Bad quality of an audio conference with opus participants

Hi Folks,

We found some strange behavior of the Asterisk that works as the audio conference bridge.
We have a solution:

  • The Hardware server
  • OS version is Linux version 3.10.0-693.21.1.el7.x86_64 ( (gcc version 4.8.5 20150623 (Red Hat 4.8.5-16)
  • Asterisk 13.18 with confbridge and chan_sip
  • Last version of opus codec 1.3.0

The confbridge has settings:

The scenario is:

  1. More then one opus participants join to a the same conference bridge
  2. Bad quality occur if one of them on mute (not server mute) or just silent. it affect only who on mute/silent.
  3. The quality is good when both of them are speaking at the same time.

The issue has been resolved by changing mixing_interval to 20.
I read documentation and found that 40 ms should cover sample rates 8-96 kHz. So, opus has 48 kHz but by some reason it doesn’t work properly.

Is it some bug our maybe it is normal behavior for opus with mixing_interval >= 40 ?

Also I tried pjsip with 13.18,13.20,13.21 and last 15.4 version. I got the same results.