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