Over the past three weeks I've tried a few different conferencing solutions, including jitsi. I'll give it another try with this update.
My use case is I take weekly music lessons, and now they are virtual. The problem is the DSP done on audio was designed for speech. If my teacher is explaining something then plays an example on his bass, it usually sounds terrible, maybe even inaudible.
I send him pre-recorded mp3s of cover songs; ideally he could listen to it and I could comment in real time about places where things could be improved. Instead, if he is playing any music on his system, I hear nothing -- no music, no talk. It seems like the software thinks "Hey, this participant is listening to non-conference audio, so I'll just mute him (at least on skype). I'd love there to be a half duplex audio button so none of the DSP shenanigans are needed, and a high quality audio stream would be sent.
If the website doesn't want to offer a control to switch this on/off, I'm confident this can be done by a browser extension in no time (which would have the benefit to work for all websites).
Audio processing is a risky move -- so hard to get right. We've been using https://team.video at work, and one thing I absolutely love about it is how they handle audio / muting.
When you're speaking, you don't have to wonder if others can hear you because your microphone pulses in green visually as you speak. If your audio isn't working it shows in yellow with no pulsing, and you and everyone else can see your audio is not flowing.
Also, if someone else forgot to mute and their kid is making a ruckus, you can just mute them. You don't have to wait for a moment to interject and ask them verbally, you can just go ahead and do it.
Or, when you see someone else in their video feed trying to speak up, but they forgot to unmute, you just unmute them. No everyone saying, "you're muted" over each other.
It takes a second to get used to the idea that everyone has all the power, but in practice it just makes everything go way smoother.
Here's something that a colleague passed along to a group of CS profs.
It's written by a music professor and geared toward using Zoom for music, but several of us Zoom newbies found it to be helpful more generally. He mentions the issue of disabling the speech-centric audio postprocessing.
Disclaimer: The author apparently makes his money selling eBooks, so you may want to skip through the several pages of promotional material at the beginning of his PDF to get to the good stuff.
Use TeamTalk[1]. If you need high audio quality, TT beats everything else you can find, maybe except very expensive software for radio stations. I've successfully used it to stream music and it works.
It's Teamspeak and Discord like, so you need to connect to a server, either public or self-hosted, join or create a channel, and then you will be able to talk to everyone on that channel. This is perfect for permanent communities where people just hang out, but works for one-offs too. It works on Windows, Linux, Mac, iOS and Android, no web access. The server is also available for Raspberry Pi. Half of it is open-source, but the core SDK needs a license if you're developing with it. The program itself is free, even for commercial use.
It uses Opus and lets you adjust the quality and processing, so you can get a lot out of it. We've been using it in our community for about 10 years now, including for radio broadcasting, and we haven't managed to find anything better since. I know of one local radio station and recording studio who successfully use it for remote work now.
To get the best experience, disable all audio processing in the preferences, on the sound system pane, so duplex mode, automatic gain control and noise reduction should be off. If you're on Windows, use Windows Audio Session as the backend for lowest latency.
Then, connect to a server, I use the German one for public stuff, as I'm close to it geographically and you don't need to register for it, but use whichever you want. After connecting, create a channel with application set to music, bitrate set to 150000 and channels set to stereo. Those are, at least, the parameters I use, and they work great. You can adjust the rest as you see fit.
There are some video and screensharing capabilities as far as I know, but I haven't used them. Audio is definitely the primary focus. If you need any assistance, my username here at gmail dot com is the way to go.
[1] bearware.dk for desktop, App Store and Google Play for iOS and Android.
ps. I'm not affiliated with the company in any way, it's just a tool I use daily and would recommend to anyone who knows his way around the computer. It definitely doesn't pass the grandma test, though.
I work for a company that builds virtual classrooms based on webrtc. Our customers are mostly business schools, but we have some music schools. For them we activate a different profile that disables all audio processing and selects the music profile of opus (opus are in fact 2 different codecs, one aimed at speech, one aimed at music). It would likely be very easy to do something like this in jitsi meet as well, since webrtc has everything onboard. The tricky part is that you also need to disable echo cancellation. So everyone must be wearing headsets and so on.
For this use case, you need selective fidelity and shared control over a sampler, with each sample having low resolution video, high quality audio and an arbitrary number of tags or notations (with a time range).
I want to like Jitsi, but I can't. I've been testing it out on a variety of servers, bare-metal and cloud with Debian and CentOS. Regardless of platform, it doesn't scale, it eats memory like peanuts, and can saturate even 10GbE network connections. The service, as opposed to the server, clearly works well. But, the service doesn't have anything like the load that Zoom, Teams, Hangouts, etc. must deal with.
I've been helping quite a few people set up Jitsi lately. The software works reasonably well, although documentation is lacking and a lot of configuration options are named terribly, but one thing I've noticed is people's expectations about bandwidth usage are way lower than actuality, especially outbound from the server. But a bit of napkin math suggests that Jitsi isn't doing anything fundamentally inefficient here; one high-res stream plus N low-res streams transmitted to N participants is just a lot of bandwidth.
There are a few of things to be aware of when deploying your own instance in jitsi-meet/config.js:
Firefox simulcast is still experimental (edit: and disabled by default), so Firefox is only sending HD to the video-bridge and no LD stream. The HD is then relayed to everyone even as a thumbnail.
If you don't need to see everyone's video all the time, set channelLastN: 5 or a similar number, and only the last N speakers video will be broadcast
If you don't need 720p (1280x720), change the constraints: video section to be something like 360p (640x360)
Enable layer suspension so that HD is not sent to the server when not needed: enableLayerSuspension: true
I haven't tried this at scale, but was pleased with 6 participants on my $200 r710 hardware with 10megabits upload connection service. Memory usage is minimal, couple GB tops. Not running high-res, but was good enough. I mean they develop this and make it freely available for self-host, I'm pleased.
It could be worth reporting some feedback to them directly, if you haven't already?
They're quite responsive on GitHub ( https://github.com/jitsi/jitsi-meet ) and I'm sure additional details regarding any bottlenecks would be appreciated.
Identifying even modest improvements now could result in large memory and bandwidth resource savings over the next few weeks and months.
I've been running multiple deployments of it on Azure, on VMs with as low as 1 core/2 GB. I literally set it up every few days from scratch using https://github.com/rcarmo/azure-ubuntu-jitsi
For my use case (around 10-15 friends, for informal meet-ups) it has worked spectacularly well.
Why is that? Is each of the participants audio/video going over the server so that you need p^2 bandwidth? Are there cool P2P tricks they are/could be using?
I've been wondering about videoconferencing scale, since Zoom seems to have handled a huge explosion in usage very well. Are they just very good at autoscaling AWS instances, or do they use cool tricks to reduce bandwidth?
What is you setup look like ? If you have largely 1:1 meetings you can enable p2p mode only , video will directly be shared between users. I have had no problems delivering 100-150 concurrent sessions at ~1 Gbps per bridge, beyond that we usually cluster the bridge ( Octo is out of the box solution jitsi uses for this)
Video stream, especially if it a group setting is always expensive. I can’t understand why people have such high expectations for video conferencing. This is the reason why Zoom and Microsoft Teams is killing it: video conferencing is inherently difficult and not a trivial task.
When I tried Jitsi with my family friends, some of them struggled to use it since they don't know English.
So I think changing the language can be easier for nonspeakers of English.
For example, language select box can be placed next to "Go" button (when creating meeting) and next to info button in the meeting.
Also using "Accept-Language" HTTP header for choosing the language can be a good default.
Another option is to add a language query parameter to the URL so that host can easily share meeting with a particular language.
Also if it would be great if some improvements can be made for low bandwitdh since there are people with slow connections and limited data plans.
(On the other hand, there is already an option to lower video quality).
Other than that, kudos to Jitsi team for their great efforts and this great project.
I've trialed several video conferencing apps in the last few weeks (Skype, Zoom, Hangouts and others), Jitsi Meet is the only one I've been able to have a video call on with my low-bandwidth, high-ping network. They've done a superb job for my use-case, I'm very happy with the result.
I see nothing in the specification that specifically mentions encryption. Are there any concerns about performance of in-JS streaming encryption? Does the encoder/decoder take promises and are they expected to use the subtle crypto APIs in the browser which return promises?
I am so happy Jitsi exists. My friends and I have a room that we regularly pop into to say hi or play games together.
The mobile app I downloaded through F-Droid works incredibly well, and for those of you Firefox users who aren't having the best experience, I recommend using the Electron desktop app [https://github.com/jitsi/jitsi-meet-electron/releases].
I've been using the Jitsi Electron app in conjunction with OBS + the VirtualCam plugin to share games, videos and my desktop. Hopefully I can convert more Zoom users.
Random fact: the 'Big Buck Bunny'[1] short film that the presenter shares during the screen-and-audio-sharing demo was made by the Blender Foundation[2], an organization that develops open and free content creation tools.
I want to like jitsi, but have yet to use it without participants in the meeting having issues. It doesn't remember hardware selection, so if the default hardware is incorrect, you have to change it every time. Often people arent able to see video, or hear audio. If people aren't using headsets, then the audio from their speaker loops and causes feedback. Its always been painful for everyone. But everyone else seems to love it, so what am I doing wrong? (Running in gcp, fwiw)
I just did a quick test using latest Jitsi meet running in Chrome and it does remember my external webcam settings (saved to browser localStorage). Looping audio is something that happens on alternative platforms too, so I'm not sure that's a fair criticism of Jitsi.
I hope this crisis causes Jitsi to get some serious funding. I've been trying Jitsi on and off for about 6 years and it's never been this good. It also could be a lot better.
Awesome new feature! That's exactly what I'm going to use it for, though I am just a tad sad that it happened before my setup game night, as I felt like a wizard using pulseaudio to route the game audio through as a microphone...
Seems a decent bunch of improvements; sharing system audio is apparently a useful feature (I don't care, but I see it mentioned enough), and device and muting changes are nice usability improvements.
The best thing about Jitsi IMHO is that it scales so well. After having problems with Nextcloud Talk, I just wrote a chat bot that returns Jitsi links (see my GitHub if you’re interested) et voilà 100+ people in one conference!
They also ask for translations at the bottom of the release notes. It's a bit unusual though that most of the time the translation project on weblate is locked, but i haven't seen any vandalism
Nice to hear about the simplified device chooser, I used Jitsi a few days ago for the first time for a family call and I lost like 2 minutes to realize where to change the mic input.
[+] [-] tasty_freeze|6 years ago|reply
My use case is I take weekly music lessons, and now they are virtual. The problem is the DSP done on audio was designed for speech. If my teacher is explaining something then plays an example on his bass, it usually sounds terrible, maybe even inaudible.
I send him pre-recorded mp3s of cover songs; ideally he could listen to it and I could comment in real time about places where things could be improved. Instead, if he is playing any music on his system, I hear nothing -- no music, no talk. It seems like the software thinks "Hey, this participant is listening to non-conference audio, so I'll just mute him (at least on skype). I'd love there to be a half duplex audio button so none of the DSP shenanigans are needed, and a high quality audio stream would be sent.
[+] [-] saghul|6 years ago|reply
https://meet.jit.si/YourRoonNameHere#config.disableAP=true
[+] [-] padenot|6 years ago|reply
> navigator.mediaDevices.getUserMedia({audio: true}).then(...)
to get a stream of the input device, one can do:
> navigator.mediaDevices.getUserMedia({audio: { autoGainControl: false, noiseSuppression: false, echoCancellation: false }}).then(...)
similarly, an _existing_ input audio stream can have its settings changed while it's running like so:
> stream.applyConstraints({ autoGainControl: true, noiseSuppression: true, echoCancellation: true })
this for examples re-enables the processing that we put on voice by default.
This probably works everywhere, we've written a blog post about this that explain a few more things: https://blog.mozilla.org/webrtc/fiddle-of-the-week-audio-con....
If the website doesn't want to offer a control to switch this on/off, I'm confident this can be done by a browser extension in no time (which would have the benefit to work for all websites).
[+] [-] montroser|6 years ago|reply
When you're speaking, you don't have to wonder if others can hear you because your microphone pulses in green visually as you speak. If your audio isn't working it shows in yellow with no pulsing, and you and everyone else can see your audio is not flowing.
Also, if someone else forgot to mute and their kid is making a ruckus, you can just mute them. You don't have to wait for a moment to interject and ask them verbally, you can just go ahead and do it.
Or, when you see someone else in their video feed trying to speak up, but they forgot to unmute, you just unmute them. No everyone saying, "you're muted" over each other.
It takes a second to get used to the idea that everyone has all the power, but in practice it just makes everything go way smoother.
[+] [-] cvwright|6 years ago|reply
It's written by a music professor and geared toward using Zoom for music, but several of us Zoom newbies found it to be helpful more generally. He mentions the issue of disabling the speech-centric audio postprocessing.
http://musictechexplained.com/
Disclaimer: The author apparently makes his money selling eBooks, so you may want to skip through the several pages of promotional material at the beginning of his PDF to get to the good stuff.
[+] [-] miki123211|6 years ago|reply
It's Teamspeak and Discord like, so you need to connect to a server, either public or self-hosted, join or create a channel, and then you will be able to talk to everyone on that channel. This is perfect for permanent communities where people just hang out, but works for one-offs too. It works on Windows, Linux, Mac, iOS and Android, no web access. The server is also available for Raspberry Pi. Half of it is open-source, but the core SDK needs a license if you're developing with it. The program itself is free, even for commercial use.
It uses Opus and lets you adjust the quality and processing, so you can get a lot out of it. We've been using it in our community for about 10 years now, including for radio broadcasting, and we haven't managed to find anything better since. I know of one local radio station and recording studio who successfully use it for remote work now.
To get the best experience, disable all audio processing in the preferences, on the sound system pane, so duplex mode, automatic gain control and noise reduction should be off. If you're on Windows, use Windows Audio Session as the backend for lowest latency.
Then, connect to a server, I use the German one for public stuff, as I'm close to it geographically and you don't need to register for it, but use whichever you want. After connecting, create a channel with application set to music, bitrate set to 150000 and channels set to stereo. Those are, at least, the parameters I use, and they work great. You can adjust the rest as you see fit.
There are some video and screensharing capabilities as far as I know, but I haven't used them. Audio is definitely the primary focus. If you need any assistance, my username here at gmail dot com is the way to go.
[1] bearware.dk for desktop, App Store and Google Play for iOS and Android.
ps. I'm not affiliated with the company in any way, it's just a tool I use daily and would recommend to anyone who knows his way around the computer. It definitely doesn't pass the grandma test, though.
[+] [-] jpdus|6 years ago|reply
[+] [-] kabes|6 years ago|reply
[+] [-] sitkack|6 years ago|reply
[+] [-] oever|6 years ago|reply
http://llcon.sourceforge.net/
[+] [-] _spduchamp|6 years ago|reply
[+] [-] CrankyBear|6 years ago|reply
[+] [-] _-___________-_|6 years ago|reply
[+] [-] spditner|6 years ago|reply
Firefox simulcast is still experimental (edit: and disabled by default), so Firefox is only sending HD to the video-bridge and no LD stream. The HD is then relayed to everyone even as a thumbnail.
If you don't need to see everyone's video all the time, set channelLastN: 5 or a similar number, and only the last N speakers video will be broadcast
If you don't need 720p (1280x720), change the constraints: video section to be something like 360p (640x360)
Enable layer suspension so that HD is not sent to the server when not needed: enableLayerSuspension: true
[+] [-] fleetside72|6 years ago|reply
[+] [-] jka|6 years ago|reply
They're quite responsive on GitHub ( https://github.com/jitsi/jitsi-meet ) and I'm sure additional details regarding any bottlenecks would be appreciated.
Identifying even modest improvements now could result in large memory and bandwidth resource savings over the next few weeks and months.
[+] [-] rcarmo|6 years ago|reply
For my use case (around 10-15 friends, for informal meet-ups) it has worked spectacularly well.
[+] [-] xiii1408|6 years ago|reply
I've been wondering about videoconferencing scale, since Zoom seems to have handled a huge explosion in usage very well. Are they just very good at autoscaling AWS instances, or do they use cool tricks to reduce bandwidth?
[+] [-] manquer|6 years ago|reply
[+] [-] BrowserMeeting|6 years ago|reply
[+] [-] reader_1000|6 years ago|reply
For example, language select box can be placed next to "Go" button (when creating meeting) and next to info button in the meeting. Also using "Accept-Language" HTTP header for choosing the language can be a good default. Another option is to add a language query parameter to the URL so that host can easily share meeting with a particular language.
Also if it would be great if some improvements can be made for low bandwitdh since there are people with slow connections and limited data plans. (On the other hand, there is already an option to lower video quality).
Other than that, kudos to Jitsi team for their great efforts and this great project.
[+] [-] bArray|6 years ago|reply
[+] [-] sandGorgon|6 years ago|reply
This is the time when its truly needed. Firefox, Chrome and Edge should just sit in a room and not come out till its done.
[+] [-] kodablah|6 years ago|reply
[+] [-] bilal4hmed|6 years ago|reply
[+] [-] telesilla|6 years ago|reply
Edge is now Chromium, you need to invite Safari.
[+] [-] dstryr|6 years ago|reply
The mobile app I downloaded through F-Droid works incredibly well, and for those of you Firefox users who aren't having the best experience, I recommend using the Electron desktop app [https://github.com/jitsi/jitsi-meet-electron/releases].
I've been using the Jitsi Electron app in conjunction with OBS + the VirtualCam plugin to share games, videos and my desktop. Hopefully I can convert more Zoom users.
[+] [-] jka|6 years ago|reply
[1] - https://en.wikipedia.org/wiki/Big_Buck_Bunny
[2] - https://blender.org/foundation/
[+] [-] saghul|6 years ago|reply
[+] [-] spoid|6 years ago|reply
https://github.com/jitsi/jitsi-meet#security
https://github.com/jitsi/jitsi-meet/issues/409
Has this changed? In the end you could at least self-host though
[+] [-] BrowserMeeting|6 years ago|reply
[+] [-] nhumrich|6 years ago|reply
[+] [-] porjo|6 years ago|reply
[+] [-] AndyMcConachie|6 years ago|reply
[+] [-] rodolphoarruda|6 years ago|reply
That pop-up alert that tells you that you are speaking while on mute is incredibly smart.
[+] [-] allset_|6 years ago|reply
[+] [-] thomcrowe|6 years ago|reply
[+] [-] MayeulC|6 years ago|reply
However, I like the product they are building. Keep it up! I hope the firefox issues will soon be a thing of the past (https://bugzilla.mozilla.org/buglist.cgi?status_whiteboard_t...)
[+] [-] electriclove|6 years ago|reply
[+] [-] fidel100|6 years ago|reply
[+] [-] unknown|6 years ago|reply
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[+] [-] miloignis|6 years ago|reply
[+] [-] sheidavanesa|6 years ago|reply
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[+] [-] yjftsjthsd-h|6 years ago|reply
[+] [-] jxcl|6 years ago|reply
[+] [-] pojntfx|6 years ago|reply
[+] [-] intsunny|6 years ago|reply
[+] [-] smichel17|6 years ago|reply
...but not well. I can see why they disable it by default, even if I wish they included a button to proceed anyway.
[+] [-] ce4|6 years ago|reply
[+] [-] darkwater|6 years ago|reply