| 123456789101112131415161718192021222324252627282930313233343536373839404142434445464748495051525354555657585960616263646566676869707172737475767778798081828384858687888990919293949596979899100101102103104105106107108109110111112113114115116117118119120121122123124125126127128129130131132133134135136137138139140141142143144145146147148149150151152153154155156157158159160161162163164165166167168169170171172173174175176177178179180181182183184185186187188189190191192193194195196197198199200201202203204205206207208209210211212213214215216217218219220221222223224225226227228229230231232233234235236237238239240241242243244245246247248249250251252253254255256257258259260261262263264265266267268269270271272273274275276277278 | 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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