Re: [SR-Users] ACK and BYE messages uses wrong socket.
Yes, it is available when kamailio started. I tried the latest version (3.2.3) and it did the same thing. If I removed listen=udp:x.x.130.34:5060 (primary IP on the NIC) from configuration file, it will report error when trying to relay() the message like this: ERROR: [forward.c:220]: ERROR: get_out_socket: no socket found It seems that kamailio does not populate default socket with secondary IP (floating IP) and you have to use force_send_socket() to send the message. Gary From: Daniel-Constantin Mierla [mailto:mico...@gmail.com] Sent: Tuesday, July 03, 2012 12:23 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Gary Chen Subject: Re: [SR-Users] ACK and BYE messages uses wrong socket. Hello, is .36 address available on the network interface when kamailio is started? Also, use at least v3.2.3 (the latest in 3.2 series) to be sure you get all the fixes since 3.2.0 and this is not related to some issue already fixed. Cheers, Daniel On 7/3/12 4:52 PM, Gary Chen wrote: Yes, I did the same thing as you mentioned and it still doing the same thing. Here is my setup: mhomed=1 listen=udp:x.x.130.36:5060 # external IP listen=udp:x.x.130.34:5060 # external IP listen=udp:10.200.1.31:5060 # internal IP If I removed .130.34, I got error saying no socket found. Gary Chen From: sr-users-boun...@lists.sip-router.org<mailto:sr-users-boun...@lists.sip-router.org> [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Vitaliy Aleksandrov Sent: Tuesday, July 03, 2012 10:27 AM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] ACK and BYE messages uses wrong socket. Have you specified interfaces with "listen" command ? I had a problem as you described and have fixed it by moving a listen directive with a "floating ip" to the top of the list. So you can try to specify interfaces you will use for SIP and set a "virtual ip" at the top of that list. Kamailio 3.2.0 I am trying to setup kamailio to do the sip trunking. It receive the sip traffic from customer and then send it to carrier. I have two NIC interface's assigned with three IP's: Interface 1: ( Public IP's) x.x.130.34 x.x.130.36 floating IP interface 2: (private IP's) 10.10.1.31 .36 is a floating IP assigned by Linux-HA (heartbeat/pacemaker). I only want to use .36 to receive and send sip traffic. I uses force_send_socket() to send INVITE with .36 IP. But the ACK message always want to use .34 IP even the Route header has .36 in it unless I force it with force_send_socket() . How can I fix this problem? See below for the SIP messages: (x.x.128.205 is customer IP, x.x.129.200 is PSTN gateway IP) U x.x.128.205:51694 -> x.x.130.36:5060 INVITE sip:5033441174@x.x.130.36:5060<mailto:sip:5033441174@x.x.130.36:5060> SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG 4bK1D3CD1..From: <mailto:sip:5024427578@x.x.128.205>;tag=24513088-D59..To: mailto:sip:5033441174@x.x.130.3> 6>..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.20<mailto:1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.20> 5..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254113-3191264919-256 0877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, CO MET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 70..Remote-P arty-ID: <mailto:sip:5024427578@x.x.128.205>;party=calling;screen=no;privacy=off..Timestamp: 1341322661 ..Contact: <mailto:sip:5024427578@x.x.128.205:5060>..Expires: 180..Allow-Events: telephone-event..Conte nt-Type: application/sdp..Content-Length: 366v=0..o=CiscoSystemsSIP-GW-UserAgent 9094 579 IN IP 4 x.x.128.205..s=SIP Call..c=IN IP4 x.x.128.205..t=0 0..m=audio 19312 RTP/AVP 125 0 18 100 10 1..c=IN IP4 x.x.128.205..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a =fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone-event/8000 ..a=fmtp:101 0-16.. # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9 hG4bK1D3CD1;rport=51694..From: <mailto:sip:5024427578@x.x.128.205>;tag=24513088-D59..To: mailto:4@x.x.130.36>>..Call-ID: 1887588d-c44b11e1-be38d697-98a3e...@x.x.128.205..cseq<mailto:1887588d-c44b11e1-be38d697-98a3e...@x.x.128.205..cseq>: 101 INVITE..Se rver: LVS Proxy 1.0..Content-Length: 0 # U x.x.130.36:5060 -> x.x.129.200:5060 INVITE sip:15033441174@x.x.129.200:5060<mailto:sip:15033441174@x.x.129.200:5060> SIP/2.0..Record-Route: ..Via: S IP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport =51694;branch=z9hG4bK1D3CD1..From: <mailto:sip:5024427578
Re: [SR-Users] ACK and BYE messages uses wrong socket.
Yes, I did the same thing as you mentioned and it still doing the same thing. Here is my setup: mhomed=1 listen=udp:x.x.130.36:5060 # external IP listen=udp:x.x.130.34:5060 # external IP listen=udp:10.200.1.31:5060 # internal IP If I removed .130.34, I got error saying no socket found. Gary Chen From: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Vitaliy Aleksandrov Sent: Tuesday, July 03, 2012 10:27 AM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] ACK and BYE messages uses wrong socket. Have you specified interfaces with "listen" command ? I had a problem as you described and have fixed it by moving a listen directive with a "floating ip" to the top of the list. So you can try to specify interfaces you will use for SIP and set a "virtual ip" at the top of that list. Kamailio 3.2.0 I am trying to setup kamailio to do the sip trunking. It receive the sip traffic from customer and then send it to carrier. I have two NIC interface's assigned with three IP's: Interface 1: ( Public IP's) x.x.130.34 x.x.130.36 floating IP interface 2: (private IP's) 10.10.1.31 .36 is a floating IP assigned by Linux-HA (heartbeat/pacemaker). I only want to use .36 to receive and send sip traffic. I uses force_send_socket() to send INVITE with .36 IP. But the ACK message always want to use .34 IP even the Route header has .36 in it unless I force it with force_send_socket() . How can I fix this problem? See below for the SIP messages: (x.x.128.205 is customer IP, x.x.129.200 is PSTN gateway IP) U x.x.128.205:51694 -> x.x.130.36:5060 INVITE sip:5033441174@x.x.130.36:5060<mailto:sip:5033441174@x.x.130.36:5060> SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG 4bK1D3CD1..From: <mailto:sip:5024427578@x.x.128.205>;tag=24513088-D59..To: mailto:sip:5033441174@x.x.130.3> 6>..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.20<mailto:1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.20> 5..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254113-3191264919-256 0877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, CO MET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 70..Remote-P arty-ID: <mailto:sip:5024427578@x.x.128.205>;party=calling;screen=no;privacy=off..Timestamp: 1341322661 ..Contact: <mailto:sip:5024427578@x.x.128.205:5060>..Expires: 180..Allow-Events: telephone-event..Conte nt-Type: application/sdp..Content-Length: 366v=0..o=CiscoSystemsSIP-GW-UserAgent 9094 579 IN IP 4 x.x.128.205..s=SIP Call..c=IN IP4 x.x.128.205..t=0 0..m=audio 19312 RTP/AVP 125 0 18 100 10 1..c=IN IP4 x.x.128.205..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a =fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone-event/8000 ..a=fmtp:101 0-16.. # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9 hG4bK1D3CD1;rport=51694..From: <mailto:sip:5024427578@x.x.128.205>;tag=24513088-D59..To: mailto:4@x.x.130.36>>..Call-ID: 1887588d-c44b11e1-be38d697-98a3e...@x.x.128.205..cseq<mailto:1887588d-c44b11e1-be38d697-98a3e...@x.x.128.205..cseq>: 101 INVITE..Se rver: LVS Proxy 1.0..Content-Length: 0 # U x.x.130.36:5060 -> x.x.129.200:5060 INVITE sip:15033441174@x.x.129.200:5060<mailto:sip:15033441174@x.x.129.200:5060> SIP/2.0..Record-Route: ..Via: S IP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport =51694;branch=z9hG4bK1D3CD1..From: <mailto:sip:5024427578@x.x.128.205>;tag=24513088-D59..To: mailto:41174@x.x.130.36>>..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3 E39A@x.x.128.205..Supported<mailto:E39A@x.x.128.205..Supported>: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254 113-3191264919-2560877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANC EL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forw ards: 69..Remote-Party-ID: <mailto:sip:5024427578@x.x.128.205>;party=calling;screen=no;privacy=off..Tim estamp: 1341322661..Contact: <mailto:sip:5024427578@x.x.128.205:51694>..Expires: 180..Allow-Events: tel ephone-event..Content-Type: application/sdp..Content-Length: 375v=0..o=CiscoSystemsSIP-GW-UserA gent 9094 579 IN IP4 10.200.1.51..s=SIP Call..c=IN IP4 10.200.1.51..t=0 0..m=audio 20464 RTP/AVP 12 5 0 18 100 101..c=IN IP4 10.200.1.51..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G7 29/8000..a=fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/800
Re: [SR-Users] ACK and BYE messages uses wrong socket.
Have you specified interfaces with "listen" command ? I had a problem as you described and have fixed it by moving a listen directive with a "floating ip" to the top of the list. So you can try to specify interfaces you will use for SIP and set a "virtual ip" at the top of that list. Kamailio 3.2.0 I am trying to setup kamailio to do the sip trunking. It receive the sip traffic from customer and then send it to carrier. I have two NIC interface's assigned with three IP's: Interface 1: ( Public IP's) x.x.130.34 x.x.130.36 floating IP interface 2: (private IP's) 10.10.1.31 .36 is a floating IP assigned by Linux-HA (heartbeat/pacemaker). I only want to use .36 to receive and send sip traffic. I uses force_send_socket() to send INVITE with .36 IP. But the ACK message always want to use .34 IP even the Route header has .36 in it unless I force it with force_send_socket() . How can I fix this problem? See below for the SIP messages: (x.x.128.205 is customer IP, x.x.129.200 is PSTN gateway IP) U x.x.128.205:51694 -> x.x.130.36:5060 INVITE sip:5033441174@x.x.130.36:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG 4bK1D3CD1..From: ;tag=24513088-D59..To: 6>..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.20 5..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254113-3191264919-256 0877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, CO MET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 70..Remote-P arty-ID: ;party=calling;screen=no;privacy=off..Timestamp: 1341322661 ..Contact: ..Expires: 180..Allow-Events: telephone-event..Conte nt-Type: application/sdp..Content-Length: 366v=0..o=CiscoSystemsSIP-GW-UserAgent 9094 579 IN IP 4 x.x.128.205..s=SIP Call..c=IN IP4 x.x.128.205..t=0 0..m=audio 19312 RTP/AVP 125 0 18 100 10 1..c=IN IP4 x.x.128.205..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a =fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone-event/8000 ..a=fmtp:101 0-16.. # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9 hG4bK1D3CD1;rport=51694..From: ;tag=24513088-D59..To: 4@x.x.130.36>..Call-ID: 1887588d-c44b11e1-be38d697-98a3e...@x.x.128.205..cseq: 101 INVITE..Se rver: LVS Proxy 1.0..Content-Length: 0 # U x.x.130.36:5060 -> x.x.129.200:5060 INVITE sip:15033441174@x.x.129.200:5060 SIP/2.0..Record-Route: ..Via: S IP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport =51694;branch=z9hG4bK1D3CD1..From: ;tag=24513088-D59..To: 41174@x.x.130.36>..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3 E39A@x.x.128.205..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254 113-3191264919-2560877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANC EL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forw ards: 69..Remote-Party-ID: ;party=calling;screen=no;privacy=off..Tim estamp: 1341322661..Contact: ..Expires: 180..Allow-Events: tel ephone-event..Content-Type: application/sdp..Content-Length: 375v=0..o=CiscoSystemsSIP-GW-UserA gent 9094 579 IN IP4 10.200.1.51..s=SIP Call..c=IN IP4 10.200.1.51..t=0 0..m=audio 20464 RTP/AVP 12 5 0 18 100 101..c=IN IP4 10.200.1.51..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G7 29/8000..a=fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone- event/8000..a=fmtp:101 0-16..a=nortpproxy:yes.. U x.x.129.200:5060 -> x.x.130.36:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/UDP 216.4 9.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: ;tag=24513088 -D59..To: ;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Cal l-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPG ateway/IOS-12.x..CSeq: 101 INVITE..Allow-Events: telephone-event..Content-Length: 0 # U x.x.129.200:5060 -> x.x.130.36:5060 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/ UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: ;ta g=24513088-D59..To: ;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:4 1 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 101 INVITE..Require: 100rel..RSeq: 6708..Allow: INVITE, OPTIONS, B YE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: tele phone-event..Contact: ..Record-Route:
[SR-Users] ACK and BYE messages uses wrong socket.
Kamailio 3.2.0 I am trying to setup kamailio to do the sip trunking. It receive the sip traffic from customer and then send it to carrier. I have two NIC interface's assigned with three IP's: Interface 1: ( Public IP's) x.x.130.34 x.x.130.36 floating IP interface 2: (private IP's) 10.10.1.31 .36 is a floating IP assigned by Linux-HA (heartbeat/pacemaker). I only want to use .36 to receive and send sip traffic. I uses force_send_socket() to send INVITE with .36 IP. But the ACK message always want to use .34 IP even the Route header has .36 in it unless I force it with force_send_socket() . How can I fix this problem? See below for the SIP messages: (x.x.128.205 is customer IP, x.x.129.200 is PSTN gateway IP) U x.x.128.205:51694 -> x.x.130.36:5060 INVITE sip:5033441174@x.x.130.36:5060 SIP/2.0..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9hG 4bK1D3CD1..From: ;tag=24513088-D59..To: ..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.20 5..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254113-3191264919-256 0877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, CO MET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forwards: 70..Remote-P arty-ID: ;party=calling;screen=no;privacy=off..Timestamp: 1341322661 ..Contact: ..Expires: 180..Allow-Events: telephone-event..Conte nt-Type: application/sdp..Content-Length: 366v=0..o=CiscoSystemsSIP-GW-UserAgent 9094 579 IN IP 4 x.x.128.205..s=SIP Call..c=IN IP4 x.x.128.205..t=0 0..m=audio 19312 RTP/AVP 125 0 18 100 10 1..c=IN IP4 x.x.128.205..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a =fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone-event/8000 ..a=fmtp:101 0-16.. # U x.x.130.36:5060 -> x.x.128.205:51694 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP x.x.128.205:5060;branch=z9 hG4bK1D3CD1;rport=51694..From: ;tag=24513088-D59..To: ..Call-ID: 1887588d-c44b11e1-be38d697-98a3e...@x.x.128.205..cseq: 101 INVITE..Se rver: LVS Proxy 1.0..Content-Length: 0 # U x.x.130.36:5060 -> x.x.129.200:5060 INVITE sip:15033441174@x.x.129.200:5060 SIP/2.0..Record-Route: ..Via: S IP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0..Via: SIP/2.0/UDP x.x.128.205:5060;rport =51694;branch=z9hG4bK1D3CD1..From: ;tag=24513088-D59..To: ..Date: Tue, 03 Jul 2012 13:37:41 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3 E39A@x.x.128.205..Supported: 100rel,timer,replaces..Min-SE: 1800..Cisco-Guid: 411443261-3293254 113-3191264919-2560877466..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANC EL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..CSeq: 101 INVITE..Max-Forw ards: 69..Remote-Party-ID: ;party=calling;screen=no;privacy=off..Tim estamp: 1341322661..Contact: ..Expires: 180..Allow-Events: tel ephone-event..Content-Type: application/sdp..Content-Length: 375v=0..o=CiscoSystemsSIP-GW-UserA gent 9094 579 IN IP4 10.200.1.51..s=SIP Call..c=IN IP4 10.200.1.51..t=0 0..m=audio 20464 RTP/AVP 12 5 0 18 100 101..c=IN IP4 10.200.1.51..a=rtpmap:125 X-CCD/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G7 29/8000..a=fmtp:18 annexb=yes..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194..a=rtpmap:101 telephone- event/8000..a=fmtp:101 0-16..a=nortpproxy:yes.. U x.x.129.200:5060 -> x.x.130.36:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/UDP 216.4 9.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: ;tag=24513088 -D59..To: ;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:41 GMT..Cal l-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPG ateway/IOS-12.x..CSeq: 101 INVITE..Allow-Events: telephone-event..Content-Length: 0 # U x.x.129.200:5060 -> x.x.130.36:5060 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP x.x.130.36;branch=z9hG4bKc2ce.17a955a3.0,SIP/2.0/ UDP x.x.128.205:5060;rport=51694;branch=z9hG4bK1D3CD1..From: ;ta g=24513088-D59..To: ;tag=F0695368-74F..Date: Tue, 03 Jul 2012 13:37:4 1 GMT..Call-ID: 1887588D-C44B11E1-BE38D697-98A3E39A@x.x.128.205..Timestamp: 1341322661..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 101 INVITE..Require: 100rel..RSeq: 6708..Allow: INVITE, OPTIONS, B YE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER..Allow-Events: tele phone-event..Contact: ..Record-Route: ..Content-Disposition: session;handling=required..Content-Type: application/sdp..Content-Length: 2 90v=0..o=CiscoSystemsSIP-GW-UserAgent 2387 2116 IN IP4 x.x.129.200..s=SIP Call..c=IN IP4 216 .49.129.200..t=0 0..m=audio 18480 RTP/AVP 0 101 100..c=IN IP4 x.x.129.200..a=rtpmap:0 PCMU/8000. .a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=rtpmap:100 X-NSE/8000..a=fmtp:100 192-194.. # U x.x.130.36:5060 -> x.x.128.2
