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

Alex Balashov abalashov at evaristesys.com
Mon Apr 9 15:56:26 CDT 2012

OpenVZ is not really "virtualisation", though for some reason people insist on throwing it into the same discursive space as Xen, VMware, HyperV, etc.

Alex Balashov - Principal 
Evariste Systems LLC 
235 E Ponce de Leon Ave 
Suite 106
Decatur, GA 30030 
Tel: +1-678-954-0670 
Fax: +1-404-961-1892 
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

Johan Wilfer <lists at jttech.se> wrote:

>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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