Re: [rtcweb] I-D Action: draft-ietf-rtcweb-rtp-usage-15.txt

[Colin Perkins <[e-mail]>]

The rtp-usage draft doesn’t mandate any payload formats, and instead defers to the rtcweb-audio draft. If RFC 3389 is to be mandated, it should probably be done in rtcweb-audio.

Colin


On 16 Jul 2014, at 23:20, Justin Uberti <[e-mail]> wrote:

> One thing that has come up while discussing JSEP is the need to specify handling of DTX in the RTP usage document. JSEP allows applications to ask for DTX, so this needs to be supported in all implementations to give deterministic behavior.
> 
> While Opus has the ability to support DTX internally, we also need to make sure this works with other codecs, including G.711. As a result, I think the RTP usage draft needs to specify RFC 3389 support as mandatory-to-implement, so that applications can deterministically enable DTX in all cases.
> 
> 
> On Wed, May 28, 2014 at 7:29 AM, Magnus Westerlund <[e-mail]> wrote:
> WG,
> 
> This version contains some changes as result of last weeks discussion.
> Especially you who wasn't there should review this to see that you agree
> with the changes proposed.
> 
> The changes are:
> 
> - Multi-stream optimization are now a MAY support, MUST signal to use.
> 
> - The T_rr_interval = 4 is now explained in
> draft-ietf-avtcore-rtp-multi-stream, so a reference has been added here.
> 
> - Signalling of extensions has been clarified to be a MUST
> 
> - The DoS potential from RTCP configuration that Martin Thomson noticed
> is now discussed and the general consideration that an WebRTC
> implementation needs safe guards among this is present. WG needs to
> consider if it needs more or are sufficient.
> 
> Cheers
> 
> Magnus
> 
> On 2014-05-28 13:21, [e-mail] wrote:
> >
> > 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 Working Group of the IETF.
> >
> >         Title           : Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
> >         Authors         : Colin Perkins
> >                           Magnus Westerlund
> >                           Joerg Ott
> >       Filename        : draft-ietf-rtcweb-rtp-usage-15.txt
> >       Pages           : 44
> >       Date            : 2014-05-28
> >
> > Abstract:
> >    The Web Real-Time Communication (WebRTC) framework provides support
> >    for direct interactive rich communication using audio, video, text,
> >    collaboration, games, etc. between two peers' web-browsers.  This
> >    memo describes the media transport aspects of the WebRTC framework.
> >    It specifies how the Real-time Transport Protocol (RTP) is used in
> >    the WebRTC context, and gives requirements for which RTP features,
> >    profiles, and extensions need to be supported.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-15
> >
> > A diff from the previous version is available at:
> > http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-rtp-usage-15
> >
> >
> > Please note that it may take a couple of minutes from the time of submission
> > until the htmlized version and diff are available at tools.ietf.org.
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > _______________________________________________
> > rtcweb mailing list
> > [e-mail]
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> >
> 
> 
> --
> 
> Magnus Westerlund
> 
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-- 
Colin Perkins
http://csperkins.org/