Trunk implementation of setting an alternate RTP source.
authorMark Michelson <mmichelson@digium.com>
Thu, 18 Jun 2009 15:20:17 +0000 (15:20 +0000)
committerMark Michelson <mmichelson@digium.com>
Thu, 18 Jun 2009 15:20:17 +0000 (15:20 +0000)
commitdce6a54a4a8bd2a36cdc72d6d111df6a252436a0
tree71dde57cae86290e715b1392b9985c98ff99908b
parenta11ac5ae2f1961cab2806dc9618dcdc7247b252f
Trunk implementation of setting an alternate RTP source.

This contains the interface by which we can let an rtp instance know
that it might start receiving audio from a new source. This is similar
in nature to revision 197588 of Asterisk 1.4.

Review: https://reviewboard.asterisk.org/r/276

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c
include/asterisk/rtp_engine.h
main/rtp_engine.c
res/res_rtp_asterisk.c