[asterisk-bugs] [JIRA] (ASTERISK-27851) app_confbridge: Opus participants have bad quality in confbridge audio conference with non-20ms mixing interval

[Aleksandr Salanov (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27851?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=243919#comment-243919 ] 

Aleksandr Salanov commented on ASTERISK-27851:
----------------------------------------------

Hello Team,

Do you have any updates?

> app_confbridge: Opus participants have bad quality in confbridge audio conference with non-20ms mixing interval
> ---------------------------------------------------------------------------------------------------------------
>
>                 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: 13.18.0, 13.20.0, 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
>              Labels: pjsip
>
> 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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