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.

config.js

@@ -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
-         * P2P test mode disables automatic switching to P2P when there are 2
+        // participants in the conference.
-         * participants in the conference.
-         */
         p2pTestMode: false,
     },
-//  getroomnode: function (path) { return 'someprefixpossiblybasedonpath'; },
+
-//  useStunTurn: true, // use XEP-0215 to fetch STUN and TURN server
+    // Disables ICE/UDP by filtering out local and remote UDP candidates in
-//  useIPv6: true, // ipv6 support. use at your own risk
+    // signalling.
-    useNicks: false,
+    //webrtcIceUdpDisable: 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
+    // Disables ICE/TCP by filtering out local and remote TCP candidates in
-    //focusUserJid: 'focus@auth.jitsi-meet.example.com', // The real JID of focus participant - can be overridden here
+    // signalling.
-    //defaultSipNumber: '', // Default SIP number
+    //webrtcIceTcpDisable: false,
-    /**
+
-     * Disables desktop sharing functionality.
+
-     */
+    // Media
-    disableDesktopSharing: false,
+    //
+
+    // 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.
+    // Try to start calls with screen-sharing instead of camera video.
-    webrtcIceUdpDisable: false,
+    //startScreenSharing: false,
-    // Disables ICE/TCP by filtering out local and remote TCP candidates in signalling.
-    webrtcIceTcpDisable: 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,
+    //disable1On1Mode: false,
-    disableStats: false,
+
-    disableAudioLevels: false,
+    // The minimum value a video's height (or width, whichever is smaller) needs
-    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
     // 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,
+    // Message to show the users. Example: 'The service will be down for
-    // disables or enables RTX (RFC 4588) (defaults to false).
+    // maintenance at 01:00 AM GMT,
-    disableRtx: false,
+    //noticeMessage: '',
-    // Sets the preferred resolution (height) for local video. Defaults to 720.
+
-    resolution: 720,
+
+    // 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
+        // Enables peer to peer mode. When enabled the system will try to
-        // direct connection given that there are exactly 2 participants in
+        // establish a direct connection when there are exactly 2 participants
-        // the room. If that succeeds the conference will stop sending data
+        // 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
+        // through the JVB and use the peer to peer connection instead. When a
-        // participant joins the conference will be moved back to the JVB
+        // 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",