Move app files into cluster subdirectories; add prod garage

1 files changed, 0 insertions, 773 deletions

diff --git a/app/jitsi/integration/meet/config.js b/app/jitsi/integration/meet/config.js
deleted file mode 100644
index 04414c3..0000000
--- a/app/jitsi/integration/meet/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: '//rayonx.machine.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: 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
-
-    // 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: 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,
-
-    // 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: 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/';
-
-    // 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 */