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

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. 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>

Received on Monday, 22 July 2013 15:45:44 UTC