On 28 Apr 2007, at 09:05, Matt Gardner wrote:
Ok this is my first post and I will try to keep it short.
I have searched everywhere and haven't found an answer to my question
I have two Trixbox servers that are connected over the Internet via
an IAX2 connection. We are experiencing very poor sound quality.
I have tried many different codecs gsm, ilbc, g729, g711 and all
seem to have the same problem. (All though g729 seems to work the
best but still isn't reliable) The problems are intermittent
sometimes the sound will cut out for 3-4 seconds and other times
the sound will just be loosing every other word, and other times it
sounds just fine.
Also, we have been using Skype over this same Internet connection
and have very good sound quality with very few lost words.
So here are my questions.
First, is it a correct assumption to say that because Skype works
well over this connection then I should be able to get asterisk to
work over this connect. I am hoping that Skype isn't "better" then
asterisk in this area.
If I should be able to get the same sound quality could you point
me in the right direction on how to achieve this. (I have tried
messing with the jitterbuffer but haven't been able to find very
good docs on how to utilize this functionality so about all I have
done is set jitterbuffer=yes)
Try making a call, and then use iax2 show netstats
(I think that is the syntax in 1.4, I'm off-line at the moment and my
memory is going :-( )
This will give you some statistics collected by the jitterbuffer code.
Post them here and we will take a look.
As a reference point, we get _perfect_ g711 calls over IAX from the
UK to new york, so it is possible.
Theoretically Skype _can_ produce better audio quality than asterisk
as it supports a wideband codec
but in practice asterisk/iax should be just as good and somewhat
more predictable as Skype's routing
varies from call to call.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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