Thanks! It's interesting that Silk is less CPU intensive than Opus. However, at this point I would be willing to spend extra CPU cycles if it gets me more consistent call quality on an iffy connection. Anyone has experience with this? On Sat, Jan 30, 2016 at 12:40 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
Of course jitter buffer is independent from codec, but its functionality affects the quality.
Yes SILK is inherited by Opus, but in general SILK is much cheaper in terms of CPU cycles for encoding and decoding.
You can see the difference here: https://txlab.wordpress.com/2015/04/10/testing-freeswitch-performance-on-sca... On 30 Jan 2016 06:41, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
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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