Hello the list, I have a problem on the proxy with the audio between two calls bridged by a UAC. When I made a normal call, no problems.
My UAC is nated. UAC > Router > KAMAILIO Frames arrives with private IP in the SDP. U 2018/01/18 21:50:16.798581 217.112.180.235:1024 -> 217.112.180.10:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 217.112.180.10;branch=z9hG4bK2d06.2f2a1856bde881173a3fc413c4136b83.0. Via: SIP/2.0/UDP 84.14.241.179:5060;rport=5060;branch=z9hG4bK0dBe3143bcf7f60a70b. Record-Route: <sip:217.112.180.10;lr=on;ftag=gK0d4dfe6f;did=227.c0d2;nat=yes>. From: <sip:32XXXXXX87@ >;tag=gK0d4dfe6f. Call-ID: [email protected]. CSeq: 29328 INVITE. Contact: <sip:[email protected]:5060;transport=udp>. Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY. Supported: timer,100rel. Server: IP Office 10.1.0.0.0 build 237. Min-SE: 1800. Require: timer. Session-Expires: 1800;refresher=uas. To: <sip:[email protected]>;tag=998e429e819ba686. Content-Type: application/sdp. Content-Length: 202. . v=0. o=UserA 3721571424 1025404311 IN IP4 192.168.2.2. s=Session SDP. c=IN IP4 192.168.2.2. t=0 0. m=audio 49154 RTP/AVP 9 101. a=rtpmap:9 G722/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. I don't understand why but the proxy (in case of an incoming call) succeed to determine the public IP. Jan 18 21:40:08 proxy1 rtpproxy[1314]: INFO:rxmit_packets:1fb3f1c6c8dfe738221842406be48450: caller's address filled in: 217.112.180.235:49154 (RTP) Jan 18 21:40:08 proxy1 rtpproxy[1314]: INFO:rxmit_packets:1fb3f1c6c8dfe738221842406be48450: guessing RTCP port for caller to be 49155 Jan 18 21:40:16 proxy1 rtpproxy[1314]: INFO:rxmit_packets:1fb3f1c6c8dfe738221842406be48450: caller's address latched in: 217.112.180.235:49155 (RTCP) Don't know why because the only information in SDP is 192.168.2.2 The Kamailio didn't send the information of the proxy to the UAC , but to the other end as this is an incoming call. So I have audio in this case. When I setup a bridge on the UAC to a second number, we have an issue. ( Kamailio 4.4.6 ) This is the same frames U 2018/01/18 21:51:26.607270 217.112.180.235:1024 -> 217.112.180.10:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 217.112.180.10;branch=z9hG4bK4181.bb7aab84b6fb0aea5836b4d8874406ec.0. Via: SIP/2.0/UDP 84.14.241.179:5060;rport=5060;branch=z9hG4bK0bB16ee5a3d300fee16. Record-Route: <sip:217.112.180.10;lr=on;ftag=gK0b5d7652;did=cc6.5452;nat=yes>. From: <sip:32XXXXXX87@ >;tag=gK0b5d7652. Call-ID: [email protected]. CSeq: 21946 INVITE. Contact: <sip:[email protected]:5060;transport=udp>. Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY. Supported: timer,100rel. Server: IP Office 10.1.0.0.0 build 237. Min-SE: 1800. Require: timer. Session-Expires: 1800;refresher=uas. To: <sip:[email protected]>;tag=559c99f5edcab5d4. Content-Type: application/sdp. Content-Length: 202. . v=0. o=UserA 1781830446 4071482272 IN IP4 192.168.2.2. s=Session SDP. c=IN IP4 192.168.2.2. t=0 0. m=audio 49156 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. But in this case no audio RTCP detected but no the RTP. He took the private ip address "192.168.2.2" and this is the reason of the "no audio". Jan 18 21:51:26 proxy1 rtpproxy[1314]: INFO:rtpp_command_ul_handle:[email protected]: lookup on ports 11264/10446, session timer restarted Jan 18 21:51:26 proxy1 rtpproxy[1314]: INFO:rtpp_command_ul_handle:[email protected]: pre-filling callee's address with 192.168.2.2:49156 Jan 18 21:51:26 proxy1 rtpproxy[1314]: INFO:rxmit_packets:e1e387824e0753522b7bc55e02090784: callee's address latched in: 79.137.49.139:39176 (RTP) Jan 18 21:51:32 proxy1 rtpproxy[1314]: INFO:rxmit_packets:e1e387824e0753522b7bc55e02090784: caller's address filled in: 217.112.180.235:49155 (RTCP) Jan 18 21:51:32 proxy1 rtpproxy[1314]: INFO:rxmit_packets:[email protected]: callee's address filled in: 217.112.180.235:49157 (RTCP) I would like to understand why with the first call, no issues to determine the RTP IP and not in the second case,
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