Sip audio session » History » Revision 17
Revision 16 (Adrian Georgescu, 04/14/2009 10:42 PM) → Revision 17/28 (Adrian Georgescu, 04/15/2009 03:21 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-oxygen:~$sip_audio_session --help
Usage: sip_audio_session [options] [user@domain] [target-user@target-domain.com]
This script can sit idle waiting for an incoming audio session, call, or initiate perform an
outgoing audio session call to a the target SIP address. account. The program will close the
session
and quit when Ctrl+D is pressed.
Options:
-h, --help show this help message and exit
-a NAME, --account=NAME
The account name to use for any outgoing traffic. If
not supplied, the default account will be used.
-c [FILE], --config-file=[FILE] --config_file=[FILE]
The path to a configuration file to use. This
overrides the default location of the configuration
file.
-s, --trace-sip -s [stdout|file|all|none], --trace-sip=[stdout|file|all|none]
Dump the raw contents of incoming and outgoing SIP
messages. The argument specifies where the messages
are to be dumped.
-j, --trace-pjsip -j [stdout|file|all|none], --trace-pjsip=[stdout|file|all|none]
Print PJSIP logging output.
-n, --trace-notifications The argument specifies
Print all notifications (disabled by default). where the messages are to be dumped.
-S, --disable-sound Disables initializing the sound card.
--auto-answer Interval after which to answer an incoming session call
(disabled by default). If the option is specified but
the interval is not, it defaults to 0 (accept (answer the call
session as soon as it starts ringing).
--auto-hangup Interval after which to hang up hangup an established session on-going call
(applies only to outgoing sessions, calls, disabled by default).
default). If the option is specified but the interval
is not, it
defaults to 0 (hangup the session call as soon
as it
connects).
-D, --daemonize Enabled running this program as a deamon. Note that
this forces --disable-sound and --auto-answer.
}}}
=== Example for incoming session ===
{{{
adigeo@ag-imac3:~$sip_audio_session
Using account 31208005169@ag-projects.com
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
Succesfully registered using contact "sip:cwntuzyl@192.168.1.6:61163"
Detected NAT type: Port Restricted
Incoming audio session from ""Adrian G." <sip:31208005169@ag-projects.com>", do you want to accept? (y/n)
Session established, using "PCMU" codec at 8000Hz
Audio RTP endpoints 192.168.1.6:50132 <-> 85.17.186.7:53358
Remote SIP User Agent is "CSCO/7"
Session ended by remote party.
Session duration was 3 seconds
}}}
=== Example for outgoing session ===
{{{
adigeo@ag-imac3:~$sip_audio_session ag@ag-projects.com
Using account 31208005169@ag-projects.com
Initiating SIP session from "Adrian G." <sip:31208005169@ag-projects.com> to
sip:ag@ag-projects.com via udp:81.23.228.150:5060 ...
Available control keys:
h: hang-up the active session
r: toggle audio recording
t: toggle SIP trace on the console
j: toggle PJSIP trace on the console
<> : adjust echo cancellation
SPACE: hold/on-hold
Ctrl-d: quit the program
?: display this help message
Succesfully registered using contact "sip:ztomvpis@192.168.1.6:61215"
Ringing...
Session established, using "speex" codec at 32000Hz
Audio RTP endpoints 192.168.1.6:50374 <-> 81.23.228.129:52156
Remote SIP User Agent is "sip2sip-0.9.0-pjsip-1.0.2-trunk-r2553"
Detected NAT type: Port Restricted
Ending session...
Session ended by local party.
Session duration was 12 seconds
}}}
=== Example for bonjour mode ===
In bonjour mode no server is used. This mode is useful for serverless ad-hoc LAN operation.
> The actual bonjour protocol that uses multicast DNS to broadcast the contact SIP URIs is not implemented.
[[Image(http://www.tech-invite.com/img/cf3665/cf3665-31.gif)]]
'''Called party'''
{{{
adigeo@ag-imac3:~$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.