Hi users,
This is ngrep report regarding my problem;
-----------------<when a sip proxy behind NAT and the user registerd to that sip-proxy calls>------------------------- U 82.102.69.105:39871 -> 81.21.33.35:5060 INVITE sip:99106883@81.21.33.35:5060 SIP/2.0. To: "99106883"sip:99106883@81.21.33.35:5060. From: "12345"sip:12345@81.21.33.35:5060;tag=c86b66ad8b9187c8. Via: SIP/2.0/UDP 192.168.1.100:5060 ;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKf638e18b56022ea3. Call-ID: a78d5c993a9dd6b4@192.168.1.102. CSeq: 47344 INVITE. Record-Route: sip:192.168.1.100:5060. Contact: sip:192.168.1.100:5060. Max-Forwards: 69. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 361. . v=0. o=line2 8000 8000 IN IP4 192.168.1.102. s=SIP Call. c=IN IP4 82.102.69.105.---------------------------------------------> this is the NAT for the sip-proxy , t=0 0. m=audio 5004 RTP/AVP 18 4 2 97 9 0 101. a=fmtp:97 mode=20. a=fmtp:101 0-11. a=ptime:20. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:9 G722/16000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv.
-----------------------------<when my SIP-SERVER standing on public ip recieved the "INV" message from the above"-----------------------------------
U 81.21.33.35:5060 -> pstngw:5060 INVITE sip:99106883@pstngw:5060 SIP/2.0. Record-Route: sip:99106883@81.21.33.35 :5060;nat=yes;ftag=c86b66ad8b9187c8;lr=on. To: "99106883"sip:99106883@81.21.33.35:5060. From: "12345"sip:12345@81.21.33.35:5060;tag=c86b66ad8b9187c8. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK0ab.9522bc25.0. Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105 ;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport=39871. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKf638e18b56022ea3. Call-ID: a78d5c993a9dd6b4@192.168.1.102. CSeq: 47344 INVITE. Record-Route: sip:192.168.1.100:5060. Contact: sip:82.102.69.105:39871. Max-Forwards: 16. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 360. . v=0. o=line2 8000 8000 IN IP4 192.168.1.102. s=SIP Call. c=IN IP4 81.21.33.35.--------------------------------------------->This is my SIP-SERVER standing on public ip (and is sending to pstngw) t=0 0. m=audio 60516 RTP/AVP 18 4 2 97 9 0 101. a=fmtp:97 mode=20. a=fmtp:101 0-11. a=ptime:20. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:9 G722/16000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv.
---------------------------------------------<and the pstngw is making call>-----------------------------------------------
U 81.21.33.35:5060 -> pstngw:39871 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105;branch=z9hG4bK-d87543-ac3b034487ba16641188e4c9e5ad0664-1--d87543-;rport=39871,SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK01c64e769ba6c176. From: "12345"sip:12345@81.21.33.35:5060;tag=9ca5c1fe3b43aad6. To: "99106883"sip:99106883@81.21.33.35:5060;tag=7E6010EC-1764. Date: Wed, 06 Sep 2006 10:27:40 GMT. Call-ID: 72d0e4e8adca2ab2@192.168.1.102. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 32352 INVITE. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 237. . v=0. o=CiscoSystemsSIP-GW-UserAgent 54 9902 IN IP4 pstngw s=SIP Call. c=IN IP4 81.21.33.35.-------------------------------------<in pstngw it noticed the contact header of SIP-SERVER>--------------------- t=0 0. m=audio 60518 RTP/AVP 18 100. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:100 X-NSE/8000. a=fmtp:100 192-194. a=ptime:20.
---------------------------------------------<after talking some time if any of UA hung the phone (here i am showing from pstn hung up) the result goes like this----------------
U pstngw:52991 -> 81.21.33.35:5060 BYE sip:99106883@81.21.33.35:5060;nat=yes;ftag=1bab08cdaad1d6e8;lr=on SIP/2.0. Via: SIP/2.0/UDP 81.21.38.15:5060. From: "99106883"sip:99106883@81.21.33.35:5060;tag=7E66E3D8-1292. To: "12345"sip:12345@81.21.33.35:5060;tag=1bab08cdaad1d6e8. Date: Wed, 06 Sep 2006 10:35:07 GMT. Call-ID: c2dd3fb9554ef6e4@192.168.1.102. User-Agent: Cisco-SIPGateway/IOS-12.x. Max-Forwards: 6. Route: sip:192.168.1.100:5060, sip:82.102.69.105:39871.------------------------> What is happening here ??????? Timestamp: 1157538919. CSeq: 101 BYE. Content-Length: 0. .
# U 81.21.33.35:5060 -> 192.168.1.100:5060--------------------------------------------------->here unexpected problem arises for me it have to use 82.102.69.105 BYE sip:192.168.1.100:5060 SIP/2.0. Record-Route: sip:81.21.33.35;ftag=7E66E3D8-1292;lr=on. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bKb0f1.3fe32691.0. Via: SIP/2.0/UDP pstngw:5060. From: "99106883"sip:99106883@81.21.33.35:5060;tag=7E66E3D8-1292. To: "12345"sip:12345@81.21.33.35:5060;tag=1bab08cdaad1d6e8. Date: Wed, 06 Sep 2006 10:35:07 GMT. Call-ID: c2dd3fb9554ef6e4@192.168.1.102. User-Agent: Cisco-SIPGateway/IOS-12.x. Max-Forwards: 5. Route: sip:82.102.69.105:39871. Timestamp: 1157538919. CSeq: 101 BYE. Content-Length: 0. P-hint: rr-enforced. So the call never ends up from SIP SERVER to other party
Make some comments on above and assist me where am I going wrong:
and i am using www.openser.org pstn default script as ser.cfg
Hope to get some help Thanks, Ravi.