Thanks Stanislav!

I understand that jitter buffer is not necessarily codec dependent and that SILK is actually the foundation of Opus, so I assume their quality is comparable.

On Fri, Jan 29, 2016 at 8:26 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
add SILK to the list (it's actually what Skype uses), and jitter buffer.

but I don't have much practical experience.

The quality of the sound depends also on the particular SIP client
implementation.



On Fri, Jan 29, 2016 at 11:32 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
> Hi guys,
>
> does anyone have experience using different codecs to reduce choppiness and
> metallic/robotic sound when connecting to FreeSWITCH from a mobile device?
>
> I know both OPUS and G.722 have packet loss concealment but I wonder if
> someone actually has practical experience with different codecs over iffy
> mobile connections.
>
> OPUS vs. G.722 vs G.711
>
>
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