Quoting Benny Prijono <bennylp at pjsip.org>: > On 1/15/08, Angelos Karageorgiou <angelos at unix.gr> wrote: >> >> hello all, >> >> I am still trying to make siprtp initiate the rtp stream for a call , >> It seems that media_thread will only wake up AFTER a packet is received >> on the rtp session. > > > That shouldn't be the case, and looking at the code I don't thing that could > happen either as media_thread() starts immediately and it only blocks on > sleep(). How did you test this? Do you have NAT in the middle? > > cheers, > -benny I am using an asterisk server bound on 127.0.0.1 with nat=yes in its sip configuration. In my extensions if I start with callednumber,1,Playback(help) the RTP stream goes back and forth fine if I start with callednumber,1,Echo() , which is the asterisk echo application i.e. it plays back anything you send it, no RTP gets send from siprtp. This was verified with tcpdump also. Any Ideas ? Angelos
