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Diffstat (limited to 'app/jitsi/build/jitsi-meet/config.js')
| -rw-r--r-- | app/jitsi/build/jitsi-meet/config.js | 517 |
1 files changed, 0 insertions, 517 deletions
diff --git a/app/jitsi/build/jitsi-meet/config.js b/app/jitsi/build/jitsi-meet/config.js deleted file mode 100644 index 18ff319..0000000 --- a/app/jitsi/build/jitsi-meet/config.js +++ /dev/null @@ -1,517 +0,0 @@ -/* 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: 480, - - // 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: 480, - 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: 'fr', - - // 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: 60 - }, - - 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 */ - |