]> git.ipfire.org Git - thirdparty/asterisk.git/commit
Fix RTP reference leak.
authorMark Michelson <mmichelson@digium.com>
Sat, 21 Jan 2012 00:04:13 +0000 (00:04 +0000)
committerMark Michelson <mmichelson@digium.com>
Sat, 21 Jan 2012 00:04:13 +0000 (00:04 +0000)
commit27d894d6249b9ea33ec7e29eb46b152ae54019c6
treeee0bba77e93db542433b8e589c35bb30ffea8d33
parent2cefb605051b4a95b3a023d9ccbf91fd76f72405
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/

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352014 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c