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config.js 46KB

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  1. /* eslint-disable no-unused-vars, no-var */
  2. var config = {
  3. // Connection
  4. //
  5. hosts: {
  6. // XMPP domain.
  7. domain: 'jitsi-meet.example.com',
  8. // When using authentication, domain for guest users.
  9. // anonymousdomain: 'guest.example.com',
  10. // Domain for authenticated users. Defaults to <domain>.
  11. // authdomain: 'jitsi-meet.example.com',
  12. // Focus component domain. Defaults to focus.<domain>.
  13. // focus: 'focus.jitsi-meet.example.com',
  14. // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
  15. muc: 'conference.jitsi-meet.example.com'
  16. },
  17. // BOSH URL. FIXME: use XEP-0156 to discover it.
  18. bosh: '//jitsi-meet.example.com/http-bind',
  19. // Websocket URL
  20. // websocket: 'wss://jitsi-meet.example.com/xmpp-websocket',
  21. // The real JID of focus participant - can be overridden here
  22. // Do not change username - FIXME: Make focus username configurable
  23. // https://github.com/jitsi/jitsi-meet/issues/7376
  24. // focusUserJid: 'focus@auth.jitsi-meet.example.com',
  25. // Testing / experimental features.
  26. //
  27. testing: {
  28. // Disables the End to End Encryption feature. Useful for debugging
  29. // issues related to insertable streams.
  30. // disableE2EE: false,
  31. // Enables/disables thumbnail reordering in the filmstrip. It is enabled by default unless explicitly
  32. // disabled by the below option.
  33. // enableThumbnailReordering: true,
  34. // Enables XMPP WebSocket (as opposed to BOSH) for the given amount of users.
  35. // mobileXmppWsThreshold: 10 // enable XMPP WebSockets on mobile for 10% of the users
  36. // P2P test mode disables automatic switching to P2P when there are 2
  37. // participants in the conference.
  38. // p2pTestMode: false,
  39. // Enables the test specific features consumed by jitsi-meet-torture
  40. // testMode: false
  41. // Disables the auto-play behavior of *all* newly created video element.
  42. // This is useful when the client runs on a host with limited resources.
  43. // noAutoPlayVideo: false
  44. // Enable / disable 500 Kbps bitrate cap on desktop tracks. When enabled,
  45. // simulcast is turned off for the desktop share. If presenter is turned
  46. // on while screensharing is in progress, the max bitrate is automatically
  47. // adjusted to 2.5 Mbps. This takes a value between 0 and 1 which determines
  48. // the probability for this to be enabled. This setting has been deprecated.
  49. // desktopSharingFrameRate.max now determines whether simulcast will be enabled
  50. // or disabled for the screenshare.
  51. // capScreenshareBitrate: 1 // 0 to disable - deprecated.
  52. // Enable callstats only for a percentage of users.
  53. // This takes a value between 0 and 100 which determines the probability for
  54. // the callstats to be enabled.
  55. // callStatsThreshold: 5 // enable callstats for 5% of the users.
  56. },
  57. // Disables moderator indicators.
  58. // disableModeratorIndicator: false,
  59. // Disables the reactions feature.
  60. // disableReactions: true,
  61. // Disables polls feature.
  62. // disablePolls: false,
  63. // Disables ICE/UDP by filtering out local and remote UDP candidates in
  64. // signalling.
  65. // webrtcIceUdpDisable: false,
  66. // Disables ICE/TCP by filtering out local and remote TCP candidates in
  67. // signalling.
  68. // webrtcIceTcpDisable: false,
  69. // Media
  70. //
  71. // Enable unified plan implementation support on Chromium based browsers.
  72. // enableUnifiedOnChrome: false,
  73. // Audio
  74. // Disable measuring of audio levels.
  75. // disableAudioLevels: false,
  76. // audioLevelsInterval: 200,
  77. // Enabling this will run the lib-jitsi-meet no audio detection module which
  78. // will notify the user if the current selected microphone has no audio
  79. // input and will suggest another valid device if one is present.
  80. enableNoAudioDetection: true,
  81. // Enabling this will show a "Save Logs" link in the GSM popover that can be
  82. // used to collect debug information (XMPP IQs, SDP offer/answer cycles)
  83. // about the call.
  84. // enableSaveLogs: false,
  85. // Enabling this will hide the "Show More" link in the GSM popover that can be
  86. // used to display more statistics about the connection (IP, Port, protocol, etc).
  87. // disableShowMoreStats: true,
  88. // Enabling this will run the lib-jitsi-meet noise detection module which will
  89. // notify the user if there is noise, other than voice, coming from the current
  90. // selected microphone. The purpose it to let the user know that the input could
  91. // be potentially unpleasant for other meeting participants.
  92. enableNoisyMicDetection: true,
  93. // Start the conference in audio only mode (no video is being received nor
  94. // sent).
  95. // startAudioOnly: false,
  96. // Every participant after the Nth will start audio muted.
  97. // startAudioMuted: 10,
  98. // Start calls with audio muted. Unlike the option above, this one is only
  99. // applied locally. FIXME: having these 2 options is confusing.
  100. // startWithAudioMuted: false,
  101. // Enabling it (with #params) will disable local audio output of remote
  102. // participants and to enable it back a reload is needed.
  103. // startSilent: false
  104. // Enables support for opus-red (redundancy for Opus).
  105. // enableOpusRed: false,
  106. // Specify audio quality stereo and opusMaxAverageBitrate values in order to enable HD audio.
  107. // Beware, by doing so, you are disabling echo cancellation, noise suppression and AGC.
  108. // audioQuality: {
  109. // stereo: false,
  110. // opusMaxAverageBitrate: null // Value to fit the 6000 to 510000 range.
  111. // },
  112. // Video
  113. // Sets the preferred resolution (height) for local video. Defaults to 720.
  114. // resolution: 720,
  115. // Specifies whether the raised hand will hide when someone becomes a dominant speaker or not
  116. // disableRemoveRaisedHandOnFocus: false,
  117. // Specifies whether there will be a search field in speaker stats or not
  118. // disableSpeakerStatsSearch: false,
  119. // Specifies whether participants in speaker stats should be ordered or not, and with what priority
  120. // speakerStatsOrder: [
  121. // 'role', <- Moderators on top
  122. // 'name', <- Alphabetically by name
  123. // 'hasLeft', <- The ones that have left in the bottom
  124. // ] <- the order of the array elements determines priority
  125. // How many participants while in the tile view mode, before the receiving video quality is reduced from HD to SD.
  126. // Use -1 to disable.
  127. // maxFullResolutionParticipants: 2,
  128. // w3c spec-compliant video constraints to use for video capture. Currently
  129. // used by browsers that return true from lib-jitsi-meet's
  130. // util#browser#usesNewGumFlow. The constraints are independent from
  131. // this config's resolution value. Defaults to requesting an ideal
  132. // resolution of 720p.
  133. // constraints: {
  134. // video: {
  135. // height: {
  136. // ideal: 720,
  137. // max: 720,
  138. // min: 240
  139. // }
  140. // }
  141. // },
  142. // Enable / disable simulcast support.
  143. // disableSimulcast: false,
  144. // Enable / disable layer suspension. If enabled, endpoints whose HD layers are not in use will be suspended
  145. // (no longer sent) until they are requested again. This is enabled by default. This must be enabled for screen
  146. // sharing to work as expected on Chrome. Disabling this might result in low resolution screenshare being sent
  147. // by the client.
  148. // enableLayerSuspension: false,
  149. // Every participant after the Nth will start video muted.
  150. // startVideoMuted: 10,
  151. // Start calls with video muted. Unlike the option above, this one is only
  152. // applied locally. FIXME: having these 2 options is confusing.
  153. // startWithVideoMuted: false,
  154. // If set to true, prefer to use the H.264 video codec (if supported).
  155. // Note that it's not recommended to do this because simulcast is not
  156. // supported when using H.264. For 1-to-1 calls this setting is enabled by
  157. // default and can be toggled in the p2p section.
  158. // This option has been deprecated, use preferredCodec under videoQuality section instead.
  159. // preferH264: true,
  160. // If set to true, disable H.264 video codec by stripping it out of the
  161. // SDP.
  162. // disableH264: false,
  163. // Desktop sharing
  164. // Optional desktop sharing frame rate options. Default value: min:5, max:5.
  165. // desktopSharingFrameRate: {
  166. // min: 5,
  167. // max: 5
  168. // },
  169. // Try to start calls with screen-sharing instead of camera video.
  170. // startScreenSharing: false,
  171. // Recording
  172. // Whether to enable file recording or not.
  173. // fileRecordingsEnabled: false,
  174. // Enable the dropbox integration.
  175. // dropbox: {
  176. // appKey: '<APP_KEY>' // Specify your app key here.
  177. // // A URL to redirect the user to, after authenticating
  178. // // by default uses:
  179. // // 'https://jitsi-meet.example.com/static/oauth.html'
  180. // redirectURI:
  181. // 'https://jitsi-meet.example.com/subfolder/static/oauth.html'
  182. // },
  183. // When integrations like dropbox are enabled only that will be shown,
  184. // by enabling fileRecordingsServiceEnabled, we show both the integrations
  185. // and the generic recording service (its configuration and storage type
  186. // depends on jibri configuration)
  187. // fileRecordingsServiceEnabled: false,
  188. // Whether to show the possibility to share file recording with other people
  189. // (e.g. meeting participants), based on the actual implementation
  190. // on the backend.
  191. // fileRecordingsServiceSharingEnabled: false,
  192. // Whether to enable live streaming or not.
  193. // liveStreamingEnabled: false,
  194. // Transcription (in interface_config,
  195. // subtitles and buttons can be configured)
  196. // transcribingEnabled: false,
  197. // If true transcriber will use the application language.
  198. // The application language is either explicitly set by participants in their settings or automatically
  199. // detected based on the environment, e.g. if the app is opened in a chrome instance which is using french as its
  200. // default language then transcriptions for that participant will be in french.
  201. // Defaults to true.
  202. // transcribeWithAppLanguage: true,
  203. // Transcriber language. This settings will only work if "transcribeWithAppLanguage" is explicitly set to false.
  204. // Available languages can be found in
  205. // ./src/react/features/transcribing/transcriber-langs.json.
  206. // preferredTranscribeLanguage: 'en-US',
  207. // Enables automatic turning on captions when recording is started
  208. // autoCaptionOnRecord: false,
  209. // Misc
  210. // Default value for the channel "last N" attribute. -1 for unlimited.
  211. channelLastN: -1,
  212. // Connection indicators
  213. // connectionIndicators: {
  214. // autoHide: true,
  215. // autoHideTimeout: 5000,
  216. // disabled: false,
  217. // inactiveDisabled: false
  218. // },
  219. // Provides a way for the lastN value to be controlled through the UI.
  220. // When startLastN is present, conference starts with a last-n value of startLastN and channelLastN
  221. // value will be used when the quality level is selected using "Manage Video Quality" slider.
  222. // startLastN: 1,
  223. // Provides a way to use different "last N" values based on the number of participants in the conference.
  224. // The keys in an Object represent number of participants and the values are "last N" to be used when number of
  225. // participants gets to or above the number.
  226. //
  227. // For the given example mapping, "last N" will be set to 20 as long as there are at least 5, but less than
  228. // 29 participants in the call and it will be lowered to 15 when the 30th participant joins. The 'channelLastN'
  229. // will be used as default until the first threshold is reached.
  230. //
  231. // lastNLimits: {
  232. // 5: 20,
  233. // 30: 15,
  234. // 50: 10,
  235. // 70: 5,
  236. // 90: 2
  237. // },
  238. // Provides a way to translate the legacy bridge signaling messages, 'LastNChangedEvent',
  239. // 'SelectedEndpointsChangedEvent' and 'ReceiverVideoConstraint' into the new 'ReceiverVideoConstraints' message
  240. // that invokes the new bandwidth allocation algorithm in the bridge which is described here
  241. // - https://github.com/jitsi/jitsi-videobridge/blob/master/doc/allocation.md.
  242. // useNewBandwidthAllocationStrategy: false,
  243. // Specify the settings for video quality optimizations on the client.
  244. // videoQuality: {
  245. // // Provides a way to prevent a video codec from being negotiated on the JVB connection. The codec specified
  246. // // here will be removed from the list of codecs present in the SDP answer generated by the client. If the
  247. // // same codec is specified for both the disabled and preferred option, the disable settings will prevail.
  248. // // Note that 'VP8' cannot be disabled since it's a mandatory codec, the setting will be ignored in this case.
  249. // disabledCodec: 'H264',
  250. //
  251. // // Provides a way to set a preferred video codec for the JVB connection. If 'H264' is specified here,
  252. // // simulcast will be automatically disabled since JVB doesn't support H264 simulcast yet. This will only
  253. // // rearrange the the preference order of the codecs in the SDP answer generated by the browser only if the
  254. // // preferred codec specified here is present. Please ensure that the JVB offers the specified codec for this
  255. // // to take effect.
  256. // preferredCodec: 'VP8',
  257. //
  258. // // Provides a way to enforce the preferred codec for the conference even when the conference has endpoints
  259. // // that do not support the preferred codec. For example, older versions of Safari do not support VP9 yet.
  260. // // This will result in Safari not being able to decode video from endpoints sending VP9 video.
  261. // // When set to false, the conference falls back to VP8 whenever there is an endpoint that doesn't support the
  262. // // preferred codec and goes back to the preferred codec when that endpoint leaves.
  263. // // enforcePreferredCodec: false,
  264. //
  265. // // Provides a way to configure the maximum bitrates that will be enforced on the simulcast streams for
  266. // // video tracks. The keys in the object represent the type of the stream (LD, SD or HD) and the values
  267. // // are the max.bitrates to be set on that particular type of stream. The actual send may vary based on
  268. // // the available bandwidth calculated by the browser, but it will be capped by the values specified here.
  269. // // This is currently not implemented on app based clients on mobile.
  270. // maxBitratesVideo: {
  271. // H264: {
  272. // low: 200000,
  273. // standard: 500000,
  274. // high: 1500000
  275. // },
  276. // VP8 : {
  277. // low: 200000,
  278. // standard: 500000,
  279. // high: 1500000
  280. // },
  281. // VP9: {
  282. // low: 100000,
  283. // standard: 300000,
  284. // high: 1200000
  285. // }
  286. // },
  287. //
  288. // // The options can be used to override default thresholds of video thumbnail heights corresponding to
  289. // // the video quality levels used in the application. At the time of this writing the allowed levels are:
  290. // // 'low' - for the low quality level (180p at the time of this writing)
  291. // // 'standard' - for the medium quality level (360p)
  292. // // 'high' - for the high quality level (720p)
  293. // // The keys should be positive numbers which represent the minimal thumbnail height for the quality level.
  294. // //
  295. // // With the default config value below the application will use 'low' quality until the thumbnails are
  296. // // at least 360 pixels tall. If the thumbnail height reaches 720 pixels then the application will switch to
  297. // // the high quality.
  298. // minHeightForQualityLvl: {
  299. // 360: 'standard',
  300. // 720: 'high'
  301. // },
  302. //
  303. // // Provides a way to resize the desktop track to 720p (if it is greater than 720p) before creating a canvas
  304. // // for the presenter mode (camera picture-in-picture mode with screenshare).
  305. // resizeDesktopForPresenter: false
  306. // },
  307. // // Options for the recording limit notification.
  308. // recordingLimit: {
  309. //
  310. // // The recording limit in minutes. Note: This number appears in the notification text
  311. // // but doesn't enforce the actual recording time limit. This should be configured in
  312. // // jibri!
  313. // limit: 60,
  314. //
  315. // // The name of the app with unlimited recordings.
  316. // appName: 'Unlimited recordings APP',
  317. //
  318. // // The URL of the app with unlimited recordings.
  319. // appURL: 'https://unlimited.recordings.app.com/'
  320. // },
  321. // Disables or enables RTX (RFC 4588) (defaults to false).
  322. // disableRtx: false,
  323. // Disables or enables TCC support in this client (default: enabled).
  324. // enableTcc: true,
  325. // Disables or enables REMB support in this client (default: enabled).
  326. // enableRemb: true,
  327. // Enables ICE restart logic in LJM and displays the page reload overlay on
  328. // ICE failure. Current disabled by default because it's causing issues with
  329. // signaling when Octo is enabled. Also when we do an "ICE restart"(which is
  330. // not a real ICE restart), the client maintains the TCC sequence number
  331. // counter, but the bridge resets it. The bridge sends media packets with
  332. // TCC sequence numbers starting from 0.
  333. // enableIceRestart: false,
  334. // Enables forced reload of the client when the call is migrated as a result of
  335. // the bridge going down.
  336. // enableForcedReload: true,
  337. // Use TURN/UDP servers for the jitsi-videobridge connection (by default
  338. // we filter out TURN/UDP because it is usually not needed since the
  339. // bridge itself is reachable via UDP)
  340. // useTurnUdp: false
  341. // Enable support for encoded transform in supported browsers. This allows
  342. // E2EE to work in Safari if the corresponding flag is enabled in the browser.
  343. // Experimental.
  344. // enableEncodedTransformSupport: false,
  345. // UI
  346. //
  347. // Disables responsive tiles.
  348. // disableResponsiveTiles: false,
  349. // Hides lobby button
  350. // hideLobbyButton: false,
  351. // Require users to always specify a display name.
  352. // requireDisplayName: true,
  353. // Whether to use a welcome page or not. In case it's false a random room
  354. // will be joined when no room is specified.
  355. enableWelcomePage: true,
  356. // Disable app shortcuts that are registered upon joining a conference
  357. // disableShortcuts: false,
  358. // Disable initial browser getUserMedia requests.
  359. // This is useful for scenarios where users might want to start a conference for screensharing only
  360. // disableInitialGUM: false,
  361. // Enabling the close page will ignore the welcome page redirection when
  362. // a call is hangup.
  363. // enableClosePage: false,
  364. // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
  365. // Setting this to null, will also disable showing the remote videos
  366. // when the toolbar is shown on mouse movements
  367. // disable1On1Mode: null | false | true,
  368. // Default language for the user interface.
  369. // defaultLanguage: 'en',
  370. // Disables profile and the edit of all fields from the profile settings (display name and email)
  371. // disableProfile: false,
  372. // Whether or not some features are checked based on token.
  373. // enableFeaturesBasedOnToken: false,
  374. // When enabled the password used for locking a room is restricted to up to the number of digits specified
  375. // roomPasswordNumberOfDigits: 10,
  376. // default: roomPasswordNumberOfDigits: false,
  377. // Message to show the users. Example: 'The service will be down for
  378. // maintenance at 01:00 AM GMT,
  379. // noticeMessage: '',
  380. // Enables calendar integration, depends on googleApiApplicationClientID
  381. // and microsoftApiApplicationClientID
  382. // enableCalendarIntegration: false,
  383. // When 'true', it shows an intermediate page before joining, where the user can configure their devices.
  384. // prejoinPageEnabled: false,
  385. // If etherpad integration is enabled, setting this to true will
  386. // automatically open the etherpad when a participant joins. This
  387. // does not affect the mobile app since opening an etherpad
  388. // obscures the conference controls -- it's better to let users
  389. // choose to open the pad on their own in that case.
  390. // openSharedDocumentOnJoin: false,
  391. // If true, shows the unsafe room name warning label when a room name is
  392. // deemed unsafe (due to the simplicity in the name) and a password is not
  393. // set or the lobby is not enabled.
  394. // enableInsecureRoomNameWarning: false,
  395. // Whether to automatically copy invitation URL after creating a room.
  396. // Document should be focused for this option to work
  397. // enableAutomaticUrlCopy: false,
  398. // Base URL for a Gravatar-compatible service. Defaults to libravatar.
  399. // gravatarBaseURL: 'https://seccdn.libravatar.org/avatar/',
  400. // App name to be displayed in the invitation email subject, as an alternative to
  401. // interfaceConfig.APP_NAME.
  402. // inviteAppName: null,
  403. // Moved from interfaceConfig(TOOLBAR_BUTTONS).
  404. // The name of the toolbar buttons to display in the toolbar, including the
  405. // "More actions" menu. If present, the button will display. Exceptions are
  406. // "livestreaming" and "recording" which also require being a moderator and
  407. // some other values in config.js to be enabled. Also, the "profile" button will
  408. // not display for users with a JWT.
  409. // Notes:
  410. // - it's impossible to choose which buttons go in the "More actions" menu
  411. // - it's impossible to control the placement of buttons
  412. // - 'desktop' controls the "Share your screen" button
  413. // - if `toolbarButtons` is undefined, we fallback to enabling all buttons on the UI
  414. // toolbarButtons: [
  415. // 'camera',
  416. // 'chat',
  417. // 'closedcaptions',
  418. // 'desktop',
  419. // 'download',
  420. // 'embedmeeting',
  421. // 'etherpad',
  422. // 'feedback',
  423. // 'filmstrip',
  424. // 'fullscreen',
  425. // 'hangup',
  426. // 'help',
  427. // 'invite',
  428. // 'livestreaming',
  429. // 'microphone',
  430. // 'mute-everyone',
  431. // 'mute-video-everyone',
  432. // 'participants-pane',
  433. // 'profile',
  434. // 'raisehand',
  435. // 'recording',
  436. // 'security',
  437. // 'select-background',
  438. // 'settings',
  439. // 'shareaudio',
  440. // 'sharedvideo',
  441. // 'shortcuts',
  442. // 'stats',
  443. // 'tileview',
  444. // 'toggle-camera',
  445. // 'videoquality',
  446. // '__end'
  447. // ],
  448. // Toolbar buttons which have their click event exposed through the API on
  449. // `toolbarButtonClicked` event instead of executing the normal click routine.
  450. // buttonsWithNotifyClick: [
  451. // 'camera',
  452. // 'chat',
  453. // 'closedcaptions',
  454. // 'desktop',
  455. // 'download',
  456. // 'embedmeeting',
  457. // 'etherpad',
  458. // 'feedback',
  459. // 'filmstrip',
  460. // 'fullscreen',
  461. // 'hangup',
  462. // 'help',
  463. // 'invite',
  464. // 'livestreaming',
  465. // 'microphone',
  466. // 'mute-everyone',
  467. // 'mute-video-everyone',
  468. // 'participants-pane',
  469. // 'profile',
  470. // 'raisehand',
  471. // 'recording',
  472. // 'security',
  473. // 'select-background',
  474. // 'settings',
  475. // 'shareaudio',
  476. // 'sharedvideo',
  477. // 'shortcuts',
  478. // 'stats',
  479. // 'tileview',
  480. // 'toggle-camera',
  481. // 'videoquality',
  482. // '__end'
  483. // ],
  484. // List of pre meeting screens buttons to hide. The values must be one or more of the 5 allowed buttons:
  485. // 'microphone', 'camera', 'select-background', 'invite', 'settings'
  486. // hiddenPremeetingButtons: [],
  487. // Stats
  488. //
  489. // Whether to enable stats collection or not in the TraceablePeerConnection.
  490. // This can be useful for debugging purposes (post-processing/analysis of
  491. // the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
  492. // estimation tests.
  493. // gatherStats: false,
  494. // The interval at which PeerConnection.getStats() is called. Defaults to 10000
  495. // pcStatsInterval: 10000,
  496. // To enable sending statistics to callstats.io you must provide the
  497. // Application ID and Secret.
  498. // callStatsID: '',
  499. // callStatsSecret: '',
  500. // The callstats initialize config params as described in the API:
  501. // https://docs.callstats.io/docs/javascript#callstatsinitialize-with-app-secret
  502. // callStatsConfigParams: {
  503. // disableBeforeUnloadHandler: true, // disables callstats.js's window.onbeforeunload parameter.
  504. // applicationVersion: "app_version", // Application version specified by the developer.
  505. // disablePrecalltest: true, // disables the pre-call test, it is enabled by default.
  506. // siteID: "siteID", // The name/ID of the site/campus from where the call/pre-call test is made.
  507. // additionalIDs: { // additionalIDs object, contains application related IDs.
  508. // customerID: "Customer Identifier. Example, walmart.",
  509. // tenantID: "Tenant Identifier. Example, monster.",
  510. // productName: "Product Name. Example, Jitsi.",
  511. // meetingsName: "Meeting Name. Example, Jitsi loves callstats.",
  512. // serverName: "Server/MiddleBox Name. Example, jvb-prod-us-east-mlkncws12.",
  513. // pbxID: "PBX Identifier. Example, walmart.",
  514. // pbxExtensionID: "PBX Extension Identifier. Example, 5625.",
  515. // fqExtensionID: "Fully qualified Extension Identifier. Example, +71 (US) +5625.",
  516. // sessionID: "Session Identifier. Example, session-12-34"
  517. // },
  518. // collectLegacyStats: true, //enables the collection of legacy stats in chrome browser
  519. // collectIP: true //enables the collection localIP address
  520. // },
  521. // Enables sending participants' display names to callstats
  522. // enableDisplayNameInStats: false,
  523. // Enables sending participants' emails (if available) to callstats and other analytics
  524. // enableEmailInStats: false,
  525. // Controls the percentage of automatic feedback shown to participants when callstats is enabled.
  526. // The default value is 100%. If set to 0, no automatic feedback will be requested
  527. // feedbackPercentage: 100,
  528. // Privacy
  529. //
  530. // If third party requests are disabled, no other server will be contacted.
  531. // This means avatars will be locally generated and callstats integration
  532. // will not function.
  533. // disableThirdPartyRequests: false,
  534. // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
  535. //
  536. p2p: {
  537. // Enables peer to peer mode. When enabled the system will try to
  538. // establish a direct connection when there are exactly 2 participants
  539. // in the room. If that succeeds the conference will stop sending data
  540. // through the JVB and use the peer to peer connection instead. When a
  541. // 3rd participant joins the conference will be moved back to the JVB
  542. // connection.
  543. enabled: true,
  544. // Enable unified plan implementation support on Chromium for p2p connection.
  545. // enableUnifiedOnChrome: false,
  546. // Sets the ICE transport policy for the p2p connection. At the time
  547. // of this writing the list of possible values are 'all' and 'relay',
  548. // but that is subject to change in the future. The enum is defined in
  549. // the WebRTC standard:
  550. // https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
  551. // If not set, the effective value is 'all'.
  552. // iceTransportPolicy: 'all',
  553. // If set to true, it will prefer to use H.264 for P2P calls (if H.264
  554. // is supported). This setting is deprecated, use preferredCodec instead.
  555. // preferH264: true,
  556. // Provides a way to set the video codec preference on the p2p connection. Acceptable
  557. // codec values are 'VP8', 'VP9' and 'H264'.
  558. // preferredCodec: 'H264',
  559. // If set to true, disable H.264 video codec by stripping it out of the
  560. // SDP. This setting is deprecated, use disabledCodec instead.
  561. // disableH264: false,
  562. // Provides a way to prevent a video codec from being negotiated on the p2p connection.
  563. // disabledCodec: '',
  564. // How long we're going to wait, before going back to P2P after the 3rd
  565. // participant has left the conference (to filter out page reload).
  566. // backToP2PDelay: 5,
  567. // The STUN servers that will be used in the peer to peer connections
  568. stunServers: [
  569. // { urls: 'stun:jitsi-meet.example.com:3478' },
  570. { urls: 'stun:meet-jit-si-turnrelay.jitsi.net:443' }
  571. ]
  572. },
  573. analytics: {
  574. // True if the analytics should be disabled
  575. // disabled: false,
  576. // The Google Analytics Tracking ID:
  577. // googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
  578. // Matomo configuration:
  579. // matomoEndpoint: 'https://your-matomo-endpoint/',
  580. // matomoSiteID: '42',
  581. // The Amplitude APP Key:
  582. // amplitudeAPPKey: '<APP_KEY>'
  583. // Configuration for the rtcstats server:
  584. // By enabling rtcstats server every time a conference is joined the rtcstats
  585. // module connects to the provided rtcstatsEndpoint and sends statistics regarding
  586. // PeerConnection states along with getStats metrics polled at the specified
  587. // interval.
  588. // rtcstatsEnabled: true,
  589. // In order to enable rtcstats one needs to provide a endpoint url.
  590. // rtcstatsEndpoint: wss://rtcstats-server-pilot.jitsi.net/,
  591. // The interval at which rtcstats will poll getStats, defaults to 1000ms.
  592. // If the value is set to 0 getStats won't be polled and the rtcstats client
  593. // will only send data related to RTCPeerConnection events.
  594. // rtcstatsPolIInterval: 1000,
  595. // Array of script URLs to load as lib-jitsi-meet "analytics handlers".
  596. // scriptURLs: [
  597. // "libs/analytics-ga.min.js", // google-analytics
  598. // "https://example.com/my-custom-analytics.js"
  599. // ],
  600. },
  601. // Logs that should go be passed through the 'log' event if a handler is defined for it
  602. // apiLogLevels: ['warn', 'log', 'error', 'info', 'debug'],
  603. // Information about the jitsi-meet instance we are connecting to, including
  604. // the user region as seen by the server.
  605. deploymentInfo: {
  606. // shard: "shard1",
  607. // region: "europe",
  608. // userRegion: "asia"
  609. },
  610. // Array<string> of disabled sounds.
  611. // Possible values:
  612. // - 'ASKED_TO_UNMUTE_SOUND'
  613. // - 'E2EE_OFF_SOUND'
  614. // - 'E2EE_ON_SOUND'
  615. // - 'INCOMING_MSG_SOUND'
  616. // - 'KNOCKING_PARTICIPANT_SOUND'
  617. // - 'LIVE_STREAMING_OFF_SOUND'
  618. // - 'LIVE_STREAMING_ON_SOUND'
  619. // - 'NO_AUDIO_SIGNAL_SOUND'
  620. // - 'NOISY_AUDIO_INPUT_SOUND'
  621. // - 'OUTGOING_CALL_EXPIRED_SOUND'
  622. // - 'OUTGOING_CALL_REJECTED_SOUND'
  623. // - 'OUTGOING_CALL_RINGING_SOUND'
  624. // - 'OUTGOING_CALL_START_SOUND'
  625. // - 'PARTICIPANT_JOINED_SOUND'
  626. // - 'PARTICIPANT_LEFT_SOUND'
  627. // - 'RAISE_HAND_SOUND'
  628. // - 'RECORDING_OFF_SOUND'
  629. // - 'RECORDING_ON_SOUND'
  630. // - 'TALK_WHILE_MUTED_SOUND'
  631. // disabledSounds: [],
  632. // DEPRECATED! Use `disabledSounds` instead.
  633. // Decides whether the start/stop recording audio notifications should play on record.
  634. // disableRecordAudioNotification: false,
  635. // DEPRECATED! Use `disabledSounds` instead.
  636. // Disables the sounds that play when other participants join or leave the
  637. // conference (if set to true, these sounds will not be played).
  638. // disableJoinLeaveSounds: false,
  639. // DEPRECATED! Use `disabledSounds` instead.
  640. // Disables the sounds that play when a chat message is received.
  641. // disableIncomingMessageSound: false,
  642. // Information for the chrome extension banner
  643. // chromeExtensionBanner: {
  644. // // The chrome extension to be installed address
  645. // url: 'https://chrome.google.com/webstore/detail/jitsi-meetings/kglhbbefdnlheedjiejgomgmfplipfeb',
  646. // // Extensions info which allows checking if they are installed or not
  647. // chromeExtensionsInfo: [
  648. // {
  649. // id: 'kglhbbefdnlheedjiejgomgmfplipfeb',
  650. // path: 'jitsi-logo-48x48.png'
  651. // }
  652. // ]
  653. // },
  654. // Local Recording
  655. //
  656. // localRecording: {
  657. // Enables local recording.
  658. // Additionally, 'localrecording' (all lowercase) needs to be added to
  659. // the `toolbarButtons`-array for the Local Recording button to show up
  660. // on the toolbar.
  661. //
  662. // enabled: true,
  663. //
  664. // The recording format, can be one of 'ogg', 'flac' or 'wav'.
  665. // format: 'flac'
  666. //
  667. // },
  668. // Options related to end-to-end (participant to participant) ping.
  669. // e2eping: {
  670. // // The interval in milliseconds at which pings will be sent.
  671. // // Defaults to 10000, set to <= 0 to disable.
  672. // pingInterval: 10000,
  673. //
  674. // // The interval in milliseconds at which analytics events
  675. // // with the measured RTT will be sent. Defaults to 60000, set
  676. // // to <= 0 to disable.
  677. // analyticsInterval: 60000,
  678. // },
  679. // If set, will attempt to use the provided video input device label when
  680. // triggering a screenshare, instead of proceeding through the normal flow
  681. // for obtaining a desktop stream.
  682. // NOTE: This option is experimental and is currently intended for internal
  683. // use only.
  684. // _desktopSharingSourceDevice: 'sample-id-or-label',
  685. // If true, any checks to handoff to another application will be prevented
  686. // and instead the app will continue to display in the current browser.
  687. // disableDeepLinking: false,
  688. // A property to disable the right click context menu for localVideo
  689. // the menu has option to flip the locally seen video for local presentations
  690. // disableLocalVideoFlip: false,
  691. // A property used to unset the default flip state of the local video.
  692. // When it is set to 'true', the local(self) video will not be mirrored anymore.
  693. // doNotFlipLocalVideo: false,
  694. // Mainly privacy related settings
  695. // Disables all invite functions from the app (share, invite, dial out...etc)
  696. // disableInviteFunctions: true,
  697. // Disables storing the room name to the recents list
  698. // doNotStoreRoom: true,
  699. // Deployment specific URLs.
  700. // deploymentUrls: {
  701. // // If specified a 'Help' button will be displayed in the overflow menu with a link to the specified URL for
  702. // // user documentation.
  703. // userDocumentationURL: 'https://docs.example.com/video-meetings.html',
  704. // // If specified a 'Download our apps' button will be displayed in the overflow menu with a link
  705. // // to the specified URL for an app download page.
  706. // downloadAppsUrl: 'https://docs.example.com/our-apps.html'
  707. // },
  708. // Options related to the remote participant menu.
  709. // remoteVideoMenu: {
  710. // // If set to true the 'Kick out' button will be disabled.
  711. // disableKick: true,
  712. // // If set to true the 'Grant moderator' button will be disabled.
  713. // disableGrantModerator: true
  714. // },
  715. // If set to true all muting operations of remote participants will be disabled.
  716. // disableRemoteMute: true,
  717. // Enables support for lip-sync for this client (if the browser supports it).
  718. // enableLipSync: false
  719. /**
  720. External API url used to receive branding specific information.
  721. If there is no url set or there are missing fields, the defaults are applied.
  722. The config file should be in JSON.
  723. None of the fields are mandatory and the response must have the shape:
  724. {
  725. // The domain url to apply (will replace the domain in the sharing conference link/embed section)
  726. inviteDomain: 'example-company.org,
  727. // The hex value for the colour used as background
  728. backgroundColor: '#fff',
  729. // The url for the image used as background
  730. backgroundImageUrl: 'https://example.com/background-img.png',
  731. // The anchor url used when clicking the logo image
  732. logoClickUrl: 'https://example-company.org',
  733. // The url used for the image used as logo
  734. logoImageUrl: 'https://example.com/logo-img.png',
  735. // Overwrite for pool of background images for avatars
  736. avatarBackgrounds: ['url(https://example.com/avatar-background-1.png)', '#FFF'],
  737. // The lobby/prejoin screen background
  738. premeetingBackground: 'url(https://example.com/premeeting-background.png)'
  739. }
  740. */
  741. // dynamicBrandingUrl: '',
  742. // Sets the background transparency level. '0' is fully transparent, '1' is opaque.
  743. // backgroundAlpha: 1,
  744. // The URL of the moderated rooms microservice, if available. If it
  745. // is present, a link to the service will be rendered on the welcome page,
  746. // otherwise the app doesn't render it.
  747. // moderatedRoomServiceUrl: 'https://moderated.jitsi-meet.example.com',
  748. // If true, tile view will not be enabled automatically when the participants count threshold is reached.
  749. // disableTileView: true,
  750. // Controls the visibility and behavior of the top header conference info labels.
  751. // If a label's id is not in any of the 2 arrays, it will not be visible at all on the header.
  752. // conferenceInfo: {
  753. // // those labels will not be hidden in tandem with the toolbox.
  754. // alwaysVisible: ['recording', 'local-recording'],
  755. // // those labels will be auto-hidden in tandem with the toolbox buttons.
  756. // autoHide: [
  757. // 'subject',
  758. // 'conference-timer',
  759. // 'participants-count',
  760. // 'e2ee',
  761. // 'transcribing',
  762. // 'video-quality',
  763. // 'insecure-room'
  764. // ]
  765. // },
  766. // Hides the conference subject
  767. // hideConferenceSubject: true,
  768. // Hides the conference timer.
  769. // hideConferenceTimer: true,
  770. // Hides the recording label
  771. // hideRecordingLabel: false,
  772. // Hides the participants stats
  773. // hideParticipantsStats: true,
  774. // Sets the conference subject
  775. // subject: 'Conference Subject',
  776. // This property is related to the use case when jitsi-meet is used via the IFrame API. When the property is true
  777. // jitsi-meet will use the local storage of the host page instead of its own. This option is useful if the browser
  778. // is not persisting the local storage inside the iframe.
  779. // useHostPageLocalStorage: true,
  780. // etherpad ("shared document") integration.
  781. //
  782. // If set, add a "Open shared document" link to the bottom right menu that
  783. // will open an etherpad document.
  784. // etherpad_base: 'https://your-etherpad-installati.on/p/',
  785. // If etherpad_base is set, and useRoomAsSharedDocumentName is set to true,
  786. // open a pad with the name of the room (lowercased) instead of a pad with a
  787. // random UUID.
  788. // useRoomAsSharedDocumentName: true,
  789. // List of undocumented settings used in jitsi-meet
  790. /**
  791. _immediateReloadThreshold
  792. debug
  793. debugAudioLevels
  794. deploymentInfo
  795. dialInConfCodeUrl
  796. dialInNumbersUrl
  797. dialOutAuthUrl
  798. dialOutCodesUrl
  799. disableRemoteControl
  800. displayJids
  801. externalConnectUrl
  802. e2eeLabels
  803. firefox_fake_device
  804. googleApiApplicationClientID
  805. iAmRecorder
  806. iAmSipGateway
  807. microsoftApiApplicationClientID
  808. peopleSearchQueryTypes
  809. peopleSearchUrl
  810. requireDisplayName
  811. tokenAuthUrl
  812. */
  813. /**
  814. * This property can be used to alter the generated meeting invite links (in combination with a branding domain
  815. * which is retrieved internally by jitsi meet) (e.g. https://meet.jit.si/someMeeting
  816. * can become https://brandedDomain/roomAlias)
  817. */
  818. // brandingRoomAlias: null,
  819. // List of undocumented settings used in lib-jitsi-meet
  820. /**
  821. _peerConnStatusOutOfLastNTimeout
  822. _peerConnStatusRtcMuteTimeout
  823. abTesting
  824. avgRtpStatsN
  825. callStatsConfIDNamespace
  826. callStatsCustomScriptUrl
  827. desktopSharingSources
  828. disableAEC
  829. disableAGC
  830. disableAP
  831. disableHPF
  832. disableNS
  833. enableTalkWhileMuted
  834. forceJVB121Ratio
  835. forceTurnRelay
  836. hiddenDomain
  837. ignoreStartMuted
  838. websocketKeepAlive
  839. websocketKeepAliveUrl
  840. */
  841. /**
  842. * Default interval (milliseconds) for triggering mouseMoved iframe API event
  843. */
  844. mouseMoveCallbackInterval: 1000,
  845. /**
  846. Use this array to configure which notifications will be shown to the user
  847. The items correspond to the title or description key of that notification
  848. Some of these notifications also depend on some other internal logic to be displayed or not,
  849. so adding them here will not ensure they will always be displayed
  850. A falsy value for this prop will result in having all notifications enabled (e.g null, undefined, false)
  851. */
  852. // notifications: [
  853. // 'connection.CONNFAIL', // shown when the connection fails,
  854. // 'dialog.cameraNotSendingData', // shown when there's no feed from user's camera
  855. // 'dialog.kickTitle', // shown when user has been kicked
  856. // 'dialog.liveStreaming', // livestreaming notifications (pending, on, off, limits)
  857. // 'dialog.lockTitle', // shown when setting conference password fails
  858. // 'dialog.maxUsersLimitReached', // shown when maximmum users limit has been reached
  859. // 'dialog.micNotSendingData', // shown when user's mic is not sending any audio
  860. // 'dialog.passwordNotSupportedTitle', // shown when setting conference password fails due to password format
  861. // 'dialog.recording', // recording notifications (pending, on, off, limits)
  862. // 'dialog.remoteControlTitle', // remote control notifications (allowed, denied, start, stop, error)
  863. // 'dialog.reservationError',
  864. // 'dialog.serviceUnavailable', // shown when server is not reachable
  865. // 'dialog.sessTerminated', // shown when there is a failed conference session
  866. // 'dialog.sessionRestarted', // show when a client reload is initiated because of bridge migration
  867. // 'dialog.tokenAuthFailed', // show when an invalid jwt is used
  868. // 'dialog.transcribing', // transcribing notifications (pending, off)
  869. // 'dialOut.statusMessage', // shown when dial out status is updated.
  870. // 'liveStreaming.busy', // shown when livestreaming service is busy
  871. // 'liveStreaming.failedToStart', // shown when livestreaming fails to start
  872. // 'liveStreaming.unavailableTitle', // shown when livestreaming service is not reachable
  873. // 'lobby.joinRejectedMessage', // shown when while in a lobby, user's request to join is rejected
  874. // 'lobby.notificationTitle', // shown when lobby is toggled and when join requests are allowed / denied
  875. // 'localRecording.localRecording', // shown when a local recording is started
  876. // 'notify.disconnected', // shown when a participant has left
  877. // 'notify.connectedOneMember', // show when a participant joined
  878. // 'notify.connectedTwoMembers', // show when two participants joined simultaneously
  879. // 'notify.connectedThreePlusMembers', // show when more than 2 participants joined simultaneously
  880. // 'notify.grantedTo', // shown when moderator rights were granted to a participant
  881. // 'notify.invitedOneMember', // shown when 1 participant has been invited
  882. // 'notify.invitedThreePlusMembers', // shown when 3+ participants have been invited
  883. // 'notify.invitedTwoMembers', // shown when 2 participants have been invited
  884. // 'notify.kickParticipant', // shown when a participant is kicked
  885. // 'notify.moderationStartedTitle', // shown when AV moderation is activated
  886. // 'notify.moderationStoppedTitle', // shown when AV moderation is deactivated
  887. // 'notify.moderationInEffectTitle', // shown when user attempts to unmute audio during AV moderation
  888. // 'notify.moderationInEffectVideoTitle', // shown when user attempts to enable video during AV moderation
  889. // 'notify.moderationInEffectCSTitle', // shown when user attempts to share content during AV moderation
  890. // 'notify.mutedRemotelyTitle', // shown when user is muted by a remote party
  891. // 'notify.mutedTitle', // shown when user has been muted upon joining,
  892. // 'notify.newDeviceAudioTitle', // prompts the user to use a newly detected audio device
  893. // 'notify.newDeviceCameraTitle', // prompts the user to use a newly detected camera
  894. // 'notify.passwordRemovedRemotely', // shown when a password has been removed remotely
  895. // 'notify.passwordSetRemotely', // shown when a password has been set remotely
  896. // 'notify.raisedHand', // shown when a partcipant used raise hand,
  897. // 'notify.startSilentTitle', // shown when user joined with no audio
  898. // 'notify.unmute', // shown to moderator when user raises hand during AV moderation
  899. // 'prejoin.errorDialOut',
  900. // 'prejoin.errorDialOutDisconnected',
  901. // 'prejoin.errorDialOutFailed',
  902. // 'prejoin.errorDialOutStatus',
  903. // 'prejoin.errorStatusCode',
  904. // 'prejoin.errorValidation',
  905. // 'recording.busy', // shown when recording service is busy
  906. // 'recording.failedToStart', // shown when recording fails to start
  907. // 'recording.unavailableTitle', // shown when recording service is not reachable
  908. // 'toolbar.noAudioSignalTitle', // shown when a broken mic is detected
  909. // 'toolbar.noisyAudioInputTitle', // shown when noise is detected for the current microphone
  910. // 'toolbar.talkWhileMutedPopup', // shown when user tries to speak while muted
  911. // 'transcribing.failedToStart' // shown when transcribing fails to start
  912. // ],
  913. // Prevent the filmstrip from autohiding when screen width is under a certain threshold
  914. // disableFilmstripAutohiding: false,
  915. // Specifies whether the chat emoticons are disabled or not
  916. // disableChatSmileys: false,
  917. // Allow all above example options to include a trailing comma and
  918. // prevent fear when commenting out the last value.
  919. makeJsonParserHappy: 'even if last key had a trailing comma'
  920. // no configuration value should follow this line.
  921. };
  922. /* eslint-enable no-unused-vars, no-var */