auto acp scripted

fqdn_conf_i0.js

@@ -0,0 +1,541 @@
+/* eslint-disable no-unused-vars, no-var */
+
+
+subdomain_test = `<!--# echo var="subdomain" default="" -->`
+
+var config = {
+    // Connection
+    //
+
+    hosts: {
+        // XMPP domain.
+        domain: 'jinnace.com',
+
+        // When using authentication, domain for guest users.
+        // anonymousdomain: 'guest.example.com',
+
+        // Domain for authenticated users. Defaults to <domain>.
+        // authdomain: 'jinnace.com',
+
+        // Jirecon recording component domain.
+        // jirecon: 'jirecon.jinnace.com',
+
+        // Call control component (Jigasi).
+        // call_control: 'callcontrol.jinnace.com',
+
+        // Focus component domain. Defaults to focus.<domain>.
+        // focus: 'focus.jinnace.com',
+
+        // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
+        muc: 'conference.<!--# echo var="subdomain" default="" -->jinnace.com'
+    },
+
+    // BOSH URL. FIXME: use XEP-0156 to discover it.
+    bosh: '//jinnace.com/http-bind',
+
+    // Websocket URL
+    // websocket: 'wss://jinnace.com/xmpp-websocket',
+
+    // 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.jinnace.com',
+
+
+    // Testing / experimental features.
+    //
+
+    testing: {
+        // 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 the auto-play behavior of *all* newly created video element.
+        // This is useful when the client runs on a host with limited resources.
+        // noAutoPlayVideo: 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,
+    // audioLevelsInterval: 200,
+
+    // Enabling this will run the lib-jitsi-meet no audio detection module which
+    // will notify the user if the current selected microphone has no audio
+    // input and will suggest another valid device if one is present.
+    enableNoAudioDetection: true,
+
+    // Enabling this will run the lib-jitsi-meet noise detection module which will
+    // notify the user if there is noise, other than voice, coming from the current
+    // selected microphone. The purpose it to let the user know that the input could
+    // be potentially unpleasant for other meeting participants.
+    enableNoisyMicDetection: true,
+
+    // 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,
+    startWithAudioMuted: true,
+
+    // Enabling it (with #params) will disable local audio output of remote
+    // participants and to enable it back a reload is needed.
+    // startSilent: 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 independent from
+    // this config's resolution value. Defaults to requesting an ideal
+    // resolution of 720p.
+    // constraints: {
+    //     video: {
+    //         height: {
+    //             ideal: 720,
+    //             max: 720,
+    //             min: 240
+    //         }
+    //     }
+    // },
+
+    // Enable / disable simulcast support.
+    // disableSimulcast: false,
+
+    // Enable / disable layer suspension.  If enabled, endpoints whose HD
+    // layers are not in use will be suspended (no longer sent) until they
+    // are requested again.
+    // enableLayerSuspension: false,
+
+    // 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,
+    startWithVideoMuted: true,
+
+    // 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
+
+    // The ID of the jidesha extension for Chrome.
+    desktopSharingChromeExtId: null,
+
+    // 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',
+
+    // 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 file recording or not.
+    // fileRecordingsEnabled: false,
+    // Enable the dropbox integration.
+    // dropbox: {
+    //     appKey: '<APP_KEY>' // Specify your app key here.
+    //     // A URL to redirect the user to, after authenticating
+    //     // by default uses:
+    //     // 'https://jinnace.com/static/oauth.html'
+    //     redirectURI:
+    //          'https://jinnace.com/subfolder/static/oauth.html'
+    // },
+    // When integrations like dropbox are enabled only that will be shown,
+    // by enabling fileRecordingsServiceEnabled, we show both the integrations
+    // and the generic recording service (its configuration and storage type
+    // depends on jibri configuration)
+    // fileRecordingsServiceEnabled: false,
+    // Whether to show the possibility to share file recording with other people
+    // (e.g. meeting participants), based on the actual implementation
+    // on the backend.
+    // fileRecordingsServiceSharingEnabled: false,
+
+    // Whether to enable live streaming or not.
+    // liveStreamingEnabled: false,
+
+    // Transcription (in interface_config,
+    // subtitles and buttons can be configured)
+    // transcribingEnabled: false,
+
+    // Enables automatic turning on captions when recording is started
+    // autoCaptionOnRecord: false,
+
+    // 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,
+
+    // 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,
+
+    // Whether or not some features are checked based on token.
+    // enableFeaturesBasedOnToken: false,
+
+    // Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests.
+    // lockRoomGuestEnabled: false,
+
+    // When enabled the password used for locking a room is restricted to up to the number of digits specified
+    // roomPasswordNumberOfDigits: 10,
+    // default: roomPasswordNumberOfDigits: false,
+
+    // Message to show the users. Example: 'The service will be down for
+    // maintenance at 01:00 AM GMT,
+    // noticeMessage: '',
+
+    // Enables calendar integration, depends on googleApiApplicationClientID
+    // and microsoftApiApplicationClientID
+    // enableCalendarIntegration: false,
+
+    // 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,
+
+    // The interval at which PeerConnection.getStats() is called. Defaults to 10000
+    // pcStatsInterval: 10000,
+
+    // To enable sending statistics to callstats.io you must provide the
+    // Application ID and Secret.
+    // callStatsID: '',
+    // callStatsSecret: '',
+
+    // enables sending participants display name to callstats
+    // enableDisplayNameInStats: false,
+
+    // enables sending participants email if available to callstats and other analytics
+    // enableEmailInStats: 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:jinnace.com:4446' },
+            { urls: 'stun:meet-jit-si-turnrelay.jitsi.net:443' }
+        ],
+
+        // 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
+    },
+
+    analytics: {
+        // The Google Analytics Tracking ID:
+        // googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
+
+        // The Amplitude APP Key:
+        // amplitudeAPPKey: '<APP_KEY>'
+
+        // Array of script URLs to load as lib-jitsi-meet "analytics handlers".
+        // scriptURLs: [
+        //      "libs/analytics-ga.min.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"
+    },
+
+    // Decides whether the start/stop recording audio notifications should play on record.
+    // disableRecordAudioNotification: false,
+
+    // Information for the chrome extension banner
+    // chromeExtensionBanner: {
+    //     // The chrome extension to be installed address
+    //     url: 'https://chrome.google.com/webstore/detail/jitsi-meetings/kglhbbefdnlheedjiejgomgmfplipfeb',
+
+    //     // Extensions info which allows checking if they are installed or not
+    //     chromeExtensionsInfo: [
+    //         {
+    //             id: 'kglhbbefdnlheedjiejgomgmfplipfeb',
+    //             path: 'jitsi-logo-48x48.png'
+    //         }
+    //     ]
+    // },
+
+    // Local Recording
+    //
+
+    // localRecording: {
+    // Enables local recording.
+    // Additionally, 'localrecording' (all lowercase) needs to be added to
+    // TOOLBAR_BUTTONS in interface_config.js for the Local Recording
+    // button to show up on the toolbar.
+    //
+    //     enabled: true,
+    //
+
+    // The recording format, can be one of 'ogg', 'flac' or 'wav'.
+    //     format: 'flac'
+    //
+
+    // },
+
+    // Options related to end-to-end (participant to participant) ping.
+    // e2eping: {
+    //   // The interval in milliseconds at which pings will be sent.
+    //   // Defaults to 10000, set to <= 0 to disable.
+    //   pingInterval: 10000,
+    //
+    //   // The interval in milliseconds at which analytics events
+    //   // with the measured RTT will be sent. Defaults to 60000, set
+    //   // to <= 0 to disable.
+    //   analyticsInterval: 60000,
+    //   },
+
+    // If set, will attempt to use the provided video input device label when
+    // triggering a screenshare, instead of proceeding through the normal flow
+    // for obtaining a desktop stream.
+    // NOTE: This option is experimental and is currently intended for internal
+    // use only.
+    // _desktopSharingSourceDevice: 'sample-id-or-label',
+
+    // If true, any checks to handoff to another application will be prevented
+    // and instead the app will continue to display in the current browser.
+    // disableDeepLinking: false,
+
+    // A property to disable the right click context menu for localVideo
+    // the menu has option to flip the locally seen video for local presentations
+    // disableLocalVideoFlip: false,
+
+    // Mainly privacy related settings
+
+    // Disables all invite functions from the app (share, invite, dial out...etc)
+    // disableInviteFunctions: true,
+
+    // Disables storing the room name to the recents list
+    // doNotStoreRoom: true,
+
+    // Deployment specific URLs.
+    // deploymentUrls: {
+    //    // If specified a 'Help' button will be displayed in the overflow menu with a link to the specified URL for
+    //    // user documentation.
+    //    userDocumentationURL: 'https://docs.example.com/video-meetings.html',
+    //    // If specified a 'Download our apps' button will be displayed in the overflow menu with a link
+    //    // to the specified URL for an app download page.
+    //    downloadAppsUrl: 'https://docs.example.com/our-apps.html'
+    // },
+
+    // Options related to the remote participant menu.
+    // remoteVideoMenu: {
+    //     // If set to true the 'Kick out' button will be disabled.
+    //     disableKick: true
+    // },
+
+    // If set to true all muting operations of remote participants will be disabled.
+    // disableRemoteMute: true,
+
+    // List of undocumented settings used in jitsi-meet
+    /**
+     _immediateReloadThreshold
+     autoRecord
+     autoRecordToken
+     debug
+     debugAudioLevels
+     deploymentInfo
+     dialInConfCodeUrl
+     dialInNumbersUrl
+     dialOutAuthUrl
+     dialOutCodesUrl
+     disableRemoteControl
+     displayJids
+     etherpad_base
+     externalConnectUrl
+     firefox_fake_device
+     googleApiApplicationClientID
+     iAmRecorder
+     iAmSipGateway
+     microsoftApiApplicationClientID
+     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
+     */
+
+
+    // Allow all above example options to include a trailing comma and
+    // prevent fear when commenting out the last value.
+    makeJsonParserHappy: 'even if last key had a trailing comma'
+
+    // no configuration value should follow this line.
+};
+
+/* eslint-enable no-unused-vars, no-var */