haskal
/
jiti-meet
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity

feat(config): reorganize and thoroughly document config.js 

- Document each option.
- Group options semantically.
- Remove no longer existing options.
- Add some missing options.
- Fix punctuation here and there.

NOTE: This is a first step towards a full configuration overhaul, once it
materializes each of the "semantic sections" will be a JS object.
This commit is contained in:
Saúl Ibarra Corretgé 2017-08-07 13:51:07 +02:00 committed by Saúl Ibarra Corretgé
parent d8cd3e75b4
commit 19d9b3f023
1 changed files with 214 additions and 72 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

286
config.js
View File

@ -1,133 +1,275 @@
/* jshint maxlen:false */
var config = { // eslint-disable-line no-unused-vars
// configLocation: './config.json', // see ./modules/HttpConfigFetch.js
// 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',
//authdomain: 'jitsi-meet.example.com', // defaults to <domain>
muc: 'conference.jitsi-meet.example.com', // FIXME: use XEP-0030
// 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: 'focus.jitsi-meet.example.com', // defaults to 'focus.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: '//jitsi-meet.example.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.jitsi-meet.example.com',
// Testing / experimental features.
//
testing: {
/**
* Enables experimental simulcast support on Firefox.
*/
// Enables experimental simulcast support on Firefox.
enableFirefoxSimulcast: false,
/**
* P2P test mode disables automatic switching to P2P when there are 2
* participants in the conference.
*/
// P2P test mode disables automatic switching to P2P when there are 2
// participants in the conference.
p2pTestMode: false,
},
// getroomnode: function (path) { return 'someprefixpossiblybasedonpath'; },
// useStunTurn: true, // use XEP-0215 to fetch STUN and TURN server
// useIPv6: true, // ipv6 support. use at your own risk
useNicks: false,
bosh: '//jitsi-meet.example.com/http-bind', // FIXME: use xep-0156 for that
clientNode: 'http://jitsi.org/jitsimeet', // The name of client node advertised in XEP-0115 'c' stanza
//focusUserJid: 'focus@auth.jitsi-meet.example.com', // The real JID of focus participant - can be overridden here
//defaultSipNumber: '', // Default SIP number
/**
* Disables desktop sharing functionality.
*/
disableDesktopSharing: 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,
// 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,
// 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,
// 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,
// Try to start calls with screen-sharing instead of camera video.
//startScreenSharing: false,
openSctp: true, // Toggle to enable/disable SCTP channels
// 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,
disableStats: false,
disableAudioLevels: false,
channelLastN: -1, // The default value of the channel attribute last-n.
enableRecording: false,
enableWelcomePage: true,
//enableClosePage: false, // enabling the close page will ignore the welcome
// page redirection when call is hangup
disableSimulcast: false,
// requireDisplayName: true, // Forces the participants that doesn't have display name to enter it when they enter the room.
startAudioOnly: false, // Will start the conference in the audio only mode (no video is being received nor sent)
startScreenSharing: false, // Will try to start with screensharing instead of camera
// startAudioMuted: 10, // every participant after the Nth will start audio muted
// startVideoMuted: 10, // every participant after the Nth will start video muted
startWithAudioMuted: false, // will start with the microphone muted
startWithVideoMuted: false, // will start with the camera turned off
// defaultLanguage: "en",
// To enable sending statistics to callstats.io you should provide Applicaiton ID and Secret.
// callStatsID: "", // Application ID for callstats.io API
// callStatsSecret: "", // Secret for callstats.io API
/*noticeMessage: 'Service update is scheduled for 16th March 2015. ' +
'During that time service will not be available. ' +
'Apologise for inconvenience.',*/
disableThirdPartyRequests: false,
// The minumum value a video's height (or width, whichever is smaller) needs
//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,
// If true - all users without token will be considered guests and all users
// 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,
// Suspending video might cause problems with audio playback. Disabling until these are fixed.
disableSuspendVideo: true,
// disables or enables RTX (RFC 4588) (defaults to false).
disableRtx: false,
// Sets the preferred resolution (height) for local video. Defaults to 720.
resolution: 720,
// 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: '',
// 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 system will try to establish
// direct connection given that 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 3rd
// participant joins the conference will be moved back to the JVB
// 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
// useStunTurn: true, // use XEP-0215 to fetch STUN and TURN server
stunServers: [
{ urls: "stun:stun.l.google.com:19302" },
{ urls: "stun:stun1.l.google.com:19302" },
{ urls: "stun:stun2.l.google.com:19302" }
],
// If set to true, it will prefer to use H.264 for P2P calls (if H.264
// is supported).
preferH264: true
// 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)
// 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
},
// Information about the jitsi-meet instance we are connecting to, including the
// user region as seen by the server.
// Information about the jitsi-meet instance we are connecting to, including
// the user region as seen by the server.
//
deploymentInfo: {
//shard: "shard1",
//region: "europe",