[Freeswitch-users] choppy sound on voice recording and conference

[Anthony Minessale]

Is it our commercial G.729 module or something else?  We only have control
over our own code.

On Wed, Jan 18, 2012 at 6:47 AM, Anestis Mavro <devel at omninet.eu> wrote:

>  Hello,****
>
> ** **
>
> I have a problem with one of my sip trunks. The provider uses g729 with a
> ptime of 40ms and all incoming calls that are being recorded or go to the
> conference have a big delay and choppy sound, like a mismatch of ptime.***
> *
>
> ** **
>
> I have already tried to put rtp-autofix-timing=false into the profiles and
> even changed the inbound-codec…. to g729 at 40i, but nothing helps.****
>
> We have done a test with the provider switching for a test call to 20ms
> and the sound was perfect.****
>
> ** **
>
> The issue now is that the provider can’t keep this setting only for us, he
> wants to send with 40ms.****
>
> ** **
>
> Is there a way to configure it on Freeswitch?****
>
> ** **
>
> Thank you****
>
> Anestis****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
>
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-- 
Anthony Minessale II

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