[OpenSIPS-Users] In dialog requests misrouted

Fri May 21 12:57:37 CEST 2010

Hi Bruce,

It is a logical problem. The chain is:  3 -> 1 -> 2 -> 4, and  when the 
reply goes back, the NAT traversal must be done by the border entity 
(first in the public net). So, if 4 is behind nat, then 2 must do it and 
not 1 (like in your case)

Because 1 (and not 2) is "fixing" the contact , the routing info gets 
lost (the 4 hop is lost). So, guilty is  2 for not fixing the contact in 
200 OK from 4.

Regards,
Bogdan

Bruce Borrett wrote:
> 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:1111111111 at 2.2.2.2 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:2222222222 at 3.3.3.3>;tag=as1a75bb38
> To: <sip:1111111111 at 2.2.2.2>
> Contact: <sip:2222222222 at 3.3.3.3>
> Call-ID: 7a98e4540899dde2053dc8a11cee1a04 at 3.3.3.3
> 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:1111111111 at 4.4.4.4 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:2222222222 at 3.3.3.3>;tag=as1a75bb38
> To: <sip:1111111111 at 2.2.2.2>
> Contact: <sip:2222222222 at 3.3.3.3>
> Call-ID: 7a98e4540899dde2053dc8a11cee1a04 at 3.3.3.3
> 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:2222222222 at 3.3.3.3>;tag=as1a75bb38
> To: <sip:1111111111 at 2.2.2.2>;tag=as5bd164c9
> Call-ID: 7a98e4540899dde2053dc8a11cee1a04 at 3.3.3.3
> 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:1111111111 at 4.4.4.4>
> 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:2222222222 at 3.3.3.3>;tag=as1a75bb38
> To: <sip:1111111111 at 2.2.2.2>;tag=as5bd164c9
> Call-ID: 7a98e4540899dde2053dc8a11cee1a04 at 3.3.3.3
> 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:1111111111 at 4.4.4.4>
> 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:1111111111 at 2.2.2.2:5060 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:2222222222 at 3.3.3.3>;tag=as1a75bb38
> To: <sip:1111111111 at 2.2.2.2>;tag=as5bd164c9
> Contact: <sip:2222222222 at 3.3.3.3>
> Call-ID: 7a98e4540899dde2053dc8a11cee1a04 at 3.3.3.3
> 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:1111111111 at 2.2.2.2:5060 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:2222222222 at 3.3.3.3>;tag=as1a75bb38
> To: <sip:1111111111 at 2.2.2.2>;tag=as5bd164c9
> Call-ID: 7a98e4540899dde2053dc8a11cee1a04 at 3.3.3.3
> 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 <bogdan at voice-system.ro>
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>
> *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:xxxxxxx7239 at xx.xxx 
> <mailto:xxxxxxx7239 at xx.xxx>.0.42>;tag=as5e3b3ce0
> > To: <sip:xxxxxx0114 at xx.xxx 
> <mailto:xxxxxx0114 at xx.xxx>.1.13>;tag=as33b0f85f
> > Call-ID: 4b3419c86b29e5ce5a387a1b74f7effc at xx.xxx 
> <mailto:4b3419c86b29e5ce5a387a1b74f7effc at xx.xxx>.0.42
> > 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:xxxxxxx0114 at xx.xxx <mailto:xxxxxxx0114 at xx.xxx>.236.105>
> > Content-Type: application/sdp
> > Content-Length: 290
> >
> >
> > ACK sip:xxxxxx0114 at xx.xxx <mailto:xxxxxx0114 at xx.xxx>.1.13:5060 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:xxxxxxx7239 at xx.xxx 
> <mailto:xxxxxxx7239 at xx.xxx>.0.42>;tag=as5e3b3ce0
> > To: <sip:xxxxxx0114 at xx.xxx 
> <mailto:xxxxxx0114 at xx.xxx>.1.13>;tag=as33b0f85f
> > Contact: <sip:xxxxxx7239 at xx.xxx <mailto:xxxxxx7239 at xx.xxx>.0.42>
> > Call-ID: 4b3419c86b29e5ce5a387a1b74f7effc at xx.xxx 
> <mailto:4b3419c86b29e5ce5a387a1b74f7effc at xx.xxx>.0.42
> > CSeq: 102 ACK
> > User-Agent: Asterisk PBX
> > Max-Forwards: 69
> > Content-Length: 0
> >
> > Thank you very much in advance..
> > Bruce
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > 
>
>
> -- 
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
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>
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>
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>   

-- 
Bogdan-Andrei Iancu
www.voice-system.ro