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config.js 19KB

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