Re: [rtcweb] I-D Action: draft-ietf-rtcweb-rtp-usage-12.txt

[Magnus Westerlund <[e-mail]>]

WG,

This version is the authors attempt to make a version ready for WG last
call. We list below the changes that we consider significant beyond
editorial improvements.

- Further clarified that draft-ietf-rtcweb-security-arch defines the
crypto suits and key-management solution for the SAVPF profile.

- Removed the discussion of the SHIM based approach (draft-westerlund-
  avtcore-transport-multiplexing)

- Changed SLI and RPSI to RECOMMENDED to implement level based on WG
  mailing list discussion.

- Clarified that the RTP packet stream association with
  MediaStramStracks is the one that MSID provides.

- Removed Simulcast as open issue. We have removed the discussion of
explicitly signaled simulcast and clarified the one that can be done
using multiple mediaStreamTracks based on the same media source. These
can be included in the same or different PeerConnections with different
parameters for basic simulcast support. I still expect that when
simulcast in RTP and SDP has been specified that adoption of that is
considered for WebRTC.

- Added an WebRTC API requirement regarding CSRC list handling.

I recommend that people review these changes and provide feedback.

The authors will continue to strive for a WG last call directly after WG
meeting in London.

Cheers

Magnus


On 2014-02-14 20:51, [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-12.txt
> 	Pages           : 41
> 	Date            : 2014-02-14
> 
> 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-12
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-rtp-usage-12
> 
> 
> 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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