[asterisk-commits] twilson: branch 1.6.2 r282730 - in /branches/1.6.2: ./ configs/sip.conf.sample

[SVN commits to the Asterisk project]

Author: twilson
Date: Wed Aug 18 21:14:28 2010
New Revision: 282730

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=282730
Log:
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:
    branches/1.6.2/   (props changed)
    branches/1.6.2/configs/sip.conf.sample

Propchange: branches/1.6.2/
------------------------------------------------------------------------------
--- branch-1.4-merged (original)
+++ branch-1.4-merged Wed Aug 18 21:14:28 2010
@@ -1,1 +1,1 @@
-/branches/1.4:1-279056,279206,279945,280088,280341,280448,280811,280982,281390,281566,281762,281819,281911,282129,282430
+/branches/1.4:1-279056,279206,279945,280088,280341,280448,280811,280982,281390,281566,281762,281819,281911,282129,282430,282729

Modified: branches/1.6.2/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/configs/sip.conf.sample?view=diff&rev=282730&r1=282729&r2=282730
==============================================================================
--- branches/1.6.2/configs/sip.conf.sample (original)
+++ branches/1.6.2/configs/sip.conf.sample Wed Aug 18 21:14:28 2010
@@ -188,6 +188,19 @@
 ;vmexten=voicemail              ; dialplan extension to reach mailbox sets the 
                                 ; 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.
+;
 ;disallow=all                   ; First disallow all codecs
 ;allow=ulaw                     ; Allow codecs in order of preference
 ;allow=ilbc                     ; see doc/rtp-packetization for framing options