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Add Jitsi help page.
Signed-off-by: Kurt Garloff <[email protected]>
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‎community/collaboration/index.mdx‎

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Our meetings are publicly announced and we are happy to welcome both newcomers and established members alike. You can navigate either through the calendar below or subscribe with your favorite client to https://sovereigncloudstack.github.io/calendar/scs.ics. The calendar is collaboratively maintained on GitHub and new entries, such as a lightning talk, are highly appreciated!
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Most of our meetings are taking place on our [own video conferencing server (Jitsi)](jitsi).
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<CommunityCalendar />
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## Collaborating with issues and pull requests

‎community/collaboration/jitsi.md‎

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# Help on Jitsi server
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## SCS Links
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* [Old SCS Website](https://scs.community/)
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* [New Forum SCS-Standards Website](https://sovereigncloudstack.org/)
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* [SCS Documentation](https://docs.scs.community/)
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* [SCS on github](https://github.com/SovereignCloudStack/)
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## SCS Jitsi Help
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### What?
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SCS runs its own Video Conference Server.
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It is a [Jitsi](https://jitsi.org/) instance run in a docker container in a VM sponsored by
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[Cleura](https://cleura.com) in its FRA datacenter.
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It is operated by the SCS community (SCS Project Board) and provides a solution
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without leaking data or metadata to outside parties.
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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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### Where?
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The server provides its VC service on
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[https://conf.scs.koeln:8443/](https://conf.scs.koeln:8443/).
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The server uses [Let's Encrypt](letsencrypt.org) TLS certificates.
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### Usage
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You can access the service with your Web Browser (supporting [WebRTC](https://webrtc.org/))
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or the Jitsi Meet App (Android, iOS).
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#### Login
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The meetings are free to be accessed by anyone that knows the room name, unless the
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moderator enables encryption and a password.
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To become a moderator, you need to have a login. Talk to [Kurt](https://scs.community/garloff).
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A moderator can open new rooms and change the room's settings.
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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.</p>
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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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We 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 distrinct 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 the dial-in.
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### Browser 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, ...) support WebRTC best.
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Safari and Firefox do work.</p>
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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
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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 most 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 them. (We do not currently have a [COTURN](https://github.com/coturn/coturn)
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server to work around.)
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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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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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