552 lines
19 KiB
JavaScript
552 lines
19 KiB
JavaScript
/* eslint-disable no-unused-vars, no-var */
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var config = {
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// Connection
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//
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hosts: {
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// XMPP domain.
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domain: 'jitsi-meet.example.com',
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// When using authentication, domain for guest users.
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// anonymousdomain: 'guest.example.com',
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// Domain for authenticated users. Defaults to <domain>.
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// authdomain: 'jitsi-meet.example.com',
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// Jirecon recording component domain.
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// jirecon: 'jirecon.jitsi-meet.example.com',
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// Call control component (Jigasi).
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// call_control: 'callcontrol.jitsi-meet.example.com',
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// Focus component domain. Defaults to focus.<domain>.
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// focus: 'focus.jitsi-meet.example.com',
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// XMPP MUC domain. FIXME: use XEP-0030 to discover it.
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muc: 'conference.jitsi-meet.example.com'
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},
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// BOSH URL. FIXME: use XEP-0156 to discover it.
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bosh: '//jitsi-meet.example.com/http-bind',
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// Websocket URL
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// websocket: 'wss://jitsi-meet.example.com/xmpp-websocket',
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// The name of client node advertised in XEP-0115 'c' stanza
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clientNode: 'http://jitsi.org/jitsimeet',
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// The real JID of focus participant - can be overridden here
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// focusUserJid: 'focus@auth.jitsi-meet.example.com',
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// Testing / experimental features.
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//
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testing: {
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// P2P test mode disables automatic switching to P2P when there are 2
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// participants in the conference.
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p2pTestMode: false
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// Enables the test specific features consumed by jitsi-meet-torture
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// testMode: false
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// Disables the auto-play behavior of *all* newly created video element.
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// This is useful when the client runs on a host with limited resources.
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// noAutoPlayVideo: false
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},
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// Disables ICE/UDP by filtering out local and remote UDP candidates in
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// signalling.
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// webrtcIceUdpDisable: false,
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// Disables ICE/TCP by filtering out local and remote TCP candidates in
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// signalling.
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// webrtcIceTcpDisable: false,
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// Media
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//
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// Audio
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// Disable measuring of audio levels.
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// disableAudioLevels: false,
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// audioLevelsInterval: 200,
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// Enabling this will run the lib-jitsi-meet no audio detection module which
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// will notify the user if the current selected microphone has no audio
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// input and will suggest another valid device if one is present.
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enableNoAudioDetection: true,
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// Enabling this will run the lib-jitsi-meet noise detection module which will
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// notify the user if there is noise, other than voice, coming from the current
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// selected microphone. The purpose it to let the user know that the input could
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// be potentially unpleasant for other meeting participants.
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enableNoisyMicDetection: true,
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// Start the conference in audio only mode (no video is being received nor
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// sent).
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// startAudioOnly: false,
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// Every participant after the Nth will start audio muted.
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// startAudioMuted: 10,
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// Start calls with audio muted. Unlike the option above, this one is only
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// applied locally. FIXME: having these 2 options is confusing.
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// startWithAudioMuted: false,
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// Enabling it (with #params) will disable local audio output of remote
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// participants and to enable it back a reload is needed.
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// startSilent: false
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// Video
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// Sets the preferred resolution (height) for local video. Defaults to 720.
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// resolution: 720,
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// w3c spec-compliant video constraints to use for video capture. Currently
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// used by browsers that return true from lib-jitsi-meet's
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// util#browser#usesNewGumFlow. The constraints are independent from
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// this config's resolution value. Defaults to requesting an ideal
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// resolution of 720p.
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// constraints: {
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// video: {
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// height: {
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// ideal: 720,
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// max: 720,
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// min: 240
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// }
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// }
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// },
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// Enable / disable simulcast support.
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// disableSimulcast: false,
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// Enable / disable layer suspension. If enabled, endpoints whose HD
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// layers are not in use will be suspended (no longer sent) until they
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// are requested again.
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// enableLayerSuspension: false,
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// Every participant after the Nth will start video muted.
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// startVideoMuted: 10,
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// Start calls with video muted. Unlike the option above, this one is only
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// applied locally. FIXME: having these 2 options is confusing.
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// startWithVideoMuted: false,
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// If set to true, prefer to use the H.264 video codec (if supported).
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// Note that it's not recommended to do this because simulcast is not
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// supported when using H.264. For 1-to-1 calls this setting is enabled by
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// default and can be toggled in the p2p section.
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// preferH264: true,
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// If set to true, disable H.264 video codec by stripping it out of the
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// SDP.
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// disableH264: false,
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// Desktop sharing
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// The ID of the jidesha extension for Chrome.
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desktopSharingChromeExtId: null,
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// Whether desktop sharing should be disabled on Chrome.
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// desktopSharingChromeDisabled: false,
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// The media sources to use when using screen sharing with the Chrome
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// extension.
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desktopSharingChromeSources: [ 'screen', 'window', 'tab' ],
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// Required version of Chrome extension
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desktopSharingChromeMinExtVersion: '0.1',
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// Whether desktop sharing should be disabled on Firefox.
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// desktopSharingFirefoxDisabled: false,
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// Optional desktop sharing frame rate options. Default value: min:5, max:5.
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// desktopSharingFrameRate: {
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// min: 5,
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// max: 5
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// },
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// Try to start calls with screen-sharing instead of camera video.
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// startScreenSharing: false,
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// Recording
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// Whether to enable file recording or not.
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// fileRecordingsEnabled: false,
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// Enable the dropbox integration.
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// dropbox: {
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// appKey: '<APP_KEY>' // Specify your app key here.
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// // A URL to redirect the user to, after authenticating
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// // by default uses:
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// // 'https://jitsi-meet.example.com/static/oauth.html'
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// redirectURI:
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// 'https://jitsi-meet.example.com/subfolder/static/oauth.html'
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// },
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// When integrations like dropbox are enabled only that will be shown,
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// by enabling fileRecordingsServiceEnabled, we show both the integrations
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// and the generic recording service (its configuration and storage type
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// depends on jibri configuration)
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// fileRecordingsServiceEnabled: false,
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// Whether to show the possibility to share file recording with other people
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// (e.g. meeting participants), based on the actual implementation
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// on the backend.
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// fileRecordingsServiceSharingEnabled: false,
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// Whether to enable live streaming or not.
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// liveStreamingEnabled: false,
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// Transcription (in interface_config,
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// subtitles and buttons can be configured)
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// transcribingEnabled: false,
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// Enables automatic turning on captions when recording is started
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// autoCaptionOnRecord: false,
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// Misc
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// Default value for the channel "last N" attribute. -1 for unlimited.
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channelLastN: -1,
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// Disables or enables RTX (RFC 4588) (defaults to false).
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// disableRtx: false,
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// Disables or enables TCC (the default is in Jicofo and set to true)
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// (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting
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// affects congestion control, it practically enables send-side bandwidth
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// estimations.
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// enableTcc: true,
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// Disables or enables REMB (the default is in Jicofo and set to false)
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// (draft-alvestrand-rmcat-remb-03). This setting affects congestion
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// control, it practically enables recv-side bandwidth estimations. When
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// both TCC and REMB are enabled, TCC takes precedence. When both are
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// disabled, then bandwidth estimations are disabled.
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// enableRemb: false,
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// Enables ICE restart logic in LJM and displays the page reload overlay on
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// ICE failure. Current disabled by default because it's causing issues with
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// signaling when Octo is enabled. Also when we do an "ICE restart"(which is
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// not a real ICE restart), the client maintains the TCC sequence number
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// counter, but the bridge resets it. The bridge sends media packets with
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// TCC sequence numbers starting from 0.
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// enableIceRestart: false,
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// Defines the minimum number of participants to start a call (the default
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// is set in Jicofo and set to 2).
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// minParticipants: 2,
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// Use XEP-0215 to fetch STUN and TURN servers.
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// useStunTurn: true,
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// Enable IPv6 support.
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// useIPv6: true,
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// Enables / disables a data communication channel with the Videobridge.
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// Values can be 'datachannel', 'websocket', true (treat it as
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// 'datachannel'), undefined (treat it as 'datachannel') and false (don't
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// open any channel).
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// openBridgeChannel: true,
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// UI
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//
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// Use display name as XMPP nickname.
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// useNicks: false,
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// Require users to always specify a display name.
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// requireDisplayName: true,
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// Whether to use a welcome page or not. In case it's false a random room
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// will be joined when no room is specified.
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enableWelcomePage: true,
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// Enabling the close page will ignore the welcome page redirection when
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// a call is hangup.
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// enableClosePage: false,
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// Disable hiding of remote thumbnails when in a 1-on-1 conference call.
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// disable1On1Mode: false,
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// Default language for the user interface.
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// defaultLanguage: 'en',
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// If true all users without a token will be considered guests and all users
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// with token will be considered non-guests. Only guests will be allowed to
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// edit their profile.
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enableUserRolesBasedOnToken: false,
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// Whether or not some features are checked based on token.
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// enableFeaturesBasedOnToken: false,
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// Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests.
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// lockRoomGuestEnabled: false,
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// When enabled the password used for locking a room is restricted to up to the number of digits specified
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// roomPasswordNumberOfDigits: 10,
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// default: roomPasswordNumberOfDigits: false,
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// Message to show the users. Example: 'The service will be down for
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// maintenance at 01:00 AM GMT,
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// noticeMessage: '',
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// Enables calendar integration, depends on googleApiApplicationClientID
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// and microsoftApiApplicationClientID
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// enableCalendarIntegration: false,
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// When 'true', it shows an intermediate page before joining, where the user can configure its devices.
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// prejoinPageEnabled: false,
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// Stats
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//
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// Whether to enable stats collection or not in the TraceablePeerConnection.
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// This can be useful for debugging purposes (post-processing/analysis of
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// the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
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// estimation tests.
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// gatherStats: false,
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// The interval at which PeerConnection.getStats() is called. Defaults to 10000
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// pcStatsInterval: 10000,
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// To enable sending statistics to callstats.io you must provide the
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// Application ID and Secret.
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// callStatsID: '',
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// callStatsSecret: '',
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// enables sending participants display name to callstats
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// enableDisplayNameInStats: false,
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// enables sending participants email if available to callstats and other analytics
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// enableEmailInStats: false,
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// Privacy
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//
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// If third party requests are disabled, no other server will be contacted.
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// This means avatars will be locally generated and callstats integration
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// will not function.
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// disableThirdPartyRequests: false,
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// Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
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//
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p2p: {
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// Enables peer to peer mode. When enabled the system will try to
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// establish a direct connection when there are exactly 2 participants
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// in the room. If that succeeds the conference will stop sending data
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// through the JVB and use the peer to peer connection instead. When a
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// 3rd participant joins the conference will be moved back to the JVB
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// connection.
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enabled: true,
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// Use XEP-0215 to fetch STUN and TURN servers.
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// useStunTurn: true,
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// The STUN servers that will be used in the peer to peer connections
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stunServers: [
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// { urls: 'stun:jitsi-meet.example.com:4446' },
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{ urls: 'stun:meet-jit-si-turnrelay.jitsi.net:443' }
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]
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// Sets the ICE transport policy for the p2p connection. At the time
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// of this writing the list of possible values are 'all' and 'relay',
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// but that is subject to change in the future. The enum is defined in
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// the WebRTC standard:
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// https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
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// If not set, the effective value is 'all'.
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// iceTransportPolicy: 'all',
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// If set to true, it will prefer to use H.264 for P2P calls (if H.264
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// is supported).
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// preferH264: true
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// If set to true, disable H.264 video codec by stripping it out of the
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// SDP.
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// disableH264: false,
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// How long we're going to wait, before going back to P2P after the 3rd
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// participant has left the conference (to filter out page reload).
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// backToP2PDelay: 5
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},
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analytics: {
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// The Google Analytics Tracking ID:
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// googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
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// Matomo configuration:
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// matomoEndpoint: 'https://your-matomo-endpoint/',
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// matomoSiteID: '42',
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// The Amplitude APP Key:
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// amplitudeAPPKey: '<APP_KEY>'
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// Array of script URLs to load as lib-jitsi-meet "analytics handlers".
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// scriptURLs: [
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// "libs/analytics-ga.min.js", // google-analytics
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// "https://example.com/my-custom-analytics.js"
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// ],
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},
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// Information about the jitsi-meet instance we are connecting to, including
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// the user region as seen by the server.
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deploymentInfo: {
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// shard: "shard1",
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// region: "europe",
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// userRegion: "asia"
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},
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// Decides whether the start/stop recording audio notifications should play on record.
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// disableRecordAudioNotification: false,
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// Information for the chrome extension banner
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// chromeExtensionBanner: {
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// // The chrome extension to be installed address
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// url: 'https://chrome.google.com/webstore/detail/jitsi-meetings/kglhbbefdnlheedjiejgomgmfplipfeb',
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// // Extensions info which allows checking if they are installed or not
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// chromeExtensionsInfo: [
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// {
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// id: 'kglhbbefdnlheedjiejgomgmfplipfeb',
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// path: 'jitsi-logo-48x48.png'
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// }
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// ]
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// },
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// Local Recording
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//
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// localRecording: {
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// Enables local recording.
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// Additionally, 'localrecording' (all lowercase) needs to be added to
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// TOOLBAR_BUTTONS in interface_config.js for the Local Recording
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// button to show up on the toolbar.
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//
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// enabled: true,
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//
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// The recording format, can be one of 'ogg', 'flac' or 'wav'.
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// format: 'flac'
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//
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// },
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// Options related to end-to-end (participant to participant) ping.
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// e2eping: {
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// // The interval in milliseconds at which pings will be sent.
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// // Defaults to 10000, set to <= 0 to disable.
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// pingInterval: 10000,
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//
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// // The interval in milliseconds at which analytics events
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// // with the measured RTT will be sent. Defaults to 60000, set
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// // to <= 0 to disable.
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// analyticsInterval: 60000,
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// },
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// If set, will attempt to use the provided video input device label when
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// triggering a screenshare, instead of proceeding through the normal flow
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// for obtaining a desktop stream.
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// NOTE: This option is experimental and is currently intended for internal
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// use only.
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// _desktopSharingSourceDevice: 'sample-id-or-label',
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// If true, any checks to handoff to another application will be prevented
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// and instead the app will continue to display in the current browser.
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// disableDeepLinking: false,
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// A property to disable the right click context menu for localVideo
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// the menu has option to flip the locally seen video for local presentations
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// disableLocalVideoFlip: false,
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// Mainly privacy related settings
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// Disables all invite functions from the app (share, invite, dial out...etc)
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// disableInviteFunctions: true,
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// Disables storing the room name to the recents list
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// doNotStoreRoom: true,
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// Deployment specific URLs.
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// deploymentUrls: {
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// // If specified a 'Help' button will be displayed in the overflow menu with a link to the specified URL for
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// // user documentation.
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// userDocumentationURL: 'https://docs.example.com/video-meetings.html',
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// // If specified a 'Download our apps' button will be displayed in the overflow menu with a link
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// // to the specified URL for an app download page.
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// downloadAppsUrl: 'https://docs.example.com/our-apps.html'
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// },
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// Options related to the remote participant menu.
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// remoteVideoMenu: {
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// // If set to true the 'Kick out' button will be disabled.
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// disableKick: true
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// },
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// If set to true all muting operations of remote participants will be disabled.
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// disableRemoteMute: true,
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// List of undocumented settings used in jitsi-meet
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/**
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_immediateReloadThreshold
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autoRecord
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autoRecordToken
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debug
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debugAudioLevels
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deploymentInfo
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dialInConfCodeUrl
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dialInNumbersUrl
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dialOutAuthUrl
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dialOutCodesUrl
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disableRemoteControl
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displayJids
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etherpad_base
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externalConnectUrl
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firefox_fake_device
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googleApiApplicationClientID
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iAmRecorder
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iAmSipGateway
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microsoftApiApplicationClientID
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peopleSearchQueryTypes
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peopleSearchUrl
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requireDisplayName
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tokenAuthUrl
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*/
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// List of undocumented settings used in lib-jitsi-meet
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/**
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_peerConnStatusOutOfLastNTimeout
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_peerConnStatusRtcMuteTimeout
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abTesting
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avgRtpStatsN
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callStatsConfIDNamespace
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callStatsCustomScriptUrl
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desktopSharingSources
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disableAEC
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disableAGC
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disableAP
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disableHPF
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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 */
|