[asterisk-users] Two Connected Servers Sound Quailty

[Noah Miller]

Hi Matt -

> 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.
>
> 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.

Yes, it's certainly possible to get good quality with asterisk.  Skype
is not better, they just build in more default latency.  I've never
measured exactly, but it seems that Skype calls typically have a built
in buffer between 250ms and 1000ms.  Asterisk, will try to use as
little latency as possible.  You've set jitterbuffer=yes, but you'll
also need to set maxjitterbuffer (probably to 1000), resyncthreshold
(probably to 1000), and maxjitterinterps (10 is a good safe value).
You can try adjusting these values to see how it affects your calls.

You'll also want to do something about QoS.  If you don't, the next
time you try to FTP a file, it will try to grab all your available
bandwidth, whether or not you are on a call.  This will surely screw
up your call quality.  If your routers have QoS options, you can
ensure that your voice traffic will get first dibs on bandwidth.
You'll need to configure QoS on both ends.


- Noah