Fix RTP reference leak.
authorMark Michelson <mmichelson@digium.com>
Sat, 21 Jan 2012 00:10:35 +0000 (00:10 +0000)
committerMark Michelson <mmichelson@digium.com>
Sat, 21 Jan 2012 00:10:35 +0000 (00:10 +0000)
commitab8ba431b6b5a29c8b0238afa8d763a87ae057e0
treea5477a084def48fe85eb3cd6e6776355e4e6db32
parentd0c765497df2fb6a076e98ab5bec1505342431e3
Fix RTP reference leak.

If a blind transfer were initiated using a REFER without a prior
reINVITE to place the call on hold, AND if Asterisk were sending
RTCP reports, then there was a reference for the RTP instance
of the transferer.

This fixes the issue by merging two similar but slightly conflicting
sections of code into a single area. It also adds a stop_media_flows()
call in the case that the transferer's UA never sends a BYE to us
like it is supposed to.

(issue ASTERISK-19192)

Review: https://reviewboard.asterisk.org/r/1681/
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Merged revisions 352014 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 352015 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c