On 1/15/08, Angelos Karageorgiou <angelos at unix.gr> wrote: > > 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 ? I don't know Asterisk well enough to say if nat=yes option makes a difference. If you give this option, does Asterisk listen to RTP packets in the ports specified in SDP? I just verified again if siprtp does send RTP packets even if it does not receive RTP packet, and it does. I'm using PJSIP SVN version, if that makes any difference. cheers, -benny -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080116/47967487/attachment.html
