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 _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
