<HTML dir=ltr><HEAD><TITLE>Re: [OpenSIPS-Users] handling multiple proxy / Record-Route</TITLE>
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<DIV dir=ltr><FONT face="Courier New" color=#000000 size=2>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 ?</FONT></DIV>
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<DIV dir=ltr><FONT face="Courier New" size=2>Maybe there is something esle wrong on my side cusaing this problem so I am including the SIP communication between the proxy this time</FONT></DIV></DIV>
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<DIV dir=ltr><FONT face="Courier New" size=2>#<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 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, 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.</FONT></DIV>
<DIV dir=ltr><FONT face="Courier New" size=2>#<BR>U 2.2.2.2:5060 -> 1.1.1.1:5060<BR>SIP/2.0 100 Trying.<BR>Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK09e6.36a0f975.0;received=1.1.1.1;rport=5060.<BR>Via: SIP/2.0/UDP 10.0.1.74:58366;received=10.0.1.74;branch=z9hG4bK-d87543-0f348609f47bda44-1--d87543-;rport=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>Contact: <sip:15148622633@64.2.142.75>.<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>.</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> Thu 30/04/2009 3:44 PM<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>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>> 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>> 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>> 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>> 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>> 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>> > 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></FONT></P></DIV></BODY></HTML>