[asterisk-commits] tzafrir: branch tzafrir/monitor-rtp r220293 - /team/tzafrir/monitor-rtp/doc/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Sep 24 14:57:05 CDT 2009


Author: tzafrir
Date: Thu Sep 24 14:57:01 2009
New Revision: 220293

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=220293
Log:
Basic documentation for the RTP monitoring

Added:
    team/tzafrir/monitor-rtp/doc/monitor-rtp.txt   (with props)

Added: team/tzafrir/monitor-rtp/doc/monitor-rtp.txt
URL: http://svnview.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/doc/monitor-rtp.txt?view=auto&rev=220293
==============================================================================
--- team/tzafrir/monitor-rtp/doc/monitor-rtp.txt (added)
+++ team/tzafrir/monitor-rtp/doc/monitor-rtp.txt Thu Sep 24 14:57:01 2009
@@ -1,0 +1,103 @@
+Asterisk RTP Monitoring
+=======================
+
+This document describes using the Monitoring To RTP feature of Asterisk.
+Note that this feature is still work in progress. 
+
+"RTP Monitoring" adds an extra backend to the current Monitor
+(recording) code: rather than recoding calls to files, send them as RTP
+streams over the network to a resording server. The RTP streams are
+proided with extra meta-data through dummy SIP INVITE (at the beginning)
+and BYE (in the end of the call) messages.
+
+
+Usage
+-----
+RTP monitoring is configured through monitor.conf (a file currently only
+used for this feature). See a sample/reference file in the configs
+directory.
+
+In order for recording to work, 'rtp_server' under [general]' must be
+set. Set it to the server to which you wish to send recordings.
+
+In order for calls to be recorded you also need to:
+
+1. Set the channel to be monitored. Use one of the following:
+   - An explicit Monitor(wav) in the dialplan. Note that the value used
+     here ('wav') will actually be ignored.
+   - Set(AUTO_MONITOR=wav) in the dialplan. As before, the actual value
+     will be ignored.
+   - setvar = AUTO_MONITOR=wav in the channel configuration (sip.conf,
+     iax.conf, chan_dahdi.conf, maybe others).
+2. (If not a DAHDI channel) set a unique port offset in the variable 
+   RTP_PORT_OFFSET . This should be unique for the run-time.
+
+The requirement for RTP_PORT_OFFSET is ugly and should hopefully be
+removed soon.
+
+There are two RTP streams per recorded channel (one for rx and one for
+tx). At the moment the rx stream is sent to port 9000+RTP_PORT_OFFSET
+and the tx stream is sent to port 11000+RTP_PORT_OFFSET (the numbers
+9000 and 11000 can be modified in monitor.conf).
+
+
+Oreka / Orecx
+-------------
+Support for recording the produced streams of this plugin has been added
+in recent SVN versions (try using at least rev. 641) of orkaudio. In
+order to use it, add the following line inside the <VoIpPlugin> tag in
+/etc/orkaudio/config.xml
+
+  <DahdiIntercept>yes</DahdiIntercept>
+
+I hope it will be renamed to something along the lines of
+'AsteriskIntercept' as the code should not have anything DAHDI-specific.
+
+
+Wireshark
+---------
+The generated streams can be shown in wireshark. Either capture in it
+directly or capture in tcpdump:
+
+  tcpdump -s 0 -w recordings.cap #add custom filter here
+
+In Wireshark use Statistics => RTP => Show All Streams
+This opens a window in which each stream is shown (there are two such
+streams per recorded channel).
+
+
+Testing
+-------
+The following can be used to generate a call without any other channels
+about in the system:
+
+In extensions.conf:
+
+ [rtp-monitor]
+ exten => genrec,1,Answer
+ exten => genrec,n,Set(AUTO_MONITOR=wav)
+ exten => genrec,n,Set(RTP_PORT_OFFSET=23)
+ exten => genrec,n,Dial(Local/answer-rec at rtp-monitor/n)
+ exten => genrec,n,Hangup
+
+ exten => answer-rec,1,Answer
+ exten => answer-rec,n,Set(AUTO_MONITOR=wav)
+ exten => answer-rec,n,Set(RTP_PORT_OFFSET=29)
+ exten => answer-rec,n,Playback(demo-thanks)
+ exten => answer-rec,n,Hangup
+     
+
+And now generate a call with:
+
+  asterisk -rx 'channel originate Local/genrec at rtp-monitor/n application Echo'
+  
+
+Bugs
+----
+Many. Did I mention work in progress?
+
+* The fact that you have to set RTP_PORT_OFFSET for channels.
+* It takes restarting Asterisk or so to re-read monitor.conf .
+* Starting a monitoring with Monitor() can crash Asterisk. Didn't look
+  into it yet.
+

Propchange: team/tzafrir/monitor-rtp/doc/monitor-rtp.txt
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: team/tzafrir/monitor-rtp/doc/monitor-rtp.txt
------------------------------------------------------------------------------
    svn:keywords = Author Date Id Revision

Propchange: team/tzafrir/monitor-rtp/doc/monitor-rtp.txt
------------------------------------------------------------------------------
    svn:mime-type = text/plain




More information about the asterisk-commits mailing list