text: /* eslint-disable no-unused-vars, no-var */
  
  var config = {
      // Connection
      //
  
      hosts: {
          // XMPP domain.
          domain: 'jitsi-meet.example.com',
  
          // When using authentication, domain for guest users.
          // anonymousdomain: 'guest.example.com',
  
          // Domain for authenticated users. Defaults to <domain>.
          // authdomain: 'jitsi-meet.example.com',
  
          // Jirecon recording component domain.
          // jirecon: 'jirecon.jitsi-meet.example.com',
  
          // Call control component (Jigasi).
          // call_control: 'callcontrol.jitsi-meet.example.com',
  
          // Focus component domain. Defaults to focus.<domain>.
          // focus: 'focus.jitsi-meet.example.com',
  
          // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
          muc: 'conference.jitsi-meet.example.com'
      },
  
      // BOSH URL. FIXME: use XEP-0156 to discover it.
      bosh: '//jitsi-meet.example.com/http-bind',
  
      // Websocket URL
      // websocket: 'wss://jitsi-meet.example.com/xmpp-websocket',
  
      // The 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
      // focusUserJid: 'focus@auth.jitsi-meet.example.com',
  
  
      // Testing / experimental features.
      //
  
      testing: {
          // 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
      },
  
      // 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 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
  
      // Video
  
      // Sets the preferred resolution (height) for local video. Defaults to 720.
      // resolution: 720,
  
      // 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.
      // preferH264: true,
  
      // If set to true, disable H.264 video codec by stripping it out of the
      // SDP.
      // disableH264: false,
  
      // Desktop sharing
  
      // The ID of the jidesha extension for Chrome.
      desktopSharingChromeExtId: null,
  
      // Whether desktop sharing should be disabled on Chrome.
      // desktopSharingChromeDisabled: false,
  
      // The media sources to use when using screen sharing with the Chrome
      // extension.
      desktopSharingChromeSources: [ 'screen', 'window', 'tab' ],
  
      // Required version of Chrome extension
      desktopSharingChromeMinExtVersion: '0.1',
  
      // Whether desktop sharing should be disabled on Firefox.
      // desktopSharingFirefoxDisabled: false,
  
      // 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,
  
      // Disables or enables RTX (RFC 4588) (defaults to false).
      // disableRtx: false,
  
      // Disables or enables TCC (the default is in Jicofo and set to true)
      // (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting
      // affects congestion control, it practically enables send-side bandwidth
      // estimations.
      // enableTcc: true,
  
      // Disables or enables REMB (the default is in Jicofo and set to false)
      // (draft-alvestrand-rmcat-remb-03). This setting affects congestion
      // control, it practically enables recv-side bandwidth estimations. When
      // both TCC and REMB are enabled, TCC takes precedence. When both are
      // disabled, then bandwidth estimations are disabled.
      // enableRemb: false,
  
      // Defines the minimum number of participants to start a call (the default
      // is set in Jicofo and set to 2).
      // minParticipants: 2,
  
      // Use XEP-0215 to fetch STUN and TURN servers.
      // useStunTurn: true,
  
      // Enable IPv6 support.
      // useIPv6: true,
  
      // Enables / disables a data communication channel with the Videobridge.
      // Values can be 'datachannel', 'websocket', true (treat it as
      // 'datachannel'), undefined (treat it as 'datachannel') and false (don't
      // open any channel).
      // openBridgeChannel: true,
  
  
      // UI
      //
  
      // Use display name as XMPP nickname.
      // useNicks: 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,
  
      // 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: 'en',
  
      // If true all users without a token will be considered guests and all users
      // with token will be considered non-guests. Only guests will be allowed to
      // edit their profile.
      enableUserRolesBasedOnToken: false,
  
      // Whether or not some features are checked based on token.
      // enableFeaturesBasedOnToken: false,
  
      // Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests.
      // lockRoomGuestEnabled: 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,
  
      // 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 name to callstats
      // enableDisplayNameInStats: false,
  
      // enables sending participants email 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,
  
          // Use XEP-0215 to fetch STUN and TURN servers.
          // useStunTurn: true,
  
          // The STUN servers that will be used in the peer to peer connections
          stunServers: [
  
              // { urls: 'stun:jitsi-meet.example.com:4446' },
              { 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).
          preferH264: true
  
          // If set to true, disable H.264 video codec by stripping it out of the
          // SDP.
          // disableH264: false,
  
          // 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'
  
          // The Amplitude APP Key:
          // amplitudeAPPKey: '<APP_KEY>'
  
          // 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"
          // ],
      },
  
      // 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,
  
      // List of undocumented settings used in jitsi-meet
      /**
       _immediateReloadThreshold
       autoRecord
       autoRecordToken
       debug
       debugAudioLevels
       deploymentInfo
       dialInConfCodeUrl
       dialInNumbersUrl
       dialOutAuthUrl
       dialOutCodesUrl
       disableRemoteControl
       displayJids
       etherpad_base
       externalConnectUrl
       firefox_fake_device
       googleApiApplicationClientID
       iAmRecorder
       iAmSipGateway
       microsoftApiApplicationClientID
       peopleSearchQueryTypes
       peopleSearchUrl
       requireDisplayName
       tokenAuthUrl
       */
  
      // List of undocumented settings used in lib-jitsi-meet
      /**
       _peerConnStatusOutOfLastNTimeout
       _peerConnStatusRtcMuteTimeout
       abTesting
       avgRtpStatsN
       callStatsConfIDNamespace
       callStatsCustomScriptUrl
       desktopSharingSources
       disableAEC
       disableAGC
       disableAP
       disableHPF
       disableNS
       enableLipSync
       enableTalkWhileMuted
       forceJVB121Ratio
       hiddenDomain
       ignoreStartMuted
       nick
       startBitrate
       */
  
  
      // 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',
  
      externalAppURL: ''
  
      // no configuration value should follow this line.
  };
  
  /* eslint-enable no-unused-vars, no-var */
  