Hi, I have a problem with pjsua-app... I try ./pjsua-i686-pc-linux-gnu --auto-answer=200 --no-tcp --stun-srv= stun.xten.com --ip-addr=10.166.113.155 and pjsua app run perfectly. But when I make a local call, for example... >m >sip:10.166.113.118 the call is established but I can't hear audio... I think that the problem is the rtp ip, these are the sip and sdp invite packets: INVITE sip:10.166.113.118 SIP/2.0 Via: SIP/2.0/UDP 10.166.113.155:5060 ;rport;branch=z9hG4bKPj54d61d49-9be0-4c7e-b3a8-a2266a068833 Max-Forwards: 70 From: <sip:10.166.113.155>;tag=386bc126-3586-4406-87d3-85ad7d094a8f To: sip:10.166.113.118 Contact: <sip:10.166.113.155:5060> Call-ID: 057e028e-e6ca-494a-91e4-9fcbe15c27ad CSeq: 3296 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: PJSUA v1.7/i686-pc-linux-gnu Content-Type: application/sdp Content-Length: 461 v=0 o=- 3492234931 3492234931 IN IP4 84.124.50.13 s=pjmedia c=IN IP4 84.124.50.13 <- why!? t=0 0 a=X-nat:8 m=audio 17390 RTP/AVP 103 102 104 109 3 0 8 9 101 a=rtcp:17391 IN IP4 84.124.50.13 <- why!? a=rtpmap:103 speex/16000 ... (more codecs) a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 does anyone know what the problem is? Thanks very much -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100831/28639ddd/attachment.html>
