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

[Carlos Alvarez]

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
TelEvolve
602-889-3003
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120411/ca7ed1ca/attachment.htm>