Merged revisions 282740 via svnmerge from
authorTerry Wilson <twilson@digium.com>
Thu, 19 Aug 2010 02:20:42 +0000 (02:20 +0000)
committerTerry Wilson <twilson@digium.com>
Thu, 19 Aug 2010 02:20:42 +0000 (02:20 +0000)
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

index 62af0e2..287e3c5 100644 (file)
@@ -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.