Sip audio session » History » Revision 25
Revision 24 (Adrian Georgescu, 01/24/2010 03:24 PM) → Revision 25/28 (Adrian Georgescu, 01/24/2010 03:34 PM)
== sip-audio-session == [[TOC(SipTesting*, sip_*, xcap*,depth=2)]] === Description === This script can be used for interactive audio session or for scripting alarms. The script returns appropriate shell response codes for failed or successful sessions. The script can be setup to auto answer and auto hangup after predefined number of seconds, detects SIP negative response codes, missing ACK and the lack of RTP media after a session has been established. > This script is available in ''sipclients'' package that must be installed separately from SIP SIMPLe client SDK package. [[Image(http://www.tech-invite.com/img/cf3665/cf3665-32.gif)]] Source code: [source:scripts/sip_audio_session.py scripts/sip_audio_session.py] {{{ adigeo@ag-blink:~$sip-audio-session -h Usage: sip-audio-session [options] [user@domain] This script can sit idle waiting for an incoming audio session, or initiate an outgoing audio session to a SIP address. The program will close the session and quit when Ctrl+D is pressed. Options: -h, --help show this help message and exit -a NAME, --account=NAME The account name to use for any outgoing traffic. If not supplied, the default account will be used. -c FILE, --config-file=FILE The path to a configuration file to use. This overrides the default location of the configuration file. -s, --trace-sip Dump the raw contents of incoming and outgoing SIP messages. -j, --trace-pjsip Print PJSIP logging output. -n, --trace-notifications Print all notifications (disabled by default). -S, --disable-sound Disables initializing the sound card. --auto-answer Interval after which to answer an incoming session (disabled by default). If the option is specified but the interval is not, it defaults to 0 (accept the session as soon as it starts ringing). --auto-hangup Interval after which to hang up an established session (disabled by default). If the option is specified but the interval is not, it defaults to 0 (hangup the session as soon as it connects). -b, --batch Run the program in batch mode: reading input from the console is disabled and the option --auto-answer is implied. This is particularly useful when running this script in a non-interactive environment. -D, --daemonize Enable running this program as a deamon. This option implies --disable-sound, --auto-answer and --batch. }}} === Example for incoming session === {{{ adigeo@ag-blink:~$sip-audio-session Using account 31208005169@ag-projects.com Logging SIP trace to file "/Users/adigeo/Library/Application Support/Blink/logs/sip_trace.txt" Logging PJSIP trace to file "/Users/adigeo/Library/Application Support/Blink/logs/pjsip_trace.txt" Available audio input devices: None, system_default, Built-in Input, Built-in Microphone Available audio output devices: None, system_default, Built-in Output Using audio input device: Built-in Microphone Using audio output device: Built-in Output Using audio alert device: Built-in Output Available control keys: s: toggle SIP trace on the console j: toggle PJSIP trace on the console n: toggle notifications trace on the console p: toggle printing RTP statistics on the console h: hang-up the active session r: toggle audio recording m: mute the microphone i: change audio input device o: change audio output device a: change audio alert device <>: adjust echo cancellation SPACE: hold/unhold Ctrl-d: quit the program ?: display this help message 2009-08-25 16:37:12 Registered contact "sip:hxsyungk@192.168.1.124:59164" for sip:31208005169@ag-projects.com at 81.23.228.150:5060;transport=udp (expires in 600 seconds). Other registered contacts: sip:31208005169@192.168.1.123:5060 (expires in 274 seconds) sip:kwbfxyvl@192.168.1.124:59116 (expires in 522 seconds) sip:ilmegvkp@192.168.1.124:59003 (expires in 339 seconds) sip:31208005169@192.168.1.1;uniq=5B2860C44383A3D6705629A7E1FB8 (expires in 1162 seconds) Detected NAT type: Port Restricted Incoming audio session from 'sip:adi@umts.ro', do you want to accept? (y/n) Audio session established using "speex" codec at 16000Hz Audio RTP endpoints 192.168.1.124:50378 <-> 85.17.186.6:58868 RTP audio stream is encrypted Remote SIP User Agent is "Blink-0.9.0" Remote party has put the audio session on hold Audio session is put on hold Audio session ended by remote party Session duration was 6 seconds 2009-08-25 16:37:44 Registration ended. }}} === Example for outgoing session === {{{ adigeo@ag-blink:~$sip-audio-session -a umts ag@ag-projects.com Using account adi@umts.ro Logging SIP trace to file "/Users/adigeo/Library/Application Support/Blink/logs/sip_trace.txt" Logging PJSIP trace to file "/Users/adigeo/Library/Application Support/Blink/logs/pjsip_trace.txt" Available audio input devices: None, system_default, Built-in Input, Built-in Microphone Available audio output devices: None, system_default, Built-in Output Using audio input device: Built-in Microphone Using audio output device: Built-in Output Using audio alert device: Built-in Output Available control keys: s: toggle SIP trace on the console j: toggle PJSIP trace on the console n: toggle notifications trace on the console p: toggle printing RTP statistics on the console h: hang-up the active session r: toggle audio recording m: mute the microphone i: change audio input device o: change audio output device a: change audio alert device <>: adjust echo cancellation SPACE: hold/unhold Ctrl-d: quit the program ?: display this help message Initiating SIP audio session from 'sip:adi@umts.ro' to 'sip:ag@ag-projects.com' via sip:85.17.186.7:5060;transport=udp... Audio session established using "speex" codec at 16000Hz Audio RTP endpoints 192.168.1.124:50054 <-> 85.17.186.6:58866 RTP audio stream is encrypted Audio session is put on hold Remote party has put the audio session on hold Detected NAT type: Port Restricted Ending audio session... Audio session ended by local party Session duration was 7 seconds }}} === Session with sip trace enabled === Use -s parameter you can see on the console detailed trace of all DNS queries/responses and SIP traffic exchanged during the session. {{{ adigeo@ag-imac3:~$sip-audio-session -s -a umts ag@ag-projects.com Using account adi@umts.ro Logging SIP trace to file "/Users/adigeo/Desktop/FileTransfers/sip_trace.txt" Logging PJSIP trace to file "/Users/adigeo/Desktop/FileTransfers/pjsip_trace.txt" Logging notifications trace to file "/Users/adigeo/Desktop/FileTransfers/notifications_trace.txt" Available audio input devices: None, system_default, Built-in Input, Built-in Microphone, Logitech Wireless Headset Available audio output devices: None, system_default, Built-in Output, Logitech Wireless Headset Using audio input device: Logitech Wireless Headset Using audio output device: Logitech Wireless Headset Using audio alert device: Built-in Output Available control keys: s: toggle SIP trace on the console j: toggle PJSIP trace on the console n: toggle notifications trace on the console p: toggle printing RTP statistics on the console h: hang-up the active session r: toggle audio recording m: mute the microphone i: change audio input device o: change audio output device a: change audio alert device <>: adjust echo cancellation SPACE: hold/unhold Ctrl-d: quit the program ?: display this help message 2009-09-24 22:31:24.118467: DNS lookup SRV _stun._udp.umts.ro succeeded, ttl=10758: 0 0 3478 stun1.dns-hosting.info., 0 0 3479 stun2.dns-hosting.info. 2009-09-24 22:31:24.120425: DNS lookup NAPTR ag-projects.com succeeded, ttl=244: 20 0 "s" "SIP+D2U" "" _sip._udp.ag-projects.com. 2009-09-24 22:31:24.126619: DNS lookup A stun1.dns-hosting.info. succeeded, ttl=845: 81.23.228.150 2009-09-24 22:31:24.128383: DNS lookup SRV _sip._udp.ag-projects.com. succeeded, ttl=18: 0 0 5060 proxy.sipthor.net. 2009-09-24 22:31:24.132502: DNS lookup A stun2.dns-hosting.info. succeeded, ttl=845: 85.17.186.6 2009-09-24 22:31:24.136754: DNS lookup A proxy.sipthor.net. succeeded, ttl=5: 85.17.186.7, 81.23.228.129 Initiating SIP audio session from '"Adrian G." <sip:adi@umts.ro>' to 'sip:ag@ag-projects.com' via sip:85.17.186.7:5060;transport=udp... 2009-09-24 22:31:24.145751: DNS lookup SRV _stun._udp.umts.ro succeeded, ttl=10758: 0 0 3478 stun1.dns-hosting.info., 0 0 3479 stun2.dns-hosting.info. 2009-09-24 22:31:24.150530: DNS lookup A stun1.dns-hosting.info. succeeded, ttl=845: 81.23.228.150 2009-09-24 22:31:24.155510: DNS lookup A stun2.dns-hosting.info. succeeded, ttl=845: 85.17.186.6 2009-09-24 22:31:24.572498: SENDING: Packet 1, +0:00:00 192.168.1.6:62054 -(SIP over UDP)-> 85.17.186.7:5060 INVITE sip:ag@ag-projects.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:62054;rport;branch=z9hG4bKPjWy0ZCjWb9Ro6Cy15cBX3FE3H.er7.wzB Max-Forwards: 70 From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF To: <sip:ag@ag-projects.com> Contact: <sip:pfxtjskq@192.168.1.6:62054> Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3 CSeq: 16887 INVITE Route: <sip:85.17.186.7;lr> Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE Supported: 100rel User-Agent: blink-0.9.0 Content-Type: application/sdp Content-Length: 1087 v=0 o=- 3462813084 3462813084 IN IP4 192.168.1.6 s=blink-0.9.0 c=IN IP4 80.101.96.20 t=0 0 m=audio 62066 RTP/AVP 104 103 102 3 9 0 8 101 a=rtcp:62067 IN IP4 80.101.96.20 a=rtpmap:104 speex/32000 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:eQ0XcBiuyy33zR2HEHLiaS5LCxA1T9rvP9J8GLw6 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:zWQU33HIZ0a7otihkQe2Y4jvqpKpXtotNwoW9Xl8 a=ice-ufrag:0aa3379a a=ice-pwd:619764ea a=candidate:S 1 UDP 31 80.101.96.20 62066 typ srflx raddr 192.168.1.6 rport 62066 a=candidate:H 1 UDP 23 192.168.1.6 62066 typ host a=candidate:H 1 UDP 23 10.211.55.2 62066 typ host a=candidate:H 1 UDP 23 10.37.129.2 62066 typ host a=candidate:S 2 UDP 30 80.101.96.20 62067 typ srflx raddr 192.168.1.6 rport 62067 a=candidate:H 2 UDP 22 192.168.1.6 62067 typ host a=candidate:H 2 UDP 22 10.211.55.2 62067 typ host a=candidate:H 2 UDP 22 10.37.129.2 62067 typ host a=sendrecv -- 2009-09-24 22:31:24.601167: RECEIVED: Packet 2, +0:00:00.028669 85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.1.6:62054;rport=62054;branch=z9hG4bKPjWy0ZCjWb9Ro6Cy15cBX3FE3H.er7.wzB;received=80.101.96.20 From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF To: <sip:ag@ag-projects.com> Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3 CSeq: 16887 INVITE Server: SIP Thor on OpenSIPS XS 1.4.5 Content-Length: 0 -- 2009-09-24 22:31:24.621860: RECEIVED: Packet 3, +0:00:00.049362 85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.6:62054;received=80.101.96.20;rport=62054;branch=z9hG4bKPjWy0ZCjWb9Ro6Cy15cBX3FE3H.er7.wzB From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF To: <sip:ag@ag-projects.com>;tag=e7d4d6b46afb9bf88242924a8d869ebf.962b Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3 CSeq: 16887 INVITE Proxy-Authenticate: Digest realm="umts.ro", nonce="4abbd73a48ba8c7fc6617208684ad122088d2207" Server: SIP Thor on OpenSIPS XS 1.4.5 Content-Length: 0 -- 2009-09-24 22:31:24.622019: SENDING: Packet 4, +0:00:00.049521 192.168.1.6:62054 -(SIP over UDP)-> 85.17.186.7:5060 ACK sip:ag@ag-projects.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:62054;rport;branch=z9hG4bKPjWy0ZCjWb9Ro6Cy15cBX3FE3H.er7.wzB Max-Forwards: 70 From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF To: <sip:ag@ag-projects.com>;tag=e7d4d6b46afb9bf88242924a8d869ebf.962b Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3 CSeq: 16887 ACK Route: <sip:85.17.186.7;lr> User-Agent: blink-0.9.0 Content-Length: 0 -- 2009-09-24 22:31:24.622214: SENDING: Packet 5, +0:00:00.049716 192.168.1.6:62054 -(SIP over UDP)-> 85.17.186.7:5060 INVITE sip:ag@ag-projects.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:62054;rport;branch=z9hG4bKPjwxj-gfiYVdjLvEWkt0l-pLfriN3gjo-T Max-Forwards: 70 From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF To: <sip:ag@ag-projects.com> Contact: <sip:pfxtjskq@192.168.1.6:62054> Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3 CSeq: 16888 INVITE Route: <sip:85.17.186.7;lr> Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE Supported: 100rel User-Agent: blink-0.9.0 Proxy-Authorization: Digest username="adi", realm="umts.ro", nonce="4abbd73a48ba8c7fc6617208684ad122088d2207", uri="sip:ag@ag-projects.com", response="cb85bbe3dbe0dcd71820c6ceaa027566" Content-Type: application/sdp Content-Length: 1087 v=0 o=- 3462813084 3462813084 IN IP4 192.168.1.6 s=blink-0.9.0 c=IN IP4 80.101.96.20 t=0 0 m=audio 62066 RTP/AVP 104 103 102 3 9 0 8 101 a=rtcp:62067 IN IP4 80.101.96.20 a=rtpmap:104 speex/32000 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:eQ0XcBiuyy33zR2HEHLiaS5LCxA1T9rvP9J8GLw6 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:zWQU33HIZ0a7otihkQe2Y4jvqpKpXtotNwoW9Xl8 a=ice-ufrag:0aa3379a a=ice-pwd:619764ea a=candidate:S 1 UDP 31 80.101.96.20 62066 typ srflx raddr 192.168.1.6 rport 62066 a=candidate:H 1 UDP 23 192.168.1.6 62066 typ host a=candidate:H 1 UDP 23 10.211.55.2 62066 typ host a=candidate:H 1 UDP 23 10.37.129.2 62066 typ host a=candidate:S 2 UDP 30 80.101.96.20 62067 typ srflx raddr 192.168.1.6 rport 62067 a=candidate:H 2 UDP 22 192.168.1.6 62067 typ host a=candidate:H 2 UDP 22 10.211.55.2 62067 typ host a=candidate:H 2 UDP 22 10.37.129.2 62067 typ host a=sendrecv -- 2009-09-24 22:31:24.656088: RECEIVED: Packet 6, +0:00:00.083590 85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.1.6:62054;rport=62054;branch=z9hG4bKPjwxj-gfiYVdjLvEWkt0l-pLfriN3gjo-T;received=80.101.96.20 From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF To: <sip:ag@ag-projects.com> Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3 CSeq: 16888 INVITE Server: SIP Thor on OpenSIPS XS 1.4.5 Content-Length: 0 -- 2009-09-24 22:31:24.721041: RECEIVED: Packet 7, +0:00:00.148543 85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.6:62054;rport=62054;received=80.101.96.20;branch=z9hG4bKPjwxj-gfiYVdjLvEWkt0l-pLfriN3gjo-T Record-Route: <sip:85.17.186.7;lr;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.9e165924> Record-Route: <sip:81.23.228.150;lr;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.e5ffeb2> Record-Route: <sip:85.17.186.7;lr;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.8e165924> Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3 From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF To: <sip:ag@ag-projects.com>;tag=FkXkUNDcrT80u8GHaUIUuF4OrIJI6O8f CSeq: 16888 INVITE Server: blink-0.9.0 Contact: <sip:iwralmqz@80.101.96.20:61962> Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE Content-Length: 0 -- 2009-09-24 22:31:24.878489: RECEIVED: Packet 8, +0:00:00.305991 85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.6:62054;received=80.101.96.20;rport=62054;branch=z9hG4bKPjwxj-gfiYVdjLvEWkt0l-pLfriN3gjo-T Record-Route: <sip:85.17.186.7;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.9e165924> Record-Route: <sip:81.23.228.150;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.e5ffeb2> Record-Route: <sip:85.17.186.7;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.8e165924> From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF To: <sip:ag@ag-projects.com>;tag=96A4E0ACA527F9AF Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3 CSeq: 16888 INVITE Contact: <sip:31208005169@80.101.96.20:5060;uniq=5B2860C44383A3D6705629A7E1FB8> User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.56 (May 1 2008) Content-Length: 0 -- 2009-09-24 22:31:25.154425: RECEIVED: Packet 9, +0:00:00.581927 85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.6:62054;received=80.101.96.20;rport=62054;branch=z9hG4bKPjwxj-gfiYVdjLvEWkt0l-pLfriN3gjo-T From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF To: <sip:ag@ag-projects.com>;tag=000c854663c02cf2799a9168-4ae390b1 Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3 CSeq: 16888 INVITE Server: CSCO/7 Contact: <sip:31208005169@80.101.96.20:61000> Record-Route: <sip:81.23.228.129;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.985cae24>,<sip:85.17.186.7;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.9e165924>,<sip:81.23.228.150;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.e5ffeb2>,<sip:85.17.186.7;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.8e165924> Content-Length: 0 -- 2009-09-24 22:31:25.368613: RECEIVED: Packet 10, +0:00:00.796115 85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.6:62054;received=80.101.96.20;rport=62054;branch=z9hG4bKPjwxj-gfiYVdjLvEWkt0l-pLfriN3gjo-T From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF To: <sip:ag@ag-projects.com>;tag=000c854663c02cf2799a9168-4ae390b1 Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3 CSeq: 16888 INVITE Server: CSCO/7 Contact: <sip:31208005169@80.101.96.20:61000> Record-Route: <sip:81.23.228.129;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.985cae24>,<sip:85.17.186.7;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.9e165924>,<sip:81.23.228.150;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.e5ffeb2>,<sip:85.17.186.7;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.8e165924> Content-Type: application/sdp Content-Length: 197 v=0 o=Cisco-SIPUA 8420 8964 IN IP4 192.168.1.123 s=SIP Call c=IN IP4 81.23.228.150 t=0 0 m=audio 51974 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 -- 2009-09-24 22:31:25.369124: SENDING: Packet 11, +0:00:00.796626 192.168.1.6:62054 -(SIP over UDP)-> 85.17.186.7:5060 ACK sip:31208005169@80.101.96.20:61000 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.6:62054;rport;branch=z9hG4bKPjkq3Y5tZfK3d.zASBBAHQHZMavNNRQw0W Max-Forwards: 70 From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF To: <sip:ag@ag-projects.com>;tag=000c854663c02cf2799a9168-4ae390b1 Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3 CSeq: 16888 ACK Route: <sip:85.17.186.7;lr;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.8e165924> Route: <sip:81.23.228.150;lr;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.e5ffeb2> Route: <sip:85.17.186.7;lr;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.9e165924> Route: <sip:81.23.228.129;lr;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.985cae24> User-Agent: blink-0.9.0 Content-Length: 0 -- Audio session established using "PCMU" codec at 8000Hz Audio RTP endpoints 80.101.96.20:62066 <-> 81.23.228.150:51974 Detected NAT type: Port Restricted 2009-09-24 22:31:40.495793: RECEIVED: Packet 12, +0:00:15.923295 85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054 NOTIFY sip:80.101.96.20:62054 SIP/2.0 Via: SIP/2.0/UDP 85.17.186.7:5060;branch=0 From: sip:keepalive@85.17.186.7;tag=7c29b7d5 To: sip:80.101.96.20:62054 Call-ID: 296fc4b6-56ba6860-24109f@85.17.186.7 CSeq: 1 NOTIFY Event: keep-alive Content-Length: 0 -- 2009-09-24 22:31:40.495929: SENDING: Packet 13, +0:00:15.923431 192.168.1.6:62054 -(SIP over UDP)-> 85.17.186.7:5060 SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 85.17.186.7:5060;received=85.17.186.7;branch=0 Call-ID: 296fc4b6-56ba6860-24109f@85.17.186.7 From: <sip:keepalive@85.17.186.7>;tag=7c29b7d5 To: <sip:80.101.96.20> CSeq: 1 NOTIFY Server: blink-0.9.0 Content-Length: 0 -- 2009-09-24 22:31:43.425393: RECEIVED: Packet 14, +0:00:18.852895 85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054 BYE sip:pfxtjskq@80.101.96.20:62054 SIP/2.0 Record-Route: <sip:85.17.186.7;lr=on;ftag=000c854663c02cf2799a9168-4ae390b1> Record-Route: <sip:81.23.228.150;lr=on;ftag=000c854663c02cf2799a9168-4ae390b1> Record-Route: <sip:85.17.186.7;lr=on;ftag=000c854663c02cf2799a9168-4ae390b1> Max-Forwards: 7 Record-Route: <sip:81.23.228.129;lr=on;ftag=000c854663c02cf2799a9168-4ae390b1> Via: SIP/2.0/UDP 85.17.186.7;branch=z9hG4bK9c22.feada044.0 Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK9c22.3cf12dd3.0 Via: SIP/2.0/UDP 85.17.186.7;branch=z9hG4bK9c22.eeada044.0 Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bK9c22.877deec6.0 Via: SIP/2.0/UDP 192.168.1.123:5060;rport=61000;received=80.101.96.20;branch=z9hG4bK63eb02c1 From: <sip:ag@ag-projects.com>;tag=000c854663c02cf2799a9168-4ae390b1 To: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3 CSeq: 101 BYE User-Agent: CSCO/7 Content-Length: 0 RTP-RxStat: Dur=18,Pkt=29,Oct=4640,LatePkt=0,LostPkt=0,AvgJit=0 RTP-TxStat: Dur=18,Pkt=889,Oct=142240 -- 2009-09-24 22:31:43.425554: SENDING: Packet 15, +0:00:18.853056 192.168.1.6:62054 -(SIP over UDP)-> 85.17.186.7:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 85.17.186.7;received=85.17.186.7;branch=z9hG4bK9c22.feada044.0 Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK9c22.3cf12dd3.0 Via: SIP/2.0/UDP 85.17.186.7;branch=z9hG4bK9c22.eeada044.0 Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bK9c22.877deec6.0 Via: SIP/2.0/UDP 192.168.1.123:5060;rport=61000;received=80.101.96.20;branch=z9hG4bK63eb02c1 Record-Route: <sip:85.17.186.7;lr;ftag=000c854663c02cf2799a9168-4ae390b1> Record-Route: <sip:81.23.228.150;lr;ftag=000c854663c02cf2799a9168-4ae390b1> Record-Route: <sip:85.17.186.7;lr;ftag=000c854663c02cf2799a9168-4ae390b1> Record-Route: <sip:81.23.228.129;lr;ftag=000c854663c02cf2799a9168-4ae390b1> Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3 From: <sip:ag@ag-projects.com>;tag=000c854663c02cf2799a9168-4ae390b1 To: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF CSeq: 101 BYE Server: blink-0.9.0 Content-Length: 0 -- Audio session ended by remote party Session duration was 18 seconds }}} === Alarm system === sip-audio-session sip_audio_session script can be used for end-to-end testing of a SIP service including the RTP media path. The following follow failures can be detected: * Timeout * Negative response code * Lack of RTP media after the SIP session has been established * Missing ACK To setup the alarm system start periodically a caller script from a monitoring software using the following arguments: {{{ sip-audio-session --auto-hangup user@domain }}} Where the user@domain has been configured as the SIP account of the listener, can be an answering machine on the PSTN network. The caller script hangs up after each call. The shell return code can be used to determine if the session setup has failed. To receive calls and answer them automatically you can also use sip_audio_session script as follows: {{{ sip-audio-session --daemonize }}} You must run the script as user root. The --daemonize option puts the client in the background and the logging goes to /var/log/syslog. The program saves its pid file to /var/run/sip_audio_session.pid.