[asterisk-users] Monitoring voice-quality with sip/rtp/rtcp

Johan Wilfer lists at jttech.se
Mon Apr 9 15:32:25 CDT 2012

2012-04-09 20:22, Carlos Alvarez skrev:
> On Mon, Apr 9, 2012 at 10:25 AM, Administrator TOOTAI
> <admin at tootai.net <mailto:admin at tootai.net>> wrote:
>     At first, if your Asterisk is in a VM install it on the real
>     server, it solved us on some installations.
> We've gone away from VMs altogether.

I use openVZ to run multiple asterisks on the same server. This works
well and has done for some time. But currently once a week for about
10-15 minutes calls sound like packetloss/jitter occurs. But a week of
traffic captures is heavy... So I need to automate this.

>     To monitor the traffic, you can use voipmonitor.org
>     <http://voipmonitor.org>
> We purchased the commercial version with a GUI and will tell you that
> the cost/benefit is very clear.  Great tool, pretty cheap ($1k I
> think).  Responsive support.

Sounds very reasonable. Do you run this on a dedicated server, and
configured the switch to duplicate the traffic to the quality server? Or
do you run this on the same server as asterisk?

Thanks for the suggestions!

Johan Wilfer                 email: johan at jttech.se
JT Tech | Developer          webb: http://jttech.se

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