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

[Administrator TOOTAI]

Le 09/04/2012 13:42, Johan Wilfer a écrit :
> 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