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June 1, 2018 14:21
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AV jitsi config
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/* 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: 'www.jitsi.agileventures.org', | |
// XMPP MUC domain. FIXME: use XEP-0030 to discover it. | |
muc: 'conference.www.jitsi.agileventures.org' | |
// When using authentication, domain for guest users. | |
// anonymousdomain: 'guest.example.com', | |
// Domain for authenticated users. Defaults to <domain>. | |
// authdomain: 'www.jitsi.agileventures.org', | |
// Jirecon recording component domain. | |
// jirecon: 'jirecon.www.jitsi.agileventures.org', | |
// Call control component (Jigasi). | |
// call_control: 'callcontrol.www.jitsi.agileventures.org', | |
// Focus component domain. Defaults to focus.<domain>. | |
// focus: 'focus.www.jitsi.agileventures.org', | |
}, | |
// BOSH URL. FIXME: use XEP-0156 to discover it. | |
bosh: '//www.jitsi.agileventures.org/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.www.jitsi.agileventures.org', | |
// 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: "jfcnlhaenjhgmkdkjfmjpnbaionhbmhg", | |
// Whether desktop sharing should be disabled on Chrome. | |
desktopSharingChromeDisabled: false, | |
// 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, | |
// 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. | |
// disableStats: 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" | |
// ], | |
// 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 */ |
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