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-/* eslint-disable no-unused-vars, no-var */
-
-var config = {
- // Connection
- //
-
- hosts: {
- // XMPP domain.
- domain: 'jitsi',
-
- // 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'
- },
-
- // BOSH URL. FIXME: use XEP-0156 to discover it.
- bosh: '//jitsi.deuxfleurs.fr/http-bind',
-
- // Websocket URL
- // websocket: 'wss://jitsi-meet.example.com/xmpp-websocket',
-
- // The name of client node advertised in XEP-0115 'c' stanza
- clientNode: 'http://jitsi.org/jitsimeet',
-
- // 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,
-
- // 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.
- // capScreenshareBitrate: 1 // 0 to disable
-
- // 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 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
- //
-
- // 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 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: false,
-
- // 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: 5,
-
- // 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
-
- // Sets the preferred target bitrate for the Opus audio codec by setting its
- // 'maxaveragebitrate' parameter. Currently not available in p2p mode.
- // Valid values are in the range 6000 to 510000
- // opusMaxAverageBitrate: 20000,
-
- // Enables support for opus-red (redundancy for Opus).
- // enableOpusRed: false
-
- // Video
-
- // Sets the preferred resolution (height) for local video. Defaults to 720.
- // resolution: 720,
-
- // 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.
- // enableLayerSuspension: false,
-
- // Every participant after the Nth will start video muted.
- startVideoMuted: 5,
-
- // 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,
-
- // 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,
-
- // 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
- // },
-
- // 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 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: {
- // low: 200000,
- // standard: 500000,
- // high: 1500000
- // },
- //
- // // 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,
-
- // 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
-
- // 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.
- // disable1On1Mode: false,
-
- // Default language for the user interface.
- defaultLanguage: 'fr',
-
- // 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: true,
-
- // 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/';
-
- // 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: '',
-
- // Enables sending participants' display names to callstats
- // enableDisplayNameInStats: false,
-
- // Enables sending participants' emails (if available) to callstats and other analytics
- // enableEmailInStats: false,
-
- // 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,
-
- // 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' }
- ]
-
- // 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,
-
- // 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,
-
- // 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: '',
-
- // 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',
-
- // 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.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 */