From 818bedf763ff4e2ec2d7f4fe849924f1f92118cd Mon Sep 17 00:00:00 2001
From: Terry Wilson <twilson@digium.com>
Date: Thu, 19 Aug 2010 02:20:42 +0000
Subject: [PATCH] Merged revisions 282740 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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
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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
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r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 Aug 2010) | 2 lines
Add some documentation about codec negotiation to sip.conf
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@282751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
configs/sip.conf.sample | 12 ++++++++++++
1 file changed, 12 insertions(+)
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 62af0e2138..287e3c52be 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -255,6 +255,18 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; 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.
--
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