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If the customers are using an ATA or other VOIP device that
supports RTCP, then you can often get packet loss and jitter stats
by
extracting the RTCP packets and analyzing them.
This will actually give you the packet loss and jitter that
the customer is seeing in the received RTP stream from you.
A combination of Tetheral and grep or perl
can get you along way in capturing and analyzing this data.
Jim
----- Original Message -----
Sent: Saturday, August 06, 2005 9:43
AM
Subject: [Asterisk-Users] sip/rtp
performance monitoring
I'm currently running asterisk to provide VoIP services to
clients of
the ISP I work for.
I would like to be able to tell if I
am loosing packets and/or are
having other issues with any of the voice
streams, so I can address them
proactively.
I'm not particularly
interested in spending oodles of money buying one
of the commercial
analysis tools. Is there some open source tool (or
something I
can monitor in asterisk) which will tell me if I'm missing
packets or
similar? I realize this will likely be only from the
customer
towards me since I can't really monitor at the customer
end.
-forrest
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