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authorAlex Auvolat <alex@adnab.me>2022-08-24 15:42:47 +0200
committerAlex Auvolat <alex@adnab.me>2022-08-24 15:42:47 +0200
commit2e8923b383eb06c53261eee8e5c442b857fb67e4 (patch)
tree0ad148f75f7b54dfed2dbac8f43f6df9badc502a /cluster/prod/app/jitsi/config/config.js
parent9848f3090f77363a2fda0f9fa673ebcf1fb8228c (diff)
downloadnixcfg-2e8923b383eb06c53261eee8e5c442b857fb67e4.tar.gz
nixcfg-2e8923b383eb06c53261eee8e5c442b857fb67e4.zip
Move app files into cluster subdirectories; add prod garage
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+/* eslint-disable no-unused-vars, no-var */
+
+var config = {
+ // Connection
+ //
+
+ hosts: {
+ // XMPP domain.
+ domain: 'jitsi',
+
+ // When using authentication, domain for guest users.
+ // anonymousdomain: 'guest.example.com',
+
+ // Domain for authenticated users. Defaults to <domain>.
+ // authdomain: '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'
+ },
+
+ // 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
+ // Do not change username - FIXME: Make focus username configurable
+ // https://github.com/jitsi/jitsi-meet/issues/7376
+ // focusUserJid: 'focus@auth.jitsi-meet.example.com',
+
+
+ // Testing / experimental features.
+ //
+
+ testing: {
+ // Disables the End to End Encryption feature. Useful for debugging
+ // issues related to insertable streams.
+ // disableE2EE: 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
+
+ // Enable / disable 500 Kbps bitrate cap on desktop tracks. When enabled,
+ // simulcast is turned off for the desktop share. If presenter is turned
+ // on while screensharing is in progress, the max bitrate is automatically
+ // adjusted to 2.5 Mbps. This takes a value between 0 and 1 which determines
+ // the probability for this to be enabled.
+ // capScreenshareBitrate: 1 // 0 to disable
+
+ // Enable callstats only for a percentage of users.
+ // This takes a value between 0 and 100 which determines the probability for
+ // the callstats to be enabled.
+ // callStatsThreshold: 5 // enable callstats for 5% of the users.
+ },
+
+ // 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 show a "Save Logs" link in the GSM popover that can be
+ // used to collect debug information (XMPP IQs, SDP offer/answer cycles)
+ // about the call.
+ // enableSaveLogs: false,
+
+ // 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: false,
+
+ // 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: 5,
+
+ // 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
+
+ // Sets the preferred target bitrate for the Opus audio codec by setting its
+ // 'maxaveragebitrate' parameter. Currently not available in p2p mode.
+ // Valid values are in the range 6000 to 510000
+ // opusMaxAverageBitrate: 20000,
+
+ // Enables support for opus-red (redundancy for Opus).
+ // enableOpusRed: false
+
+ // Video
+
+ // Sets the preferred resolution (height) for local video. Defaults to 720.
+ // resolution: 720,
+
+ // How many participants while in the tile view mode, before the receiving video quality is reduced from HD to SD.
+ // Use -1 to disable.
+ // maxFullResolutionParticipants: 2,
+
+ // 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 independent from
+ // this config's resolution value. Defaults to requesting an ideal
+ // resolution of 720p.
+ // constraints: {
+ // video: {
+ // 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: 5,
+
+ // 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.
+ // This option has been deprecated, use preferredCodec under videoQuality section instead.
+ // preferH264: true,
+
+ // If set to true, disable H.264 video codec by stripping it out of the
+ // SDP.
+ // disableH264: false,
+
+ // Desktop sharing
+
+ // 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,
+
+ // Provides a way to use different "last N" values based on the number of participants in the conference.
+ // The keys in an Object represent number of participants and the values are "last N" to be used when number of
+ // participants gets to or above the number.
+ //
+ // For the given example mapping, "last N" will be set to 20 as long as there are at least 5, but less than
+ // 29 participants in the call and it will be lowered to 15 when the 30th participant joins. The 'channelLastN'
+ // will be used as default until the first threshold is reached.
+ //
+ // lastNLimits: {
+ // 5: 20,
+ // 30: 15,
+ // 50: 10,
+ // 70: 5,
+ // 90: 2
+ // },
+
+ // Specify the settings for video quality optimizations on the client.
+ // videoQuality: {
+ // // Provides a way to prevent a video codec from being negotiated on the JVB connection. The codec specified
+ // // here will be removed from the list of codecs present in the SDP answer generated by the client. If the
+ // // same codec is specified for both the disabled and preferred option, the disable settings will prevail.
+ // // Note that 'VP8' cannot be disabled since it's a mandatory codec, the setting will be ignored in this case.
+ // disabledCodec: 'H264',
+ //
+ // // Provides a way to set a preferred video codec for the JVB connection. If 'H264' is specified here,
+ // // simulcast will be automatically disabled since JVB doesn't support H264 simulcast yet. This will only
+ // // rearrange the the preference order of the codecs in the SDP answer generated by the browser only if the
+ // // preferred codec specified here is present. Please ensure that the JVB offers the specified codec for this
+ // // to take effect.
+ // preferredCodec: 'VP8',
+ //
+ // // Provides a way to configure the maximum bitrates that will be enforced on the simulcast streams for
+ // // video tracks. The keys in the object represent the type of the stream (LD, SD or HD) and the values
+ // // are the max.bitrates to be set on that particular type of stream. The actual send may vary based on
+ // // the available bandwidth calculated by the browser, but it will be capped by the values specified here.
+ // // This is currently not implemented on app based clients on mobile.
+ // maxBitratesVideo: {
+ // low: 200000,
+ // standard: 500000,
+ // high: 1500000
+ // },
+ //
+ // // The options can be used to override default thresholds of video thumbnail heights corresponding to
+ // // the video quality levels used in the application. At the time of this writing the allowed levels are:
+ // // 'low' - for the low quality level (180p at the time of this writing)
+ // // 'standard' - for the medium quality level (360p)
+ // // 'high' - for the high quality level (720p)
+ // // The keys should be positive numbers which represent the minimal thumbnail height for the quality level.
+ // //
+ // // With the default config value below the application will use 'low' quality until the thumbnails are
+ // // at least 360 pixels tall. If the thumbnail height reaches 720 pixels then the application will switch to
+ // // the high quality.
+ // minHeightForQualityLvl: {
+ // 360: 'standard',
+ // 720: 'high'
+ // },
+ //
+ // // Provides a way to resize the desktop track to 720p (if it is greater than 720p) before creating a canvas
+ // // for the presenter mode (camera picture-in-picture mode with screenshare).
+ // resizeDesktopForPresenter: false
+ // },
+
+ // // Options for the recording limit notification.
+ // recordingLimit: {
+ //
+ // // The recording limit in minutes. Note: This number appears in the notification text
+ // // but doesn't enforce the actual recording time limit. This should be configured in
+ // // jibri!
+ // limit: 60,
+ //
+ // // The name of the app with unlimited recordings.
+ // appName: 'Unlimited recordings APP',
+ //
+ // // The URL of the app with unlimited recordings.
+ // appURL: 'https://unlimited.recordings.app.com/'
+ // },
+
+ // Disables or enables RTX (RFC 4588) (defaults to false).
+ // disableRtx: false,
+
+ // Disables or enables TCC support in this client (default: enabled).
+ // enableTcc: true,
+
+ // Disables or enables REMB support in this client (default: enabled).
+ // enableRemb: true,
+
+ // Enables ICE restart logic in LJM and displays the page reload overlay on
+ // ICE failure. Current disabled by default because it's causing issues with
+ // signaling when Octo is enabled. Also when we do an "ICE restart"(which is
+ // not a real ICE restart), the client maintains the TCC sequence number
+ // counter, but the bridge resets it. The bridge sends media packets with
+ // TCC sequence numbers starting from 0.
+ // enableIceRestart: false,
+
+ // Use TURN/UDP servers for the jitsi-videobridge connection (by default
+ // we filter out TURN/UDP because it is usually not needed since the
+ // bridge itself is reachable via UDP)
+ // useTurnUdp: false
+
+ // UI
+ //
+
+ // Disables responsive tiles.
+ // disableResponsiveTiles: false,
+
+ // Hides lobby button
+ // hideLobbyButton: 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,
+
+ // Disable app shortcuts that are registered upon joining a conference
+ // disableShortcuts: false,
+
+ // Disable initial browser getUserMedia requests.
+ // This is useful for scenarios where users might want to start a conference for screensharing only
+ // disableInitialGUM: false,
+
+ // 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',
+
+ // Disables profile and the edit of all fields from the profile settings (display name and email)
+ // disableProfile: false,
+
+ // Whether or not some features are checked based on token.
+ // enableFeaturesBasedOnToken: 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,
+
+ // When 'true', it shows an intermediate page before joining, where the user can configure their devices.
+ prejoinPageEnabled: true,
+
+ // If etherpad integration is enabled, setting this to true will
+ // automatically open the etherpad when a participant joins. This
+ // does not affect the mobile app since opening an etherpad
+ // obscures the conference controls -- it's better to let users
+ // choose to open the pad on their own in that case.
+ // openSharedDocumentOnJoin: false,
+
+ // If true, shows the unsafe room name warning label when a room name is
+ // deemed unsafe (due to the simplicity in the name) and a password is not
+ // set or the lobby is not enabled.
+ // enableInsecureRoomNameWarning: false,
+
+ // Whether to automatically copy invitation URL after creating a room.
+ // Document should be focused for this option to work
+ // enableAutomaticUrlCopy: false,
+
+ // Base URL for a Gravatar-compatible service. Defaults to libravatar.
+ // gravatarBaseURL: 'https://seccdn.libravatar.org/avatar/';
+
+ // 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 names to callstats
+ // enableDisplayNameInStats: false,
+
+ // Enables sending participants' emails (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,
+
+ // The STUN servers that will be used in the peer to peer connections
+ stunServers: [
+
+ // { urls: 'stun:jitsi-meet.example.com:3478' },
+ { urls: 'stun:meet-jit-si-turnrelay.jitsi.net:443' }
+ ]
+
+ // 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). This setting is deprecated, use preferredCodec instead.
+ // preferH264: true
+
+ // Provides a way to set the video codec preference on the p2p connection. Acceptable
+ // codec values are 'VP8', 'VP9' and 'H264'.
+ // preferredCodec: 'H264',
+
+ // If set to true, disable H.264 video codec by stripping it out of the
+ // SDP. This setting is deprecated, use disabledCodec instead.
+ // disableH264: false,
+
+ // Provides a way to prevent a video codec from being negotiated on the p2p connection.
+ // disabledCodec: '',
+
+ // 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'
+
+ // Matomo configuration:
+ // matomoEndpoint: 'https://your-matomo-endpoint/',
+ // matomoSiteID: '42',
+
+ // The Amplitude APP Key:
+ // amplitudeAPPKey: '<APP_KEY>'
+
+ // Configuration for the rtcstats server:
+ // By enabling rtcstats server every time a conference is joined the rtcstats
+ // module connects to the provided rtcstatsEndpoint and sends statistics regarding
+ // PeerConnection states along with getStats metrics polled at the specified
+ // interval.
+ // rtcstatsEnabled: true,
+
+ // In order to enable rtcstats one needs to provide a endpoint url.
+ // rtcstatsEndpoint: wss://rtcstats-server-pilot.jitsi.net/,
+
+ // The interval at which rtcstats will poll getStats, defaults to 1000ms.
+ // If the value is set to 0 getStats won't be polled and the rtcstats client
+ // will only send data related to RTCPeerConnection events.
+ // rtcstatsPolIInterval: 1000
+
+ // 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"
+ // ],
+ },
+
+ // Logs that should go be passed through the 'log' event if a handler is defined for it
+ // apiLogLevels: ['warn', 'log', 'error', 'info', 'debug'],
+
+ // 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"
+ },
+
+ // Decides whether the start/stop recording audio notifications should play on record.
+ // disableRecordAudioNotification: false,
+
+ // 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,
+
+ // Mainly privacy related settings
+
+ // Disables all invite functions from the app (share, invite, dial out...etc)
+ // disableInviteFunctions: true,
+
+ // Disables storing the room name to the recents list
+ // doNotStoreRoom: true,
+
+ // 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'
+ // },
+
+ // Options related to the remote participant menu.
+ // remoteVideoMenu: {
+ // // If set to true the 'Kick out' button will be disabled.
+ // disableKick: true
+ // },
+
+ // If set to true all muting operations of remote participants will be disabled.
+ // disableRemoteMute: true,
+
+ // Enables support for lip-sync for this client (if the browser supports it).
+ // enableLipSync: false
+
+ /**
+ External API url used to receive branding specific information.
+ If there is no url set or there are missing fields, the defaults are applied.
+ None of the fields are mandatory and the response must have the shape:
+ {
+ // The hex value for the colour used as background
+ backgroundColor: '#fff',
+ // The url for the image used as background
+ backgroundImageUrl: 'https://example.com/background-img.png',
+ // The anchor url used when clicking the logo image
+ logoClickUrl: 'https://example-company.org',
+ // The url used for the image used as logo
+ logoImageUrl: 'https://example.com/logo-img.png'
+ }
+ */
+ // dynamicBrandingUrl: '',
+
+ // The URL of the moderated rooms microservice, if available. If it
+ // is present, a link to the service will be rendered on the welcome page,
+ // otherwise the app doesn't render it.
+ // moderatedRoomServiceUrl: 'https://moderated.jitsi-meet.example.com',
+
+ // If true, tile view will not be enabled automatically when the participants count threshold is reached.
+ // disableTileView: true,
+
+ // Hides the conference subject
+ // hideConferenceSubject: true
+
+ // Hides the conference timer.
+ // hideConferenceTimer: true,
+
+ // Hides the participants stats
+ // hideParticipantsStats: true
+
+ // Sets the conference subject
+ // subject: 'Conference Subject',
+
+ // List of undocumented settings used in jitsi-meet
+ /**
+ _immediateReloadThreshold
+ debug
+ debugAudioLevels
+ deploymentInfo
+ dialInConfCodeUrl
+ dialInNumbersUrl
+ dialOutAuthUrl
+ dialOutCodesUrl
+ disableRemoteControl
+ displayJids
+ etherpad_base
+ externalConnectUrl
+ firefox_fake_device
+ googleApiApplicationClientID
+ iAmRecorder
+ iAmSipGateway
+ microsoftApiApplicationClientID
+ peopleSearchQueryTypes
+ peopleSearchUrl
+ requireDisplayName
+ tokenAuthUrl
+ */
+
+ /**
+ * This property can be used to alter the generated meeting invite links (in combination with a branding domain
+ * which is retrieved internally by jitsi meet) (e.g. https://meet.jit.si/someMeeting
+ * can become https://brandedDomain/roomAlias)
+ */
+ // brandingRoomAlias: null,
+
+ // List of undocumented settings used in lib-jitsi-meet
+ /**
+ _peerConnStatusOutOfLastNTimeout
+ _peerConnStatusRtcMuteTimeout
+ abTesting
+ avgRtpStatsN
+ callStatsConfIDNamespace
+ callStatsCustomScriptUrl
+ desktopSharingSources
+ disableAEC
+ disableAGC
+ disableAP
+ disableHPF
+ disableNS
+ enableTalkWhileMuted
+ forceJVB121Ratio
+ forceTurnRelay
+ hiddenDomain
+ ignoreStartMuted
+ websocketKeepAlive
+ websocketKeepAliveUrl
+ */
+
+ /**
+ Use this array to configure which notifications will be shown to the user
+ The items correspond to the title or description key of that notification
+ Some of these notifications also depend on some other internal logic to be displayed or not,
+ so adding them here will not ensure they will always be displayed
+
+ A falsy value for this prop will result in having all notifications enabled (e.g null, undefined, false)
+ */
+ // notifications: [
+ // 'connection.CONNFAIL', // shown when the connection fails,
+ // 'dialog.cameraNotSendingData', // shown when there's no feed from user's camera
+ // 'dialog.kickTitle', // shown when user has been kicked
+ // 'dialog.liveStreaming', // livestreaming notifications (pending, on, off, limits)
+ // 'dialog.lockTitle', // shown when setting conference password fails
+ // 'dialog.maxUsersLimitReached', // shown when maximmum users limit has been reached
+ // 'dialog.micNotSendingData', // shown when user's mic is not sending any audio
+ // 'dialog.passwordNotSupportedTitle', // shown when setting conference password fails due to password format
+ // 'dialog.recording', // recording notifications (pending, on, off, limits)
+ // 'dialog.remoteControlTitle', // remote control notifications (allowed, denied, start, stop, error)
+ // 'dialog.reservationError',
+ // 'dialog.serviceUnavailable', // shown when server is not reachable
+ // 'dialog.sessTerminated', // shown when there is a failed conference session
+ // 'dialog.tokenAuthFailed', // show when an invalid jwt is used
+ // 'dialog.transcribing', // transcribing notifications (pending, off)
+ // 'dialOut.statusMessage', // shown when dial out status is updated.
+ // 'liveStreaming.busy', // shown when livestreaming service is busy
+ // 'liveStreaming.failedToStart', // shown when livestreaming fails to start
+ // 'liveStreaming.unavailableTitle', // shown when livestreaming service is not reachable
+ // 'lobby.joinRejectedMessage', // shown when while in a lobby, user's request to join is rejected
+ // 'lobby.notificationTitle', // shown when lobby is toggled and when join requests are allowed / denied
+ // 'localRecording.localRecording', // shown when a local recording is started
+ // 'notify.disconnected', // shown when a participant has left
+ // 'notify.grantedTo', // shown when moderator rights were granted to a participant
+ // 'notify.invitedOneMember', // shown when 1 participant has been invited
+ // 'notify.invitedThreePlusMembers', // shown when 3+ participants have been invited
+ // 'notify.invitedTwoMembers', // shown when 2 participants have been invited
+ // 'notify.kickParticipant', // shown when a participant is kicked
+ // 'notify.mutedRemotelyTitle', // shown when user is muted by a remote party
+ // 'notify.mutedTitle', // shown when user has been muted upon joining,
+ // 'notify.newDeviceAudioTitle', // prompts the user to use a newly detected audio device
+ // 'notify.newDeviceCameraTitle', // prompts the user to use a newly detected camera
+ // 'notify.passwordRemovedRemotely', // shown when a password has been removed remotely
+ // 'notify.passwordSetRemotely', // shown when a password has been set remotely
+ // 'notify.raisedHand', // shown when a partcipant used raise hand,
+ // 'notify.startSilentTitle', // shown when user joined with no audio
+ // 'prejoin.errorDialOut',
+ // 'prejoin.errorDialOutDisconnected',
+ // 'prejoin.errorDialOutFailed',
+ // 'prejoin.errorDialOutStatus',
+ // 'prejoin.errorStatusCode',
+ // 'prejoin.errorValidation',
+ // 'recording.busy', // shown when recording service is busy
+ // 'recording.failedToStart', // shown when recording fails to start
+ // 'recording.unavailableTitle', // shown when recording service is not reachable
+ // 'toolbar.noAudioSignalTitle', // shown when a broken mic is detected
+ // 'toolbar.noisyAudioInputTitle', // shown when noise is detected for the current microphone
+ // 'toolbar.talkWhileMutedPopup', // shown when user tries to speak while muted
+ // 'transcribing.failedToStart' // shown when transcribing fails to start
+ // ]
+
+ // Allow all above example options to include a trailing comma and
+ // prevent fear when commenting out the last value.
+ makeJsonParserHappy: 'even if last key had a trailing comma'
+
+ // no configuration value should follow this line.
+};
+
+/* eslint-enable no-unused-vars, no-var */