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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