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Add documentation on rtp packetization.

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Technically, ooh323 doesn't support it yet, but there is a patch that should be committed very soon.

Issue #7989, patch by DEA, slightly modified.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Jason Parker
2006-09-20 17:39:59 +00:00
parent 3c6d5053ba
commit 8bd82ebc0d
4 changed files with 76 additions and 2 deletions
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2
configs/sip.conf.sample
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@@ -70,7 +70,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; defaults to "asterisk"
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc ;
;allow=ilbc ; see doc/rtp-packetization for framing options
;
; This option specifies a preference for which music on hold class this channel
; should listen to when put on hold if the music class has not been set on the