Move app files into cluster subdirectories; add prod garage

5 files changed, 0 insertions, 1604 deletions

diff --git a/app/jitsi/config/config.js b/app/jitsi/config/config.js
deleted file mode 100644
index 9464f37..0000000
--- a/app/jitsi/config/config.js
+++ /dev/null
@@ -1,773 +0,0 @@
-/* eslint-disable no-unused-vars, no-var */
-
-var config = {
-    // Connection
-    //
-
-    hosts: {
-        // XMPP domain.
-        domain: 'jitsi',
-
-        // When using authentication, domain for guest users.
-        // anonymousdomain: 'guest.example.com',
-
-        // Domain for authenticated users. Defaults to <domain>.
-        // authdomain: 'jitsi-meet.example.com',
-
-        // Focus component domain. Defaults to focus.<domain>.
-        // focus: 'focus.jitsi-meet.example.com',
-
-        // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
-        muc: 'conference.jitsi'
-    },
-
-    // BOSH URL. FIXME: use XEP-0156 to discover it.
-    bosh: '//jitsi.deuxfleurs.fr/http-bind',
-
-    // Websocket URL
-    // websocket: 'wss://jitsi-meet.example.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
-    // Do not change username - FIXME: Make focus username configurable
-    // https://github.com/jitsi/jitsi-meet/issues/7376
-    // focusUserJid: 'focus@auth.jitsi-meet.example.com',
-
-
-    // Testing / experimental features.
-    //
-
-    testing: {
-        // Disables the End to End Encryption feature. Useful for debugging
-        // issues related to insertable streams.
-        // disableE2EE: false,
-
-        // 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
-
-        // Enable / disable 500 Kbps bitrate cap on desktop tracks. When enabled,
-        // simulcast is turned off for the desktop share. If presenter is turned
-        // on while screensharing is in progress, the max bitrate is automatically
-        // adjusted to 2.5 Mbps. This takes a value between 0 and 1 which determines
-        // the probability for this to be enabled.
-        // capScreenshareBitrate: 1 // 0 to disable
-
-        // Enable callstats only for a percentage of users.
-        // This takes a value between 0 and 100 which determines the probability for
-        // the callstats to be enabled.
-        // callStatsThreshold: 5 // enable callstats for 5% of the users.
-    },
-
-    // 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 show a "Save Logs" link in the GSM popover that can be
-    // used to collect debug information (XMPP IQs, SDP offer/answer cycles)
-    // about the call.
-    // enableSaveLogs: false,
-
-    // 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: 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: 5,
-
-    // Start calls with audio muted. Unlike the option above, this one is only
-    // applied locally. FIXME: having these 2 options is confusing.
-    // startWithAudioMuted: false,
-
-    // Enabling it (with #params) will disable local audio output of remote
-    // participants and to enable it back a reload is needed.
-    // startSilent: false
-
-    // Sets the preferred target bitrate for the Opus audio codec by setting its
-    // 'maxaveragebitrate' parameter. Currently not available in p2p mode.
-    // Valid values are in the range 6000 to 510000
-    // opusMaxAverageBitrate: 20000,
-
-    // Enables support for opus-red (redundancy for Opus).
-    // enableOpusRed: false
-
-    // Video
-
-    // Sets the preferred resolution (height) for local video. Defaults to 720.
-    // resolution: 720,
-
-    // How many participants while in the tile view mode, before the receiving video quality is reduced from HD to SD.
-    // Use -1 to disable.
-    // maxFullResolutionParticipants: 2,
-
-    // 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: 5,
-
-    // 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.
-    // This option has been deprecated, use preferredCodec under videoQuality section instead.
-    // preferH264: true,
-
-    // If set to true, disable H.264 video codec by stripping it out of the
-    // SDP.
-    // disableH264: false,
-
-    // Desktop sharing
-
-    // 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://jitsi-meet.example.com/static/oauth.html'
-    //     redirectURI:
-    //          'https://jitsi-meet.example.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,
-
-    // Provides a way to use different "last N" values based on the number of participants in the conference.
-    // The keys in an Object represent number of participants and the values are "last N" to be used when number of
-    // participants gets to or above the number.
-    //
-    // For the given example mapping, "last N" will be set to 20 as long as there are at least 5, but less than
-    // 29 participants in the call and it will be lowered to 15 when the 30th participant joins. The 'channelLastN'
-    // will be used as default until the first threshold is reached.
-    //
-    // lastNLimits: {
-    //     5: 20,
-    //     30: 15,
-    //     50: 10,
-    //     70: 5,
-    //     90: 2
-    // },
-
-    // Specify the settings for video quality optimizations on the client.
-    // videoQuality: {
-    //    // Provides a way to prevent a video codec from being negotiated on the JVB connection. The codec specified
-    //    // here will be removed from the list of codecs present in the SDP answer generated by the client. If the
-    //    // same codec is specified for both the disabled and preferred option, the disable settings will prevail.
-    //    // Note that 'VP8' cannot be disabled since it's a mandatory codec, the setting will be ignored in this case.
-    //    disabledCodec: 'H264',
-    //
-    //    // Provides a way to set a preferred video codec for the JVB connection. If 'H264' is specified here,
-    //    // simulcast will be automatically disabled since JVB doesn't support H264 simulcast yet. This will only
-    //    // rearrange the the preference order of the codecs in the SDP answer generated by the browser only if the
-    //    // preferred codec specified here is present. Please ensure that the JVB offers the specified codec for this
-    //    // to take effect.
-    //    preferredCodec: 'VP8',
-    //
-    //    // Provides a way to configure the maximum bitrates that will be enforced on the simulcast streams for
-    //    // video tracks. The keys in the object represent the type of the stream (LD, SD or HD) and the values
-    //    // are the max.bitrates to be set on that particular type of stream. The actual send may vary based on
-    //    // the available bandwidth calculated by the browser, but it will be capped by the values specified here.
-    //    // This is currently not implemented on app based clients on mobile.
-    //    maxBitratesVideo: {
-    //        low: 200000,
-    //        standard: 500000,
-    //        high: 1500000
-    //    },
-    //
-    //    // The options can be used to override default thresholds of video thumbnail heights corresponding to
-    //    // the video quality levels used in the application. At the time of this writing the allowed levels are:
-    //    //     'low' - for the low quality level (180p at the time of this writing)
-    //    //     'standard' - for the medium quality level (360p)
-    //    //     'high' - for the high quality level (720p)
-    //    // The keys should be positive numbers which represent the minimal thumbnail height for the quality level.
-    //    //
-    //    // With the default config value below the application will use 'low' quality until the thumbnails are
-    //    // at least 360 pixels tall. If the thumbnail height reaches 720 pixels then the application will switch to
-    //    // the high quality.
-    //    minHeightForQualityLvl: {
-    //        360: 'standard',
-    //        720: 'high'
-    //    },
-    //
-    //    // Provides a way to resize the desktop track to 720p (if it is greater than 720p) before creating a canvas
-    //    // for the presenter mode (camera picture-in-picture mode with screenshare).
-    //    resizeDesktopForPresenter: false
-    // },
-
-    // // Options for the recording limit notification.
-    // recordingLimit: {
-    //
-    //    // The recording limit in minutes. Note: This number appears in the notification text
-    //    // but doesn't enforce the actual recording time limit. This should be configured in
-    //    // jibri!
-    //    limit: 60,
-    //
-    //    // The name of the app with unlimited recordings.
-    //    appName: 'Unlimited recordings APP',
-    //
-    //    // The URL of the app with unlimited recordings.
-    //    appURL: 'https://unlimited.recordings.app.com/'
-    // },
-
-    // Disables or enables RTX (RFC 4588) (defaults to false).
-    // disableRtx: false,
-
-    // Disables or enables TCC support in this client (default: enabled).
-    // enableTcc: true,
-
-    // Disables or enables REMB support in this client (default: enabled).
-    // enableRemb: true,
-
-    // Enables ICE restart logic in LJM and displays the page reload overlay on
-    // ICE failure. Current disabled by default because it's causing issues with
-    // signaling when Octo is enabled. Also when we do an "ICE restart"(which is
-    // not a real ICE restart), the client maintains the TCC sequence number
-    // counter, but the bridge resets it. The bridge sends media packets with
-    // TCC sequence numbers starting from 0.
-    // enableIceRestart: false,
-
-    // Use TURN/UDP servers for the jitsi-videobridge connection (by default
-    // we filter out TURN/UDP because it is usually not needed since the
-    // bridge itself is reachable via UDP)
-    // useTurnUdp: false
-
-    // UI
-    //
-
-    // Disables responsive tiles.
-    // disableResponsiveTiles: false,
-
-    // Hides lobby button
-    // hideLobbyButton: 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,
-
-    // Disable app shortcuts that are registered upon joining a conference
-    // disableShortcuts: false,
-
-    // Disable initial browser getUserMedia requests.
-    // This is useful for scenarios where users might want to start a conference for screensharing only
-    // disableInitialGUM: false,
-
-    // 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: 'fr',
-
-    // Disables profile and the edit of all fields from the profile settings (display name and email)
-    // disableProfile: false,
-
-    // Whether or not some features are checked based on token.
-    // enableFeaturesBasedOnToken: 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,
-
-    // When 'true', it shows an intermediate page before joining, where the user can configure their devices.
-    prejoinPageEnabled: true,
-
-    // If etherpad integration is enabled, setting this to true will
-    // automatically open the etherpad when a participant joins.  This
-    // does not affect the mobile app since opening an etherpad
-    // obscures the conference controls -- it's better to let users
-    // choose to open the pad on their own in that case.
-    // openSharedDocumentOnJoin: false,
-
-    // If true, shows the unsafe room name warning label when a room name is
-    // deemed unsafe (due to the simplicity in the name) and a password is not
-    // set or the lobby is not enabled.
-    // enableInsecureRoomNameWarning: false,
-
-    // Whether to automatically copy invitation URL after creating a room.
-    // Document should be focused for this option to work
-    // enableAutomaticUrlCopy: false,
-
-    // Base URL for a Gravatar-compatible service. Defaults to libravatar.
-    // gravatarBaseURL: 'https://seccdn.libravatar.org/avatar/';
-
-    // 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 names to callstats
-    // enableDisplayNameInStats: false,
-
-    // Enables sending participants' emails (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,
-
-        // The STUN servers that will be used in the peer to peer connections
-        stunServers: [
-
-            // { urls: 'stun:jitsi-meet.example.com:3478' },
-            { 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). This setting is deprecated, use preferredCodec instead.
-        // preferH264: true
-
-        // Provides a way to set the video codec preference on the p2p connection. Acceptable
-        // codec values are 'VP8', 'VP9' and 'H264'.
-        // preferredCodec: 'H264',
-
-        // If set to true, disable H.264 video codec by stripping it out of the
-        // SDP. This setting is deprecated, use disabledCodec instead.
-        // disableH264: false,
-
-        // Provides a way to prevent a video codec from being negotiated on the p2p connection.
-        // disabledCodec: '',
-
-        // 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'
-
-        // Matomo configuration:
-        // matomoEndpoint: 'https://your-matomo-endpoint/',
-        // matomoSiteID: '42',
-
-        // The Amplitude APP Key:
-        // amplitudeAPPKey: '<APP_KEY>'
-
-        // Configuration for the rtcstats server:
-        // By enabling rtcstats server every time a conference is joined the rtcstats
-        // module connects to the provided rtcstatsEndpoint and sends statistics regarding
-        // PeerConnection states along with getStats metrics polled at the specified
-        // interval.
-        // rtcstatsEnabled: true,
-
-        // In order to enable rtcstats one needs to provide a endpoint url.
-        // rtcstatsEndpoint: wss://rtcstats-server-pilot.jitsi.net/,
-
-        // The interval at which rtcstats will poll getStats, defaults to 1000ms.
-        // If the value is set to 0 getStats won't be polled and the rtcstats client
-        // will only send data related to RTCPeerConnection events.
-        // rtcstatsPolIInterval: 1000
-
-        // 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"
-        // ],
-    },
-
-    // Logs that should go be passed through the 'log' event if a handler is defined for it
-    // apiLogLevels: ['warn', 'log', 'error', 'info', 'debug'],
-
-    // 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,
-
-    // Enables support for lip-sync for this client (if the browser supports it).
-    // enableLipSync: false
-
-    /**
-     External API url used to receive branding specific information.
-     If there is no url set or there are missing fields, the defaults are applied.
-     None of the fields are mandatory and the response must have the shape:
-     {
-         // The hex value for the colour used as background
-         backgroundColor: '#fff',
-         // The url for the image used as background
-         backgroundImageUrl: 'https://example.com/background-img.png',
-         // The anchor url used when clicking the logo image
-         logoClickUrl: 'https://example-company.org',
-         // The url used for the image used as logo
-         logoImageUrl: 'https://example.com/logo-img.png'
-     }
-    */
-    // dynamicBrandingUrl: '',
-
-    // The URL of the moderated rooms microservice, if available. If it
-    // is present, a link to the service will be rendered on the welcome page,
-    // otherwise the app doesn't render it.
-    // moderatedRoomServiceUrl: 'https://moderated.jitsi-meet.example.com',
-
-    // If true, tile view will not be enabled automatically when the participants count threshold is reached.
-    // disableTileView: true,
-
-    // Hides the conference subject
-    // hideConferenceSubject: true
-
-    // Hides the conference timer.
-    // hideConferenceTimer: true,
-
-    // Hides the participants stats
-    // hideParticipantsStats: true
-
-    // Sets the conference subject
-    // subject: 'Conference Subject',
-
-    // List of undocumented settings used in jitsi-meet
-    /**
-     _immediateReloadThreshold
-     debug
-     debugAudioLevels
-     deploymentInfo
-     dialInConfCodeUrl
-     dialInNumbersUrl
-     dialOutAuthUrl
-     dialOutCodesUrl
-     disableRemoteControl
-     displayJids
-     etherpad_base
-     externalConnectUrl
-     firefox_fake_device
-     googleApiApplicationClientID
-     iAmRecorder
-     iAmSipGateway
-     microsoftApiApplicationClientID
-     peopleSearchQueryTypes
-     peopleSearchUrl
-     requireDisplayName
-     tokenAuthUrl
-     */
-
-    /**
-     * This property can be used to alter the generated meeting invite links (in combination with a branding domain
-     * which is retrieved internally by jitsi meet) (e.g. https://meet.jit.si/someMeeting
-     * can become https://brandedDomain/roomAlias)
-     */
-    // brandingRoomAlias: null,
-
-    // List of undocumented settings used in lib-jitsi-meet
-    /**
-     _peerConnStatusOutOfLastNTimeout
-     _peerConnStatusRtcMuteTimeout
-     abTesting
-     avgRtpStatsN
-     callStatsConfIDNamespace
-     callStatsCustomScriptUrl
-     desktopSharingSources
-     disableAEC
-     disableAGC
-     disableAP
-     disableHPF
-     disableNS
-     enableTalkWhileMuted
-     forceJVB121Ratio
-     forceTurnRelay
-     hiddenDomain
-     ignoreStartMuted
-     websocketKeepAlive
-     websocketKeepAliveUrl
-     */
-
-    /**
-        Use this array to configure which notifications will be shown to the user
-        The items correspond to the title or description key of that notification
-        Some of these notifications also depend on some other internal logic to be displayed or not,
-        so adding them here will not ensure they will always be displayed
-
-        A falsy value for this prop will result in having all notifications enabled (e.g null, undefined, false)
-    */
-    // notifications: [
-    //     'connection.CONNFAIL', // shown when the connection fails,
-    //     'dialog.cameraNotSendingData', // shown when there's no feed from user's camera
-    //     'dialog.kickTitle', // shown when user has been kicked
-    //     'dialog.liveStreaming', // livestreaming notifications (pending, on, off, limits)
-    //     'dialog.lockTitle', // shown when setting conference password fails
-    //     'dialog.maxUsersLimitReached', // shown when maximmum users limit has been reached
-    //     'dialog.micNotSendingData', // shown when user's mic is not sending any audio
-    //     'dialog.passwordNotSupportedTitle', // shown when setting conference password fails due to password format
-    //     'dialog.recording', // recording notifications (pending, on, off, limits)
-    //     'dialog.remoteControlTitle', // remote control notifications (allowed, denied, start, stop, error)
-    //     'dialog.reservationError',
-    //     'dialog.serviceUnavailable', // shown when server is not reachable
-    //     'dialog.sessTerminated', // shown when there is a failed conference session
-    //     'dialog.tokenAuthFailed', // show when an invalid jwt is used
-    //     'dialog.transcribing', // transcribing notifications (pending, off)
-    //     'dialOut.statusMessage', // shown when dial out status is updated.
-    //     'liveStreaming.busy', // shown when livestreaming service is busy
-    //     'liveStreaming.failedToStart', // shown when livestreaming fails to start
-    //     'liveStreaming.unavailableTitle', // shown when livestreaming service is not reachable
-    //     'lobby.joinRejectedMessage', // shown when while in a lobby, user's request to join is rejected
-    //     'lobby.notificationTitle', // shown when lobby is toggled and when join requests are allowed / denied
-    //     'localRecording.localRecording', // shown when a local recording is started
-    //     'notify.disconnected', // shown when a participant has left
-    //     'notify.grantedTo', // shown when moderator rights were granted to a participant
-    //     'notify.invitedOneMember', // shown when 1 participant has been invited
-    //     'notify.invitedThreePlusMembers', // shown when 3+ participants have been invited
-    //     'notify.invitedTwoMembers', // shown when 2 participants have been invited
-    //     'notify.kickParticipant', // shown when a participant is kicked
-    //     'notify.mutedRemotelyTitle', // shown when user is muted by a remote party
-    //     'notify.mutedTitle', // shown when user has been muted upon joining,
-    //     'notify.newDeviceAudioTitle', // prompts the user to use a newly detected audio device
-    //     'notify.newDeviceCameraTitle', // prompts the user to use a newly detected camera
-    //     'notify.passwordRemovedRemotely', // shown when a password has been removed remotely
-    //     'notify.passwordSetRemotely', // shown when a password has been set remotely
-    //     'notify.raisedHand', // shown when a partcipant used raise hand,
-    //     'notify.startSilentTitle', // shown when user joined with no audio
-    //     'prejoin.errorDialOut',
-    //     'prejoin.errorDialOutDisconnected',
-    //     'prejoin.errorDialOutFailed',
-    //     'prejoin.errorDialOutStatus',
-    //     'prejoin.errorStatusCode',
-    //     'prejoin.errorValidation',
-    //     'recording.busy', // shown when recording service is busy
-    //     'recording.failedToStart', // shown when recording fails to start
-    //     'recording.unavailableTitle', // shown when recording service is not reachable
-    //     'toolbar.noAudioSignalTitle', // shown when a broken mic is detected
-    //     'toolbar.noisyAudioInputTitle', // shown when noise is detected for the current microphone
-    //     'toolbar.talkWhileMutedPopup', // shown when user tries to speak while muted
-    //     'transcribing.failedToStart' // shown when transcribing fails to start
-    // ]
-
-    // 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 */
diff --git a/app/jitsi/config/jicofo.conf b/app/jitsi/config/jicofo.conf
deleted file mode 100644
index 5586348..0000000
--- a/app/jitsi/config/jicofo.conf
+++ /dev/null
@@ -1,273 +0,0 @@
-jicofo {
-  // Authentication with external services
-  authentication {
-    enabled = false
-    // The type of authentication. Supported values are XMPP, JWT or SHIBBOLETH (default).
-    type = SHIBBOLETH
-
-    // The pattern of authentication URL. See ShibbolethAuthAuthority for more information.
-    # login-url =
-
-    # logout-url =
-
-    authentication-lifetime = 24 hours
-    enable-auto-login = true
-  }
-  // Configuration related to jitsi-videobridge
-  bridge {
-    // The maximum number of participants in a single conference to put on one bridge (use -1 for no maximum).
-    max-bridge-participants = -1
-    // The assumed maximum packet rate that a bridge can handle.
-    max-bridge-packet-rate = 50000
-    // The assumed average packet rate per participant.
-    average-participant-packet-rate-pps = 500
-    // The assumed average stress per participant.
-    average-participant-stress = 0.01
-    // The assumed time that an endpoint takes to start contributing fully to the load on a bridge. To avoid allocating
-    // a burst of endpoints to the same bridge, the bridge stress is adjusted by adding the number of new endpoints
-    // in the last [participant-rampup-time] multiplied by [average-participant-stress].
-    participant-rampup-interval = 20 seconds
-    // The stress level above which a bridge is considered overstressed.
-    stress-threshold = 0.8
-    // The amount of to wait before retrying using a failed bridge.
-    failure-reset-threshold = 1 minute
-    // The bridge selection strategy. The built-in strategies are:
-    // SingleBridgeSelectionStrategy: Use the least loaded bridge, do not split a conference between bridges (Octo).
-    // SplitBridgeSelectionStrategy: Use a separate bridge for each participant (for testing).
-    // RegionBasedBridgeSelectionStrategy: Attempt to put each participant in a bridge in their local region (i.e. use
-    //    Octo for geo-location).
-    // IntraRegionBridgeSelectionStrategy: Use additional bridges when a bridge becomes overloaded (i.e. use Octo for
-    //    load balancing).
-    //
-    // Additionally, you can use the fully qualified class name for custom BridgeSelectionStrategy implementations.
-    selection-strategy = SingleBridgeSelectionStrategy
-    health-checks {
-      // Whether jicofo should perform periodic health checks to the connected bridges.
-      enabled = true
-      // The interval at which to perform health checks.
-      interval = 10 seconds
-      // When a health checks times out, jicofo will retry and only consider it fail after the retry fails. This
-      // configures the delay between the original health check timing out and the second health check being sent.
-      // It is a duration and defaults to half the [interval].
-      # retry-delay = 5 seconds
-    }
-
-    // The JID of the MUC to be used as a brewery for bridge instances.
-    brewery-jid = "jvbbrewery@internal.auth.jitsi"
-  }
-  // Configure the codecs and RTP extensions to be used in the offer sent to clients.
-  codec {
-    video {
-      vp8 {
-        enabled = true
-        pt = 100
-        // Payload type for the associated RTX stream. Set to -1 to disable RTX.
-        rtx-pt = 96
-      }
-      vp9 {
-        enabled = true
-        pt = 101
-        // Payload type for the associated RTX stream. Set to -1 to disable RTX.
-        rtx-pt = 97
-      }
-      h264 {
-        enabled = true
-        pt = 107
-        // Payload type for the associated RTX stream. Set to -1 to disable RTX.
-        rtx-pt = 99
-      }
-    }
-
-    audio {
-      isac-16000 {
-        enabled = true
-        pt = 103
-      }
-      isac-32000 {
-        enabled = true
-        pt = 104
-      }
-      opus {
-        enabled = true
-        pt = 111
-        minptime = 10
-        use-inband-fec = true
-        red {
-          enabled = false
-          pt = 112
-        }
-      }
-      telephone-event {
-        enabled = true
-        pt = 126
-      }
-    }
-
-    // RTP header extensions
-    rtp-extensions {
-      audio-level {
-        enabled = true
-        id = 1
-      }
-      tof {
-        // TOF is currently disabled, because we don't support it in the bridge
-        // (and currently clients seem to not use it when abs-send-time is
-        // available).
-        enabled = false
-        id = 2
-      }
-      abs-send-time {
-        enabled = true
-        id = 3
-      }
-      rid {
-        enabled = false
-        id = 4
-      }
-      tcc {
-        enabled = true
-        id = 5
-      }
-      video-content-type {
-        enabled = false
-        id = 7
-      }
-      framemarking {
-        enabled = false
-        id = 9
-      }
-    }
-  }
-
-  conference {
-    // Whether to automatically grant the 'owner' role to the first participant in the conference (and subsequently to
-    // the next in line when the current owner leaves).
-    enable-auto-owner = true
-
-    // How long to wait for the initial participant in a conference.
-    initial-timeout = 15 seconds
-
-    // Whether jicofo should inject a random SSRC for endpoints which don't advertise any SSRCs. This is a temporary
-    // workaround for an issue with signaling endpoints for Octo.
-    inject-ssrc-for-recv-only-endpoints = false
-
-    max-ssrcs-per-user = 20
-
-    // How long a participant's media session will be kept alive once it remains the only participant in the room.
-    single-participant-timeout = 20 seconds
-
-    // The minimum number of participants required for the conference to be started.
-    min-participants = 2
-
-    // Experimental.
-    enable-lip-sync = false
-
-    shared-document {
-      // If `true` the shared document uses a random name. Otherwise, it uses the conference name.
-      use-random-name = false
-    }
-  }
-
-  // Configuration for the internal health checks performed by jicofo.
-  health {
-    // Whether to perform health checks.
-    enabled = false
-
-    // The interval between health checks. If set to 0, periodic health checks will not be performed.
-    interval = 10 seconds
-
-    # The timeout for a health check
-    timeout = 30 seconds
-
-    # If performing a health check takes longer than this, it is considered unsuccessful.
-    max-check-duration = 20 seconds
-
-    # The prefix to use when creating MUC rooms for the purpose of health checks.
-    room-name-prefix = "__jicofo-health-check"
-  }
-
-  jibri {
-    // The JID of the MUC to be used as a brewery for jibri instances for streaming.
-    # brewery-jid = "jibribrewery@example.com"
-
-    // How many times to retry a given Jibri request before giving up. Set to -1 to allow infinite retries.
-    num-retries = 5
-
-    // How long to wait for Jibri to start recording from the time it accepts a START request.
-    pending-timeout = 90 seconds
-  }
-
-  jibri-sip {
-    // The JID of the MUC to be used as a brewery for jibri instances for SIP.
-    # brewery-jid = "jibrisipbrewery@example.com"
-  }
-
-  jigasi {
-    // The JID of the MUC to be used as a brewery for jigasi instances.
-    # brewery-jid = "jigasibrewery@example.com"
-  }
-
-  // The region in which the machine is running.
-  #local-region="us-east-1"
-
-  octo {
-    // Whether or not to use Octo. Note that when enabled, its use will be determined by
-    // $jicofo.bridge.selection-strategy.
-    enabled = false
-
-    // An identifier of the Jicofo instance, used for the purpose of generating conference IDs unique across a set of
-    // Jicofo instances. Valid values are [1, 65535]. The value 0 is used when none is explicitly configured.
-    id = 1
-  }
-
-  rest {
-    port = 8888
-    tls-port = 8843
-  }
-
-  sctp {
-    // Whether to allocate SCTP channels on the bridge (only when the client advertises support, and SCTP is
-    // enabled in the per-conference configuration).
-    enabled = true
-  }
-
-  task-pools {
-    shared-pool-max-threads = 1500
-  }
-
-  xmpp {
-    // The separate XMPP connection used for communication with clients (endpoints).
-    client {
-      enabled = true
-      hostname = "{{ env "NOMAD_IP_xmpp_port" }}"
-      port = {{ env "NOMAD_PORT_xmpp_port" }}
-      domain = "auth.jitsi"
-      username = "focus"
-      password = "{{ key "secrets/jitsi/jicofo_pass" | trimSpace }}"
-
-      // How long to wait for a response to a stanza before giving up.
-      reply-timeout = 15 seconds
-
-      // The JID/domain of the MUC service used for conferencing.
-      conference-muc-jid = conference.jitsi
-
-      // A flag to suppress the TLS certificate verification.
-      disable-certificate-verification = false
-    }
-    // The separate XMPP connection used for internal services (currently only jitsi-videobridge).
-    service {
-      enabled = false
-      hostname = "jitsi-xmpp"
-      port = 5222
-      domain = "auth.jitsi"
-      username = "focus"
-      password = "jicofopass"
-
-      // How long to wait for a response to a stanza before giving up.
-      reply-timeout = 15 seconds
-
-      // A flag to suppress the TLS certificate verification.
-      disable-certificate-verification = false
-    }
-  }
-}
diff --git a/app/jitsi/config/nginx.conf b/app/jitsi/config/nginx.conf
deleted file mode 100644
index 32cc3c1..0000000
--- a/app/jitsi/config/nginx.conf
+++ /dev/null
@@ -1,133 +0,0 @@
-# some doc: https://www.nginx.com/resources/wiki/start/topics/examples/full/
-error_log  /dev/stderr info;
-
-events {}
-
-http {
-  ##
-  # Basic Settings
-  ##
-  sendfile on;
-  tcp_nopush on;
-  tcp_nodelay on;
-  keepalive_timeout 65;
-  types_hash_max_size 2048;
-
-
-  # mimetypes, required by jitsi!
-  include /etc/nginx/mime.types;
-  default_type application/octet-stream;
-
-  types {
-    application/wasm wasm;
-  }
-
-  ##
-  # SSL Settings
-  ##
-
-  ssl_protocols TLSv1 TLSv1.1 TLSv1.2; # Dropping SSLv3, ref: POODLE
-  ssl_prefer_server_ciphers on;
-       
-  ##
-  # Gzip Settings
-  ##
-  gzip on;
-
-  access_log /dev/stdout;
-  server_names_hash_bucket_size 64;
-
-  # inspired by https://raw.githubusercontent.com/jitsi/docker-jitsi-meet/master/web/rootfs/defaults/meet.conf
-  server {
-    #listen 0.0.0.0:{{ env "NOMAD_PORT_https_port" }} ssl http2 default_server;
-    #listen [::]:{{ env "NOMAD_PORT_https_port" }} ssl http2 default_server;
-    listen 0.0.0.0:{{ env "NOMAD_PORT_https_port" }} default_server;
-    listen [::]:{{ env "NOMAD_PORT_https_port" }} default_server;
-    client_max_body_size 0;
-    server_name _;
-
-    # ssi on with javascript for multidomain variables in config.js
-    ssi on;
-    ssi_types application/x-javascript application/javascript;
-
-    #ssl_certificate /etc/nginx/jitsi.crt;
-    #ssl_certificate_key /etc/nginx/jitsi.key;
-    root /srv/jitsi-meet;
-    index index.html;
-    error_page 404 /static/404.html;
-
-    location = /config.js {
-      alias /srv/jitsi-meet/config.js;
-    }
-
-    location = /interface_config.js {
-      alias /srv/jitsi-meet/interface_config.js;
-    }
-
-    location = /external_api.js {
-      alias /srv/jitsi-meet/libs/external_api.min.js;
-    }
-
-    # ensure all static content can always be found first
-    location ~ ^/(libs|css|static|images|fonts|lang|sounds|connection_optimization|.well-known)/(.*)$
-    {
-      add_header 'Access-Control-Allow-Origin' '*';
-      alias /srv/jitsi-meet/$1/$2;
-    }
-
-    # not used yet VVV
-    # colibri (JVB) websockets
-    #location ~ ^/colibri-ws/([a-zA-Z0-9-\.]+)/(.*) {
-    #  proxy_pass http://$1:9090/colibri-ws/$1/$2$is_args$args;
-    #  proxy_http_version 1.1;
-    #  proxy_set_header Upgrade $http_upgrade;
-    #  proxy_set_header Connection "upgrade";
-    #  tcp_nodelay on;
-    #}
-
-    location = /http-bind {
-        # We add CORS to use a different frontend which is useful for load testing as we do not want to advertise too much our URL
-        add_header 'Access-Control-Allow-Headers' 'content-type';
-        add_header 'Access-Control-Allow-Methods' 'GET,POST,PUT,DELETE,OPTIONS';
-        add_header 'Access-Control-Allow-Origin' '*';
-        proxy_pass      http://{{ env "NOMAD_ADDR_bosh_port" }}/http-bind;
-        proxy_set_header X-Forwarded-For \$remote_addr;
-        #proxy_set_header Host \$http_host;
-    }
-
-    # not used yet VVV
-    # xmpp websockets
-    #location = /xmpp-websocket {
-    #    proxy_pass {{ .Env.XMPP_BOSH_URL_BASE }}/xmpp-websocket;
-    #    proxy_http_version 1.1;
-    #    proxy_set_header Connection "upgrade";
-    #    proxy_set_header Upgrade $http_upgrade;
-    #    proxy_set_header Host {{ .Env.XMPP_DOMAIN }};
-    #    proxy_set_header X-Forwarded-For $remote_addr;
-    #    tcp_nodelay on;
-    #}
-
-    location ~ ^/([^/?&:'"]+)$ {
-      try_files $uri @root_path;
-    }
-
-    location @root_path {
-      rewrite ^/(.*)$ / break;
-    }
-
-  # Not used yet VVVV
-  # Etherpad-lite
-  # location /etherpad/ {
-  #    proxy_http_version 1.1;
-  #    proxy_set_header Upgrade $http_upgrade;
-  #    proxy_set_header Connection 'upgrade';
-  #    proxy_set_header Host $host;
-  #    proxy_cache_bypass $http_upgrade;
-  #    proxy_pass {{ .Env.ETHERPAD_URL_BASE }}/;
-  #    proxy_set_header X-Forwarded-For $remote_addr;
-  #    proxy_buffering off;
-  #    proxy_set_header Host {{ .Env.XMPP_DOMAIN }};
-  #  }
-
-  }
-}
diff --git a/app/jitsi/config/prosody.cfg.lua b/app/jitsi/config/prosody.cfg.lua
deleted file mode 100644
index 7141f8b..0000000
--- a/app/jitsi/config/prosody.cfg.lua
+++ /dev/null
@@ -1,135 +0,0 @@
-modules_enabled = {
-  "roster"; -- Allow users to have a roster. Recommended ;)
-  "saslauth"; -- Authentication for clients and servers. Recommended if you want to log in.
-  "tls"; -- Add support for secure TLS on c2s/s2s connections
-  "dialback"; -- s2s dialback support
-  "disco"; -- Service discovery
-  "posix"; -- POSIX functionality, sends server to background, enables syslog, etc.
-  "version"; -- Replies to server version requests
-  "uptime"; -- Report how long server has been running
-  "time"; -- Let others know the time here on this server
-  "ping"; -- Replies to XMPP pings with pongs
-  "pep"; -- Enables users to publish their mood, activity, playing music and more
-  -- jitsi
-  --"smacks"; -- not shipped with prosody
-  "carbons";
-  "mam";
-  "lastactivity";
-  "offline";
-  "pubsub";
-  "adhoc";
-  "websocket";
-  --"http_altconnect"; -- not shipped with prosody
-}
-modules_disabled = { "s2s" }
-
-plugin_paths = { "/usr/share/jitsi-meet/prosody-plugins/" }
-
-log = {
-  --log less on console with warn="*console"; or err="*console" or more with debug="*console"
-  info="*console";
-}
-daemonize = false
-use_libevent = true
-
--- domain mapper options, must at least have domain base set to use the mapper
-muc_mapper_domain_base = "jitsi.deuxfleurs.fr";
-
---@FIXME would be great to configure it
---turncredentials_secret = "__turnSecret__";
-
---turncredentials = {
---    { type = "stun", host = "jitmeet.example.com", port = "3478" },
---    { type = "turn", host = "jitmeet.example.com", port = "3478", transport = "udp" },
---    { type = "turns", host = "jitmeet.example.com", port = "5349", transport = "tcp" }
---};
-
-cross_domain_bosh = false;
-consider_bosh_secure = true;
-component_ports = { } -- it seems we don't need external components for now...
-https_ports = { } -- we don't need https
-http_ports = { {{env "NOMAD_PORT_bosh_port" }} }
-c2s_ports = { {{env "NOMAD_PORT_xmpp_port" }} }
-
-
--- https://ssl-config.mozilla.org/#server=haproxy&version=2.1&config=intermediate&openssl=1.1.0g&guideline=5.4
-ssl = {
-    protocol = "tlsv1_2+";
-    ciphers = "ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-ECDSA-CHACHA20-POLY1305:ECDHE-RSA-CHACHA20-POLY1305:DHE-RSA-AES128-GCM-SHA256:DHE-RSA-AES256-GCM-SHA384"
-}
-
-VirtualHost "jitsi"
-    enabled = true -- Remove this line to enable this host
-    authentication = "anonymous"
-    -- Properties below are modified by jitsi-meet-tokens package config
-    -- and authentication above is switched to "token"
-    --app_id="example_app_id"
-    --app_secret="example_app_secret"
-    -- Assign this host a certificate for TLS, otherwise it would use the one
-    -- set in the global section (if any).
-    -- Note that old-style SSL on port 5223 only supports one certificate, and will always
-    -- use the global one.
-    ssl = {
-        key = "/var/lib/prosody/jitsi.key";
-        certificate = "/var/lib/prosody/jitsi.crt";
-    }
-    speakerstats_component = "speakerstats.jitsi"
-    conference_duration_component = "conferenceduration.jitsi"
-    -- we need bosh
-    modules_enabled = {
-        "bosh";
-        "pubsub";
-        "ping"; -- Enable mod_ping
-        "speakerstats";
-        --"turncredentials"; not supported yet
-        "conference_duration";
-        "muc_lobby_rooms";
-    }
-    c2s_require_encryption = false
-    lobby_muc = "lobby.jitsi"
-    main_muc = "conference.jitsi"
-    -- muc_lobby_whitelist = { "recorder.jitmeet.example.com" } -- Here we can whitelist jibri to enter lobby enabled rooms
-
-Component "conference.jitsi" "muc"
-    storage = "memory"
-    modules_enabled = {
-        "muc_meeting_id";
-        "muc_domain_mapper";
-        --"token_verification";
-    }
-    admins = { "focus@auth.jitsi" }
-    muc_room_locking = false
-    muc_room_default_public_jids = true
-
--- internal muc component
-Component "internal.auth.jitsi" "muc"
-    storage = "memory"
-    modules_enabled = {
-        "ping";
-    }
-    admins = { "focus@auth.jitsi", "jvb@auth.jitsi" }
-    muc_room_locking = false
-    muc_room_default_public_jids = true
-
-VirtualHost "auth.jitsi"
-    ssl = {
-        key = "/var/lib/prosody/auth.jitsi.key";
-        certificate = "/var/lib/prosody/auth.jitsi.crt";
-    }
-    authentication = "internal_plain"
-
-Component "focus.jitsi" "client_proxy"
-    target_address = "focus@auth.jitsi"
-
-Component "speakerstats.jitsi" "speakerstats_component"
-    muc_component = "conference.jitsi"
-
-Component "conferenceduration.jitsi" "conference_duration_component"
-    muc_component = "conference.jitsi"
-
-Component "lobby.jitsi" "muc"
-    storage = "memory"
-    restrict_room_creation = true
-    muc_room_locking = false
-    muc_room_default_public_jids = true
-
diff --git a/app/jitsi/config/videobridge.conf b/app/jitsi/config/videobridge.conf
deleted file mode 100644
index a7c166a..0000000
--- a/app/jitsi/config/videobridge.conf
+++ /dev/null
@@ -1,290 +0,0 @@
-videobridge {
-  entity-expiration {
-    # If an entity has no activity after this timeout, it is expired
-    timeout=1 minute
-
-    # The interval at which the videobridge will check for expired entities
-    check-interval=${videobridge.entity-expiration.timeout}
-  }
-  health {
-    # The interval between health checks
-    interval=10 seconds
-
-    # The timeout for a health check
-    timeout=30 seconds
-
-    # If performing a health check takes longer than this, it is considered unsuccessful.
-    max-check-duration=3 seconds
-
-    # Whether or not health check failures should be 'sticky'
-    # (i.e. once the bridge becomes unhealthy, it will never
-    # go back to a healthy state)
-    sticky-failures=false
-  }
-  ep-connection-status {
-    # How long we'll wait for an endpoint to *start* sending
-    # data before we consider it 'inactive'
-    first-transfer-timeout=15 seconds
-
-    # How long an endpoint can be 'inactive' before it will
-    # be considered disconnected
-    max-inactivity-limit=3 seconds
-
-    # How often we check endpoint's connectivity status
-    check-interval=500 milliseconds
-  }
-  cc {
-    bwe-change-threshold=0.15
-    thumbnail-max-height-px=180
-    onstage-ideal-height-px=1080
-    onstage-preferred-height-px=360
-    onstage-preferred-framerate=30
-    enable-onstage-video-suspend=false
-    trust-bwe=true
-
-    # How often we check to send probing data
-    padding-period=15ms
-
-    # How often we'll force recalculations of forwarded
-    # streams
-    max-time-between-calculations = 15 seconds
-
-    # A JVB-wide last-n value, observed by all endpoints.  Endpoints
-    # will take the minimum of their setting and this one (-1 implies
-    # no last-n limit)
-    jvb-last-n = -1
-  }
-  # The APIs by which the JVB can be controlled
-  apis {
-    xmpp-client {
-      # The interval at which presence is published in the configured MUCs.
-      presence-interval = ${videobridge.stats.interval}
-
-      configs {
-        unique-xmpp-server {
-          hostname="{{ env "NOMAD_IP_xmpp_port" }}"
-          port = {{ env "NOMAD_PORT_xmpp_port" }}
-          domain = "auth.jitsi"
-          username = "jvb"
-          password = "{{ key "secrets/jitsi/jvb_pass" | trimSpace }}"
-          muc_jids = "jvbbrewery@internal.auth.jitsi"
-          # The muc_nickname must be unique across all jitsi-videobridge instances
-          muc_nickname = "unique-jvb-server"
-          disable_certificate_verification = false
-        }
-        # example-connection-id {
-        #   For the properties which should be
-        #   filled out here, see MucClientConfiguration
-        # }
-      }
-    }
-    # The COLIBRI REST API
-    rest {
-      enabled = true
-    }
-    jvb-api {
-      enabled = true
-    }
-  }
-  # Configuration of the different REST APIs.
-  # Note that the COLIBRI REST API is configured under videobridge.apis.rest instead.
-  rest {
-    debug {
-      enabled = true
-    }
-    health {
-      enabled = true
-    }
-    shutdown {
-      # Note that the shutdown API requires the COLIBRI API to also be enabled.
-      enabled = false
-    }
-    version {
-      enabled = true
-    }
-  }
-  http-servers {
-    # The HTTP server which hosts services intended for 'public' use
-    # (e.g. websockets for the bridge channel connection)
-    public {
-      # See JettyBundleActivatorConfig in Jicoco for values
-      port = -1
-      tls-port = -1
-    }
-    # The HTTP server which hosts services intended for 'private' use
-    # (e.g. health or debug stats)
-    private {
-      # See JettyBundleActivatorConfig in Jicoco for values
-      host = 127.0.0.1
-    }
-  }
-  octo {
-    # Whether or not Octo is enabled
-    enabled=false
-
-    # A string denoting the 'region' of this JVB.  This region
-    # will be used by Jicofo in the selection of a bridge for
-    # a client by comparing it to the client's region.
-    # Must be set when 'enabled' is true
-    #region="us-west-1"
-
-    # The address on which the Octo relay should bind
-    # Must be set when 'enabled' is true
-    #bind-address=198.51.100.1
-
-    # The port to which the Octo relay should bind
-    bind-port=4096
-
-    # The address which controls the public address which
-    # will be part of the Octo relayId
-    #public-address=198.51.100.1
-
-    # The size of the incoming octo queue. This queue is per-remote-endpoint,
-    # so it matches what we use for local endpoints
-    recv-queue-size=1024
-
-    # The size of the outgoing octo queue.  This is a per-originating-endpoint
-    # queue, so assuming all packets are routed (as they currently are for Octo)
-    # it should be the same size as the transceiver recv queue in
-    # jitsi-media-transform.  Repeating the description from there:
-    #   Assuming 300pps for high-definition, 200pps for standard-definition,
-    #   100pps for low-definition and 50pps for audio, this queue is fed
-    #   650pps, so its size in terms of millis is 1024/650*1000 ~= 1575ms.
-    send-queue-size=1024
-  }
-  load-management {
-    # Whether or not the reducer will be enabled to take actions to mitigate load
-    reducer-enabled = false
-    load-measurements {
-      packet-rate {
-        # The packet rate at which we'll consider the bridge overloaded
-        load-threshold = 50000
-        # The packet rate at which we'll consider the bridge 'underloaded' enough
-        # to start recovery
-        recovery-threshold = 40000
-      }
-    }
-    load-reducers {
-      last-n {
-        # The factor by which we'll reduce the current last-n when trying to reduce load
-        reduction-scale = .75
-        # The factor by which we'll increase the current last-n when trying to recover
-        recover-scale = 1.25
-        # The minimum time in between runs of the last-n reducer to reduce or recover from
-        # load
-        impact-time = 1 minute
-        # The lowest value we'll set for last-n
-        minimum-last-n-value = 0
-        # The highest last-n value we'll enforce.  Once the enforced last-n exceeds this value
-        # we'll remove the limit entirely
-        maximum-enforced-last-n-value = 40
-      }
-    }
-  }
-  sctp {
-    # Whether SCTP data channels are enabled.
-    enabled=true
-  }
-  stats {
-    # Whether periodic collection of statistics is enabled or not. When enabled they are accessible through the REST
-    # API (at `/colibri/stats`), and are available to other modules (e.g. to be pushed to callstats or in a MUC).
-    enabled = true
-
-    # The interval at which stats are gathered.
-    interval = 5 seconds
-
-    # Configuration related to pushing statistics to callstats.io.
-    callstats {
-      # An integer application ID (use 0 to disable pushing stats to callstats).
-      app-id = 0
-
-      # The shared secred to authentication with callstats.io.
-      //app-secret = "s3cret"
-
-      # ID of the key that was used to generate token.
-      //key-id = "abcd"
-
-      # The path to private key file.
-      //key-path = "/etc/jitsi/videobridge/ecpriv.jwk"
-
-      # The ID of the server instance to be used when reporting to callstats.
-      bridge-id = "jitsi"
-
-      # TODO: document
-      //conference-id-prefix = "abcd"
-
-      # The interval at which statististics will be published to callstats. This affects both per-conference and global
-      # statistics.
-      # Note that this value will be overriden if a "callstatsio" transport is defined in the parent "stats" section.
-      interval = ${videobridge.stats.interval}
-    }
-  }
-  websockets {
-    enabled=false
-    server-id="default-id"
-
-    # Optional, even when 'enabled' is set to true
-    # tls=true
-    # Must be set when enabled = true
-    #domain="some-domain"
-  }
-  ice {
-    tcp {
-      # Whether ICE/TCP is enabled.
-      enabled = true
-
-      # The port to bind to for ICE/TCP.
-      port = {{ env "NOMAD_PORT_video_port" }}
-
-      # An optional additional port to advertise.
-      # mapped-port = 8443
-      # Whether to use "ssltcp" or plain "tcp".
-      ssltcp = true
-    }
-
-    udp {
-        # The port for ICE/UDP.
-        port = {{ env "NOMAD_PORT_video_port" }}
-    }
-
-    # An optional prefix to include in STUN username fragments generated by the bridge.
-    #ufrag-prefix = "jvb-123:"
-
-    # Which candidate pairs to keep alive. The accepted values are defined in ice4j's KeepAliveStrategy:
-    # "selected_and_tcp", "selected_only", or "all_succeeded".
-    keep-alive-strategy = "selected_and_tcp"
-
-    # Whether to use the "component socket" feature of ice4j.
-    use-component-socket = true
-
-    # Whether to attempt DNS resolution for remote candidates that contain a non-literal address. When set to 'false'
-    # such candidates will be ignored.
-    resolve-remote-candidates = false
-
-    # The nomination strategy to use for ICE.  THe accepted values are defined in ice4j's NominationStrategy:
-    # "NominateFirstValid", "NominateHighestPriority", "NominateFirstHostOrReflexiveValid", or "NominateBestRTT".
-    nomination-strategy = "NominateFirstValid"
-  }
-
-  transport {
-    send {
-      # The size of the dtls-transport outgoing queue. This is a per-participant
-      # queue. Packets from the egress end-up in this queue right before
-      # transmission by the outgoing srtp pipeline (which mainly consists of the
-      # packet sender).
-      #
-      # Its size needs to be of the same order of magnitude as the rtp sender
-      # queue. In a 100 participant call, assuming 300pps for the on-stage and
-      # 100pps for low-definition, last-n 20 and 2 participants talking, so
-      # 2*50pps for audio, this queue is fed 300+19*100+2*50 = 2300pps, so its
-      # size in terms of millis is 1024/2300*1000 ~= 445ms.
-      queue-size=1024
-    }
-  }
-
-  version {
-    // Wheather to announe the jitsi-videobridge version to clients in the ServerHello message.
-    announce = false
-  }
-}
-