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

[Marc De Corny]

My apologies, i thought i had.
Let me try again
Marc

On 2 Feb 2011, at 18:11, David Ponzone <david.ponzone at ipeva.fr> wrote:

> Marc,
> 
> I would recommend you resend your message without replying to another one, so it has its own thread.
> 
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
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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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