media, aside from
>> sending
>>>     >    non-interactive media content via the data channel.
>>>     >
>>>     > This ought to cite a W3C reference for the WebRTC API - the
>> following
>>>     > appears plausible:
>>>     >
>>>     >    [W3C.WD-webrtc-20120209]
>>>     >               Bergkvist, A., Burnett, D., Jennings, C., and A.
>>>     >               Narayanan, "WebRTC 1.0: Real-time Communication
>> Between
>>>     >               Browsers", World Wide Web Consortium WD WD-webrtc-
>>>     >               20120209, February 2012,
>>>     >               <http://www.w3.org/TR/2012/WD-webrtc-20120209>.
>>>     >
>>>     > -- WebRTC QoS draft - proposed changes
>>>     >
>>>     > In Section 5 DSCP Mappings, after the following existing text:
>>>     >
>>>     >    The browser SHOULD first select the flow type of the flow.
>> Within
>>>     >    the flow type, the relative importance of the flow SHOULD be
>> used to
>>>     >    select the appropriate DSCP value.
>>>     >
>>>     > Add a new paragraph:
>>>     >
>>>     >    The current WebRTC browser API does not support non-interactive
>>>     >    video; all video is assumed to be interactive
>> [I-D.ietf-rtcweb-transports].
>>>     >    Browsers MUST NOT use the DSCP mappings for Non-Interactive
>> Video
>>>     >    in Table 1.  Non-browser implementations of WebRTC MAY use the
>>>     >    DSCP mappings for Non-Interactive Video in Table 1 for video
>> that is
>>>     >    known to not be interactive, e.g., all video in a video
>> playback application
>>>     >    based on a custom implementation of the WebRTC protocols would
>> not
>>>     >    be interactive.
>>>     >
>>>     > ------
>>>     >
>>>     > Comments are welcome - as noted, above,  we should resolve this in
>>>     > the next few days in order to avoid pulling the WebRTC QoS draft
>> off
>>>     > of the May 19 IESG telechat agenda. Please reply to both the TSVWG
>> and
>>>     > RTCWEB lists, as we need  a coherent solution across both drafts.
>>>     >
>>>     > I want to thank Colin Perkins for suggesting the video playback
>> example.
>>>     >
>>>     > Thanks, --David (as WebRTC QoS draft shepherd)
>>>     >
>>>     >
>>>
>>>     _______________________________________________
>>>     rtcweb mailing list
>>>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>>     https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>
> 


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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Brian E Carpenter <brian.e.carpenter@gmail.com>, "Black, David" <david.black@emc.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] [tsvwg] Diffserv QoS for Video
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Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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On 04/05/16 22:28, Brian E Carpenter wrote:=0A=
> I fear I have an up-level question: what is the definition of "interactiv=
e"?=0A=
>=0A=
> draft-ietf-tsvwg-rtcweb-qos doesn't define it, presumably because it was=
=0A=
> assumed to be obvious, but thinking about David's suggestions, I decided=
=0A=
> that it isn't so obvious.=0A=
>=0A=
> https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-4 says=
:=0A=
>=0A=
>     The following are the inputs provided by the WebRTC application to=0A=
>     the browser:=0A=
>=0A=
>     o  Flow Type: The browser provides this input as it knows if the flow=
=0A=
>        is audio, interactive video with or without audio, non-interactive=
=0A=
>        video with or without audio, or data.=0A=
>=0A=
> Firstly, I'm puzzled by the apparent contradiction between the first=0A=
> sentence (in which the app provides input to the browser) and the second=
=0A=
> sentence (in which to the browser provides input to something else).=0A=
=0A=
The sentences contradict, but with the API defined the second sentence =0A=
is true.=0A=
=0A=
>=0A=
> Then, is it clear how the browser "knows" this? If David's statement belo=
w=0A=
> about the API is correct, the browser actually cannot "provide this input=
"=0A=
> as stated in the above quotation. Or I am confused.=0A=
=0A=
There is no browser API to define if the video (or audio) is interactive =
=0A=
or not, and the reason is that we've dealt only with interactive flows.=0A=
=0A=
I've heard people talk about more robust flows with higher latency for =0A=
situations like lectures and so on, but that is not part of WebRTC 1.0 =0A=
(the API standard).=0A=
=0A=
>=0A=
> Finally, why is audio not also subdivided into interactive and=0A=
> non-interactive? As far as I can see, both are logically possible.=0A=
>=0A=
> Regards=0A=
>     Brian=0A=
>=0A=
> On 05/05/2016 03:26, Black, David wrote:=0A=
>> Greetings,=0A=
>>=0A=
>> I write as the shepherd for the WebRTC QoS draft (draft-ietf-tsvwg-rtcwe=
b-qos).=0A=
>> IETF Last Call on this draft uncovered an issue that appears to also req=
uire=0A=
>> changes in the Web RTC transports draft (draft-ietf-rtcweb-transports).=
=0A=
>>=0A=
>> The Web RTC draft  specifies different Diffserv Codepoints (DSCPs) for u=
se with=0A=
>> Interactive vs. Non-Interactive Video (with or without audio in both cas=
es):=0A=
>> https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-5=0A=
>>=0A=
>> The IETF LC issue is: How does an implementer determine whether video is=
=0A=
>> interactive vs. non-interactive?=0A=
>>=0A=
>> The answer (unexpected to at least me) is that this can't be done for cu=
rrent=0A=
>> WebRTC - all video has to be treated as interactive because there's no A=
PI=0A=
>> support to distinguish interactive video from non-interactive video.  Th=
at=0A=
>> doesn't appear to be stated anywhere obvious, and a TSVWG draft seems=0A=
>> like the wrong place to do so.=0A=
>>=0A=
>> Hence, this message contains initial proposed changes to both the WebRTC=
=0A=
>> Transports draft and WebRTC QoS draft to resolve this issue without remo=
ving=0A=
>> any of the DSCPs currently specified for video in the WebRTC QoS draft.=
=0A=
>>=0A=
>> We should resolve this within the next few days, as the WebRTC QoS draft=
 is=0A=
>> (finally!) on the IESG telechat agenda for May 19.=0A=
>>=0A=
>> --- WebRTC Transports draft - proposed changes=0A=
>>=0A=
>> In Section 4 Media Prioritization, after the following existing text:=0A=
>>=0A=
>>     For media, a "media flow", which can be an "audio flow" or a "video=
=0A=
>>     flow", is what [RFC7656] calls a "media source", which results in a=
=0A=
>>     "source RTP stream" and one or more "redundancy RTP streams".  This=
=0A=
>>     specification does not describe prioritization between the RTP=0A=
>>     streams that come from a single "media source".=0A=
>>=0A=
>> Add a new paragraph:=0A=
>>=0A=
>>     All media flows in WebRTC are assumed to be interactive; there is=0A=
>>     no browser API support for non-interactive media, aside from sending=
=0A=
>>     non-interactive media content via the data channel.=0A=
>>=0A=
>> This ought to cite a W3C reference for the WebRTC API - the following=0A=
>> appears plausible:=0A=
>>=0A=
>>     [W3C.WD-webrtc-20120209]=0A=
>>                Bergkvist, A., Burnett, D., Jennings, C., and A.=0A=
>>                Narayanan, "WebRTC 1.0: Real-time Communication Between=
=0A=
>>                Browsers", World Wide Web Consortium WD WD-webrtc-=0A=
>>                20120209, February 2012,=0A=
>>                <http://www.w3.org/TR/2012/WD-webrtc-20120209>.=0A=
>>=0A=
>> -- WebRTC QoS draft - proposed changes=0A=
>>=0A=
>> In Section 5 DSCP Mappings, after the following existing text:=0A=
>>=0A=
>>     The browser SHOULD first select the flow type of the flow.  Within=
=0A=
>>     the flow type, the relative importance of the flow SHOULD be used to=
=0A=
>>     select the appropriate DSCP value.=0A=
>>=0A=
>> Add a new paragraph:=0A=
>>=0A=
>>     The current WebRTC browser API does not support non-interactive=0A=
>>     video; all video is assumed to be interactive [I-D.ietf-rtcweb-trans=
ports].=0A=
>>     Browsers MUST NOT use the DSCP mappings for Non-Interactive Video=0A=
>>     in Table 1.  Non-browser implementations of WebRTC MAY use the=0A=
>>     DSCP mappings for Non-Interactive Video in Table 1 for video that is=
=0A=
>>     known to not be interactive, e.g., all video in a video playback app=
lication=0A=
>>     based on a custom implementation of the WebRTC protocols would not=
=0A=
>>     be interactive.=0A=
>>=0A=
>> ------=0A=
>>=0A=
>> Comments are welcome - as noted, above,  we should resolve this in=0A=
>> the next few days in order to avoid pulling the WebRTC QoS draft off=0A=
>> of the May 19 IESG telechat agenda. Please reply to both the TSVWG and=
=0A=
>> RTCWEB lists, as we need  a coherent solution across both drafts.=0A=
>>=0A=
>> I want to thank Colin Perkins for suggesting the video playback example.=
=0A=
>>=0A=
>> Thanks, --David (as WebRTC QoS draft shepherd)=0A=
>>=0A=
>>=0A=
>=0A=
> _______________________________________________=0A=
> rtcweb mailing list=0A=
> rtcweb@ietf.org=0A=
> https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
=0A=


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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Brian E Carpenter" <brian.e.carpenter@gmail.com>, "Black, David" <david.black@emc.com>, "tsvwg@ietf.org" <tsvwg@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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Brian,

>I fear I have an up-level question: what is the definition of=20
>"interactive"?

That comes from RFC 4594, which comes from G.1010.

>draft-ietf-tsvwg-rtcweb-qos doesn't define it, presumably because it=20
>was
>assumed to be obvious, but thinking about David's suggestions, I=20
>decided
>that it isn't so obvious.
>
>https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-4=20
>says:
>
>    The following are the inputs provided by the WebRTC application to
>    the browser:
>
>    o  Flow Type: The browser provides this input as it knows if the=20
>flow
>       is audio, interactive video with or without audio,=20
>non-interactive
>       video with or without audio, or data.

This paragraph has been modified during the course of discussions during=
=20
LC.  The text now reads:

     Flow Type: The application provides this input because it knows
     if the flow is audio, interactive video with or without
     audio, non-interactive video with or without audio, or data.
     For audio that is associated with a video flow, the flow
     type of the associated video MAY be used instead of the
     associated audio type.

>Firstly, I'm puzzled by the apparent contradiction between the first
>sentence (in which the app provides input to the browser) and the=20
>second
>sentence (in which to the browser provides input to something else).

This should have been "application", as it's providing this input to the=
=20
browser, in spite of the fact that the current API does not allow the=20
application to indicate interactive or not.

>Then, is it clear how the browser "knows" this? If David's statement=20
>below
>about the API is correct, the browser actually cannot "provide this=20
>input"
>as stated in the above quotation. Or I am confused.

Yeah, the browser would not know.  At present, it can only assume it's=20
interactive based on the fact it is an RTP video and audio flow.

>Finally, why is audio not also subdivided into interactive and
>non-interactive? As far as I can see, both are logically possible.

For WebRTC, audio alone is "interactive" in nature (which is why it's=20
marked EF).  However, if one is sending audio and video it makes sense=20
to mark them both same way to get the same PHB and hopefully have them=20
queued in the same buffers along the path.  Sending audio as EF and=20
possibly having a PHB that results in packets arriving much faster than=20
corresponding video packets marked as AF42 is not at all helpful for=20
applications that have to synchronize the audio and video flows.

I can think of examples of potentially non-interactive video (e.g.,=20
whiteboard, app sharing, or presentation video).  Likewise, if audio is=20
sent with a presentation video it would make sense for the audio to be=20
marked similarly to the corresponding video.

Paul


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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           : Application Layer Protocol Negotiation for Web Real-Time Communications (WebRTC) 
        Author          : Martin Thomson
	Filename        : draft-ietf-rtcweb-alpn-04.txt
	Pages           : 7
	Date            : 2016-05-05

Abstract:
   This document specifies two Application Layer Protocol Negotiation
   (ALPN) labels for use with Web Real-Time Communications (WebRTC).
   The "webrtc" label identifies regular WebRTC communications: a DTLS
   session that is used establish keys for Secure Real-time Transport
   Protocol (SRTP) or to establish data channels using SCTP over DTLS.
   The "c-webrtc" label describes the same protocol, but the peers also
   agree to maintain the confidentiality of the media by not sharing it
   with other applications.


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

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

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


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/


From nobody Fri May  6 09:41:27 2016
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A new meeting session request has just been submitted by Ted Hardie, a Chair of the rtcweb working group.


---------------------------------------------------------
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:
  There are two possible BoFs that we need to add as First priority conflicts:  SPUD and QUIC (if approved).

thanks,

Ted
---------------------------------------------------------


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From: "Black, David" <david.black@emc.com>
To: "tsvwg@ietf.org" <tsvwg@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Many thanks for the productive comments - I'm replying to my own message in=
 an attempt to summarize the discussion and what ought to be done to the dr=
afts as a result.

-- [A] Brian Carpenter asked: what is the definition of "interactive"?

Paul Jones answered: from RFC 4594, which comes from G.1010.

IMHO, both drafts should cite RFC 4594 and G.1010 for this discussion and d=
efinition.

-- [B] Brian Carpenter asked about whether the WebRTC API supports a parame=
ter to distinguish interactive video from non-interactive video.

The definitive answer from a number of sources (e.g., Stefan H=E5kansson) i=
s "No."   There is additional text in the tsvwg-rtcweb-qos draft that will =
need to be revised to align with this answer (e.g., for WebRTC, there is no=
 input to the browser to distinguish interactive vs. non-interactive video =
and the browser SHOULD NOT try to do so on its own).

-- [C] Bernard Aboba wondered  whether a "non-interactive" marking/priority=
 would deliver anything useful to application developers independent of whe=
ther or not it's supported in the WebRTC API..

My perception is that the answer here is "Yes" which leads to leaving the N=
on-Interactive Video flow type in the tsvwg-rtcweb-qos draft, but making it=
 clear that use of Web RTC via the browser API currently does not (and cann=
ot) use that flow type.

-- [D] Finally, Brian Carpenter asked about non-interactive audio:

  Finally, why is audio not also subdivided into interactive and
  non-interactive? As far as I can see, both are logically possible.

I believe the answer is a lack of interest combined with absence of support=
 in WebRTC "running code".

Further comments?

Absent further discussion, I'll work offline with the draft authors on text=
 changes for both drafts in accordance with this summary.

Thanks, --David

> -----Original Message-----
> From: Black, David
> Sent: Wednesday, May 04, 2016 11:26 AM
> To: tsvwg@ietf.org; rtcweb@ietf.org
> Cc: Black, David
> Subject: Diffserv QoS for Video
>=20
> Greetings,
>=20
> I write as the shepherd for the WebRTC QoS draft (draft-ietf-tsvwg-rtcweb=
-qos).
> IETF Last Call on this draft uncovered an issue that appears to also requ=
ire
> changes in the Web RTC transports draft (draft-ietf-rtcweb-transports).
>=20
> The Web RTC draft  specifies different Diffserv Codepoints (DSCPs) for us=
e with
> Interactive vs. Non-Interactive Video (with or without audio in both case=
s):
> https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-15#section-5
>=20
> The IETF LC issue is: How does an implementer determine whether video is
> interactive vs. non-interactive?
>=20
> The answer (unexpected to at least me) is that this can't be done for cur=
rent
> WebRTC - all video has to be treated as interactive because there's no AP=
I
> support to distinguish interactive video from non-interactive video.  Tha=
t
> doesn't appear to be stated anywhere obvious, and a TSVWG draft seems
> like the wrong place to do so.
>=20
> Hence, this message contains initial proposed changes to both the WebRTC
> Transports draft and WebRTC QoS draft to resolve this issue without remov=
ing
> any of the DSCPs currently specified for video in the WebRTC QoS draft.
>=20
> We should resolve this within the next few days, as the WebRTC QoS draft =
is
> (finally!) on the IESG telechat agenda for May 19.
>=20
> --- WebRTC Transports draft - proposed changes
>=20
> In Section 4 Media Prioritization, after the following existing text:
>=20
>    For media, a "media flow", which can be an "audio flow" or a "video
>    flow", is what [RFC7656] calls a "media source", which results in a
>    "source RTP stream" and one or more "redundancy RTP streams".  This
>    specification does not describe prioritization between the RTP
>    streams that come from a single "media source".
>=20
> Add a new paragraph:
>=20
>    All media flows in WebRTC are assumed to be interactive; there is
>    no browser API support for non-interactive media, aside from sending
>    non-interactive media content via the data channel.
>=20
> This ought to cite a W3C reference for the WebRTC API - the following
> appears plausible:
>=20
>    [W3C.WD-webrtc-20120209]
>               Bergkvist, A., Burnett, D., Jennings, C., and A.
>               Narayanan, "WebRTC 1.0: Real-time Communication Between
>               Browsers", World Wide Web Consortium WD WD-webrtc-
>               20120209, February 2012,
>               <http://www.w3.org/TR/2012/WD-webrtc-20120209>.
>=20
> -- WebRTC QoS draft - proposed changes
>=20
> In Section 5 DSCP Mappings, after the following existing text:
>=20
>    The browser SHOULD first select the flow type of the flow.  Within
>    the flow type, the relative importance of the flow SHOULD be used to
>    select the appropriate DSCP value.
>=20
> Add a new paragraph:
>=20
>    The current WebRTC browser API does not support non-interactive
>    video; all video is assumed to be interactive [I-D.ietf-rtcweb-transpo=
rts].
>    Browsers MUST NOT use the DSCP mappings for Non-Interactive Video
>    in Table 1.  Non-browser implementations of WebRTC MAY use the
>    DSCP mappings for Non-Interactive Video in Table 1 for video that is
>    known to not be interactive, e.g., all video in a video playback appli=
cation
>    based on a custom implementation of the WebRTC protocols would not
>    be interactive.
>=20
> ------
>=20
> Comments are welcome - as noted, above,  we should resolve this in
> the next few days in order to avoid pulling the WebRTC QoS draft off
> of the May 19 IESG telechat agenda. Please reply to both the TSVWG and
> RTCWEB lists, as we need  a coherent solution across both drafts.
>=20
> I want to thank Colin Perkins for suggesting the video playback example.
>=20
> Thanks, --David (as WebRTC QoS draft shepherd)


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From nobody Mon May  9 12:04:05 2016
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From: "Paul E. Jones" <paulej@packetizer.com>
To: Ruediger.Geib@telekom.de
Date: Mon, 09 May 2016 19:03:10 +0000
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Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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Ruediger,

Perhaps an example might be helpful.  How about I add this text for=20
illustrative purposes?

     To illustrate the use of the above table, let us assume the
     application assigns a priority of "medium" to audio and video
     flows.  Given that assumption, if the application wishes to send
     only audio then packets would be marked EF.  However, if the
     application wishes to send both interactive video and audio,
     then audio and video packets would both be marked as AF42 or
     AF43.  The intent is to ensure that when both audio and video
     are being sent together that they receive similar per-hop
     behavior.

This doesn't get into the preference for AF42 vs. AF43. If it were me,=20
I'd mark all audio as AF42 and only key video frames as AF42.  All=20
predictive frames would be sent with an AF43 marking.  I might even take=
=20
it a step further and classify all audio as "high".  However, I've seen=20
a tremendous amount of debate on this before, so I'd prefer to not go=20
too far in dictating audio markings vs. video.  I do think most people=20
generally agree about at least ensuring the class is the same, otherwise=
=20
the wildly different PHB introduces skew between A/V packet arrival,=20
thus inflating the size of buffers managing the A/V streams.  However,=20
we do not want to dictate that audio should be treated significantly=20
better than audio.  For deaf users, for example, the audio really isn't=20
important at all.  That is perhaps an extreme example, but it=20
nonetheless highlights why we should be cautious about exactly what we=20
normatively mandate.

Paul

------ Original Message ------
From: Ruediger.Geib@telekom.de
To: paulej@packetizer.com
Cc: brian.e.carpenter@gmail.com; david.black@emc.com; tsvwg@ietf.org;=20
rtcweb@ietf.org
Sent: 5/9/2016 3:34:25 AM
Subject: AW: [tsvwg] Diffserv QoS for Video

>Hi Paul,
>
>I've talked with audio/video experts of Deutsche Telekom and they too=20
>favored what you recommend below: transport audio and video by the same=
=20
>queue. Your statement below however stops there and the draft text=20
>doesn't clarify the issue:
>
>If there's interactive video with audio, then they both should be=20
>marked for the same PHB which is:
>- EF ?
>- AF4? Like AF41 Audio, AF42 Video (AF43 in addition, if P or B frames=20
>are to receive a lower priority /
>   higher drop precedence PHB)?
>
>I personally prefer AF4 if audio and video are to be transported in the=
=20
>same queue.
>
>I'd also ask for the draft text to be clear about the issue when to=20
>mark audio by the EF PHB. My understanding after reading your statement=
=20
>below is: Audio marked EF if there's no video flow only.
>
>...
>BC>Finally, why is audio not also subdivided into interactive and
>BC>non-interactive? As far as I can see, both are logically possible.
>
>[PJ] For WebRTC, audio alone is "interactive" in nature (which is
>   why it's marked EF).  However, if one is sending audio and video
>   it makes sense to mark them both same way to get the same PHB and
>   hopefully have them queued in the same buffers along the path.
>   Sending audio as EF and possibly having a PHB that results in
>   packets arriving much faster than corresponding video packets
>   marked as AF42 is not at all helpful for applications that have
>   to synchronize the audio and video flows.
>...
>
>   Paul
>
>
>Regards,
>
>Ruediger
>


From nobody Mon May  9 13:31:37 2016
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I've been looking at how WebRTC Identity and STIR work together and put =
together a worked out example at=20

http://www.ietf.org/id/draft-jennings-stir-rtcweb-identity-00.txt

It does not propose any significant changes to WebRTC but it does point =
at a few syntax changes that might make things easier.=20

Probably the key change for RTCWeb would be to unify the syntax we use =
for DTLS-SRTP fingerprints.=20=


From nobody Mon May  9 23:57:08 2016
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From nobody Tue May 10 00:21:28 2016
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From: Harald Alvestrand <harald@alvestrand.no>
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Den 09. mai 2016 22:31, skrev Cullen Jennings:
> 
> 
> I've been looking at how WebRTC Identity and STIR work together and put together a worked out example at 
> 
> http://www.ietf.org/id/draft-jennings-stir-rtcweb-identity-00.txt
> 
> It does not propose any significant changes to WebRTC but it does point at a few syntax changes that might make things easier. 
> 
> Probably the key change for RTCWeb would be to unify the syntax we use for DTLS-SRTP fingerprints. 


The fingerprint syntax comes from SDP/DTLS, not from RTCWeb, I believe.

It might be best for STIR to either get with the party or propose an SDP
change for DTLS.

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


From nobody Tue May 10 00:35:53 2016
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To: Ruediger.Geib@telekom.de, paulej@packetizer.com
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FTR: I don't see such an agreement at all.

On the contrary, my perception is that people want the ability to
deliver audio with a lower loss probability and lower delay probability
than video - it's more important to the conversation, and there are
fewer things the recipient can do to hide the losses. If the sender
chose to send them on separate flows, they shold have different DSCP
markings.

I believe this is what draft-ietf-tsvwg-rtcweb-qos-15 section 5 states,
and I believe that this is what TSVWG has declared consensus on and
wrote in the document that passed WG last call and is currently in
"waiting for writeup" state.

Changing this determination would, at minimum, require reopening the WG
Last Call.
And I'd object.

Harald


Den 10. mai 2016 08:56, skrev Ruediger.Geib@telekom.de:
> Hi Paul,
> 
> I think we agree, that audio and video frames, if both are part of the same (interactive) media flow should be transported by the same PHB [PJ] or the same queue [RG]. The latter is ensured, if the same PHB is picked for audio and video. To me the text of the draft so far doesn't express that both audio and video are supposed to use an "Interactive Video..." PHB, if both are present. I'd prefer to have text with a non binding standard requirement saying  
> 
>      However, if the application wishes to send both interactive 
>      video and audio, it is RECOMMENDED to transport audio 
>      and video packets by the same per hop behavior. For example, 
>      audio and video packets would both be marked as AF42 or
>      AF43. 
> 
> I don't insist on descriptive text proposing to transport audio by an AF4 PHB offering a lower drop ratio than that used to transport video. My audio/video experts support this and I'm pretty sure, that also Cisco representatives mentioned that audio quality ranks above video quality in telepresence sessions.
> 
> Regards,
> 
> Ruediger
> 
> 
> -----Ursprüngliche Nachricht-----
> Von: Paul E. Jones [mailto:paulej@packetizer.com] 
> Gesendet: Montag, 9. Mai 2016 21:03
> An: Geib, Rüdiger
> Cc: brian.e.carpenter@gmail.com; david.black@emc.com; tsvwg@ietf.org; rtcweb@ietf.org
> Betreff: Re: AW: [tsvwg] Diffserv QoS for Video
> 
> Ruediger,
> 
> Perhaps an example might be helpful.  How about I add this text for illustrative purposes?
> 
>      To illustrate the use of the above table, let us assume the
>      application assigns a priority of "medium" to audio and video
>      flows.  Given that assumption, if the application wishes to send
>      only audio then packets would be marked EF.  However, if the
>      application wishes to send both interactive video and audio,
>      then audio and video packets would both be marked as AF42 or
>      AF43.  The intent is to ensure that when both audio and video
>      are being sent together that they receive similar per-hop
>      behavior.
> 
> This doesn't get into the preference for AF42 vs. AF43. If it were me, I'd mark all audio as AF42 and only key video frames as AF42.  All predictive frames would be sent with an AF43 marking.  I might even take it a step further and classify all audio as "high".  However, I've seen a tremendous amount of debate on this before, so I'd prefer to not go too far in dictating audio markings vs. video.  I do think most people generally agree about at least ensuring the class is the same, otherwise the wildly different PHB introduces skew between A/V packet arrival, thus inflating the size of buffers managing the A/V streams.  However, we do not want to dictate that audio should be treated significantly better than audio.  For deaf users, for example, the audio really isn't important at all.  That is perhaps an extreme example, but it nonetheless highlights why we should be cautious about exactly what we normatively mandate.
> 
> Paul
> 
> ------ Original Message ------
> From: Ruediger.Geib@telekom.de
> To: paulej@packetizer.com
> Cc: brian.e.carpenter@gmail.com; david.black@emc.com; tsvwg@ietf.org; rtcweb@ietf.org
> Sent: 5/9/2016 3:34:25 AM
> Subject: AW: [tsvwg] Diffserv QoS for Video
> 
>> Hi Paul,
>>
>> I've talked with audio/video experts of Deutsche Telekom and they too 
>> favored what you recommend below: transport audio and video by the same 
>> queue. Your statement below however stops there and the draft text 
>> doesn't clarify the issue:
>>
>> If there's interactive video with audio, then they both should be 
>> marked for the same PHB which is:
>> - EF ?
>> - AF4? Like AF41 Audio, AF42 Video (AF43 in addition, if P or B frames 
>> are to receive a lower priority /
>>   higher drop precedence PHB)?
>>
>> I personally prefer AF4 if audio and video are to be transported in the 
>> same queue.
>>
>> I'd also ask for the draft text to be clear about the issue when to 
>> mark audio by the EF PHB. My understanding after reading your statement 
>> below is: Audio marked EF if there's no video flow only.
>>
>> ...
>> BC>Finally, why is audio not also subdivided into interactive and 
>> BC>non-interactive? As far as I can see, both are logically possible.
>>
>> [PJ] For WebRTC, audio alone is "interactive" in nature (which is
>>   why it's marked EF).  However, if one is sending audio and video
>>   it makes sense to mark them both same way to get the same PHB and
>>   hopefully have them queued in the same buffers along the path.
>>   Sending audio as EF and possibly having a PHB that results in
>>   packets arriving much faster than corresponding video packets
>>   marked as AF42 is not at all helpful for applications that have
>>   to synchronize the audio and video flows.
>> ...
>>
>>   Paul
>>
>>
>> Regards,
>>
>> Ruediger
>>
> 


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From: <Ruediger.Geib@telekom.de>
To: <harald@alvestrand.no>
Date: Tue, 10 May 2016 10:18:30 +0200
Thread-Topic: [tsvwg] Diffserv QoS for Video
Thread-Index: AdGqjqcAtUfKmRHmTSOs4NfsI9nIYQAARuzg
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Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: Harald Alvestrand <harald@alvestrand.no>, "Ruediger.Geib@telekom.de" <Ruediger.Geib@telekom.de>, "paulej@packetizer.com" <paulej@packetizer.com>
Thread-Topic: [rtcweb] [tsvwg] Diffserv QoS for Video
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Date: Tue, 10 May 2016 09:20:04 +0000
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Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Harald Alvestrand" <harald@alvestrand.no>, Ruediger.Geib@telekom.de
Date: Wed, 11 May 2016 03:31:34 +0000
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Cc: rtcweb@ietf.org, tsvwg@ietf.org
Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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Yeah, that's a good point.   Let's not go down the path of re-opening a=20
new round of changes that would require yet another LC and more time.

Paul

------ Original Message ------
From: "Harald Alvestrand" <harald@alvestrand.no>
To: Ruediger.Geib@telekom.de; paulej@packetizer.com
Cc: rtcweb@ietf.org; tsvwg@ietf.org
Sent: 5/10/2016 3:35:42 AM
Subject: Re: [tsvwg] Diffserv QoS for Video

>FTR: I don't see such an agreement at all.
>
>On the contrary, my perception is that people want the ability to
>deliver audio with a lower loss probability and lower delay probability
>than video - it's more important to the conversation, and there are
>fewer things the recipient can do to hide the losses. If the sender
>chose to send them on separate flows, they shold have different DSCP
>markings.
>
>I believe this is what draft-ietf-tsvwg-rtcweb-qos-15 section 5 states,
>and I believe that this is what TSVWG has declared consensus on and
>wrote in the document that passed WG last call and is currently in
>"waiting for writeup" state.
>
>Changing this determination would, at minimum, require reopening the WG
>Last Call.
>And I'd object.
>
>Harald
>
>
>Den 10. mai 2016 08:56, skrev Ruediger.Geib@telekom.de:
>>  Hi Paul,
>>
>>  I think we agree, that audio and video frames, if both are part of=20
>>the same (interactive) media flow should be transported by the same=20
>>PHB [PJ] or the same queue [RG]. The latter is ensured, if the same=20
>>PHB is picked for audio and video. To me the text of the draft so far=20
>>doesn't express that both audio and video are supposed to use an=20
>>"Interactive Video..." PHB, if both are present. I'd prefer to have=20
>>text with a non binding standard requirement saying
>>
>>       However, if the application wishes to send both interactive
>>       video and audio, it is RECOMMENDED to transport audio
>>       and video packets by the same per hop behavior. For example,
>>       audio and video packets would both be marked as AF42 or
>>       AF43.
>>
>>  I don't insist on descriptive text proposing to transport audio by an=
=20
>>AF4 PHB offering a lower drop ratio than that used to transport video.=
=20
>>My audio/video experts support this and I'm pretty sure, that also=20
>>Cisco representatives mentioned that audio quality ranks above video=20
>>quality in telepresence sessions.
>>
>>  Regards,
>>
>>  Ruediger
>>
>>
>>  -----Urspr=C3=BCngliche Nachricht-----
>>  Von: Paul E. Jones [mailto:paulej@packetizer.com]
>>  Gesendet: Montag, 9. Mai 2016 21:03
>>  An: Geib, R=C3=BCdiger
>>  Cc: brian.e.carpenter@gmail.com; david.black@emc.com; tsvwg@ietf.org;=
=20
>>rtcweb@ietf.org
>>  Betreff: Re: AW: [tsvwg] Diffserv QoS for Video
>>
>>  Ruediger,
>>
>>  Perhaps an example might be helpful.  How about I add this text for=20
>>illustrative purposes?
>>
>>       To illustrate the use of the above table, let us assume the
>>       application assigns a priority of "medium" to audio and video
>>       flows.  Given that assumption, if the application wishes to send
>>       only audio then packets would be marked EF.  However, if the
>>       application wishes to send both interactive video and audio,
>>       then audio and video packets would both be marked as AF42 or
>>       AF43.  The intent is to ensure that when both audio and video
>>       are being sent together that they receive similar per-hop
>>       behavior.
>>
>>  This doesn't get into the preference for AF42 vs. AF43. If it were=20
>>me, I'd mark all audio as AF42 and only key video frames as AF42.  All=
=20
>>predictive frames would be sent with an AF43 marking.  I might even=20
>>take it a step further and classify all audio as "high".  However,=20
>>I've seen a tremendous amount of debate on this before, so I'd prefer=20
>>to not go too far in dictating audio markings vs. video.  I do think=20
>>most people generally agree about at least ensuring the class is the=20
>>same, otherwise the wildly different PHB introduces skew between A/V=20
>>packet arrival, thus inflating the size of buffers managing the A/V=20
>>streams.  However, we do not want to dictate that audio should be=20
>>treated significantly better than audio.  For deaf users, for example,=
=20
>>the audio really isn't important at all.  That is perhaps an extreme=20
>>example, but it nonetheless highlights why we should be cautious about=
=20
>>exactly what we normatively mandate.
>>
>>  Paul
>>
>>  ------ Original Message ------
>>  From: Ruediger.Geib@telekom.de
>>  To: paulej@packetizer.com
>>  Cc: brian.e.carpenter@gmail.com; david.black@emc.com; tsvwg@ietf.org;=
=20
>>rtcweb@ietf.org
>>  Sent: 5/9/2016 3:34:25 AM
>>  Subject: AW: [tsvwg] Diffserv QoS for Video
>>
>>>  Hi Paul,
>>>
>>>  I've talked with audio/video experts of Deutsche Telekom and they=20
>>>too
>>>  favored what you recommend below: transport audio and video by the=20
>>>same
>>>  queue. Your statement below however stops there and the draft text
>>>  doesn't clarify the issue:
>>>
>>>  If there's interactive video with audio, then they both should be
>>>  marked for the same PHB which is:
>>>  - EF ?
>>>  - AF4? Like AF41 Audio, AF42 Video (AF43 in addition, if P or B=20
>>>frames
>>>  are to receive a lower priority /
>>>    higher drop precedence PHB)?
>>>
>>>  I personally prefer AF4 if audio and video are to be transported in=
=20
>>>the
>>>  same queue.
>>>
>>>  I'd also ask for the draft text to be clear about the issue when to
>>>  mark audio by the EF PHB. My understanding after reading your=20
>>>statement
>>>  below is: Audio marked EF if there's no video flow only.
>>>
>>>  ...
>>>  BC>Finally, why is audio not also subdivided into interactive and
>>>  BC>non-interactive? As far as I can see, both are logically=20
>>>possible.
>>>
>>>  [PJ] For WebRTC, audio alone is "interactive" in nature (which is
>>>    why it's marked EF).  However, if one is sending audio and video
>>>    it makes sense to mark them both same way to get the same PHB and
>>>    hopefully have them queued in the same buffers along the path.
>>>    Sending audio as EF and possibly having a PHB that results in
>>>    packets arriving much faster than corresponding video packets
>>>    marked as AF42 is not at all helpful for applications that have
>>>    to synchronize the audio and video flows.
>>>  ...
>>>
>>>    Paul
>>>
>>>
>>>  Regards,
>>>
>>>  Ruediger
>>>
>>


From nobody Wed May 11 08:55:38 2016
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From: "Hutton, Andrew" <andrew.hutton@unify.com>
To: Philipp Hancke <fippo@goodadvice.pages.de>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] which bandwidth modifiers need to be supported?
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I was also looking at this again my assumption is that TIAS is what a WebRT=
C endpoint should generate TIAS as Philipp stated this is hinted at in the =
W3C spec but probably also should be specified in JSEP and an example in ht=
tps://tools.ietf.org/html/draft-ietf-rtcweb-sdp-01#section-5.2.2 would also=
 be useful.

For parsing then both TIAS and AS needs to be supported this is specified i=
n https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-14#section-5.8.=20

Regards
Andy




> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Philipp
> Hancke
> Sent: 25 April 2016 10:24
> To: rtcweb@ietf.org
> Subject: [rtcweb] which bandwidth modifiers need to be supported?
>=20
> which bandwidth modifiers for b=3D lines does a client need to support?
>=20
> Chrome currently supports b=3DAS, however the implementation is closer to
> TIAS semantically. Firefox does not implement either currently.
>=20
> http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-
> maxBitrate
> uses TIAS as a definition.
>=20
> For parsing, supporting both is fine (for a while at least).
> For serialization I would prefer picking one (TIAS) to having to track
> which variant a peer supports.
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Wed May 11 09:40:30 2016
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From: <amina.boubendir@orange.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] On ICE improvement / Switching Between Pairs Based on RTT
Thread-Index: AdGrnkrp7cI9TydpTh23sxm4WUHMFw==
Date: Wed, 11 May 2016 16:40:25 +0000
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Dear all,=20

I am studying delay reduction between WebRTC pairs and TURN servers.=20
This draft https://tools.ietf.org/html/draft-uberti-mmusic-nombis-00#page-9=
  on ICE improvement mentions :=20
Switching Between Pairs Based on RTT and Switching To A New TURN Server as =
examples.=20
Does anyone have more information about any concrete measurements about the=
se aspects?=20

Thank you=20

Regards

Amina Boubendir

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

___________________________________________________________________________=
______________________________________________

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From nobody Wed May 11 13:54:28 2016
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] RTCWeb and STIR
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Date: Wed, 11 May 2016 20:54:15 +0000
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Subject: Re: [rtcweb] RTCWeb and STIR
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STIR is planning on moving the fingerprint syntax to a JSON array anyway.
I think we could easily get the PASSporT object aligned with the syntax in
rtcweb-security-arch 5.6.4. So I think the fixes in 3.2 can probably be
made - though I agree (per 3.7) we should try to make the encoding of the
digest as compact as we can across the two mechanisms, which might warrant
some more discussion.

The really interesting interworking question, as Cullen's draft suggests,
is how to prevent the "grossness" that results from the fact that STIR
today carries its identity assertion in a SIP header, whereas WebRTC
carries its assertion as an SDP attribute. The last thing we'd want is to
do both redunantly. It does look like, provided we can get fingerprints
into alignment, it would at least be possible to translate the SDP
attribute into a SIP header and vice versa.

A number of the smaller fixes that Cullen proposes for WebRTC seem like
they could be helpful: as some are just adding extensions to the existing
a=3Didentity attribute, which the rtcweb-security-arch spec allows, so much
of that is painless. Support for multiple identity assertions in one SDP
session seems like it would be valuable for non-STIR IdP use cases even.
Questions like Cullen's 3.3.3, whether or not it would be better to break
the domain and protocol out of the base64 encoded assertion, are things
that would be nice to have, though I imagine people are reluctant to
reopen rtcweb-security-arch at this point.

Most of the questions about the identifiers carried in the assertion are
cases where, at least on the PASSporT side, extensibility should be able
to take care of them. The 3.6 question of whether identity should focus on
RFC822-style names versus URIs is an interesting one: I'd venture it would
be perfectly valid to define a PASSporT claim type for those, in addition
to having URIs and telephone numbers as options (also, there's already a
baseline JWS "email" claim that might serve). Per 3.9, I've been imagining
a PASSporT extension specifically for conferencing architectures -
possibly more than one will be required, to support centralized vs. mesh.

This is a very handy document to have though, I think I understand the
interworking story a lot better than I did before.

Jon Peterson
Neustar, Inc.

On 5/10/16, 12:21 AM, "rtcweb on behalf of Harald Alvestrand"
<rtcweb-bounces@ietf.org on behalf of harald@alvestrand.no> wrote:

>Den 09. mai 2016 22:31, skrev Cullen Jennings:
>>=20
>>=20
>> I've been looking at how WebRTC Identity and STIR work together and put
>>together a worked out example at
>>=20
>> http://www.ietf.org/id/draft-jennings-stir-rtcweb-identity-00.txt
>>=20
>> It does not propose any significant changes to WebRTC but it does point
>>at a few syntax changes that might make things easier.
>>=20
>> Probably the key change for RTCWeb would be to unify the syntax we use
>>for DTLS-SRTP fingerprints.
>
>
>The fingerprint syntax comes from SDP/DTLS, not from RTCWeb, I believe.
>
>It might be best for STIR to either get with the party or propose an SDP
>change for DTLS.
>
>> _______________________________________________
>> 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: "Black, David" <david.black@emc.com>
To: "Ruediger.Geib@telekom.de" <Ruediger.Geib@telekom.de>, "harald@alvestrand.no" <harald@alvestrand.no>
Thread-Topic: [tsvwg] Diffserv QoS for Video
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From nobody Wed May 11 18:43:56 2016
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In-Reply-To: <D358DF9E.18BA35%jon.peterson@neustar.biz>
References: <1D53717F-3679-4E9C-B612-FA75BFF13032@iii.ca> <57318BF1.4050006@alvestrand.no> <D358DF9E.18BA35%jon.peterson@neustar.biz>
Date: Thu, 12 May 2016 11:43:43 +1000
Message-ID: <CABkgnnWeuXPm05nvaNe6ctstEHQ9a7metZJGyTC1=yVFu73cyw@mail.gmail.com>
From: Martin Thomson <martin.thomson@gmail.com>
To: "Peterson, Jon" <jon.peterson@neustar.biz>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "Wendt, Chris" <Chris_Wendt@cable.comcast.com>
Subject: Re: [rtcweb] RTCWeb and STIR
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On 12 May 2016 at 06:54, Peterson, Jon <jon.peterson@neustar.biz> wrote:
> This is a very handy document to have though, I think I understand the
> interworking story a lot better than I did before.

wot jon sed

These are likely to be breaking changes, but I think that there are
changes in this direction that would improve the functioning of the
WebRTC identity stuff.  Some potentially a lot. Getting good STIR
PASSporT integration might be a big deal.


From nobody Thu May 12 09:48:26 2016
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Subject: [rtcweb] Protocol Action: 'Application Layer Protocol Negotiation for Web Real-Time Communications (WebRTC)' to Proposed Standard (draft-ietf-rtcweb-alpn-04.txt)
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The IESG has approved the following document:
- 'Application Layer Protocol Negotiation for Web Real-Time
   Communications (WebRTC)'
  (draft-ietf-rtcweb-alpn-04.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-alpn/




This is pretty simple/short draft that defines/registers two Application 
Layer Protocol Negotiation (ALPN) [RFC7301] values to enable an endpoint 
to positively identify WebRTC uses and distinguish them from other DTLS 
uses (i.e., provide media isolation).  The two values are webrtc and c-
webrtc; the first identifies a session that uses a combination of WebRTC 
compatible media and data, and the second identifies a session requiring 
confidentiality protection.

As far as where you should point your fingers:
- Sean Turner is the document shepherd, and;
- Alissa Cooper is the responsible Area Director.


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To: "Black, David" <david.black@emc.com>, "Ruediger.Geib@telekom.de" <Ruediger.Geib@telekom.de>, "harald@alvestrand.no" <harald@alvestrand.no>
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From: Brian E Carpenter <brian.e.carpenter@gmail.com>
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Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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> Also, in addition to the -16 version of the WebRTC QoS draft that was j=
ust uploaded,

fwiw, as one of the people who's been commenting, I am happy with the cha=
nges
that have been made in that version.

Regards
   Brian

On 12/05/2016 12:19, Black, David wrote:
>> I'm working on transport only and my rtcweb input is draft-ietf-tsvwg-=
rtcweb-
>> qos-15. It states little for audience like me, if the subject is audio=
 and video
>> related to a talking person. I think the relevant statement are lines =
in a table. I'd
>> appreciate text helping with the interpretation of table. If we can't =
agree on
>> useful text - leave it as is.
>=20
> As draft shepherd, I think the table text is ok as-is; other WebRTC spe=
cs own
> the definition of what a media flow is, and one of them would be the ap=
propriate
> location for text (if appropriate) to point out that audio for video co=
uld be sent
> on the same or different or different media flow as the video.   Among =
the reasons
> for this is that deciding whether to use one or two media flows has RTP=
 impacts
> that occur well before DSCP assignment to outbound traffic.
>=20
> Also, in addition to the -16 version of the WebRTC QoS draft that was j=
ust uploaded,
> I've edited that draft's shepherd writeup to reflect the resolution of =
the issue around
> interactive vs. non-interactive video, including recording the expectat=
ion that
> there will be some additional text on this topic in the next version of=
 the WebRTC
> Transports  draft.
>=20
> Thanks, --David
>=20
>> -----Original Message-----
>> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of
>> Ruediger.Geib@telekom.de
>> Sent: Tuesday, May 10, 2016 4:19 AM
>> To: harald@alvestrand.no
>> Cc: rtcweb@ietf.org; tsvwg@ietf.org
>> Subject: Re: [tsvwg] Diffserv QoS for Video
>>
>> Hi Harald,
>>
>> We should avoid to re-open last call. That's not my intent.
>>
>> I'm working on transport only and my rtcweb input is draft-ietf-tsvwg-=
rtcweb-
>> qos-15. It states little for audience like me, if the subject is audio=
 and video
>> related to a talking person. I think the relevant statement are lines =
in a table. I'd
>> appreciate text helping with the interpretation of table. If we can't =
agree on
>> useful text - leave it as is.
>>
>> A backbone supporting RTP based TV distribution tends to be engineered=
 for
>> support of an AF4 like QoS class for low loss. I think, telephony code=
cs, whose
>> RTP streams will be transported by EF, are able to deal with higher pa=
cket loss,
>> than multicast based IPTV.
>>
>> To me, an audio or telepresence/videoconference call with QoS require
>> admission control. This ensures that they are transported without queu=
ing or
>> drops.
>>
>> If there's congestion however:
>>
>> An audio flow marked AF41 should face a lower drop rate than a video f=
low
>> marked AF42 in the case of congestion at an network edge node. The pac=
kets
>> arrive in the same order as they were sent (independent of the flow th=
ey belong
>> to).
>>
>> An EF marked audio flow may experience loss events independent from a =
video
>> flow marked AF4x. In the case of queuing, one of the flows will be ear=
lier. If both
>> are transported by QoS classes optimized for rtp/udp transport, the di=
fference is
>> a few [ms] per congestion point. If one of the queues is optimized for=
 general
>> data transport, the delay difference is likely to be a double digit [m=
s] number.
>> The packets of each flow arrive in order as sent, the packets of one f=
low are
>> delayed against those of the other.
>>
>> Regards,
>>
>> Ruediger
>>
>> -----Urspr=C3=BCngliche Nachricht-----
>> Von: Harald Alvestrand [mailto:harald@alvestrand.no]
>> Gesendet: Dienstag, 10. Mai 2016 09:36
>> An: Geib, R=C3=BCdiger; paulej@packetizer.com
>> Cc: rtcweb@ietf.org; tsvwg@ietf.org
>> Betreff: Re: [tsvwg] Diffserv QoS for Video
>>
>> FTR: I don't see such an agreement at all.
>>
>> On the contrary, my perception is that people want the ability to deli=
ver audio
>> with a lower loss probability and lower delay probability than video -=
 it's more
>> important to the conversation, and there are fewer things the recipien=
t can do to
>> hide the losses. If the sender chose to send them on separate flows, t=
hey shold
>> have different DSCP markings.
>>
>> I believe this is what draft-ietf-tsvwg-rtcweb-qos-15 section 5 states=
, and I
>> believe that this is what TSVWG has declared consensus on and wrote in=
 the
>> document that passed WG last call and is currently in "waiting for wri=
teup" state.
>>
>> Changing this determination would, at minimum, require reopening the W=
G Last
>> Call.
>> And I'd object.
>>
>> Harald
>>
>>
>> Den 10. mai 2016 08:56, skrev Ruediger.Geib@telekom.de:
>>> Hi Paul,
>>>
>>> I think we agree, that audio and video frames, if both are part of th=
e
>>> same (interactive) media flow should be transported by the same PHB
>>> [PJ] or the same queue [RG]. The latter is ensured, if the same PHB i=
s
>>> picked for audio and video. To me the text of the draft so far doesn'=
t
>>> express that both audio and video are supposed to use an "Interactive=

>>> Video..." PHB, if both are present. I'd prefer to have text with a no=
n
>>> binding standard requirement saying
>>>
>>>      However, if the application wishes to send both interactive
>>>      video and audio, it is RECOMMENDED to transport audio
>>>      and video packets by the same per hop behavior. For example,
>>>      audio and video packets would both be marked as AF42 or
>>>      AF43.
>>>
>>> I don't insist on descriptive text proposing to transport audio by an=
 AF4 PHB
>> offering a lower drop ratio than that used to transport video. My audi=
o/video
>> experts support this and I'm pretty sure, that also Cisco representati=
ves
>> mentioned that audio quality ranks above video quality in telepresence=
 sessions.
>>>
>>> Regards,
>>>
>>> Ruediger
>>>
>>>
>>> -----Urspr=C3=BCngliche Nachricht-----
>>> Von: Paul E. Jones [mailto:paulej@packetizer.com]
>>> Gesendet: Montag, 9. Mai 2016 21:03
>>> An: Geib, R=C3=BCdiger
>>> Cc: brian.e.carpenter@gmail.com; david.black@emc.com; tsvwg@ietf.org;=

>>> rtcweb@ietf.org
>>> Betreff: Re: AW: [tsvwg] Diffserv QoS for Video
>>>
>>> Ruediger,
>>>
>>> Perhaps an example might be helpful.  How about I add this text for i=
llustrative
>> purposes?
>>>
>>>      To illustrate the use of the above table, let us assume the
>>>      application assigns a priority of "medium" to audio and video
>>>      flows.  Given that assumption, if the application wishes to send=

>>>      only audio then packets would be marked EF.  However, if the
>>>      application wishes to send both interactive video and audio,
>>>      then audio and video packets would both be marked as AF42 or
>>>      AF43.  The intent is to ensure that when both audio and video
>>>      are being sent together that they receive similar per-hop
>>>      behavior.
>>>
>>> This doesn't get into the preference for AF42 vs. AF43. If it were me=
, I'd mark all
>> audio as AF42 and only key video frames as AF42.  All predictive frame=
s would be
>> sent with an AF43 marking.  I might even take it a step further and cl=
assify all
>> audio as "high".  However, I've seen a tremendous amount of debate on =
this
>> before, so I'd prefer to not go too far in dictating audio markings vs=
=2E video.  I do
>> think most people generally agree about at least ensuring the class is=
 the same,
>> otherwise the wildly different PHB introduces skew between A/V packet =
arrival,
>> thus inflating the size of buffers managing the A/V streams.  However,=
 we do not
>> want to dictate that audio should be treated significantly better than=
 audio.  For
>> deaf users, for example, the audio really isn't important at all.  Tha=
t is perhaps an
>> extreme example, but it nonetheless highlights why we should be cautio=
us about
>> exactly what we normatively mandate.
>>>
>>> Paul
>>>
>>> ------ Original Message ------
>>> From: Ruediger.Geib@telekom.de
>>> To: paulej@packetizer.com
>>> Cc: brian.e.carpenter@gmail.com; david.black@emc.com; tsvwg@ietf.org;=

>>> rtcweb@ietf.org
>>> Sent: 5/9/2016 3:34:25 AM
>>> Subject: AW: [tsvwg] Diffserv QoS for Video
>>>
>>>> Hi Paul,
>>>>
>>>> I've talked with audio/video experts of Deutsche Telekom and they to=
o
>>>> favored what you recommend below: transport audio and video by the
>>>> same queue. Your statement below however stops there and the draft
>>>> text doesn't clarify the issue:
>>>>
>>>> If there's interactive video with audio, then they both should be
>>>> marked for the same PHB which is:
>>>> - EF ?
>>>> - AF4? Like AF41 Audio, AF42 Video (AF43 in addition, if P or B
>>>> frames are to receive a lower priority /
>>>>   higher drop precedence PHB)?
>>>>
>>>> I personally prefer AF4 if audio and video are to be transported in
>>>> the same queue.
>>>>
>>>> I'd also ask for the draft text to be clear about the issue when to
>>>> mark audio by the EF PHB. My understanding after reading your
>>>> statement below is: Audio marked EF if there's no video flow only.
>>>>
>>>> ...
>>>> BC>Finally, why is audio not also subdivided into interactive and
>>>> BC>non-interactive? As far as I can see, both are logically possible=
=2E
>>>>
>>>> [PJ] For WebRTC, audio alone is "interactive" in nature (which is
>>>>   why it's marked EF).  However, if one is sending audio and video
>>>>   it makes sense to mark them both same way to get the same PHB and
>>>>   hopefully have them queued in the same buffers along the path.
>>>>   Sending audio as EF and possibly having a PHB that results in
>>>>   packets arriving much faster than corresponding video packets
>>>>   marked as AF42 is not at all helpful for applications that have
>>>>   to synchronize the audio and video flows.
>>>> ...
>>>>
>>>>   Paul
>>>>
>>>>
>>>> Regards,
>>>>
>>>> Ruediger
>>>>
>>>


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Cc: rtcweb@ietf.org, tsvwg@ietf.org, brian.e.carpenter@gmail.com
Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Ruediger.Geib@telekom.de" <Ruediger.Geib@telekom.de>
Thread-Topic: [tsvwg] Diffserv QoS for Video
Thread-Index: AQHRrS2dZ/47y6ykhEqiDL26Kx1AUA==
Date: Fri, 13 May 2016 15:39:19 +0000
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References: <828773FE19B05B4581311493A046E85E6ECC3084D0@HE111642.emea1.cds.t-internal.com> <em88678e54-c513-4d74-8bbd-ba0785d70b36@sydney> <828773FE19B05B4581311493A046E85E6ECC4090FD@HE111642.emea1.cds.t-internal.com>
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Subject: Re: [rtcweb] [tsvwg] Diffserv QoS for Video
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> On May 10, 2016, at 12:56 AM, Ruediger.Geib@telekom.de wrote:
>=20
> I think we agree, that audio and video frames, if both are part of the sa=
me (interactive) media flow should be transported by the same PHB [PJ] or t=
he same queue [RG].=20

I don't think we have consensus on that in the ART area but my understandin=
g of this spec was that it allowed people that wanted to do that to give th=
em the same and it allowed people that wanted to implement the different qu=
eues to do that.=20



From nobody Wed May 18 12:47:34 2016
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From: "Black, David" <david.black@emc.com>
To: Suresh Krishnan <suresh.krishnan@ericsson.com>, The IESG <iesg@ietf.org>
Thread-Topic: Suresh Krishnan's No Objection on draft-ietf-tsvwg-rtcweb-qos-16: (with COMMENT)
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Subject: Re: [rtcweb] Suresh Krishnan's No Objection on draft-ietf-tsvwg-rtcweb-qos-16: (with COMMENT)
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From nobody Mon May 23 09:58:07 2016
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To: "Hutton, Andrew" <andrew.hutton@unify.com>
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Subject: Re: [rtcweb] which bandwidth modifiers need to be supported?
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I=E2=80=99m not sure there is a requirement for a WebRTC endpoint to =
generate them but if there is, agree that we should say TIAS is =
preferred.=20

> On May 11, 2016, at 8:55 AM, Hutton, Andrew <andrew.hutton@unify.com> =
wrote:
>=20
> I was also looking at this again my assumption is that TIAS is what a =
WebRTC endpoint should generate TIAS as Philipp stated this is hinted at =
in the W3C spec but probably also should be specified in JSEP and an =
example in =
https://tools.ietf.org/html/draft-ietf-rtcweb-sdp-01#section-5.2.2 would =
also be useful.
>=20
> For parsing then both TIAS and AS needs to be supported this is =
specified in =
https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-14#section-5.8.=20
>=20
> Regards
> Andy
>=20
>=20
>=20
>=20
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Philipp
>> Hancke
>> Sent: 25 April 2016 10:24
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] which bandwidth modifiers need to be supported?
>>=20
>> which bandwidth modifiers for b=3D lines does a client need to =
support?
>>=20
>> Chrome currently supports b=3DAS, however the implementation is =
closer to
>> TIAS semantically. Firefox does not implement either currently.
>>=20
>> http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-
>> maxBitrate
>> uses TIAS as a definition.
>>=20
>> For parsing, supporting both is fine (for a while at least).
>> For serialization I would prefer picking one (TIAS) to having to =
track
>> which variant a peer supports.
>>=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


From nobody Wed May 25 03:04:49 2016
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Security architecture: Making ECDSA mandatory
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This is a multi-part message in MIME format.
--------------040204050708010808010908
Content-Type: text/plain; charset=windows-1252
Content-Transfer-Encoding: 7bit

In my search for status on ECDSA (we're in the process of switching the
Chrome default), I came across this in the current draft:

   All implementations MUST implement DTLS 1.0, with the cipher suite
   TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA and the DTLS-SRTP protection
   profile SRTP_AES128_CM_HMAC_SHA1_80.  Implementations SHOULD
   implement DTLS 1.2 with the TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
   cipher suite.  Implementations SHOULD favor cipher suites which
   support PFS over non-PFS cipher suites and GCM over CBC cipher
   suites.  [[OPEN ISSUE: Should we require ECDSA?  Waiting for WG
   Consensus.]]

I also found Martin's PR. It's 11 months old, still open.

Can we merge this now?


On 06/13/2015 12:06 AM, Martin Thomson wrote:
> I've opened https://github.com/rtcweb-wg/security-arch/pull/33
>
> This changes the MTI cipher suites to ECDSA and does a little cleanup
> on the corresponding API requirements to more closely match what has
> just landed in the W3C specification.
>
> We discussed ECDSA and the only concerns raised were with
> compatibility.  I've done some testing with other implementations with
> no issues, and ECDSA seems to be well supported on all those
> hard-to-upgrade PSTN gateways (thanks to Cullen and Ethan for helping
> out with checks there and to NIST for creating certification pressure
> with FIPS-2).
>
> I have an implementation that switches Firefox to ECDSA with P-256 by
> default.  It's much, much faster.  http://bench.cr.yp.to/ claims that
> it's 150 times faster on mobile devices for keygen.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--------------040204050708010808010908
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<html>
  <head>
    <meta content="text/html; charset=windows-1252"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">In my search for status on ECDSA (we're
      in the process of switching the Chrome default), I came across
      this in the current draft:<br>
      <br>
      <meta http-equiv="content-type" content="text/html;
        charset=windows-1252">
      <pre class="newpage" style="font-size: 13.3333px; margin-top: 0px; margin-bottom: 0px; break-before: page; color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; widows: 1; word-spacing: 0px; -webkit-text-stroke-width: 0px;">   All implementations MUST implement DTLS 1.0, with the cipher suite
   TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA and the DTLS-SRTP protection
   profile SRTP_AES128_CM_HMAC_SHA1_80.  Implementations SHOULD
   implement DTLS 1.2 with the TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
   cipher suite.  Implementations SHOULD favor cipher suites which
   support PFS over non-PFS cipher suites and GCM over CBC cipher
   suites.  [[OPEN ISSUE: Should we require ECDSA?  Waiting for WG
   Consensus.]]

</pre>
      I also found Martin's PR. It's 11 months old, still open.<br>
      <br>
      Can we merge this now?<br class="Apple-interchange-newline">
      <br>
      <br>
      On 06/13/2015 12:06 AM, Martin Thomson wrote:<br>
    </div>
    <blockquote
cite="mid:CABkgnnWjaBqVdNurt+sd3w9U_rpTi0WJKFce12KfA2W1mrnsTA@mail.gmail.com"
      type="cite">
      <pre wrap="">I've opened <a class="moz-txt-link-freetext" href="https://github.com/rtcweb-wg/security-arch/pull/33">https://github.com/rtcweb-wg/security-arch/pull/33</a>

This changes the MTI cipher suites to ECDSA and does a little cleanup
on the corresponding API requirements to more closely match what has
just landed in the W3C specification.

We discussed ECDSA and the only concerns raised were with
compatibility.  I've done some testing with other implementations with
no issues, and ECDSA seems to be well supported on all those
hard-to-upgrade PSTN gateways (thanks to Cullen and Ethan for helping
out with checks there and to NIST for creating certification pressure
with FIPS-2).

I have an implementation that switches Firefox to ECDSA with P-256 by
default.  It's much, much faster.  <a class="moz-txt-link-freetext" href="http://bench.cr.yp.to/">http://bench.cr.yp.to/</a> claims that
it's 150 times faster on mobile devices for keygen.

_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------040204050708010808010908--


From nobody Wed May 25 03:17:10 2016
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Subject: Re: [rtcweb] Security architecture: Making ECDSA mandatory
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On 13/06/15 00:06, Martin Thomson wrote:
> I've opened https://github.com/rtcweb-wg/security-arch/pull/33
>
> This changes the MTI cipher suites to ECDSA and does a little cleanup
> on the corresponding API requirements to more closely match what has
> just landed in the W3C specification.
>
> We discussed ECDSA and the only concerns raised were with
> compatibility.  I've done some testing with other implementations with
> no issues, and ECDSA seems to be well supported on all those
> hard-to-upgrade PSTN gateways (thanks to Cullen and Ethan for helping
> out with checks there and to NIST for creating certification pressure
> with FIPS-2).
>
> I have an implementation that switches Firefox to ECDSA with P-256 by
> default.  It's much, much faster.  http://bench.cr.yp.to/ claims that
> it's 150 times faster on mobile devices for keygen.
>

I can confirm this. Here at Wire.com we are using ECDSA certs for the
DTLS connections. Generating self-signed certs takes around 1 millisecond.



/alfred


From nobody Wed May 25 08:12:59 2016
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From: Michael Davey <md84419@gmail.com>
Date: Wed, 25 May 2016 16:10:53 +0100
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Subject: Re: [rtcweb] Security architecture: Making ECDSA mandatory
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--001a114fb5a28dafb10533ac158c
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I would recommend referencing IETF BCP 195.  The comments about ECDHE in
that document (and of course the wider issues with weak DH key exchange)
may also be noteworthy.

-- 
Michael


On 25 May 2016 at 11:17, Alfred E. Heggestad <aeh@db.org> wrote:

>
>
> On 13/06/15 00:06, Martin Thomson wrote:
>
>> I've opened https://github.com/rtcweb-wg/security-arch/pull/33
>>
>> This changes the MTI cipher suites to ECDSA and does a little cleanup
>> on the corresponding API requirements to more closely match what has
>> just landed in the W3C specification.
>>
>> We discussed ECDSA and the only concerns raised were with
>> compatibility.  I've done some testing with other implementations with
>> no issues, and ECDSA seems to be well supported on all those
>> hard-to-upgrade PSTN gateways (thanks to Cullen and Ethan for helping
>> out with checks there and to NIST for creating certification pressure
>> with FIPS-2).
>>
>> I have an implementation that switches Firefox to ECDSA with P-256 by
>> default.  It's much, much faster.  http://bench.cr.yp.to/ claims that
>> it's 150 times faster on mobile devices for keygen.
>>
>>
> I can confirm this. Here at Wire.com we are using ECDSA certs for the
> DTLS connections. Generating self-signed certs takes around 1 millisecond.
>
>
>
> /alfred
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a114fb5a28dafb10533ac158c
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<div dir=3D"ltr"><div>I would recommend referencing IETF BCP 195.=C2=A0 The=
 comments about ECDHE in that document (and of course the wider issues with=
 weak DH key exchange) may also be noteworthy.<br><br>-- <br></div>Michael<=
br><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On 25 May =
2016 at 11:17, Alfred E. Heggestad <span dir=3D"ltr">&lt;<a href=3D"mailto:=
aeh@db.org" target=3D"_blank">aeh@db.org</a>&gt;</span> wrote:<br><blockquo=
te class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc so=
lid;padding-left:1ex"><span class=3D""><br>
<br>
On 13/06/15 00:06, Martin Thomson wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I&#39;ve opened <a href=3D"https://github.com/rtcweb-wg/security-arch/pull/=
33" rel=3D"noreferrer" target=3D"_blank">https://github.com/rtcweb-wg/secur=
ity-arch/pull/33</a><br>
<br>
This changes the MTI cipher suites to ECDSA and does a little cleanup<br>
on the corresponding API requirements to more closely match what has<br>
just landed in the W3C specification.<br>
<br>
We discussed ECDSA and the only concerns raised were with<br>
compatibility.=C2=A0 I&#39;ve done some testing with other implementations =
with<br>
no issues, and ECDSA seems to be well supported on all those<br>
hard-to-upgrade PSTN gateways (thanks to Cullen and Ethan for helping<br>
out with checks there and to NIST for creating certification pressure<br>
with FIPS-2).<br>
<br>
I have an implementation that switches Firefox to ECDSA with P-256 by<br>
default.=C2=A0 It&#39;s much, much faster.=C2=A0 <a href=3D"http://bench.cr=
.yp.to/" rel=3D"noreferrer" target=3D"_blank">http://bench.cr.yp.to/</a> cl=
aims that<br>
it&#39;s 150 times faster on mobile devices for keygen.<br>
<br>
</blockquote>
<br></span>
I can confirm this. Here at Wire.com we are using ECDSA certs for the<br>
DTLS connections. Generating self-signed certs takes around 1 millisecond.<=
span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br>
<br>
<br>
/alfred</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
_______________________________________________<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/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--001a114fb5a28dafb10533ac158c--


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A new Request for Comments is now available in online RFC libraries.

        
        RFC 7874

        Title:      WebRTC Audio Codec and Processing 
                    Requirements 
        Author:     JM. Valin, C. Bran
        Status:     Standards Track
        Stream:     IETF
        Date:       May 2016
        Mailbox:    jmvalin@jmvalin.ca, 
                    cary.bran@plantronics.com
        Pages:      7
        Characters: 16160
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-rtcweb-audio-11.txt

        URL:        https://www.rfc-editor.org/info/rfc7874

        DOI:        http://dx.doi.org/10.17487/RFC7874

This document outlines the audio codec and processing requirements
for WebRTC endpoints.

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

This is now a Proposed Standard.

STANDARDS TRACK: This document specifies an Internet Standards Track
protocol for the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the Official
Internet Protocol Standards (https://www.rfc-editor.org/standards) for the 
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 7875

        Title:      Additional WebRTC Audio Codecs for 
                    Interoperability 
        Author:     S. Proust, Ed.
        Status:     Informational
        Stream:     IETF
        Date:       May 2016
        Mailbox:    stephane.proust@orange.com
        Pages:      12
        Characters: 26626
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-rtcweb-audio-codecs-for-interop-06.txt

        URL:        https://www.rfc-editor.org/info/rfc7875

        DOI:        http://dx.doi.org/10.17487/RFC7875

To ensure a baseline of interoperability between WebRTC endpoints, a
minimum set of required codecs is specified.  However, to maximize
the possibility of establishing the session without the need for
audio transcoding, it is also recommended to include in the offer
other suitable audio codecs that are available to the browser.

This document provides some guidelines on the suitable codecs to be
considered for WebRTC endpoints to address the use cases most
relevant to interoperability.

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


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Subject: [rtcweb] partial review of draft-ietf-rtcweb-sdp-01
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Ted asked if I would do a review of draft-ietf-rtcweb-sdp-01. I started, 
but it is taking a long time. I'm sending what I have so far to give you 
something to consider/work on.

In order to simplify this review for both me and readers, I will in 
general only mention an issue once, the first time I find it, and be 
silent on future occurrences of the same issue.

* In general in the examples it would be very helpful if you would 
include a blank line prior to each m-line. When reading these it is 
currently hard to find the m-lines, which is something needed when 
trying understand and compare things.

* Section 3:

I realize this is for a webrtc oriented reader, so I am trying to 
understand it from that perspective. Even so, I have trouble with 
classifying a=ice* as "Security Descriptions". Given the existing 
categories I would think they better belong in "Stream Description".

* Section 5.1:

    o  The term "Session" is used rather loosely in this document to
       refer to either a "Communication Session" or a "RTP Session" or a
       "RTP Stream" depending on the context.

This seems like a bad idea. It may be forgivable to mix communication 
session and RTP session in examples where a single bundle is used for 
all media, or there is only one m-line. But not if multiple m-lines 
aren't bundled, and not with RTP Stream.

OTOH, so far I haven't noticed any place where this is an issue.

* Section 5.2.1:

The SDP line-wrapping convention described in 5.1 isn't being used in 
the example.

Is there a reason for including the IP address in the a=rtcp line, when 
it is the same as the address in the c= line, which is the default?

Is there any expectation that a=ptime:60 only applies to opus? IIUC this 
should be interpreted ad applying to all codecs mentioned in the m-line. 
(There is no order-dependence among attributes within a single media 
section. AFAIK there is no way of specifying a separate ptime for 
individual codecs in an m-line.)

Use of sha-1 in a=fingerprint is about to become incorrect according to 
draft-ietf-mmusic-4572-update-03. That work is being driven by rtcweb, 
so I would think you would want to reflect it here.

The way tables are formatted in RFC format can make the examples 
confusing to readers: the label for table 1 falls exactly between the 
bottom of the table it describes and the following table. When viewed on 
a screen it is difficult to tell what each table is. I suggest tweaking 
the contents of the tables to make this clearer. E.g.,

    +----------------------------------+--------------------------------+
    | SDP Contents - Offer             | RFC#/Notes                     |
    +----------------------------------+--------------------------------+

In Table 2, the o-line has two spaces after the "-". It should only be one.

* Section 5.2.3:

Table 5 has ice and fingerprint attributes prior to the first m-line.

The m-line and associated attributes for the data channel is 
inconsistent with draft-ietf-mmusic-sctp-sdp-16. I think it ought to be 
(more or less):

m=application 56966 UDP/DTLS/SCTP webrtc-datachannel
c=IN IP4 24.23.204.141
a=mid:data
a=sctp-port:5000
a=setup:actpass
...

If you really want to negotiate the usage of channel 1, then it also 
needs to include:

a=dcmap:1 subprotocol="chat";label="channel 1"

(Note: this presumes that "chat" is a registered subprotocol. It isn't!)

* Section 5.2.4:

This seems fine if Bob intends to send audio (e.g. MoH). If not then it 
would be better to answer a=inactive.

* Section 5.2.5:

Has there been any discussion of pros/cons of having DTMF on a separate 
m-line and a separate ssrc? If one or both ends actually think of DTMF 
as being part of another audio stream then this might present problems. 
This could especially be true for interoperation with telephony devices.

Is there anything in rtcweb that prevents having these on the same 
m-line and ssrc?

* Section 5.2.7:

Note that the BAS concept has been removed from Bundle, so you at least 
shouldn't refer to it. (You can still do the O/A if you want.)

In Table 14 (answer), why are ice attributes and fingerprints given 
separately (and differently) for the two m-lines? (Even though the ports 
and addresses are the same.) This is contrary to current bundle (-29) 
procedures.

In the 2nd (BAS) offer there is a gratuitous change in ordering of 
several of the attributes. (Makes it difficult to see what has changed.) 
I also don't understand the other differences between the ice and 
fingerprints in audio and video, and between this and the first offer.

* Section 5.2.8:

This example *assumes* bundle support, and so uses the same addr/port 
for bundled m-lines. But this will fail badly if the assumption turns 
out to be wrong. When making this assumption you ought to use 
bundle-only in the initial offer to ensure nothing bad happens if the 
assumption of bundle support by answerer turns out to be wrong. AFAIK 
there is no downside to doing this.

Again this doesn't follow current bundle rules.

=======

That's as far as I have gotten. (Slow going.) I'll try to get you more 
later.

	Thanks,
	Paul


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To: rtcweb@ietf.org
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Date: Tue, 31 May 2016 15:09:35 -0400
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Subject: Re: [rtcweb] partial review of draft-ietf-rtcweb-sdp-01
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Another general issue:

I have been *trying* to verify the consistency in use of ports, IP 
addresses, fingerprints, ice ufrag info, ice passwords, ssrcs, cnames, etc.

Of course this is excruciating to do. (For a reviewer like me, and also 
for a potential implementer trying to understand the examples.)

My preliminary take is that there are a number of errors. A lot of them 
are probably cut/paste errors. With sufficient analysis it is sometimes 
possible to figure out what was intended vs. what is written. But it is 
hard with this much stuff.

It would be easier if, instead of using random sample values, the values 
were replaced with semantically relevant and easily recognized values. 
(E.g., "ALICE-SSRC-1", "TED-CNAME-2") Unfortunately I don't think this 
can be done while also having those values by syntactically valid.

I think I will end up making such a replacement myself, in a local copy, 
simply to assist in analyzing what is there.

	Thanks,
	Paul

On 5/30/16 6:14 PM, Paul Kyzivat wrote:
> Ted asked if I would do a review of draft-ietf-rtcweb-sdp-01. I started,
> but it is taking a long time. I'm sending what I have so far to give you
> something to consider/work on.
>
> In order to simplify this review for both me and readers, I will in
> general only mention an issue once, the first time I find it, and be
> silent on future occurrences of the same issue.
>
> * In general in the examples it would be very helpful if you would
> include a blank line prior to each m-line. When reading these it is
> currently hard to find the m-lines, which is something needed when
> trying understand and compare things.
>
> * Section 3:
>
> I realize this is for a webrtc oriented reader, so I am trying to
> understand it from that perspective. Even so, I have trouble with
> classifying a=ice* as "Security Descriptions". Given the existing
> categories I would think they better belong in "Stream Description".
>
> * Section 5.1:
>
>    o  The term "Session" is used rather loosely in this document to
>       refer to either a "Communication Session" or a "RTP Session" or a
>       "RTP Stream" depending on the context.
>
> This seems like a bad idea. It may be forgivable to mix communication
> session and RTP session in examples where a single bundle is used for
> all media, or there is only one m-line. But not if multiple m-lines
> aren't bundled, and not with RTP Stream.
>
> OTOH, so far I haven't noticed any place where this is an issue.
>
> * Section 5.2.1:
>
> The SDP line-wrapping convention described in 5.1 isn't being used in
> the example.
>
> Is there a reason for including the IP address in the a=rtcp line, when
> it is the same as the address in the c= line, which is the default?
>
> Is there any expectation that a=ptime:60 only applies to opus? IIUC this
> should be interpreted ad applying to all codecs mentioned in the m-line.
> (There is no order-dependence among attributes within a single media
> section. AFAIK there is no way of specifying a separate ptime for
> individual codecs in an m-line.)
>
> Use of sha-1 in a=fingerprint is about to become incorrect according to
> draft-ietf-mmusic-4572-update-03. That work is being driven by rtcweb,
> so I would think you would want to reflect it here.
>
> The way tables are formatted in RFC format can make the examples
> confusing to readers: the label for table 1 falls exactly between the
> bottom of the table it describes and the following table. When viewed on
> a screen it is difficult to tell what each table is. I suggest tweaking
> the contents of the tables to make this clearer. E.g.,
>
>    +----------------------------------+--------------------------------+
>    | SDP Contents - Offer             | RFC#/Notes                     |
>    +----------------------------------+--------------------------------+
>
> In Table 2, the o-line has two spaces after the "-". It should only be one.
>
> * Section 5.2.3:
>
> Table 5 has ice and fingerprint attributes prior to the first m-line.
>
> The m-line and associated attributes for the data channel is
> inconsistent with draft-ietf-mmusic-sctp-sdp-16. I think it ought to be
> (more or less):
>
> m=application 56966 UDP/DTLS/SCTP webrtc-datachannel
> c=IN IP4 24.23.204.141
> a=mid:data
> a=sctp-port:5000
> a=setup:actpass
> ...
>
> If you really want to negotiate the usage of channel 1, then it also
> needs to include:
>
> a=dcmap:1 subprotocol="chat";label="channel 1"
>
> (Note: this presumes that "chat" is a registered subprotocol. It isn't!)
>
> * Section 5.2.4:
>
> This seems fine if Bob intends to send audio (e.g. MoH). If not then it
> would be better to answer a=inactive.
>
> * Section 5.2.5:
>
> Has there been any discussion of pros/cons of having DTMF on a separate
> m-line and a separate ssrc? If one or both ends actually think of DTMF
> as being part of another audio stream then this might present problems.
> This could especially be true for interoperation with telephony devices.
>
> Is there anything in rtcweb that prevents having these on the same
> m-line and ssrc?
>
> * Section 5.2.7:
>
> Note that the BAS concept has been removed from Bundle, so you at least
> shouldn't refer to it. (You can still do the O/A if you want.)
>
> In Table 14 (answer), why are ice attributes and fingerprints given
> separately (and differently) for the two m-lines? (Even though the ports
> and addresses are the same.) This is contrary to current bundle (-29)
> procedures.
>
> In the 2nd (BAS) offer there is a gratuitous change in ordering of
> several of the attributes. (Makes it difficult to see what has changed.)
> I also don't understand the other differences between the ice and
> fingerprints in audio and video, and between this and the first offer.
>
> * Section 5.2.8:
>
> This example *assumes* bundle support, and so uses the same addr/port
> for bundled m-lines. But this will fail badly if the assumption turns
> out to be wrong. When making this assumption you ought to use
> bundle-only in the initial offer to ensure nothing bad happens if the
> assumption of bundle support by answerer turns out to be wrong. AFAIK
> there is no downside to doing this.
>
> Again this doesn't follow current bundle rules.
>
> =======
>
> That's as far as I have gotten. (Slow going.) I'll try to get you more
> later.
>
>     Thanks,
>     Paul
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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Thanks Paul for reviewing the draft.  We will go through your comments and
get back to you. In the meanwhile, I suggest people to HTML rendering of
the draft for ease of readability.

Here is the link: https://tools.ietf.org/id/draft-ietf-rtcweb-sdp-01.html

Thanks
Suhas

On Tue, May 31, 2016 at 12:09 PM, Paul Kyzivat <pkyzivat@alum.mit.edu>
wrote:

> Another general issue:
>
> I have been *trying* to verify the consistency in use of ports, IP
> addresses, fingerprints, ice ufrag info, ice passwords, ssrcs, cnames, etc.
>
> Of course this is excruciating to do. (For a reviewer like me, and also
> for a potential implementer trying to understand the examples.)
>
> My preliminary take is that there are a number of errors. A lot of them
> are probably cut/paste errors. With sufficient analysis it is sometimes
> possible to figure out what was intended vs. what is written. But it is
> hard with this much stuff.
>
> It would be easier if, instead of using random sample values, the values
> were replaced with semantically relevant and easily recognized values.
> (E.g., "ALICE-SSRC-1", "TED-CNAME-2") Unfortunately I don't think this can
> be done while also having those values by syntactically valid.
>
> I think I will end up making such a replacement myself, in a local copy,
> simply to assist in analyzing what is there.
>
>         Thanks,
>         Paul
>
>
> On 5/30/16 6:14 PM, Paul Kyzivat wrote:
>
>> Ted asked if I would do a review of draft-ietf-rtcweb-sdp-01. I started,
>> but it is taking a long time. I'm sending what I have so far to give you
>> something to consider/work on.
>>
>> In order to simplify this review for both me and readers, I will in
>> general only mention an issue once, the first time I find it, and be
>> silent on future occurrences of the same issue.
>>
>> * In general in the examples it would be very helpful if you would
>> include a blank line prior to each m-line. When reading these it is
>> currently hard to find the m-lines, which is something needed when
>> trying understand and compare things.
>>
>> * Section 3:
>>
>> I realize this is for a webrtc oriented reader, so I am trying to
>> understand it from that perspective. Even so, I have trouble with
>> classifying a=ice* as "Security Descriptions". Given the existing
>> categories I would think they better belong in "Stream Description".
>>
>> * Section 5.1:
>>
>>    o  The term "Session" is used rather loosely in this document to
>>       refer to either a "Communication Session" or a "RTP Session" or a
>>       "RTP Stream" depending on the context.
>>
>> This seems like a bad idea. It may be forgivable to mix communication
>> session and RTP session in examples where a single bundle is used for
>> all media, or there is only one m-line. But not if multiple m-lines
>> aren't bundled, and not with RTP Stream.
>>
>> OTOH, so far I haven't noticed any place where this is an issue.
>>
>> * Section 5.2.1:
>>
>> The SDP line-wrapping convention described in 5.1 isn't being used in
>> the example.
>>
>> Is there a reason for including the IP address in the a=rtcp line, when
>> it is the same as the address in the c= line, which is the default?
>>
>> Is there any expectation that a=ptime:60 only applies to opus? IIUC this
>> should be interpreted ad applying to all codecs mentioned in the m-line.
>> (There is no order-dependence among attributes within a single media
>> section. AFAIK there is no way of specifying a separate ptime for
>> individual codecs in an m-line.)
>>
>> Use of sha-1 in a=fingerprint is about to become incorrect according to
>> draft-ietf-mmusic-4572-update-03. That work is being driven by rtcweb,
>> so I would think you would want to reflect it here.
>>
>> The way tables are formatted in RFC format can make the examples
>> confusing to readers: the label for table 1 falls exactly between the
>> bottom of the table it describes and the following table. When viewed on
>> a screen it is difficult to tell what each table is. I suggest tweaking
>> the contents of the tables to make this clearer. E.g.,
>>
>>    +----------------------------------+--------------------------------+
>>    | SDP Contents - Offer             | RFC#/Notes                     |
>>    +----------------------------------+--------------------------------+
>>
>> In Table 2, the o-line has two spaces after the "-". It should only be
>> one.
>>
>> * Section 5.2.3:
>>
>> Table 5 has ice and fingerprint attributes prior to the first m-line.
>>
>> The m-line and associated attributes for the data channel is
>> inconsistent with draft-ietf-mmusic-sctp-sdp-16. I think it ought to be
>> (more or less):
>>
>> m=application 56966 UDP/DTLS/SCTP webrtc-datachannel
>> c=IN IP4 24.23.204.141
>> a=mid:data
>> a=sctp-port:5000
>> a=setup:actpass
>> ...
>>
>> If you really want to negotiate the usage of channel 1, then it also
>> needs to include:
>>
>> a=dcmap:1 subprotocol="chat";label="channel 1"
>>
>> (Note: this presumes that "chat" is a registered subprotocol. It isn't!)
>>
>> * Section 5.2.4:
>>
>> This seems fine if Bob intends to send audio (e.g. MoH). If not then it
>> would be better to answer a=inactive.
>>
>> * Section 5.2.5:
>>
>> Has there been any discussion of pros/cons of having DTMF on a separate
>> m-line and a separate ssrc? If one or both ends actually think of DTMF
>> as being part of another audio stream then this might present problems.
>> This could especially be true for interoperation with telephony devices.
>>
>> Is there anything in rtcweb that prevents having these on the same
>> m-line and ssrc?
>>
>> * Section 5.2.7:
>>
>> Note that the BAS concept has been removed from Bundle, so you at least
>> shouldn't refer to it. (You can still do the O/A if you want.)
>>
>> In Table 14 (answer), why are ice attributes and fingerprints given
>> separately (and differently) for the two m-lines? (Even though the ports
>> and addresses are the same.) This is contrary to current bundle (-29)
>> procedures.
>>
>> In the 2nd (BAS) offer there is a gratuitous change in ordering of
>> several of the attributes. (Makes it difficult to see what has changed.)
>> I also don't understand the other differences between the ice and
>> fingerprints in audio and video, and between this and the first offer.
>>
>> * Section 5.2.8:
>>
>> This example *assumes* bundle support, and so uses the same addr/port
>> for bundled m-lines. But this will fail badly if the assumption turns
>> out to be wrong. When making this assumption you ought to use
>> bundle-only in the initial offer to ensure nothing bad happens if the
>> assumption of bundle support by answerer turns out to be wrong. AFAIK
>> there is no downside to doing this.
>>
>> Again this doesn't follow current bundle rules.
>>
>> =======
>>
>> That's as far as I have gotten. (Slow going.) I'll try to get you more
>> later.
>>
>>     Thanks,
>>     Paul
>>
>> _______________________________________________
>> 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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Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Thanks Paul for reviewing the draft.=C2=A0 We will go thro=
ugh your comments and get back to you. In the meanwhile, I suggest people t=
o HTML rendering of the draft for ease of readability.<div><br></div><div>H=
ere is the link:=C2=A0<a href=3D"https://tools.ietf.org/id/draft-ietf-rtcwe=
b-sdp-01.html">https://tools.ietf.org/id/draft-ietf-rtcweb-sdp-01.html</a><=
/div><div><br></div><div>Thanks</div><div>Suhas</div></div><div class=3D"gm=
ail_extra"><br><div class=3D"gmail_quote">On Tue, May 31, 2016 at 12:09 PM,=
 Paul Kyzivat <span dir=3D"ltr">&lt;<a href=3D"mailto:pkyzivat@alum.mit.edu=
" target=3D"_blank">pkyzivat@alum.mit.edu</a>&gt;</span> wrote:<br><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc s=
olid;padding-left:1ex">Another general issue:<br>
<br>
I have been *trying* to verify the consistency in use of ports, IP addresse=
s, fingerprints, ice ufrag info, ice passwords, ssrcs, cnames, etc.<br>
<br>
Of course this is excruciating to do. (For a reviewer like me, and also for=
 a potential implementer trying to understand the examples.)<br>
<br>
My preliminary take is that there are a number of errors. A lot of them are=
 probably cut/paste errors. With sufficient analysis it is sometimes possib=
le to figure out what was intended vs. what is written. But it is hard with=
 this much stuff.<br>
<br>
It would be easier if, instead of using random sample values, the values we=
re replaced with semantically relevant and easily recognized values. (E.g.,=
 &quot;ALICE-SSRC-1&quot;, &quot;TED-CNAME-2&quot;) Unfortunately I don&#39=
;t think this can be done while also having those values by syntactically v=
alid.<br>
<br>
I think I will end up making such a replacement myself, in a local copy, si=
mply to assist in analyzing what is there.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Thanks,<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Paul<div class=3D"HOEnZb"><div class=3D"h5"><br=
>
<br>
On 5/30/16 6:14 PM, Paul Kyzivat wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Ted asked if I would do a review of draft-ietf-rtcweb-sdp-01. I started,<br=
>
but it is taking a long time. I&#39;m sending what I have so far to give yo=
u<br>
something to consider/work on.<br>
<br>
In order to simplify this review for both me and readers, I will in<br>
general only mention an issue once, the first time I find it, and be<br>
silent on future occurrences of the same issue.<br>
<br>
* In general in the examples it would be very helpful if you would<br>
include a blank line prior to each m-line. When reading these it is<br>
currently hard to find the m-lines, which is something needed when<br>
trying understand and compare things.<br>
<br>
* Section 3:<br>
<br>
I realize this is for a webrtc oriented reader, so I am trying to<br>
understand it from that perspective. Even so, I have trouble with<br>
classifying a=3Dice* as &quot;Security Descriptions&quot;. Given the existi=
ng<br>
categories I would think they better belong in &quot;Stream Description&quo=
t;.<br>
<br>
* Section 5.1:<br>
<br>
=C2=A0 =C2=A0o=C2=A0 The term &quot;Session&quot; is used rather loosely in=
 this document to<br>
=C2=A0 =C2=A0 =C2=A0 refer to either a &quot;Communication Session&quot; or=
 a &quot;RTP Session&quot; or a<br>
=C2=A0 =C2=A0 =C2=A0 &quot;RTP Stream&quot; depending on the context.<br>
<br>
This seems like a bad idea. It may be forgivable to mix communication<br>
session and RTP session in examples where a single bundle is used for<br>
all media, or there is only one m-line. But not if multiple m-lines<br>
aren&#39;t bundled, and not with RTP Stream.<br>
<br>
OTOH, so far I haven&#39;t noticed any place where this is an issue.<br>
<br>
* Section 5.2.1:<br>
<br>
The SDP line-wrapping convention described in 5.1 isn&#39;t being used in<b=
r>
the example.<br>
<br>
Is there a reason for including the IP address in the a=3Drtcp line, when<b=
r>
it is the same as the address in the c=3D line, which is the default?<br>
<br>
Is there any expectation that a=3Dptime:60 only applies to opus? IIUC this<=
br>
should be interpreted ad applying to all codecs mentioned in the m-line.<br=
>
(There is no order-dependence among attributes within a single media<br>
section. AFAIK there is no way of specifying a separate ptime for<br>
individual codecs in an m-line.)<br>
<br>
Use of sha-1 in a=3Dfingerprint is about to become incorrect according to<b=
r>
draft-ietf-mmusic-4572-update-03. That work is being driven by rtcweb,<br>
so I would think you would want to reflect it here.<br>
<br>
The way tables are formatted in RFC format can make the examples<br>
confusing to readers: the label for table 1 falls exactly between the<br>
bottom of the table it describes and the following table. When viewed on<br=
>
a screen it is difficult to tell what each table is. I suggest tweaking<br>
the contents of the tables to make this clearer. E.g.,<br>
<br>
=C2=A0 =C2=A0+----------------------------------+--------------------------=
------+<br>
=C2=A0 =C2=A0| SDP Contents - Offer=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0| RFC#/Notes=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0|<br>
=C2=A0 =C2=A0+----------------------------------+--------------------------=
------+<br>
<br>
In Table 2, the o-line has two spaces after the &quot;-&quot;. It should on=
ly be one.<br>
<br>
* Section 5.2.3:<br>
<br>
Table 5 has ice and fingerprint attributes prior to the first m-line.<br>
<br>
The m-line and associated attributes for the data channel is<br>
inconsistent with draft-ietf-mmusic-sctp-sdp-16. I think it ought to be<br>
(more or less):<br>
<br>
m=3Dapplication 56966 UDP/DTLS/SCTP webrtc-datachannel<br>
c=3DIN IP4 24.23.204.141<br>
a=3Dmid:data<br>
a=3Dsctp-port:5000<br>
a=3Dsetup:actpass<br>
...<br>
<br>
If you really want to negotiate the usage of channel 1, then it also<br>
needs to include:<br>
<br>
a=3Ddcmap:1 subprotocol=3D&quot;chat&quot;;label=3D&quot;channel 1&quot;<br=
>
<br>
(Note: this presumes that &quot;chat&quot; is a registered subprotocol. It =
isn&#39;t!)<br>
<br>
* Section 5.2.4:<br>
<br>
This seems fine if Bob intends to send audio (e.g. MoH). If not then it<br>
would be better to answer a=3Dinactive.<br>
<br>
* Section 5.2.5:<br>
<br>
Has there been any discussion of pros/cons of having DTMF on a separate<br>
m-line and a separate ssrc? If one or both ends actually think of DTMF<br>
as being part of another audio stream then this might present problems.<br>
This could especially be true for interoperation with telephony devices.<br=
>
<br>
Is there anything in rtcweb that prevents having these on the same<br>
m-line and ssrc?<br>
<br>
* Section 5.2.7:<br>
<br>
Note that the BAS concept has been removed from Bundle, so you at least<br>
shouldn&#39;t refer to it. (You can still do the O/A if you want.)<br>
<br>
In Table 14 (answer), why are ice attributes and fingerprints given<br>
separately (and differently) for the two m-lines? (Even though the ports<br=
>
and addresses are the same.) This is contrary to current bundle (-29)<br>
procedures.<br>
<br>
In the 2nd (BAS) offer there is a gratuitous change in ordering of<br>
several of the attributes. (Makes it difficult to see what has changed.)<br=
>
I also don&#39;t understand the other differences between the ice and<br>
fingerprints in audio and video, and between this and the first offer.<br>
<br>
* Section 5.2.8:<br>
<br>
This example *assumes* bundle support, and so uses the same addr/port<br>
for bundled m-lines. But this will fail badly if the assumption turns<br>
out to be wrong. When making this assumption you ought to use<br>
bundle-only in the initial offer to ensure nothing bad happens if the<br>
assumption of bundle support by answerer turns out to be wrong. AFAIK<br>
there is no downside to doing this.<br>
<br>
Again this doesn&#39;t follow current bundle rules.<br>
<br>
=3D=3D=3D=3D=3D=3D=3D<br>
<br>
That&#39;s as far as I have gotten. (Slow going.) I&#39;ll try to get you m=
ore<br>
later.<br>
<br>
=C2=A0 =C2=A0 Thanks,<br>
=C2=A0 =C2=A0 Paul<br>
<br>
_______________________________________________<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/listinfo/rtcweb</a><br>
<br>
</blockquote>
<br>
_______________________________________________<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/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--001a114914f0e611c50534292cf0--

