On Mon, Apr 3, 2017 at 1:28 PM, Yury Tsaregorodtsev <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/ 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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