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

Carlos Alvarez carlos at televolve.com
Wed Apr 11 08:15:25 CDT 2012

On Wed, Apr 11, 2012 at 4:29 AM, Johan Wilfer <lists at jttech.se> wrote:

> 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?
Cheap dedicated server with a span port on the switch.  We *never* run
anything other than Asterisk on a production voice server.

Carlos Alvarez
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