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+/* eslint-disable no-unused-vars, no-var */
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+
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+var config = {
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+ // Configuration
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+ //
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+
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+ // Alternative location for the configuration.
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+ // configLocation: './config.json',
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+
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+ // Custom function which given the URL path should return a room name.
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+ // getroomnode: function (path) { return 'someprefixpossiblybasedonpath'; },
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+
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+
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+ // Connection
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+ //
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+
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+ hosts: {
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+ // XMPP domain.
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+ domain: 'jitsi-meet.example.com',
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+
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+ // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
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+ muc: 'conference.jitsi-meet.example.com'
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+
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+ // When using authentication, domain for guest users.
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+ // anonymousdomain: 'guest.example.com',
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+
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+ // Domain for authenticated users. Defaults to <domain>.
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+ // authdomain: 'jitsi-meet.example.com',
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+
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+ // Jirecon recording component domain.
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+ // jirecon: 'jirecon.jitsi-meet.example.com',
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+
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+ // Call control component (Jigasi).
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+ // call_control: 'callcontrol.jitsi-meet.example.com',
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+
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+ // Focus component domain. Defaults to focus.<domain>.
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+ // focus: 'focus.jitsi-meet.example.com',
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+ },
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+
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+ // BOSH URL. FIXME: use XEP-0156 to discover it.
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+ bosh: '/http-bind',
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+
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+ // The name of client node advertised in XEP-0115 'c' stanza
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+ clientNode: 'http://jitsi.org/jitsimeet',
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+
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+ // The real JID of focus participant - can be overridden here
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+ // focusUserJid: 'focus@auth.jitsi-meet.example.com',
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+
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+
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+ // Testing / experimental features.
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+ //
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+
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+ testing: {
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+ // Enables experimental simulcast support on Firefox.
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+ enableFirefoxSimulcast: false,
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+
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+ // P2P test mode disables automatic switching to P2P when there are 2
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+ // participants in the conference.
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+ p2pTestMode: false
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+ },
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+
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+ // Disables ICE/UDP by filtering out local and remote UDP candidates in
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+ // signalling.
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+ // webrtcIceUdpDisable: false,
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+
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+ // Disables ICE/TCP by filtering out local and remote TCP candidates in
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+ // signalling.
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+ // webrtcIceTcpDisable: false,
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+
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+
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+ // Media
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+ //
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+
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+ // Audio
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+
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+ // Disable measuring of audio levels.
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+ // disableAudioLevels: false,
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+
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+ // Start the conference in audio only mode (no video is being received nor
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+ // sent).
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+ // startAudioOnly: false,
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+
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+ // Every participant after the Nth will start audio muted.
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+ // startAudioMuted: 10,
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+
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+ // Start calls with audio muted. Unlike the option above, this one is only
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+ // applied locally. FIXME: having these 2 options is confusing.
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+ // startWithAudioMuted: false,
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+
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+ // Video
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+
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+ // Sets the preferred resolution (height) for local video. Defaults to 720.
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+ // resolution: 720,
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+
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+ // w3c spec-compliant video constraints to use for video capture. Currently
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+ // used by browsers that return true from lib-jitsi-meet's
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+ // util#browser#usesNewGumFlow. The constraints are independency from
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+ // this config's resolution value. Defaults to requesting an ideal aspect
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+ // ratio of 16:9 with an ideal resolution of 1080p.
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+ // constraints: {
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+ // video: {
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+ // aspectRatio: 16 / 9,
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+ // height: {
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+ // ideal: 1080,
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+ // max: 1080,
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+ // min: 240
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+ // }
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+ // }
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+ // },
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+
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+ // Enable / disable simulcast support.
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+ // disableSimulcast: false,
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+
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+ // Suspend sending video if bandwidth estimation is too low. This may cause
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+ // problems with audio playback. Disabled until these are fixed.
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+ disableSuspendVideo: true,
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+
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+ // Every participant after the Nth will start video muted.
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+ // startVideoMuted: 10,
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+
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+ // Start calls with video muted. Unlike the option above, this one is only
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+ // applied locally. FIXME: having these 2 options is confusing.
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+ // startWithVideoMuted: false,
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+
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+ // If set to true, prefer to use the H.264 video codec (if supported).
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+ // Note that it's not recommended to do this because simulcast is not
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+ // supported when using H.264. For 1-to-1 calls this setting is enabled by
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+ // default and can be toggled in the p2p section.
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+ // preferH264: true,
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+
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+ // If set to true, disable H.264 video codec by stripping it out of the
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+ // SDP.
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+ // disableH264: false,
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+
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+ // Desktop sharing
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+
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+ // Enable / disable desktop sharing
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+ // disableDesktopSharing: false,
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+
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+ // The ID of the jidesha extension for Chrome.
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+ desktopSharingChromeExtId: null,
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+
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+ // Whether desktop sharing should be disabled on Chrome.
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+ desktopSharingChromeDisabled: true,
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+
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+ // The media sources to use when using screen sharing with the Chrome
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+ // extension.
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+ desktopSharingChromeSources: [ 'screen', 'window', 'tab' ],
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+
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+ // Required version of Chrome extension
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+ desktopSharingChromeMinExtVersion: '0.1',
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+
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+ // The ID of the jidesha extension for Firefox. If null, we assume that no
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+ // extension is required.
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+ desktopSharingFirefoxExtId: null,
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+
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+ // Whether desktop sharing should be disabled on Firefox.
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+ desktopSharingFirefoxDisabled: false,
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+
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+ // The maximum version of Firefox which requires a jidesha extension.
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+ // Example: if set to 41, we will require the extension for Firefox versions
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+ // up to and including 41. On Firefox 42 and higher, we will run without the
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+ // extension.
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+ // If set to -1, an extension will be required for all versions of Firefox.
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+ desktopSharingFirefoxMaxVersionExtRequired: 51,
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+
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+ // The URL to the Firefox extension for desktop sharing.
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+ desktopSharingFirefoxExtensionURL: null,
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+
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+ // Optional desktop sharing frame rate options. Default value: min:5, max:5.
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+ // desktopSharingFrameRate: {
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+ // min: 5,
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+ // max: 5
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+ // },
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+
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+ // Try to start calls with screen-sharing instead of camera video.
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+ // startScreenSharing: false,
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+
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+ // Recording
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+
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+ // Whether to enable recording or not.
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+ // enableRecording: false,
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+
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+ // Type for recording: one of jibri or jirecon.
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+ // recordingType: 'jibri',
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+
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+ // Misc
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+
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+ // Default value for the channel "last N" attribute. -1 for unlimited.
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+ channelLastN: -1,
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+
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+ // Disables or enables RTX (RFC 4588) (defaults to false).
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+ // disableRtx: false,
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+
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+ // Use XEP-0215 to fetch STUN and TURN servers.
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+ // useStunTurn: true,
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+
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+ // Enable IPv6 support.
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+ // useIPv6: true,
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+
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+ // Enables / disables a data communication channel with the Videobridge.
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+ // Values can be 'datachannel', 'websocket', true (treat it as
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+ // 'datachannel'), undefined (treat it as 'datachannel') and false (don't
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+ // open any channel).
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+ // openBridgeChannel: true,
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+
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+
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+ // UI
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+ //
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+
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+ // Use display name as XMPP nickname.
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+ // useNicks: false,
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+
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+ // Require users to always specify a display name.
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+ // requireDisplayName: true,
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+
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+ // Whether to use a welcome page or not. In case it's false a random room
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+ // will be joined when no room is specified.
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+ enableWelcomePage: true,
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+
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+ // Enabling the close page will ignore the welcome page redirection when
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+ // a call is hangup.
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+ // enableClosePage: false,
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+
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+ // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
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+ // disable1On1Mode: false,
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+
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+ // The minimum value a video's height (or width, whichever is smaller) needs
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+ // to be in order to be considered high-definition.
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+ minHDHeight: 540,
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+
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+ // Default language for the user interface.
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+ // defaultLanguage: 'en',
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+
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+ // If true all users without a token will be considered guests and all users
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+ // with token will be considered non-guests. Only guests will be allowed to
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+ // edit their profile.
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+ enableUserRolesBasedOnToken: false,
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+
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+ // Message to show the users. Example: 'The service will be down for
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+ // maintenance at 01:00 AM GMT,
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+ // noticeMessage: '',
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+
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+
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+ // Stats
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+ //
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+
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+ // Whether to enable stats collection or not in the TraceablePeerConnection.
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+ // This can be useful for debugging purposes (post-processing/analysis of
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+ // the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
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+ // estimation tests.
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+ // gatherStats: false,
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+
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+ // To enable sending statistics to callstats.io you must provide the
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+ // Application ID and Secret.
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+ // callStatsID: '',
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+ // callStatsSecret: '',
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+
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+ // enables callstatsUsername to be reported as statsId and used
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+ // by callstats as repoted remote id
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+ // enableStatsID: false
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+
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+ // enables sending participants display name to callstats
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+ // enableDisplayNameInStats: false
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+
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+
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+ // Privacy
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+ //
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+
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+ // If third party requests are disabled, no other server will be contacted.
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+ // This means avatars will be locally generated and callstats integration
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+ // will not function.
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+ // disableThirdPartyRequests: false,
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+
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+
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+ // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
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+ //
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+
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+ p2p: {
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+ // Enables peer to peer mode. When enabled the system will try to
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+ // establish a direct connection when there are exactly 2 participants
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+ // in the room. If that succeeds the conference will stop sending data
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+ // through the JVB and use the peer to peer connection instead. When a
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+ // 3rd participant joins the conference will be moved back to the JVB
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+ // connection.
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+ enabled: true,
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+
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+ // Use XEP-0215 to fetch STUN and TURN servers.
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+ // useStunTurn: true,
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+
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+ // The STUN servers that will be used in the peer to peer connections
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+ stunServers: [
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+ { urls: 'stun:stun.l.google.com:19302' },
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+ { urls: 'stun:stun1.l.google.com:19302' },
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+ { urls: 'stun:stun2.l.google.com:19302' }
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+ ],
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+
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+ // Sets the ICE transport policy for the p2p connection. At the time
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+ // of this writing the list of possible values are 'all' and 'relay',
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+ // but that is subject to change in the future. The enum is defined in
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+ // the WebRTC standard:
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+ // https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
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+ // If not set, the effective value is 'all'.
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+ // iceTransportPolicy: 'all',
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+
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+ // If set to true, it will prefer to use H.264 for P2P calls (if H.264
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+ // is supported).
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+ preferH264: true
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+
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+ // If set to true, disable H.264 video codec by stripping it out of the
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+ // SDP.
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+ // disableH264: false,
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+
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+ // How long we're going to wait, before going back to P2P after the 3rd
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+ // participant has left the conference (to filter out page reload).
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+ // backToP2PDelay: 5
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+ },
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+
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+ // A list of scripts to load as lib-jitsi-meet "analytics handlers".
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+ // analyticsScriptUrls: [
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+ // "libs/analytics-ga.js", // google-analytics
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+ // "https://example.com/my-custom-analytics.js"
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+ // ],
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+
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+ // The Google Analytics Tracking ID
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+ // googleAnalyticsTrackingId = 'your-tracking-id-here-UA-123456-1',
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+
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+ // Information about the jitsi-meet instance we are connecting to, including
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+ // the user region as seen by the server.
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+ deploymentInfo: {
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+ // shard: "shard1",
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+ // region: "europe",
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+ // userRegion: "asia"
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+ }
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+
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+
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+ // List of undocumented settings used in jitsi-meet
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+ /**
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+ alwaysVisibleToolbar
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+ autoEnableDesktopSharing
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+ autoRecord
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+ autoRecordToken
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+ debug
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+ debugAudioLevels
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+ deploymentInfo
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+ dialInConfCodeUrl
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+ dialInNumbersUrl
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+ dialOutAuthUrl
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+ dialOutCodesUrl
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+ disableRemoteControl
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+ displayJids
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+ enableLocalVideoFlip
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+ etherpad_base
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+ externalConnectUrl
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+ firefox_fake_device
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+ iAmRecorder
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+ iAmSipGateway
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+ peopleSearchQueryTypes
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+ peopleSearchUrl
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+ requireDisplayName
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+ tokenAuthUrl
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+ */
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+
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+ // List of undocumented settings used in lib-jitsi-meet
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+ /**
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+ _peerConnStatusOutOfLastNTimeout
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+ _peerConnStatusRtcMuteTimeout
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+ abTesting
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+ avgRtpStatsN
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|
|
|
+ callStatsConfIDNamespace
|
|
|
|
+ callStatsCustomScriptUrl
|
|
|
|
+ desktopSharingSources
|
|
|
|
+ disableAEC
|
|
|
|
+ disableAGC
|
|
|
|
+ disableAP
|
|
|
|
+ disableHPF
|
|
|
|
+ disableNS
|
|
|
|
+ enableLipSync
|
|
|
|
+ enableTalkWhileMuted
|
|
|
|
+ forceJVB121Ratio
|
|
|
|
+ hiddenDomain
|
|
|
|
+ ignoreStartMuted
|
|
|
|
+ nick
|
|
|
|
+ startBitrate
|
|
|
|
+ */
|
|
|
|
+};
|
|
|
|
+
|
|
|
|
+/* eslint-enable no-unused-vars, no-var */
|