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Diffstat (limited to 'app/jitsi/config/config.js')
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diff --git a/app/jitsi/config/config.js b/app/jitsi/config/config.js deleted file mode 100644 index 9464f37..0000000 --- a/app/jitsi/config/config.js +++ /dev/null @@ -1,773 +0,0 @@ -/* 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 */ |