[asterisk-commits] tzafrir: branch tzafrir/monitor-rtp r224490 - /team/tzafrir/monitor-rtp/doc/

[SVN commits to the Asterisk project]

Author: tzafrir
Date: Mon Oct 19 01:04:29 2009
New Revision: 224490

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=224490
Log:
Better explain why this branch is needed

Modified:
    team/tzafrir/monitor-rtp/doc/monitor-rtp.txt

Modified: team/tzafrir/monitor-rtp/doc/monitor-rtp.txt
URL: http://svnview.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/doc/monitor-rtp.txt?view=diff&rev=224490&r1=224489&r2=224490
==============================================================================
--- team/tzafrir/monitor-rtp/doc/monitor-rtp.txt (original)
+++ team/tzafrir/monitor-rtp/doc/monitor-rtp.txt Mon Oct 19 01:04:29 2009
@@ -9,6 +9,20 @@
 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.
+
+
+Why?
+----
+Q: Why not use audio_hooks?
+A: I'd like to have monitoring as trasparent as possible. AUTO_MONITOR
+makes this simple and requires very minimal configuration changes. There
+is still quite complete control in the dialplan.
+
+Q: Why not just run a local Orkaudio (Oreka) recording server?
+A: There are a number of cases where this won't handle:
+   - Non-VoIP calls
+   - Call-center agents, for which all their session is one long "call".
+   - Ability not to record me when I do my secret stuff.
 
 
 Usage