I've traced the problem. The host is not handling VAD properly. With VAD enabled, pjsua does not send audio samples to the host, and he is expecting audio samples (even if it is silence). So the parser on the other side exits abnormally and do not end the sip call properly, so I'm not getting the BYE. I've disabled VAD, ( no_vad = 1 ) and it's working fine now ( not dropping the calls when i do not produce sound ). Regarding the question I made about the multiple invite sessions, im using pjsua_make_call, so, a new invite session is properly made inside of this function and i don't have to worry about it. I miss confused because we have 2 possibilities of work. Directly with pjsip and pjmedia (where we have to worry about those things), or with pjsua-api. Thanks On Tue, Jun 3, 2008 at 8:46 PM, Benny Prijono <bennylp at pjsip.org> wrote: > On Tue, Jun 3, 2008 at 6:26 PM, Norman Franke <norman at myasd.com> wrote: > > This may not be related, but using the latest PJSIP from subversion, I > get > > something similar. If I make a call, terminate it, then make another, > I'll > > end up with two calls and garbled audio. (Like it mixes the RTP from > both.) > > However, this only happens if I compile with debug (-g), otherwise, > things > > work fine. > > I'm not sure what's happening there. If you could reproduce it with > pjsua, or have some trace info, then I can give it a look (I tried it > and it works fine here). > > Cheers > Benny > > _______________________________________________ > 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 > -- Joao Cesar msn: jpcesar at gmail.com gtalk: jpcesar at gmail.com icq: 13790802 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080604/9a0edfe9/attachment-0001.html
