Re: [asterisk-dev] OPUS horrible quality with packet loss

On Mon, Apr 3, 2017 at 1:28 PM, Yury Tsaregorodtsev wrote: > after fixing attr->fec in open source edition of OPUS and applying > Alexanders patch: ASTERISK-25629 (Native Packet-Loss Concealment) > I have following results: > > I setup 2 hosts with asterisk 13.14.0 and made

Re: [asterisk-dev] OPUS horrible quality with packet loss

On Mon, Apr 3, 2017 at 3:38 PM, Yury Tsaregorodtsev wrote: > Even forced enabled jitter doesn't make asterisk to ignore late arrived > packets. > During my tests jb was always enabled (forced). > Hrm I'd think the jitter buffer should ignore or drop the late packets. That

Re: [asterisk-dev] OPUS horrible quality with packet loss

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.

Re: [asterisk-dev] OPUS horrible quality with packet loss

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

Re: [asterisk-dev] OPUS horrible quality with packet loss

On Mon, Apr 3, 2017 at 1:28 PM, Yury Tsaregorodtsev wrote: > > 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%

Re: [asterisk-dev] OPUS horrible quality with packet loss

after fixing attr->fec in open source edition of OPUS and applying Alexanders patch: ASTERISK-25629 (Native Packet-Loss Concealment) I have following results: I setup 2 hosts with asterisk 13.14.0 and made dozen calls in OPUS between them. On 2nd host I've simulated packet delay and loss: tc

Re: [asterisk-dev] OPUS horrible quality with packet loss

On Fri, Mar 31, 2017 at 6:54 PM, Yury Tsaregorodtsev wrote: > Examples of OPUS with 30% of LOSS with FEC on http://opus-codec.org/ > examples/ > Sounds too perfect. Why I can't achieve even similar quality on asterisk > with built codec by digium? > To simulate 30% packet

Re: [asterisk-dev] OPUS horrible quality with packet loss

> maybe version is too high [for that Native-PLC patch] You can answer that question yourself by adding debug output into codecs/codec_opus_open_source.c. For example, put one ast_log(…) for each "/* Case x". Case 5 and 6 are the perfect case without loss and therefore without activating

Re: [asterisk-dev] OPUS horrible quality with packet loss

Hi Alexander, I did try your opus version including native PLC patch (on asterisk 13.14.0, everything compiled without issues but anyway maybe version is too high ?) I wouldn't see much improvements on quality. From my personal point of view it's sounds little bit better, from point of view

Re: [asterisk-dev] OPUS horrible quality with packet loss

With Opus Codec, speech frames interdepend. When one RTP packet is lost, several frames are corrupted. Furthermore, when the RTP packet is not lost but just late, Asterisk simply forwards it down to the transcoding modules. Consequently, a late packet corrupts even more frames. The Opus Codec

[asterisk-dev] OPUS horrible quality with packet loss

Examples of OPUS with 30% of LOSS with FEC on http://opus-codec.org/examples/ Sounds too perfect. Why I can't achieve even similar quality on asterisk with built codec by digium? To simulate 30% packet loss I use: # tc qdisc add dev eth0 root netem delay 100ms