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

[Johan Wilfer]

2012-04-09 22:32, Johan Wilfer skrev:
> 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!

I contacted them and will use a server connected to a switch-port in
mirroring mode. The gui seems like a great tool in troubleshooting.

Nobody uses the rtcp-stats in asterisk for quality monitoring?

Other suggestions?

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

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