I think I will try the option to use the "textops" module to enforce the correct order of Record-Route to validate this is my problem etc.
________________________________ From: users-boun...@lists.opensips.org on behalf of Julien Chavanton Sent: Thu 30/04/2009 3:44 PM To: Bogdan-Andrei Iancu Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] handling multiple proxy / Record-Route thank you, this is a problem as I do not control this proxy (2.2.2.2), is there a suggested way of handling this problem ? Maybe there is something esle wrong on my side cusaing this problem so I am including the SIP communication between the proxy this time # U 1.1.1.1:5060 -> 2.2.2.2:5060 INVITE sip:15148622...@2.2.2.2 SIP/2.0. Record-Route: <sip:1.1.1.1;lr>. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK09e6.36a0f975.0. Via: SIP/2.0/UDP 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366. Max-Forwards: 69. Contact: <sip:7...@10.0.1.74:58366>. To: "15141234567"<sip:15148622...@osip.dev.com>. From: "777"<sip:7...@osip.dev.com>;tag=a030735d. Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 1003s stamp 31159. Content-Length: 478. P-hint: Route[6]: mediaproxy . . v=0. o=- 8 2 IN IP4 10.0.1.74. s=CounterPath eyeBeam 1.5. c=IN IP4 1.1.1.1. t=0 0. m=audio 52550 RTP/AVP 0 8 18 101. a=alt:1 4 : LM6OZaAl 4x8r9qea 192.168.1.101 50006. a=alt:2 3 : 84SVypDj oi4PbxZ7 192.168.114.1 50006. a=alt:3 2 : L4wf6+MH s4gK5GAV 192.168.146.1 50006. a=alt:4 1 : cg2pbkCG WDFvj29+ 10.0.1.74 50006. a=fmtp:18 annexb=no. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=x-rtp-session-id:D56BCBC26473491FA111854E4C9F3575. a=direction:active. # U 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK09e6.36a0f975.0;received=1.1.1.1;rport=5060. Via: SIP/2.0/UDP 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366. To: "15141234567" <sip:15141234...@osip.dev.com>. From: "777" <sip:7...@osip.dev.com>;tag=a030735d. Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc.. CSeq: 1 INVITE. Contact: <sip:15141234...@2.2.2.2>. Content-Length: 0. Record-Route: <sip:1.1.1.1;lr>. User-Agent: Packetrino. Supported: replaces. Record-Route: <sip:2.2.2.2:5060;lr>. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. . ________________________________ From: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] Sent: Thu 30/04/2009 3:44 PM To: Julien Chavanton Cc: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] handling multiple proxy / Record-Route Hi Julian, Julien Chavanton wrote: > > > UA --> PROXY 1.1.1.1 --> PROXY 2.2.2.2 --> UA > > P1 --> P2 > INVITE > Record-Route: <sip:1.1.1.1;lr=on;nat=yes> > > P2 --> P1 > 100 Trying > Record-Route: <sip:1.1.1.1;lr=on;nat=yes> > Record-Route: <sip:2.2.2.2:5060;lr> > ^^^^^^^^^^^^ This is not correct. The RR of P2 most me on top of RR of P1 - adding RR headers works as a stack. Regards, Bogdan > > Is there something wrong ? shouldn't proxy 2.2.2.2 add his > Record-Route on top of the existing Record-Route ? > > ------------------------------------------------------------------------ > *From:* Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro] > *Sent:* Thu 30/04/2009 8:12 AM > *To:* Julien Chavanton > *Cc:* users@lists.opensips.org > *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route > > Hi Julien, > > I think Asterisk is doing the job properly. As you see the 200 OK has: > Contact: <sip:15141234...@2.2.2.2:5060>. > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>. > Record-Route: <sip:2.2.2.2:5060;lr>. > > So, Asterisk is generating the ACK with the Contact in RURI and the > Route set in the reverted order (correct loose routing). > -> RURI: sip:15141234...@2.2.2.2:5060 > Destination: sip:2.2.2.2:5060;lr > Route: sip:2.2.2.2:5060;lr + sip:1.1.1.1;lr=on;nat=yes > > I think the problem here is who and why adding the bottom RR in 200 OK > (why 2 of them ?) > > Regards, > Bogdan > > Julien Chavanton wrote: > > > > Hi, > > > > I have a situation whit multiple proxy where ACK is not sent as I > > would expect. > > > > if we look at the following "200 OK", I am expecting ACK to be sent to > > 1.1.1.1 but the "Asterisk PBX 1.6.0.6." is selecting 2.2.2.2 is this > > normal ? > > > > Do I have to handle Record-Route differently ? > > > > > > > > > > > > U 1.1.1.1:5060 -> 192.168.1.108:5060 > > SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > > 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060. > > To: <sip:15141234...@osip.dev.com>;tag=as664de2c2. > > From: "15141234567" <sip:15141234...@192.168.1.108>;tag=as55bd7355. > > Call-ID: 641cab3f73fa37a871818d1a70c40...@192.168.1.108 > > <mailto:641cab3f73fa37a871818d1a70c40...@192.168.1.108>. > > CSeq: 102 INVITE. > > Content-Type: application/sdp. > > Contact: <sip:15141234...@2.2.2.2:5060>. > > Content-Length: 241. > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>. > > User-Agent: Packetrino. > > Supported: replaces. > > Record-Route: <sip:2.2.2.2:5060;lr>. > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > > > > > > > > > > > > > > > > > > > > --------------------------------------------------------- > > > > complete SIP signaling > > > > --------------------------------------------------------- > > > > # > > U 192.168.1.108:5060 -> 1.1.1.1:5060 > > INVITE sip:15141234...@osip.dev.com SIP/2.0. > > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport. > > Max-Forwards: 70. > > From: "15141234567" <sip:15141234...@192.168.1.108>;tag=as55bd7355. > > To: <sip:15141234...@osip.dev.com>. > > Contact: <sip:15141234...@192.168.1.108>. > > Call-ID: 641cab3f73fa37a871818d1a70c40...@192.168.1.108 > > <mailto:641cab3f73fa37a871818d1a70c40...@192.168.1.108>. > > CSeq: 102 INVITE. > > User-Agent: Asterisk PBX 1.6.0.6. > > Date: Wed, 29 Apr 2009 15:38:18 GMT. > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > > Supported: replaces, timer. > > Content-Type: application/sdp. > > Content-Length: 265. > > . > > v=0. > > o=root 1992389746 1992389746 IN IP4 192.168.1.108. > > s=Asterisk PBX 1.6.0.6. > > c=IN IP4 192.168.1.108. > > t=0 0. > > m=audio 11232 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-16. > > a=silenceSupp:off - - - -. > > a=ptime:20. > > a=sendrecv. > > > > # > > U 1.1.1.1:5060 -> 192.168.1.108:5060 > > SIP/2.0 100 Giving a try. > > Via: SIP/2.0/UDP > > > 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport=5060;received=74.56.45.88. > > From: "15141234567" <sip:15141234...@192.168.1.108>;tag=as55bd7355. > > To: <sip:15141234...@osip.dev.com>. > > Call-ID: 641cab3f73fa37a871818d1a70c40...@192.168.1.108 > > <mailto:641cab3f73fa37a871818d1a70c40...@192.168.1.108>. > > CSeq: 102 INVITE. > > Server: OpenSIPS (1.4.4-notls (x86_64/linux)). > > Content-Length: 0. > > . > > > > # > > U 1.1.1.1:5060 -> 192.168.1.108:5060 > > SIP/2.0 183 Session Progress. > > Via: SIP/2.0/UDP > > > 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060. > > To: <sip:15141234...@osip.dev.com>;tag=as664de2c2. > > From: "15141234567" <sip:15141234...@192.168.1.108>;tag=as55bd7355. > > Call-ID: 641cab3f73fa37a871818d1a70c40...@192.168.1.108 > > <mailto:641cab3f73fa37a871818d1a70c40...@192.168.1.108>. > > CSeq: 102 INVITE. > > Content-Type: application/sdp. > > Contact: <sip:15141234...@2.2.2.2:5060>. > > Content-Length: 241. > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>. > > User-Agent: Packetrino. > > Supported: replaces. > > Record-Route: <sip:2.2.2.2:5060;lr>. > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > > . > > v=0. > > o=root 29378 29378 IN IP4 64.2.142.160. > > s=session. > > c=IN IP4 1.1.1.1. > > t=0 0. > > m=audio 52528 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-16. > > a=silenceSupp:off - - - -. > > a=ptime:20. > > a=sendrecv. > > > > # > > U 1.1.1.1:5060 -> 192.168.1.108:5060 > > SIP/2.0 180 Ringing. > > Via: SIP/2.0/UDP > > > 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060. > > To: <sip:15141234...@osip.dev.com>;tag=as664de2c2. > > From: "15141234567" <sip:15141234...@192.168.1.108>;tag=as55bd7355. > > Call-ID: 641cab3f73fa37a871818d1a70c40...@192.168.1.108 > > <mailto:641cab3f73fa37a871818d1a70c40...@192.168.1.108>. > > CSeq: 102 INVITE. > > Contact: <sip:15141234...@2.2.2.2:5060>. > > Content-Length: 0. > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>. > > User-Agent: Packetrino. > > Supported: replaces. > > Record-Route: <sip:2.2.2.2:5060;lr>. > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > > . > > > > # > > U 1.1.1.1:5060 -> 192.168.1.108:5060 > > SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > > 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060. > > To: <sip:15141234...@osip.dev.com>;tag=as664de2c2. > > From: "15141234567" <sip:15141234...@192.168.1.108>;tag=as55bd7355. > > Call-ID: 641cab3f73fa37a871818d1a70c40...@192.168.1.108 > > <mailto:641cab3f73fa37a871818d1a70c40...@192.168.1.108>. > > CSeq: 102 INVITE. > > Content-Type: application/sdp. > > Contact: <sip:15141234...@2.2.2.2:5060>. > > Content-Length: 241. > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>. > > User-Agent: Packetrino. > > Supported: replaces. > > Record-Route: <sip:2.2.2.2:5060;lr>. > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > > . > > v=0. > > o=root 29378 29379 IN IP4 64.2.142.160. > > s=session. > > c=IN IP4 1.1.1.1. > > t=0 0. > > m=audio 52528 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-16. > > a=silenceSupp:off - - - -. > > a=ptime:20. > > a=sendrecv. > > > > # > > U 192.168.1.108:5060 -> 2.2.2.2:5060 > > ACK sip:15141234...@2.2.2.2:5060 SIP/2.0. > > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK04335252;rport. > > Route: <sip:2.2.2.2:5060;lr>,<sip:1.1.1.1;lr=on;nat=yes>. > > Max-Forwards: 70. > > From: "15141234567" <sip:15141234...@192.168.1.108>;tag=as55bd7355. > > To: <sip:15141234...@osip.dev.com>;tag=as664de2c2. > > Contact: <sip:15141234...@192.168.1.108>. > > Call-ID: 641cab3f73fa37a871818d1a70c40...@192.168.1.108 > > <mailto:641cab3f73fa37a871818d1a70c40...@192.168.1.108>. > > CSeq: 102 ACK. > > User-Agent: Asterisk PBX 1.6.0.6. > > Content-Length: 0. > > . > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users@lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >
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