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config.js
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/* eslint-disable comma-dangle, no-unused-vars, no-var, prefer-template, vars-on-top */
/*
* NOTE: If you add a new option please remember to document it here:
* https://jitsi.github.io/handbook/docs/dev-guide/dev-guide-configuration
*/
var subdir = '<!--# echo var="subdir" default="" -->';
var subdomain = '<!--# echo var="subdomain" default="" -->';
if (subdomain) {
subdomain = subdomain.substr(0, subdomain.length - 1).split('.')
.join('_')
.toLowerCase() + '.';
}
// In case of no ssi provided by the webserver, use empty strings
if (subdir.startsWith('<!--')) {
subdir = '';
}
if (subdomain.startsWith('<!--')) {
subdomain = '';
}
var enableJaaS = false;
var config = {
// Connection
//
hosts: {
// XMPP domain.
domain: 'jitsi-meet.example.com',
// When using authentication, domain for guest users.
// anonymousdomain: 'guest.example.com',
// Domain for authenticated users. Defaults to <domain>.
// authdomain: 'jitsi-meet.example.com',
// Focus component domain. Defaults to focus.<domain>.
// focus: 'focus.jitsi-meet.example.com',
// XMPP MUC domain. FIXME: use XEP-0030 to discover it.
muc: 'conference.' + subdomain + 'jitsi-meet.example.com',
},
// BOSH URL. FIXME: use XEP-0156 to discover it.
bosh: '//jitsi-meet.example.com/' + subdir + 'http-bind',
// Websocket URL
// websocket: 'wss://jitsi-meet.example.com/' + subdir + 'xmpp-websocket',
// The real JID of focus participant - can be overridden here
// Do not change username - FIXME: Make focus username configurable
// https://github.com/jitsi/jitsi-meet/issues/7376
// focusUserJid: 'focus@auth.jitsi-meet.example.com',
// Testing / experimental features.
//
testing: {
// Disables the End to End Encryption feature. Useful for debugging
// issues related to insertable streams.
// disableE2EE: false,
// Enables XMPP WebSocket (as opposed to BOSH) for the given amount of users.
// mobileXmppWsThreshold: 10, // enable XMPP WebSockets on mobile for 10% of the users
// P2P test mode disables automatic switching to P2P when there are 2
// participants in the conference.
// p2pTestMode: false,
// Enables the test specific features consumed by jitsi-meet-torture
// testMode: false,
// Disables the auto-play behavior of *all* newly created video element.
// This is useful when the client runs on a host with limited resources.
// noAutoPlayVideo: false,
// Whether to use fake constraints (height: 99999, width: 99999) when calling getDisplayMedia on
// Chromium based browsers. This is intended as a workaround for
// https://bugs.chromium.org/p/chromium/issues/detail?id=1056311
// setScreenSharingResolutionConstraints: true,
// Enable callstats only for a percentage of users.
// This takes a value between 0 and 100 which determines the probability for
// the callstats to be enabled.
// callStatsThreshold: 5, // enable callstats for 5% of the users.
},
// Disables moderator indicators.
// disableModeratorIndicator: false,
// Disables the reactions feature.
// disableReactions: true,
// Disables the reactions moderation feature.
// disableReactionsModeration: false,
// Disables polls feature.
// disablePolls: false,
// Disables self-view tile. (hides it from tile view and from filmstrip)
// disableSelfView: false,
// Disables self-view settings in UI
// disableSelfViewSettings: false,
// screenshotCapture : {
// Enables the screensharing capture feature.
// enabled: false,
//
// The mode for the screenshot capture feature.
// Can be either 'recording' - screensharing screenshots are taken
// only when the recording is also on,
// or 'always' - screensharing screenshots are always taken.
// mode: 'recording',
// }
// Disables ICE/UDP by filtering out local and remote UDP candidates in
// signalling.
// webrtcIceUdpDisable: false,
// Disables ICE/TCP by filtering out local and remote TCP candidates in
// signalling.
// webrtcIceTcpDisable: false,
// Media
//
// Enable unified plan implementation support on Chromium based browsers.
// enableUnifiedOnChrome: false,
// Audio
// Disable measuring of audio levels.
// disableAudioLevels: false,
// audioLevelsInterval: 200,
// Enabling this will run the lib-jitsi-meet no audio detection module which
// will notify the user if the current selected microphone has no audio
// input and will suggest another valid device if one is present.
enableNoAudioDetection: true,
// Enabling this will show a "Save Logs" link in the GSM popover that can be
// used to collect debug information (XMPP IQs, SDP offer/answer cycles)
// about the call.
// enableSaveLogs: false,
// Enabling this will hide the "Show More" link in the GSM popover that can be
// used to display more statistics about the connection (IP, Port, protocol, etc).
// disableShowMoreStats: true,
// Enabling this will run the lib-jitsi-meet noise detection module which will
// notify the user if there is noise, other than voice, coming from the current
// selected microphone. The purpose it to let the user know that the input could
// be potentially unpleasant for other meeting participants.
enableNoisyMicDetection: true,
// Start the conference in audio only mode (no video is being received nor
// sent).
// startAudioOnly: false,
// Every participant after the Nth will start audio muted.
// startAudioMuted: 10,
// Start calls with audio muted. Unlike the option above, this one is only
// applied locally. FIXME: having these 2 options is confusing.
// startWithAudioMuted: false,
// Enabling it (with #params) will disable local audio output of remote
// participants and to enable it back a reload is needed.
// startSilent: false,
// Enables support for opus-red (redundancy for Opus).
// enableOpusRed: false,
// Specify audio quality stereo and opusMaxAverageBitrate values in order to enable HD audio.
// Beware, by doing so, you are disabling echo cancellation, noise suppression and AGC.
// Specify enableOpusDtx to enable support for opus-dtx where
// audio packets won’t be transmitted while participant is silent or muted.
// audioQuality: {
// stereo: false,
// opusMaxAverageBitrate: null, // Value to fit the 6000 to 510000 range.
// enableOpusDtx: false,
// },
// Video
// Sets the preferred resolution (height) for local video. Defaults to 720.
// resolution: 720,
// Specifies whether the raised hand will hide when someone becomes a dominant speaker or not
// disableRemoveRaisedHandOnFocus: false,
// speakerStats: {
// // Specifies whether the speaker stats is enable or not.
// disabled: false,
// // Specifies whether there will be a search field in speaker stats or not.
// disableSearch: false,
// // Specifies whether participants in speaker stats should be ordered or not, and with what priority.
// // 'role', <- Moderators on top.
// // 'name', <- Alphabetically by name.
// // 'hasLeft', <- The ones that have left in the bottom.
// order: [
// 'role',
// 'name',
// 'hasLeft',
// ],
// },
// DEPRECATED. Please use speakerStats.disableSearch instead.
// Specifies whether there will be a search field in speaker stats or not
// disableSpeakerStatsSearch: false,
// DEPRECATED. Please use speakerStats.order .
// Specifies whether participants in speaker stats should be ordered or not, and with what priority
// speakerStatsOrder: [
// 'role', <- Moderators on top
// 'name', <- Alphabetically by name
// 'hasLeft', <- The ones that have left in the bottom
// ], <- the order of the array elements determines priority
// How many participants while in the tile view mode, before the receiving video quality is reduced from HD to SD.
// Use -1 to disable.
// maxFullResolutionParticipants: 2,
// w3c spec-compliant video constraints to use for video capture. Currently
// used by browsers that return true from lib-jitsi-meet's
// util#browser#usesNewGumFlow. The constraints are independent from
// this config's resolution value. Defaults to requesting an ideal
// resolution of 720p.
// constraints: {
// video: {
// height: {
// ideal: 720,
// max: 720,
// min: 240,
// },
// },
// },
// Enable / disable simulcast support.
// disableSimulcast: false,
// Enable / disable layer suspension. If enabled, endpoints whose HD layers are not in use will be suspended
// (no longer sent) until they are requested again. This is enabled by default. This must be enabled for screen
// sharing to work as expected on Chrome. Disabling this might result in low resolution screenshare being sent
// by the client.
// enableLayerSuspension: false,
// Every participant after the Nth will start video muted.
// startVideoMuted: 10,
// Start calls with video muted. Unlike the option above, this one is only
// applied locally. FIXME: having these 2 options is confusing.
// startWithVideoMuted: false,
// If set to true, prefer to use the H.264 video codec (if supported).
// Note that it's not recommended to do this because simulcast is not
// supported when using H.264. For 1-to-1 calls this setting is enabled by
// default and can be toggled in the p2p section.
// This option has been deprecated, use preferredCodec under videoQuality section instead.
// preferH264: true,
// If set to true, disable H.264 video codec by stripping it out of the
// SDP.
// disableH264: false,
// Desktop sharing
// Optional desktop sharing frame rate options. Default value: min:5, max:5.
// desktopSharingFrameRate: {
// min: 5,
// max: 5,
// },
// This option has been deprecated since it is no longer supported as per the w3c spec.
// https://w3c.github.io/mediacapture-screen-share/#dom-mediadevices-getdisplaymedia. If the user has not
// interacted with the webpage before the getDisplayMedia call, the promise will be rejected by the browser. This
// has already been implemented in Firefox and Safari and will be implemented in Chrome soon.
// https://bugs.chromium.org/p/chromium/issues/detail?id=1198918
// startScreenSharing: false,
// Recording
// DEPRECATED. Use recordingService.enabled instead.
// fileRecordingsEnabled: false,
// Enable the dropbox integration.
// dropbox: {
// appKey: '<APP_KEY>', // Specify your app key here.
// // A URL to redirect the user to, after authenticating
// // by default uses:
// // 'https://jitsi-meet.example.com/static/oauth.html'
// redirectURI:
// 'https://jitsi-meet.example.com/subfolder/static/oauth.html',
// },
// recordingService: {
// // When integrations like dropbox are enabled only that will be shown,
// // by enabling fileRecordingsServiceEnabled, we show both the integrations
// // and the generic recording service (its configuration and storage type
// // depends on jibri configuration)
// enabled: false,
// // Whether to show the possibility to share file recording with other people
// // (e.g. meeting participants), based on the actual implementation
// // on the backend.
// sharingEnabled: false,
// // Hide the warning that says we only store the recording for 24 hours.
// hideStorageWarning: false,
// },
// DEPRECATED. Use recordingService.enabled instead.
// fileRecordingsServiceEnabled: false,
// DEPRECATED. Use recordingService.sharingEnabled instead.
// fileRecordingsServiceSharingEnabled: false,
// Local recording configuration.
// localRecording: {
// // Whether to disable local recording or not.
// disable: false,
// // Whether to notify all participants when a participant is recording locally.
// notifyAllParticipants: false,
// // Whether to disable the self recording feature (only local participant streams).
// disableSelfRecording: false,
// },
// Customize the Live Streaming dialog. Can be modified for a non-YouTube provider.
// liveStreaming: {
// // Whether to enable live streaming or not.
// enabled: false,
// // Terms link
// termsLink: 'https://www.youtube.com/t/terms',
// // Data privacy link
// dataPrivacyLink: 'https://policies.google.com/privacy',
// // RegExp string that validates the stream key input field
// validatorRegExpString: '^(?:[a-zA-Z0-9]{4}(?:-(?!$)|$)){4}',
// // Documentation reference for the live streaming feature.
// helpLink: 'https://jitsi.org/live'
// },
// DEPRECATED. Use liveStreaming.enabled instead.
// liveStreamingEnabled: false,
// DEPRECATED. Use transcription.enabled instead.
// transcribingEnabled: false,
// DEPRECATED. Use transcription.useAppLanguage instead.
// transcribeWithAppLanguage: true,
// DEPRECATED. Use transcription.preferredLanguage instead.
// preferredTranscribeLanguage: 'en-US',
// DEPRECATED. Use transcription.autoCaptionOnRecord instead.
// autoCaptionOnRecord: false,
// Transcription options.
// transcription: {
// // Whether the feature should be enabled or not.
// enabled: false,
// // Translation languages.
// // Available languages can be found in
// // ./src/react/features/transcribing/translation-languages.json.
// translationLanguages: ['en', 'es', 'fr', 'ro'],
// // Important languages to show on the top of the language list.
// translationLanguagesHead: ['en'],
// // If true transcriber will use the application language.
// // The application language is either explicitly set by participants in their settings or automatically
// // detected based on the environment, e.g. if the app is opened in a chrome instance which
// // is using french as its default language then transcriptions for that participant will be in french.
// // Defaults to true.
// useAppLanguage: true,
// // Transcriber language. This settings will only work if "useAppLanguage"
// // is explicitly set to false.
// // Available languages can be found in
// // ./src/react/features/transcribing/transcriber-langs.json.
// preferredLanguage: 'en-US',
// // Disable start transcription for all participants.
// disableStartForAll: false,
// // Enables automatic turning on captions when recording is started
// autoCaptionOnRecord: false,
// },
// Misc
// Default value for the channel "last N" attribute. -1 for unlimited.
channelLastN: -1,
// Connection indicators
// connectionIndicators: {
// autoHide: true,
// autoHideTimeout: 5000,
// disabled: false,
// disableDetails: false,
// inactiveDisabled: false
// },
// Provides a way for the lastN value to be controlled through the UI.
// When startLastN is present, conference starts with a last-n value of startLastN and channelLastN
// value will be used when the quality level is selected using "Manage Video Quality" slider.
// startLastN: 1,
// Provides a way to use different "last N" values based on the number of participants in the conference.
// The keys in an Object represent number of participants and the values are "last N" to be used when number of
// participants gets to or above the number.
//
// For the given example mapping, "last N" will be set to 20 as long as there are at least 5, but less than
// 29 participants in the call and it will be lowered to 15 when the 30th participant joins. The 'channelLastN'
// will be used as default until the first threshold is reached.
//
// lastNLimits: {
// 5: 20,
// 30: 15,
// 50: 10,
// 70: 5,
// 90: 2,
// },
// Provides a way to translate the legacy bridge signaling messages, 'LastNChangedEvent',
// 'SelectedEndpointsChangedEvent' and 'ReceiverVideoConstraint' into the new 'ReceiverVideoConstraints' message
// that invokes the new bandwidth allocation algorithm in the bridge which is described here
// - https://github.com/jitsi/jitsi-videobridge/blob/master/doc/allocation.md.
// useNewBandwidthAllocationStrategy: false,
// Specify the settings for video quality optimizations on the client.
// videoQuality: {
// // Provides a way to prevent a video codec from being negotiated on the JVB connection. The codec specified
// // here will be removed from the list of codecs present in the SDP answer generated by the client. If the
// // same codec is specified for both the disabled and preferred option, the disable settings will prevail.
// // Note that 'VP8' cannot be disabled since it's a mandatory codec, the setting will be ignored in this case.
// disabledCodec: 'H264',
//
// // Provides a way to set a preferred video codec for the JVB connection. If 'H264' is specified here,
// // simulcast will be automatically disabled since JVB doesn't support H264 simulcast yet. This will only
// // rearrange the the preference order of the codecs in the SDP answer generated by the browser only if the
// // preferred codec specified here is present. Please ensure that the JVB offers the specified codec for this
// // to take effect.
// preferredCodec: 'VP8',
//
// // Provides a way to enforce the preferred codec for the conference even when the conference has endpoints
// // that do not support the preferred codec. For example, older versions of Safari do not support VP9 yet.
// // This will result in Safari not being able to decode video from endpoints sending VP9 video.
// // When set to false, the conference falls back to VP8 whenever there is an endpoint that doesn't support the
// // preferred codec and goes back to the preferred codec when that endpoint leaves.
// enforcePreferredCodec: false,
//
// // Provides a way to configure the maximum bitrates that will be enforced on the simulcast streams for
// // video tracks. The keys in the object represent the type of the stream (LD, SD or HD) and the values
// // are the max.bitrates to be set on that particular type of stream. The actual send may vary based on
// // the available bandwidth calculated by the browser, but it will be capped by the values specified here.
// // This is currently not implemented on app based clients on mobile.
// maxBitratesVideo: {
// H264: {
// low: 200000,
// standard: 500000,
// high: 1500000,
// },
// VP8 : {
// low: 200000,
// standard: 500000,
// high: 1500000,
// },
// VP9: {
// low: 100000,
// standard: 300000,
// high: 1200000,
// },
// },
//
// // The options can be used to override default thresholds of video thumbnail heights corresponding to
// // the video quality levels used in the application. At the time of this writing the allowed levels are:
// // 'low' - for the low quality level (180p at the time of this writing)
// // 'standard' - for the medium quality level (360p)
// // 'high' - for the high quality level (720p)
// // The keys should be positive numbers which represent the minimal thumbnail height for the quality level.
// //
// // With the default config value below the application will use 'low' quality until the thumbnails are
// // at least 360 pixels tall. If the thumbnail height reaches 720 pixels then the application will switch to
// // the high quality.
// minHeightForQualityLvl: {
// 360: 'standard',
// 720: 'high',
// },
//
// },
// Notification timeouts
// notificationTimeouts: {
// short: 2500,
// medium: 5000,
// long: 10000,
// },
// // Options for the recording limit notification.
// recordingLimit: {
//
// // The recording limit in minutes. Note: This number appears in the notification text
// // but doesn't enforce the actual recording time limit. This should be configured in
// // jibri!
// limit: 60,
//
// // The name of the app with unlimited recordings.
// appName: 'Unlimited recordings APP',
//
// // The URL of the app with unlimited recordings.
// appURL: 'https://unlimited.recordings.app.com/',
// },
// Disables or enables RTX (RFC 4588) (defaults to false).
// disableRtx: false,
// Moves all Jitsi Meet 'beforeunload' logic (cleanup, leaving, disconnecting, etc) to the 'unload' event.
// disableBeforeUnloadHandlers: true,
// Disables or enables TCC support in this client (default: enabled).
// enableTcc: true,
// Disables or enables REMB support in this client (default: enabled).
// enableRemb: true,
// Enables ICE restart logic in LJM and displays the page reload overlay on
// ICE failure. Current disabled by default because it's causing issues with
// signaling when Octo is enabled. Also when we do an "ICE restart"(which is
// not a real ICE restart), the client maintains the TCC sequence number
// counter, but the bridge resets it. The bridge sends media packets with
// TCC sequence numbers starting from 0.
// enableIceRestart: false,
// Enables forced reload of the client when the call is migrated as a result of
// the bridge going down.
// enableForcedReload: true,
// Use TURN/UDP servers for the jitsi-videobridge connection (by default
// we filter out TURN/UDP because it is usually not needed since the
// bridge itself is reachable via UDP)
// useTurnUdp: false
// Enable support for encoded transform in supported browsers. This allows
// E2EE to work in Safari if the corresponding flag is enabled in the browser.
// Experimental.
// enableEncodedTransformSupport: false,
// UI
//
// Disables responsive tiles.
// disableResponsiveTiles: false,
// Hides lobby button
// hideLobbyButton: false,
// If Lobby is enabled starts knocking automatically.
// autoKnockLobby: false,
// Enable lobby chat.
// enableLobbyChat: true,
// DEPRECATED! Use `breakoutRooms.hideAddRoomButton` instead.
// Hides add breakout room button
// hideAddRoomButton: false,
// Require users to always specify a display name.
// requireDisplayName: true,
// DEPRECATED! Use 'welcomePage.disabled' instead.
// Whether to use a welcome page or not. In case it's false a random room
// will be joined when no room is specified.
// enableWelcomePage: true,
// Configs for welcome page.
// welcomePage: {
// // Whether to disable welcome page. In case it's disabled a random room
// // will be joined when no room is specified.
// disabled: false,
// // If set,landing page will redirect to this URL.
// customUrl: ''
// },
// Disable app shortcuts that are registered upon joining a conference
// disableShortcuts: false,
// Disable initial browser getUserMedia requests.
// This is useful for scenarios where users might want to start a conference for screensharing only
// disableInitialGUM: false,
// Enabling the close page will ignore the welcome page redirection when
// a call is hangup.
// enableClosePage: false,
// Disable hiding of remote thumbnails when in a 1-on-1 conference call.
// Setting this to null, will also disable showing the remote videos
// when the toolbar is shown on mouse movements
// disable1On1Mode: null | false | true,
// Default local name to be displayed
// defaultLocalDisplayName: 'me',
// Default remote name to be displayed
// defaultRemoteDisplayName: 'Fellow Jitster',
// Hides the display name from the participant thumbnail
// hideDisplayName: false,
// Hides the dominant speaker name badge that hovers above the toolbox
// hideDominantSpeakerBadge: false,
// Default language for the user interface. Cannot be overwritten.
// defaultLanguage: 'en',
// Disables profile and the edit of all fields from the profile settings (display name and email)
// disableProfile: false,
// Hides the email section under profile settings.
// hideEmailInSettings: false,
// When enabled the password used for locking a room is restricted to up to the number of digits specified
// default: roomPasswordNumberOfDigits: false,
// roomPasswordNumberOfDigits: 10,
// Message to show the users. Example: 'The service will be down for
// maintenance at 01:00 AM GMT,
// noticeMessage: '',
// Enables calendar integration, depends on googleApiApplicationClientID
// and microsoftApiApplicationClientID
// enableCalendarIntegration: false,
// Configs for prejoin page.
// prejoinConfig: {
// // When 'true', it shows an intermediate page before joining, where the user can configure their devices.
// // This replaces `prejoinPageEnabled`.
// enabled: true,
// // Hides the participant name editing field in the prejoin screen.
// // If requireDisplayName is also set as true, a name should still be provided through
// // either the jwt or the userInfo from the iframe api init object in order for this to have an effect.
// hideDisplayName: false,
// // List of buttons to hide from the extra join options dropdown.
// hideExtraJoinButtons: ['no-audio', 'by-phone'],
// },
// When 'true', the user cannot edit the display name.
// (Mainly useful when used in conjunction with the JWT so the JWT name becomes read only.)
// readOnlyName: false,
// If etherpad integration is enabled, setting this to true will
// automatically open the etherpad when a participant joins. This
// does not affect the mobile app since opening an etherpad
// obscures the conference controls -- it's better to let users
// choose to open the pad on their own in that case.
// openSharedDocumentOnJoin: false,
// If true, shows the unsafe room name warning label when a room name is
// deemed unsafe (due to the simplicity in the name) and a password is not
// set or the lobby is not enabled.
// enableInsecureRoomNameWarning: false,
// Whether to automatically copy invitation URL after creating a room.
// Document should be focused for this option to work
// enableAutomaticUrlCopy: false,
// Array with avatar URL prefixes that need to use CORS.
// corsAvatarURLs: [ 'https://www.gravatar.com/avatar/' ],
// Base URL for a Gravatar-compatible service. Defaults to Gravatar.
// DEPRECATED! Use `gravatar.baseUrl` instead.
// gravatarBaseURL: 'https://www.gravatar.com/avatar/',
// Setup for Gravatar-compatible services.
// gravatar: {
// // Defaults to Gravatar.
// baseUrl: 'https://www.gravatar.com/avatar/',
// // True if Gravatar should be disabled.
// disabled: false,
// },
// App name to be displayed in the invitation email subject, as an alternative to
// interfaceConfig.APP_NAME.
// inviteAppName: null,
// Moved from interfaceConfig(TOOLBAR_BUTTONS).
// The name of the toolbar buttons to display in the toolbar, including the
// "More actions" menu. If present, the button will display. Exceptions are
// "livestreaming" and "recording" which also require being a moderator and
// some other values in config.js to be enabled. Also, the "profile" button will
// not display for users with a JWT.
// Notes:
// - it's impossible to choose which buttons go in the "More actions" menu
// - it's impossible to control the placement of buttons
// - 'desktop' controls the "Share your screen" button
// - if `toolbarButtons` is undefined, we fallback to enabling all buttons on the UI
// toolbarButtons: [
// 'camera',
// 'chat',
// 'closedcaptions',
// 'desktop',
// 'download',
// 'embedmeeting',
// 'etherpad',
// 'feedback',
// 'filmstrip',
// 'fullscreen',
// 'hangup',
// 'help',
// 'highlight',
// 'invite',
// 'linktosalesforce',
// 'livestreaming',
// 'microphone',
// 'noisesuppression',
// 'participants-pane',
// 'profile',
// 'raisehand',
// 'recording',
// 'security',
// 'select-background',
// 'settings',
// 'shareaudio',
// 'sharedvideo',
// 'shortcuts',
// 'stats',
// 'tileview',
// 'toggle-camera',
// 'videoquality',
// 'whiteboard',
// ],
// Holds values related to toolbar visibility control.
// toolbarConfig: {
// // Moved from interfaceConfig.INITIAL_TOOLBAR_TIMEOUT
// // The initial number of milliseconds for the toolbar buttons to be visible on screen.
// initialTimeout: 20000,
// // Moved from interfaceConfig.TOOLBAR_TIMEOUT
// // Number of milliseconds for the toolbar buttons to be visible on screen.
// timeout: 4000,
// // Moved from interfaceConfig.TOOLBAR_ALWAYS_VISIBLE
// // Whether toolbar should be always visible or should hide after x milliseconds.
// alwaysVisible: false,
// // Indicates whether the toolbar should still autohide when chat is open
// autoHideWhileChatIsOpen: false,
// },
// Toolbar buttons which have their click/tap event exposed through the API on
// `toolbarButtonClicked`. Passing a string for the button key will
// prevent execution of the click/tap routine; passing an object with `key` and
// `preventExecution` flag on false will not prevent execution of the click/tap
// routine. Below array with mixed mode for passing the buttons.
// buttonsWithNotifyClick: [
// 'camera',
// {
// key: 'chat',
// preventExecution: false
// },
// {
// key: 'closedcaptions',
// preventExecution: true
// },
// 'desktop',
// 'download',
// 'embedmeeting',
// 'end-meeting',
// 'etherpad',
// 'feedback',
// 'filmstrip',
// 'fullscreen',
// 'hangup',
// 'hangup-menu',
// 'help',
// {
// key: 'invite',
// preventExecution: false
// },
// 'livestreaming',
// 'microphone',
// 'mute-everyone',
// 'mute-video-everyone',
// 'noisesuppression',
// 'participants-pane',
// 'profile',
// {
// key: 'raisehand',
// preventExecution: true
// },
// 'recording',
// 'security',
// 'select-background',
// 'settings',
// 'shareaudio',
// 'sharedvideo',
// 'shortcuts',
// 'stats',
// 'tileview',
// 'toggle-camera',
// 'videoquality',
// // The add passcode button from the security dialog.
// {
// key: 'add-passcode',
// preventExecution: false
// },
// 'whiteboard',
// ],
// List of pre meeting screens buttons to hide. The values must be one or more of the 5 allowed buttons:
// 'microphone', 'camera', 'select-background', 'invite', 'settings'
// hiddenPremeetingButtons: [],
// Stats
//
// Whether to enable stats collection or not in the TraceablePeerConnection.
// This can be useful for debugging purposes (post-processing/analysis of
// the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
// estimation tests.
// gatherStats: false,
// The interval at which PeerConnection.getStats() is called. Defaults to 10000
// pcStatsInterval: 10000,
// To enable sending statistics to callstats.io you must provide the
// Application ID and Secret.
// callStatsID: '',
// callStatsSecret: '',
// callStatsApplicationLogsDisabled: false,
// The callstats initialize config params as described in the API:
// https://docs.callstats.io/docs/javascript#callstatsinitialize-with-app-secret
// callStatsConfigParams: {
// disableBeforeUnloadHandler: true, // disables callstats.js's window.onbeforeunload parameter.
// applicationVersion: "app_version", // Application version specified by the developer.
// disablePrecalltest: true, // disables the pre-call test, it is enabled by default.
// siteID: "siteID", // The name/ID of the site/campus from where the call/pre-call test is made.
// additionalIDs: { // additionalIDs object, contains application related IDs.
// customerID: "Customer Identifier. Example, walmart.",
// tenantID: "Tenant Identifier. Example, monster.",
// productName: "Product Name. Example, Jitsi.",
// meetingsName: "Meeting Name. Example, Jitsi loves callstats.",
// serverName: "Server/MiddleBox Name. Example, jvb-prod-us-east-mlkncws12.",
// pbxID: "PBX Identifier. Example, walmart.",
// pbxExtensionID: "PBX Extension Identifier. Example, 5625.",
// fqExtensionID: "Fully qualified Extension Identifier. Example, +71 (US) +5625.",
// sessionID: "Session Identifier. Example, session-12-34",
// },
// collectLegacyStats: true, //enables the collection of legacy stats in chrome browser
// collectIP: true, //enables the collection localIP address
// },
// Enables sending participants' display names to callstats
// enableDisplayNameInStats: false,
// Enables sending participants' emails (if available) to callstats and other analytics
// enableEmailInStats: false,
// faceLandmarks: {
// // Enables sharing your face coordinates. Used for centering faces within a video.
// enableFaceCentering: false,
// // Enables detecting face expressions and sharing data with other participants
// enableFaceExpressionsDetection: false,
// // Enables displaying face expressions in speaker stats
// enableDisplayFaceExpressions: false,
// // Enable rtc stats for face landmarks
// enableRTCStats: false,
// // Minimum required face movement percentage threshold for sending new face centering coordinates data.
// faceCenteringThreshold: 10,
// // Milliseconds for processing a new image capture in order to detect face coordinates if they exist.
// captureInterval: 1000,
// },
// Controls the percentage of automatic feedback shown to participants when callstats is enabled.
// The default value is 100%. If set to 0, no automatic feedback will be requested
// feedbackPercentage: 100,
// Privacy
//
// If third party requests are disabled, no other server will be contacted.
// This means avatars will be locally generated and callstats integration
// will not function.
// disableThirdPartyRequests: false,
// Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
//
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,
// Enable unified plan implementation support on Chromium for p2p connection.
// enableUnifiedOnChrome: false,
// 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,
// The STUN servers that will be used in the peer to peer connections
stunServers: [
// { urls: 'stun:jitsi-meet.example.com:3478' },
{ urls: 'stun:meet-jit-si-turnrelay.jitsi.net:443' },
],
},
analytics: {
// True if the analytics should be disabled
// disabled: false,
// 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>',
// Obfuscates room name sent to analytics (amplitude, rtcstats)
// Default value is false.
// obfuscateRoomName: false,
// 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: false,
// rtcstatsStoreLogs: false,
// 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 10000ms.
// If the value is set to 0 getStats won't be polled and the rtcstats client
// will only send data related to RTCPeerConnection events.
// rtcstatsPollInterval: 10000,
// This determines if rtcstats sends the SDP to the rtcstats server or replaces
// all SDPs with an empty string instead.
// rtcstatsSendSdp: false,
// 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'],