Hi guys, because of some problems with my SIP-Client, I have traced the traffic of a snom SIP-phone. Now I'm a little confused about the Record-Route header.
I've made two incoming testcalls with different providers. At the first test the snom-phone received an invite-message with record-route with ftag-field. When the phone hangup it sends it's BYE-Request to the IP-address defined in the last route-header-field. At the second test (2nd provider) the snom-phone received an invite-message with record-route and NO ftag-field. When the phone hangup it sends it's BYE-Request to the IP-address of the registrar. Is it correct, that a SIP-application which receives a record-route with ftag must send the traffic depending on this call to that destination? Regards Marc :-) Trace-Snapshot: Call 1 with re-routing -------------------------------------------------------------------------------- Received from udp:212.227.15.197:5060 at 18/12/2007 14:00:09:980 (1392 bytes): INVITE sip:aaaaa at xx.xxx.229.7:2054;line=j3eted9k SIP/2.0 Record-Route: <sip:212.227.15.232;ftag=1937134676;lr=on> Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bK59f8248d2c568739bc7dc8a7de9c4f82 Via: SIP/2.0/UDP 212.227.15.232;branch=z9hG4bK589.5362a815.0 Via: SIP/2.0/UDP 212.227.15.197;branch=z9hG4bK202ed21ce8126c48fd64f9f8458a705d Via: SIP/2.0/UDP sipgw01.bmcag.com:5060 ;received=62.206.6.140;branch=z9hG4bKterm-2ab0b6-496151951416-496214255114 From: caller <sip:caller at sipgw01.bmcag.com;user=phone>;tag=1937134676 To: called <sip:called at 212.227.15.197;user=phone> Contact: <sip:caller at sipgw01.bmcag.com:5060> ... Sent to udp:212.227.15.232:5060 at 18/12/2007 14:00:32:380 (695 bytes): BYE sip:caller at sipgw01.bmcag.com:5060 SIP/2.0 Via: SIP/2.0/UDP xx.xxx.229.7:2054;branch=z9hG4bK-54bqxvq49o3d;rport Route: <sip:212.227.15.232;ftag=1937134676;lr=on> From: "called" <sip:called at 212.227.15.197;user=phone>;tag=ckf1qvklvd To: "caller" <sip:caller at sipgw01.bmcag.com;user=phone>;tag=1937134676 Call-ID: 7e38a35-4274d41-7d136ce2-b115 at sipgw01.bmcag.com CSeq: 1 BYE -------------------------------------------------------------------------------- Call 2 without re-routing -------------------------------------------------------------------------------- Received from udp:213.218.28.202:5060 at 18/12/2007 18:02:31:170 (1422 bytes): INVITE sip:user at xx.xxx.229.7:2060;line=i8vjxde1 SIP/2.0 Record-Route: <sip:213.218.28.202;lr=on> Record-Route: <sip:213.218.28.102;lr=on> Via: SIP/2.0/UDP 213.218.28.202;branch=z9hG4bKbd69.7040b7b5.0 Via: SIP/2.0/UDP 213.218.28.102;branch=z9hG4bKbd69.b2ae7011.0 Via: SIP/2.0/UDP 213.218.28.164:5060;branch=z9hG4bK00E0F510055C3A From: "caller" <sip:caller at provider.net;user=phone>;tag=00E0F5100 To: <sip:called at provider.net;user=phone> Call-ID: [EMAIL PROTECTED] CSeq: 34207 INVITE Contact: <sip:caller at ip> ... Sent to udp:213.218.28.202:5060 at 18/12/2007 18:02:43:310 (731 bytes): BYE sip:called at ip SIP/2.0 Via: SIP/2.0/UDP 80.152.229.7:2060;branch=z9hG4bK-t7jlwh3sb054;rport Route: <sip:213.218.28.202;lr=on> Route: <sip:213.218.28.102;lr=on> -------------------------------------------------------------------------------- _______________________________________________ Sip mailing list https://www1.ietf.org/mailman/listinfo/sip This list is for NEW development of the core SIP Protocol Use [EMAIL PROTECTED] for questions on current sip Use [EMAIL PROTECTED] for new developments on the application of sip
