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

[Johan Wilfer]

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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