[svn-commits] twilson: trunk r282751 - in /trunk: ./ configs/sip.conf.sample

[SVN commits to the Digium repositories]

Author: twilson
Date: Wed Aug 18 21:20:42 2010
New Revision: 282751

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=282751
Log:
Merged revisions 282740 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r282740 | twilson | 2010-08-18 21:18:50 -0500 (Wed, 18 Aug 2010) | 16 lines
  
  Merged revisions 282730 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r282730 | twilson | 2010-08-18 21:14:28 -0500 (Wed, 18 Aug 2010) | 9 lines
    
    Merged revisions 282729 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 Aug 2010) | 2 lines
      
      Add some documentation about codec negotiation to sip.conf
    ........
  ................
................

Modified:
    trunk/   (props changed)
    trunk/configs/sip.conf.sample

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Modified: trunk/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=282751&r1=282750&r2=282751
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Wed Aug 18 21:20:42 2010
@@ -255,6 +255,18 @@
                                 ; Message-Account in the MWI notify message
                                 ; defaults to "asterisk"
 
+; Codec negotiation
+;
+; When Asterisk is receiving a call, the codec will initially be set to the
+; first codec in the allowed codecs defined for the user receiving the call
+; that the caller also indicates that it supports. But, after the caller
+; starts sending RTP, Asterisk will switch to using whatever codec the caller
+; is sending.
+;
+; When Asterisk is placing a call, the codec used will be the first codec in
+; the allowed codecs that the callee indicates that it supports. Asterisk will
+; *not* switch to whatever codec the callee is sending.
+;
 ;preferred_codec_only=yes       ; Respond to a SIP invite with the single most preferred codec
                                 ; rather than advertising all joint codec capabilities. This
                                 ; limits the other side's codec choice to exactly what we prefer.