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0fa2f5914e
RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions. The ability to disable RTCP streams in res_rtp_asterisk was missing, so this code was added to support the bug fix. (closes issue ASTERISK-18400) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340970 65c4cc65-6c06-0410-ace0-fbb531ad65f3