Telephone connection to is garbled

I stumbled across whilst installing as it was included in it. I’ve been testing it for a couple of weeks and at first it worked very well. Recently however, when calling into a conf call using the telephone numbers the sound has been garbled with an ongoing background noise of pops and clicks that make it of little use to the person calling in by phone.

Also, in a series of test calls I made today the conf is not picking up any sound made by the person calling in by phone. They could hear (pops squeeks clicks n all) but not speak.

So there seems to be some issue with the telephone call in aspect.

Any update as to what the issue is and plans to fix it much appreciated.

PS I used Chrome, Vivaldi and Brave browsers in these tests.

Is it like robotic sound? Can you send me the meeting name you were using (privately if you want) so I can investigate and report it to our provider.

Never mind I just reproduced it.

Yes - like a robotic sound. I recorded it. Would you like the file ?

I have had that result on several meeting names - I was trying different ones to see if it would go away - and it was the same in all instances.

I’ve just done one on:

The robotic clicks were there, this time the person calling in could be heard but of course through all the clicks and pops.

Yep, I reported it and will come back when I have more info.
Thank you for the report!

Thank you for the prompt response. The service is v good, particularly with the PSTN call in as it lends itself perfectly as a base, a sort of virtual studio, for running online radio shows with a call in facility for listeners. The volume slider on each caller is also v useful - an audio mixing desk inside the call.

The issue of garbled noise interference when phoning in to a conference has been cleared up. I have had clean audio on all calls over the past 5 days. V good call quality.

One last question. Phone callers into the conference can be muted. Once they have been muted, how do they unmute themselves ? Is there a phone key or keys they need to press to unmute themselves ?

Glad to hear it.

There is no such option for now.

A follow on from the questions I was asking earlier in the year.

On there is an option under Settings > More titled Everyone starts muted. This works when someone joins the call via the url but when they join via calling on over one of the PSTN numbers it does not work. Also the option to mute/unmute each phone caller does not work although it is possible to mute them by sliding their volume down to zero.

Is there a way for the admin to mute/unmute attendees and particularly those that are connecting by telephone ? I would like it so that whoever connects to the call, either by internet or by phone, starts off being muted and that I (the admin) can mute/unmute them as and when I want to bring them into the call.

If this is not doable at right now are there any plans to make it a feature soon ? And if not, can this feature be achieved if I were to install jitsi on my own server ? … although I guess I am then faced with setting up all my own pstn call in numbers which would sort of defeat the whole object of the exercise.

Any pointers etc much appreciated.

Pstn connected participants cannot be remotely muted as their is no way for them to unmute. There are no plans for changing that, those participants needs to manage that on their own.

You can achieve that by implementing some IVR on your sip side so people from phones can press a number or combination of numbers to unmute and modify jicofo so pstn users can be remotely muted.
But I don’t see a point of doing that as why would those participant use the IVR to unmute when they have such a button on their phone and basically the current situation is what makes sense from users point of view.

The thinking behind it is that you could run a training session etc where everyone other than the main speakers are muted, a sort of Lecture mode. The admin would unmute any phone caller that had indicated they wanted to ask a question but o course to do that there would need to be some IVR or dtmf tone to allow phone callers to indicate to the admin that they had a question etc.

Its standard fare on most conference call systems (ie freeconferencecall) where people call in by phone, are auto-muted from the start and only the admin can unmute them. That would be lecture mode I guess. Q&A is where the callers can unmute themselves and talk away.

Anyway, thank you for your answer and comments.

Unmuting people has privacy concerns, that’s the reason we are against it and people needs to be in control of their own privacy. When someone is asked a question he controls his own mic and can unmute, answer and mute again.

Understand the privacy concern. If they could just join the call by phone and be auto muted when they join that would be useful - similar to the way people dial in to a conference call simply to listen to it - if you have a lot of people phone in and they all have their mics unmuted then the background noise makes it unworkable. This looks like something I’d need to do on our own install and with IVR etc.

Thx again for your comments etc.

Is there a way to achieve both goals. I agree that the moderator unmuting can be a privacy issue but it is also true that large conference calls with unmuted dial in users can become unusable.
Can we make it that the moderator can mute everyone including dial-in users but cannot unmute them. The dial-in users should then be able to unmute themselves with a key combination.

We just murged such un option in jicofo and jigasi. You need to enable it in your deployment with some configs. Jigasi is sending sip INFO messages to inform for muting and is processing sip INFO messages to be informed when the IVR mutes it.
For jicofo this is: org.jitsi.jicofo.SIP_MUTE_ENABLED
And for jigasi: org.jitsi.jigasi.ENABLE_SIP_STARTMUTED

That’s great news! Will that appear soon in the Jitsi Meet service? That is what we are using. Thanks

It may take some time before we deploy it on, there are more priority tasks right now.

What! My issue is not the most important one :wink: Just kidding of course. But, if a beer gets it slipped in sooner rather than later, I’m buying. Dial-in is very important in our application. Thanks and I look forward to the change whenever it arrives.

Hmm . . . it looks like we are going to choose Jitsi Meet as our solution and this is a big issue for us. We don’t have much of a budget but how much would it cost to sponsor this as an addition to Jitsi Meet?

Are you asking about muting and unmuting pstn users? If this is the case, it may come the following days … I’m working on that.