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

[Johan Wilfer]

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