Hi all,
With Asterisk 1.6.1.6
Trying to have statistic concerning Rtp audio quality, I use
${CHANNEL(rtpqos,audio,all)}
having also tried AUDIORTPQOS and ${CHANNEL(rtpqos,audio,...)}
Sometimes, it works and I have results.
Most of the time I get strange or no results even when the call was
succesfull.
rtpdest set at 0.0.0.0:0, no Joitter information, no packetlosts,...
It seems that when the channel is hungup, some informations are lost
(often the cas with rtpdest) depending on the party hanging-up.
Also, some info are not clear for me, like what are the meaning of
-rtt? (Delay?)
-ssrc=1271016709 (what is the meaning of this number?
-themssrc
Any clue, docs, informations to make the rtp statistics working?
What do I wrong?
Olivier
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