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| 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: 'meet.demo.com',
// XMPP MUC domain. FIXME: use XEP-0030 to discover it.
muc: 'conference.meet.demo.com',
// When using authentication, domain for guest users.
// anonymousdomain: 'guest.example.com',
// Domain for authenticated users. Defaults to <domain>.
// authdomain: 'meet.demo.com',
// Jirecon recording component domain.
// jirecon: 'jirecon.meet.demo.com',
// Call control component (Jigasi).
call_control: 'callcontrol.meet.demo.com',
// Focus component domain. Defaults to focus.<domain>.
focus: 'focus.meet.demo.com',
bridge: 'jitsi-videobridge.meet.demo.com'
},
// BOSH URL. FIXME: use XEP-0156 to discover it.
bosh: '//meet.demo.com/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.meet.demo.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
// Enables the test specific features consumed by jitsi-meet-torture
// testMode: 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',
// Whether desktop sharing should be disabled on Firefox.
desktopSharingFirefoxDisabled: false,
// 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,
// Disables or enables TCC (the default is in Jicofo and set to true)
// (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting
// affects congestion control, it practically enables send-side bandwidth
// estimations.
// enableTcc: true,
// Disables or enables REMB (the default is in Jicofo and set to false)
// (draft-alvestrand-rmcat-remb-03). This setting affects congestion
// control, it practically enables recv-side bandwidth estimations. When
// both TCC and REMB are enabled, TCC takes precedence. When both are
// disabled, then bandwidth estimations are disabled.
// enableRemb: false,
// Defines the minimum number of participants to start a call (the default
// is set in Jicofo and set to 2).
// minParticipants: 2,
// 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
autoRecord
autoRecordToken
debug
debugAudioLevels
deploymentInfo
dialInConfCodeUrl
dialInNumbersUrl
dialOutAuthUrl
dialOutCodesUrl
disableRemoteControl
displayJids
enableLocalVideoFlip
etherpad_base
externalConnectUrl
firefox_fake_device
googleApiApplicationClientID
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
*/
};
|