Occassional posts about VoIP, SIP, WebRTC and Bitcoin.
I’ve been half heartedly seeing if any softphone makers are interested in supporting the pseudo ICE mechanism that GTalk uses when setting up the media on an XMPP call. It doesn’t look promising at this stage. Google did make a change around the start of this month to their XMPP call set up mechanism, which broke the ability for Asterisk 1.8 to place outgoing calls through GTalk, so maybe they are working on the service and will have full Jingle support soon which in theory would allow ICE compatible phones such as Counterpath’s softphone range and others to be able to place SIP calls through sipsorcery and have them terminated via GTalk/Google Voice’s XMPP/Jingle service. That would be neat as it would be a validation of SIP and XMPP signalling working together and interconnecting two technologies which both support a large number of users.
However Jingle for GTalk isn’t here yet and in the interests of encouraging any developers that are involved with writing softphones to look at supporting the Google STUN requests on the RTP sockets I’ve created a prototype application that shows how to do it. The application is written in C# and hosted on codeplex here. What the application does is listen on a socket for a SIP INVITE request and when it gets one translates it to an XMPP request which it sends off to GTalk. As well as handling the SIP and XMPP signalling the prototype application also fires up two media sockets, one that talks to the Google XMPP end and one that talks to the SIP phone end. The media sockets are needed so that the STUN requests and responses required by the Google XMPP end can be handled correctly and that’s the bit that’s missing from the softphones.