Hi Saul,
OK, aside the TCP part (which anyhow is scheduled for fixing) and some
extra parsing, does supporting WebRTC imply something more on the
OpenSIPS side ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 10/29/2012 12:42 PM, Saúl Ibarra Corretgé wrote:
Hi Bogdan,
On Oct 29, 2012, at 11:26 AM, Bogdan-Andrei Iancu wrote:
Hi Ali,
I have to admit I'm not really familiar with the WebRTC and what are the
requirements for this to work directly over OpenSIPS - there is pending patch
for compatibility with WebRTC (parsing and detecting more VIA params, specific
to WebRTC), but we will need to have a clear view on what needs to be done for
a complete integration - and at that point to say if makes sense to do it or
not.
In a nutshell:
WebRTC is a combined effort between IETF and W3C to bring realtime
communications to web browsers. WebRTC only specifies the media plane, so
signaling is up to applications. Here is the draft specifying the WebSocket
transport for SIP:
http://tools.ietf.org/html/draft-ietf-sipcore-sip-websocket-05 which allows
applications written in JavaScript and running in web browsers to connect to
existing SIP infrastructure.
Since OpenSIPS is a SIP proxy and thus is not concerned by the media plane, supporting
the WebSocket transport is what would be required for it to enter the "WebRTC
game". That said, IMHIO, makes little to no sense to do it until TCP works properly,
given that WebSocket is a transport protocol on top of TCP.
Regards,
--
Saúl Ibarra Corretgé
AG Projects
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