Sip audio session
Version 18 (Adrian Georgescu, 04/16/2009 09:38 am)
1 | 1 | Adrian Georgescu | == sip_audio_session == |
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2 | 2 | Adrian Georgescu | [[TOC(SipTesting*, sip_*, xcap*,depth=2)]] |
3 | 1 | Adrian Georgescu | |
4 | 1 | Adrian Georgescu | === Description === |
5 | 1 | Adrian Georgescu | |
6 | 1 | Adrian Georgescu | 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. |
7 | 6 | Adrian Georgescu | |
8 | 10 | Adrian Georgescu | [[Image(http://www.tech-invite.com/img/cf3665/cf3665-32.gif)]] |
9 | 1 | Adrian Georgescu | |
10 | 1 | Adrian Georgescu | Source code: [source:scripts/sip_audio_session.py scripts/sip_audio_session.py] |
11 | 1 | Adrian Georgescu | |
12 | 1 | Adrian Georgescu | {{{ |
13 | 5 | Adrian Georgescu | adigeo@ag-oxygen:~$sip_audio_session --help |
14 | 17 | Adrian Georgescu | Usage: sip_audio_session [options] [user@domain] |
15 | 1 | Adrian Georgescu | |
16 | 17 | Adrian Georgescu | This script can sit idle waiting for an incoming audio session, or initiate an |
17 | 17 | Adrian Georgescu | outgoing audio session to a SIP address. The program will close the session |
18 | 17 | Adrian Georgescu | and quit when Ctrl+D is pressed. |
19 | 1 | Adrian Georgescu | |
20 | 1 | Adrian Georgescu | Options: |
21 | 1 | Adrian Georgescu | -h, --help show this help message and exit |
22 | 5 | Adrian Georgescu | -a NAME, --account=NAME |
23 | 5 | Adrian Georgescu | The account name to use for any outgoing traffic. If |
24 | 5 | Adrian Georgescu | not supplied, the default account will be used. |
25 | 17 | Adrian Georgescu | -c [FILE], --config-file=[FILE] |
26 | 1 | Adrian Georgescu | The path to a configuration file to use. This |
27 | 5 | Adrian Georgescu | overrides the default location of the configuration |
28 | 5 | Adrian Georgescu | file. |
29 | 17 | Adrian Georgescu | -s, --trace-sip Dump the raw contents of incoming and outgoing SIP |
30 | 17 | Adrian Georgescu | messages. |
31 | 17 | Adrian Georgescu | -j, --trace-pjsip Print PJSIP logging output. |
32 | 17 | Adrian Georgescu | -n, --trace-notifications |
33 | 17 | Adrian Georgescu | Print all notifications (disabled by default). |
34 | 5 | Adrian Georgescu | -S, --disable-sound Disables initializing the sound card. |
35 | 17 | Adrian Georgescu | --auto-answer Interval after which to answer an incoming session |
36 | 1 | Adrian Georgescu | (disabled by default). If the option is specified but |
37 | 17 | Adrian Georgescu | the interval is not, it defaults to 0 (accept the |
38 | 17 | Adrian Georgescu | session as soon as it starts ringing). |
39 | 17 | Adrian Georgescu | --auto-hangup Interval after which to hang up an established session |
40 | 17 | Adrian Georgescu | (applies only to outgoing sessions, disabled by |
41 | 17 | Adrian Georgescu | default). If the option is specified but the interval |
42 | 17 | Adrian Georgescu | is not, it defaults to 0 (hangup the session as soon |
43 | 17 | Adrian Georgescu | as it connects). |
44 | 17 | Adrian Georgescu | -D, --daemonize Enabled running this program as a deamon. Note that |
45 | 17 | Adrian Georgescu | this forces --disable-sound and --auto-answer. |
46 | 17 | Adrian Georgescu | |
47 | 1 | Adrian Georgescu | }}} |
48 | 1 | Adrian Georgescu | |
49 | 1 | Adrian Georgescu | |
50 | 1 | Adrian Georgescu | === Example for incoming session === |
51 | 1 | Adrian Georgescu | |
52 | 1 | Adrian Georgescu | {{{ |
53 | 7 | Adrian Georgescu | adigeo@ag-imac3:~$sip_audio_session |
54 | 7 | Adrian Georgescu | Using account 31208005169@ag-projects.com |
55 | 1 | Adrian Georgescu | Available control keys: |
56 | 1 | Adrian Georgescu | h: hang-up the active session |
57 | 1 | Adrian Georgescu | r: toggle audio recording |
58 | 1 | Adrian Georgescu | t: toggle SIP trace on the console |
59 | 7 | Adrian Georgescu | j: toggle PJSIP trace on the console |
60 | 1 | Adrian Georgescu | <> : adjust echo cancellation |
61 | 1 | Adrian Georgescu | SPACE: hold/on-hold |
62 | 1 | Adrian Georgescu | Ctrl-d: quit the program |
63 | 1 | Adrian Georgescu | ?: display this help message |
64 | 7 | Adrian Georgescu | Succesfully registered using contact "sip:cwntuzyl@192.168.1.6:61163" |
65 | 1 | Adrian Georgescu | Detected NAT type: Port Restricted |
66 | 7 | Adrian Georgescu | Incoming audio session from ""Adrian G." <sip:31208005169@ag-projects.com>", do you want to accept? (y/n) |
67 | 7 | Adrian Georgescu | Session established, using "PCMU" codec at 8000Hz |
68 | 7 | Adrian Georgescu | Audio RTP endpoints 192.168.1.6:50132 <-> 85.17.186.7:53358 |
69 | 7 | Adrian Georgescu | Remote SIP User Agent is "CSCO/7" |
70 | 7 | Adrian Georgescu | Session ended by remote party. |
71 | 7 | Adrian Georgescu | Session duration was 3 seconds |
72 | 7 | Adrian Georgescu | |
73 | 1 | Adrian Georgescu | }}} |
74 | 1 | Adrian Georgescu | |
75 | 1 | Adrian Georgescu | === Example for outgoing session === |
76 | 1 | Adrian Georgescu | |
77 | 1 | Adrian Georgescu | {{{ |
78 | 7 | Adrian Georgescu | adigeo@ag-imac3:~$sip_audio_session ag@ag-projects.com |
79 | 7 | Adrian Georgescu | Using account 31208005169@ag-projects.com |
80 | 8 | Adrian Georgescu | Initiating SIP session from "Adrian G." <sip:31208005169@ag-projects.com> to |
81 | 8 | Adrian Georgescu | sip:ag@ag-projects.com via udp:81.23.228.150:5060 ... |
82 | 1 | Adrian Georgescu | Available control keys: |
83 | 1 | Adrian Georgescu | h: hang-up the active session |
84 | 1 | Adrian Georgescu | r: toggle audio recording |
85 | 1 | Adrian Georgescu | t: toggle SIP trace on the console |
86 | 7 | Adrian Georgescu | j: toggle PJSIP trace on the console |
87 | 1 | Adrian Georgescu | <> : adjust echo cancellation |
88 | 1 | Adrian Georgescu | SPACE: hold/on-hold |
89 | 1 | Adrian Georgescu | Ctrl-d: quit the program |
90 | 1 | Adrian Georgescu | ?: display this help message |
91 | 7 | Adrian Georgescu | Succesfully registered using contact "sip:ztomvpis@192.168.1.6:61215" |
92 | 1 | Adrian Georgescu | Ringing... |
93 | 1 | Adrian Georgescu | Session established, using "speex" codec at 32000Hz |
94 | 7 | Adrian Georgescu | Audio RTP endpoints 192.168.1.6:50374 <-> 81.23.228.129:52156 |
95 | 7 | Adrian Georgescu | Remote SIP User Agent is "sip2sip-0.9.0-pjsip-1.0.2-trunk-r2553" |
96 | 7 | Adrian Georgescu | Detected NAT type: Port Restricted |
97 | 7 | Adrian Georgescu | Ending session... |
98 | 7 | Adrian Georgescu | Session ended by local party. |
99 | 7 | Adrian Georgescu | Session duration was 12 seconds |
100 | 1 | Adrian Georgescu | }}} |
101 | 11 | Adrian Georgescu | |
102 | 11 | Adrian Georgescu | |
103 | 11 | Adrian Georgescu | === Example for bonjour mode === |
104 | 11 | Adrian Georgescu | |
105 | 11 | Adrian Georgescu | In bonjour mode no server is used. This mode is useful for serverless ad-hoc LAN operation. |
106 | 11 | Adrian Georgescu | |
107 | 11 | Adrian Georgescu | > The actual bonjour protocol that uses multicast DNS to broadcast the contact SIP URIs is not implemented. |
108 | 11 | Adrian Georgescu | |
109 | 11 | Adrian Georgescu | [[Image(http://www.tech-invite.com/img/cf3665/cf3665-31.gif)]] |
110 | 11 | Adrian Georgescu | |
111 | 12 | Adrian Georgescu | '''Called party''' |
112 | 12 | Adrian Georgescu | |
113 | 1 | Adrian Georgescu | {{{ |
114 | 12 | Adrian Georgescu | adigeo@ag-imac3:~$sip_audio_session -a bonjour@local |
115 | 11 | Adrian Georgescu | Using account bonjour@local |
116 | 12 | Adrian Georgescu | Listening on "sip:wjnrczhi@192.168.1.6:57624;transport=tls" |
117 | 12 | Adrian Georgescu | Listening on "sip:wjnrczhi@192.168.1.6:57623;transport=tcp" |
118 | 12 | Adrian Georgescu | Listening on "sip:wjnrczhi@192.168.1.6:61994" |
119 | 11 | Adrian Georgescu | Available control keys: |
120 | 11 | Adrian Georgescu | h: hang-up the active session |
121 | 11 | Adrian Georgescu | r: toggle audio recording |
122 | 1 | Adrian Georgescu | t: toggle SIP trace on the console |
123 | 1 | Adrian Georgescu | j: toggle PJSIP trace on the console |
124 | 11 | Adrian Georgescu | <> : adjust echo cancellation |
125 | 11 | Adrian Georgescu | SPACE: hold/on-hold |
126 | 11 | Adrian Georgescu | Ctrl-d: quit the program |
127 | 11 | Adrian Georgescu | ?: display this help message |
128 | 12 | Adrian Georgescu | Incoming audio session from "sip:imdyzosg@192.168.1.6", do you want to accept? (y/n) |
129 | 1 | Adrian Georgescu | Session established, using "speex" codec at 32000Hz |
130 | 12 | Adrian Georgescu | Audio RTP endpoints 192.168.1.6:50276 <-> 192.168.1.6:50100 |
131 | 1 | Adrian Georgescu | RTP audio stream is encrypted |
132 | 1 | Adrian Georgescu | Remote SIP User Agent is "sip2sip-0.9.0-pjsip-1.0.2-trunk-r2553" |
133 | 12 | Adrian Georgescu | Session ended by remote party. |
134 | 11 | Adrian Georgescu | Session duration was 5 seconds |
135 | 11 | Adrian Georgescu | }}} |
136 | 11 | Adrian Georgescu | |
137 | 12 | Adrian Georgescu | '''Calling party''' |
138 | 12 | Adrian Georgescu | |
139 | 11 | Adrian Georgescu | {{{ |
140 | 12 | Adrian Georgescu | adigeo@ag-imac3:~$sip_audio_session -a bonjour@local "sip:wjnrczhi@192.168.1.6:57624;transport=tls" |
141 | 1 | Adrian Georgescu | Using account bonjour@local |
142 | 12 | Adrian Georgescu | Listening on "sip:imdyzosg@192.168.1.6:57626;transport=tls" |
143 | 12 | Adrian Georgescu | Listening on "sip:imdyzosg@192.168.1.6:57625;transport=tcp" |
144 | 12 | Adrian Georgescu | Listening on "sip:imdyzosg@192.168.1.6:62008" |
145 | 12 | Adrian Georgescu | 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 ... |
146 | 11 | Adrian Georgescu | Available control keys: |
147 | 11 | Adrian Georgescu | h: hang-up the active session |
148 | 11 | Adrian Georgescu | r: toggle audio recording |
149 | 11 | Adrian Georgescu | t: toggle SIP trace on the console |
150 | 11 | Adrian Georgescu | j: toggle PJSIP trace on the console |
151 | 11 | Adrian Georgescu | <> : adjust echo cancellation |
152 | 11 | Adrian Georgescu | SPACE: hold/on-hold |
153 | 11 | Adrian Georgescu | Ctrl-d: quit the program |
154 | 11 | Adrian Georgescu | ?: display this help message |
155 | 12 | Adrian Georgescu | Ringing... |
156 | 11 | Adrian Georgescu | Session established, using "speex" codec at 32000Hz |
157 | 12 | Adrian Georgescu | Audio RTP endpoints 192.168.1.6:50100 <-> 192.168.1.6:50276 |
158 | 11 | Adrian Georgescu | RTP audio stream is encrypted |
159 | 11 | Adrian Georgescu | Remote SIP User Agent is "sip2sip-0.9.0-pjsip-1.0.2-trunk-r2553" |
160 | 12 | Adrian Georgescu | Ending session... |
161 | 12 | Adrian Georgescu | Session ended by local party. |
162 | 1 | Adrian Georgescu | Session duration was 5 seconds |
163 | 1 | Adrian Georgescu | }}} |
164 | 14 | Adrian Georgescu | |
165 | 14 | Adrian Georgescu | |
166 | 15 | Adrian Georgescu | === Alarm system === |
167 | 14 | Adrian Georgescu | |
168 | 16 | Adrian Georgescu | 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: |
169 | 14 | Adrian Georgescu | {{{ |
170 | 14 | Adrian Georgescu | sip_audio_session --auto-hangup user@domain |
171 | 14 | Adrian Georgescu | }}} |
172 | 14 | Adrian Georgescu | |
173 | 16 | Adrian Georgescu | 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. |
174 | 18 | Adrian Georgescu | |
175 | 18 | Adrian Georgescu | To receive calls and answer them automatically you can also use sip_audio_session script as follows: |
176 | 18 | Adrian Georgescu | |
177 | 18 | Adrian Georgescu | {{{ |
178 | 18 | Adrian Georgescu | sip_audio_session --daemonize |
179 | 18 | Adrian Georgescu | }}} |
180 | 18 | Adrian Georgescu | |
181 | 18 | Adrian Georgescu | 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. |