[asterisk-users] Monitoring voice-quality with sip/rtp/rtcp
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
Decatur, GA 30030
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
>> 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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