845 lines
35 KiB
Bash
845 lines
35 KiB
Bash
#!/bin/sh -e
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# shellcheck disable=SC2034 # This is intended to be included
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JITSI_CONFIG_JS="$(cat <<EOF
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/* eslint-disable no-unused-vars, no-var */
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var config = {
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// Connection
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//
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hosts: {
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// XMPP domain.
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domain: '${JITSI_HOST}',
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// When using authentication, domain for guest users.
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$( if [ -n "${SECURED_DOMAINS}" ]; then cat<<EOF2
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anonymousdomain: 'guest.${JITSI_HOST}',
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EOF2
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else cat <<EOF2
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// anonymousdomain: 'guest.example.com',
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EOF2
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fi
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)
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// Domain for authenticated users. Defaults to <domain>.
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// authdomain: '${JITSI_HOST}',
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// Focus component domain. Defaults to focus.<domain>.
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// focus: 'focus.${JITSI_HOST}',
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// XMPP MUC domain. FIXME: use XEP-0030 to discover it.
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muc: 'conference.${JITSI_HOST}'
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},
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// BOSH URL. FIXME: use XEP-0156 to discover it.
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bosh: '//${JITSI_HOST}/http-bind',
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// Websocket URL
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// websocket: 'wss://${JITSI_HOST}/xmpp-websocket',
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// The name of client node advertised in XEP-0115 'c' stanza
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clientNode: 'http://jitsi.org/jitsimeet',
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// The real JID of focus participant - can be overridden here
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// Do not change username - FIXME: Make focus username configurable
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// https://github.com/jitsi/jitsi-meet/issues/7376
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// focusUserJid: 'focus@auth.${JITSI_HOST}',
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// Testing / experimental features.
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//
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testing: {
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// Disables the End to End Encryption feature. Useful for debugging
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// issues related to insertable streams.
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// disableE2EE: false,
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// P2P test mode disables automatic switching to P2P when there are 2
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// participants in the conference.
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p2pTestMode: false
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// Enables the test specific features consumed by jitsi-meet-torture
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// testMode: false
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// Disables the auto-play behavior of *all* newly created video element.
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// This is useful when the client runs on a host with limited resources.
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// noAutoPlayVideo: false
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// Enable / disable 500 Kbps bitrate cap on desktop tracks. When enabled,
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// simulcast is turned off for the desktop share. If presenter is turned
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// on while screensharing is in progress, the max bitrate is automatically
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// adjusted to 2.5 Mbps. This takes a value between 0 and 1 which determines
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// the probability for this to be enabled.
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// capScreenshareBitrate: 1 // 0 to disable
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// Enable callstats only for a percentage of users.
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// This takes a value between 0 and 100 which determines the probability for
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// the callstats to be enabled.
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// callStatsThreshold: 5 // enable callstats for 5% of the users.
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},
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// Disables ICE/UDP by filtering out local and remote UDP candidates in
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// signalling.
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// webrtcIceUdpDisable: false,
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// Disables ICE/TCP by filtering out local and remote TCP candidates in
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// signalling.
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// webrtcIceTcpDisable: false,
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// Media
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//
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// Audio
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// Disable measuring of audio levels.
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disableAudioLevels: $(if [ -n "${DISABLE_AUDIO_LEVELS}" ]; then printf "true"; else printf "false"; fi),
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// audioLevelsInterval: 200,
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// Enabling this will run the lib-jitsi-meet no audio detection module which
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// will notify the user if the current selected microphone has no audio
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// input and will suggest another valid device if one is present.
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enableNoAudioDetection: true,
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// Enabling this will show a "Save Logs" link in the GSM popover that can be
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// used to collect debug information (XMPP IQs, SDP offer/answer cycles)
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// about the call.
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// enableSaveLogs: false,
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// Enabling this will run the lib-jitsi-meet noise detection module which will
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// notify the user if there is noise, other than voice, coming from the current
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// selected microphone. The purpose it to let the user know that the input could
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// be potentially unpleasant for other meeting participants.
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enableNoisyMicDetection: true,
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// Start the conference in audio only mode (no video is being received nor
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// sent).
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// startAudioOnly: false,
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// Every participant after the Nth will start audio muted.
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// startAudioMuted: 10,
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// Start calls with audio muted. Unlike the option above, this one is only
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// applied locally. FIXME: having these 2 options is confusing.
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// startWithAudioMuted: false,
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// Enabling it (with #params) will disable local audio output of remote
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// participants and to enable it back a reload is needed.
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// startSilent: false
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// Sets the preferred target bitrate for the Opus audio codec by setting its
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// 'maxaveragebitrate' parameter. Currently not available in p2p mode.
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// Valid values are in the range 6000 to 510000
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// opusMaxAverageBitrate: 20000,
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// Enables support for opus-red (redundancy for Opus).
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// enableOpusRed: false,
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// Video
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// Sets the preferred resolution (height) for local video. Defaults to 720.
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// resolution: 720,
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// How many participants while in the tile view mode, before the receiving video quality is reduced from HD to SD.
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// Use -1 to disable.
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// maxFullResolutionParticipants: 2,
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// w3c spec-compliant video constraints to use for video capture. Currently
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// used by browsers that return true from lib-jitsi-meet's
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// util#browser#usesNewGumFlow. The constraints are independent from
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// this config's resolution value. Defaults to requesting an ideal
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// resolution of 720p.
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// constraints: {
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// video: {
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// height: {
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// ideal: 720,
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// max: 720,
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// min: 240
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// }
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// }
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// },
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$(if [ -n "${VIDEO_CONSTRAINTS}" ]; then echo "${VIDEO_CONSTRAINTS},"; fi)
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// Enable / disable simulcast support.
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// disableSimulcast: false,
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// Enable / disable layer suspension. If enabled, endpoints whose HD
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// layers are not in use will be suspended (no longer sent) until they
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// are requested again.
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// enableLayerSuspension: false,
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// Every participant after the Nth will start video muted.
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startVideoMuted: ${START_VIDEO_MUTED},
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// Start calls with video muted. Unlike the option above, this one is only
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// applied locally. FIXME: having these 2 options is confusing.
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// startWithVideoMuted: false,
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// If set to true, prefer to use the H.264 video codec (if supported).
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// Note that it's not recommended to do this because simulcast is not
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// supported when using H.264. For 1-to-1 calls this setting is enabled by
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// default and can be toggled in the p2p section.
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// This option has been deprecated, use preferredCodec under videoQuality section instead.
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// preferH264: true,
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// If set to true, disable H.264 video codec by stripping it out of the
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// SDP.
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// disableH264: false,
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// Desktop sharing
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// Optional desktop sharing frame rate options. Default value: min:5, max:5.
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// desktopSharingFrameRate: {
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// min: 5,
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// max: 5
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// },
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// Try to start calls with screen-sharing instead of camera video.
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// startScreenSharing: false,
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// Recording
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// Whether to enable file recording or not.
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// fileRecordingsEnabled: false,
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// Enable the dropbox integration.
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// dropbox: {
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// appKey: '<APP_KEY>' // Specify your app key here.
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// // A URL to redirect the user to, after authenticating
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// // by default uses:
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// // 'https://${JITSI_HOST}/static/oauth.html'
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// redirectURI:
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// 'https://${JITSI_HOST}/subfolder/static/oauth.html'
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// },
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// When integrations like dropbox are enabled only that will be shown,
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// by enabling fileRecordingsServiceEnabled, we show both the integrations
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// and the generic recording service (its configuration and storage type
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// depends on jibri configuration)
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// fileRecordingsServiceEnabled: false,
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// Whether to show the possibility to share file recording with other people
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// (e.g. meeting participants), based on the actual implementation
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// on the backend.
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// fileRecordingsServiceSharingEnabled: false,
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// Whether to enable live streaming or not.
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// liveStreamingEnabled: false,
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// Transcription (in interface_config,
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// subtitles and buttons can be configured)
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// transcribingEnabled: false,
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// Enables automatic turning on captions when recording is started
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// autoCaptionOnRecord: false,
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// Misc
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// Default value for the channel "last N" attribute. -1 for unlimited.
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channelLastN: ${CHANNEL_LAST_N},
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// Provides a way to use different "last N" values based on the number of participants in the conference.
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// The keys in an Object represent number of participants and the values are "last N" to be used when number of
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// participants gets to or above the number.
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//
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// For the given example mapping, "last N" will be set to 20 as long as there are at least 5, but less than
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// 29 participants in the call and it will be lowered to 15 when the 30th participant joins. The 'channelLastN'
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// will be used as default until the first threshold is reached.
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//
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// lastNLimits: {
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// 5: 20,
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// 30: 15,
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// 50: 10,
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// 70: 5,
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// 90: 2
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// },
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// Provides a way to translate the legacy bridge signaling messages, 'LastNChangedEvent',
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// 'SelectedEndpointsChangedEvent' and 'ReceiverVideoConstraint' into the new 'ReceiverVideoConstraints' message
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// that invokes the new bandwidth allocation algorithm in the bridge which is described here
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// - https://github.com/jitsi/jitsi-videobridge/blob/master/doc/allocation.md.
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// useNewBandwidthAllocationStrategy: false,
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// Specify the settings for video quality optimizations on the client.
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// videoQuality: {
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// // Provides a way to prevent a video codec from being negotiated on the JVB connection. The codec specified
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// // here will be removed from the list of codecs present in the SDP answer generated by the client. If the
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// // same codec is specified for both the disabled and preferred option, the disable settings will prevail.
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// // Note that 'VP8' cannot be disabled since it's a mandatory codec, the setting will be ignored in this case.
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// disabledCodec: 'H264',
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//
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// // Provides a way to set a preferred video codec for the JVB connection. If 'H264' is specified here,
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// // simulcast will be automatically disabled since JVB doesn't support H264 simulcast yet. This will only
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// // rearrange the the preference order of the codecs in the SDP answer generated by the browser only if the
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// // preferred codec specified here is present. Please ensure that the JVB offers the specified codec for this
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// // to take effect.
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// preferredCodec: 'VP8',
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//
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// // Provides a way to configure the maximum bitrates that will be enforced on the simulcast streams for
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// // video tracks. The keys in the object represent the type of the stream (LD, SD or HD) and the values
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// // are the max.bitrates to be set on that particular type of stream. The actual send may vary based on
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// // the available bandwidth calculated by the browser, but it will be capped by the values specified here.
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// // This is currently not implemented on app based clients on mobile.
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// maxBitratesVideo: {
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// VP8 : {
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// low: 200000,
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// standard: 500000,
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// high: 1500000
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// },
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// VP9: {
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// low: 100000,
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// standard: 300000,
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// high: 1200000
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// }
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// },
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//
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// // The options can be used to override default thresholds of video thumbnail heights corresponding to
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// // the video quality levels used in the application. At the time of this writing the allowed levels are:
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// // 'low' - for the low quality level (180p at the time of this writing)
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// // 'standard' - for the medium quality level (360p)
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// // 'high' - for the high quality level (720p)
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// // The keys should be positive numbers which represent the minimal thumbnail height for the quality level.
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// //
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// // With the default config value below the application will use 'low' quality until the thumbnails are
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// // at least 360 pixels tall. If the thumbnail height reaches 720 pixels then the application will switch to
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// // the high quality.
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// minHeightForQualityLvl: {
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// 360: 'standard',
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// 720: 'high'
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// },
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//
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// // Provides a way to resize the desktop track to 720p (if it is greater than 720p) before creating a canvas
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// // for the presenter mode (camera picture-in-picture mode with screenshare).
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// resizeDesktopForPresenter: false
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// },
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// // Options for the recording limit notification.
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// recordingLimit: {
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//
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// // The recording limit in minutes. Note: This number appears in the notification text
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// // but doesn't enforce the actual recording time limit. This should be configured in
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// // jibri!
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// limit: 60,
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//
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// // The name of the app with unlimited recordings.
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// appName: 'Unlimited recordings APP',
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//
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// // The URL of the app with unlimited recordings.
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// appURL: 'https://unlimited.recordings.app.com/'
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// },
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// Disables or enables RTX (RFC 4588) (defaults to false).
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// disableRtx: false,
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// Disables or enables TCC support in this client (default: enabled).
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// enableTcc: true,
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// Disables or enables REMB support in this client (default: enabled).
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// enableRemb: true,
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// Enables ICE restart logic in LJM and displays the page reload overlay on
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// ICE failure. Current disabled by default because it's causing issues with
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// signaling when Octo is enabled. Also when we do an "ICE restart"(which is
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// not a real ICE restart), the client maintains the TCC sequence number
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// counter, but the bridge resets it. The bridge sends media packets with
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// TCC sequence numbers starting from 0.
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// enableIceRestart: false,
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// Enables forced reload of the client when the call is migrated as a result of
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// the bridge going down. Currently enabled by default as call migration through
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// session-terminate is causing siganling issues when Octo is enabled.
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// enableForcedReload: true,
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// Use TURN/UDP servers for the jitsi-videobridge connection (by default
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// we filter out TURN/UDP because it is usually not needed since the
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// bridge itself is reachable via UDP)
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// useTurnUdp: false
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// UI
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//
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// Disables responsive tiles.
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// disableResponsiveTiles: false,
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// Hides lobby button
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// hideLobbyButton: false,
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// Require users to always specify a display name.
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// requireDisplayName: true,
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// Whether to use a welcome page or not. In case it's false a random room
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// will be joined when no room is specified.
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enableWelcomePage: true,
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// Disable app shortcuts that are registered upon joining a conference
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// disableShortcuts: false,
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// Disable initial browser getUserMedia requests.
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// This is useful for scenarios where users might want to start a conference for screensharing only
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// disableInitialGUM: false,
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// Enabling the close page will ignore the welcome page redirection when
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// a call is hangup.
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// enableClosePage: false,
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// Disable hiding of remote thumbnails when in a 1-on-1 conference call.
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// disable1On1Mode: false,
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// Default language for the user interface.
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defaultLanguage: '${DEFAULT_LANGUAGE}',
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// Disables profile and the edit of all fields from the profile settings (display name and email)
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// disableProfile: false,
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// Whether or not some features are checked based on token.
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// enableFeaturesBasedOnToken: false,
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// When enabled the password used for locking a room is restricted to up to the number of digits specified
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// roomPasswordNumberOfDigits: 10,
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// default: roomPasswordNumberOfDigits: false,
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// Message to show the users. Example: 'The service will be down for
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// maintenance at 01:00 AM GMT,
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noticeMessage: '${NOTICE_MESSAGE}',
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// Enables calendar integration, depends on googleApiApplicationClientID
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// and microsoftApiApplicationClientID
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// enableCalendarIntegration: false,
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// When 'true', it shows an intermediate page before joining, where the user can configure their devices.
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// prejoinPageEnabled: false,
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// If etherpad integration is enabled, setting this to true will
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// automatically open the etherpad when a participant joins. This
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// does not affect the mobile app since opening an etherpad
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// obscures the conference controls -- it's better to let users
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// choose to open the pad on their own in that case.
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// openSharedDocumentOnJoin: false,
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// If true, shows the unsafe room name warning label when a room name is
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// deemed unsafe (due to the simplicity in the name) and a password is not
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// set or the lobby is not enabled.
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// enableInsecureRoomNameWarning: false,
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// Whether to automatically copy invitation URL after creating a room.
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// Document should be focused for this option to work
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// enableAutomaticUrlCopy: false,
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// Base URL for a Gravatar-compatible service. Defaults to libravatar.
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// gravatarBaseURL: 'https://seccdn.libravatar.org/avatar/',
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// Moved from interfaceConfig(TOOLBAR_BUTTONS).
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// The name of the toolbar buttons to display in the toolbar, including the
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// "More actions" menu. If present, the button will display. Exceptions are
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// "livestreaming" and "recording" which also require being a moderator and
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// some other values in config.js to be enabled. Also, the "profile" button will
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// not display for users with a JWT.
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// Notes:
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// - it's impossible to choose which buttons go in the "More actions" menu
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// - it's impossible to control the placement of buttons
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// - 'desktop' controls the "Share your screen" button
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// - if \`toolbarButtons\` is undefined, we fallback to enabling all buttons on the UI
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// toolbarButtons: [
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// 'microphone', 'camera', 'closedcaptions', 'desktop', 'embedmeeting', 'fullscreen',
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// 'fodeviceselection', 'hangup', 'profile', 'chat', 'recording',
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// 'livestreaming', 'etherpad', 'sharedvideo', 'shareaudio', 'settings', 'raisehand',
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// 'videoquality', 'filmstrip', 'invite', 'feedback', 'stats', 'shortcuts',
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// 'tileview', 'select-background', 'download', 'help', 'mute-everyone', 'mute-video-everyone', 'security'
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// ],
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// Stats
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//
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// Whether to enable stats collection or not in the TraceablePeerConnection.
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// This can be useful for debugging purposes (post-processing/analysis of
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// the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
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// estimation tests.
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// gatherStats: false,
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// The interval at which PeerConnection.getStats() is called. Defaults to 10000
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// pcStatsInterval: 10000,
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// To enable sending statistics to callstats.io you must provide the
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// Application ID and Secret.
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// callStatsID: '',
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// callStatsSecret: '',
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// Enables sending participants' display names to callstats
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// enableDisplayNameInStats: false,
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// Enables sending participants' emails (if available) to callstats and other analytics
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// enableEmailInStats: false,
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// Controls the percentage of automatic feedback shown to participants when callstats is enabled.
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// The default value is 100%. If set to 0, no automatic feedback will be requested
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// feedbackPercentage: 100,
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// Privacy
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//
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// If third party requests are disabled, no other server will be contacted.
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// This means avatars will be locally generated and callstats integration
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// will not function.
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disableThirdPartyRequests: $(if [ -z "${ENABLE_THIRD_PARTY_REQUESTS}" ]; then printf "true"; else printf "false"; fi),
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// Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
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//
|
|
|
|
p2p: {
|
|
// Enables peer to peer mode. When enabled the system will try to
|
|
// establish a direct connection when there are exactly 2 participants
|
|
// in the room. If that succeeds the conference will stop sending data
|
|
// through the JVB and use the peer to peer connection instead. When a
|
|
// 3rd participant joins the conference will be moved back to the JVB
|
|
// connection.
|
|
enabled: true,
|
|
|
|
// The STUN servers that will be used in the peer to peer connections
|
|
stunServers: [
|
|
|
|
{ urls: 'stun:${TURN_SERVER}:443' }
|
|
]
|
|
|
|
// Sets the ICE transport policy for the p2p connection. At the time
|
|
// of this writing the list of possible values are 'all' and 'relay',
|
|
// but that is subject to change in the future. The enum is defined in
|
|
// the WebRTC standard:
|
|
// https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
|
|
// If not set, the effective value is 'all'.
|
|
// iceTransportPolicy: 'all',
|
|
|
|
// If set to true, it will prefer to use H.264 for P2P calls (if H.264
|
|
// is supported). This setting is deprecated, use preferredCodec instead.
|
|
// preferH264: true,
|
|
|
|
// Provides a way to set the video codec preference on the p2p connection. Acceptable
|
|
// codec values are 'VP8', 'VP9' and 'H264'.
|
|
// preferredCodec: 'H264',
|
|
|
|
// If set to true, disable H.264 video codec by stripping it out of the
|
|
// SDP. This setting is deprecated, use disabledCodec instead.
|
|
// disableH264: false,
|
|
|
|
// Provides a way to prevent a video codec from being negotiated on the p2p connection.
|
|
// disabledCodec: '',
|
|
|
|
// How long we're going to wait, before going back to P2P after the 3rd
|
|
// participant has left the conference (to filter out page reload).
|
|
// backToP2PDelay: 5
|
|
},
|
|
|
|
analytics: {
|
|
// The Google Analytics Tracking ID:
|
|
// googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
|
|
|
|
// Matomo configuration:
|
|
// matomoEndpoint: 'https://your-matomo-endpoint/',
|
|
// matomoSiteID: '42',
|
|
|
|
// The Amplitude APP Key:
|
|
// amplitudeAPPKey: '<APP_KEY>'
|
|
|
|
// Configuration for the rtcstats server:
|
|
// By enabling rtcstats server every time a conference is joined the rtcstats
|
|
// module connects to the provided rtcstatsEndpoint and sends statistics regarding
|
|
// PeerConnection states along with getStats metrics polled at the specified
|
|
// interval.
|
|
// rtcstatsEnabled: true,
|
|
|
|
// In order to enable rtcstats one needs to provide a endpoint url.
|
|
// rtcstatsEndpoint: wss://rtcstats-server-pilot.jitsi.net/,
|
|
|
|
// The interval at which rtcstats will poll getStats, defaults to 1000ms.
|
|
// If the value is set to 0 getStats won't be polled and the rtcstats client
|
|
// will only send data related to RTCPeerConnection events.
|
|
// rtcstatsPolIInterval: 1000,
|
|
|
|
// Array of script URLs to load as lib-jitsi-meet "analytics handlers".
|
|
// scriptURLs: [
|
|
// "libs/analytics-ga.min.js", // google-analytics
|
|
// "https://example.com/my-custom-analytics.js"
|
|
// ],
|
|
},
|
|
|
|
// Logs that should go be passed through the 'log' event if a handler is defined for it
|
|
// apiLogLevels: ['warn', 'log', 'error', 'info', 'debug'],
|
|
|
|
// Information about the jitsi-meet instance we are connecting to, including
|
|
// the user region as seen by the server.
|
|
deploymentInfo: {
|
|
// shard: "shard1",
|
|
// region: "europe",
|
|
// userRegion: "asia"
|
|
},
|
|
|
|
// Decides whether the start/stop recording audio notifications should play on record.
|
|
// disableRecordAudioNotification: false,
|
|
|
|
// Disables the sounds that play when other participants join or leave the
|
|
// conference (if set to true, these sounds will not be played).
|
|
// disableJoinLeaveSounds: false,
|
|
|
|
// Information for the chrome extension banner
|
|
// chromeExtensionBanner: {
|
|
// // The chrome extension to be installed address
|
|
// url: 'https://chrome.google.com/webstore/detail/jitsi-meetings/kglhbbefdnlheedjiejgomgmfplipfeb',
|
|
|
|
// // Extensions info which allows checking if they are installed or not
|
|
// chromeExtensionsInfo: [
|
|
// {
|
|
// id: 'kglhbbefdnlheedjiejgomgmfplipfeb',
|
|
// path: 'jitsi-logo-48x48.png'
|
|
// }
|
|
// ]
|
|
// },
|
|
|
|
// Local Recording
|
|
//
|
|
|
|
// localRecording: {
|
|
// Enables local recording.
|
|
// Additionally, 'localrecording' (all lowercase) needs to be added to
|
|
// TOOLBAR_BUTTONS in interface_config.js for the Local Recording
|
|
// button to show up on the toolbar.
|
|
//
|
|
// enabled: true,
|
|
//
|
|
|
|
// The recording format, can be one of 'ogg', 'flac' or 'wav'.
|
|
// format: 'flac'
|
|
//
|
|
|
|
// },
|
|
|
|
// Options related to end-to-end (participant to participant) ping.
|
|
// e2eping: {
|
|
// // The interval in milliseconds at which pings will be sent.
|
|
// // Defaults to 10000, set to <= 0 to disable.
|
|
// pingInterval: 10000,
|
|
//
|
|
// // The interval in milliseconds at which analytics events
|
|
// // with the measured RTT will be sent. Defaults to 60000, set
|
|
// // to <= 0 to disable.
|
|
// analyticsInterval: 60000,
|
|
// },
|
|
|
|
// If set, will attempt to use the provided video input device label when
|
|
// triggering a screenshare, instead of proceeding through the normal flow
|
|
// for obtaining a desktop stream.
|
|
// NOTE: This option is experimental and is currently intended for internal
|
|
// use only.
|
|
// _desktopSharingSourceDevice: 'sample-id-or-label',
|
|
|
|
// If true, any checks to handoff to another application will be prevented
|
|
// and instead the app will continue to display in the current browser.
|
|
// disableDeepLinking: false,
|
|
|
|
// A property to disable the right click context menu for localVideo
|
|
// the menu has option to flip the locally seen video for local presentations
|
|
// disableLocalVideoFlip: false,
|
|
|
|
// A property used to unset the default flip state of the local video.
|
|
// When it is set to 'true', the local(self) video will not be mirrored anymore.
|
|
// doNotFlipLocalVideo: false,
|
|
|
|
// Mainly privacy related settings
|
|
|
|
// Disables all invite functions from the app (share, invite, dial out...etc)
|
|
// disableInviteFunctions: true,
|
|
|
|
// Disables storing the room name to the recents list
|
|
// doNotStoreRoom: true,
|
|
|
|
// Deployment specific URLs.
|
|
// deploymentUrls: {
|
|
// // If specified a 'Help' button will be displayed in the overflow menu with a link to the specified URL for
|
|
// // user documentation.
|
|
// userDocumentationURL: 'https://docs.example.com/video-meetings.html',
|
|
// // If specified a 'Download our apps' button will be displayed in the overflow menu with a link
|
|
// // to the specified URL for an app download page.
|
|
// downloadAppsUrl: 'https://docs.example.com/our-apps.html'
|
|
// },
|
|
|
|
// Options related to the remote participant menu.
|
|
// remoteVideoMenu: {
|
|
// // If set to true the 'Kick out' button will be disabled.
|
|
// disableKick: true
|
|
// },
|
|
|
|
// If set to true all muting operations of remote participants will be disabled.
|
|
// disableRemoteMute: true,
|
|
|
|
// Enables support for lip-sync for this client (if the browser supports it).
|
|
// enableLipSync: false
|
|
|
|
/**
|
|
External API url used to receive branding specific information.
|
|
If there is no url set or there are missing fields, the defaults are applied.
|
|
None of the fields are mandatory and the response must have the shape:
|
|
{
|
|
// The hex value for the colour used as background
|
|
backgroundColor: '#fff',
|
|
// The url for the image used as background
|
|
backgroundImageUrl: 'https://example.com/background-img.png',
|
|
// The anchor url used when clicking the logo image
|
|
logoClickUrl: 'https://example-company.org',
|
|
// The url used for the image used as logo
|
|
logoImageUrl: 'https://example.com/logo-img.png'
|
|
}
|
|
*/
|
|
dynamicBrandingUrl: "${DYNAMIC_BRANDING_URL}",
|
|
|
|
// Sets the background transparency level. '0' is fully transparent, '1' is opaque.
|
|
// backgroundAlpha: 1,
|
|
|
|
// The URL of the moderated rooms microservice, if available. If it
|
|
// is present, a link to the service will be rendered on the welcome page,
|
|
// otherwise the app doesn't render it.
|
|
// moderatedRoomServiceUrl: 'https://moderated.jitsi-meet.example.com',
|
|
|
|
// If true, tile view will not be enabled automatically when the participants count threshold is reached.
|
|
// disableTileView: true,
|
|
|
|
// Hides the conference subject
|
|
// hideConferenceSubject: true,
|
|
|
|
// Hides the conference timer.
|
|
// hideConferenceTimer: true,
|
|
|
|
// Hides the participants stats
|
|
// hideParticipantsStats: true,
|
|
|
|
// Sets the conference subject
|
|
// subject: 'Conference Subject',
|
|
|
|
// This property is related to the use case when jitsi-meet is used via the IFrame API. When the property is true
|
|
// jitsi-meet will use the local storage of the host page instead of its own. This option is useful if the browser
|
|
// is not persisting the local storage inside the iframe.
|
|
// useHostPageLocalStorage: true,
|
|
|
|
// List of undocumented settings used in jitsi-meet
|
|
/**
|
|
_immediateReloadThreshold
|
|
debug
|
|
debugAudioLevels
|
|
deploymentInfo
|
|
dialInConfCodeUrl
|
|
dialInNumbersUrl
|
|
dialOutAuthUrl
|
|
dialOutCodesUrl
|
|
disableRemoteControl
|
|
displayJids
|
|
etherpad_base
|
|
externalConnectUrl
|
|
firefox_fake_device
|
|
googleApiApplicationClientID
|
|
iAmRecorder
|
|
iAmSipGateway
|
|
microsoftApiApplicationClientID
|
|
peopleSearchQueryTypes
|
|
peopleSearchUrl
|
|
requireDisplayName
|
|
tokenAuthUrl
|
|
*/
|
|
|
|
/**
|
|
* This property can be used to alter the generated meeting invite links (in combination with a branding domain
|
|
* which is retrieved internally by jitsi meet) (e.g. https://meet.jit.si/someMeeting
|
|
* can become https://brandedDomain/roomAlias)
|
|
*/
|
|
// brandingRoomAlias: null,
|
|
|
|
// List of undocumented settings used in lib-jitsi-meet
|
|
/**
|
|
_peerConnStatusOutOfLastNTimeout
|
|
_peerConnStatusRtcMuteTimeout
|
|
abTesting
|
|
avgRtpStatsN
|
|
callStatsConfIDNamespace
|
|
callStatsCustomScriptUrl
|
|
desktopSharingSources
|
|
disableAEC
|
|
disableAGC
|
|
disableAP
|
|
disableHPF
|
|
disableNS
|
|
enableTalkWhileMuted
|
|
forceJVB121Ratio
|
|
forceTurnRelay
|
|
hiddenDomain
|
|
ignoreStartMuted
|
|
websocketKeepAlive
|
|
websocketKeepAliveUrl
|
|
*/
|
|
|
|
/**
|
|
Use this array to configure which notifications will be shown to the user
|
|
The items correspond to the title or description key of that notification
|
|
Some of these notifications also depend on some other internal logic to be displayed or not,
|
|
so adding them here will not ensure they will always be displayed
|
|
|
|
A falsy value for this prop will result in having all notifications enabled (e.g null, undefined, false)
|
|
*/
|
|
// notifications: [
|
|
// 'connection.CONNFAIL', // shown when the connection fails,
|
|
// 'dialog.cameraNotSendingData', // shown when there's no feed from user's camera
|
|
// 'dialog.kickTitle', // shown when user has been kicked
|
|
// 'dialog.liveStreaming', // livestreaming notifications (pending, on, off, limits)
|
|
// 'dialog.lockTitle', // shown when setting conference password fails
|
|
// 'dialog.maxUsersLimitReached', // shown when maximmum users limit has been reached
|
|
// 'dialog.micNotSendingData', // shown when user's mic is not sending any audio
|
|
// 'dialog.passwordNotSupportedTitle', // shown when setting conference password fails due to password format
|
|
// 'dialog.recording', // recording notifications (pending, on, off, limits)
|
|
// 'dialog.remoteControlTitle', // remote control notifications (allowed, denied, start, stop, error)
|
|
// 'dialog.reservationError',
|
|
// 'dialog.serviceUnavailable', // shown when server is not reachable
|
|
// 'dialog.sessTerminated', // shown when there is a failed conference session
|
|
// 'dialog.sessionRestarted', // show when a client reload is initiated because of bridge migration
|
|
// 'dialog.tokenAuthFailed', // show when an invalid jwt is used
|
|
// 'dialog.transcribing', // transcribing notifications (pending, off)
|
|
// 'dialOut.statusMessage', // shown when dial out status is updated.
|
|
// 'liveStreaming.busy', // shown when livestreaming service is busy
|
|
// 'liveStreaming.failedToStart', // shown when livestreaming fails to start
|
|
// 'liveStreaming.unavailableTitle', // shown when livestreaming service is not reachable
|
|
// 'lobby.joinRejectedMessage', // shown when while in a lobby, user's request to join is rejected
|
|
// 'lobby.notificationTitle', // shown when lobby is toggled and when join requests are allowed / denied
|
|
// 'localRecording.localRecording', // shown when a local recording is started
|
|
// 'notify.disconnected', // shown when a participant has left
|
|
// 'notify.grantedTo', // shown when moderator rights were granted to a participant
|
|
// 'notify.invitedOneMember', // shown when 1 participant has been invited
|
|
// 'notify.invitedThreePlusMembers', // shown when 3+ participants have been invited
|
|
// 'notify.invitedTwoMembers', // shown when 2 participants have been invited
|
|
// 'notify.kickParticipant', // shown when a participant is kicked
|
|
// 'notify.mutedRemotelyTitle', // shown when user is muted by a remote party
|
|
// 'notify.mutedTitle', // shown when user has been muted upon joining,
|
|
// 'notify.newDeviceAudioTitle', // prompts the user to use a newly detected audio device
|
|
// 'notify.newDeviceCameraTitle', // prompts the user to use a newly detected camera
|
|
// 'notify.passwordRemovedRemotely', // shown when a password has been removed remotely
|
|
// 'notify.passwordSetRemotely', // shown when a password has been set remotely
|
|
// 'notify.raisedHand', // shown when a partcipant used raise hand,
|
|
// 'notify.startSilentTitle', // shown when user joined with no audio
|
|
// 'prejoin.errorDialOut',
|
|
// 'prejoin.errorDialOutDisconnected',
|
|
// 'prejoin.errorDialOutFailed',
|
|
// 'prejoin.errorDialOutStatus',
|
|
// 'prejoin.errorStatusCode',
|
|
// 'prejoin.errorValidation',
|
|
// 'recording.busy', // shown when recording service is busy
|
|
// 'recording.failedToStart', // shown when recording fails to start
|
|
// 'recording.unavailableTitle', // shown when recording service is not reachable
|
|
// 'toolbar.noAudioSignalTitle', // shown when a broken mic is detected
|
|
// 'toolbar.noisyAudioInputTitle', // shown when noise is detected for the current microphone
|
|
// 'toolbar.talkWhileMutedPopup', // shown when user tries to speak while muted
|
|
// 'transcribing.failedToStart' // shown when transcribing fails to start
|
|
// ]
|
|
|
|
// Allow all above example options to include a trailing comma and
|
|
// prevent fear when commenting out the last value.
|
|
makeJsonParserHappy: 'even if last key had a trailing comma'
|
|
|
|
// no configuration value should follow this line.
|
|
};
|
|
|
|
/* eslint-enable no-unused-vars, no-var */
|
|
EOF
|
|
)"
|