[asterisk-users] Two Connected Servers Sound Quailty

Tim Panton tim at mexuar.com
Sun Apr 29 03:55:43 MST 2007

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


More information about the asterisk-users mailing list