594 lines
21 KiB
JavaScript
594 lines
21 KiB
JavaScript
/* eslint-disable no-unused-vars, no-var */
|
|
|
|
var config = {
|
|
// Connection
|
|
//
|
|
|
|
hosts: {
|
|
// XMPP domain.
|
|
domain: 'jitsi-meet.example.com',
|
|
|
|
// 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-meet.example.com'
|
|
},
|
|
|
|
// BOSH URL. FIXME: use XEP-0156 to discover it.
|
|
bosh: '//jitsi-meet.example.com/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: {
|
|
// 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
|
|
},
|
|
|
|
// 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
|
|
|
|
// 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
|
|
// opusMaxAvgBitrate: 20000,
|
|
|
|
// 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 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: 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
|
|
|
|
// 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,
|
|
|
|
// // 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 (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,
|
|
|
|
// 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,
|
|
|
|
// Defines the minimum number of participants to start a call (the default
|
|
// is set in Jicofo and set to 2).
|
|
// minParticipants: 2,
|
|
|
|
// Use the TURN servers discovered via XEP-0215 for the jitsi-videobridge
|
|
// connection
|
|
// useStunTurn: true,
|
|
|
|
// 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
|
|
|
|
// 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
|
|
//
|
|
|
|
// 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,
|
|
|
|
// When 'true', it shows an intermediate page before joining, where the user can configure their devices.
|
|
// prejoinPageEnabled: 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,
|
|
|
|
// 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,
|
|
|
|
// 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: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).
|
|
// 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'
|
|
|
|
// Matomo configuration:
|
|
// matomoEndpoint: 'https://your-matomo-endpoint/',
|
|
// matomoSiteID: '42',
|
|
|
|
// 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"
|
|
},
|
|
|
|
// 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,
|
|
|
|
/**
|
|
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'
|
|
}
|
|
*/
|
|
// brandingDataUrl: '',
|
|
|
|
// 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',
|
|
|
|
// 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
|
|
*/
|
|
|
|
|
|
// 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 */
|