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							- var config = { // eslint-disable-line no-unused-vars
 -     // Configuration
 -     //
 - 
 -     // Alternative location for the configuration.
 -     //configLocation: './config.json',
 - 
 -     // Custom function which given the URL path should return a room name.
 -     //getroomnode: function (path) { return 'someprefixpossiblybasedonpath'; },
 - 
 - 
 -     // Connection
 -     //
 - 
 -     hosts: {
 -         // XMPP domain.
 -         domain: 'jitsi-meet.example.com',
 - 
 -         // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
 -         muc: 'conference.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',
 -     },
 - 
 -     // BOSH URL. FIXME: use XEP-0156 to discover it.
 -     bosh: '//jitsi-meet.example.com/http-bind',
 - 
 -     // 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: {
 -         // Enables experimental simulcast support on Firefox.
 -         enableFirefoxSimulcast: false,
 -         // P2P test mode disables automatic switching to P2P when there are 2
 -         // participants in the conference.
 -         p2pTestMode: 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,
 - 
 -     // 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,
 - 
 -     // Video
 - 
 -     // Sets the preferred resolution (height) for local video. Defaults to 720.
 -     //resolution: 720,
 - 
 -     // Enable / disable simulcast support.
 -     //disableSimulcast: false,
 - 
 -     // Suspend sending video if bandwidth estimation is too low. This may cause
 -     // problems with audio playback. Disabled until these are fixed.
 -     disableSuspendVideo: true,
 - 
 -     // 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,
 - 
 -     // Desktop sharing
 - 
 -     // Enable / disable desktop sharing
 -     //disableDesktopSharing: false,
 - 
 -     // The ID of the jidesha extension for Chrome.
 -     desktopSharingChromeExtId: null,
 - 
 -     // Whether desktop sharing should be disabled on Chrome.
 -     desktopSharingChromeDisabled: true,
 - 
 -     // 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',
 - 
 -     // The ID of the jidesha extension for Firefox. If null, we assume that no
 -     // extension is required.
 -     desktopSharingFirefoxExtId: null,
 - 
 -     // Whether desktop sharing should be disabled on Firefox.
 -     desktopSharingFirefoxDisabled: false,
 - 
 -     // The maximum version of Firefox which requires a jidesha extension.
 -     // Example: if set to 41, we will require the extension for Firefox versions
 -     // up to and including 41. On Firefox 42 and higher, we will run without the
 -     // extension.
 -     // If set to -1, an extension will be required for all versions of Firefox.
 -     desktopSharingFirefoxMaxVersionExtRequired: 51,
 - 
 -     // The URL to the Firefox extension for desktop sharing.
 -     desktopSharingFirefoxExtensionURL: null,
 - 
 -     // Try to start calls with screen-sharing instead of camera video.
 -     //startScreenSharing: false,
 - 
 -     // Recording
 - 
 -     // Whether to enable recording or not.
 -     //enableRecording: false,
 - 
 -     // Type for recording: one of jibri or jirecon.
 -     //recordingType: 'jibri',
 - 
 -     // 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,
 - 
 -     // 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,
 - 
 -     // The minimum value a video's height (or width, whichever is smaller) needs
 -     // to be in order to be considered high-definition.
 -     minHDHeight: 540,
 - 
 -     // 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,
 - 
 -     // Message to show the users. Example: 'The service will be down for
 -     // maintenance at 01:00 AM GMT,
 -     //noticeMessage: '',
 - 
 - 
 -     // Stats
 -     //
 - 
 -     // Whether to enable stats collection or not.
 -     //disableStats: false,
 - 
 -     // To enable sending statistics to callstats.io you must provide the
 -     // Application ID and Secret.
 -     //callStatsID: '',
 -     //callStatsSecret: '',
 - 
 - 
 -     // 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:stun.l.google.com:19302" },
 -             { urls: "stun:stun1.l.google.com:19302" },
 -             { urls: "stun:stun2.l.google.com:19302" }
 -         ],
 - 
 -         // If set to true, it will prefer to use H.264 for P2P calls (if H.264
 -         // is supported).
 -         preferH264: true
 - 
 -         // 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
 -     },
 - 
 - 
 -     // 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"
 -     }
 - };
 
 
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