Is it possible to somehow monitor/log packet loss and/or jitter in RTP? I want to know how things look if someone complains about audio.
ethereal can do some of this for rtp, I think. At the very least, if the endpoint supports RTCP (most do, except for asterisk), it can show you the contents of the RTCP RRs, which should contain this information.
Getting this stuff into asterisk would be in bug 2532, bug 2863, and bug 3236. [and not just getting stats, but actually improving quality under these conditions].
Can't use ethereal for realtime monitoring, though. Too heavy. This should be in rtp.c and possibly reporting to the manager or at least logging the stuff (to ODBC?)
roy
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