sip-audio-session¶
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. Once the media stream is connected, the outcome of the ICE negotiation and the selected RTP candidates are displayed.
This script is available in sipclients package that must be installed separately from SIP SIMPLe client SDK package.
adigeo@ag-blink:~$sip-audio-session -h
Usage: sip-audio-session [options] [user@domain]
This script can sit idle waiting for an incoming audio session, or initiate an
outgoing audio session to a SIP address. The program will close the session
and quit when Ctrl+D is pressed.
Options:
-h, --help show this help message and exit
-a NAME, --account=NAME
The account name to use for any outgoing traffic. If
not supplied, the default account will be used.
-c FILE, --config-file=FILE
The path to a configuration file to use. This
overrides the default location of the configuration
file.
-s, --trace-sip Dump the raw contents of incoming and outgoing SIP
messages.
-j, --trace-pjsip Print PJSIP logging output.
-n, --trace-notifications
Print all notifications (disabled by default).
-S, --disable-sound Disables initializing the sound card.
--auto-answer Interval after which to answer an incoming session
(disabled by default). If the option is specified but
the interval is not, it defaults to 0 (accept the
session as soon as it starts ringing).
--auto-hangup Interval after which to hang up an established session
(disabled by default). If the option is specified but
the interval is not, it defaults to 0 (hangup the
session as soon as it connects).
-b, --batch Run the program in batch mode: reading input from the
console is disabled and the option --auto-answer is
implied. This is particularly useful when running this
script in a non-interactive environment.
-D, --daemonize Enable running this program as a deamon. This option
implies --disable-sound, --auto-answer and --batch.
Incoming Session¶
adigeo@ag-blink:~$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.
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 ICE negotiation succeeded in 1s:412 Audio RTP endpoints 192.168.1.124:50852 (ICE type host) <-> 192.168.1.124:50871 (ICE type host) 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
Alarm System¶
sip-audio-session script can be used for end-to-end testing of a SIP service including the RTP media path. The following 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.