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config.js 6.7KB

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  1. /* jshint maxlen:false */
  2. var config = { // eslint-disable-line no-unused-vars
  3. // configLocation: './config.json', // see ./modules/HttpConfigFetch.js
  4. hosts: {
  5. domain: 'jitsi-meet.example.com',
  6. //anonymousdomain: 'guest.example.com',
  7. //authdomain: 'jitsi-meet.example.com', // defaults to <domain>
  8. muc: 'conference.jitsi-meet.example.com', // FIXME: use XEP-0030
  9. //jirecon: 'jirecon.jitsi-meet.example.com',
  10. //call_control: 'callcontrol.jitsi-meet.example.com',
  11. //focus: 'focus.jitsi-meet.example.com', // defaults to 'focus.jitsi-meet.example.com'
  12. },
  13. testing: {
  14. /**
  15. * Enables experimental simulcast support on Firefox.
  16. */
  17. enableFirefoxSimulcast: false,
  18. /**
  19. * P2P test mode disables automatic switching to P2P when there are 2
  20. * participants in the conference.
  21. */
  22. p2pTestMode: false,
  23. },
  24. // getroomnode: function (path) { return 'someprefixpossiblybasedonpath'; },
  25. // useStunTurn: true, // use XEP-0215 to fetch STUN and TURN server
  26. // useIPv6: true, // ipv6 support. use at your own risk
  27. useNicks: false,
  28. bosh: '//jitsi-meet.example.com/http-bind', // FIXME: use xep-0156 for that
  29. clientNode: 'http://jitsi.org/jitsimeet', // The name of client node advertised in XEP-0115 'c' stanza
  30. //focusUserJid: 'focus@auth.jitsi-meet.example.com', // The real JID of focus participant - can be overridden here
  31. //defaultSipNumber: '', // Default SIP number
  32. /**
  33. * Disables desktop sharing functionality.
  34. */
  35. disableDesktopSharing: false,
  36. // The ID of the jidesha extension for Chrome.
  37. desktopSharingChromeExtId: null,
  38. // Whether desktop sharing should be disabled on Chrome.
  39. desktopSharingChromeDisabled: true,
  40. // The media sources to use when using screen sharing with the Chrome
  41. // extension.
  42. desktopSharingChromeSources: ['screen', 'window', 'tab'],
  43. // Required version of Chrome extension
  44. desktopSharingChromeMinExtVersion: '0.1',
  45. // The ID of the jidesha extension for Firefox. If null, we assume that no
  46. // extension is required.
  47. desktopSharingFirefoxExtId: null,
  48. // Whether desktop sharing should be disabled on Firefox.
  49. desktopSharingFirefoxDisabled: false,
  50. // The maximum version of Firefox which requires a jidesha extension.
  51. // Example: if set to 41, we will require the extension for Firefox versions
  52. // up to and including 41. On Firefox 42 and higher, we will run without the
  53. // extension.
  54. // If set to -1, an extension will be required for all versions of Firefox.
  55. desktopSharingFirefoxMaxVersionExtRequired: 51,
  56. // The URL to the Firefox extension for desktop sharing.
  57. desktopSharingFirefoxExtensionURL: null,
  58. // Disables ICE/UDP by filtering out local and remote UDP candidates in signalling.
  59. webrtcIceUdpDisable: false,
  60. // Disables ICE/TCP by filtering out local and remote TCP candidates in signalling.
  61. webrtcIceTcpDisable: false,
  62. openSctp: true, // Toggle to enable/disable SCTP channels
  63. // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
  64. disable1On1Mode: false,
  65. disableStats: false,
  66. disableAudioLevels: false,
  67. channelLastN: -1, // The default value of the channel attribute last-n.
  68. enableRecording: false,
  69. enableWelcomePage: true,
  70. //enableClosePage: false, // enabling the close page will ignore the welcome
  71. // page redirection when call is hangup
  72. disableSimulcast: false,
  73. // requireDisplayName: true, // Forces the participants that doesn't have display name to enter it when they enter the room.
  74. startAudioOnly: false, // Will start the conference in the audio only mode (no video is being received nor sent)
  75. startScreenSharing: false, // Will try to start with screensharing instead of camera
  76. // startAudioMuted: 10, // every participant after the Nth will start audio muted
  77. // startVideoMuted: 10, // every participant after the Nth will start video muted
  78. startWithAudioMuted: false, // will start with the microphone muted
  79. startWithVideoMuted: false, // will start with the camera turned off
  80. // defaultLanguage: "en",
  81. // To enable sending statistics to callstats.io you should provide Applicaiton ID and Secret.
  82. // callStatsID: "", // Application ID for callstats.io API
  83. // callStatsSecret: "", // Secret for callstats.io API
  84. /*noticeMessage: 'Service update is scheduled for 16th March 2015. ' +
  85. 'During that time service will not be available. ' +
  86. 'Apologise for inconvenience.',*/
  87. disableThirdPartyRequests: false,
  88. // The minumum value a video's height (or width, whichever is smaller) needs
  89. // to be in order to be considered high-definition.
  90. minHDHeight: 540,
  91. // If true - all users without token will be considered guests and all users
  92. // with token will be considered non-guests. Only guests will be allowed to
  93. // edit their profile.
  94. enableUserRolesBasedOnToken: false,
  95. // Suspending video might cause problems with audio playback. Disabling until these are fixed.
  96. disableSuspendVideo: true,
  97. // disables or enables RTX (RFC 4588) (defaults to false).
  98. disableRtx: false,
  99. // Sets the preferred resolution (height) for local video. Defaults to 720.
  100. resolution: 720,
  101. // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
  102. p2p: {
  103. // Enables peer to peer mode. When enabled system will try to establish
  104. // direct connection given that there are exactly 2 participants in
  105. // the room. If that succeeds the conference will stop sending data
  106. // through the JVB and use the peer to peer connection instead. When 3rd
  107. // participant joins the conference will be moved back to the JVB
  108. // connection.
  109. enabled: true,
  110. // The STUN servers that will be used in the peer to peer connections
  111. // useStunTurn: true, // use XEP-0215 to fetch STUN and TURN server
  112. stunServers: [
  113. { urls: "stun:stun.l.google.com:19302" },
  114. { urls: "stun:stun1.l.google.com:19302" },
  115. { urls: "stun:stun2.l.google.com:19302" }
  116. ],
  117. // If set to true, it will prefer to use H.264 for P2P calls (if H.264
  118. // is supported).
  119. preferH264: true
  120. // How long we're going to wait, before going back to P2P after
  121. // the 3rd participant has left the conference (to filter out page reload)
  122. //backToP2PDelay: 5
  123. },
  124. // Information about the jitsi-meet instance we are connecting to, including the
  125. // user region as seen by the server.
  126. deploymentInfo: {
  127. //shard: "shard1",
  128. //region: "europe",
  129. //userRegion: "asia"
  130. }
  131. };