1125 lines
47 KiB
JavaScript
1125 lines
47 KiB
JavaScript
/* eslint-disable no-unused-vars, no-var */
|
|
|
|
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.jitsi-meet.example.com'
|
|
},
|
|
|
|
// BOSH URL. FIXME: use XEP-0156 to discover it.
|
|
bosh: '//jitsi-meet.example.com/http-bind',
|
|
|
|
// Websocket URL
|
|
// websocket: 'wss://jitsi-meet.example.com/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/disables thumbnail reordering in the filmstrip. It is enabled by default unless explicitly
|
|
// disabled by the below option.
|
|
// enableThumbnailReordering: true,
|
|
|
|
// 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
|
|
|
|
// Enable / disable 500 Kbps bitrate cap on desktop tracks. When enabled,
|
|
// simulcast is turned off for the desktop share. If presenter is turned
|
|
// on while screensharing is in progress, the max bitrate is automatically
|
|
// adjusted to 2.5 Mbps. This takes a value between 0 and 1 which determines
|
|
// the probability for this to be enabled. This setting has been deprecated.
|
|
// desktopSharingFrameRate.max now determines whether simulcast will be enabled
|
|
// or disabled for the screenshare.
|
|
// capScreenshareBitrate: 1 // 0 to disable - deprecated.
|
|
|
|
// 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 polls feature.
|
|
// disablePolls: false,
|
|
|
|
// 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.
|
|
// audioQuality: {
|
|
// stereo: false,
|
|
// opusMaxAverageBitrate: null // Value to fit the 6000 to 510000 range.
|
|
// },
|
|
|
|
// 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,
|
|
|
|
// Specifies whether there will be a search field in speaker stats or not
|
|
// disableSpeakerStatsSearch: false,
|
|
|
|
// 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
|
|
// },
|
|
|
|
// Try to start calls with screen-sharing instead of camera video.
|
|
// startScreenSharing: false,
|
|
|
|
// Recording
|
|
|
|
// Whether to enable file recording or not.
|
|
// 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'
|
|
// },
|
|
// 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)
|
|
// fileRecordingsServiceEnabled: 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.
|
|
// fileRecordingsServiceSharingEnabled: false,
|
|
|
|
// Whether to enable live streaming or not.
|
|
// liveStreamingEnabled: false,
|
|
|
|
// Transcription (in interface_config,
|
|
// subtitles and buttons can be configured)
|
|
// transcribingEnabled: false,
|
|
|
|
// 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.
|
|
// transcribeWithAppLanguage: true,
|
|
|
|
// Transcriber language. This settings will only work if "transcribeWithAppLanguage" is explicitly set to false.
|
|
// Available languages can be found in
|
|
// ./src/react/features/transcribing/transcriber-langs.json.
|
|
// preferredTranscribeLanguage: 'en-US',
|
|
|
|
// 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,
|
|
// 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'
|
|
// },
|
|
//
|
|
// // Provides a way to resize the desktop track to 720p (if it is greater than 720p) before creating a canvas
|
|
// // for the presenter mode (camera picture-in-picture mode with screenshare).
|
|
// resizeDesktopForPresenter: false
|
|
// },
|
|
|
|
// // 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,
|
|
|
|
// 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,
|
|
|
|
// Require users to always specify a display name.
|
|
// requireDisplayName: true,
|
|
|
|
// 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,
|
|
|
|
// 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 language for the user interface.
|
|
// defaultLanguage: 'en',
|
|
|
|
// Disables profile and the edit of all fields from the profile settings (display name and email)
|
|
// disableProfile: false,
|
|
|
|
// Whether or not some features are checked based on token.
|
|
// enableFeaturesBasedOnToken: false,
|
|
|
|
// When enabled the password used for locking a room is restricted to up to the number of digits specified
|
|
// roomPasswordNumberOfDigits: 10,
|
|
// default: roomPasswordNumberOfDigits: false,
|
|
|
|
// 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,
|
|
|
|
// When 'true', it shows an intermediate page before joining, where the user can configure their devices.
|
|
// prejoinPageEnabled: false,
|
|
|
|
// When 'true', the user cannot edit the display name.
|
|
// (Mainly useful when used in conjuction 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,
|
|
|
|
// Base URL for a Gravatar-compatible service. Defaults to libravatar.
|
|
// gravatarBaseURL: 'https://seccdn.libravatar.org/avatar/',
|
|
|
|
// 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',
|
|
// 'invite',
|
|
// 'livestreaming',
|
|
// 'microphone',
|
|
// 'mute-everyone',
|
|
// 'mute-video-everyone',
|
|
// 'participants-pane',
|
|
// 'profile',
|
|
// 'raisehand',
|
|
// 'recording',
|
|
// 'security',
|
|
// 'select-background',
|
|
// 'settings',
|
|
// 'shareaudio',
|
|
// 'sharedvideo',
|
|
// 'shortcuts',
|
|
// 'stats',
|
|
// 'tileview',
|
|
// 'toggle-camera',
|
|
// 'videoquality',
|
|
// '__end'
|
|
// ],
|
|
|
|
// Holds values related to toolbar visibility control.
|
|
// toolbarConfig: {
|
|
// // Moved from interfaceConfig.INITIAL_TOOLBAR_TIMEOUT
|
|
// // The initial numer of miliseconds for the toolbar buttons to be visible on screen.
|
|
// initialTimeout: 20000,
|
|
// // Moved from interfaceConfig.TOOLBAR_TIMEOUT
|
|
// // Number of miliseconds 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 miliseconds.
|
|
// alwaysVisible: false
|
|
// },
|
|
|
|
// Toolbar buttons which have their click event exposed through the API on
|
|
// `toolbarButtonClicked` event instead of executing the normal click routine.
|
|
// buttonsWithNotifyClick: [
|
|
// 'camera',
|
|
// 'chat',
|
|
// 'closedcaptions',
|
|
// 'desktop',
|
|
// 'download',
|
|
// 'embedmeeting',
|
|
// 'etherpad',
|
|
// 'feedback',
|
|
// 'filmstrip',
|
|
// 'fullscreen',
|
|
// 'hangup',
|
|
// 'help',
|
|
// 'invite',
|
|
// 'livestreaming',
|
|
// 'microphone',
|
|
// 'mute-everyone',
|
|
// 'mute-video-everyone',
|
|
// 'participants-pane',
|
|
// 'profile',
|
|
// 'raisehand',
|
|
// 'recording',
|
|
// 'security',
|
|
// 'select-background',
|
|
// 'settings',
|
|
// 'shareaudio',
|
|
// 'sharedvideo',
|
|
// 'shortcuts',
|
|
// 'stats',
|
|
// 'tileview',
|
|
// 'toggle-camera',
|
|
// 'videoquality',
|
|
// '__end'
|
|
// ],
|
|
|
|
// 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: '',
|
|
|
|
// 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,
|
|
|
|
// 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>'
|
|
|
|
// 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"
|
|
},
|
|
|
|
// Array<string> of disabled sounds.
|
|
// Possible values:
|
|
// - 'ASKED_TO_UNMUTE_SOUND'
|
|
// - 'E2EE_OFF_SOUND'
|
|
// - 'E2EE_ON_SOUND'
|
|
// - 'INCOMING_MSG_SOUND'
|
|
// - 'KNOCKING_PARTICIPANT_SOUND'
|
|
// - 'LIVE_STREAMING_OFF_SOUND'
|
|
// - 'LIVE_STREAMING_ON_SOUND'
|
|
// - 'NO_AUDIO_SIGNAL_SOUND'
|
|
// - 'NOISY_AUDIO_INPUT_SOUND'
|
|
// - 'OUTGOING_CALL_EXPIRED_SOUND'
|
|
// - 'OUTGOING_CALL_REJECTED_SOUND'
|
|
// - 'OUTGOING_CALL_RINGING_SOUND'
|
|
// - 'OUTGOING_CALL_START_SOUND'
|
|
// - 'PARTICIPANT_JOINED_SOUND'
|
|
// - 'PARTICIPANT_LEFT_SOUND'
|
|
// - 'RAISE_HAND_SOUND'
|
|
// - 'REACTION_SOUND'
|
|
// - 'RECORDING_OFF_SOUND'
|
|
// - 'RECORDING_ON_SOUND'
|
|
// - 'TALK_WHILE_MUTED_SOUND'
|
|
// disabledSounds: [],
|
|
|
|
// DEPRECATED! Use `disabledSounds` instead.
|
|
// Decides whether the start/stop recording audio notifications should play on record.
|
|
// disableRecordAudioNotification: false,
|
|
|
|
// DEPRECATED! Use `disabledSounds` instead.
|
|
// 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,
|
|
|
|
// DEPRECATED! Use `disabledSounds` instead.
|
|
// Disables the sounds that play when a chat message is received.
|
|
// disableIncomingMessageSound: 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
|
|
// the `toolbarButtons`-array 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 the 'Grant moderator' button will be disabled.
|
|
// disableGrantModerator: 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.
|
|
The config file should be in JSON.
|
|
None of the fields are mandatory and the response must have the shape:
|
|
{
|
|
// The domain url to apply (will replace the domain in the sharing conference link/embed section)
|
|
inviteDomain: 'example-company.org,
|
|
// 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',
|
|
// Overwrite for pool of background images for avatars
|
|
avatarBackgrounds: ['url(https://example.com/avatar-background-1.png)', '#FFF'],
|
|
// The lobby/prejoin screen background
|
|
premeetingBackground: 'url(https://example.com/premeeting-background.png)'
|
|
}
|
|
*/
|
|
// dynamicBrandingUrl: '',
|
|
|
|
// When true the user cannot add more images to be used as virtual background.
|
|
// Only the default ones from will be available.
|
|
// disableAddingBackgroundImages: false,
|
|
|
|
// 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,
|
|
|
|
// Controls the visibility and behavior of the top header conference info labels.
|
|
// If a label's id is not in any of the 2 arrays, it will not be visible at all on the header.
|
|
// conferenceInfo: {
|
|
// // those labels will not be hidden in tandem with the toolbox.
|
|
// alwaysVisible: ['recording', 'local-recording'],
|
|
// // those labels will be auto-hidden in tandem with the toolbox buttons.
|
|
// autoHide: [
|
|
// 'subject',
|
|
// 'conference-timer',
|
|
// 'participants-count',
|
|
// 'e2ee',
|
|
// 'transcribing',
|
|
// 'video-quality',
|
|
// 'insecure-room'
|
|
// ]
|
|
// },
|
|
|
|
// Hides the conference subject
|
|
// hideConferenceSubject: true,
|
|
|
|
// Hides the conference timer.
|
|
// hideConferenceTimer: true,
|
|
|
|
// Hides the recording label
|
|
// hideRecordingLabel: false,
|
|
|
|
// 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,
|
|
|
|
// etherpad ("shared document") integration.
|
|
//
|
|
|
|
// If set, add a "Open shared document" link to the bottom right menu that
|
|
// will open an etherpad document.
|
|
// etherpad_base: 'https://your-etherpad-installati.on/p/',
|
|
|
|
// If etherpad_base is set, and useRoomAsSharedDocumentName is set to true,
|
|
// open a pad with the name of the room (lowercased) instead of a pad with a
|
|
// random UUID.
|
|
// useRoomAsSharedDocumentName: true,
|
|
|
|
// List of undocumented settings used in jitsi-meet
|
|
/**
|
|
_immediateReloadThreshold
|
|
debug
|
|
debugAudioLevels
|
|
deploymentInfo
|
|
dialInConfCodeUrl
|
|
dialInNumbersUrl
|
|
dialOutAuthUrl
|
|
dialOutCodesUrl
|
|
disableRemoteControl
|
|
displayJids
|
|
externalConnectUrl
|
|
e2eeLabels
|
|
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
|
|
*/
|
|
|
|
/**
|
|
* Default interval (milliseconds) for triggering mouseMoved iframe API event
|
|
*/
|
|
mouseMoveCallbackInterval: 1000,
|
|
|
|
/**
|
|
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.connectedOneMember', // show when a participant joined
|
|
// 'notify.connectedTwoMembers', // show when two participants joined simultaneously
|
|
// 'notify.connectedThreePlusMembers', // show when more than 2 participants joined simultaneously
|
|
// '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.moderationStartedTitle', // shown when AV moderation is activated
|
|
// 'notify.moderationStoppedTitle', // shown when AV moderation is deactivated
|
|
// 'notify.moderationInEffectTitle', // shown when user attempts to unmute audio during AV moderation
|
|
// 'notify.moderationInEffectVideoTitle', // shown when user attempts to enable video during AV moderation
|
|
// 'notify.moderationInEffectCSTitle', // shown when user attempts to share content during AV moderation
|
|
// '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
|
|
// 'notify.unmute', // shown to moderator when user raises hand during AV moderation
|
|
// '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
|
|
// ],
|
|
|
|
// Prevent the filmstrip from autohiding when screen width is under a certain threshold
|
|
// disableFilmstripAutohiding: false,
|
|
|
|
// Specifies whether the chat emoticons are disabled or not
|
|
// disableChatSmileys: false,
|
|
|
|
// 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 */
|