[Freeswitch-users] Call quality

[Hynek Cihlar]

The motivation is to fix the audio distortions.

The architecture is as follows. One freeswitch acting as a "gateway" with
configured sip gateway connected to a VOIP provider. Second freeswitch
acting as a "controller" conects to the "gateway". Incoming call passes the
"gateway" to the "controller" where it is handled with ESL. Routing is
decided in an ESL application and in its simplest case is forwarded back to
the "gateway" through the VOIP provider to another PSTN number. The
forwarding is implemented in the "controller" such that uuid_originate is
issued and then bridged with the incoming call. No transocding should take
place, neither between the"gateway" and VOIP provider nor between the
"controller" and the "gateway".

Although no transcoding takes place and the system is idle, still there are
little distortions in the audio. Distortions that don't exist if I connect
to the VOIP directly with a SIP phone.

I'm looking for a way to find out, where the distortion is introduced.

Hynek



On Wed, Nov 2, 2011 at 1:40 PM, Sammy Govind <govoiper at gmail.com> wrote:

> 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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