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config.js 41KB

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