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Merge branch 'feat/add-jitsi-help' into staging
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‎community/tools/jitsi.md‎

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# Jitsi
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We use a self-hosted [Jitsi Meet](https://jitsi.org) instance for video conferencing.
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Thanks go to Cleura for providing the server for it.
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Thanks go to Cleura for providing the server VM for it.
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Jitsi has served us well, providing good quality and reliable VC service while allowing
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multiple screen shares and conferences with (at least) up to 50 video participants.
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The server uses an automated deployment based on the
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[heat-docker-jitsi-meet](https://github.com/garloff/heat-docker-jitsi-meet) project.
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Configuration is such everyone who knows the room can connect, unless the moderator
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sets a password/PIN. Opening a new room requires authentication. (Contact Kurt if
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you need a password.)
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sets a password/PIN. Opening a new room requires authentication. (Contact
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[Kurt](https://scs.community/garloff) if you need a password.)
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Links to the meeting room (as well as dial-in information) are in the appointments
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in the public calendar.
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## Usage
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Connect with a desktop browser (Chrome/Chromium or other blink based browser
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recommended due to superior WebRTC implementation with SimulCast/SVC for VP8/VP9 --
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Safari & Firefox work, but cause higher data traffic). For mobile devices use
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the Jitsi Meet App.
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Connect with a desktop browser or (for mobile devices) the Jitsi Meet App.
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Use the little arrows in the control bar at the bottom to select speaker, microphone
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and camera in case you lack audio/video. Occasionally, you can not hear all but
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one participant; in this case reconnecting typically helps.
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and camera in case you lack audio/video.
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## Features
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### Whiteboard and Etherpad
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The Jitsi instance has an etherpad and a whiteboard enabled.
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These tools can be used for collaborative creation and collection of content.
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Don't forget to save the contents to a persistent place after the meeting.
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### Codecs
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It is configured to prefer video codecs [AV1](https://en.wikipedia.org/wiki/AV1)
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over [VP9](https://en.wikipedia.org/wiki/VP9)
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over [VP8](https://en.wikipedia.org/wiki/VP8)
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over [H.264](https://en.wikipedia.org/wiki/H.264/MPEG-4_AVC).
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It prefers the [opus](https://opus-codec.org/) audio codec.
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These settings are chosen to provide good video and audio quality for clients
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with modern hardware at moderate bandwidth requirements.
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Clients can chose to use older codecs without impacting audio or video streams
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of others.
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## Dial-In
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Dial-In may be more stable for participants that have a stable phone connection, but
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not a reliable internet connection.
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We thus have an audio bridge using jigasi and [asterisk](https://www.asterisk.org/)
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connected to a [SIP](https://en.wikipedia.org/wiki/Session_Initiation_Protocol) provider.
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This allows a distinct set of rooms to be provided with phone dial-in.
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Here's the setup:
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| Room Name | Dialin Suffix |
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| --------------- | ------------- |
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| SCS-Tech | 611 |
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| SCS-Governance | 612 |
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| Open-Operations | 613 |
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| SCS-OSISM | 614 |
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| SCS-Project | 615 |
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| SCS-Forum | 616 |
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| SCS-Kurt | 617 |
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| SCS-Taskforce | 618 |
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| SCS-ProjectTeam | 619 |
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Dial +49-221-292772-Suffix to connect.
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Rooms protected with a PIN would use 60x instead of 61x as suffix.
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Rooms with a three or four-digit number as room name would be connected to -61XXX or -61XXXX.
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Note that dial-in is not super-reliable due to occasional trouble with the SIP provider.
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So double-check ahead of important conference calls that require phone dial-in. Talk to Kurt
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to change room assignment or to resolve issues with dial-in.
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## Browser specific hints
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Traditionally, the [blink](<https://en.wikipedia.org/wiki/Blink_(browser_engine)-based>>
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browsers (like Google Chrome, Chromium, Edge, ...) supported WebRTC best.
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Safari and Firefox do work, but at the cost of inferior codecs or increased CPU or
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bandwidth requirements (e.g. due to missing [SimulCast](https://en.wikipedia.org/wiki/Simulcast)
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support or missing hardware acceleration).
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### Firefox and VP9 / AV1
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On [Firefox](https://www.mozilla.org), in `about:config`,
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you can enable `media.peerconnection.video.vp9_preferred` and
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`media.webrtc.simulcast.vp9.enabled` for using VP9 video codec (which is better than VP8).
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By enabling experimental `media.webrtc.codec.video.av1.experimental_preferred` you even get AV1
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(which is even better) in Firefox 139+. Depending on whether your hardware has hardware support for VP9
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or AV1 encoding support and on whether that is exposed by your graphics driver stack, this may or may
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not create high CPU usage which you may not consider welcome as mobile user.
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## Limitations
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### Firewalls blocking UDP traffic
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While the web interface uses https (port 8443) which most firewalls find acceptable, the audio and
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video is transmitted via UDP (port 10000+). Some corporate and many public sector firewalls believe
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that outgoing(!) UDP traffic is dangerous and needs to be intercepted. This means that our Jitsi
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setup will not work for participants behind such firewalls.
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(We do not currently have a [COTURN](https://github.com/coturn/coturn) server to work around this;
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instead we use other VC tools such as BB or OpenTalk or the tool of the partner.))
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### Large conferences
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For large conferences, it is recommended that participants stay muted and raise their hand
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in order to talk, so a moderator can ensure a somewhat structured discussion. While Jitsi can route
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a few dozens of video streams without trouble, the combines bandwidth may become a challenge for
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some of the participants and it is recommended to only switch on videos for the active participants.
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We have not tested much above 50 participants in the SCS community, so we don't know the precise limits
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of the server connection or capacity we use.
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## Known Issues
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### Local audio
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A lack of audio is often in the local audio setup (mixer volumes turned to zero etc.).
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On Linux systems, the `pavucontrol` mixer may be the best starting point to resolve issues.
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### Selective Stream forwarding failure
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Jitsi receives one or several audio and video streams from every participiant and selectively
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forwards those to all recipients that have subscribed to these streams. (Typically, a low-res video
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stream is sent in addition to a medium-res and a high-res one — if any high-res subscribers exist).
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This approach to video-conferencing is called
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[selective forwarding unit (SFU)](https://bloggeek.me/webrtcglossary/sfu/).
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Occasionally, one of the participants can not hear one other (out of many) participants but everyone
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else can hear echa other - a subscription to an audio (or video) stream may have gotten lost.
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In this case, a reconnect by the one not hearing is the best remedy.
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### Screen sharing frame rate
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We have an asterisk connected to some conference rooms to provide dial-in capabilities
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for folks that lack internet connectivity (but have a working phone connection).
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Some browsers seem to ignore the FPS setting and try to transmit a shared window (or a shared
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desktop) at high resolution (e.g. 2560x1600) with 30fps. This requires more bandwidth than ADSL
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links typically handle. This can result in low-resolution streams rather than the (wanted) low-fps
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high-resolution stream.

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