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config.js 8.8KB

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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. // If set to true, disable H.264 video codec by stripping it out of the
  79. // SDP.
  80. //disableH264: false,
  81. // Desktop sharing
  82. // Enable / disable desktop sharing
  83. //disableDesktopSharing: false,
  84. // The ID of the jidesha extension for Chrome.
  85. desktopSharingChromeExtId: null,
  86. // Whether desktop sharing should be disabled on Chrome.
  87. desktopSharingChromeDisabled: true,
  88. // The media sources to use when using screen sharing with the Chrome
  89. // extension.
  90. desktopSharingChromeSources: ['screen', 'window', 'tab'],
  91. // Required version of Chrome extension
  92. desktopSharingChromeMinExtVersion: '0.1',
  93. // The ID of the jidesha extension for Firefox. If null, we assume that no
  94. // extension is required.
  95. desktopSharingFirefoxExtId: null,
  96. // Whether desktop sharing should be disabled on Firefox.
  97. desktopSharingFirefoxDisabled: false,
  98. // The maximum version of Firefox which requires a jidesha extension.
  99. // Example: if set to 41, we will require the extension for Firefox versions
  100. // up to and including 41. On Firefox 42 and higher, we will run without the
  101. // extension.
  102. // If set to -1, an extension will be required for all versions of Firefox.
  103. desktopSharingFirefoxMaxVersionExtRequired: 51,
  104. // The URL to the Firefox extension for desktop sharing.
  105. desktopSharingFirefoxExtensionURL: null,
  106. // Try to start calls with screen-sharing instead of camera video.
  107. //startScreenSharing: false,
  108. // Recording
  109. // Whether to enable recording or not.
  110. //enableRecording: false,
  111. // Type for recording: one of jibri or jirecon.
  112. //recordingType: 'jibri',
  113. // Misc
  114. // Default value for the channel "last N" attribute. -1 for unlimited.
  115. channelLastN: -1,
  116. // Disables or enables RTX (RFC 4588) (defaults to false).
  117. //disableRtx: false,
  118. // Use XEP-0215 to fetch STUN and TURN servers.
  119. //useStunTurn: true,
  120. // Enable IPv6 support.
  121. //useIPv6: true,
  122. // Enables / disables a data communication channel with the Videobridge.
  123. // Values can be 'datachannel', 'websocket', true (treat it as
  124. // 'datachannel'), undefined (treat it as 'datachannel') and false (don't
  125. // open any channel).
  126. //openBridgeChannel: true,
  127. // UI
  128. //
  129. // Use display name as XMPP nickname.
  130. //useNicks: false,
  131. // Require users to always specify a display name.
  132. //requireDisplayName: true,
  133. // Whether to use a welcome page or not. In case it's false a random room
  134. // will be joined when no room is specified.
  135. enableWelcomePage: true,
  136. // Enabling the close page will ignore the welcome page redirection when
  137. // a call is hangup.
  138. //enableClosePage: false,
  139. // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
  140. //disable1On1Mode: false,
  141. // The minimum value a video's height (or width, whichever is smaller) needs
  142. // to be in order to be considered high-definition.
  143. minHDHeight: 540,
  144. // Default language for the user interface.
  145. //defaultLanguage: 'en',
  146. // If true all users without a token will be considered guests and all users
  147. // with token will be considered non-guests. Only guests will be allowed to
  148. // edit their profile.
  149. enableUserRolesBasedOnToken: false,
  150. // Message to show the users. Example: 'The service will be down for
  151. // maintenance at 01:00 AM GMT,
  152. //noticeMessage: '',
  153. // Stats
  154. //
  155. // Whether to enable stats collection or not.
  156. //disableStats: false,
  157. // To enable sending statistics to callstats.io you must provide the
  158. // Application ID and Secret.
  159. //callStatsID: '',
  160. //callStatsSecret: '',
  161. // Privacy
  162. //
  163. // If third party requests are disabled, no other server will be contacted.
  164. // This means avatars will be locally generated and callstats integration
  165. // will not function.
  166. //disableThirdPartyRequests: false,
  167. // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
  168. //
  169. p2p: {
  170. // Enables peer to peer mode. When enabled the system will try to
  171. // establish a direct connection when there are exactly 2 participants
  172. // in the room. If that succeeds the conference will stop sending data
  173. // through the JVB and use the peer to peer connection instead. When a
  174. // 3rd participant joins the conference will be moved back to the JVB
  175. // connection.
  176. enabled: true,
  177. // Use XEP-0215 to fetch STUN and TURN servers.
  178. //useStunTurn: true,
  179. // The STUN servers that will be used in the peer to peer connections
  180. stunServers: [
  181. { urls: "stun:stun.l.google.com:19302" },
  182. { urls: "stun:stun1.l.google.com:19302" },
  183. { urls: "stun:stun2.l.google.com:19302" }
  184. ],
  185. // If set to true, it will prefer to use H.264 for P2P calls (if H.264
  186. // is supported).
  187. preferH264: true
  188. // If set to true, disable H.264 video codec by stripping it out of the
  189. // SDP.
  190. //disableH264: false,
  191. // How long we're going to wait, before going back to P2P after the 3rd
  192. // participant has left the conference (to filter out page reload).
  193. //backToP2PDelay: 5
  194. },
  195. // Information about the jitsi-meet instance we are connecting to, including
  196. // the user region as seen by the server.
  197. //
  198. deploymentInfo: {
  199. //shard: "shard1",
  200. //region: "europe",
  201. //userRegion: "asia"
  202. }
  203. };