On 10/23/2012 10:59 AM, Bernd Petrovitsch wrote: > 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 > Just a quick shot here, but how about initially answering with audio a=recvonly and after receiving the information sending a re-invite with the new SDP and audio a=sendrevc? -- Mit freundlichen Gr??en / Best regards Andreas Wehrmann CENTER COMMUNICATION SYSTEMS GMBH Ein Unternehmen der STRABAG SE Software Development Ignaz-K?ck-Str. 19 A-1210 Wien, ?sterreich Tel.: +43 (0) 190 199 - 3616 Fax: +43 (0) 190 199 - 2110 Mobil: +43 (0) 664 884 75916 a.wehrmann at centersystems.com FN 796 88p, Sitz in Wien Firmenbuchgericht Wien <http://www.centersystems.com/> www.centersystems.com Gesch?ftsf?hrung: Ing. Gerhard Jelinek, Josef-Eduard Burger
