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Commit dce6a54a authored by Mark Michelson's avatar Mark Michelson
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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
parent a11ac5ae
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