From: Terry Wilson <twilson@digium.com>
Date: Thu, 19 Aug 2010 02:14:28 +0000 (+0000)
Subject: Merged revisions 282729 via svnmerge from
X-Git-Tag: 1.6.2.12-rc1~9
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=0faad5f43b2c7ab55c4ea9326e30c661b39528d1;p=thirdparty%2Fasterisk.git

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
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 516ec7058a..24f03b95bb 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -188,6 +188,19 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;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