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config.js 18KB

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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. // Enables experimental simulcast support on Firefox.
  33. enableFirefoxSimulcast: false,
  34. // P2P test mode disables automatic switching to P2P when there are 2
  35. // participants in the conference.
  36. p2pTestMode: false
  37. // Enables the test specific features consumed by jitsi-meet-torture
  38. // testMode: false
  39. // Disables the auto-play behavior of *all* newly created video element.
  40. // This is useful when the client runs on a host with limited resources.
  41. // noAutoPlayVideo: false
  42. },
  43. // Disables ICE/UDP by filtering out local and remote UDP candidates in
  44. // signalling.
  45. // webrtcIceUdpDisable: false,
  46. // Disables ICE/TCP by filtering out local and remote TCP candidates in
  47. // signalling.
  48. // webrtcIceTcpDisable: false,
  49. // Media
  50. //
  51. // Audio
  52. // Disable measuring of audio levels.
  53. // disableAudioLevels: false,
  54. // audioLevelsInterval: 200,
  55. // Enabling this will run the lib-jitsi-meet no audio detection module which
  56. // will notify the user if the current selected microphone has no audio
  57. // input and will suggest another valid device if one is present.
  58. enableNoAudioDetection: true,
  59. // Enabling this will run the lib-jitsi-meet noise detection module which will
  60. // notify the user if there is noise, other than voice, coming from the current
  61. // selected microphone. The purpose it to let the user know that the input could
  62. // be potentially unpleasant for other meeting participants.
  63. enableNoisyMicDetection: true,
  64. // Start the conference in audio only mode (no video is being received nor
  65. // sent).
  66. // startAudioOnly: false,
  67. // Every participant after the Nth will start audio muted.
  68. // startAudioMuted: 10,
  69. // Start calls with audio muted. Unlike the option above, this one is only
  70. // applied locally. FIXME: having these 2 options is confusing.
  71. // startWithAudioMuted: false,
  72. // Enabling it (with #params) will disable local audio output of remote
  73. // participants and to enable it back a reload is needed.
  74. // startSilent: false
  75. // Video
  76. // Sets the preferred resolution (height) for local video. Defaults to 720.
  77. // resolution: 720,
  78. // w3c spec-compliant video constraints to use for video capture. Currently
  79. // used by browsers that return true from lib-jitsi-meet's
  80. // util#browser#usesNewGumFlow. The constraints are independent from
  81. // this config's resolution value. Defaults to requesting an ideal aspect
  82. // ratio of 16:9 with an ideal resolution of 720.
  83. // constraints: {
  84. // video: {
  85. // aspectRatio: 16 / 9,
  86. // height: {
  87. // ideal: 720,
  88. // max: 720,
  89. // min: 240
  90. // }
  91. // }
  92. // },
  93. // Enable / disable simulcast support.
  94. // disableSimulcast: false,
  95. // Enable / disable layer suspension. If enabled, endpoints whose HD
  96. // layers are not in use will be suspended (no longer sent) until they
  97. // are requested again.
  98. // enableLayerSuspension: false,
  99. // Every participant after the Nth will start video muted.
  100. // startVideoMuted: 10,
  101. // Start calls with video muted. Unlike the option above, this one is only
  102. // applied locally. FIXME: having these 2 options is confusing.
  103. // startWithVideoMuted: false,
  104. // If set to true, prefer to use the H.264 video codec (if supported).
  105. // Note that it's not recommended to do this because simulcast is not
  106. // supported when using H.264. For 1-to-1 calls this setting is enabled by
  107. // default and can be toggled in the p2p section.
  108. // preferH264: true,
  109. // If set to true, disable H.264 video codec by stripping it out of the
  110. // SDP.
  111. // disableH264: false,
  112. // Desktop sharing
  113. // The ID of the jidesha extension for Chrome.
  114. desktopSharingChromeExtId: null,
  115. // Whether desktop sharing should be disabled on Chrome.
  116. // desktopSharingChromeDisabled: false,
  117. // The media sources to use when using screen sharing with the Chrome
  118. // extension.
  119. desktopSharingChromeSources: [ 'screen', 'window', 'tab' ],
  120. // Required version of Chrome extension
  121. desktopSharingChromeMinExtVersion: '0.1',
  122. // Whether desktop sharing should be disabled on Firefox.
  123. // desktopSharingFirefoxDisabled: false,
  124. // Optional desktop sharing frame rate options. Default value: min:5, max:5.
  125. // desktopSharingFrameRate: {
  126. // min: 5,
  127. // max: 5
  128. // },
  129. // Try to start calls with screen-sharing instead of camera video.
  130. // startScreenSharing: false,
  131. // Recording
  132. // Whether to enable file recording or not.
  133. // fileRecordingsEnabled: false,
  134. // Enable the dropbox integration.
  135. // dropbox: {
  136. // appKey: '<APP_KEY>' // Specify your app key here.
  137. // // A URL to redirect the user to, after authenticating
  138. // // by default uses:
  139. // // 'https://jitsi-meet.example.com/static/oauth.html'
  140. // redirectURI:
  141. // 'https://jitsi-meet.example.com/subfolder/static/oauth.html'
  142. // },
  143. // When integrations like dropbox are enabled only that will be shown,
  144. // by enabling fileRecordingsServiceEnabled, we show both the integrations
  145. // and the generic recording service (its configuration and storage type
  146. // depends on jibri configuration)
  147. // fileRecordingsServiceEnabled: false,
  148. // Whether to show the possibility to share file recording with other people
  149. // (e.g. meeting participants), based on the actual implementation
  150. // on the backend.
  151. // fileRecordingsServiceSharingEnabled: false,
  152. // Whether to enable live streaming or not.
  153. // liveStreamingEnabled: false,
  154. // Transcription (in interface_config,
  155. // subtitles and buttons can be configured)
  156. // transcribingEnabled: false,
  157. // Enables automatic turning on captions when recording is started
  158. // autoCaptionOnRecord: false,
  159. // Misc
  160. // Default value for the channel "last N" attribute. -1 for unlimited.
  161. channelLastN: -1,
  162. // Disables or enables RTX (RFC 4588) (defaults to false).
  163. // disableRtx: false,
  164. // Disables or enables TCC (the default is in Jicofo and set to true)
  165. // (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting
  166. // affects congestion control, it practically enables send-side bandwidth
  167. // estimations.
  168. // enableTcc: true,
  169. // Disables or enables REMB (the default is in Jicofo and set to false)
  170. // (draft-alvestrand-rmcat-remb-03). This setting affects congestion
  171. // control, it practically enables recv-side bandwidth estimations. When
  172. // both TCC and REMB are enabled, TCC takes precedence. When both are
  173. // disabled, then bandwidth estimations are disabled.
  174. // enableRemb: false,
  175. // Defines the minimum number of participants to start a call (the default
  176. // is set in Jicofo and set to 2).
  177. // minParticipants: 2,
  178. // Use XEP-0215 to fetch STUN and TURN servers.
  179. // useStunTurn: true,
  180. // Enable IPv6 support.
  181. // useIPv6: true,
  182. // Enables / disables a data communication channel with the Videobridge.
  183. // Values can be 'datachannel', 'websocket', true (treat it as
  184. // 'datachannel'), undefined (treat it as 'datachannel') and false (don't
  185. // open any channel).
  186. // openBridgeChannel: true,
  187. // UI
  188. //
  189. // Use display name as XMPP nickname.
  190. // useNicks: false,
  191. // Require users to always specify a display name.
  192. // requireDisplayName: true,
  193. // Whether to use a welcome page or not. In case it's false a random room
  194. // will be joined when no room is specified.
  195. enableWelcomePage: true,
  196. // Enabling the close page will ignore the welcome page redirection when
  197. // a call is hangup.
  198. // enableClosePage: false,
  199. // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
  200. // disable1On1Mode: false,
  201. // Default language for the user interface.
  202. // defaultLanguage: 'en',
  203. // If true all users without a token will be considered guests and all users
  204. // with token will be considered non-guests. Only guests will be allowed to
  205. // edit their profile.
  206. enableUserRolesBasedOnToken: false,
  207. // Whether or not some features are checked based on token.
  208. // enableFeaturesBasedOnToken: false,
  209. // Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests.
  210. // lockRoomGuestEnabled: false,
  211. // When enabled the password used for locking a room is restricted to up to the number of digits specified
  212. // roomPasswordNumberOfDigits: 10,
  213. // default: roomPasswordNumberOfDigits: false,
  214. // Message to show the users. Example: 'The service will be down for
  215. // maintenance at 01:00 AM GMT,
  216. // noticeMessage: '',
  217. // Enables calendar integration, depends on googleApiApplicationClientID
  218. // and microsoftApiApplicationClientID
  219. // enableCalendarIntegration: false,
  220. // Stats
  221. //
  222. // Whether to enable stats collection or not in the TraceablePeerConnection.
  223. // This can be useful for debugging purposes (post-processing/analysis of
  224. // the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
  225. // estimation tests.
  226. // gatherStats: false,
  227. // The interval at which PeerConnection.getStats() is called. Defaults to 10000
  228. // pcStatsInterval: 10000,
  229. // To enable sending statistics to callstats.io you must provide the
  230. // Application ID and Secret.
  231. // callStatsID: '',
  232. // callStatsSecret: '',
  233. // enables sending participants display name to callstats
  234. // enableDisplayNameInStats: false,
  235. // enables sending participants email if available to callstats and other analytics
  236. // enableEmailInStats: false,
  237. // Privacy
  238. //
  239. // If third party requests are disabled, no other server will be contacted.
  240. // This means avatars will be locally generated and callstats integration
  241. // will not function.
  242. // disableThirdPartyRequests: false,
  243. // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
  244. //
  245. p2p: {
  246. // Enables peer to peer mode. When enabled the system will try to
  247. // establish a direct connection when there are exactly 2 participants
  248. // in the room. If that succeeds the conference will stop sending data
  249. // through the JVB and use the peer to peer connection instead. When a
  250. // 3rd participant joins the conference will be moved back to the JVB
  251. // connection.
  252. enabled: true,
  253. // Use XEP-0215 to fetch STUN and TURN servers.
  254. // useStunTurn: true,
  255. // The STUN servers that will be used in the peer to peer connections
  256. stunServers: [
  257. // { urls: 'stun:jitsi-meet.example.com:443' },
  258. { urls: 'stun:stun.l.google.com:19302' },
  259. { urls: 'stun:stun1.l.google.com:19302' },
  260. { urls: 'stun:stun2.l.google.com:19302' }
  261. ],
  262. // Sets the ICE transport policy for the p2p connection. At the time
  263. // of this writing the list of possible values are 'all' and 'relay',
  264. // but that is subject to change in the future. The enum is defined in
  265. // the WebRTC standard:
  266. // https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
  267. // If not set, the effective value is 'all'.
  268. // iceTransportPolicy: 'all',
  269. // If set to true, it will prefer to use H.264 for P2P calls (if H.264
  270. // is supported).
  271. preferH264: true
  272. // If set to true, disable H.264 video codec by stripping it out of the
  273. // SDP.
  274. // disableH264: false,
  275. // How long we're going to wait, before going back to P2P after the 3rd
  276. // participant has left the conference (to filter out page reload).
  277. // backToP2PDelay: 5
  278. },
  279. analytics: {
  280. // The Google Analytics Tracking ID:
  281. // googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
  282. // The Amplitude APP Key:
  283. // amplitudeAPPKey: '<APP_KEY>'
  284. // Array of script URLs to load as lib-jitsi-meet "analytics handlers".
  285. // scriptURLs: [
  286. // "libs/analytics-ga.min.js", // google-analytics
  287. // "https://example.com/my-custom-analytics.js"
  288. // ],
  289. },
  290. // Information about the jitsi-meet instance we are connecting to, including
  291. // the user region as seen by the server.
  292. deploymentInfo: {
  293. // shard: "shard1",
  294. // region: "europe",
  295. // userRegion: "asia"
  296. },
  297. // Information for the chrome extension banner
  298. // chromeExtensionBanner: {
  299. // // The chrome extension to be installed address
  300. // url: 'https://chrome.google.com/webstore/detail/jitsi-meetings/kglhbbefdnlheedjiejgomgmfplipfeb',
  301. // // Extensions info which allows checking if they are installed or not
  302. // chromeExtensionsInfo: [
  303. // {
  304. // id: 'kglhbbefdnlheedjiejgomgmfplipfeb',
  305. // path: 'jitsi-logo-48x48.png'
  306. // }
  307. // ]
  308. // },
  309. // Local Recording
  310. //
  311. // localRecording: {
  312. // Enables local recording.
  313. // Additionally, 'localrecording' (all lowercase) needs to be added to
  314. // TOOLBAR_BUTTONS in interface_config.js for the Local Recording
  315. // button to show up on the toolbar.
  316. //
  317. // enabled: true,
  318. //
  319. // The recording format, can be one of 'ogg', 'flac' or 'wav'.
  320. // format: 'flac'
  321. //
  322. // },
  323. // Options related to end-to-end (participant to participant) ping.
  324. // e2eping: {
  325. // // The interval in milliseconds at which pings will be sent.
  326. // // Defaults to 10000, set to <= 0 to disable.
  327. // pingInterval: 10000,
  328. //
  329. // // The interval in milliseconds at which analytics events
  330. // // with the measured RTT will be sent. Defaults to 60000, set
  331. // // to <= 0 to disable.
  332. // analyticsInterval: 60000,
  333. // },
  334. // If set, will attempt to use the provided video input device label when
  335. // triggering a screenshare, instead of proceeding through the normal flow
  336. // for obtaining a desktop stream.
  337. // NOTE: This option is experimental and is currently intended for internal
  338. // use only.
  339. // _desktopSharingSourceDevice: 'sample-id-or-label',
  340. // If true, any checks to handoff to another application will be prevented
  341. // and instead the app will continue to display in the current browser.
  342. // disableDeepLinking: false,
  343. // A property to disable the right click context menu for localVideo
  344. // the menu has option to flip the locally seen video for local presentations
  345. // disableLocalVideoFlip: false,
  346. // Deployment specific URLs.
  347. // deploymentUrls: {
  348. // // If specified a 'Help' button will be displayed in the overflow menu with a link to the specified URL for
  349. // // user documentation.
  350. // userDocumentationURL: 'https://docs.example.com/video-meetings.html',
  351. // // If specified a 'Download our apps' button will be displayed in the overflow menu with a link
  352. // // to the specified URL for an app download page.
  353. // downloadAppsUrl: 'https://docs.example.com/our-apps.html'
  354. // },
  355. // List of undocumented settings used in jitsi-meet
  356. /**
  357. _immediateReloadThreshold
  358. autoRecord
  359. autoRecordToken
  360. debug
  361. debugAudioLevels
  362. deploymentInfo
  363. dialInConfCodeUrl
  364. dialInNumbersUrl
  365. dialOutAuthUrl
  366. dialOutCodesUrl
  367. disableRemoteControl
  368. displayJids
  369. etherpad_base
  370. externalConnectUrl
  371. firefox_fake_device
  372. googleApiApplicationClientID
  373. iAmRecorder
  374. iAmSipGateway
  375. microsoftApiApplicationClientID
  376. peopleSearchQueryTypes
  377. peopleSearchUrl
  378. requireDisplayName
  379. tokenAuthUrl
  380. */
  381. // List of undocumented settings used in lib-jitsi-meet
  382. /**
  383. _peerConnStatusOutOfLastNTimeout
  384. _peerConnStatusRtcMuteTimeout
  385. abTesting
  386. avgRtpStatsN
  387. callStatsConfIDNamespace
  388. callStatsCustomScriptUrl
  389. desktopSharingSources
  390. disableAEC
  391. disableAGC
  392. disableAP
  393. disableHPF
  394. disableNS
  395. enableLipSync
  396. enableTalkWhileMuted
  397. forceJVB121Ratio
  398. hiddenDomain
  399. ignoreStartMuted
  400. nick
  401. startBitrate
  402. */
  403. // Allow all above example options to include a trailing comma and
  404. // prevent fear when commenting out the last value.
  405. makeJsonParserHappy: 'even if last key had a trailing comma'
  406. // no configuration value should follow this line.
  407. };
  408. /* eslint-enable no-unused-vars, no-var */