I'm running a SIP server and have been investigating integrating with Google's GTalk service over XMPP. The integration turned out to be fairly painless and I can translate between SIP and XMPP to handle call signalling. The problem is the ICE format Google use to set up the media. Google are a little bit behind with their ICE implementation and instead of utilising RFC5389 STUN requests and the sequence of them specified in the ICE RFC 5245 specification they rely on RFC 3489 STUN requests. Google's STUN request flow is actually very simple, the client needs to send a STUN (RFC3489) binding request to which the Google media server will respond and then send its own STUN binding request to which the client must respond. So that's one STUN binding request each from the client and server. After that RTP starts flowing exactly the same as for a standard SIP call. The specification detailing the Google approach is http://code.google.com/apis/talk/call_signaling.html. The reason I'm sending this email is to see if any pjsip developers are interested in adding the ability for the pjsip stack to set up media with the Google GTalk service by adding the ability to send the STUN requests and responses it requires. AS far as I know there is no current client SIP/media stack that can interoperate with Google's GTalk service (as far as B2BUA's go Asterisk and FreeSWITCH can but I'm after a UA). Regards, Aaron
