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

[David Ponzone]

Marc,

I would recommend you resend your message without replying to another one, so it has its own thread.


David Ponzone  Direction Technique
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Le 02/02/2011 à 19:03, Marc De Corny a écrit :

> 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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