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config.js 8.6KB

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  1. var config = { // eslint-disable-line no-unused-vars
  2. // Configuration
  3. //
  4. // Alternative location for the configuration.
  5. //configLocation: './config.json',
  6. // Custom function which given the URL path should return a room name.
  7. //getroomnode: function (path) { return 'someprefixpossiblybasedonpath'; },
  8. // Connection
  9. //
  10. hosts: {
  11. // XMPP domain.
  12. domain: 'jitsi-meet.example.com',
  13. // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
  14. muc: 'conference.jitsi-meet.example.com',
  15. // When using authentication, domain for guest users.
  16. //anonymousdomain: 'guest.example.com',
  17. // Domain for authenticated users. Defaults to <domain>.
  18. //authdomain: 'jitsi-meet.example.com',
  19. // Jirecon recording component domain.
  20. //jirecon: 'jirecon.jitsi-meet.example.com',
  21. // Call control component (Jigasi).
  22. //call_control: 'callcontrol.jitsi-meet.example.com',
  23. // Focus component domain. Defaults to focus.<domain>.
  24. //focus: 'focus.jitsi-meet.example.com',
  25. },
  26. // BOSH URL. FIXME: use XEP-0156 to discover it.
  27. bosh: '//jitsi-meet.example.com/http-bind',
  28. // The name of client node advertised in XEP-0115 'c' stanza
  29. clientNode: 'http://jitsi.org/jitsimeet',
  30. // The real JID of focus participant - can be overridden here
  31. //focusUserJid: 'focus@auth.jitsi-meet.example.com',
  32. // Testing / experimental features.
  33. //
  34. testing: {
  35. // Enables experimental simulcast support on Firefox.
  36. enableFirefoxSimulcast: false,
  37. // P2P test mode disables automatic switching to P2P when there are 2
  38. // participants in the conference.
  39. p2pTestMode: false,
  40. },
  41. // Disables ICE/UDP by filtering out local and remote UDP candidates in
  42. // signalling.
  43. //webrtcIceUdpDisable: false,
  44. // Disables ICE/TCP by filtering out local and remote TCP candidates in
  45. // signalling.
  46. //webrtcIceTcpDisable: false,
  47. // Media
  48. //
  49. // Audio
  50. // Disable measuring of audio levels.
  51. //disableAudioLevels: false,
  52. // Start the conference in audio only mode (no video is being received nor
  53. // sent).
  54. //startAudioOnly: false,
  55. // Every participant after the Nth will start audio muted.
  56. //startAudioMuted: 10,
  57. // Start calls with audio muted. Unlike the option above, this one is only
  58. // applied locally. FIXME: having these 2 options is confusing.
  59. //startWithAudioMuted: false,
  60. // Video
  61. // Sets the preferred resolution (height) for local video. Defaults to 720.
  62. //resolution: 720,
  63. // Enable / disable simulcast support.
  64. //disableSimulcast: false,
  65. // Suspend sending video if bandwidth estimation is too low. This may cause
  66. // problems with audio playback. Disabled until these are fixed.
  67. disableSuspendVideo: true,
  68. // Every participant after the Nth will start video muted.
  69. //startVideoMuted: 10,
  70. // Start calls with video muted. Unlike the option above, this one is only
  71. // applied locally. FIXME: having these 2 options is confusing.
  72. //startWithVideoMuted: false,
  73. // If set to true, prefer to use the H.264 video codec (if supported).
  74. // Note that it's not recommended to do this because simulcast is not
  75. // supported when using H.264. For 1-to-1 calls this setting is enabled by
  76. // default and can be toggled in the p2p section.
  77. //preferH264: true,
  78. // Desktop sharing
  79. // Enable / disable desktop sharing
  80. //disableDesktopSharing: false,
  81. // The ID of the jidesha extension for Chrome.
  82. desktopSharingChromeExtId: null,
  83. // Whether desktop sharing should be disabled on Chrome.
  84. desktopSharingChromeDisabled: true,
  85. // The media sources to use when using screen sharing with the Chrome
  86. // extension.
  87. desktopSharingChromeSources: ['screen', 'window', 'tab'],
  88. // Required version of Chrome extension
  89. desktopSharingChromeMinExtVersion: '0.1',
  90. // The ID of the jidesha extension for Firefox. If null, we assume that no
  91. // extension is required.
  92. desktopSharingFirefoxExtId: null,
  93. // Whether desktop sharing should be disabled on Firefox.
  94. desktopSharingFirefoxDisabled: false,
  95. // The maximum version of Firefox which requires a jidesha extension.
  96. // Example: if set to 41, we will require the extension for Firefox versions
  97. // up to and including 41. On Firefox 42 and higher, we will run without the
  98. // extension.
  99. // If set to -1, an extension will be required for all versions of Firefox.
  100. desktopSharingFirefoxMaxVersionExtRequired: 51,
  101. // The URL to the Firefox extension for desktop sharing.
  102. desktopSharingFirefoxExtensionURL: null,
  103. // Try to start calls with screen-sharing instead of camera video.
  104. //startScreenSharing: false,
  105. // Recording
  106. // Whether to enable recording or not.
  107. //enableRecording: false,
  108. // Type for recording: one of jibri or jirecon.
  109. //recordingType: 'jibri',
  110. // Misc
  111. // Default value for the channel "last N" attribute. -1 for unlimited.
  112. channelLastN: -1,
  113. // Disables or enables RTX (RFC 4588) (defaults to false).
  114. //disableRtx: false,
  115. // Use XEP-0215 to fetch STUN and TURN servers.
  116. //useStunTurn: true,
  117. // Enable IPv6 support.
  118. //useIPv6: true,
  119. // Enables / disables a data communication channel with the Videobridge.
  120. // Values can be 'datachannel', 'websocket', true (treat it as
  121. // 'datachannel'), undefined (treat it as 'datachannel') and false (don't
  122. // open any channel).
  123. //openBridgeChannel: true,
  124. // UI
  125. //
  126. // Use display name as XMPP nickname.
  127. //useNicks: false,
  128. // Require users to always specify a display name.
  129. //requireDisplayName: true,
  130. // Whether to use a welcome page or not. In case it's false a random room
  131. // will be joined when no room is specified.
  132. enableWelcomePage: true,
  133. // Enabling the close page will ignore the welcome page redirection when
  134. // a call is hangup.
  135. //enableClosePage: false,
  136. // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
  137. //disable1On1Mode: false,
  138. // The minimum value a video's height (or width, whichever is smaller) needs
  139. // to be in order to be considered high-definition.
  140. minHDHeight: 540,
  141. // Default language for the user interface.
  142. //defaultLanguage: 'en',
  143. // If true all users without a token will be considered guests and all users
  144. // with token will be considered non-guests. Only guests will be allowed to
  145. // edit their profile.
  146. enableUserRolesBasedOnToken: false,
  147. // Message to show the users. Example: 'The service will be down for
  148. // maintenance at 01:00 AM GMT,
  149. //noticeMessage: '',
  150. // Stats
  151. //
  152. // Whether to enable stats collection or not.
  153. //disableStats: false,
  154. // To enable sending statistics to callstats.io you must provide the
  155. // Application ID and Secret.
  156. //callStatsID: '',
  157. //callStatsSecret: '',
  158. // Privacy
  159. //
  160. // If third party requests are disabled, no other server will be contacted.
  161. // This means avatars will be locally generated and callstats integration
  162. // will not function.
  163. //disableThirdPartyRequests: false,
  164. // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
  165. //
  166. p2p: {
  167. // Enables peer to peer mode. When enabled the system will try to
  168. // establish a direct connection when there are exactly 2 participants
  169. // in the room. If that succeeds the conference will stop sending data
  170. // through the JVB and use the peer to peer connection instead. When a
  171. // 3rd participant joins the conference will be moved back to the JVB
  172. // connection.
  173. enabled: true,
  174. // Use XEP-0215 to fetch STUN and TURN servers.
  175. //useStunTurn: true,
  176. // The STUN servers that will be used in the peer to peer connections
  177. stunServers: [
  178. { urls: "stun:stun.l.google.com:19302" },
  179. { urls: "stun:stun1.l.google.com:19302" },
  180. { urls: "stun:stun2.l.google.com:19302" }
  181. ],
  182. // If set to true, it will prefer to use H.264 for P2P calls (if H.264
  183. // is supported).
  184. preferH264: true
  185. // How long we're going to wait, before going back to P2P after the 3rd
  186. // participant has left the conference (to filter out page reload).
  187. //backToP2PDelay: 5
  188. },
  189. // Information about the jitsi-meet instance we are connecting to, including
  190. // the user region as seen by the server.
  191. //
  192. deploymentInfo: {
  193. //shard: "shard1",
  194. //region: "europe",
  195. //userRegion: "asia"
  196. }
  197. };