Re: [rtcweb] Making progress on the signaling discussion (NB: Action items enclosed!)

Harald Alvestrand <harald@alvestrand.no> Wed, 12 October 2011 09:01 UTC

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Date: Wed, 12 Oct 2011 11:01:41 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Making progress on the signaling discussion (NB: Action items enclosed!)
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On 10/11/11 10:37, Iñaki Baz Castillo wrote:
> 2011/10/7 Iñaki Baz Castillo<ibc@aliax.net>;:
>>> To clarify, what I see the question for the WG is how the signalling
>>> model for RTCWEB session establishment is intended to work. Thus the
>>> drafts intended for answering this needs to be focused on describing a
>>> signalling model.
>> Yes, agreed.
>>
>>
>> Regards.
>>
>>
>> PS: SIP over WebSocket just works fine. A demonstration next week.
>
> Hi, as told in other thread, the demonstration and video of SIP over
> WebSocket is already available:
>
>    http://sip-on-the-web.aliax.net/
>
> So this becomes a signaling protocol example for RTCweb, a *known*
> signaling protocol in fact (so we can figure what we need from the
> future RTCweb JS API in order to deal with SDP's and media sessions).
>
the video has a caveat of course:

we don't know that the signalling can be successfully connected to the 
browser's API, because there's no API to connect to yet.

One of the worries I have with doing a "low level spec" unconstrained by 
our present competence (ignorance?) in  is that I'm reasonably sure we 
have the knowledge to generate and parse SDP, because the codebases we 
are building on already generate and parse SDP, and the information 
present there is enough to set up calls, because we're already setting 
up calls using that information.

I'm less sure of our "getting things right" if we start off by 
describing the capabilities and control knobs to be exposed in an 
unconstrained API.

                    Harald