[RTW] [dispatch] RTCWEB I-D with thoughts on the framework
Stefan Håkansson LK
stefan.lk.hakansson at ericsson.com
Fri Feb 11 15:25:35 CET 2011
I agree to large parts of the document, and I think that the section on interoperability with existing VoIP gear brings up important aspects.
However, I am not completely comfortable with a couple of things:
* The application would be responsible for rate adaptation, but what is the incentive for the application developer to do something that is fair to other applications or users? I would prefer that this is part of the browser (and with the auto update feature most browsers use the update cycles could be short anyway)
* Are you suggesting that the <video> (and consequently also the <audio>) tag should not be used? I don't think this is a good idea. Since they are accepted and used by the web community we should use them also for RTC-Web rather than trying to invent something new.
--Stefan
> -----Original Message-----
> From: dispatch-bounces at ietf.org
> [mailto:dispatch-bounces at ietf.org] On Behalf Of Jonathan Rosenberg
> Sent: den 8 februari 2011 15:49
> To: 'DISPATCH list'
> Subject: [dispatch] RTCWEB I-D with thoughts on the framework
>
> Some of my colleagues from Skype and I have put together a
> draft which gives our thoughts on the scope of the protocols
> and functionality for enabling browser RTC. I've just
> submitted the draft, which you can find
> here:
>
> http://www.ietf.org/id/draft-rosenberg-rtcweb-framework-00.txt
>
> Some of the key points:
>
> * keep it minimal
> * APIs for controlling behaviors of the various media
> components, with defaults for those that dont care
> * no SIP or Jingle; leave that to proprietary over websockets/HTTP
> * no ICE - just STUN. ICE can be a javascript library.
>
> Thanks,
> Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D. SkypeID: jdrosen
> Skype Chief Technology Strategist
> jdrosen at skype.net http://www.skype.com
> jdrosen at jdrosen.net http://www.jdrosen.net
>
>
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