[asterisk-bugs] [JIRA] (ASTERISK-27851) Opus participants have bad quality in confbridge audio conference

[Aleksandr Salanov (JIRA)]

Aleksandr Salanov created ASTERISK-27851:
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             Summary: Opus participants have bad quality in confbridge audio conference
                 Key: ASTERISK-27851
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27851
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Applications/app_confbridge, Codecs/codec_opus
    Affects Versions: 13.20.0, 13.18.0, 13.21.0, 15.4.0
         Environment: PowerEdge R630/2 x Intel Xeon E5-2667 v4 3.2GHz()/RAM 32Gb/
            Reporter: Aleksandr Salanov
            Severity: Minor


We caught some strange behavior of the Asterisk that works as the audio conference bridge.

We have a solution:
1) The Hardware server
2) OS version is Linux version 3.10.0-693.21.1.el7.x86_64 (mockbuild at x86-ol7-builder-02.us.oracle.com) (gcc version 4.8.5 20150623 (Red Hat 4.8.5-16)
3) Asterisk 13.18 with confbridge and chan_sip
4) Last version of opus codec 1.3.0
5) The confbridge has settings:
[default_bridge]
type=bridge
video_mode=none
mixing_interval=40
sound_join=en/beep
sound_only_person=en/beep
sound_leave=en/nc_custom/confbridge-leave

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.

Testing other version of the Asterisk:
I tried pjsip with 13.18,13.20,13.21 and last 15.4 version. I got the same results.

Workaround is:
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 this the expected behavior or a bug?



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