Hi! I'm working on a SIP-Client which handles account registration etc. against a SIP-Server but the voice streams run elsewhere. What do we do now? We get the information about IP address and ports from the voice side and rewrite the SDP in PjSUAs "on_call_sdp_created" callback. This works as long as we get the IP address and ports before we get the SIP-INVITE. The problem is now: What do we do if the SIP-INVITE comes before the voice information? Well, we could answer the call with "RINGING" (or not at all), but the SDP has been created already without knowing the data and the "on_call_sdp_created" callback is not called later on. Are we allowed to just rewrite it - the dirty way - via the data structures? I failed to find a useful API function to rewrite/regenerate the SDP, change media parameters or similar. Any hints or suggestions? FWIW, we use pjsip-2.0.1. Bernd -- mobile: +43 664 4416156 http://www.sysprog.at/ Linux Software Development, Consulting and Services
