W3C home > Mailing lists > Public > public-webrtc@w3.org > July 2013

Re: [rtcweb] Proposed message to send to the IETF rtcweb and W3C WebRTC working groups.

From: Daniel-Constantin Mierla <miconda@gmail.com>
Date: Mon, 22 Jul 2013 18:21:59 +0200
Message-ID: <51ED5C27.6040406@gmail.com>
To: Iñaki Baz Castillo <ibc@aliax.net>
CC: Ralph Meijer <ralphm@ik.nu>, stox <stox@ietf.org>, XMPP Jingle <jingle@xmpp.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
On 7/22/13 5:44 PM, Iñaki Baz Castillo wrote:
> 2013/7/22 Daniel-Constantin Mierla <miconda@gmail.com>:
>> On 7/22/13 5:14 PM, Iñaki Baz Castillo wrote:
>>> Great. First thing you should complain about is the fact that current
>>> WebRTC specification makes unfeasible for a browser to use SDP-XML as
>>> defined by XEP-0167. So if you have a SIP server you will be able to
>>> directly connect from the browser, but if you have a Jingle server you
>>> will need a gateway.
>> You are obviously misinforming here. SIP is the signaling protocol and a SIP
>> server has really little to deal with SDP -- I'm sure you know that.
> I was talking about a SIP device also implementing WebRTC in the media
> plane.
You wrote a SIP server, just read above.

And producing a xml blob instead of text plain blob does not make much 
difference from the architecture point of view, if that was your 
concern, nor simplifies things.


>   Current WebRTC spec mandates plain-SDP usage in the wire to
> signal your media description and transport/ICE information to the
> peer. So if you want to communicate with a XEP-0167 compliant
> server/endpoint, then you need a gateway to convert the plain-SDP
> generated by the browser into the SDP-XML version defined by XEP-0167.
>> And one
>> cannot call directly a SIP endpoint from the browser, as SIP is not a
>> mandatory signaling protocol, so there is extensive need of coding a
>> javascript SIP stack (or reusing an existing one).
> Reusing existing JavaScript SIP stacks is something good, don't you agree?
> Please, let's focus:
> Today from a browser you can speak SIP over WebSocket and connect to a
> SIP media server/gateway understanding the SDP of WebRTC. So yes, you
> can talk SIP fom a browser.
> Today fom a browser you cannot speak XMPP/Jingle (XEP-0167) over
> WebSocket (or over AJAX) because the browser produces a plain SDP
> blob, while you need a XML based SDP as XEP-1067 states. You can parse
> such a SDP blob string in JavaScript and map it into a XML body,
> but... good luck with that...
> Hope it is clear now.
> Regards.
> --
> Iñaki Baz Castillo
> <ibc@aliax.net>

Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Received on Thursday, 25 July 2013 10:44:15 UTC

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