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. .
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