asterisk users - Apr 2012 - Monitoring voice-quality with sip/rtp/rtcp

After some customer complaints I find myself tcpdumping, gzipping and
transferring large packagedumps over the network to be analyzed.

While this manual process isn't a long-term solution, I'm evaluating
different options. Aside from the manual thing I could see two variants:
 - Dump the traffic (on the server or another via switch port
mirroring/monitoring) and analyze it with tshark
 - Analyze the traffic in asterisk

How do you monitor call quality for you services? (Right now I use
asterisk 1.4, but are in the process to migrate to 1.8 or 10.) I looking
for some ideas to setup this so I can eliminate this manual and
time-consuming process in the future. And know about the problems before
the customer complains about the quality..


Thanks in advance!

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

Le 09/04/2012 13:42, Johan Wilfer a ?crit :
> After some customer complaints I find myself tcpdumping, gzipping and
> transferring large packagedumps over the network to be analyzed.
>
> While this manual process isn't a long-term solution, I'm
evaluating
> different options. Aside from the manual thing I could see two variants:
>   - Dump the traffic (on the server or another via switch port
> mirroring/monitoring) and analyze it with tshark
>   - Analyze the traffic in asterisk
>
> How do you monitor call quality for you services? (Right now I use
> asterisk 1.4, but are in the process to migrate to 1.8 or 10.) I looking
> for some ideas to setup this so I can eliminate this manual and
> time-consuming process in the future. And know about the problems before
> the customer complains about the quality..
>
At first, if your Asterisk is in a VM install it on the real server, it 
solved us on some installations.

To monitor the traffic, you can use voipmonitor.org

-- 
Daniel

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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