Hi all! Please tell me know if this behaviour is intentional: Problem with proxying rtp: UAC receives ip in the UAS' subnet while UAS receives the ip of UAC's subnet of rtp proxy by default.
IP-Phone is on 10.200.10.195. Network configuration: ast1.local <-----------> opensips+rtpproxy <---------> ast2.local 192.168.56.3 192.168.56.2/192.168.58.2 192.168.58.3 10.200.10.something for ip phone. SIP: ast1 configured with outboundproxy .56.2 ast2 configured with outboundproxy .58.2 no ip routing is done on the ALG OpenSIPS 1.6.4: configured to rtpproxy_offer(); on INVITE and to rtpproxy_answer(); on reply to it. rtpproxy 1.2.1: in bridge mode 192.168.56.2/192.168.58.2 SIP works fine: <ast1> Reliably Transmitting (no NAT) to 192.168.56.2:5060: OPTIONS sip:ast2.local SIP/2.0 Via: SIP/2.0/UDP 192.168.56.3:5060;branch=z9hG4bK2a985f57;rport From: "asterisk" <sip:asterisk@192.168.56.3>;tag=as7d368e85 To: <sip:ast2.local> Contact: <sip:asterisk@192.168.56.3> Call-ID: 1a0cd5082af8bd525e2071c10edf2920@192.168.56.3 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 Mar 2011 21:54:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <ast2> <--- Transmitting (no NAT) to 192.168.58.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.58.2;branch=z9hG4bK5054.135a9ff1.0;received=192.168.58.2 Via: SIP/2.0/UDP 192.168.56.3:5060;received=192.168.56.3;branch=z9hG4bK2a985f57;rport=5060 Record-Route: <sip:192.168.58.2;r2=on;lr=on> Record-Route: <sip:192.168.56.2;r2=on;lr=on> From: "asterisk" <sip:asterisk@192.168.56.3>;tag=as7d368e85 To: <sip:ast2.local>;tag=as63b00332 Call-ID: 1a0cd5082af8bd525e2071c10edf2920@192.168.56.3 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:192.168.58.3> Accept: application/sdp Content-Length: 0 THE CALL: from ast1 to ast2 <ast1> <------------> -- Executing [565656@incoming:1] Dial("SIP/bratner-000000a9", "SIP/ast2/565656") in new stack Audio is at 192.168.56.3 port 18922 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.56.2:5060: INVITE sip:565656@ast2.local SIP/2.0 Via: SIP/2.0/UDP 192.168.56.3:5060;branch=z9hG4bK391f49a8;rport From: "Extension 1001" <sip:bratner@192.168.56.3>;tag=as6826385c To: <sip:565656@ast2.local> Contact: <sip:bratner@192.168.56.3> Call-ID: 2d167b8f4108d4467ff3ea4e1cff7218@192.168.56.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 Mar 2011 21:58:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 19589 19589 IN IP4 192.168.56.3 s=session c=IN IP4 192.168.56.3 t=0 0 m=audio 18922 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <RTP PROXY> DBUG:handle_command: received command "U 2d167b8f4108d4467ff3ea4e1cff7218@192.168.56.3 192.168.56.3 18922 as6826385c;1" INFO:handle_command: new session 2d167b8f4108d4467ff3ea4e1cff7218@192.168.56.3, tag as6826385c;1 requested, type strong INFO:handle_command: BRAT: given remote address 192.168.56.3 INFO:create_twinlistener: BINDING TO 0.0.0.0 INFO:create_twinlistener: BINDING TO 0.0.0.0 INFO:handle_command: new session on a port 50026 created, tag as6826385c;1 INFO:handle_command: pre-filling caller's address with 192.168.56.3:18922 DBUG:doreply: sending reply "50026" <ast2> <--- SIP read from 192.168.58.2:5060 ---> INVITE sip:565656@ast2.local SIP/2.0 Record-Route: <sip:192.168.58.2;r2=on;lr=on> Record-Route: <sip:192.168.56.2;r2=on;lr=on> Via: SIP/2.0/UDP 192.168.58.2;branch=z9hG4bKc05.3917bd94.0 Via: SIP/2.0/UDP 192.168.56.3:5060;received=192.168.56.3;branch=z9hG4bK391f49a8;rport=5060 From: "Extension 1001" <sip:bratner@192.168.56.3>;tag=as6826385c To: <sip:565656@ast2.local> Contact: <sip:bratner@192.168.56.3> Call-ID: 2d167b8f4108d4467ff3ea4e1cff7218@192.168.56.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Mon, 07 Mar 2011 21:58:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 305 P-hint: thehelldoiknow v=0 o=root 19589 19589 IN IP4 192.168.56.3 s=session c=IN IP4 192.168.56.2 t=0 0 m=audio 50026 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes <--- Reliably Transmitting (no NAT) to 192.168.58.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.58.2;branch=z9hG4bKc05.3917bd94.0;received=192.168.58.2 Via: SIP/2.0/UDP 192.168.56.3:5060;received=192.168.56.3;branch=z9hG4bK391f49a8;rport=5060 Record-Route: <sip:192.168.58.2;r2=on;lr=on> Record-Route: <sip:192.168.56.2;r2=on;lr=on> From: "Extension 1001" <sip:bratner@192.168.56.3>;tag=as6826385c To: <sip:565656@ast2.local>;tag=as112d0057 Call-ID: 2d167b8f4108d4467ff3ea4e1cff7218@192.168.56.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:565656@192.168.58.3> Content-Type: application/sdp Content-Length: 285 v=0 o=root 4755 4755 IN IP4 192.168.58.3 s=session c=IN IP4 192.168.58.3 t=0 0 m=audio 19452 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <RTPPROXY> DBUG:handle_command: received command "L 2d167b8f4108d4467ff3ea4e1cff7218@192.168.56.3 192.168.58.3 19452 as6826385c;1 as112d0057;1" INFO:handle_command: BRAT: given internal address 192.168.58.3 INFO:create_twinlistener: BINDING TO 0.0.0.0 INFO:create_twinlistener: BINDING TO 0.0.0.0 INFO:handle_command: lookup on ports 50026/52556, session timer restarted INFO:handle_command: pre-filling callee's address with 192.168.58.3:19452 DBUG:doreply: sending reply "52556" <back at ast1> <--- SIP read from 192.168.56.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.56.3:5060;received=192.168.56.3;branch=z9hG4bK391f49a8;rport=5060 Record-Route: <sip:192.168.58.2;r2=on;lr=on> Record-Route: <sip:192.168.56.2;r2=on;lr=on> From: "Extension 1001" <sip:bratner@192.168.56.3>;tag=as6826385c To: <sip:565656@ast2.local>;tag=as112d0057 Call-ID: 2d167b8f4108d4467ff3ea4e1cff7218@192.168.56.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:565656@192.168.58.3> Content-Type: application/sdp Content-Length: 303unforce_rtp_proxy(); v=0 o=root 4755 4755 IN IP4 192.168.58.3 s=session c=IN IP4 192.168.58.2 t=0 0 m=audio 52556 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes rtp debug on ast1: Got RTP packet from 10.200.10.195:24466 (type 18, seq 052716, ts 1254981560, len 000020) Sent RTP packet to 192.168.58.2:52556 (type 00, seq 062080, ts 1254981560, len 000160) Got RTP packet from 10.200.10.195:24466 (type 18, seq 052717, ts 1254981720, len 000020) Sent RTP packet to 192.168.58.2:52556 (type 00, seq 062081, ts 1254981720, len 000160) Got RTP packet from 10.200.10.195:24466 (type 18, seq 052718, ts 1254981880, len 000020) Sent RTP packet to 192.168.58.2:52556 (type 00, seq 062082, ts 1254981880, len 000160) Got RTP packet from 10.200.10.195:24466 (type 18, seq 052719, ts 1254982040, len 000020) Sent RTP packet to 192.168.58.2:52556 (type 00, seq 062083, ts 1254982040, len 000160) there is no route to .58.2 from ast1.local rtp debug on ast2: <-------------> --- (11 headers 0 lines) --- Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005309, ts 000320, len 000160) Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005310, ts 000480, len 000160) Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005311, ts 000640, len 000160) Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005312, ts 000800, len 000160) Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005313, ts 000960, len 000160) Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005314, ts 001120, len 000160) Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005315, ts 001280, len 000160) Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005316, ts 001440, len 000160) Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005317, ts 001600, len 000160) Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005318, ts 001760, len 000160) Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005319, ts 001920, len 000160) there is no route to .56.2 from this host. _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users