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

Magnus Westerlund <magnus.westerlund@ericsson.com> Thu, 06 October 2011 08:18 UTC

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Date: Thu, 6 Oct 2011 10:21:27 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] Making progress on the signaling discussion (NB: Action items enclosed!)
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On 2011-10-05 19:03, Iñaki Baz Castillo wrote:
> 2011/10/5 Ted Hardie <ted.ietf@gmail.com>;:
>> Sorry, but do you intend to provide a draft?  It was hard for me to parse
>> whether the statement above is a commitment or a statement that you don't
>> have time.
> 
> Maybe not a draft, but a complete explanation and presentation of what
> we have working (SIP over WebSocket + JavaScript code implementing a
> SIP stack + SIP proxy implementing the WebSocket transport). And it
> just works.
> 
> The draft is already done:
> 
>   http://tools.ietf.org/html/draft-ibc-rtcweb-sip-websocket-00
> 
> In fact it defines a concrete signaling protocol (well, not exactly as
> it uses pure SIP, but on top of WebSocket). Of course it aim is to be
> used with rtcweb for media sessions.
> 

Iñaki,

could you please clarify your intention of your proposal?

My interpretation of your draft is that it is a description of how you
can deploy SIP implemented in JS over websocket. Thus enabling SIP
interworking when needed by having the JS SIP stack convert regular SIP
with SDP into interactions with the WEBRTC API. Thus your proposal
really is primarily an argument that you don't need SIP implemented in
the browser, as long as the API allows the JS to create SIP with SDP
messages. Thus I would interpret this as something that could work both
over something similar to the current PeerConnection API proposals and a
more low level API. Where PeerConnection likely is less work as it does
provide the SDP that in most cases don't need modifications.

Is this a correct interpretation and thus your draft is an extension
that simplifies life for the people who like SIP interworking by getting
the SIP functionality into the end-point rather than a gateway?

Or is it something else?

If it is the first I don't see your proposal as complete proposal for
how signaling for RTCWEB, as you still need APIs to something in some
form and with possibly certain semantics that interact with the SIP in
JS implementation.

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.

Cheers

Magnus Westerlund

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