[Freeswitch-users] Call quality

[Sammy Govind]

Please be sure that no-transcoding is done. Make sure the codecs of inbound
and outbound legs are same on provider<->gateway and then
gateway<->controller and way back.
If transcoding isn't the case, in my experiences virtual machines show
these little distortions in call.

On Wed, Nov 2, 2011 at 5:56 PM, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:

> 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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