Re: [rtcweb] SIP vs Websocket in RTCWeb [was RE: SIP MUST NOT be used in browser?]

Peter Saint-Andre <stpeter@stpeter.im> Mon, 12 September 2011 21:37 UTC

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Date: Mon, 12 Sep 2011 15:39:23 -0600
From: Peter Saint-Andre <stpeter@stpeter.im>
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To: Ravindran Parthasarathi <pravindran@sonusnet.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] SIP vs Websocket in RTCWeb [was RE: SIP MUST NOT be used in browser?]
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On 9/12/11 12:41 PM, Ravindran Parthasarathi wrote:
> Peter,
> 
> <snip> > I think that the protocol used for server-to-server federation
> is a 
>> matter for the service providers and thus is not in scope for RTCWeb.
> </snip>
> 
> Sec 4.2.5 of draft-ietf-rtcweb-use-cases-and-requirements-04 includes
> the usecase for federation. My assumption is based on this usecases.
> Till now, there is no better protocol than SIP exists for this
> discussion.
> 
>> Some s2s links might use SIP, some might use XMPP/Jingle, etc.
>> </snip>
> 
> I agree with you that signaling interop is not focus of RTCWeb and only
> one webserver is solution space then your solution looks like working.
> Even then, I prefer single signaling protocol rather than ad-hoc
> signaling protocol by each web developer.

Using DNS SRV records, it's easy enough to figure out what protocols are
supported by the peer service with which you want to federate.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/