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 <kharw...@digium.com > <mailto:kharw...@digium.com>> wrote: > > > > On Mon, Apr 3, 2017 at 1:28 PM, Yury Tsaregorodtsev <aero.1...@icloud.com > <mailto:aero.1...@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. > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com > <http://www.api-digital.com/> -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > <http://lists.digium.com/mailman/listinfo/asterisk-dev>
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