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:[email protected]] > *Sent:* Thu 30/04/2009 8:12 AM > *To:* Julien Chavanton > *Cc:* [email protected] > *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:[email protected]: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:[email protected]: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:[email protected]>;tag=as664de2c2. > > From: "15141234567" <sip:[email protected]>;tag=as55bd7355. > > Call-ID: [email protected] > > <mailto:[email protected]>. > > CSeq: 102 INVITE. > > Content-Type: application/sdp. > > Contact: <sip:[email protected]: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:[email protected] SIP/2.0. > > Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK2e975bf5;rport. > > Max-Forwards: 70. > > From: "15141234567" <sip:[email protected]>;tag=as55bd7355. > > To: <sip:[email protected]>. > > Contact: <sip:[email protected]>. > > Call-ID: [email protected] > > <mailto:[email protected]>. > > 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:[email protected]>;tag=as55bd7355. > > To: <sip:[email protected]>. > > Call-ID: [email protected] > > <mailto:[email protected]>. > > 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:[email protected]>;tag=as664de2c2. > > From: "15141234567" <sip:[email protected]>;tag=as55bd7355. > > Call-ID: [email protected] > > <mailto:[email protected]>. > > CSeq: 102 INVITE. > > Content-Type: application/sdp. > > Contact: <sip:[email protected]: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:[email protected]>;tag=as664de2c2. > > From: "15141234567" <sip:[email protected]>;tag=as55bd7355. > > Call-ID: [email protected] > > <mailto:[email protected]>. > > CSeq: 102 INVITE. > > Contact: <sip:[email protected]: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:[email protected]>;tag=as664de2c2. > > From: "15141234567" <sip:[email protected]>;tag=as55bd7355. > > Call-ID: [email protected] > > <mailto:[email protected]>. > > CSeq: 102 INVITE. > > Content-Type: application/sdp. > > Contact: <sip:[email protected]: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:[email protected]: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:[email protected]>;tag=as55bd7355. > > To: <sip:[email protected]>;tag=as664de2c2. > > Contact: <sip:[email protected]>. > > Call-ID: [email protected] > > <mailto:[email protected]>. > > CSeq: 102 ACK. > > User-Agent: Asterisk PBX 1.6.0.6. > > Content-Length: 0. > > . > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > [email protected] > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
