[asterisk-users] Voice quality is bad from one side and good from another side

[Steve Totaro]

On Wed, Mar 5, 2008 at 6:53 PM, Steve Totaro
<stotaro at totarotechnologies.com> wrote:
> On Wed, Mar 5, 2008 at 6:09 PM, Alex Balashov <abalashov at evaristesys.com> wrote:
>  > bilal ghayyad wrote:
>  >
>  >  > Hi all;
>  >  >
>  >  > I have two asterisk boxes installed in two separated
>  >  > sites, the internet bandwidth between them is very
>  >  > good and I am using G729 codec to communicate with
>  >  > them and IAX.
>  >
>  >  Try playing around with the adaptive / fixed jitter buffer settings for
>  >  IAX2.  Also, if it's consistently a problem found on one side, consider
>  >  any resource consumption and/or processing issues that may be occuring
>  >  on the sending side that is creating the problem.
>  >
>  >  Is this "very good bandwidth" just good bandwidth, or also good latency,
>  >  by which we really mean _consistent_ latency?  Or is it highly bursty
>  >  and variable?  This will create jitter and break voice, even if the
>  >  amount of potential throughput is very high.  Consider satellite links
>  >  if you need a good example.
>  >
>  >  --
>  >  Alex Balashov
>  >  Evariste Systems
>  >  Web    : http://www.evaristesys.com/
>  >  Tel    : (+1) (678) 954-0670
>  >  Direct : (+1) (678) 954-0671
>  >  Mobile : (+1) (706) 338-8599
>  >
>
>  Try using SIP.  Post back with results.
>
>  Thanks,
>  Steve Totaro
>

Sorry to reply to my own post but if NAT is an issue, consider OpenVPN
and SIP.  I bet your calls become crystal clear, if not, try GSM
instead of G729.  Actually, that might be where you want to start if
changing to SIP is non-trivial in your setup.

Thanks,
Steve Totaro