Move Jitsi hints to existing Jitsi documentation.

Signed-off-by: Kurt Garloff <[email protected]>

Author: garloff

community/collaboration/index.mdx

@@ -6,8 +6,6 @@ import CommunityCalendar from '../../src/components/CommunityCalendar'
 
 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!
 
-Most of our meetings are taking place on our [own video conferencing server (Jitsi)](jitsi).
-
 <CommunityCalendar />
 
 ## Collaborating with issues and pull requests

community/collaboration/jitsi.md

@@ -1,28 +1,140 @@
 # Jitsi
 
 We use a self-hosted [Jitsi Meet](https://jitsi.org) instance for video conferencing.
-Thanks go to Cleura for providing the server for it.
+Thanks go to Cleura for providing the server VM for it.
+
+Jitsi has served us well, providing good quality and reliable VC service while allowing
+multiple screen shares and conferences with (at least) up to 50 video participants.
 
 The server uses an automated deployment based on the
 [heat-docker-jitsi-meet](https://github.com/garloff/heat-docker-jitsi-meet) project.
 
 Configuration is such everyone who knows the room can connect, unless the moderator
-sets a password/PIN. Opening a new room requires authentication. (Contact Kurt if
-you need a password.)
+sets a password/PIN. Opening a new room requires authentication. (Contact
+[Kurt](https://scs.community/garloff) if you need a password.)
 
 Links to the meeting room (as well as dial-in information) are in the appointments
 in the public calendar.
 
 ## Usage
 
-Connect with a desktop browser (Chrome/Chromium or other blink based browser
-recommended due to superior WebRTC implementation with SimulCast/SVC for VP8/VP9 --
-Safari & Firefox work, but cause higher data traffic). For mobile devices use
-the Jitsi Meet App.
+Connect with a desktop browser or (for mobile devices) the Jitsi Meet App.
 
 Use the little arrows in the control bar at the bottom to select speaker, microphone
-and camera in case you lack audio/video. Occasionally, you can not hear all but
-one participant; in this case reconnecting typically helps.
+and camera in case you lack audio/video.
+
+## Features
+
+### Whiteboard and Etherpad
+
+The Jitsi instance has an etherpad and a whiteboard enabled.
+These tools can be used for collaborative creation and collection of content.
+Don't forget to save the contents to a persistent place after the meeting.
+
+### Codecs
+
+It is configured to prefer video codecs [AV1](https://en.wikipedia.org/wiki/AV1)
+over [VP9](https://en.wikipedia.org/wiki/VP9)
+over [VP8](https://en.wikipedia.org/wiki/VP8)
+over [H.264](https://en.wikipedia.org/wiki/H.264/MPEG-4_AVC).
+It prefers the [opus](https://opus-codec.org/) audio codec.
+
+These settings are chosen to provide good video and audio quality for clients
+with modern hardware at moderate bandwidth requirements.
+Clients can chose to use older codecs without impacting audio or video streams
+of others.
+
+## Dial-In
+
+Dial-In may be more stable for participants that have a stable phone connection, but
+not a reliable internet connection.
+
+We thus have an audio bridge using jigasi and [asterisk](https://www.asterisk.org/)
+connected to a [SIP](https://en.wikipedia.org/wiki/Session_Initiation_Protocol) provider.
+This allows a distinct set of rooms to be provided with phone dial-in.
+
+Here's the setup:
+
+| Room Name       | Dialin Suffix |
+| --------------- | ------------- |
+| SCS-Tech        | 611           |
+| SCS-Governance  | 612           |
+| Open-Operations | 613           |
+| SCS-OSISM       | 614           |
+| SCS-Project     | 615           |
+| SCS-Forum       | 616           |
+| SCS-Kurt        | 617           |
+| SCS-Taskforce   | 618           |
+| SCS-ProjectTeam | 619           |
+
+Dial +49-221-292772-Suffix to connect.
+
+Rooms protected with a PIN would use 60x instead of 61x as suffix.
+Rooms with a three or four-digit number as room name would be connected to -61XXX or -61XXXX.
+Note that dial-in is not super-reliable due to occasional trouble with the SIP provider.
+So double-check ahead of important conference calls that require phone dial-in. Talk to Kurt
+to change room assignment or to resolve issues with dial-in.
+
+## Browser specific hints
+
+Traditionally, the [blink](<https://en.wikipedia.org/wiki/Blink_(browser_engine)-based>>
+browsers (like Google Chrome, Chromium, Edge, ...) supported WebRTC best.
+Safari and Firefox do work, but at the cost of inferior codecs or increased CPU or
+bandwidth requirements (e.g. due to missing [SimulCast](https://en.wikipedia.org/wiki/Simulcast)
+support or missing hardware acceleration).
+
+### Firefox and VP9 / AV1
+
+On [Firefox](https://www.mozilla.org), in `about:config`,
+you can enable `media.peerconnection.video.vp9_preferred` and
+`media.webrtc.simulcast.vp9.enabled` for using VP9 video codec (which is better than VP8).
+
+By enabling experimental `media.webrtc.codec.video.av1.experimental_preferred` you even get AV1
+(which is even better) in Firefox 139+. Depending on whether your hardware has hardware support for VP9
+or AV1 encoding support and on whether that is exposed by your graphics driver stack, this may or may
+not create high CPU usage which you may not consider welcome as mobile user.
+
+## Limitations
+
+### Firewalls blocking UDP traffic
+
+While the web interface uses https (port 8443) which most firewalls find acceptable, the audio and
+video is transmitted via UDP (port 10000+). Some corporate and many public sector firewalls believe
+that outgoing(!) UDP traffic is dangerous and needs to be intercepted. This means that our Jitsi
+setup will not work for participants behind such firewalls.
+(We do not currently have a [COTURN](https://github.com/coturn/coturn) server to work around this;
+instead we use other VC tools such as BB or OpenTalk or the tool of the partner.))
+
+### Large conferences
+
+For large conferences, it is recommended that participants stay muted and raise their hand
+in order to talk, so a moderator can ensure a somewhat structured discussion. While Jitsi can route
+a few dozens of video streams without trouble, the combines bandwidth may become a challenge for
+some of the participants and it is recommended to only switch on videos for the active participants.
+We have not tested much above 50 participants in the SCS community, so we don't know the precise limits
+of the server connection or capacity we use.
+
+## Known Issues
+
+### Local audio
+
+A lack of audio is often in the local audio setup (mixer volumes turned to zero etc.).
+On Linux systems, the `pavucontrol` mixer may be the best starting point to resolve issues.
+
+### Selective Stream forwarding failure
+
+Jitsi receives one or several audio and video streams from every participiant and selectively
+forwards those to all recipients that have subscribed to these streams. (Typically, a low-res video
+stream is sent in addition to a medium-res and a high-res one — if any high-res subscribers exist).
+This approach to video-conferencing is called
+[selective forwarding unit (SFU)](https://bloggeek.me/webrtcglossary/sfu/).
+Occasionally, one of the participants can not hear one other (out of many) participants but everyone
+else can hear echa other - a subscription to an audio (or video) stream may have gotten lost.
+In this case, a reconnect by the one not hearing is the best remedy.
+
+### Screen sharing frame rate
 
-We have an asterisk connected to some conference rooms to provide dial-in capabilities
-for folks that lack internet connectivity (but have a working phone connection).
+Some browsers seem to ignore the FPS setting and try to transmit a shared window (or a shared
+desktop) at high resolution (e.g. 2560x1600) with 30fps. This requires more bandwidth than ADSL
+links typically handle. This can result in low-resolution streams rather than the (wanted) low-fps
+high-resolution stream.