From 1c0a4847633f6b0950c57a1e2cf3258ece1c5bcd Mon Sep 17 00:00:00 2001 From: Terry Wilson Date: Thu, 19 Aug 2010 02:12:55 +0000 Subject: [PATCH] Add some documentation about codec negotiation to sip.conf git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@282729 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- configs/sip.conf.sample | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 9a420f7686..deb40781eb 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -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 -- 2.47.2