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@@ -1,481 +0,0 @@
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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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- // 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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- // 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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-
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- // BOSH URL. FIXME: use XEP-0156 to discover it.
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- bosh: '//jitsi-meet.example.com/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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- // Enables the test specific features consumed by jitsi-meet-torture
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- // testMode: 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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- // Enabling it (with #params) will disable local audio output of remote
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- // participants and to enable it back a reload is needed.
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- // startSilent: 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 720.
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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: 720,
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- // max: 720,
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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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- // Enable / disable layer suspension. If enabled, endpoints whose HD
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- // layers are not in use will be suspended (no longer sent) until they
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- // are requested again.
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- // enableLayerSuspension: 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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- // 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: false,
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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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- // Whether desktop sharing should be disabled on Firefox.
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- // desktopSharingFirefoxDisabled: false,
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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 file recording or not.
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- // fileRecordingsEnabled: false,
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- // Enable the dropbox integration.
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- // dropbox: {
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- // appKey: '<APP_KEY>' // Specify your app key here.
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- // // A URL to redirect the user to, after authenticating
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- // // by default uses:
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- // // 'https://jitsi-meet.example.com/static/oauth.html'
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- // redirectURI:
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- // 'https://jitsi-meet.example.com/subfolder/static/oauth.html'
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- // },
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- // When integrations like dropbox are enabled only that will be shown,
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- // by enabling fileRecordingsServiceEnabled, we show both the integrations
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- // and the generic recording service (its configuration and storage type
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- // depends on jibri configuration)
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- // fileRecordingsServiceEnabled: false,
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- // Whether to show the possibility to share file recording with other people
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- // (e.g. meeting participants), based on the actual implementation
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- // on the backend.
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- // fileRecordingsServiceSharingEnabled: false,
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-
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- // Whether to enable live streaming or not.
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- // liveStreamingEnabled: false,
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-
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- // Transcription (in interface_config,
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- // subtitles and buttons can be configured)
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- // transcribingEnabled: false,
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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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- // Disables or enables TCC (the default is in Jicofo and set to true)
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- // (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting
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- // affects congestion control, it practically enables send-side bandwidth
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- // estimations.
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- // enableTcc: true,
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-
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- // Disables or enables REMB (the default is in Jicofo and set to false)
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- // (draft-alvestrand-rmcat-remb-03). This setting affects congestion
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- // control, it practically enables recv-side bandwidth estimations. When
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- // both TCC and REMB are enabled, TCC takes precedence. When both are
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- // disabled, then bandwidth estimations are disabled.
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- // enableRemb: false,
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-
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- // Defines the minimum number of participants to start a call (the default
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- // is set in Jicofo and set to 2).
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- // minParticipants: 2,
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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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- // 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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- // Whether or not some features are checked based on token.
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- // enableFeaturesBasedOnToken: false,
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-
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- // Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests.
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- // lockRoomGuestEnabled: false,
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-
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- // When enabled the password used for locking a room is restricted to up to the number of digits specified
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- // roomPasswordNumberOfDigits: 10,
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- // default: roomPasswordNumberOfDigits: 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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- // Enables calendar integration, depends on googleApiApplicationClientID
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- // and microsoftApiApplicationClientID
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- // enableCalendarIntegration: false,
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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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- analytics: {
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- // The Google Analytics Tracking ID:
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- // googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
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-
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- // The Amplitude APP Key:
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- // amplitudeAPPKey: '<APP_KEY>'
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-
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- // Array of script URLs to load as lib-jitsi-meet "analytics handlers".
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- // scriptURLs: [
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- // "libs/analytics-ga.min.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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-
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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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- // Local Recording
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- //
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-
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- // localRecording: {
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- // Enables local recording.
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- // Additionally, 'localrecording' (all lowercase) needs to be added to
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- // TOOLBAR_BUTTONS in interface_config.js for the Local Recording
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- // button to show up on the toolbar.
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- //
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- // enabled: true,
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- //
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-
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- // The recording format, can be one of 'ogg', 'flac' or 'wav'.
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- // format: 'flac'
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- //
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-
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- // }
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-
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- // Options related to end-to-end (participant to participant) ping.
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- // e2eping: {
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- // // The interval in milliseconds at which pings will be sent.
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- // // Defaults to 10000, set to <= 0 to disable.
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- // pingInterval: 10000,
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- //
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- // // The interval in milliseconds at which analytics events
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- // // with the measured RTT will be sent. Defaults to 60000, set
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- // // to <= 0 to disable.
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- // analyticsInterval: 60000,
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- // }
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-
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- // If set, will attempt to use the provided video input device label when
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|
- // 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
|
|
|
-
|
|
|
- // 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
|
|
|
- */
|
|
|
-
|
|
|
-};
|
|
|
-
|
|
|
-/* eslint-enable no-unused-vars, no-var */
|