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authorQuentin <quentin@deuxfleurs.fr>2020-09-12 10:03:48 +0200
committerQuentin <quentin@deuxfleurs.fr>2020-09-12 10:03:48 +0200
commitc4a6cf1534b864d3941c839d4a4dca7e505bd828 (patch)
tree3e866768cf39ae947def5b205f74fddfb435725d /docker/jitsi/jitsi-front/config.js
parent0550647b9348d1e36431400e0832b4340564c658 (diff)
downloadinfrastructure-c4a6cf1534b864d3941c839d4a4dca7e505bd828.tar.gz
infrastructure-c4a6cf1534b864d3941c839d4a4dca7e505bd828.zip
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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: 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 */
-