[asterisk-users] TCP port, VPN and resolving the cutting voice problem

[Mark Deneen]

Any idea what is it about SIP over IAX2 that made such an improvement?

-M

On Thu, Dec 2, 2010 at 6:01 AM, Steve Totaro
<stotaro at asteriskhelpdesk.com> wrote:
>
> If getting a second circuit is out of the question.
>
> 1.  Switch to SIP
> 2.  Install and Learn Vyatta for QoS (Squid may help you quite a bit
> as well) as your router (or whatever you prefer)  I use the paid
> versions of Vyatta but the free edition should be sufficient.
>
> I did the same setup over OpenVPN VSAT links in Iraq, 700ms ping
> times.  I used GSM and some tricks on the Vyatta box.
>
> Originally, before I deployed the above, it was a wild west situation
> like what you have now.  Going from G729 to GSM made a big improvement
> in conjunction with QoS.
>
> My theory on that is that G729 is already a very lossy codec, so any
> more loss, garbled audio.  GSM is less lossy.
>
> Switch from IAX to SIP was another huge improvement, and then finally
> putting Vyatta and QoS as my router made calls almost crystal clear.
>
> There was the obvious lag time but users get used to that and wait a
> second or two before speaking so they don't talk over each other and
> the quality was five by five, except for solar flares, sandstorms,
> rain.  Things beyond my control.
>
> Thanks,
> Steve T
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>