Sip audio session » History » Revision 20
Revision 19 (Adrian Georgescu, 07/23/2009 01:42 PM) → Revision 20/28 (Adrian Georgescu, 08/19/2009 02:08 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.
[[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-imac3:~$sip_audio_session -h
Usage: sip_audio_session [options] [user@domain]
This script can sit idle waiting for an incoming audio session, or initiate an
outgoing audio session to a SIP address. The program will close the session
and quit when Ctrl+D is pressed.
Options:
-h, --help show this help message and exit
-a NAME, --account=NAME
The account name to use for any outgoing traffic. If
not supplied, the default account will be used.
-c FILE, --config-file=FILE
The path to a configuration file to use. This
overrides the default location of the configuration
file.
-s, --trace-sip Dump the raw contents of incoming and outgoing SIP
messages.
-j, --trace-pjsip Print PJSIP logging output.
-n, --trace-notifications
Print all notifications (disabled by default).
-S, --disable-sound Disables initializing the sound card.
--auto-answer Interval after which to answer an incoming session
(disabled by default). If the option is specified but
the interval is not, it defaults to 0 (accept the
session as soon as it starts ringing).
--auto-hangup Interval after which to hang up an established session
(disabled by default). If the option is specified but
the interval is not, it defaults to 0 (hangup the
session as soon as it connects).
-b, --batch Run the program in batch mode: reading input from the
console is disabled and the option --auto-answer is
implied. This is particularly useful when running this
script in a non-interactive environment.
-D, --daemonize Enable running this program as a deamon. This option
implies --disable-sound, --auto-answer and --batch.
}}}
=== Example for incoming session ===
{{{
adigeo@ag-imac3:~$sip_audio_session
Using account 31208005169@ag-projects.com
Logging SIP trace to file "/Users/adigeo/.sipclient/logs/sip_trace.txt"
Logging PJSIP trace to file "/Users/adigeo/.sipclient/logs/pjsip_trace.txt"
Available audio input devices: Built-in Input, Built-in Microphone, Logitech Wireless Headset
Available audio output devices: Built-in Output, Logitech Wireless Headset
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
<>: adjust echo cancellation
SPACE: hold/unhold
Ctrl-d: quit the program
?: display this help message
2009-07-23 13:40:02 Registered contact "sip:oedtbzgw@192.168.1.6:50361" for sip:31208005169@ag-projects.com at 81.23.228.129:5060;transport=udp (expires in 600 seconds).
Other registered contacts:
sip:froghdmq@192.168.1.6:50334 (expires in 547 seconds)
sip:31208005169@192.168.1.123:5060 (expires in 234 seconds)
sip:zegoxqlw@192.168.1.6:50298 (expires in 468 seconds)
sip:31208005169@192.168.1.1;uniq=5B2860C44383A3D6705629A7E1FB8 (expires in 813 seconds)
Detected NAT type: Port Restricted
Incoming audio session from 'sip:31208005169@ag-projects.com', do you want to accept? (y/n)
Audio session established using "speex" codec at 32000Hz
Audio RTP endpoints 80.101.96.20:50406 <-> 81.23.228.150:52916
RTP audio stream is encrypted
Remote SIP User Agent is "sipsimple 0.9.1"
Audio session ended by remote party
Call duration was 4 seconds
}}}
=== Example for outgoing session ===
{{{
adigeo@ag-imac3:~$sip_audio_session ag@ag-projects.com
Using account 31208005169@ag-projects.com
Logging SIP trace to file "/Users/adigeo/.sipclient/logs/sip_trace.txt"
Logging PJSIP trace to file "/Users/adigeo/.sipclient/logs/pjsip_trace.txt"
Available audio input devices: Built-in Input, Built-in Microphone, Logitech Wireless Headset
Available audio output devices: Built-in Output, Logitech Wireless Headset
Using audio input device: Built-in Microphone
Using audio output device: Built-in Output
Using audio alert device: Built-in Output
Available control keys:
s: toggle SIP trace on the console
j: toggle PJSIP trace on the console
n: toggle notifications trace on the console
p: toggle printing RTP statistics on the console
h: hang-up the active session
r: toggle audio recording
<>: adjust echo cancellation
SPACE: hold/unhold
Ctrl-d: quit the program
?: display this help message
Initiating SIP audio session from 'sip:31208005169@ag-projects.com' to 'sip:ag@ag-projects.com' via sip:81.23.228.150:5060;transport=udp...
Audio session established using "speex" codec at 32000Hz
Audio RTP endpoints 80.101.96.20:50400 <-> 81.23.228.150:53734
RTP audio stream is encrypted
Ending audio session...
Audio session ended by local party
Call duration was 5 seconds
}}}
=== Example for bonjour mode ===
In bonjour mode no server is used. This mode is useful for serverless ad-hoc LAN operation.
> The actual bonjour protocol that uses multicast DNS to broadcast the contact SIP URIs is not implemented.
[[Image(http://www.tech-invite.com/img/cf3665/cf3665-31.gif)]]
'''Called party'''
{{{
adigeo@ag-imac3:~$sip_audio_session -a bonjour@local
Using account bonjour@local
Listening on "sip:wjnrczhi@192.168.1.6:57624;transport=tls"
Listening on "sip:wjnrczhi@192.168.1.6:57623;transport=tcp"
Listening on "sip:wjnrczhi@192.168.1.6:61994"
Available control keys:
h: hang-up the active session
r: toggle audio recording
t: toggle SIP trace on the console
j: toggle PJSIP trace on the console
<> : adjust echo cancellation
SPACE: hold/on-hold
Ctrl-d: quit the program
?: display this help message
Incoming audio session from "sip:imdyzosg@192.168.1.6", do you want to accept? (y/n)
Session established, using "speex" codec at 32000Hz
Audio RTP endpoints 192.168.1.6:50276 <-> 192.168.1.6:50100
RTP audio stream is encrypted
Remote SIP User Agent is "sip2sip-0.9.0-pjsip-1.0.2-trunk-r2553"
Session ended by remote party.
Session duration was 5 seconds
}}}
'''Calling party'''
{{{
adigeo@ag-imac3:~$sip_audio_session -a bonjour@local "sip:wjnrczhi@192.168.1.6:57624;transport=tls"
Using account bonjour@local
Listening on "sip:imdyzosg@192.168.1.6:57626;transport=tls"
Listening on "sip:imdyzosg@192.168.1.6:57625;transport=tcp"
Listening on "sip:imdyzosg@192.168.1.6:62008"
Initiating SIP session from sip:imdyzosg@192.168.1.6 to sip:wjnrczhi@192.168.1.6:57624;transport=tls via tls:192.168.1.6:57624 ...
Available control keys:
h: hang-up the active session
r: toggle audio recording
t: toggle SIP trace on the console
j: toggle PJSIP trace on the console
<> : adjust echo cancellation
SPACE: hold/on-hold
Ctrl-d: quit the program
?: display this help message
Ringing...
Session established, using "speex" codec at 32000Hz
Audio RTP endpoints 192.168.1.6:50100 <-> 192.168.1.6:50276
RTP audio stream is encrypted
Remote SIP User Agent is "sip2sip-0.9.0-pjsip-1.0.2-trunk-r2553"
Ending session...
Session ended by local party.
Session duration was 5 seconds
}}}
=== Alarm system ===
sip_audio_session script can be used for end-to-end testing of a SIP service. 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. The failure can be caused by timeout, a negative response code or lack of RTP media after the SIP session has been established.
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.