[asterisk-users] Monitoring voice-quality with sip/rtp/rtcp
lists at jttech.se
Mon Apr 9 06:42:12 CDT 2012
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
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