Hi Bogdan Thank you for your reply, that is exactly how I understood it.
Here is another fuller trace for a better understanding of the problem, which I believe to be uac_nat_test. On both servers , when a 200 is received, uac_nat_test is returning true because it is finding the top via (next hop for reply) to be different to the source ip (previous hop of reply): Internet Protocol, Src: 1.1.1.1 (1.1.1.1), Dst: 2.2.2.2 (2.2.2.2) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) INVITE sip:[email protected] SIP/2.0 Record-Route: <sip:1.1.1.1;lr=on;ftag=as1a75bb38> Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK5be8.eeb63911.0 Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK55985828;rport=5060 From: "22222222222" <sip:[email protected]>;tag=as1a75bb38 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Tue, 18 May 2010 06:29:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-CRE-ID: "sbc-tp-04-1274164140.30823.0 X-DIALSTRING: SIP/ico-bry-001/1111111111 Content-Type: application/sdp Content-Length: 259 v=0 o=root 4296 4296 IN IP4 3.3.3.3 s=session c=IN IP4 3.3.3.3 t=0 0 m=audio 10060 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:60 a=sendrecv Internet Protocol, Src: 2.2.2.2 (2.2.2.2), Dst: 4.4.4.4 (4.4.4.4) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) INVITE sip:[email protected] SIP/2.0 Record-Route: <sip:2.2.2.2;lr=on;ftag=as1a75bb38> Record-Route: <sip:1.1.1.1;lr=on;ftag=as1a75bb38> Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK5be8.91dcf496.0 Via: SIP/2.0/UDP 1.1.1.1;rport=5060;received=1.1.1.1;branch=z9hG4bK5be8.eeb63911.0 Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK55985828;rport=5060 From: "2222222222" <sip:[email protected]>;tag=as1a75bb38 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 68 Date: Tue, 18 May 2010 06:29:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-CRE-ID: "sbc-tp-04-1274164140.30823.0 X-DIALSTRING: SIP/ico-bry-001/1111111111 Content-Type: application/sdp Content-Length: 259 v=0 o=root 4296 4296 IN IP4 3.3.3.3 s=session c=IN IP4 3.3.3.3 t=0 0 m=audio 10060 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:60 a=sendrecv Internet Protocol, Src: 4.4.4.4 (4.4.4.4), Dst: 2.2.2.2 (2.2.2.2) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) SIP/2.0 200 OK Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK5be8.91dcf496.0;received=2.2.2.2 Via: SIP/2.0/UDP 1.1.1.1;rport=5060;received=1.1.1.1;branch=z9hG4bK5be8.eeb63911.0 Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK55985828;rport=5060 Record-Route: <sip:2.2.2.2;lr=on;ftag=as1a75bb38> Record-Route: <sip:1.1.1.1;lr=on;ftag=as1a75bb38> From: "2222222222" <sip:[email protected]>;tag=as1a75bb38 To: <sip:[email protected]>;tag=as5bd164c9 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 290 v=0 o=root 2115801714 2115801714 IN IP4 4.4.4.4 s=Asterisk PBX 1.6.0.9 c=IN IP4 4.4.4.4 t=0 0 m=audio 12004 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:60 a=sendrecv Internet Protocol, Src: 2.2.2.2 (2.2.2.2), Dst: 1.1.1.1 (1.1.1.1) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) SIP/2.0 200 OK Via: SIP/2.0/UDP 1.1.1.1;rport=5060;received=1.1.1.1;branch=z9hG4bK5be8.eeb63911.0 Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK55985828;rport=5060 Record-Route: <sip:2.2.2.2;lr=on;ftag=as1a75bb38> Record-Route: <sip:1.1.1.1;lr=on;ftag=as1a75bb38> From: "2222222222" <sip:[email protected]>;tag=as1a75bb38 To: <sip:[email protected]>;tag=as5bd164c9 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 290 v=0 o=root 2115801714 2115801714 IN IP4 4.4.4.4 s=Asterisk PBX 1.6.0.9 c=IN IP4 4.4.4.4 t=0 0 m=audio 12004 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:60 a=sendrecv ACK is sent with 2.2.2.2 in ruri, instead of 4.4.4.4: Internet Protocol, Src: 1.1.1.1 (1.1.1.1), Dst: 2.2.2.2 (2.2.2.2) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) ACK sip:[email protected]:5060 SIP/2.0 Record-Route: <sip:1.1.1.1;lr=on;ftag=as1a75bb38> Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK5be8.eeb63911.2 Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK07b01eaf;rport=5060 Route: <sip:2.2.2.2;lr=on;ftag=as1a75bb38> From: "2222222222" <sip:[email protected]>;tag=as1a75bb38 To: <sip:[email protected]>;tag=as5bd164c9 Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 The ACK is not relayed on to 4.4.4.4, and so 4.4.4.4 just keeps retransmitting 200 replies. Later the BYE from 1.1.1.1 also has an incorrect ruri and so it is also not sent on to 4.4.4.4 as follows: Internet Protocol, Src: 1.1.1.1 (1.1.1.1), Dst: 2.2.2.2 (2.2.2.2) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) BYE sip:[email protected]:5060 SIP/2.0 Record-Route: <sip:1.1.1.1;lr=on;ftag=as1a75bb38> Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK6be8.2fa52b26.0 Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK02874f97;rport=5060 Route: <sip:2.2.2.2;lr=on;ftag=as1a75bb38> From: "2222222222" <sip:[email protected]>;tag=as1a75bb38 To: <sip:[email protected]>;tag=as5bd164c9 Call-ID: [email protected] CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 69 Reason: Q.850 ;cause=16; text="Normal Clearing" X-Asterisk-HangupCauseCode: 16 Content-Length: 0 I believe we would have been able to fix this issue by using the dialoq module on both servers, but I do not know much about the dialog module yet, and unfortunately we have no control over one of the opensips servers. i also thought of trying to use the b2bua modules on just our server, but once again i will first need to learn more about those, but for now, we have managed to create a workaround whereby we rewrite the ruri for all acks, byes and cancels with the ip retrieved from the location table (using avp_db_query), which seems to be working, for now. I hope to find a more reliable fix. Thanks for the help. Bruce ________________________________ From: Bogdan-Andrei Iancu <[email protected]> To: OpenSIPS users mailling list <[email protected]> Sent: Tue, 18 May, 2010 17:54:01 Subject: Re: [OpenSIPS-Users] In dialog requests misrouted Hi Bruce, The ACK for a 200OK is routed based on the route set - this the RR set + the contact of the other party. So, the ACK will have in RURI the contact of the other party (from 200 OK) and the RR set as Route hdrs. Regards, Bogdan Bruce Borrett wrote: > Hi > > We are trying to migrate from an SBC to Opensips 1.6. When we are > sending calls to another provider who are using Openser, they are not > taking the contact address from our 200 replies, instead they are > putting our Openser address in the RURI of all Acks, Byes and Cancels. > Am I right in saying that this is incorrect? Im not sure where they > are getting this address either, maybe from the To: field, or from the > record route header? > > Is there a way to match the message received to a transaction and > route the message to the contact in the original invite stored by TM? > Or perhaps some better way of solving this? > > Here are the messages: > > SIP/2.0 200 OK > Via: SIP/2.0/UDP > xx.xxx.0.33;rport=5060;received=41.221.0.33;branch=z9hG4bKb9fc.f1cf4e03.0 > Via: SIP/2.0/UDP xx.xxx.0.42:5060;branch=z9hG4bK3d3a5800;rport=5060 > Record-Route: <sip:xx.xxx.1.13;lr=on;ftag=as5e3b3ce0> > Record-Route: <sip:xx.xxx.0.33;lr=on;ftag=as5e3b3ce0> > From: "xxxxxxx7239" <sip:[email protected]>;tag=as5e3b3ce0 > To: <sip:[email protected]>;tag=as33b0f85f > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: Asterisk PBX 1.6.0.9 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces, timer > Contact: <sip:[email protected]> > Content-Type: application/sdp > Content-Length: 290 > > > ACK sip:[email protected]:5060 SIP/2.0 > Record-Route: <sip:xx.xxx.0.33;lr=on;ftag=as5e3b3ce0> > Via: SIP/2.0/UDP xx.xxx.0.33;branch=z9hG4bKb9fc.f1cf4e03.2 > Via: SIP/2.0/UDP xx.xxx.0.42:5060;branch=z9hG4bK5e0350ba;rport=5060 > Route: <sip:xx.xxx.1.13;lr=on;ftag=as5e3b3ce0> > From: "xxxxxxx7239" <sip:[email protected]>;tag=as5e3b3ce0 > To: <sip:[email protected]>;tag=as33b0f85f > Contact: <sip:[email protected]> > Call-ID: [email protected] > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 69 > Content-Length: 0 > > Thank you very much in advance.. > Bruce > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Bogdan-Andrei Iancu www.voice-system.ro _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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