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config.js 9.1KB

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  1. /* eslint-disable no-unused-vars, no-var */
  2. var config = { // eslint-disable-line no-unused-vars
  3. // Configuration
  4. //
  5. // Alternative location for the configuration.
  6. // configLocation: './config.json',
  7. // Custom function which given the URL path should return a room name.
  8. // getroomnode: function (path) { return 'someprefixpossiblybasedonpath'; },
  9. // Connection
  10. //
  11. hosts: {
  12. // XMPP domain.
  13. domain: 'jitsi-meet.example.com',
  14. // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
  15. muc: 'conference.jitsi-meet.example.com'
  16. // When using authentication, domain for guest users.
  17. // anonymousdomain: 'guest.example.com',
  18. // Domain for authenticated users. Defaults to <domain>.
  19. // authdomain: 'jitsi-meet.example.com',
  20. // Jirecon recording component domain.
  21. // jirecon: 'jirecon.jitsi-meet.example.com',
  22. // Call control component (Jigasi).
  23. // call_control: 'callcontrol.jitsi-meet.example.com',
  24. // Focus component domain. Defaults to focus.<domain>.
  25. // focus: 'focus.jitsi-meet.example.com',
  26. },
  27. // BOSH URL. FIXME: use XEP-0156 to discover it.
  28. bosh: '//jitsi-meet.example.com/http-bind',
  29. // The name of client node advertised in XEP-0115 'c' stanza
  30. clientNode: 'http://jitsi.org/jitsimeet',
  31. // The real JID of focus participant - can be overridden here
  32. // focusUserJid: 'focus@auth.jitsi-meet.example.com',
  33. // Testing / experimental features.
  34. //
  35. testing: {
  36. // Enables experimental simulcast support on Firefox.
  37. enableFirefoxSimulcast: false,
  38. // P2P test mode disables automatic switching to P2P when there are 2
  39. // participants in the conference.
  40. p2pTestMode: false
  41. },
  42. // Disables ICE/UDP by filtering out local and remote UDP candidates in
  43. // signalling.
  44. // webrtcIceUdpDisable: false,
  45. // Disables ICE/TCP by filtering out local and remote TCP candidates in
  46. // signalling.
  47. // webrtcIceTcpDisable: false,
  48. // Media
  49. //
  50. // Audio
  51. // Disable measuring of audio levels.
  52. // disableAudioLevels: false,
  53. // Start the conference in audio only mode (no video is being received nor
  54. // sent).
  55. // startAudioOnly: false,
  56. // Every participant after the Nth will start audio muted.
  57. // startAudioMuted: 10,
  58. // Start calls with audio muted. Unlike the option above, this one is only
  59. // applied locally. FIXME: having these 2 options is confusing.
  60. // startWithAudioMuted: false,
  61. // Video
  62. // Sets the preferred resolution (height) for local video. Defaults to 720.
  63. // resolution: 720,
  64. // Enable / disable simulcast support.
  65. // disableSimulcast: false,
  66. // Suspend sending video if bandwidth estimation is too low. This may cause
  67. // problems with audio playback. Disabled until these are fixed.
  68. disableSuspendVideo: true,
  69. // Every participant after the Nth will start video muted.
  70. // startVideoMuted: 10,
  71. // Start calls with video muted. Unlike the option above, this one is only
  72. // applied locally. FIXME: having these 2 options is confusing.
  73. // startWithVideoMuted: false,
  74. // If set to true, prefer to use the H.264 video codec (if supported).
  75. // Note that it's not recommended to do this because simulcast is not
  76. // supported when using H.264. For 1-to-1 calls this setting is enabled by
  77. // default and can be toggled in the p2p section.
  78. // preferH264: true,
  79. // If set to true, disable H.264 video codec by stripping it out of the
  80. // SDP.
  81. // disableH264: false,
  82. // Desktop sharing
  83. // Enable / disable desktop sharing
  84. // disableDesktopSharing: false,
  85. // The ID of the jidesha extension for Chrome.
  86. desktopSharingChromeExtId: null,
  87. // Whether desktop sharing should be disabled on Chrome.
  88. desktopSharingChromeDisabled: true,
  89. // The media sources to use when using screen sharing with the Chrome
  90. // extension.
  91. desktopSharingChromeSources: [ 'screen', 'window', 'tab' ],
  92. // Required version of Chrome extension
  93. desktopSharingChromeMinExtVersion: '0.1',
  94. // The ID of the jidesha extension for Firefox. If null, we assume that no
  95. // extension is required.
  96. desktopSharingFirefoxExtId: null,
  97. // Whether desktop sharing should be disabled on Firefox.
  98. desktopSharingFirefoxDisabled: false,
  99. // The maximum version of Firefox which requires a jidesha extension.
  100. // Example: if set to 41, we will require the extension for Firefox versions
  101. // up to and including 41. On Firefox 42 and higher, we will run without the
  102. // extension.
  103. // If set to -1, an extension will be required for all versions of Firefox.
  104. desktopSharingFirefoxMaxVersionExtRequired: 51,
  105. // The URL to the Firefox extension for desktop sharing.
  106. desktopSharingFirefoxExtensionURL: null,
  107. // Try to start calls with screen-sharing instead of camera video.
  108. // startScreenSharing: false,
  109. // Recording
  110. // Whether to enable recording or not.
  111. // enableRecording: false,
  112. // Type for recording: one of jibri or jirecon.
  113. // recordingType: 'jibri',
  114. // Misc
  115. // Default value for the channel "last N" attribute. -1 for unlimited.
  116. channelLastN: -1,
  117. // Disables or enables RTX (RFC 4588) (defaults to false).
  118. // disableRtx: false,
  119. // Use XEP-0215 to fetch STUN and TURN servers.
  120. // useStunTurn: true,
  121. // Enable IPv6 support.
  122. // useIPv6: true,
  123. // Enables / disables a data communication channel with the Videobridge.
  124. // Values can be 'datachannel', 'websocket', true (treat it as
  125. // 'datachannel'), undefined (treat it as 'datachannel') and false (don't
  126. // open any channel).
  127. // openBridgeChannel: true,
  128. // UI
  129. //
  130. // Use display name as XMPP nickname.
  131. // useNicks: false,
  132. // Require users to always specify a display name.
  133. // requireDisplayName: true,
  134. // Whether to use a welcome page or not. In case it's false a random room
  135. // will be joined when no room is specified.
  136. enableWelcomePage: true,
  137. // Enabling the close page will ignore the welcome page redirection when
  138. // a call is hangup.
  139. // enableClosePage: false,
  140. // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
  141. // disable1On1Mode: false,
  142. // The minumum value a video's height (or width, whichever is smaller) needs
  143. // The minimum value a video's height (or width, whichever is smaller) needs
  144. // to be in order to be considered high-definition.
  145. minHDHeight: 540,
  146. // Default language for the user interface.
  147. // defaultLanguage: 'en',
  148. // If true all users without a token will be considered guests and all users
  149. // with token will be considered non-guests. Only guests will be allowed to
  150. // edit their profile.
  151. enableUserRolesBasedOnToken: false,
  152. // Message to show the users. Example: 'The service will be down for
  153. // maintenance at 01:00 AM GMT,
  154. // noticeMessage: '',
  155. // Stats
  156. //
  157. // Whether to enable stats collection or not.
  158. // disableStats: false,
  159. // To enable sending statistics to callstats.io you must provide the
  160. // Application ID and Secret.
  161. // callStatsID: '',
  162. // callStatsSecret: '',
  163. // Privacy
  164. //
  165. // If third party requests are disabled, no other server will be contacted.
  166. // This means avatars will be locally generated and callstats integration
  167. // will not function.
  168. // disableThirdPartyRequests: false,
  169. // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
  170. //
  171. p2p: {
  172. // Enables peer to peer mode. When enabled the system will try to
  173. // establish a direct connection when there are exactly 2 participants
  174. // in the room. If that succeeds the conference will stop sending data
  175. // through the JVB and use the peer to peer connection instead. When a
  176. // 3rd participant joins the conference will be moved back to the JVB
  177. // connection.
  178. enabled: true,
  179. // Use XEP-0215 to fetch STUN and TURN servers.
  180. // useStunTurn: true,
  181. // The STUN servers that will be used in the peer to peer connections
  182. stunServers: [
  183. { urls: 'stun:stun.l.google.com:19302' },
  184. { urls: 'stun:stun1.l.google.com:19302' },
  185. { urls: 'stun:stun2.l.google.com:19302' }
  186. ],
  187. // If set to true, it will prefer to use H.264 for P2P calls (if H.264
  188. // is supported).
  189. preferH264: true
  190. // If set to true, disable H.264 video codec by stripping it out of the
  191. // SDP.
  192. // disableH264: false,
  193. // How long we're going to wait, before going back to P2P after the 3rd
  194. // participant has left the conference (to filter out page reload).
  195. // backToP2PDelay: 5
  196. },
  197. // Information about the jitsi-meet instance we are connecting to, including
  198. // the user region as seen by the server.
  199. //
  200. deploymentInfo: {
  201. // shard: "shard1",
  202. // region: "europe",
  203. // userRegion: "asia"
  204. }
  205. };
  206. /* eslint-enable no-unused-vars, no-var */