[asterisk-dev] OPUS horrible quality with packet loss

Yury Tsaregorodtsev aero.1080 at icloud.com
Mon Apr 3 15:38:43 CDT 2017


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 <mailto:kharwell at digium.com>> wrote:
> 
> 
> 
> On Mon, Apr 3, 2017 at 1:28 PM, Yury Tsaregorodtsev <aero.1080 at icloud.com <mailto: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/ <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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