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Spellchecker.
Signed-off-by: Kurt Garloff <[email protected]>
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community/tools/jitsi.md

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@@ -55,17 +55,17 @@ 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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| Room Name | Dial-in 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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@@ -94,7 +94,7 @@ By enabling experimental `media.webrtc.codec.video.av1.experimental_preferred` y
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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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## Limitations and issues
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### Firewalls blocking UDP traffic
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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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instead we use other VC tools such as BB or OpenTalk or the tool of the partner.)
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### Local audio
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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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Jitsi receives one or several audio and video streams from every participant 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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else can hear each 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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### 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 combined 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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### 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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