|
|
@@ -0,0 +1,388 @@
|
|
|
+/* eslint-disable no-unused-vars, no-var */
|
|
|
+
|
|
|
+var config = {
|
|
|
+ // Configuration
|
|
|
+ //
|
|
|
+
|
|
|
+ // Alternative location for the configuration.
|
|
|
+ // configLocation: './config.json',
|
|
|
+
|
|
|
+ // Custom function which given the URL path should return a room name.
|
|
|
+ // getroomnode: function (path) { return 'someprefixpossiblybasedonpath'; },
|
|
|
+
|
|
|
+
|
|
|
+ // Connection
|
|
|
+ //
|
|
|
+
|
|
|
+ hosts: {
|
|
|
+ // XMPP domain.
|
|
|
+ domain: 'jitsi-meet.example.com',
|
|
|
+
|
|
|
+ // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
|
|
|
+ muc: 'conference.jitsi-meet.example.com'
|
|
|
+
|
|
|
+ // When using authentication, domain for guest users.
|
|
|
+ // anonymousdomain: 'guest.example.com',
|
|
|
+
|
|
|
+ // Domain for authenticated users. Defaults to <domain>.
|
|
|
+ // authdomain: 'jitsi-meet.example.com',
|
|
|
+
|
|
|
+ // Jirecon recording component domain.
|
|
|
+ // jirecon: 'jirecon.jitsi-meet.example.com',
|
|
|
+
|
|
|
+ // Call control component (Jigasi).
|
|
|
+ // call_control: 'callcontrol.jitsi-meet.example.com',
|
|
|
+
|
|
|
+ // Focus component domain. Defaults to focus.<domain>.
|
|
|
+ // focus: 'focus.jitsi-meet.example.com',
|
|
|
+ },
|
|
|
+
|
|
|
+ // BOSH URL. FIXME: use XEP-0156 to discover it.
|
|
|
+ bosh: '/http-bind',
|
|
|
+
|
|
|
+ // The name of client node advertised in XEP-0115 'c' stanza
|
|
|
+ clientNode: 'http://jitsi.org/jitsimeet',
|
|
|
+
|
|
|
+ // The real JID of focus participant - can be overridden here
|
|
|
+ // focusUserJid: 'focus@auth.jitsi-meet.example.com',
|
|
|
+
|
|
|
+
|
|
|
+ // Testing / experimental features.
|
|
|
+ //
|
|
|
+
|
|
|
+ testing: {
|
|
|
+ // Enables experimental simulcast support on Firefox.
|
|
|
+ enableFirefoxSimulcast: false,
|
|
|
+
|
|
|
+ // P2P test mode disables automatic switching to P2P when there are 2
|
|
|
+ // participants in the conference.
|
|
|
+ p2pTestMode: false
|
|
|
+ },
|
|
|
+
|
|
|
+ // Disables ICE/UDP by filtering out local and remote UDP candidates in
|
|
|
+ // signalling.
|
|
|
+ // webrtcIceUdpDisable: false,
|
|
|
+
|
|
|
+ // Disables ICE/TCP by filtering out local and remote TCP candidates in
|
|
|
+ // signalling.
|
|
|
+ // webrtcIceTcpDisable: false,
|
|
|
+
|
|
|
+
|
|
|
+ // Media
|
|
|
+ //
|
|
|
+
|
|
|
+ // Audio
|
|
|
+
|
|
|
+ // Disable measuring of audio levels.
|
|
|
+ // disableAudioLevels: false,
|
|
|
+
|
|
|
+ // Start the conference in audio only mode (no video is being received nor
|
|
|
+ // sent).
|
|
|
+ // startAudioOnly: false,
|
|
|
+
|
|
|
+ // Every participant after the Nth will start audio muted.
|
|
|
+ // startAudioMuted: 10,
|
|
|
+
|
|
|
+ // Start calls with audio muted. Unlike the option above, this one is only
|
|
|
+ // applied locally. FIXME: having these 2 options is confusing.
|
|
|
+ // startWithAudioMuted: false,
|
|
|
+
|
|
|
+ // Video
|
|
|
+
|
|
|
+ // Sets the preferred resolution (height) for local video. Defaults to 720.
|
|
|
+ // resolution: 720,
|
|
|
+
|
|
|
+ // w3c spec-compliant video constraints to use for video capture. Currently
|
|
|
+ // used by browsers that return true from lib-jitsi-meet's
|
|
|
+ // util#browser#usesNewGumFlow. The constraints are independency from
|
|
|
+ // this config's resolution value. Defaults to requesting an ideal aspect
|
|
|
+ // ratio of 16:9 with an ideal resolution of 1080p.
|
|
|
+ // constraints: {
|
|
|
+ // video: {
|
|
|
+ // aspectRatio: 16 / 9,
|
|
|
+ // height: {
|
|
|
+ // ideal: 1080,
|
|
|
+ // max: 1080,
|
|
|
+ // min: 240
|
|
|
+ // }
|
|
|
+ // }
|
|
|
+ // },
|
|
|
+
|
|
|
+ // Enable / disable simulcast support.
|
|
|
+ // disableSimulcast: false,
|
|
|
+
|
|
|
+ // Suspend sending video if bandwidth estimation is too low. This may cause
|
|
|
+ // problems with audio playback. Disabled until these are fixed.
|
|
|
+ disableSuspendVideo: true,
|
|
|
+
|
|
|
+ // Every participant after the Nth will start video muted.
|
|
|
+ // startVideoMuted: 10,
|
|
|
+
|
|
|
+ // Start calls with video muted. Unlike the option above, this one is only
|
|
|
+ // applied locally. FIXME: having these 2 options is confusing.
|
|
|
+ // startWithVideoMuted: false,
|
|
|
+
|
|
|
+ // If set to true, prefer to use the H.264 video codec (if supported).
|
|
|
+ // Note that it's not recommended to do this because simulcast is not
|
|
|
+ // supported when using H.264. For 1-to-1 calls this setting is enabled by
|
|
|
+ // default and can be toggled in the p2p section.
|
|
|
+ // preferH264: true,
|
|
|
+
|
|
|
+ // If set to true, disable H.264 video codec by stripping it out of the
|
|
|
+ // SDP.
|
|
|
+ // disableH264: false,
|
|
|
+
|
|
|
+ // Desktop sharing
|
|
|
+
|
|
|
+ // Enable / disable desktop sharing
|
|
|
+ // disableDesktopSharing: false,
|
|
|
+
|
|
|
+ // The ID of the jidesha extension for Chrome.
|
|
|
+ desktopSharingChromeExtId: null,
|
|
|
+
|
|
|
+ // Whether desktop sharing should be disabled on Chrome.
|
|
|
+ desktopSharingChromeDisabled: true,
|
|
|
+
|
|
|
+ // The media sources to use when using screen sharing with the Chrome
|
|
|
+ // extension.
|
|
|
+ desktopSharingChromeSources: [ 'screen', 'window', 'tab' ],
|
|
|
+
|
|
|
+ // Required version of Chrome extension
|
|
|
+ desktopSharingChromeMinExtVersion: '0.1',
|
|
|
+
|
|
|
+ // The ID of the jidesha extension for Firefox. If null, we assume that no
|
|
|
+ // extension is required.
|
|
|
+ desktopSharingFirefoxExtId: null,
|
|
|
+
|
|
|
+ // Whether desktop sharing should be disabled on Firefox.
|
|
|
+ desktopSharingFirefoxDisabled: false,
|
|
|
+
|
|
|
+ // The maximum version of Firefox which requires a jidesha extension.
|
|
|
+ // Example: if set to 41, we will require the extension for Firefox versions
|
|
|
+ // up to and including 41. On Firefox 42 and higher, we will run without the
|
|
|
+ // extension.
|
|
|
+ // If set to -1, an extension will be required for all versions of Firefox.
|
|
|
+ desktopSharingFirefoxMaxVersionExtRequired: 51,
|
|
|
+
|
|
|
+ // The URL to the Firefox extension for desktop sharing.
|
|
|
+ desktopSharingFirefoxExtensionURL: null,
|
|
|
+
|
|
|
+ // Optional desktop sharing frame rate options. Default value: min:5, max:5.
|
|
|
+ // desktopSharingFrameRate: {
|
|
|
+ // min: 5,
|
|
|
+ // max: 5
|
|
|
+ // },
|
|
|
+
|
|
|
+ // Try to start calls with screen-sharing instead of camera video.
|
|
|
+ // startScreenSharing: false,
|
|
|
+
|
|
|
+ // Recording
|
|
|
+
|
|
|
+ // Whether to enable recording or not.
|
|
|
+ // enableRecording: false,
|
|
|
+
|
|
|
+ // Type for recording: one of jibri or jirecon.
|
|
|
+ // recordingType: 'jibri',
|
|
|
+
|
|
|
+ // Misc
|
|
|
+
|
|
|
+ // Default value for the channel "last N" attribute. -1 for unlimited.
|
|
|
+ channelLastN: -1,
|
|
|
+
|
|
|
+ // Disables or enables RTX (RFC 4588) (defaults to false).
|
|
|
+ // disableRtx: false,
|
|
|
+
|
|
|
+ // Use XEP-0215 to fetch STUN and TURN servers.
|
|
|
+ // useStunTurn: true,
|
|
|
+
|
|
|
+ // Enable IPv6 support.
|
|
|
+ // useIPv6: true,
|
|
|
+
|
|
|
+ // Enables / disables a data communication channel with the Videobridge.
|
|
|
+ // Values can be 'datachannel', 'websocket', true (treat it as
|
|
|
+ // 'datachannel'), undefined (treat it as 'datachannel') and false (don't
|
|
|
+ // open any channel).
|
|
|
+ // openBridgeChannel: true,
|
|
|
+
|
|
|
+
|
|
|
+ // UI
|
|
|
+ //
|
|
|
+
|
|
|
+ // Use display name as XMPP nickname.
|
|
|
+ // useNicks: false,
|
|
|
+
|
|
|
+ // Require users to always specify a display name.
|
|
|
+ // requireDisplayName: true,
|
|
|
+
|
|
|
+ // Whether to use a welcome page or not. In case it's false a random room
|
|
|
+ // will be joined when no room is specified.
|
|
|
+ enableWelcomePage: true,
|
|
|
+
|
|
|
+ // Enabling the close page will ignore the welcome page redirection when
|
|
|
+ // a call is hangup.
|
|
|
+ // enableClosePage: false,
|
|
|
+
|
|
|
+ // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
|
|
|
+ // disable1On1Mode: false,
|
|
|
+
|
|
|
+ // The minimum value a video's height (or width, whichever is smaller) needs
|
|
|
+ // to be in order to be considered high-definition.
|
|
|
+ minHDHeight: 540,
|
|
|
+
|
|
|
+ // Default language for the user interface.
|
|
|
+ // defaultLanguage: 'en',
|
|
|
+
|
|
|
+ // If true all users without a token will be considered guests and all users
|
|
|
+ // with token will be considered non-guests. Only guests will be allowed to
|
|
|
+ // edit their profile.
|
|
|
+ enableUserRolesBasedOnToken: false,
|
|
|
+
|
|
|
+ // Message to show the users. Example: 'The service will be down for
|
|
|
+ // maintenance at 01:00 AM GMT,
|
|
|
+ // noticeMessage: '',
|
|
|
+
|
|
|
+
|
|
|
+ // Stats
|
|
|
+ //
|
|
|
+
|
|
|
+ // Whether to enable stats collection or not in the TraceablePeerConnection.
|
|
|
+ // This can be useful for debugging purposes (post-processing/analysis of
|
|
|
+ // the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
|
|
|
+ // estimation tests.
|
|
|
+ // gatherStats: false,
|
|
|
+
|
|
|
+ // To enable sending statistics to callstats.io you must provide the
|
|
|
+ // Application ID and Secret.
|
|
|
+ // callStatsID: '',
|
|
|
+ // callStatsSecret: '',
|
|
|
+
|
|
|
+ // enables callstatsUsername to be reported as statsId and used
|
|
|
+ // by callstats as repoted remote id
|
|
|
+ // enableStatsID: false
|
|
|
+
|
|
|
+ // enables sending participants display name to callstats
|
|
|
+ // enableDisplayNameInStats: false
|
|
|
+
|
|
|
+
|
|
|
+ // Privacy
|
|
|
+ //
|
|
|
+
|
|
|
+ // If third party requests are disabled, no other server will be contacted.
|
|
|
+ // This means avatars will be locally generated and callstats integration
|
|
|
+ // will not function.
|
|
|
+ // disableThirdPartyRequests: false,
|
|
|
+
|
|
|
+
|
|
|
+ // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
|
|
|
+ //
|
|
|
+
|
|
|
+ p2p: {
|
|
|
+ // Enables peer to peer mode. When enabled the system will try to
|
|
|
+ // establish a direct connection when there are exactly 2 participants
|
|
|
+ // in the room. If that succeeds the conference will stop sending data
|
|
|
+ // through the JVB and use the peer to peer connection instead. When a
|
|
|
+ // 3rd participant joins the conference will be moved back to the JVB
|
|
|
+ // connection.
|
|
|
+ enabled: true,
|
|
|
+
|
|
|
+ // Use XEP-0215 to fetch STUN and TURN servers.
|
|
|
+ // useStunTurn: true,
|
|
|
+
|
|
|
+ // The STUN servers that will be used in the peer to peer connections
|
|
|
+ stunServers: [
|
|
|
+ { urls: 'stun:stun.l.google.com:19302' },
|
|
|
+ { urls: 'stun:stun1.l.google.com:19302' },
|
|
|
+ { urls: 'stun:stun2.l.google.com:19302' }
|
|
|
+ ],
|
|
|
+
|
|
|
+ // Sets the ICE transport policy for the p2p connection. At the time
|
|
|
+ // of this writing the list of possible values are 'all' and 'relay',
|
|
|
+ // but that is subject to change in the future. The enum is defined in
|
|
|
+ // the WebRTC standard:
|
|
|
+ // https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
|
|
|
+ // If not set, the effective value is 'all'.
|
|
|
+ // iceTransportPolicy: 'all',
|
|
|
+
|
|
|
+ // If set to true, it will prefer to use H.264 for P2P calls (if H.264
|
|
|
+ // is supported).
|
|
|
+ preferH264: true
|
|
|
+
|
|
|
+ // If set to true, disable H.264 video codec by stripping it out of the
|
|
|
+ // SDP.
|
|
|
+ // disableH264: false,
|
|
|
+
|
|
|
+ // How long we're going to wait, before going back to P2P after the 3rd
|
|
|
+ // participant has left the conference (to filter out page reload).
|
|
|
+ // backToP2PDelay: 5
|
|
|
+ },
|
|
|
+
|
|
|
+ // A list of scripts to load as lib-jitsi-meet "analytics handlers".
|
|
|
+ // analyticsScriptUrls: [
|
|
|
+ // "libs/analytics-ga.js", // google-analytics
|
|
|
+ // "https://example.com/my-custom-analytics.js"
|
|
|
+ // ],
|
|
|
+
|
|
|
+ // The Google Analytics Tracking ID
|
|
|
+ // googleAnalyticsTrackingId = 'your-tracking-id-here-UA-123456-1',
|
|
|
+
|
|
|
+ // Information about the jitsi-meet instance we are connecting to, including
|
|
|
+ // the user region as seen by the server.
|
|
|
+ deploymentInfo: {
|
|
|
+ // shard: "shard1",
|
|
|
+ // region: "europe",
|
|
|
+ // userRegion: "asia"
|
|
|
+ }
|
|
|
+
|
|
|
+
|
|
|
+ // List of undocumented settings used in jitsi-meet
|
|
|
+ /**
|
|
|
+ alwaysVisibleToolbar
|
|
|
+ autoEnableDesktopSharing
|
|
|
+ autoRecord
|
|
|
+ autoRecordToken
|
|
|
+ debug
|
|
|
+ debugAudioLevels
|
|
|
+ deploymentInfo
|
|
|
+ dialInConfCodeUrl
|
|
|
+ dialInNumbersUrl
|
|
|
+ dialOutAuthUrl
|
|
|
+ dialOutCodesUrl
|
|
|
+ disableRemoteControl
|
|
|
+ displayJids
|
|
|
+ enableLocalVideoFlip
|
|
|
+ etherpad_base
|
|
|
+ externalConnectUrl
|
|
|
+ firefox_fake_device
|
|
|
+ iAmRecorder
|
|
|
+ iAmSipGateway
|
|
|
+ peopleSearchQueryTypes
|
|
|
+ peopleSearchUrl
|
|
|
+ requireDisplayName
|
|
|
+ tokenAuthUrl
|
|
|
+ */
|
|
|
+
|
|
|
+ // List of undocumented settings used in lib-jitsi-meet
|
|
|
+ /**
|
|
|
+ _peerConnStatusOutOfLastNTimeout
|
|
|
+ _peerConnStatusRtcMuteTimeout
|
|
|
+ abTesting
|
|
|
+ avgRtpStatsN
|
|
|
+ callStatsConfIDNamespace
|
|
|
+ callStatsCustomScriptUrl
|
|
|
+ desktopSharingSources
|
|
|
+ disableAEC
|
|
|
+ disableAGC
|
|
|
+ disableAP
|
|
|
+ disableHPF
|
|
|
+ disableNS
|
|
|
+ enableLipSync
|
|
|
+ enableTalkWhileMuted
|
|
|
+ forceJVB121Ratio
|
|
|
+ hiddenDomain
|
|
|
+ ignoreStartMuted
|
|
|
+ nick
|
|
|
+ startBitrate
|
|
|
+ */
|
|
|
+};
|
|
|
+
|
|
|
+/* eslint-enable no-unused-vars, no-var */
|
Saúl Ibarra Corretgé
il y a 7 ans