[Freeswitch-users] Call quality

Sammy Govind govoiper at gmail.com
Wed Nov 2 15:40:48 MSK 2011


Please explain your query bit more, keeping in mind the ideal world
scenario as your's, what exactly you require to know.

On Wed, Nov 2, 2011 at 5:09 PM, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:

> In this case, it's the "ideal" world. Testing environment, all media
> servers on one OS, one native server box, CPU idle, no load, no
> transcoding. One media switch is connected to a VOIP provider though
> through SIP. But the provider sits in the same telehouse and again no
> transcoding takes place.
>
> Any idea where to direct the investigation?
>
> Hynek
>
>
>
>
> On Wed, Nov 2, 2011 at 6:42 AM, Sammy Govind <govoiper at gmail.com> wrote:
>
>> Network latency, transcoding processing, jitter handling, call session
>> establishment on each Media-Server/switches etc all add up in overall call
>> quality factor.
>>
>> On Tue, Nov 1, 2011 at 11:39 PM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>>
>>> Would anybody know, if the call quality could be dependant on the
>>> number of media switches the call is routed through?
>>>
>>> Currently I'm testing a system with several freeswitches in the call
>>> way (up to 4) and I'm noticing very short gaps, not very disturbing
>>> but noticable.
>>>
>>> What's your experience?
>>>
>>> Sent from my mobile device
>>>
>>>
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>>
>>
>>
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>>
>
>
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