Nelze vybrat více než 25 témat Téma musí začínat písmenem nebo číslem, může obsahovat pomlčky („-“) a může být dlouhé až 35 znaků.

config.js 18KB

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  1. /* eslint-disable no-unused-vars, no-var */
  2. var config = {
  3. // Connection
  4. //
  5. hosts: {
  6. // XMPP domain.
  7. domain: 'jitsi-meet.example.com',
  8. // When using authentication, domain for guest users.
  9. // anonymousdomain: 'guest.example.com',
  10. // Domain for authenticated users. Defaults to <domain>.
  11. // authdomain: 'jitsi-meet.example.com',
  12. // Jirecon recording component domain.
  13. // jirecon: 'jirecon.jitsi-meet.example.com',
  14. // Call control component (Jigasi).
  15. // call_control: 'callcontrol.jitsi-meet.example.com',
  16. // Focus component domain. Defaults to focus.<domain>.
  17. // focus: 'focus.jitsi-meet.example.com',
  18. // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
  19. muc: 'conference.jitsi-meet.example.com'
  20. },
  21. // BOSH URL. FIXME: use XEP-0156 to discover it.
  22. bosh: '//jitsi-meet.example.com/http-bind',
  23. // Websocket URL
  24. // websocket: 'wss://jitsi-meet.example.com/xmpp-websocket',
  25. // The name of client node advertised in XEP-0115 'c' stanza
  26. clientNode: 'http://jitsi.org/jitsimeet',
  27. // The real JID of focus participant - can be overridden here
  28. // focusUserJid: 'focus@auth.jitsi-meet.example.com',
  29. // Testing / experimental features.
  30. //
  31. testing: {
  32. // Enables experimental simulcast support on Firefox.
  33. enableFirefoxSimulcast: false,
  34. // P2P test mode disables automatic switching to P2P when there are 2
  35. // participants in the conference.
  36. p2pTestMode: false
  37. // Enables the test specific features consumed by jitsi-meet-torture
  38. // testMode: false
  39. // Disables the auto-play behavior of *all* newly created video element.
  40. // This is useful when the client runs on a host with limited resources.
  41. // noAutoPlayVideo: false
  42. },
  43. // Disables ICE/UDP by filtering out local and remote UDP candidates in
  44. // signalling.
  45. // webrtcIceUdpDisable: false,
  46. // Disables ICE/TCP by filtering out local and remote TCP candidates in
  47. // signalling.
  48. // webrtcIceTcpDisable: false,
  49. // Media
  50. //
  51. // Audio
  52. // Disable measuring of audio levels.
  53. // disableAudioLevels: false,
  54. // audioLevelsInterval: 200,
  55. // Enabling this will run the lib-jitsi-meet no audio detection module which
  56. // will notify the user if the current selected microphone has no audio
  57. // input and will suggest another valid device if one is present.
  58. enableNoAudioDetection: true,
  59. // Enabling this will run the lib-jitsi-meet noise detection module which will
  60. // notify the user if there is noise, other than voice, coming from the current
  61. // selected microphone. The purpose it to let the user know that the input could
  62. // be potentially unpleasant for other meeting participants.
  63. enableNoisyMicDetection: true,
  64. // Start the conference in audio only mode (no video is being received nor
  65. // sent).
  66. // startAudioOnly: false,
  67. // Every participant after the Nth will start audio muted.
  68. // startAudioMuted: 10,
  69. // Start calls with audio muted. Unlike the option above, this one is only
  70. // applied locally. FIXME: having these 2 options is confusing.
  71. // startWithAudioMuted: false,
  72. // Enabling it (with #params) will disable local audio output of remote
  73. // participants and to enable it back a reload is needed.
  74. // startSilent: false
  75. // Video
  76. // Sets the preferred resolution (height) for local video. Defaults to 720.
  77. // resolution: 720,
  78. // w3c spec-compliant video constraints to use for video capture. Currently
  79. // used by browsers that return true from lib-jitsi-meet's
  80. // util#browser#usesNewGumFlow. The constraints are independent from
  81. // this config's resolution value. Defaults to requesting an ideal aspect
  82. // ratio of 16:9 with an ideal resolution of 720.
  83. // constraints: {
  84. // video: {
  85. // aspectRatio: 16 / 9,
  86. // height: {
  87. // ideal: 720,
  88. // max: 720,
  89. // min: 240
  90. // }
  91. // }
  92. // },
  93. // Enable / disable simulcast support.
  94. // disableSimulcast: false,
  95. // Enable / disable layer suspension. If enabled, endpoints whose HD
  96. // layers are not in use will be suspended (no longer sent) until they
  97. // are requested again.
  98. // enableLayerSuspension: false,
  99. // Every participant after the Nth will start video muted.
  100. // startVideoMuted: 10,
  101. // Start calls with video muted. Unlike the option above, this one is only
  102. // applied locally. FIXME: having these 2 options is confusing.
  103. // startWithVideoMuted: false,
  104. // If set to true, prefer to use the H.264 video codec (if supported).
  105. // Note that it's not recommended to do this because simulcast is not
  106. // supported when using H.264. For 1-to-1 calls this setting is enabled by
  107. // default and can be toggled in the p2p section.
  108. // preferH264: true,
  109. // If set to true, disable H.264 video codec by stripping it out of the
  110. // SDP.
  111. // disableH264: false,
  112. // Desktop sharing
  113. // The ID of the jidesha extension for Chrome.
  114. desktopSharingChromeExtId: null,
  115. // Whether desktop sharing should be disabled on Chrome.
  116. // desktopSharingChromeDisabled: false,
  117. // The media sources to use when using screen sharing with the Chrome
  118. // extension.
  119. desktopSharingChromeSources: [ 'screen', 'window', 'tab' ],
  120. // Required version of Chrome extension
  121. desktopSharingChromeMinExtVersion: '0.1',
  122. // Whether desktop sharing should be disabled on Firefox.
  123. // desktopSharingFirefoxDisabled: false,
  124. // Optional desktop sharing frame rate options. Default value: min:5, max:5.
  125. // desktopSharingFrameRate: {
  126. // min: 5,
  127. // max: 5
  128. // },
  129. // Try to start calls with screen-sharing instead of camera video.
  130. // startScreenSharing: false,
  131. // Recording
  132. // Whether to enable file recording or not.
  133. // fileRecordingsEnabled: false,
  134. // Enable the dropbox integration.
  135. // dropbox: {
  136. // appKey: '<APP_KEY>' // Specify your app key here.
  137. // // A URL to redirect the user to, after authenticating
  138. // // by default uses:
  139. // // 'https://jitsi-meet.example.com/static/oauth.html'
  140. // redirectURI:
  141. // 'https://jitsi-meet.example.com/subfolder/static/oauth.html'
  142. // },
  143. // When integrations like dropbox are enabled only that will be shown,
  144. // by enabling fileRecordingsServiceEnabled, we show both the integrations
  145. // and the generic recording service (its configuration and storage type
  146. // depends on jibri configuration)
  147. // fileRecordingsServiceEnabled: false,
  148. // Whether to show the possibility to share file recording with other people
  149. // (e.g. meeting participants), based on the actual implementation
  150. // on the backend.
  151. // fileRecordingsServiceSharingEnabled: false,
  152. // Whether to enable live streaming or not.
  153. // liveStreamingEnabled: false,
  154. // Transcription (in interface_config,
  155. // subtitles and buttons can be configured)
  156. // transcribingEnabled: false,
  157. // Enables automatic turning on captions when recording is started
  158. // autoCaptionOnRecord: false,
  159. // Misc
  160. // Default value for the channel "last N" attribute. -1 for unlimited.
  161. channelLastN: -1,
  162. // Disables or enables RTX (RFC 4588) (defaults to false).
  163. // disableRtx: false,
  164. // Disables or enables TCC (the default is in Jicofo and set to true)
  165. // (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting
  166. // affects congestion control, it practically enables send-side bandwidth
  167. // estimations.
  168. // enableTcc: true,
  169. // Disables or enables REMB (the default is in Jicofo and set to false)
  170. // (draft-alvestrand-rmcat-remb-03). This setting affects congestion
  171. // control, it practically enables recv-side bandwidth estimations. When
  172. // both TCC and REMB are enabled, TCC takes precedence. When both are
  173. // disabled, then bandwidth estimations are disabled.
  174. // enableRemb: false,
  175. // Defines the minimum number of participants to start a call (the default
  176. // is set in Jicofo and set to 2).
  177. // minParticipants: 2,
  178. // Use XEP-0215 to fetch STUN and TURN servers.
  179. // useStunTurn: true,
  180. // Enable IPv6 support.
  181. // useIPv6: true,
  182. // Enables / disables a data communication channel with the Videobridge.
  183. // Values can be 'datachannel', 'websocket', true (treat it as
  184. // 'datachannel'), undefined (treat it as 'datachannel') and false (don't
  185. // open any channel).
  186. // openBridgeChannel: true,
  187. // UI
  188. //
  189. // Use display name as XMPP nickname.
  190. // useNicks: false,
  191. // Require users to always specify a display name.
  192. // requireDisplayName: true,
  193. // Whether to use a welcome page or not. In case it's false a random room
  194. // will be joined when no room is specified.
  195. enableWelcomePage: true,
  196. // Enabling the close page will ignore the welcome page redirection when
  197. // a call is hangup.
  198. // enableClosePage: false,
  199. // Enabling pre join page will add an additional step before starting the meeting,
  200. // where the user can configure its devices and choose the way he
  201. // joins audio (by phone/or web).
  202. // prejoinPageEnabled: false,
  203. // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
  204. // disable1On1Mode: false,
  205. // Default language for the user interface.
  206. // defaultLanguage: 'en',
  207. // If true all users without a token will be considered guests and all users
  208. // with token will be considered non-guests. Only guests will be allowed to
  209. // edit their profile.
  210. enableUserRolesBasedOnToken: false,
  211. // Whether or not some features are checked based on token.
  212. // enableFeaturesBasedOnToken: false,
  213. // Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests.
  214. // lockRoomGuestEnabled: false,
  215. // When enabled the password used for locking a room is restricted to up to the number of digits specified
  216. // roomPasswordNumberOfDigits: 10,
  217. // default: roomPasswordNumberOfDigits: false,
  218. // Message to show the users. Example: 'The service will be down for
  219. // maintenance at 01:00 AM GMT,
  220. // noticeMessage: '',
  221. // Enables calendar integration, depends on googleApiApplicationClientID
  222. // and microsoftApiApplicationClientID
  223. // enableCalendarIntegration: false,
  224. // Stats
  225. //
  226. // Whether to enable stats collection or not in the TraceablePeerConnection.
  227. // This can be useful for debugging purposes (post-processing/analysis of
  228. // the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
  229. // estimation tests.
  230. // gatherStats: false,
  231. // The interval at which PeerConnection.getStats() is called. Defaults to 10000
  232. // pcStatsInterval: 10000,
  233. // To enable sending statistics to callstats.io you must provide the
  234. // Application ID and Secret.
  235. // callStatsID: '',
  236. // callStatsSecret: '',
  237. // enables sending participants display name to callstats
  238. // enableDisplayNameInStats: false,
  239. // enables sending participants email if available to callstats and other analytics
  240. // enableEmailInStats: false,
  241. // Privacy
  242. //
  243. // If third party requests are disabled, no other server will be contacted.
  244. // This means avatars will be locally generated and callstats integration
  245. // will not function.
  246. // disableThirdPartyRequests: false,
  247. // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
  248. //
  249. p2p: {
  250. // Enables peer to peer mode. When enabled the system will try to
  251. // establish a direct connection when there are exactly 2 participants
  252. // in the room. If that succeeds the conference will stop sending data
  253. // through the JVB and use the peer to peer connection instead. When a
  254. // 3rd participant joins the conference will be moved back to the JVB
  255. // connection.
  256. enabled: true,
  257. // Use XEP-0215 to fetch STUN and TURN servers.
  258. // useStunTurn: true,
  259. // The STUN servers that will be used in the peer to peer connections
  260. stunServers: [
  261. // { urls: 'stun:jitsi-meet.example.com:443' },
  262. { urls: 'stun:meet-jit-si-turnrelay.jitsi.net:443' }
  263. ],
  264. // Sets the ICE transport policy for the p2p connection. At the time
  265. // of this writing the list of possible values are 'all' and 'relay',
  266. // but that is subject to change in the future. The enum is defined in
  267. // the WebRTC standard:
  268. // https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
  269. // If not set, the effective value is 'all'.
  270. // iceTransportPolicy: 'all',
  271. // If set to true, it will prefer to use H.264 for P2P calls (if H.264
  272. // is supported).
  273. preferH264: true
  274. // If set to true, disable H.264 video codec by stripping it out of the
  275. // SDP.
  276. // disableH264: false,
  277. // How long we're going to wait, before going back to P2P after the 3rd
  278. // participant has left the conference (to filter out page reload).
  279. // backToP2PDelay: 5
  280. },
  281. analytics: {
  282. // The Google Analytics Tracking ID:
  283. // googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
  284. // The Amplitude APP Key:
  285. // amplitudeAPPKey: '<APP_KEY>'
  286. // Array of script URLs to load as lib-jitsi-meet "analytics handlers".
  287. // scriptURLs: [
  288. // "libs/analytics-ga.min.js", // google-analytics
  289. // "https://example.com/my-custom-analytics.js"
  290. // ],
  291. },
  292. // Information about the jitsi-meet instance we are connecting to, including
  293. // the user region as seen by the server.
  294. deploymentInfo: {
  295. // shard: "shard1",
  296. // region: "europe",
  297. // userRegion: "asia"
  298. },
  299. // Information for the chrome extension banner
  300. // chromeExtensionBanner: {
  301. // // The chrome extension to be installed address
  302. // url: 'https://chrome.google.com/webstore/detail/jitsi-meetings/kglhbbefdnlheedjiejgomgmfplipfeb',
  303. // // Extensions info which allows checking if they are installed or not
  304. // chromeExtensionsInfo: [
  305. // {
  306. // id: 'kglhbbefdnlheedjiejgomgmfplipfeb',
  307. // path: 'jitsi-logo-48x48.png'
  308. // }
  309. // ]
  310. // },
  311. // Local Recording
  312. //
  313. // localRecording: {
  314. // Enables local recording.
  315. // Additionally, 'localrecording' (all lowercase) needs to be added to
  316. // TOOLBAR_BUTTONS in interface_config.js for the Local Recording
  317. // button to show up on the toolbar.
  318. //
  319. // enabled: true,
  320. //
  321. // The recording format, can be one of 'ogg', 'flac' or 'wav'.
  322. // format: 'flac'
  323. //
  324. // },
  325. // Options related to end-to-end (participant to participant) ping.
  326. // e2eping: {
  327. // // The interval in milliseconds at which pings will be sent.
  328. // // Defaults to 10000, set to <= 0 to disable.
  329. // pingInterval: 10000,
  330. //
  331. // // The interval in milliseconds at which analytics events
  332. // // with the measured RTT will be sent. Defaults to 60000, set
  333. // // to <= 0 to disable.
  334. // analyticsInterval: 60000,
  335. // },
  336. // If set, will attempt to use the provided video input device label when
  337. // triggering a screenshare, instead of proceeding through the normal flow
  338. // for obtaining a desktop stream.
  339. // NOTE: This option is experimental and is currently intended for internal
  340. // use only.
  341. // _desktopSharingSourceDevice: 'sample-id-or-label',
  342. // If true, any checks to handoff to another application will be prevented
  343. // and instead the app will continue to display in the current browser.
  344. // disableDeepLinking: false,
  345. // A property to disable the right click context menu for localVideo
  346. // the menu has option to flip the locally seen video for local presentations
  347. // disableLocalVideoFlip: false,
  348. // Deployment specific URLs.
  349. // deploymentUrls: {
  350. // // If specified a 'Help' button will be displayed in the overflow menu with a link to the specified URL for
  351. // // user documentation.
  352. // userDocumentationURL: 'https://docs.example.com/video-meetings.html',
  353. // // If specified a 'Download our apps' button will be displayed in the overflow menu with a link
  354. // // to the specified URL for an app download page.
  355. // downloadAppsUrl: 'https://docs.example.com/our-apps.html'
  356. // },
  357. // Options related to the remote participant menu.
  358. // remoteVideoMenu: {
  359. // // If set to true the 'Kick out' button will be disabled.
  360. // disableKick: true
  361. // },
  362. // If set to true all muting operations of remote participants will be disabled.
  363. // disableRemoteMute: true,
  364. // List of undocumented settings used in jitsi-meet
  365. /**
  366. _immediateReloadThreshold
  367. autoRecord
  368. autoRecordToken
  369. debug
  370. debugAudioLevels
  371. deploymentInfo
  372. dialInConfCodeUrl
  373. dialInNumbersUrl
  374. dialOutAuthUrl
  375. dialOutCodesUrl
  376. disableRemoteControl
  377. displayJids
  378. etherpad_base
  379. externalConnectUrl
  380. firefox_fake_device
  381. googleApiApplicationClientID
  382. iAmRecorder
  383. iAmSipGateway
  384. microsoftApiApplicationClientID
  385. peopleSearchQueryTypes
  386. peopleSearchUrl
  387. requireDisplayName
  388. tokenAuthUrl
  389. */
  390. // List of undocumented settings used in lib-jitsi-meet
  391. /**
  392. _peerConnStatusOutOfLastNTimeout
  393. _peerConnStatusRtcMuteTimeout
  394. abTesting
  395. avgRtpStatsN
  396. callStatsConfIDNamespace
  397. callStatsCustomScriptUrl
  398. desktopSharingSources
  399. disableAEC
  400. disableAGC
  401. disableAP
  402. disableHPF
  403. disableNS
  404. enableLipSync
  405. enableTalkWhileMuted
  406. forceJVB121Ratio
  407. hiddenDomain
  408. ignoreStartMuted
  409. nick
  410. startBitrate
  411. */
  412. // Allow all above example options to include a trailing comma and
  413. // prevent fear when commenting out the last value.
  414. makeJsonParserHappy: 'even if last key had a trailing comma'
  415. // no configuration value should follow this line.
  416. };
  417. /* eslint-enable no-unused-vars, no-var */