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config.js 21KB

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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. // Jirecon recording component domain.
  13. // jirecon: 'jirecon.jitsi-meet.example.com',
  14. // Call control component (Jigasi).
  15. // call_control: 'callcontrol.jitsi-meet.example.com',
  16. // Focus component domain. Defaults to focus.<domain>.
  17. // focus: 'focus.jitsi-meet.example.com',
  18. // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
  19. muc: 'conference.jitsi-meet.example.com'
  20. },
  21. // BOSH URL. FIXME: use XEP-0156 to discover it.
  22. bosh: '//jitsi-meet.example.com/http-bind',
  23. // Websocket URL
  24. // websocket: 'wss://jitsi-meet.example.com/xmpp-websocket',
  25. // The name of client node advertised in XEP-0115 'c' stanza
  26. clientNode: 'http://jitsi.org/jitsimeet',
  27. // The real JID of focus participant - can be overridden here
  28. // focusUserJid: 'focus@auth.jitsi-meet.example.com',
  29. // Testing / experimental features.
  30. //
  31. testing: {
  32. // P2P test mode disables automatic switching to P2P when there are 2
  33. // participants in the conference.
  34. p2pTestMode: false
  35. // Enables the test specific features consumed by jitsi-meet-torture
  36. // testMode: false
  37. // Disables the auto-play behavior of *all* newly created video element.
  38. // This is useful when the client runs on a host with limited resources.
  39. // noAutoPlayVideo: false
  40. // Enable / disable 500 Kbps bitrate cap on desktop tracks. When enabled,
  41. // simulcast is turned off for the desktop share. If presenter is turned
  42. // on while screensharing is in progress, the max bitrate is automatically
  43. // adjusted to 2.5 Mbps. This takes a value between 0 and 1 which determines
  44. // the probability for this to be enabled.
  45. // capScreenshareBitrate: 1 // 0 to disable
  46. },
  47. // Disables ICE/UDP by filtering out local and remote UDP candidates in
  48. // signalling.
  49. // webrtcIceUdpDisable: false,
  50. // Disables ICE/TCP by filtering out local and remote TCP candidates in
  51. // signalling.
  52. // webrtcIceTcpDisable: false,
  53. // Media
  54. //
  55. // Audio
  56. // Disable measuring of audio levels.
  57. // disableAudioLevels: false,
  58. // audioLevelsInterval: 200,
  59. // Enabling this will run the lib-jitsi-meet no audio detection module which
  60. // will notify the user if the current selected microphone has no audio
  61. // input and will suggest another valid device if one is present.
  62. enableNoAudioDetection: true,
  63. // Enabling this will run the lib-jitsi-meet noise detection module which will
  64. // notify the user if there is noise, other than voice, coming from the current
  65. // selected microphone. The purpose it to let the user know that the input could
  66. // be potentially unpleasant for other meeting participants.
  67. enableNoisyMicDetection: true,
  68. // Start the conference in audio only mode (no video is being received nor
  69. // sent).
  70. // startAudioOnly: false,
  71. // Every participant after the Nth will start audio muted.
  72. // startAudioMuted: 10,
  73. // Start calls with audio muted. Unlike the option above, this one is only
  74. // applied locally. FIXME: having these 2 options is confusing.
  75. // startWithAudioMuted: false,
  76. // Enabling it (with #params) will disable local audio output of remote
  77. // participants and to enable it back a reload is needed.
  78. // startSilent: false
  79. // Video
  80. // Sets the preferred resolution (height) for local video. Defaults to 720.
  81. // resolution: 720,
  82. // w3c spec-compliant video constraints to use for video capture. Currently
  83. // used by browsers that return true from lib-jitsi-meet's
  84. // util#browser#usesNewGumFlow. The constraints are independent from
  85. // this config's resolution value. Defaults to requesting an ideal
  86. // resolution of 720p.
  87. // constraints: {
  88. // video: {
  89. // height: {
  90. // ideal: 720,
  91. // max: 720,
  92. // min: 240
  93. // }
  94. // }
  95. // },
  96. // Enable / disable simulcast support.
  97. // disableSimulcast: false,
  98. // Enable / disable layer suspension. If enabled, endpoints whose HD
  99. // layers are not in use will be suspended (no longer sent) until they
  100. // are requested again.
  101. // enableLayerSuspension: false,
  102. // Every participant after the Nth will start video muted.
  103. // startVideoMuted: 10,
  104. // Start calls with video muted. Unlike the option above, this one is only
  105. // applied locally. FIXME: having these 2 options is confusing.
  106. // startWithVideoMuted: false,
  107. // If set to true, prefer to use the H.264 video codec (if supported).
  108. // Note that it's not recommended to do this because simulcast is not
  109. // supported when using H.264. For 1-to-1 calls this setting is enabled by
  110. // default and can be toggled in the p2p section.
  111. // preferH264: true,
  112. // If set to true, disable H.264 video codec by stripping it out of the
  113. // SDP.
  114. // disableH264: false,
  115. // Desktop sharing
  116. // The ID of the jidesha extension for Chrome.
  117. desktopSharingChromeExtId: null,
  118. // Whether desktop sharing should be disabled on Chrome.
  119. // desktopSharingChromeDisabled: false,
  120. // The media sources to use when using screen sharing with the Chrome
  121. // extension.
  122. desktopSharingChromeSources: [ 'screen', 'window', 'tab' ],
  123. // Required version of Chrome extension
  124. desktopSharingChromeMinExtVersion: '0.1',
  125. // Whether desktop sharing should be disabled on Firefox.
  126. // desktopSharingFirefoxDisabled: false,
  127. // Optional desktop sharing frame rate options. Default value: min:5, max:5.
  128. // desktopSharingFrameRate: {
  129. // min: 5,
  130. // max: 5
  131. // },
  132. // Try to start calls with screen-sharing instead of camera video.
  133. // startScreenSharing: false,
  134. // Recording
  135. // Whether to enable file recording or not.
  136. // fileRecordingsEnabled: false,
  137. // Enable the dropbox integration.
  138. // dropbox: {
  139. // appKey: '<APP_KEY>' // Specify your app key here.
  140. // // A URL to redirect the user to, after authenticating
  141. // // by default uses:
  142. // // 'https://jitsi-meet.example.com/static/oauth.html'
  143. // redirectURI:
  144. // 'https://jitsi-meet.example.com/subfolder/static/oauth.html'
  145. // },
  146. // When integrations like dropbox are enabled only that will be shown,
  147. // by enabling fileRecordingsServiceEnabled, we show both the integrations
  148. // and the generic recording service (its configuration and storage type
  149. // depends on jibri configuration)
  150. // fileRecordingsServiceEnabled: false,
  151. // Whether to show the possibility to share file recording with other people
  152. // (e.g. meeting participants), based on the actual implementation
  153. // on the backend.
  154. // fileRecordingsServiceSharingEnabled: false,
  155. // Whether to enable live streaming or not.
  156. // liveStreamingEnabled: false,
  157. // Transcription (in interface_config,
  158. // subtitles and buttons can be configured)
  159. // transcribingEnabled: false,
  160. // Enables automatic turning on captions when recording is started
  161. // autoCaptionOnRecord: false,
  162. // Misc
  163. // Default value for the channel "last N" attribute. -1 for unlimited.
  164. channelLastN: -1,
  165. // // Options for the recording limit notification.
  166. // recordingLimit: {
  167. //
  168. // // The recording limit in minutes. Note: This number appears in the notification text
  169. // // but doesn't enforce the actual recording time limit. This should be configured in
  170. // // jibri!
  171. // limit: 60,
  172. //
  173. // // The name of the app with unlimited recordings.
  174. // appName: 'Unlimited recordings APP',
  175. //
  176. // // The URL of the app with unlimited recordings.
  177. // appURL: 'https://unlimited.recordings.app.com/'
  178. // },
  179. // Disables or enables RTX (RFC 4588) (defaults to false).
  180. // disableRtx: false,
  181. // Disables or enables TCC (the default is in Jicofo and set to true)
  182. // (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting
  183. // affects congestion control, it practically enables send-side bandwidth
  184. // estimations.
  185. // enableTcc: true,
  186. // Disables or enables REMB (the default is in Jicofo and set to false)
  187. // (draft-alvestrand-rmcat-remb-03). This setting affects congestion
  188. // control, it practically enables recv-side bandwidth estimations. When
  189. // both TCC and REMB are enabled, TCC takes precedence. When both are
  190. // disabled, then bandwidth estimations are disabled.
  191. // enableRemb: false,
  192. // Enables ICE restart logic in LJM and displays the page reload overlay on
  193. // ICE failure. Current disabled by default because it's causing issues with
  194. // signaling when Octo is enabled. Also when we do an "ICE restart"(which is
  195. // not a real ICE restart), the client maintains the TCC sequence number
  196. // counter, but the bridge resets it. The bridge sends media packets with
  197. // TCC sequence numbers starting from 0.
  198. // enableIceRestart: false,
  199. // Defines the minimum number of participants to start a call (the default
  200. // is set in Jicofo and set to 2).
  201. // minParticipants: 2,
  202. // Use the TURN servers discovered via XEP-0215 for the jitsi-videobridge
  203. // connection
  204. // useStunTurn: true,
  205. // Use TURN/UDP servers for the jitsi-videobridge connection (by default
  206. // we filter out TURN/UDP because it is usually not needed since the
  207. // bridge itself is reachable via UDP)
  208. // useTurnUdp: false
  209. // Enable IPv6 support.
  210. // useIPv6: true,
  211. // Enables / disables a data communication channel with the Videobridge.
  212. // Values can be 'datachannel', 'websocket', true (treat it as
  213. // 'datachannel'), undefined (treat it as 'datachannel') and false (don't
  214. // open any channel).
  215. // openBridgeChannel: true,
  216. // UI
  217. //
  218. // Use display name as XMPP nickname.
  219. // useNicks: false,
  220. // Require users to always specify a display name.
  221. // requireDisplayName: true,
  222. // Whether to use a welcome page or not. In case it's false a random room
  223. // will be joined when no room is specified.
  224. enableWelcomePage: true,
  225. // Enabling the close page will ignore the welcome page redirection when
  226. // a call is hangup.
  227. // enableClosePage: false,
  228. // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
  229. // disable1On1Mode: false,
  230. // Default language for the user interface.
  231. // defaultLanguage: 'en',
  232. // If true all users without a token will be considered guests and all users
  233. // with token will be considered non-guests. Only guests will be allowed to
  234. // edit their profile.
  235. enableUserRolesBasedOnToken: false,
  236. // Whether or not some features are checked based on token.
  237. // enableFeaturesBasedOnToken: false,
  238. // Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests.
  239. // lockRoomGuestEnabled: false,
  240. // When enabled the password used for locking a room is restricted to up to the number of digits specified
  241. // roomPasswordNumberOfDigits: 10,
  242. // default: roomPasswordNumberOfDigits: false,
  243. // Message to show the users. Example: 'The service will be down for
  244. // maintenance at 01:00 AM GMT,
  245. // noticeMessage: '',
  246. // Enables calendar integration, depends on googleApiApplicationClientID
  247. // and microsoftApiApplicationClientID
  248. // enableCalendarIntegration: false,
  249. // When 'true', it shows an intermediate page before joining, where the user can configure its devices.
  250. // prejoinPageEnabled: false,
  251. // Stats
  252. //
  253. // Whether to enable stats collection or not in the TraceablePeerConnection.
  254. // This can be useful for debugging purposes (post-processing/analysis of
  255. // the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
  256. // estimation tests.
  257. // gatherStats: false,
  258. // The interval at which PeerConnection.getStats() is called. Defaults to 10000
  259. // pcStatsInterval: 10000,
  260. // To enable sending statistics to callstats.io you must provide the
  261. // Application ID and Secret.
  262. // callStatsID: '',
  263. // callStatsSecret: '',
  264. // enables sending participants display name to callstats
  265. // enableDisplayNameInStats: false,
  266. // enables sending participants email if available to callstats and other analytics
  267. // enableEmailInStats: false,
  268. // Privacy
  269. //
  270. // If third party requests are disabled, no other server will be contacted.
  271. // This means avatars will be locally generated and callstats integration
  272. // will not function.
  273. // disableThirdPartyRequests: false,
  274. // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
  275. //
  276. p2p: {
  277. // Enables peer to peer mode. When enabled the system will try to
  278. // establish a direct connection when there are exactly 2 participants
  279. // in the room. If that succeeds the conference will stop sending data
  280. // through the JVB and use the peer to peer connection instead. When a
  281. // 3rd participant joins the conference will be moved back to the JVB
  282. // connection.
  283. enabled: true,
  284. // Use XEP-0215 to fetch STUN and TURN servers.
  285. // useStunTurn: true,
  286. // The STUN servers that will be used in the peer to peer connections
  287. stunServers: [
  288. // { urls: 'stun:jitsi-meet.example.com:4446' },
  289. { urls: 'stun:meet-jit-si-turnrelay.jitsi.net:443' }
  290. ]
  291. // Sets the ICE transport policy for the p2p connection. At the time
  292. // of this writing the list of possible values are 'all' and 'relay',
  293. // but that is subject to change in the future. The enum is defined in
  294. // the WebRTC standard:
  295. // https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
  296. // If not set, the effective value is 'all'.
  297. // iceTransportPolicy: 'all',
  298. // If set to true, it will prefer to use H.264 for P2P calls (if H.264
  299. // is supported).
  300. // preferH264: true
  301. // If set to true, disable H.264 video codec by stripping it out of the
  302. // SDP.
  303. // disableH264: false,
  304. // How long we're going to wait, before going back to P2P after the 3rd
  305. // participant has left the conference (to filter out page reload).
  306. // backToP2PDelay: 5
  307. },
  308. analytics: {
  309. // The Google Analytics Tracking ID:
  310. // googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
  311. // Matomo configuration:
  312. // matomoEndpoint: 'https://your-matomo-endpoint/',
  313. // matomoSiteID: '42',
  314. // The Amplitude APP Key:
  315. // amplitudeAPPKey: '<APP_KEY>'
  316. // Array of script URLs to load as lib-jitsi-meet "analytics handlers".
  317. // scriptURLs: [
  318. // "libs/analytics-ga.min.js", // google-analytics
  319. // "https://example.com/my-custom-analytics.js"
  320. // ],
  321. },
  322. // Information about the jitsi-meet instance we are connecting to, including
  323. // the user region as seen by the server.
  324. deploymentInfo: {
  325. // shard: "shard1",
  326. // region: "europe",
  327. // userRegion: "asia"
  328. },
  329. // Decides whether the start/stop recording audio notifications should play on record.
  330. // disableRecordAudioNotification: false,
  331. // Information for the chrome extension banner
  332. // chromeExtensionBanner: {
  333. // // The chrome extension to be installed address
  334. // url: 'https://chrome.google.com/webstore/detail/jitsi-meetings/kglhbbefdnlheedjiejgomgmfplipfeb',
  335. // // Extensions info which allows checking if they are installed or not
  336. // chromeExtensionsInfo: [
  337. // {
  338. // id: 'kglhbbefdnlheedjiejgomgmfplipfeb',
  339. // path: 'jitsi-logo-48x48.png'
  340. // }
  341. // ]
  342. // },
  343. // Local Recording
  344. //
  345. // localRecording: {
  346. // Enables local recording.
  347. // Additionally, 'localrecording' (all lowercase) needs to be added to
  348. // TOOLBAR_BUTTONS in interface_config.js for the Local Recording
  349. // button to show up on the toolbar.
  350. //
  351. // enabled: true,
  352. //
  353. // The recording format, can be one of 'ogg', 'flac' or 'wav'.
  354. // format: 'flac'
  355. //
  356. // },
  357. // Options related to end-to-end (participant to participant) ping.
  358. // e2eping: {
  359. // // The interval in milliseconds at which pings will be sent.
  360. // // Defaults to 10000, set to <= 0 to disable.
  361. // pingInterval: 10000,
  362. //
  363. // // The interval in milliseconds at which analytics events
  364. // // with the measured RTT will be sent. Defaults to 60000, set
  365. // // to <= 0 to disable.
  366. // analyticsInterval: 60000,
  367. // },
  368. // If set, will attempt to use the provided video input device label when
  369. // triggering a screenshare, instead of proceeding through the normal flow
  370. // for obtaining a desktop stream.
  371. // NOTE: This option is experimental and is currently intended for internal
  372. // use only.
  373. // _desktopSharingSourceDevice: 'sample-id-or-label',
  374. // If true, any checks to handoff to another application will be prevented
  375. // and instead the app will continue to display in the current browser.
  376. // disableDeepLinking: false,
  377. // A property to disable the right click context menu for localVideo
  378. // the menu has option to flip the locally seen video for local presentations
  379. // disableLocalVideoFlip: false,
  380. // Mainly privacy related settings
  381. // Disables all invite functions from the app (share, invite, dial out...etc)
  382. // disableInviteFunctions: true,
  383. // Disables storing the room name to the recents list
  384. // doNotStoreRoom: true,
  385. // Deployment specific URLs.
  386. // deploymentUrls: {
  387. // // If specified a 'Help' button will be displayed in the overflow menu with a link to the specified URL for
  388. // // user documentation.
  389. // userDocumentationURL: 'https://docs.example.com/video-meetings.html',
  390. // // If specified a 'Download our apps' button will be displayed in the overflow menu with a link
  391. // // to the specified URL for an app download page.
  392. // downloadAppsUrl: 'https://docs.example.com/our-apps.html'
  393. // },
  394. // Options related to the remote participant menu.
  395. // remoteVideoMenu: {
  396. // // If set to true the 'Kick out' button will be disabled.
  397. // disableKick: true
  398. // },
  399. // If set to true all muting operations of remote participants will be disabled.
  400. // disableRemoteMute: true,
  401. /**
  402. External API url used to receive branding specific information.
  403. If there is no url set or there are missing fields, the defaults are applied.
  404. None of the fieds are mandatory and the response must have the shape:
  405. {
  406. // The hex value for the colour used as background
  407. backgroundColor: '#fff',
  408. // The url for the image used as background
  409. backgroundImageUrl: 'https://example.com/background-img.png',
  410. // The anchor url used when clicking the logo image
  411. logoClickUrl: 'https://example-company.org',
  412. // The url used for the image used as logo
  413. logoImageUrl: 'https://example.com/logo-img.png'
  414. }
  415. */
  416. // brandingDataUrl: '',
  417. // List of undocumented settings used in jitsi-meet
  418. /**
  419. _immediateReloadThreshold
  420. autoRecord
  421. autoRecordToken
  422. debug
  423. debugAudioLevels
  424. deploymentInfo
  425. dialInConfCodeUrl
  426. dialInNumbersUrl
  427. dialOutAuthUrl
  428. dialOutCodesUrl
  429. disableRemoteControl
  430. displayJids
  431. etherpad_base
  432. externalConnectUrl
  433. firefox_fake_device
  434. googleApiApplicationClientID
  435. iAmRecorder
  436. iAmSipGateway
  437. microsoftApiApplicationClientID
  438. peopleSearchQueryTypes
  439. peopleSearchUrl
  440. requireDisplayName
  441. tokenAuthUrl
  442. */
  443. // List of undocumented settings used in lib-jitsi-meet
  444. /**
  445. _peerConnStatusOutOfLastNTimeout
  446. _peerConnStatusRtcMuteTimeout
  447. abTesting
  448. avgRtpStatsN
  449. callStatsConfIDNamespace
  450. callStatsCustomScriptUrl
  451. desktopSharingSources
  452. disableAEC
  453. disableAGC
  454. disableAP
  455. disableHPF
  456. disableNS
  457. enableLipSync
  458. enableTalkWhileMuted
  459. forceJVB121Ratio
  460. hiddenDomain
  461. ignoreStartMuted
  462. nick
  463. startBitrate
  464. */
  465. // Allow all above example options to include a trailing comma and
  466. // prevent fear when commenting out the last value.
  467. makeJsonParserHappy: 'even if last key had a trailing comma'
  468. // no configuration value should follow this line.
  469. };
  470. /* eslint-enable no-unused-vars, no-var */