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<DIV dir=ltr><FONT face=Verdana color=#000000 size=2>Thanks ,a programmer of packetrino have requested a trace I am waiting for there answer.</FONT></DIV>
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<FONT face=Tahoma size=2><B>From:</B> Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro]<BR><B>Sent:</B> Sat 02/05/2009 8:17 AM<BR><B>To:</B> Julien Chavanton<BR><B>Cc:</B> users@lists.opensips.org<BR><B>Subject:</B> Re: [OpenSIPS-Users] handling multiple proxy / Record-Route<BR></FONT><BR></DIV>
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<P><FONT size=2>I think the only thing you can do is (if this path is fixed), to simply<BR>ignore the Route headers and to do a static routing.<BR><BR>Regards,<BR>Bogdan<BR><BR>Julien Chavanton wrote:<BR>> I think I will try the option to use the "textops" module to enforce<BR>> the correct order of Record-Route to validate this is my problem etc.<BR>> <BR>> <BR>><BR>> ------------------------------------------------------------------------<BR>> *From:* users-bounces@lists.opensips.org on behalf of Julien Chavanton<BR>> *Sent:* Thu 30/04/2009 3:44 PM<BR>> *To:* Bogdan-Andrei Iancu<BR>> *Cc:* users@lists.opensips.org<BR>> *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route<BR>><BR>> thank you, this is a problem as I do not control this proxy (2.2.2.2),<BR>> is there a suggested way of handling this problem ?<BR>> <BR>> Maybe there is something esle wrong on my side cusaing this problem so<BR>> I am including the SIP communication between the proxy this time<BR>> <BR>> <BR>> <BR>> #<BR>> U 1.1.1.1:5060 -> 2.2.2.2:5060<BR>> INVITE sip:15148622633@2.2.2.2 SIP/2.0.<BR>> Record-Route: <sip:1.1.1.1;lr>.<BR>> Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK09e6.36a0f975.0.<BR>> Via: SIP/2.0/UDP<BR>> 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366.<BR>> Max-Forwards: 69.<BR>> Contact: <sip:777@10.0.1.74:58366>.<BR>> To: "15141234567"<sip:15148622633@osip.dev.com>.<BR>> From: "777"<sip:777@osip.dev.com>;tag=a030735d.<BR>> Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..<BR>> CSeq: 1 INVITE.<BR>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,<BR>> SUBSCRIBE, INFO.<BR>> Content-Type: application/sdp.<BR>> User-Agent: eyeBeam release 1003s stamp 31159.<BR>> Content-Length: 478.<BR>> P-hint: Route[6]: mediaproxy .<BR>> .<BR>> v=0.<BR>> o=- 8 2 IN IP4 10.0.1.74.<BR>> s=CounterPath eyeBeam 1.5.<BR>> c=IN IP4 1.1.1.1.<BR>> t=0 0.<BR>> m=audio 52550 RTP/AVP 0 8 18 101.<BR>> a=alt:1 4 : LM6OZaAl 4x8r9qea 192.168.1.101 50006.<BR>> a=alt:2 3 : 84SVypDj oi4PbxZ7 192.168.114.1 50006.<BR>> a=alt:3 2 : L4wf6+MH s4gK5GAV 192.168.146.1 50006.<BR>> a=alt:4 1 : cg2pbkCG WDFvj29+ 10.0.1.74 50006.<BR>> a=fmtp:18 annexb=no.<BR>> a=fmtp:101 0-15.<BR>> a=rtpmap:101 telephone-event/8000.<BR>> a=sendrecv.<BR>> a=x-rtp-session-id:D56BCBC26473491FA111854E4C9F3575.<BR>> a=direction:active.<BR>> #<BR>> U 2.2.2.2:5060 -> 1.1.1.1:5060<BR>> SIP/2.0 100 Trying.<BR>> Via: SIP/2.0/UDP<BR>> 1.1.1.1:5060;branch=z9hG4bK09e6.36a0f975.0;received=1.1.1.1;rport=5060.<BR>> Via: SIP/2.0/UDP<BR>> 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=58366.<BR>> To: "15141234567" <sip:15141234567@osip.dev.com>.<BR>> From: "777" <sip:777@osip.dev.com>;tag=a030735d.<BR>> Call-ID: 8116f933cc4ea03fMjYzN2Q1MGQ5Y2M1ZDc5Yzk4OTRjN2Y5YzEwYWMwMzc..<BR>> CSeq: 1 INVITE.<BR>> Contact: <sip:15141234567@2.2.2.2>.<BR>> Content-Length: 0.<BR>> Record-Route: <sip:1.1.1.1;lr>.<BR>> User-Agent: Packetrino.<BR>> Supported: replaces.<BR>> Record-Route: <sip:2.2.2.2:5060;lr>.<BR>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<BR>> .<BR>> <BR>><BR>> ------------------------------------------------------------------------<BR>> *From:* Bogdan-Andrei Iancu [<A href="mailto:bogdan@voice-system.ro">mailto:bogdan@voice-system.ro</A>]<BR>> *Sent:* Thu 30/04/2009 3:44 PM<BR>> *To:* Julien Chavanton<BR>> *Cc:* users@lists.opensips.org<BR>> *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route<BR>><BR>> Hi Julian,<BR>><BR>> Julien Chavanton wrote:<BR>> ><BR>> ><BR>> > UA --> PROXY 1.1.1.1 --> PROXY 2.2.2.2 --> UA<BR>> ><BR>> > P1 --> P2<BR>> > INVITE<BR>> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes><BR>> ><BR>> > P2 --> P1<BR>> > 100 Trying<BR>> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes><BR>> > Record-Route: <sip:2.2.2.2:5060;lr><BR>> ><BR>> ^^^^^^^^^^^^<BR>><BR>> This is not correct. The RR of P2 most me on top of RR of P1 - adding RR<BR>> headers works as a stack.<BR>><BR>> Regards,<BR>> Bogdan<BR>> ><BR>> > Is there something wrong ? shouldn't proxy 2.2.2.2 add his<BR>> > Record-Route on top of the existing Record-Route ?<BR>> ><BR>> > ------------------------------------------------------------------------<BR>> > *From:* Bogdan-Andrei Iancu [<A href="mailto:bogdan@voice-system.ro">mailto:bogdan@voice-system.ro</A>]<BR>> > *Sent:* Thu 30/04/2009 8:12 AM<BR>> > *To:* Julien Chavanton<BR>> > *Cc:* users@lists.opensips.org<BR>> > *Subject:* Re: [OpenSIPS-Users] handling multiple proxy / Record-Route<BR>> ><BR>> > Hi Julien,<BR>> ><BR>> > I think Asterisk is doing the job properly. As you see the 200 OK has:<BR>> > Contact: <sip:15141234567@2.2.2.2:5060>.<BR>> > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.<BR>> > Record-Route: <sip:2.2.2.2:5060;lr>.<BR>> ><BR>> > So, Asterisk is generating the ACK with the Contact in RURI and the<BR>> > Route set in the reverted order (correct loose routing).<BR>> > -> RURI: sip:15141234567@2.2.2.2:5060<BR>> > Destination: sip:2.2.2.2:5060;lr<BR>> > Route: sip:2.2.2.2:5060;lr + sip:1.1.1.1;lr=on;nat=yes<BR>> ><BR>> > I think the problem here is who and why adding the bottom RR in 200 OK<BR>> > (why 2 of them ?)<BR>> ><BR>> > Regards,<BR>> > Bogdan<BR>> ><BR>> > Julien Chavanton wrote:<BR>> > ><BR>> > > Hi,<BR>> > ><BR>> > > I have a situation whit multiple proxy where ACK is not sent as I<BR>> > > would expect.<BR>> > ><BR>> > > if we look at the following "200 OK", I am expecting ACK to be sent to<BR>> > > 1.1.1.1 but the "Asterisk PBX 1.6.0.6." is selecting 2.2.2.2 is this<BR>> > > normal ?<BR>> > ><BR>> > > Do I have to handle Record-Route differently ?<BR>> > ><BR>> > ><BR>> > ><BR>> > ><BR>> > ><BR>> > > U 1.1.1.1:5060 -> 192.168.1.108:5060<BR>> > > SIP/2.0 200 OK.<BR>> > > Via: SIP/2.0/UDP<BR>> > ><BR>> ><BR>> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.<BR>> > > To: <sip:15141234567@osip.dev.com>;tag=as664de2c2.<BR>> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.<BR>> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108<BR>> > > <<A href="mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108">mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108</A>>.<BR>> > > CSeq: 102 INVITE.<BR>> > > Content-Type: application/sdp.<BR>> > > Contact: <sip:15141234567@2.2.2.2:5060>.<BR>> > > Content-Length: 241.<BR>> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.<BR>> > > User-Agent: Packetrino.<BR>> > > Supported: replaces.<BR>> > > Record-Route: <sip:2.2.2.2:5060;lr>.<BR>> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<BR>> > ><BR>> > ><BR>> > ><BR>> > ><BR>> > ><BR>> > ><BR>> > ><BR>> > ><BR>> > ><BR>> > > ---------------------------------------------------------<BR>> > ><BR>> > > complete SIP signaling<BR>> > ><BR>> > > ---------------------------------------------------------<BR>> > ><BR>> > > #<BR>> > > U 192.168.1.108:5060 -> 1.1.1.1:5060<BR>> > > INVITE sip:15141234567@osip.dev.com SIP/2.0.<BR>> > > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport.<BR>> > > Max-Forwards: 70.<BR>> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.<BR>> > > To: <sip:15141234567@osip.dev.com>.<BR>> > > Contact: <sip:15141234567@192.168.1.108>.<BR>> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108<BR>> > > <<A href="mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108">mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108</A>>.<BR>> > > CSeq: 102 INVITE.<BR>> > > User-Agent: Asterisk PBX 1.6.0.6.<BR>> > > Date: Wed, 29 Apr 2009 15:38:18 GMT.<BR>> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<BR>> > > Supported: replaces, timer.<BR>> > > Content-Type: application/sdp.<BR>> > > Content-Length: 265.<BR>> > > .<BR>> > > v=0.<BR>> > > o=root 1992389746 1992389746 IN IP4 192.168.1.108.<BR>> > > s=Asterisk PBX 1.6.0.6.<BR>> > > c=IN IP4 192.168.1.108.<BR>> > > t=0 0.<BR>> > > m=audio 11232 RTP/AVP 0 101.<BR>> > > a=rtpmap:0 PCMU/8000.<BR>> > > a=rtpmap:101 telephone-event/8000.<BR>> > > a=fmtp:101 0-16.<BR>> > > a=silenceSupp:off - - - -.<BR>> > > a=ptime:20.<BR>> > > a=sendrecv.<BR>> > ><BR>> > > #<BR>> > > U 1.1.1.1:5060 -> 192.168.1.108:5060<BR>> > > SIP/2.0 100 Giving a try.<BR>> > > Via: SIP/2.0/UDP<BR>> > ><BR>> ><BR>> 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport=5060;received=74.56.45.88.<BR>> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.<BR>> > > To: <sip:15141234567@osip.dev.com>.<BR>> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108<BR>> > > <<A href="mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108">mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108</A>>.<BR>> > > CSeq: 102 INVITE.<BR>> > > Server: OpenSIPS (1.4.4-notls (x86_64/linux)).<BR>> > > Content-Length: 0.<BR>> > > .<BR>> > ><BR>> > > #<BR>> > > U 1.1.1.1:5060 -> 192.168.1.108:5060<BR>> > > SIP/2.0 183 Session Progress.<BR>> > > Via: SIP/2.0/UDP<BR>> > ><BR>> ><BR>> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.<BR>> > > To: <sip:15141234567@osip.dev.com>;tag=as664de2c2.<BR>> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.<BR>> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108<BR>> > > <<A href="mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108">mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108</A>>.<BR>> > > CSeq: 102 INVITE.<BR>> > > Content-Type: application/sdp.<BR>> > > Contact: <sip:15141234567@2.2.2.2:5060>.<BR>> > > Content-Length: 241.<BR>> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.<BR>> > > User-Agent: Packetrino.<BR>> > > Supported: replaces.<BR>> > > Record-Route: <sip:2.2.2.2:5060;lr>.<BR>> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<BR>> > > .<BR>> > > v=0.<BR>> > > o=root 29378 29378 IN IP4 64.2.142.160.<BR>> > > s=session.<BR>> > > c=IN IP4 1.1.1.1.<BR>> > > t=0 0.<BR>> > > m=audio 52528 RTP/AVP 0 101.<BR>> > > a=rtpmap:0 PCMU/8000.<BR>> > > a=rtpmap:101 telephone-event/8000.<BR>> > > a=fmtp:101 0-16.<BR>> > > a=silenceSupp:off - - - -.<BR>> > > a=ptime:20.<BR>> > > a=sendrecv.<BR>> > ><BR>> > > #<BR>> > > U 1.1.1.1:5060 -> 192.168.1.108:5060<BR>> > > SIP/2.0 180 Ringing.<BR>> > > Via: SIP/2.0/UDP<BR>> > ><BR>> ><BR>> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.<BR>> > > To: <sip:15141234567@osip.dev.com>;tag=as664de2c2.<BR>> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.<BR>> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108<BR>> > > <<A href="mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108">mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108</A>>.<BR>> > > CSeq: 102 INVITE.<BR>> > > Contact: <sip:15141234567@2.2.2.2:5060>.<BR>> > > Content-Length: 0.<BR>> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.<BR>> > > User-Agent: Packetrino.<BR>> > > Supported: replaces.<BR>> > > Record-Route: <sip:2.2.2.2:5060;lr>.<BR>> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<BR>> > > .<BR>> > ><BR>> > > #<BR>> > > U 1.1.1.1:5060 -> 192.168.1.108:5060<BR>> > > SIP/2.0 200 OK.<BR>> > > Via: SIP/2.0/UDP<BR>> > ><BR>> ><BR>> 192.168.1.108:5060;received=74.56.45.88;branch=z9hG4bK2e975bf5;rport=5060.<BR>> > > To: <sip:15141234567@osip.dev.com>;tag=as664de2c2.<BR>> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.<BR>> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108<BR>> > > <<A href="mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108">mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108</A>>.<BR>> > > CSeq: 102 INVITE.<BR>> > > Content-Type: application/sdp.<BR>> > > Contact: <sip:15141234567@2.2.2.2:5060>.<BR>> > > Content-Length: 241.<BR>> > > Record-Route: <sip:1.1.1.1;lr=on;nat=yes>.<BR>> > > User-Agent: Packetrino.<BR>> > > Supported: replaces.<BR>> > > Record-Route: <sip:2.2.2.2:5060;lr>.<BR>> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<BR>> > > .<BR>> > > v=0.<BR>> > > o=root 29378 29379 IN IP4 64.2.142.160.<BR>> > > s=session.<BR>> > > c=IN IP4 1.1.1.1.<BR>> > > t=0 0.<BR>> > > m=audio 52528 RTP/AVP 0 101.<BR>> > > a=rtpmap:0 PCMU/8000.<BR>> > > a=rtpmap:101 telephone-event/8000.<BR>> > > a=fmtp:101 0-16.<BR>> > > a=silenceSupp:off - - - -.<BR>> > > a=ptime:20.<BR>> > > a=sendrecv.<BR>> > ><BR>> > > #<BR>> > > U 192.168.1.108:5060 -> 2.2.2.2:5060<BR>> > > ACK sip:15141234567@2.2.2.2:5060 SIP/2.0.<BR>> > > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK04335252;rport.<BR>> > > Route: <sip:2.2.2.2:5060;lr>,<sip:1.1.1.1;lr=on;nat=yes>.<BR>> > > Max-Forwards: 70.<BR>> > > From: "15141234567" <sip:15141234567@192.168.1.108>;tag=as55bd7355.<BR>> > > To: <sip:15141234567@osip.dev.com>;tag=as664de2c2.<BR>> > > Contact: <sip:15141234567@192.168.1.108>.<BR>> > > Call-ID: 641cab3f73fa37a871818d1a70c4061b@192.168.1.108<BR>> > > <<A href="mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108">mailto:641cab3f73fa37a871818d1a70c4061b@192.168.1.108</A>>.<BR>> > > CSeq: 102 ACK.<BR>> > > User-Agent: Asterisk PBX 1.6.0.6.<BR>> > > Content-Length: 0.<BR>> > > .<BR>> > ><BR>> > ><BR>> > ><BR>> ------------------------------------------------------------------------<BR>> > ><BR>> > > _______________________________________________<BR>> > > Users mailing list<BR>> > > Users@lists.opensips.org<BR>> > > <A href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</A><BR>> > ><BR>> ><BR>><BR><BR></FONT></P></DIV></BODY></HTML>