[asterisk-dev] OPUS horrible quality with packet loss

[Matt Fredrickson]

Hey Yury,

Thanks for letting us know about the challenges you're having with the
Digium sanctioned version of codec_opus.  We're going to try to lab up
the scenario you have found and see if there is a bug in our
encoder/decoder processing that's making it perform worse under your
scenarios.

Sorry about the trouble - as you know, with software, it tends to
improve as it goes forward but sometimes it takes time.  Hopefully we
can figure out what's different in the Digium codec quickly and
release a new version of it with the fix.

Best wishes,
Matthew Fredrickson

On Mon, Apr 3, 2017 at 3:38 PM, Yury Tsaregorodtsev
<aero.1080 at icloud.com> wrote:
> Even forced enabled jitter doesn't make asterisk to ignore late arrived
> packets.
> During my tests jb was always enabled (forced).
> I made also tests without delay, only with drops - quality of Digium Opus
> not acceptable for voice conversation anyway.
> The fact is open source opus handles better dropped packets.
> You can't disagree with quality after all,
> I can send you recorded samples, you can compare.
>
> Yury
>
>
> On 3 Apr 2017, at 21:22, Kevin Harwell <kharwell at digium.com> wrote:
>
>
>
> On Mon, Apr 3, 2017 at 1:28 PM, Yury Tsaregorodtsev <aero.1080 at icloud.com>
> wrote:
>>
>> <snip>
>>
>> MOS on calls using open source opus higher almost twice.
>> Subjective opinion regarding audio quality: using open source codec
>> quality almost same as in example on http://opus-codec.org/examples/ with
>> 30% loss and FEC, acceptable for ears, but
>> using digium opus quality is not acceptable, a lot of spikes,
>> interruptions.
>>
>> I also double checked the fact before applying ASTERISK-25629 patch
>> asterisk don't drop lately arrived RTP.
>
>
> Dropped packets and late arriving packets are two separate issue and are
> handled as such in Asterisk. The Digium Opus codec can handle dropped
> packets by enabling FEC. If the problem is late arriving packets than
> applying a jitter buffer to the audio stream exhibiting the problem should
> help alleviate that.
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-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA