Files
asterisk/include/asterisk
Olle Johansson 0a52297cf0 Add new SIP cli command "sip show channelstats" that displays some QoS data (if we have RTCP reports
and not use the p2p rtp bridge). I could not find a way to detect us using the p2p bridge, which
would be nice.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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