You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.

config.js 20KB

123456789101112131415161718192021222324252627282930313233343536373839404142434445464748495051525354555657585960616263646566676869707172737475767778798081828384858687888990919293949596979899100101102103104105106107108109110111112113114115116117118119120121122123124125126127128129130131132133134135136137138139140141142143144145146147148149150151152153154155156157158159160161162163164165166167168169170171172173174175176177178179180181182183184185186187188189190191192193194195196197198199200201202203204205206207208209210211212213214215216217218219220221222223224225226227228229230231232233234235236237238239240241242243244245246247248249250251252253254255256257258259260261262263264265266267268269270271272273274275276277278279280281282283284285286287288289290291292293294295296297298299300301302303304305306307308309310311312313314315316317318319320321322323324325326327328329330331332333334335336337338339340341342343344345346347348349350351352353354355356357358359360361362363364365366367368369370371372373374375376377378379380381382383384385386387388389390391392393394395396397398399400401402403404405406407408409410411412413414415416417418419420421422423424425426427428429430431432433434435436437438439440441442443444445446447448449450451452453454455456457458459460461462463464465466467468469470471472473474475476477478479480481482483484485486487488489490491492493494495496497498499500501502503504505506507508509510511512513514515516517518519520521522523524525526527528529530531532533534535536537538539540541542543544545546547548549550551552553554555556557558559560561562563564565566567568569570571572573574575576577578579
  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. // Optional desktop sharing frame rate options. Default value: min:5, max:5.
  117. // desktopSharingFrameRate: {
  118. // min: 5,
  119. // max: 5
  120. // },
  121. // Try to start calls with screen-sharing instead of camera video.
  122. // startScreenSharing: false,
  123. // Recording
  124. // Whether to enable file recording or not.
  125. // fileRecordingsEnabled: false,
  126. // Enable the dropbox integration.
  127. // dropbox: {
  128. // appKey: '<APP_KEY>' // Specify your app key here.
  129. // // A URL to redirect the user to, after authenticating
  130. // // by default uses:
  131. // // 'https://jitsi-meet.example.com/static/oauth.html'
  132. // redirectURI:
  133. // 'https://jitsi-meet.example.com/subfolder/static/oauth.html'
  134. // },
  135. // When integrations like dropbox are enabled only that will be shown,
  136. // by enabling fileRecordingsServiceEnabled, we show both the integrations
  137. // and the generic recording service (its configuration and storage type
  138. // depends on jibri configuration)
  139. // fileRecordingsServiceEnabled: false,
  140. // Whether to show the possibility to share file recording with other people
  141. // (e.g. meeting participants), based on the actual implementation
  142. // on the backend.
  143. // fileRecordingsServiceSharingEnabled: false,
  144. // Whether to enable live streaming or not.
  145. // liveStreamingEnabled: false,
  146. // Transcription (in interface_config,
  147. // subtitles and buttons can be configured)
  148. // transcribingEnabled: false,
  149. // Enables automatic turning on captions when recording is started
  150. // autoCaptionOnRecord: false,
  151. // Misc
  152. // Default value for the channel "last N" attribute. -1 for unlimited.
  153. channelLastN: -1,
  154. // // Options for the recording limit notification.
  155. // recordingLimit: {
  156. //
  157. // // The recording limit in minutes. Note: This number appears in the notification text
  158. // // but doesn't enforce the actual recording time limit. This should be configured in
  159. // // jibri!
  160. // limit: 60,
  161. //
  162. // // The name of the app with unlimited recordings.
  163. // appName: 'Unlimited recordings APP',
  164. //
  165. // // The URL of the app with unlimited recordings.
  166. // appURL: 'https://unlimited.recordings.app.com/'
  167. // },
  168. // Disables or enables RTX (RFC 4588) (defaults to false).
  169. // disableRtx: false,
  170. // Disables or enables TCC (the default is in Jicofo and set to true)
  171. // (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting
  172. // affects congestion control, it practically enables send-side bandwidth
  173. // estimations.
  174. // enableTcc: true,
  175. // Disables or enables REMB (the default is in Jicofo and set to false)
  176. // (draft-alvestrand-rmcat-remb-03). This setting affects congestion
  177. // control, it practically enables recv-side bandwidth estimations. When
  178. // both TCC and REMB are enabled, TCC takes precedence. When both are
  179. // disabled, then bandwidth estimations are disabled.
  180. // enableRemb: false,
  181. // Enables ICE restart logic in LJM and displays the page reload overlay on
  182. // ICE failure. Current disabled by default because it's causing issues with
  183. // signaling when Octo is enabled. Also when we do an "ICE restart"(which is
  184. // not a real ICE restart), the client maintains the TCC sequence number
  185. // counter, but the bridge resets it. The bridge sends media packets with
  186. // TCC sequence numbers starting from 0.
  187. // enableIceRestart: false,
  188. // Defines the minimum number of participants to start a call (the default
  189. // is set in Jicofo and set to 2).
  190. // minParticipants: 2,
  191. // Use the TURN servers discovered via XEP-0215 for the jitsi-videobridge
  192. // connection
  193. // useStunTurn: true,
  194. // Use TURN/UDP servers for the jitsi-videobridge connection (by default
  195. // we filter out TURN/UDP because it is usually not needed since the
  196. // bridge itself is reachable via UDP)
  197. // useTurnUdp: false
  198. // Enables / disables a data communication channel with the Videobridge.
  199. // Values can be 'datachannel', 'websocket', true (treat it as
  200. // 'datachannel'), undefined (treat it as 'datachannel') and false (don't
  201. // open any channel).
  202. // openBridgeChannel: true,
  203. // UI
  204. //
  205. // Require users to always specify a display name.
  206. // requireDisplayName: true,
  207. // Whether to use a welcome page or not. In case it's false a random room
  208. // will be joined when no room is specified.
  209. enableWelcomePage: true,
  210. // Enabling the close page will ignore the welcome page redirection when
  211. // a call is hangup.
  212. // enableClosePage: false,
  213. // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
  214. // disable1On1Mode: false,
  215. // Default language for the user interface.
  216. // defaultLanguage: 'en',
  217. // If true all users without a token will be considered guests and all users
  218. // with token will be considered non-guests. Only guests will be allowed to
  219. // edit their profile.
  220. enableUserRolesBasedOnToken: false,
  221. // Whether or not some features are checked based on token.
  222. // enableFeaturesBasedOnToken: false,
  223. // Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests.
  224. // lockRoomGuestEnabled: false,
  225. // When enabled the password used for locking a room is restricted to up to the number of digits specified
  226. // roomPasswordNumberOfDigits: 10,
  227. // default: roomPasswordNumberOfDigits: false,
  228. // Message to show the users. Example: 'The service will be down for
  229. // maintenance at 01:00 AM GMT,
  230. // noticeMessage: '',
  231. // Enables calendar integration, depends on googleApiApplicationClientID
  232. // and microsoftApiApplicationClientID
  233. // enableCalendarIntegration: false,
  234. // When 'true', it shows an intermediate page before joining, where the user can configure their devices.
  235. // prejoinPageEnabled: false,
  236. // If true, shows the unsafe roon name warning label when a room name is
  237. // deemed unsafe (due to the simplicity in the name) and a password is not
  238. // set or the lobby is not enabled.
  239. // enableInsecureRoomNameWarning: false,
  240. // Stats
  241. //
  242. // Whether to enable stats collection or not in the TraceablePeerConnection.
  243. // This can be useful for debugging purposes (post-processing/analysis of
  244. // the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
  245. // estimation tests.
  246. // gatherStats: false,
  247. // The interval at which PeerConnection.getStats() is called. Defaults to 10000
  248. // pcStatsInterval: 10000,
  249. // To enable sending statistics to callstats.io you must provide the
  250. // Application ID and Secret.
  251. // callStatsID: '',
  252. // callStatsSecret: '',
  253. // Enables sending participants' display names to callstats
  254. // enableDisplayNameInStats: false,
  255. // Enables sending participants' emails (if available) to callstats and other analytics
  256. // enableEmailInStats: false,
  257. // Privacy
  258. //
  259. // If third party requests are disabled, no other server will be contacted.
  260. // This means avatars will be locally generated and callstats integration
  261. // will not function.
  262. // disableThirdPartyRequests: false,
  263. // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
  264. //
  265. p2p: {
  266. // Enables peer to peer mode. When enabled the system will try to
  267. // establish a direct connection when there are exactly 2 participants
  268. // in the room. If that succeeds the conference will stop sending data
  269. // through the JVB and use the peer to peer connection instead. When a
  270. // 3rd participant joins the conference will be moved back to the JVB
  271. // connection.
  272. enabled: true,
  273. // Use XEP-0215 to fetch STUN and TURN servers.
  274. // useStunTurn: true,
  275. // The STUN servers that will be used in the peer to peer connections
  276. stunServers: [
  277. // { urls: 'stun:jitsi-meet.example.com:4446' },
  278. { urls: 'stun:meet-jit-si-turnrelay.jitsi.net:443' }
  279. ]
  280. // Sets the ICE transport policy for the p2p connection. At the time
  281. // of this writing the list of possible values are 'all' and 'relay',
  282. // but that is subject to change in the future. The enum is defined in
  283. // the WebRTC standard:
  284. // https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
  285. // If not set, the effective value is 'all'.
  286. // iceTransportPolicy: 'all',
  287. // If set to true, it will prefer to use H.264 for P2P calls (if H.264
  288. // is supported).
  289. // preferH264: true
  290. // If set to true, disable H.264 video codec by stripping it out of the
  291. // SDP.
  292. // disableH264: false,
  293. // How long we're going to wait, before going back to P2P after the 3rd
  294. // participant has left the conference (to filter out page reload).
  295. // backToP2PDelay: 5
  296. },
  297. analytics: {
  298. // The Google Analytics Tracking ID:
  299. // googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
  300. // Matomo configuration:
  301. // matomoEndpoint: 'https://your-matomo-endpoint/',
  302. // matomoSiteID: '42',
  303. // The Amplitude APP Key:
  304. // amplitudeAPPKey: '<APP_KEY>'
  305. // Array of script URLs to load as lib-jitsi-meet "analytics handlers".
  306. // scriptURLs: [
  307. // "libs/analytics-ga.min.js", // google-analytics
  308. // "https://example.com/my-custom-analytics.js"
  309. // ],
  310. },
  311. // Information about the jitsi-meet instance we are connecting to, including
  312. // the user region as seen by the server.
  313. deploymentInfo: {
  314. // shard: "shard1",
  315. // region: "europe",
  316. // userRegion: "asia"
  317. },
  318. // Decides whether the start/stop recording audio notifications should play on record.
  319. // disableRecordAudioNotification: false,
  320. // Information for the chrome extension banner
  321. // chromeExtensionBanner: {
  322. // // The chrome extension to be installed address
  323. // url: 'https://chrome.google.com/webstore/detail/jitsi-meetings/kglhbbefdnlheedjiejgomgmfplipfeb',
  324. // // Extensions info which allows checking if they are installed or not
  325. // chromeExtensionsInfo: [
  326. // {
  327. // id: 'kglhbbefdnlheedjiejgomgmfplipfeb',
  328. // path: 'jitsi-logo-48x48.png'
  329. // }
  330. // ]
  331. // },
  332. // Local Recording
  333. //
  334. // localRecording: {
  335. // Enables local recording.
  336. // Additionally, 'localrecording' (all lowercase) needs to be added to
  337. // TOOLBAR_BUTTONS in interface_config.js for the Local Recording
  338. // button to show up on the toolbar.
  339. //
  340. // enabled: true,
  341. //
  342. // The recording format, can be one of 'ogg', 'flac' or 'wav'.
  343. // format: 'flac'
  344. //
  345. // },
  346. // Options related to end-to-end (participant to participant) ping.
  347. // e2eping: {
  348. // // The interval in milliseconds at which pings will be sent.
  349. // // Defaults to 10000, set to <= 0 to disable.
  350. // pingInterval: 10000,
  351. //
  352. // // The interval in milliseconds at which analytics events
  353. // // with the measured RTT will be sent. Defaults to 60000, set
  354. // // to <= 0 to disable.
  355. // analyticsInterval: 60000,
  356. // },
  357. // If set, will attempt to use the provided video input device label when
  358. // triggering a screenshare, instead of proceeding through the normal flow
  359. // for obtaining a desktop stream.
  360. // NOTE: This option is experimental and is currently intended for internal
  361. // use only.
  362. // _desktopSharingSourceDevice: 'sample-id-or-label',
  363. // If true, any checks to handoff to another application will be prevented
  364. // and instead the app will continue to display in the current browser.
  365. // disableDeepLinking: false,
  366. // A property to disable the right click context menu for localVideo
  367. // the menu has option to flip the locally seen video for local presentations
  368. // disableLocalVideoFlip: false,
  369. // Mainly privacy related settings
  370. // Disables all invite functions from the app (share, invite, dial out...etc)
  371. // disableInviteFunctions: true,
  372. // Disables storing the room name to the recents list
  373. // doNotStoreRoom: true,
  374. // Deployment specific URLs.
  375. // deploymentUrls: {
  376. // // If specified a 'Help' button will be displayed in the overflow menu with a link to the specified URL for
  377. // // user documentation.
  378. // userDocumentationURL: 'https://docs.example.com/video-meetings.html',
  379. // // If specified a 'Download our apps' button will be displayed in the overflow menu with a link
  380. // // to the specified URL for an app download page.
  381. // downloadAppsUrl: 'https://docs.example.com/our-apps.html'
  382. // },
  383. // Options related to the remote participant menu.
  384. // remoteVideoMenu: {
  385. // // If set to true the 'Kick out' button will be disabled.
  386. // disableKick: true
  387. // },
  388. // If set to true all muting operations of remote participants will be disabled.
  389. // disableRemoteMute: true,
  390. /**
  391. External API url used to receive branding specific information.
  392. If there is no url set or there are missing fields, the defaults are applied.
  393. None of the fieds are mandatory and the response must have the shape:
  394. {
  395. // The hex value for the colour used as background
  396. backgroundColor: '#fff',
  397. // The url for the image used as background
  398. backgroundImageUrl: 'https://example.com/background-img.png',
  399. // The anchor url used when clicking the logo image
  400. logoClickUrl: 'https://example-company.org',
  401. // The url used for the image used as logo
  402. logoImageUrl: 'https://example.com/logo-img.png'
  403. }
  404. */
  405. // brandingDataUrl: '',
  406. // List of undocumented settings used in jitsi-meet
  407. /**
  408. _immediateReloadThreshold
  409. autoRecord
  410. autoRecordToken
  411. debug
  412. debugAudioLevels
  413. deploymentInfo
  414. dialInConfCodeUrl
  415. dialInNumbersUrl
  416. dialOutAuthUrl
  417. dialOutCodesUrl
  418. disableRemoteControl
  419. displayJids
  420. etherpad_base
  421. externalConnectUrl
  422. firefox_fake_device
  423. googleApiApplicationClientID
  424. iAmRecorder
  425. iAmSipGateway
  426. microsoftApiApplicationClientID
  427. peopleSearchQueryTypes
  428. peopleSearchUrl
  429. requireDisplayName
  430. tokenAuthUrl
  431. */
  432. // List of undocumented settings used in lib-jitsi-meet
  433. /**
  434. _peerConnStatusOutOfLastNTimeout
  435. _peerConnStatusRtcMuteTimeout
  436. abTesting
  437. avgRtpStatsN
  438. callStatsConfIDNamespace
  439. callStatsCustomScriptUrl
  440. desktopSharingSources
  441. disableAEC
  442. disableAGC
  443. disableAP
  444. disableHPF
  445. disableNS
  446. enableLipSync
  447. enableTalkWhileMuted
  448. forceJVB121Ratio
  449. hiddenDomain
  450. ignoreStartMuted
  451. nick
  452. startBitrate
  453. */
  454. // Allow all above example options to include a trailing comma and
  455. // prevent fear when commenting out the last value.
  456. makeJsonParserHappy: 'even if last key had a trailing comma'
  457. // no configuration value should follow this line.
  458. };
  459. /* eslint-enable no-unused-vars, no-var */