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OUR_DOMAIN.-config.js

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  1. /* eslint-disable no-unused-vars, no-var */
  2.  
  3. var config = {
  4. // Connection
  5. //
  6.  
  7. hosts: {
  8. // XMPP domain.
  9. domain: 'jitsi-vm.OUR_DOMAIN',
  10.  
  11. // When using authentication, domain for guest users.
  12. // anonymousdomain: 'guest.example.com',
  13.  
  14. // Domain for authenticated users. Defaults to <domain>.
  15. // authdomain: 'jitsi-vm.OUR_DOMAIN',
  16.  
  17. // Focus component domain. Defaults to focus.<domain>.
  18. // focus: 'focus.jitsi-vm.OUR_DOMAIN',
  19.  
  20. // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
  21. muc: 'conference.<!--# echo var="subdomain" default="" -->jitsi-vm.OUR_DOMAIN'
  22. },
  23.  
  24. // BOSH URL. FIXME: use XEP-0156 to discover it.
  25. bosh: '//jitsi-vm.OUR_DOMAIN/http-bind',
  26.  
  27. // Websocket URL
  28. // websocket: 'wss://jitsi-vm.OUR_DOMAIN/xmpp-websocket',
  29.  
  30. // The name of client node advertised in XEP-0115 'c' stanza
  31. clientNode: 'http://jitsi.org/jitsimeet',
  32.  
  33. // The real JID of focus participant - can be overridden here
  34. // Do not change username - FIXME: Make focus username configurable
  35. // https://github.com/jitsi/jitsi-meet/issues/7376
  36. // focusUserJid: 'focus@auth.jitsi-vm.OUR_DOMAIN',
  37.  
  38.  
  39. // Testing / experimental features.
  40. //
  41.  
  42. testing: {
  43. octo: {
  44. probability: 1
  45. },
  46. // Disables the End to End Encryption feature. Useful for debugging
  47. // issues related to insertable streams.
  48. // disableE2EE: false,
  49.  
  50. // P2P test mode disables automatic switching to P2P when there are 2
  51. // participants in the conference.
  52. p2pTestMode: false
  53.  
  54. // Enables the test specific features consumed by jitsi-meet-torture
  55. // testMode: false
  56.  
  57. // Disables the auto-play behavior of *all* newly created video element.
  58. // This is useful when the client runs on a host with limited resources.
  59. // noAutoPlayVideo: false
  60.  
  61. // Enable / disable 500 Kbps bitrate cap on desktop tracks. When enabled,
  62. // simulcast is turned off for the desktop share. If presenter is turned
  63. // on while screensharing is in progress, the max bitrate is automatically
  64. // adjusted to 2.5 Mbps. This takes a value between 0 and 1 which determines
  65. // the probability for this to be enabled. This setting has been deprecated.
  66. // desktopSharingFrameRate.max now determines whether simulcast will be enabled
  67. // or disabled for the screenshare.
  68. // capScreenshareBitrate: 1 // 0 to disable - deprecated.
  69.  
  70. // Enable callstats only for a percentage of users.
  71. // This takes a value between 0 and 100 which determines the probability for
  72. // the callstats to be enabled.
  73. // callStatsThreshold: 5 // enable callstats for 5% of the users.
  74. },
  75.  
  76. // Enables reactions feature.
  77. // enableReactions: false,
  78.  
  79. // Disables ICE/UDP by filtering out local and remote UDP candidates in
  80. // signalling.
  81. // webrtcIceUdpDisable: false,
  82.  
  83. // Disables ICE/TCP by filtering out local and remote TCP candidates in
  84. // signalling.
  85. // webrtcIceTcpDisable: false,
  86.  
  87.  
  88. // Media
  89. //
  90.  
  91. // Enable unified plan implementation support on Chromium based browsers.
  92. // enableUnifiedOnChrome: false,
  93.  
  94. // Audio
  95.  
  96. // Disable measuring of audio levels.
  97. // disableAudioLevels: false,
  98. // audioLevelsInterval: 200,
  99.  
  100. // Enabling this will run the lib-jitsi-meet no audio detection module which
  101. // will notify the user if the current selected microphone has no audio
  102. // input and will suggest another valid device if one is present.
  103. enableNoAudioDetection: true,
  104.  
  105. // Enabling this will show a "Save Logs" link in the GSM popover that can be
  106. // used to collect debug information (XMPP IQs, SDP offer/answer cycles)
  107. // about the call.
  108. // enableSaveLogs: false,
  109.  
  110. // Enabling this will hide the "Show More" link in the GSM popover that can be
  111. // used to display more statistics about the connection (IP, Port, protocol, etc).
  112. // disableShowMoreStats: true,
  113.  
  114. // Enabling this will run the lib-jitsi-meet noise detection module which will
  115. // notify the user if there is noise, other than voice, coming from the current
  116. // selected microphone. The purpose it to let the user know that the input could
  117. // be potentially unpleasant for other meeting participants.
  118. enableNoisyMicDetection: true,
  119.  
  120. // Start the conference in audio only mode (no video is being received nor
  121. // sent).
  122. // startAudioOnly: false,
  123.  
  124. // Every participant after the Nth will start audio muted.
  125. // startAudioMuted: 10,
  126.  
  127. // Start calls with audio muted. Unlike the option above, this one is only
  128. // applied locally. FIXME: having these 2 options is confusing.
  129. // startWithAudioMuted: false,
  130.  
  131. // Enabling it (with #params) will disable local audio output of remote
  132. // participants and to enable it back a reload is needed.
  133. // startSilent: false
  134.  
  135. // Enables support for opus-red (redundancy for Opus).
  136. // enableOpusRed: false,
  137.  
  138. // Specify audio quality stereo and opusMaxAverageBitrate values in order to enable HD audio.
  139. // Beware, by doing so, you are disabling echo cancellation, noise suppression and AGC.
  140. // audioQuality: {
  141. // stereo: false,
  142. // opusMaxAverageBitrate: null // Value to fit the 6000 to 510000 range.
  143. // },
  144.  
  145. // Video
  146.  
  147. // Sets the preferred resolution (height) for local video. Defaults to 720.
  148. // resolution: 720,
  149.  
  150. // How many participants while in the tile view mode, before the receiving video quality is reduced from HD to SD.
  151. // Use -1 to disable.
  152. // maxFullResolutionParticipants: 2,
  153.  
  154. // w3c spec-compliant video constraints to use for video capture. Currently
  155. // used by browsers that return true from lib-jitsi-meet's
  156. // util#browser#usesNewGumFlow. The constraints are independent from
  157. // this config's resolution value. Defaults to requesting an ideal
  158. // resolution of 720p.
  159. // constraints: {
  160. // video: {
  161. // height: {
  162. // ideal: 720,
  163. // max: 720,
  164. // min: 240
  165. // }
  166. // }
  167. // },
  168.  
  169. // Enable / disable simulcast support.
  170. // disableSimulcast: false,
  171.  
  172. // Enable / disable layer suspension. If enabled, endpoints whose HD
  173. // layers are not in use will be suspended (no longer sent) until they
  174. // are requested again.
  175. // enableLayerSuspension: false,
  176.  
  177. // Every participant after the Nth will start video muted.
  178. // startVideoMuted: 10,
  179.  
  180. // Start calls with video muted. Unlike the option above, this one is only
  181. // applied locally. FIXME: having these 2 options is confusing.
  182. // startWithVideoMuted: false,
  183.  
  184. // If set to true, prefer to use the H.264 video codec (if supported).
  185. // Note that it's not recommended to do this because simulcast is not
  186. // supported when using H.264. For 1-to-1 calls this setting is enabled by
  187. // default and can be toggled in the p2p section.
  188. // This option has been deprecated, use preferredCodec under videoQuality section instead.
  189. // preferH264: true,
  190.  
  191. // If set to true, disable H.264 video codec by stripping it out of the
  192. // SDP.
  193. // disableH264: false,
  194.  
  195. // Desktop sharing
  196.  
  197. // Optional desktop sharing frame rate options. Default value: min:5, max:5.
  198. // desktopSharingFrameRate: {
  199. // min: 5,
  200. // max: 5
  201. // },
  202.  
  203. // Try to start calls with screen-sharing instead of camera video.
  204. // startScreenSharing: false,
  205.  
  206. // Recording
  207.  
  208. // Whether to enable file recording or not.
  209. // fileRecordingsEnabled: false,
  210. // Enable the dropbox integration.
  211. // dropbox: {
  212. // appKey: '<APP_KEY>' // Specify your app key here.
  213. // // A URL to redirect the user to, after authenticating
  214. // // by default uses:
  215. // // 'https://jitsi-vm.OUR_DOMAIN/static/oauth.html'
  216. // redirectURI:
  217. // 'https://jitsi-vm.OUR_DOMAIN/subfolder/static/oauth.html'
  218. // },
  219. // When integrations like dropbox are enabled only that will be shown,
  220. // by enabling fileRecordingsServiceEnabled, we show both the integrations
  221. // and the generic recording service (its configuration and storage type
  222. // depends on jibri configuration)
  223. // fileRecordingsServiceEnabled: false,
  224. // Whether to show the possibility to share file recording with other people
  225. // (e.g. meeting participants), based on the actual implementation
  226. // on the backend.
  227. // fileRecordingsServiceSharingEnabled: false,
  228.  
  229. // Whether to enable live streaming or not.
  230. // liveStreamingEnabled: false,
  231.  
  232. // Transcription (in interface_config,
  233. // subtitles and buttons can be configured)
  234. // transcribingEnabled: false,
  235.  
  236. // If true transcriber will use the application language.
  237. // The application language is either explicitly set by participants in their settings or automatically
  238. // detected based on the environment, e.g. if the app is opened in a chrome instance which is using french as its
  239. // default language then transcriptions for that participant will be in french.
  240. // Defaults to true.
  241. // transcribeWithAppLanguage: true,
  242.  
  243. // Transcriber language. This settings will only work if "transcribeWithAppLanguage" is explicitly set to false.
  244. // Available languages can be found in lang/language.json.
  245. // preferredTranscribeLanguage: 'en',
  246.  
  247. // Enables automatic turning on captions when recording is started
  248. // autoCaptionOnRecord: false,
  249.  
  250. // Misc
  251.  
  252. // Default value for the channel "last N" attribute. -1 for unlimited.
  253. channelLastN: 40,
  254.  
  255. // Provides a way for the lastN value to be controlled through the UI.
  256. // When startLastN is present, conference starts with a last-n value of startLastN and channelLastN
  257. // value will be used when the quality level is selected using "Manage Video Quality" slider.
  258. // startLastN: 1,
  259.  
  260. // Provides a way to use different "last N" values based on the number of participants in the conference.
  261. // The keys in an Object represent number of participants and the values are "last N" to be used when number of
  262. // participants gets to or above the number.
  263. //
  264. // For the given example mapping, "last N" will be set to 20 as long as there are at least 5, but less than
  265. // 29 participants in the call and it will be lowered to 15 when the 30th participant joins. The 'channelLastN'
  266. // will be used as default until the first threshold is reached.
  267. //
  268. // lastNLimits: {
  269. // 5: 20,
  270. // 30: 15,
  271. // 50: 10,
  272. // 70: 5,
  273. // 90: 2
  274. // },
  275.  
  276. // Provides a way to translate the legacy bridge signaling messages, 'LastNChangedEvent',
  277. // 'SelectedEndpointsChangedEvent' and 'ReceiverVideoConstraint' into the new 'ReceiverVideoConstraints' message
  278. // that invokes the new bandwidth allocation algorithm in the bridge which is described here
  279. // - https://github.com/jitsi/jitsi-videobridge/blob/master/doc/allocation.md.
  280. // useNewBandwidthAllocationStrategy: false,
  281.  
  282. // Specify the settings for video quality optimizations on the client.
  283. // videoQuality: {
  284. // // Provides a way to prevent a video codec from being negotiated on the JVB connection. The codec specified
  285. // // here will be removed from the list of codecs present in the SDP answer generated by the client. If the
  286. // // same codec is specified for both the disabled and preferred option, the disable settings will prevail.
  287. // // Note that 'VP8' cannot be disabled since it's a mandatory codec, the setting will be ignored in this case.
  288. // disabledCodec: 'H264',
  289. //
  290. // // Provides a way to set a preferred video codec for the JVB connection. If 'H264' is specified here,
  291. // // simulcast will be automatically disabled since JVB doesn't support H264 simulcast yet. This will only
  292. // // rearrange the the preference order of the codecs in the SDP answer generated by the browser only if the
  293. // // preferred codec specified here is present. Please ensure that the JVB offers the specified codec for this
  294. // // to take effect.
  295. // preferredCodec: 'VP8',
  296. //
  297. // // Provides a way to enforce the preferred codec for the conference even when the conference has endpoints
  298. // // that do not support the preferred codec. For example, older versions of Safari do not support VP9 yet.
  299. // // This will result in Safari not being able to decode video from endpoints sending VP9 video.
  300. // // When set to false, the conference falls back to VP8 whenever there is an endpoint that doesn't support the
  301. // // preferred codec and goes back to the preferred codec when that endpoint leaves.
  302. // // enforcePreferredCodec: false,
  303. //
  304. // // Provides a way to configure the maximum bitrates that will be enforced on the simulcast streams for
  305. // // video tracks. The keys in the object represent the type of the stream (LD, SD or HD) and the values
  306. // // are the max.bitrates to be set on that particular type of stream. The actual send may vary based on
  307. // // the available bandwidth calculated by the browser, but it will be capped by the values specified here.
  308. // // This is currently not implemented on app based clients on mobile.
  309. // maxBitratesVideo: {
  310. // H264: {
  311. // low: 200000,
  312. // standard: 500000,
  313. // high: 1500000
  314. // },
  315. // VP8 : {
  316. // low: 200000,
  317. // standard: 500000,
  318. // high: 1500000
  319. // },
  320. // VP9: {
  321. // low: 100000,
  322. // standard: 300000,
  323. // high: 1200000
  324. // }
  325. // },
  326. //
  327. // // The options can be used to override default thresholds of video thumbnail heights corresponding to
  328. // // the video quality levels used in the application. At the time of this writing the allowed levels are:
  329. // // 'low' - for the low quality level (180p at the time of this writing)
  330. // // 'standard' - for the medium quality level (360p)
  331. // // 'high' - for the high quality level (720p)
  332. // // The keys should be positive numbers which represent the minimal thumbnail height for the quality level.
  333. // //
  334. // // With the default config value below the application will use 'low' quality until the thumbnails are
  335. // // at least 360 pixels tall. If the thumbnail height reaches 720 pixels then the application will switch to
  336. // // the high quality.
  337. // minHeightForQualityLvl: {
  338. // 360: 'standard',
  339. // 720: 'high'
  340. // },
  341. //
  342. // // Provides a way to resize the desktop track to 720p (if it is greater than 720p) before creating a canvas
  343. // // for the presenter mode (camera picture-in-picture mode with screenshare).
  344. // resizeDesktopForPresenter: false
  345. // },
  346.  
  347. // // Options for the recording limit notification.
  348. // recordingLimit: {
  349. //
  350. // // The recording limit in minutes. Note: This number appears in the notification text
  351. // // but doesn't enforce the actual recording time limit. This should be configured in
  352. // // jibri!
  353. // limit: 60,
  354. //
  355. // // The name of the app with unlimited recordings.
  356. // appName: 'Unlimited recordings APP',
  357. //
  358. // // The URL of the app with unlimited recordings.
  359. // appURL: 'https://unlimited.recordings.app.com/'
  360. // },
  361.  
  362. // Disables or enables RTX (RFC 4588) (defaults to false).
  363. // disableRtx: false,
  364.  
  365. // Disables or enables TCC support in this client (default: enabled).
  366. // enableTcc: true,
  367.  
  368. // Disables or enables REMB support in this client (default: enabled).
  369. // enableRemb: true,
  370.  
  371. // Enables ICE restart logic in LJM and displays the page reload overlay on
  372. // ICE failure. Current disabled by default because it's causing issues with
  373. // signaling when Octo is enabled. Also when we do an "ICE restart"(which is
  374. // not a real ICE restart), the client maintains the TCC sequence number
  375. // counter, but the bridge resets it. The bridge sends media packets with
  376. // TCC sequence numbers starting from 0.
  377. // enableIceRestart: false,
  378.  
  379. // Enables forced reload of the client when the call is migrated as a result of
  380. // the bridge going down.
  381. // enableForcedReload: true,
  382.  
  383. // Use TURN/UDP servers for the jitsi-videobridge connection (by default
  384. // we filter out TURN/UDP because it is usually not needed since the
  385. // bridge itself is reachable via UDP)
  386. // useTurnUdp: false
  387.  
  388. // UI
  389. //
  390.  
  391. // Disables responsive tiles.
  392. // disableResponsiveTiles: false,
  393.  
  394. // Hides lobby button
  395. // hideLobbyButton: false,
  396.  
  397. // Require users to always specify a display name.
  398. // requireDisplayName: true,
  399.  
  400. // Whether to use a welcome page or not. In case it's false a random room
  401. // will be joined when no room is specified.
  402. enableWelcomePage: true,
  403.  
  404. // Disable app shortcuts that are registered upon joining a conference
  405. // disableShortcuts: false,
  406.  
  407. // Disable initial browser getUserMedia requests.
  408. // This is useful for scenarios where users might want to start a conference for screensharing only
  409. // disableInitialGUM: false,
  410.  
  411. // Enabling the close page will ignore the welcome page redirection when
  412. // a call is hangup.
  413. // enableClosePage: false,
  414.  
  415. // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
  416. // Setting this to null, will also disable showing the remote videos
  417. // when the toolbar is shown on mouse movements
  418. // disable1On1Mode: null | false | true,
  419.  
  420. // Default language for the user interface.
  421. // defaultLanguage: 'en',
  422.  
  423. // Disables profile and the edit of all fields from the profile settings (display name and email)
  424. // disableProfile: false,
  425.  
  426. // Whether or not some features are checked based on token.
  427. // enableFeaturesBasedOnToken: false,
  428.  
  429. // When enabled the password used for locking a room is restricted to up to the number of digits specified
  430. // roomPasswordNumberOfDigits: 10,
  431. // default: roomPasswordNumberOfDigits: false,
  432.  
  433. // Message to show the users. Example: 'The service will be down for
  434. // maintenance at 01:00 AM GMT,
  435. // noticeMessage: '',
  436.  
  437. // Enables calendar integration, depends on googleApiApplicationClientID
  438. // and microsoftApiApplicationClientID
  439. // enableCalendarIntegration: false,
  440.  
  441. // When 'true', it shows an intermediate page before joining, where the user can configure their devices.
  442. // prejoinPageEnabled: false,
  443.  
  444. // If etherpad integration is enabled, setting this to true will
  445. // automatically open the etherpad when a participant joins. This
  446. // does not affect the mobile app since opening an etherpad
  447. // obscures the conference controls -- it's better to let users
  448. // choose to open the pad on their own in that case.
  449. // openSharedDocumentOnJoin: false,
  450.  
  451. // If true, shows the unsafe room name warning label when a room name is
  452. // deemed unsafe (due to the simplicity in the name) and a password is not
  453. // set or the lobby is not enabled.
  454. // enableInsecureRoomNameWarning: false,
  455.  
  456. // Whether to automatically copy invitation URL after creating a room.
  457. // Document should be focused for this option to work
  458. // enableAutomaticUrlCopy: false,
  459.  
  460. // Base URL for a Gravatar-compatible service. Defaults to libravatar.
  461. // gravatarBaseURL: 'https://seccdn.libravatar.org/avatar/',
  462.  
  463. // App name to be displayed in the invitation email subject, as an alternative to
  464. // interfaceConfig.APP_NAME.
  465. // inviteAppName: null,
  466.  
  467. // Moved from interfaceConfig(TOOLBAR_BUTTONS).
  468. // The name of the toolbar buttons to display in the toolbar, including the
  469. // "More actions" menu. If present, the button will display. Exceptions are
  470. // "livestreaming" and "recording" which also require being a moderator and
  471. // some other values in config.js to be enabled. Also, the "profile" button will
  472. // not display for users with a JWT.
  473. // Notes:
  474. // - it's impossible to choose which buttons go in the "More actions" menu
  475. // - it's impossible to control the placement of buttons
  476. // - 'desktop' controls the "Share your screen" button
  477. // - if `toolbarButtons` is undefined, we fallback to enabling all buttons on the UI
  478. // toolbarButtons: [
  479. // 'camera',
  480. // 'chat',
  481. // 'closedcaptions',
  482. // 'desktop',
  483. // 'download',
  484. // 'embedmeeting',
  485. // 'etherpad',
  486. // 'feedback',
  487. // 'filmstrip',
  488. // 'fullscreen',
  489. // 'hangup',
  490. // 'help',
  491. // 'invite',
  492. // 'livestreaming',
  493. // 'microphone',
  494. // 'mute-everyone',
  495. // 'mute-video-everyone',
  496. // 'participants-pane',
  497. // 'profile',
  498. // 'raisehand',
  499. // 'recording',
  500. // 'security',
  501. // 'select-background',
  502. // 'settings',
  503. // 'shareaudio',
  504. // 'sharedvideo',
  505. // 'shortcuts',
  506. // 'stats',
  507. // 'tileview',
  508. // 'toggle-camera',
  509. // 'videoquality',
  510. // '__end'
  511. // ],
  512.  
  513. // Stats
  514. //
  515.  
  516. // Whether to enable stats collection or not in the TraceablePeerConnection.
  517. // This can be useful for debugging purposes (post-processing/analysis of
  518. // the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
  519. // estimation tests.
  520. // gatherStats: false,
  521.  
  522. // The interval at which PeerConnection.getStats() is called. Defaults to 10000
  523. // pcStatsInterval: 10000,
  524.  
  525. // To enable sending statistics to callstats.io you must provide the
  526. // Application ID and Secret.
  527. // callStatsID: '',
  528. // callStatsSecret: '',
  529.  
  530. // Enables sending participants' display names to callstats
  531. // enableDisplayNameInStats: false,
  532.  
  533. // Enables sending participants' emails (if available) to callstats and other analytics
  534. // enableEmailInStats: false,
  535.  
  536. // Controls the percentage of automatic feedback shown to participants when callstats is enabled.
  537. // The default value is 100%. If set to 0, no automatic feedback will be requested
  538. // feedbackPercentage: 100,
  539.  
  540. // Privacy
  541. //
  542.  
  543. // If third party requests are disabled, no other server will be contacted.
  544. // This means avatars will be locally generated and callstats integration
  545. // will not function.
  546. // disableThirdPartyRequests: false,
  547.  
  548.  
  549. // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
  550. //
  551.  
  552. p2p: {
  553. // Enables peer to peer mode. When enabled the system will try to
  554. // establish a direct connection when there are exactly 2 participants
  555. // in the room. If that succeeds the conference will stop sending data
  556. // through the JVB and use the peer to peer connection instead. When a
  557. // 3rd participant joins the conference will be moved back to the JVB
  558. // connection.
  559. enabled: true,
  560.  
  561. // Enable unified plan implementation support on Chromium for p2p connection.
  562. // enableUnifiedOnChrome: false,
  563.  
  564. // Sets the ICE transport policy for the p2p connection. At the time
  565. // of this writing the list of possible values are 'all' and 'relay',
  566. // but that is subject to change in the future. The enum is defined in
  567. // the WebRTC standard:
  568. // https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
  569. // If not set, the effective value is 'all'.
  570. // iceTransportPolicy: 'all',
  571.  
  572. // If set to true, it will prefer to use H.264 for P2P calls (if H.264
  573. // is supported). This setting is deprecated, use preferredCodec instead.
  574. // preferH264: true,
  575.  
  576. // Provides a way to set the video codec preference on the p2p connection. Acceptable
  577. // codec values are 'VP8', 'VP9' and 'H264'.
  578. // preferredCodec: 'H264',
  579.  
  580. // If set to true, disable H.264 video codec by stripping it out of the
  581. // SDP. This setting is deprecated, use disabledCodec instead.
  582. // disableH264: false,
  583.  
  584. // Provides a way to prevent a video codec from being negotiated on the p2p connection.
  585. // disabledCodec: '',
  586.  
  587. // How long we're going to wait, before going back to P2P after the 3rd
  588. // participant has left the conference (to filter out page reload).
  589. // backToP2PDelay: 5,
  590.  
  591. // The STUN servers that will be used in the peer to peer connections
  592. stunServers: [
  593.  
  594. // { urls: 'stun:jitsi-vm.OUR_DOMAIN:3478' },
  595. { urls: 'stun:meet-jit-si-turnrelay.jitsi.net:443' }
  596. ]
  597. },
  598.  
  599. analytics: {
  600. // The Google Analytics Tracking ID:
  601. // googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
  602.  
  603. // Matomo configuration:
  604. // matomoEndpoint: 'https://your-matomo-endpoint/',
  605. // matomoSiteID: '42',
  606.  
  607. // The Amplitude APP Key:
  608. // amplitudeAPPKey: '<APP_KEY>'
  609.  
  610. // Configuration for the rtcstats server:
  611. // By enabling rtcstats server every time a conference is joined the rtcstats
  612. // module connects to the provided rtcstatsEndpoint and sends statistics regarding
  613. // PeerConnection states along with getStats metrics polled at the specified
  614. // interval.
  615. // rtcstatsEnabled: true,
  616.  
  617. // In order to enable rtcstats one needs to provide a endpoint url.
  618. // rtcstatsEndpoint: wss://rtcstats-server-pilot.jitsi.net/,
  619.  
  620. // The interval at which rtcstats will poll getStats, defaults to 1000ms.
  621. // If the value is set to 0 getStats won't be polled and the rtcstats client
  622. // will only send data related to RTCPeerConnection events.
  623. // rtcstatsPolIInterval: 1000,
  624.  
  625. // Array of script URLs to load as lib-jitsi-meet "analytics handlers".
  626. // scriptURLs: [
  627. // "libs/analytics-ga.min.js", // google-analytics
  628. // "https://example.com/my-custom-analytics.js"
  629. // ],
  630. },
  631.  
  632. // Logs that should go be passed through the 'log' event if a handler is defined for it
  633. // apiLogLevels: ['warn', 'log', 'error', 'info', 'debug'],
  634.  
  635. // Information about the jitsi-meet instance we are connecting to, including
  636. // the user region as seen by the server.
  637. deploymentInfo: {
  638. // shard: "shard1",
  639. // region: "europe",
  640. // userRegion: "asia"
  641. },
  642.  
  643. // Decides whether the start/stop recording audio notifications should play on record.
  644. // disableRecordAudioNotification: false,
  645.  
  646. // Disables the sounds that play when other participants join or leave the
  647. // conference (if set to true, these sounds will not be played).
  648. // disableJoinLeaveSounds: false,
  649.  
  650. // Disables the sounds that play when a chat message is received.
  651. // disableIncomingMessageSound: false,
  652.  
  653. // Information for the chrome extension banner
  654. // chromeExtensionBanner: {
  655. // // The chrome extension to be installed address
  656. // url: 'https://chrome.google.com/webstore/detail/jitsi-meetings/kglhbbefdnlheedjiejgomgmfplipfeb',
  657.  
  658. // // Extensions info which allows checking if they are installed or not
  659. // chromeExtensionsInfo: [
  660. // {
  661. // id: 'kglhbbefdnlheedjiejgomgmfplipfeb',
  662. // path: 'jitsi-logo-48x48.png'
  663. // }
  664. // ]
  665. // },
  666.  
  667. // Local Recording
  668. //
  669.  
  670. // localRecording: {
  671. // Enables local recording.
  672. // Additionally, 'localrecording' (all lowercase) needs to be added to
  673. // the `toolbarButtons`-array for the Local Recording button to show up
  674. // on the toolbar.
  675. //
  676. // enabled: true,
  677. //
  678.  
  679. // The recording format, can be one of 'ogg', 'flac' or 'wav'.
  680. // format: 'flac'
  681. //
  682.  
  683. // },
  684.  
  685. // Options related to end-to-end (participant to participant) ping.
  686. // e2eping: {
  687. // // The interval in milliseconds at which pings will be sent.
  688. // // Defaults to 10000, set to <= 0 to disable.
  689. // pingInterval: 10000,
  690. //
  691. // // The interval in milliseconds at which analytics events
  692. // // with the measured RTT will be sent. Defaults to 60000, set
  693. // // to <= 0 to disable.
  694. // analyticsInterval: 60000,
  695. // },
  696.  
  697. // If set, will attempt to use the provided video input device label when
  698. // triggering a screenshare, instead of proceeding through the normal flow
  699. // for obtaining a desktop stream.
  700. // NOTE: This option is experimental and is currently intended for internal
  701. // use only.
  702. // _desktopSharingSourceDevice: 'sample-id-or-label',
  703.  
  704. // If true, any checks to handoff to another application will be prevented
  705. // and instead the app will continue to display in the current browser.
  706. // disableDeepLinking: false,
  707.  
  708. // A property to disable the right click context menu for localVideo
  709. // the menu has option to flip the locally seen video for local presentations
  710. // disableLocalVideoFlip: false,
  711.  
  712. // A property used to unset the default flip state of the local video.
  713. // When it is set to 'true', the local(self) video will not be mirrored anymore.
  714. // doNotFlipLocalVideo: false,
  715.  
  716. // Mainly privacy related settings
  717.  
  718. // Disables all invite functions from the app (share, invite, dial out...etc)
  719. // disableInviteFunctions: true,
  720.  
  721. // Disables storing the room name to the recents list
  722. // doNotStoreRoom: true,
  723.  
  724. // Deployment specific URLs.
  725. // deploymentUrls: {
  726. // // If specified a 'Help' button will be displayed in the overflow menu with a link to the specified URL for
  727. // // user documentation.
  728. // userDocumentationURL: 'https://docs.example.com/video-meetings.html',
  729. // // If specified a 'Download our apps' button will be displayed in the overflow menu with a link
  730. // // to the specified URL for an app download page.
  731. // downloadAppsUrl: 'https://docs.example.com/our-apps.html'
  732. // },
  733.  
  734. // Options related to the remote participant menu.
  735. // remoteVideoMenu: {
  736. // // If set to true the 'Kick out' button will be disabled.
  737. // disableKick: true,
  738. // // If set to true the 'Grant moderator' button will be disabled.
  739. // disableGrantModerator: true
  740. // },
  741.  
  742. // If set to true all muting operations of remote participants will be disabled.
  743. // disableRemoteMute: true,
  744.  
  745. // Enables support for lip-sync for this client (if the browser supports it).
  746. // enableLipSync: false
  747.  
  748. /**
  749. External API url used to receive branding specific information.
  750. If there is no url set or there are missing fields, the defaults are applied.
  751. The config file should be in JSON.
  752. None of the fields are mandatory and the response must have the shape:
  753. {
  754. // The domain url to apply (will replace the domain in the sharing conference link/embed section)
  755. inviteDomain: 'example-company.org,
  756. // The hex value for the colour used as background
  757. backgroundColor: '#fff',
  758. // The url for the image used as background
  759. backgroundImageUrl: 'https://example.com/background-img.png',
  760. // The anchor url used when clicking the logo image
  761. logoClickUrl: 'https://example-company.org',
  762. // The url used for the image used as logo
  763. logoImageUrl: 'https://example.com/logo-img.png'
  764. }
  765. */
  766. // dynamicBrandingUrl: '',
  767.  
  768. // Sets the background transparency level. '0' is fully transparent, '1' is opaque.
  769. // backgroundAlpha: 1,
  770.  
  771. // The URL of the moderated rooms microservice, if available. If it
  772. // is present, a link to the service will be rendered on the welcome page,
  773. // otherwise the app doesn't render it.
  774. // moderatedRoomServiceUrl: 'https://moderated.jitsi-vm.OUR_DOMAIN',
  775.  
  776. // If true, tile view will not be enabled automatically when the participants count threshold is reached.
  777. // disableTileView: true,
  778.  
  779. // Hides the conference subject
  780. // hideConferenceSubject: true,
  781.  
  782. // Hides the recording label
  783. // hideRecordingLabel: false,
  784.  
  785. // Hides the conference timer.
  786. // hideConferenceTimer: true,
  787.  
  788. // Hides the participants stats
  789. // hideParticipantsStats: true,
  790.  
  791. // Sets the conference subject
  792. // subject: 'Conference Subject',
  793.  
  794. // This property is related to the use case when jitsi-meet is used via the IFrame API. When the property is true
  795. // jitsi-meet will use the local storage of the host page instead of its own. This option is useful if the browser
  796. // is not persisting the local storage inside the iframe.
  797. // useHostPageLocalStorage: true,
  798.  
  799. // etherpad ("shared document") integration.
  800. //
  801.  
  802. // If set, add a "Open shared document" link to the bottom right menu that
  803. // will open an etherpad document.
  804. // etherpad_base: 'https://your-etherpad-installati.on/p/',
  805.  
  806. // If etherpad_base is set, and useRoomAsSharedDocumentName is set to true,
  807. // open a pad with the name of the room (lowercased) instead of a pad with a
  808. // random UUID.
  809. // useRoomAsSharedDocumentName: true,
  810.  
  811. // List of undocumented settings used in jitsi-meet
  812. /**
  813. _immediateReloadThreshold
  814. debug
  815. debugAudioLevels
  816. deploymentInfo
  817. dialInConfCodeUrl
  818. dialInNumbersUrl
  819. dialOutAuthUrl
  820. dialOutCodesUrl
  821. disableRemoteControl
  822. displayJids
  823. externalConnectUrl
  824. firefox_fake_device
  825. googleApiApplicationClientID
  826. iAmRecorder
  827. iAmSipGateway
  828. microsoftApiApplicationClientID
  829. peopleSearchQueryTypes
  830. peopleSearchUrl
  831. requireDisplayName
  832. tokenAuthUrl
  833. */
  834.  
  835. /**
  836. * This property can be used to alter the generated meeting invite links (in combination with a branding domain
  837. * which is retrieved internally by jitsi meet) (e.g. https://meet.jit.si/someMeeting
  838. * can become https://brandedDomain/roomAlias)
  839. */
  840. // brandingRoomAlias: null,
  841.  
  842. // List of undocumented settings used in lib-jitsi-meet
  843. /**
  844. _peerConnStatusOutOfLastNTimeout
  845. _peerConnStatusRtcMuteTimeout
  846. abTesting
  847. avgRtpStatsN
  848. callStatsConfIDNamespace
  849. callStatsCustomScriptUrl
  850. desktopSharingSources
  851. disableAEC
  852. disableAGC
  853. disableAP
  854. disableHPF
  855. disableNS
  856. enableTalkWhileMuted
  857. forceJVB121Ratio
  858. forceTurnRelay
  859. hiddenDomain
  860. ignoreStartMuted
  861. websocketKeepAlive
  862. websocketKeepAliveUrl
  863. */
  864.  
  865. /**
  866. * Default interval (milliseconds) for triggering mouseMoved iframe API event
  867. */
  868. mouseMoveCallbackInterval: 1000,
  869.  
  870. /**
  871. Use this array to configure which notifications will be shown to the user
  872. The items correspond to the title or description key of that notification
  873. Some of these notifications also depend on some other internal logic to be displayed or not,
  874. so adding them here will not ensure they will always be displayed
  875.  
  876. A falsy value for this prop will result in having all notifications enabled (e.g null, undefined, false)
  877. */
  878. // notifications: [
  879. // 'connection.CONNFAIL', // shown when the connection fails,
  880. // 'dialog.cameraNotSendingData', // shown when there's no feed from user's camera
  881. // 'dialog.kickTitle', // shown when user has been kicked
  882. // 'dialog.liveStreaming', // livestreaming notifications (pending, on, off, limits)
  883. // 'dialog.lockTitle', // shown when setting conference password fails
  884. // 'dialog.maxUsersLimitReached', // shown when maximmum users limit has been reached
  885. // 'dialog.micNotSendingData', // shown when user's mic is not sending any audio
  886. // 'dialog.passwordNotSupportedTitle', // shown when setting conference password fails due to password format
  887. // 'dialog.recording', // recording notifications (pending, on, off, limits)
  888. // 'dialog.remoteControlTitle', // remote control notifications (allowed, denied, start, stop, error)
  889. // 'dialog.reservationError',
  890. // 'dialog.serviceUnavailable', // shown when server is not reachable
  891. // 'dialog.sessTerminated', // shown when there is a failed conference session
  892. // 'dialog.sessionRestarted', // show when a client reload is initiated because of bridge migration
  893. // 'dialog.tokenAuthFailed', // show when an invalid jwt is used
  894. // 'dialog.transcribing', // transcribing notifications (pending, off)
  895. // 'dialOut.statusMessage', // shown when dial out status is updated.
  896. // 'liveStreaming.busy', // shown when livestreaming service is busy
  897. // 'liveStreaming.failedToStart', // shown when livestreaming fails to start
  898. // 'liveStreaming.unavailableTitle', // shown when livestreaming service is not reachable
  899. // 'lobby.joinRejectedMessage', // shown when while in a lobby, user's request to join is rejected
  900. // 'lobby.notificationTitle', // shown when lobby is toggled and when join requests are allowed / denied
  901. // 'localRecording.localRecording', // shown when a local recording is started
  902. // 'notify.disconnected', // shown when a participant has left
  903. // 'notify.grantedTo', // shown when moderator rights were granted to a participant
  904. // 'notify.invitedOneMember', // shown when 1 participant has been invited
  905. // 'notify.invitedThreePlusMembers', // shown when 3+ participants have been invited
  906. // 'notify.invitedTwoMembers', // shown when 2 participants have been invited
  907. // 'notify.kickParticipant', // shown when a participant is kicked
  908. // 'notify.mutedRemotelyTitle', // shown when user is muted by a remote party
  909. // 'notify.mutedTitle', // shown when user has been muted upon joining,
  910. // 'notify.newDeviceAudioTitle', // prompts the user to use a newly detected audio device
  911. // 'notify.newDeviceCameraTitle', // prompts the user to use a newly detected camera
  912. // 'notify.passwordRemovedRemotely', // shown when a password has been removed remotely
  913. // 'notify.passwordSetRemotely', // shown when a password has been set remotely
  914. // 'notify.raisedHand', // shown when a partcipant used raise hand,
  915. // 'notify.startSilentTitle', // shown when user joined with no audio
  916. // 'prejoin.errorDialOut',
  917. // 'prejoin.errorDialOutDisconnected',
  918. // 'prejoin.errorDialOutFailed',
  919. // 'prejoin.errorDialOutStatus',
  920. // 'prejoin.errorStatusCode',
  921. // 'prejoin.errorValidation',
  922. // 'recording.busy', // shown when recording service is busy
  923. // 'recording.failedToStart', // shown when recording fails to start
  924. // 'recording.unavailableTitle', // shown when recording service is not reachable
  925. // 'toolbar.noAudioSignalTitle', // shown when a broken mic is detected
  926. // 'toolbar.noisyAudioInputTitle', // shown when noise is detected for the current microphone
  927. // 'toolbar.talkWhileMutedPopup', // shown when user tries to speak while muted
  928. // 'transcribing.failedToStart' // shown when transcribing fails to start
  929. // ],
  930.  
  931. // Prevent the filmstrip from autohiding when screen width is under a certain threshold
  932. // disableFilmstripAutohiding: false,
  933.  
  934. // Allow all above example options to include a trailing comma and
  935. // prevent fear when commenting out the last value.
  936. makeJsonParserHappy: 'even if last key had a trailing comma'
  937.  
  938. // no configuration value should follow this line.
  939. };
  940.  
  941. /* eslint-enable no-unused-vars, no-var */
  942.  
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