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 (firstname.lastname@example.org) (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:
- More then one opus participants join to a the same conference bridge
- Bad quality occur if one of them on mute (not server mute) or just silent. it affect only who on mute/silent.
- 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.