[Asterisk-Users] sip/rtp performance monitoring

Andres andres at telesip.net
Sat Aug 6 15:06:17 MST 2005


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
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-- 
Andres
Network Admin
http://www.telesip.net





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