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+/* eslint-disable no-unused-vars, no-var */
+
+var config = {
+ // Connection
+ //
+
+ hosts: {
+ // XMPP domain.
+ domain: 'jitsi.deuxfleurs.fr',
+
+ // 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.deuxfleurs.fr'
+ },
+
+ // BOSH URL. FIXME: use XEP-0156 to discover it.
+ bosh: '//jitsi.deuxfleurs.fr/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: {
+ // 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
+
+ // 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 independency from
+ // this config's resolution value. Defaults to requesting an ideal aspect
+ // ratio of 16:9 with an ideal resolution of 720.
+ // constraints: {
+ // video: {
+ // aspectRatio: 16 / 9,
+ // 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:443' },
+ { urls: 'stun:stun.l.google.com:19302' },
+ { urls: 'stun:stun1.l.google.com:19302' },
+ { urls: 'stun:stun2.l.google.com:19302' }
+ ],
+
+ // 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"
+ }
+
+ // 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
+
+ // 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'
+ // }
+
+ // 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 */
+
generated by cgit v1.2.3 (git 2.25.1) at 2024-10-18 22:29:20 +0000