poor voice quality in sip communicator

Hi guys,
you're doing really great job on sip communicator.
Me and another 3 friends tried voice communication over sip
communicator and it's cool, but we have a problem with
quality of
sound. We tried to connect to the conference room and chat
together.
But some of us were heard very silently or interruptedly :(
We were
testing this on our server with asterisk installed on it. 3
of us were
behind some firewalls and NAT, so this might be problem.
Do you have some observations about this? Or could this
problem be caused
with bad asterisk configuration or java media framework
configuration?

Thanx much,
Zdenek Louzensky

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Zdeněk Louženský wrote:
> Hi guys,
> you're doing really great job on sip communicator.
> Me and another 3 friends tried voice communication over
sip
> communicator and it's cool, but we have a problem with
quality of
> sound. We tried to connect to the conference room and
chat together.
> But some of us were heard very silently or
interruptedly :( We were
> testing this on our server with asterisk installed on
it. 3 of us were
> behind some firewalls and NAT, so this might be
problem.
> Do you have some observations about this? Or could this
problem be caused
> with bad asterisk configuration or java media framework
configuration?
>
>   
What codec are you using? If you are in one network g711a/u
is ok. But 
when you are connection through internet
low bandwidth codec must be used. Right now we are
implementing ilbc and 
speex. The speex implementation will be committed in few
hours )
What is the problem with quality - lost packets(interrupting
in the 
sound) or noise ?

damencho

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> Zdeněk Louženský wrote:
>> Hi guys,
>> you're doing really great job on sip communicator.
>> Me and another 3 friends tried voice communication
over sip
>> communicator and it's cool, but we have a problem
with quality of
>> sound. We tried to connect to the conference room
and chat together.
>> But some of us were heard very silently or
interruptedly :( We were
>> testing this on our server with asterisk installed
on it. 3 of us were
>> behind some firewalls and NAT, so this might be
problem.
>> Do you have some observations about this? Or could
this problem be caused
>> with bad asterisk configuration or java media
framework configuration?
>>
>>   
> What codec are you using? If you are in one network
g711a/u is ok. But
> when you are connection through internet
> low bandwidth codec must be used. Right now we are
implementing ilbc and
> speex. The speex implementation will be committed in
few hours )
> What is the problem with quality - lost
packets(interrupting in the 
> sound) or noise ?

> damencho

We are using g711 u codec, but we also tried Xlite and it
used the
same codec and with Xlite the quality of sound was much
higher, almost
perfect (so I think this is not the problem).
The problem with quality is mainly the quality of noise,
it's hard
to understand what are others saying. Sometimes also happens
some
interruptions, but not so often.

We will try speex codec and will see whether the quality is
higher.


Thanx much danencho.

zdenek

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