Benny- > we're probably not interested in adding GTalk's "ICE" protocol, since > we can't talk directly with it anyway. Can you clarify a little... is this because of RTP encryption? Non-supported codec? Thanks. -Jeff > But it should be fairly simple > to create a media transport adapter for this (similar to SRTP; see > [1]), and add STUN send/receive capability into it (see STUN Session > in [2]). Then plug your adapter on top of existing UDP media transport > using https://trac.pjsip.org/repos/ticket/1173. > > [1] http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__TRANSPORT.htm > [2] http://www.pjsip.org/pjnath/docs/html/group__PJNATH__STUN.htm > > ?Benny > > > On Fri, Dec 10, 2010 at 6:56 PM, Aaron Clauson <aaron at sipsorcery.com> wrote: >> 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 >> >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >
