Re: WebRTC2SIP w/o SIPML5?

[James Mortensen]

I almost feel like you could just take the http://apprtc.appspot.com demos, use JSON to transport the SDP's, and leave out sipML5 and webrtc2sip altogether.  The main reason you'd want webrtc2sip is to speak to legacy SIP clients, and in order to do that you're going to need a SIP stack of some sort.  

A colleague and I were just talking about this yesterday. If we used a non-SIP method for sending data to our server, we'd then need to read all of those dry RFC specs, become SIP experts, and write a converter to convert our custom transport to SIP so that it can talk to everything else that uses the standard. :)  We didn't think that sounded too appealing. :D

With that said, another option is to take the JsSIP library, which is 564K uncompressed, and hack it to add in the RTCWebBreaker SIP headers and UDP Proxy header so that the SIP messages will be interpreted as RTCWebBreaker being ON.  

Hope this helps!

James

On Friday, May 17, 2013 11:24:01 AM UTC-7, Jim Li wrote:

Hi!  Is it possible to setup webrtc2sip to allow for simpler call control from the browser? Basically, it would be great if I could set up webrtc2sip to have some kind of REST interface that I could ping from the browser to set up a call. The output from webrtc2sip would be, of course, SIP. I'm just trying to "lighten up" the interface between the browser to the proxy. 

The sipML5 library is pretty large (> 1MB) and I only need to set up simple outbound voice calls to my own servers. I don't need to accept inbound calls, REGISTER with my servers, or much of the other stuff that a full blown SIP stack offers.

If anyone has hacked such a modification to webrtc2sip, or can provide pointers into the code for accomplishing that, please let me know.

Thanks in advance,

Jim