[sip-comm-dev] poor voice quality in sip communicator


#1

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 :frowning: 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

路路路

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#2

Zden臎k Lou啪ensk媒 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 :frowning: 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

路路路

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#3

Zden臎k Lou啪ensk媒 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 :frowning: 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

路路路

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