Hello Everyone, I am having problem getting RTP packets flowing smoothly. The setup is
NAT Box (192.168.2.1) <-> OpenSIPS/RTPProxy (192.168.2.5) <-> Asterisk (192.168.2.10) I know that media is reaching the boxes since I see: OpenSIPS (192.168.2.5) 0.000000 192.168.2.10 -> 192.168.2.5 UDP 214 Source port: 24454 Destination port: 20198 0.000099 192.168.2.5 -> 81.201.85.45 UDP 214 Source port: 39810 Destination port: 13272 0.017956 192.168.2.10 -> 192.168.2.5 UDP 214 Source port: 24454 Destination port: 20198 0.018028 192.168.2.5 -> 81.201.85.45 UDP 214 Source port: 39810 Destination port: 13272 0.037760 192.168.2.10 -> 192.168.2.5 UDP 214 Source port: 24454 Destination port: 20198 0.037814 192.168.2.5 -> 81.201.85.45 UDP 214 Source port: 39810 Destination port: 13272 Asterisk CLI (192.168.2.10) Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160) Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160) Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160) Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160) Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160) Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160) Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160) RTPProxy Messages: INFO:handle_command: new session [email protected], tag 86219;1 requested, type strong INFO:handle_command: new session on a port 20198 created, tag 86219;1 INFO:handle_command: pre-filling caller's address with 81.201.85.45:13272 INFO:handle_command: lookup on ports 20198/39810, session timer restarted INFO:handle_command: pre-filling callee's address with 192.168.2.10:24454 INFO:handle_delete: forcefully deleting session 1 on ports 20198/39810 INFO:remove_session: RTP stats: 86 in from callee, 0 in from caller, 86 relayed, 0 dropped INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped INFO:remove_session: session on ports 20198/39810 is cleaned up It says 86 in from callee but we do not even have incoming audio. I'm pretty sure it's "rtpproxy_offer/answer" issue so bellow is my configuration: route[1] { xlog("Start Call Route For: [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]\n"); if (has_body("application/sdp")) { xlog("Has SDP: $fu\n"); rtpproxy_offer(); } } onreply_route[1] { xlog("Reply Route 1: [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]\n"); if (has_body("application/sdp")) { xlog("Answering RTP Proxy: $fu\n"); rtpproxy_answer(); } } Your help is greatly appreciated, Nick. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
