[Freeswitch-users] Voice quality monitoring via loopback on sip phone

[Marc De Corny]

Hi all,
I am thinking of a way of testing the quality of a voice call. 

Ideally i would like to create a call from FS with a loopback parameter that tells phone to answer the call automatically and loopback the RTP. This is a common functionality in most phoned (for example cisco)
The sent and received packets could be compared for jitter, latency and packet loss and a result extracted.

Does this already exist? I know there some specific tool that accomplish this bit they are expensive and awkward to use. Starting and retrieving the result from an api would be fantastic.


On 2 Feb 2011, at 17:43, Massimiliano Ravelli <massimiliano.ravelli at gmail.com> wrote:

> I need to move our current pbx to a new hardware.
> How can I disable the g729  licences on the old server and enable them on the new one ?
> 
> The current production server has an old installation of mod_com_g729: can I upgrade it to a more recent one without upgrading freeswitch ?
> Does mod_com_g729 upgrade need to stop running calls ?
> 
> Thanks in advance
> Massimiliano
> 
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