[asterisk-dev] OPUS horrible quality with packet loss

Kevin Harwell kharwell at digium.com
Mon Apr 3 14:22:06 CDT 2017


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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