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etc_jitsi_meet_jitsi_domain-config.js.j2 36KB

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