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author | Terry Wilson <twilson@digium.com> | 2010-08-19 02:20:42 +0000 |
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committer | Terry Wilson <twilson@digium.com> | 2010-08-19 02:20:42 +0000 |
commit | 818bedf763ff4e2ec2d7f4fe849924f1f92118cd (patch) | |
tree | c4b4d2d5dfeddd02faa56775fc62ce8af19f38b8 /configs/sip.conf.sample | |
parent | 0e5b6069f44b7120ae41523b862fa0023099714b (diff) |
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
Diffstat (limited to 'configs/sip.conf.sample')
-rw-r--r-- | configs/sip.conf.sample | 12 |
1 files changed, 12 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 62af0e213..287e3c52b 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. |