Well I was trying to implement HTTP tunneling with pjsip port on symbian. So far I am successfull. Yet litte problem. I incorported the tunnel client application with the symbian_ua where client is acting as a proxy for the symbian_ua. I am tunneling both SIP and RTP through this proxy. I can now make call from the symbian application through the tunnel to the remote client which is a eyebeam. I can disconnect the call from symbian with no problem. But whenever I try to disconnect from remote client I receive BYE at symbian and symbian_ua is responsing with OK. It is good for every time but symbian Hangup the call for the first time or second time but from 3rd time and onward it never disconnects or hangup but remote client does hangup. Please look at the log file in eyebeam which is same for 3 calls but for the last call eyebeam hangs up fine but symbian does not. It keeps on sending RTP. 05:53:13.1 RECEIVING FROM: 192.168.1.100:5061 INVITE sip:5002 at 192.168.1.101 SIP/2.0 CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.1.100:5061 From: <sip:voipswitch@192.168.1.100:5061>;tag=25065209175132503031 Call-ID: GWaboiQks-F7iLznPGYX4BBFmN0ub5ABPm at 192.168.1.100 To: <sip:5002 at 192.168.1.101> Contact: <sip:192.168.1.100:5061;transport=udp> Content-Type: application/sdp Content-Length: 211 v=0 o=VoipSwitch 7294 7294 IN IP4 192.168.1.100 s=VoipSIP i=Audio Session c=IN IP4 192.168.1.100 t=0 0 m=audio 6294 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 05:53:13.2 Call (l:'xxxxxxxxx' r:'sip:voipswitch at 192.168.1.100:5061') - Incoming call. 05:53:13.4 SENDING TO: 192.168.1.100:5061 SIP/2.0 180 Ringing To: <sip:5002 at 192.168.1.101>;tag=1862d251 From: <sip:voipswitch@192.168.1.100:5061>;tag=25065209175132503031 Via: SIP/2.0/UDP 192.168.1.100:5061 Call-ID: GWaboiQks-F7iLznPGYX4BBFmN0ub5ABPm at 192.168.1.100 CSeq: 1 INVITE Contact: <sip:5002 at 192.168.1.101> Content-Length: 0 05:53:16.2 SENDING TO: 192.168.1.100:5061 SIP/2.0 200 OK To: <sip:5002 at 192.168.1.101>;tag=1862d251 From: <sip:voipswitch@192.168.1.100:5061>;tag=25065209175132503031 Via: SIP/2.0/UDP 192.168.1.100:5061 Call-ID: GWaboiQks-F7iLznPGYX4BBFmN0ub5ABPm at 192.168.1.100 CSeq: 1 INVITE Contact: <sip:5002 at 192.168.1.101> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Content-Length: 255 v=0 o=- 31062339 31062477 IN IP4 192.168.1.101 s=eyeBeam c=IN IP4 192.168.1.101 t=0 0 m=audio 6920 RTP/AVP 0 101 a=alt:1 1 : 9ECC179C 00000023 192.168.1.101 6920 a=fmtp:101 0-15 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 05:53:16.5 RECEIVING FROM: 192.168.1.100:5061 ACK sip:5002 at 192.168.1.101 SIP/2.0 CSeq: 1 ACK Via: SIP/2.0/UDP 192.168.1.100:5061 From: <sip:voipswitch@192.168.1.100:5061>;tag=25065209175132503031 Call-ID: GWaboiQks-F7iLznPGYX4BBFmN0ub5ABPm at 192.168.1.100 To: <sip:5002 at 192.168.1.101>;tag=1862d251 Contact: <sip:192.168.1.100:5061;transport=udp> Content-Length: 0 05:53:20.5 SENDING TO: 192.168.1.100:5061 BYE sip:192.168.1.100:5061;transport=udp SIP/2.0 To: <sip:voipswitch at 192.168.1.100:5061>;tag=25065209175132503031 From: <sip:5002@192.168.1.101>;tag=1862d251 Via: SIP/2.0/UDP 192.168.1.101:6913;branch=z9hG4bK-d87543-392054145-1--d87543-;rport Call-ID: GWaboiQks-F7iLznPGYX4BBFmN0ub5ABPm at 192.168.1.100 CSeq: 2 BYE Contact: <sip:5002 at 192.168.1.101> Max-Forwards: 70 User-Agent: eyeBeam release 3007n stamp 17816 Content-Length: 0 05:53:21.1 RECEIVING FROM: 192.168.1.100:5061 SIP/2.0 200 OK CSeq: 2 BYE Via: SIP/2.0/UDP 192.168.1.101:6913;branch=z9hG4bK-d87543-392054145-1--d87543-;rport From: <sip:5002@192.168.1.101>;tag=1862d251 Call-ID: GWaboiQks-F7iLznPGYX4BBFmN0ub5ABPm at 192.168.1.100 To: <sip:voipswitch at 192.168.1.100:5061>;tag=25065209175132503031 Contact: <sip:192.168.1.100:5061;transport=udp> Content-Length: 0 05:53:21.7 Call (l:'xxxxxxxxx' r:'sip:voipswitch at 192.168.1.100:5061') - Call being terminated. Reasons: "OK", (code: 200) 05:54:01.7 RECEIVING FROM: 192.168.1.100:5061 INVITE sip:5002 at 192.168.1.101 SIP/2.0 CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.1.100:5061 From: <sip:voipswitch@192.168.1.100:5061>;tag=25065309174032551609 Call-ID: GWkz3EXbigJc3CEywl5rH5Ju7T5dWcbvaE at 192.168.1.100 To: <sip:5002 at 192.168.1.101> Contact: <sip:192.168.1.100:5061;transport=udp> Content-Type: application/sdp Content-Length: 211 v=0 o=VoipSwitch 7298 7298 IN IP4 192.168.1.100 s=VoipSIP i=Audio Session c=IN IP4 192.168.1.100 t=0 0 m=audio 6298 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 05:54:02.3 Call (l:'xxxxxxxxx' r:'sip:voipswitch at 192.168.1.100:5061') - Incoming call. 05:54:03.0 SENDING TO: 192.168.1.100:5061 SIP/2.0 180 Ringing To: <sip:5002 at 192.168.1.101>;tag=a0386977 From: <sip:voipswitch@192.168.1.100:5061>;tag=25065309174032551609 Via: SIP/2.0/UDP 192.168.1.100:5061 Call-ID: GWkz3EXbigJc3CEywl5rH5Ju7T5dWcbvaE at 192.168.1.100 CSeq: 1 INVITE Contact: <sip:5002 at 192.168.1.101> Content-Length: 0 05:54:06.4 SENDING TO: 192.168.1.100:5061 SIP/2.0 200 OK To: <sip:5002 at 192.168.1.101>;tag=a0386977 From: <sip:voipswitch@192.168.1.100:5061>;tag=25065309174032551609 Via: SIP/2.0/UDP 192.168.1.100:5061 Call-ID: GWkz3EXbigJc3CEywl5rH5Ju7T5dWcbvaE at 192.168.1.100 CSeq: 1 INVITE Contact: <sip:5002 at 192.168.1.101> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Content-Length: 255 v=0 o=- 31110911 31112201 IN IP4 192.168.1.101 s=eyeBeam c=IN IP4 192.168.1.101 t=0 0 m=audio 6920 RTP/AVP 0 101 a=alt:1 1 : 03321852 00000045 192.168.1.101 6920 a=fmtp:101 0-15 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 05:54:07.0 RECEIVING FROM: 192.168.1.100:5061 ACK sip:5002 at 192.168.1.101 SIP/2.0 CSeq: 1 ACK Via: SIP/2.0/UDP 192.168.1.100:5061 From: <sip:voipswitch@192.168.1.100:5061>;tag=25065309174032551609 Call-ID: GWkz3EXbigJc3CEywl5rH5Ju7T5dWcbvaE at 192.168.1.100 To: <sip:5002 at 192.168.1.101>;tag=a0386977 Contact: <sip:192.168.1.100:5061;transport=udp> Content-Length: 0 05:54:11.7 SENDING TO: 192.168.1.100:5061 BYE sip:192.168.1.100:5061;transport=udp SIP/2.0 To: <sip:voipswitch at 192.168.1.100:5061>;tag=25065309174032551609 From: <sip:5002@192.168.1.101>;tag=a0386977 Via: SIP/2.0/UDP 192.168.1.101:6913;branch=z9hG4bK-d87543-865019333-1--d87543-;rport Call-ID: GWkz3EXbigJc3CEywl5rH5Ju7T5dWcbvaE at 192.168.1.100 CSeq: 2 BYE Contact: <sip:5002 at 192.168.1.101> Max-Forwards: 70 User-Agent: eyeBeam release 3007n stamp 17816 Content-Length: 0 05:54:12.3 RECEIVING FROM: 192.168.1.100:5061 SIP/2.0 200 OK CSeq: 2 BYE Via: SIP/2.0/UDP 192.168.1.101:6913;branch=z9hG4bK-d87543-865019333-1--d87543-;rport From: <sip:5002@192.168.1.101>;tag=a0386977 Call-ID: GWkz3EXbigJc3CEywl5rH5Ju7T5dWcbvaE at 192.168.1.100 To: <sip:voipswitch at 192.168.1.100:5061>;tag=25065309174032551609 Contact: <sip:192.168.1.100:5061;transport=udp> Content-Length: 0 05:54:12.9 Call (l:'xxxxxxxxx' r:'sip:voipswitch at 192.168.1.100:5061') - Call being terminated. Reasons: "OK", (code: 200) 05:54:19.0 RECEIVING FROM: 192.168.1.100:5061 INVITE sip:5002 at 192.168.1.101 SIP/2.0 CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.1.100:5061 From: <sip:voipswitch@192.168.1.100:5061>;tag=25065309175732568906 Call-ID: GWPbAcS70F3ta.nLKx7QskKFmSWFuuhhaM at 192.168.1.100 To: <sip:5002 at 192.168.1.101> Contact: <sip:192.168.1.100:5061;transport=udp> Content-Type: application/sdp Content-Length: 211 v=0 o=VoipSwitch 7302 7302 IN IP4 192.168.1.100 s=VoipSIP i=Audio Session c=IN IP4 192.168.1.100 t=0 0 m=audio 6302 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 05:54:19.5 Call (l:'xxxxxxxxx' r:'sip:voipswitch at 192.168.1.100:5061') - Incoming call. 05:54:20.3 SENDING TO: 192.168.1.100:5061 SIP/2.0 180 Ringing To: <sip:5002 at 192.168.1.101>;tag=e9570569 From: <sip:voipswitch@192.168.1.100:5061>;tag=25065309175732568906 Via: SIP/2.0/UDP 192.168.1.100:5061 Call-ID: GWPbAcS70F3ta.nLKx7QskKFmSWFuuhhaM at 192.168.1.100 CSeq: 1 INVITE Contact: <sip:5002 at 192.168.1.101> Content-Length: 0 05:54:22.5 SENDING TO: 192.168.1.100:5061 SIP/2.0 200 OK To: <sip:5002 at 192.168.1.101>;tag=e9570569 From: <sip:voipswitch@192.168.1.100:5061>;tag=25065309175732568906 Via: SIP/2.0/UDP 192.168.1.100:5061 Call-ID: GWPbAcS70F3ta.nLKx7QskKFmSWFuuhhaM at 192.168.1.100 CSeq: 1 INVITE Contact: <sip:5002 at 192.168.1.101> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Content-Length: 255 v=0 o=- 31128282 31129460 IN IP4 192.168.1.101 s=eyeBeam c=IN IP4 192.168.1.101 t=0 0 m=audio 6920 RTP/AVP 0 101 a=alt:1 1 : 1C489F77 0000005E 192.168.1.101 6920 a=fmtp:101 0-15 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 05:54:23.1 RECEIVING FROM: 192.168.1.100:5061 ACK sip:5002 at 192.168.1.101 SIP/2.0 CSeq: 1 ACK Via: SIP/2.0/UDP 192.168.1.100:5061 From: <sip:voipswitch@192.168.1.100:5061>;tag=25065309175732568906 Call-ID: GWPbAcS70F3ta.nLKx7QskKFmSWFuuhhaM at 192.168.1.100 To: <sip:5002 at 192.168.1.101>;tag=e9570569 Contact: <sip:192.168.1.100:5061;transport=udp> Content-Length: 0 05:54:28.7 SENDING TO: 192.168.1.100:5061 BYE sip:192.168.1.100:5061;transport=udp SIP/2.0 To: <sip:voipswitch at 192.168.1.100:5061>;tag=25065309175732568906 From: <sip:5002@192.168.1.101>;tag=e9570569 Via: SIP/2.0/UDP 192.168.1.101:6913;branch=z9hG4bK-d87543-366831874-1--d87543-;rport Call-ID: GWPbAcS70F3ta.nLKx7QskKFmSWFuuhhaM at 192.168.1.100 CSeq: 2 BYE Contact: <sip:5002 at 192.168.1.101> Max-Forwards: 70 User-Agent: eyeBeam release 3007n stamp 17816 Content-Length: 0 05:54:29.7 RECEIVING FROM: 192.168.1.100:5061 SIP/2.0 200 OK CSeq: 2 BYE Via: SIP/2.0/UDP 192.168.1.101:6913;branch=z9hG4bK-d87543-366831874-1--d87543-;rport From: <sip:5002@192.168.1.101>;tag=e9570569 Call-ID: GWPbAcS70F3ta.nLKx7QskKFmSWFuuhhaM at 192.168.1.100 To: <sip:voipswitch at 192.168.1.100:5061>;tag=25065309175732568906 Contact: <sip:192.168.1.100:5061;transport=udp> Content-Length: 0 05:54:30.4 Call (l:'xxxxxxxxx' r:'sip:voipswitch at 192.168.1.100:5061') - Call being terminated. Reasons: "OK", (code: 200)
