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> 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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