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Add some documentation about codec negotiation to sip.conf

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@282729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Terry Wilson
2010-08-19 02:12:55 +00:00
parent c5278fc111
commit 1c0a484763
1 changed files with 13 additions and 0 deletions
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13
configs/sip.conf.sample
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@@ -91,6 +91,19 @@ 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