<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>UA --> PROXY 1.1.1.1 --> PROXY 2.2.2.2 --> UA</FONT></DIV>
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<DIV dir=ltr><FONT face="Courier New" size=2>P1 --> P2</FONT></DIV>
<DIV dir=ltr><FONT face="Courier New" size=2>INVITE </FONT></DIV>
<DIV dir=ltr><FONT face="Courier New" size=2>Record-Route: <sip:1.1.1.1;lr=on;nat=yes></FONT></DIV>
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<DIV dir=ltr><FONT face="Courier New" size=2>P2 --> P1</FONT></DIV>
<DIV dir=ltr><FONT face="Courier New" size=2>100 Trying</FONT></DIV>
<DIV dir=ltr><FONT face="Courier New" size=2>Record-Route: <sip:1.1.1.1;lr=on;nat=yes></FONT></DIV>
<DIV dir=ltr><FONT face="Courier New" size=2>Record-Route: <sip:2.2.2.2:5060;lr></FONT></DIV>
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<DIV dir=ltr><FONT face="Courier New" size=2>Is there something wrong ? shouldn't proxy 2.2.2.2 add his Record-Route on top of the existing Record-Route ?</FONT></DIV></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 8:12 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>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>> 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>> 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>> 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>> 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>> 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></FONT></P></DIV></BODY></HTML>