[Freeswitch-users] Audio Codecs over Mobile Connection

Stanislav Sinyagin ssinyagin at gmail.com
Sat Jan 30 11:40:27 MSK 2016


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-scaleway-c1/
On 30 Jan 2016 06:41, "Oleg Stolyar" <olegstolyar at 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 at 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 at 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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>
>
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