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

[Steve Totaro]

No but if google my posts about IAX2, you will see that I have seen
IAX2 cause so many problems with audio, I have made a good amount of
money just switching customers to SIP.  Even a large ITSP.

I have found it to be responsible for poor audio in over a dozen cases
and after switching to SIP, the audio was five by.

Several people that work for Digium that will remain anonymous, have
said to only use IAX when absolutely needed.

You will also see people agreeing with me and others that have no issues.

I just use SIP.

Thanks,
Steve T

On Thu, Dec 2, 2010 at 9:27 AM, Mark Deneen <mdeneen at gmail.com> wrote:
> 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
>>
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