Jicofo does not mute Jigasi clients

[weltfahr]

Description

Our Jigasi clients cannot mute themselves and start unmuted although in our Jigasi config we have set:

org.jitsi.jigasi.ENABLE_SIP_STARTMUTED=true

The log does show:

jicofo-1   | Jicofo 2025-12-02 11:55:34.392 INFO: [37] [room=<the room>@muc.meet.jitsi meeting_id=dcbe906f-847d-48a3-aff2-797bb05acf4a] JitsiMeetConferenceImpl.handleMuteRequest#1620: Will mute <the room>@muc.meet.jitsi/<the Jigasi client id> on behalf of <the room>@muc.meet.jitsi/<the Jigasi client id> for AUDIO

But the calculated colibriMuteState is false.

I am wondering whether this line is wrong:

jicofo/jicofo/src/main/java/org/jitsi/jicofo/conference/JitsiMeetConferenceImpl.java

Line 1607 in 6301519

colibriMuteState = !chatRoom.isMemberAllowedToUnmute(toBeMutedJid, MediaType.AUDIO);

The mute state only depends on the permission to unmute. Shouldn't it also depend on doMute? Or am I missing something?

[damencho]

I guess you are not running using translator?

[weltfahr]

Thanks for your reply, @damencho!

You are right. We only use Jigasi for phone users to dial in to the meetings.

[damencho]

For dial-in you can do mixer moder and translator mode where the mixing happens on the sip side, requirement is that the sip side needs to support multistreams.
For the translator mode we do actually mute the stream. The mixer mode is something we use only for transcriber and havn't been tested with all the features.

[weltfahr]

Thank you for the explanation, @damencho!

We are currently using jitsi-contrib/jigate, which relies on FreeSWITCH. Based on that, I assume FreeSWITCH does not support translator mode.

Could you confirm if that assessment is correct?

[damencho]

FreeSWITCH supports webrtc so I suspect it does support multiple streams, but I don't have experience with it to advice you how to configure it. I know asterisk supports that but also I have never used that ...