This would've been appropriate on sip-implementers mailer. The behavior below looks correct to me and has nothing to do with ftag field in RR header.
In the 1st case since UAS is getting a request with a RR header, next request on that dialog is sent to the address in RR header and since the proxy adding RR supports loose routing, request uri of BYE is contact address from previous request. In the 2nd case, since there are 2 RR headers, route set for UAS is: <sip:213.218.28.202;lr=on>, <sip:213.218.28.102;lr=on>. And the next request on that dialog is correctly sent to 213.218.28.202. Sanjay >-----Original Message----- >From: Frank, Marc [mailto:[EMAIL PROTECTED] >Sent: Tuesday, December 18, 2007 1:04 PM >To: [email protected] >Subject: [Sip] Record-Route-Header ftag-field and routing > >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-2ab0b6-496151951416-4 >96214255114 >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 > _______________________________________________ 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
