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
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
James H. Thompson
jht@lj.net
----- Original Message -----
From: Forrest Christian
To: Asterisk Users Mailing List - Non-Commercial Discussion
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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Forrest Christian wrote:> 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.You could use Ethereal. It has an RTP tool that tells what the jitter and packet loss is. And by the way, if your customers have Sipura units then you can indeed monitor their end as well. The latest firmware versions include a feature where they send all call statistics(jitter, packet loss, ..etc) in a header with the BYE message. We have integrated it into our system so when our support people open up the customers account, then can click on a link to see all the RTP stats of all calls made and received by the customer. Its quite nice and quickly gives a snapshot of the quality of service the customer is receiving.> > -forrest > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-- Andres Network Admin http://www.telesip.net