Prepare Nomad deployment

6 files changed, 1513 insertions, 10 deletions

diff --git a/app/jitsi/config/config.js b/app/jitsi/config/config.js
new file mode 100644
index 0000000..9464f37
--- /dev/null
+++ b/app/jitsi/config/config.js
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+/* 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/global_env.tpl b/app/jitsi/config/global_env.tpl
deleted file mode 100644
index d78975d..0000000
--- a/app/jitsi/config/global_env.tpl
+++ /dev/null
@@ -1,10 +0,0 @@
-JITSI_SECRET_VIDEOBRIDGE={{ key "secrets/jitsi/jitsi_secret_videobridge" }}
-JITSI_SECRET_JICOFO_COMPONENT={{ key "secrets/jitsi/jitsi_secret_jicofo_component" }}
-JITSI_SECRET_JICOFO_USER={{ key "secrets/jitsi/jitsi_secret_jicofo_user" }}
-JITSI_PROSODY_BOSH_PORT={{ env "NOMAD_PORT_bosh_port" }}
-JITSI_PROSODY_BOSH_HOST=127.0.0.1
-JITSI_PROSODY_HOST=127.0.0.1
-JITSI_CERTS_FOLDER=/secrets/certs/
-JITSI_NAT_PUBLIC_IP=78.197.205.190
-JITSI_NAT_LOCAL_IP={{ env "NOMAD_IP_video1_port" }}
-NGINX_PORT={{ env "NOMAD_PORT_https_port" }}
diff --git a/app/jitsi/config/jicofo.conf b/app/jitsi/config/jicofo.conf
new file mode 100644
index 0000000..1d33b9c
--- /dev/null
+++ b/app/jitsi/config/jicofo.conf
@@ -0,0 +1,273 @@
+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/jitsi_secret_jicofo_user" }}
+
+      // 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
new file mode 100644
index 0000000..b2213e9
--- /dev/null
+++ b/app/jitsi/config/nginx.conf
@@ -0,0 +1,42 @@
+# some doc: https://www.nginx.com/resources/wiki/start/topics/examples/full/
+error_log  /dev/stderr;
+
+events {}
+
+http {
+  access_log /dev/stdout;
+  server_names_hash_bucket_size 64;
+
+  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;
+    server_name _;
+    ssl_certificate /etc/nginx/jitsi.crt;
+    ssl_certificate_key /etc/nginx/jitsi.key;
+    root /srv/jitsi-meet;
+    index index.html;
+
+    # lot of work would be needed to improve location rules
+    #  - in order to allow - and _ in the URL, even space
+    #  - while not shadowing other files (.js and following locations)
+    #  - passed some times twice on the problem, not as easy as it seems
+    location ~ ^/([a-zA-Z0-9=\?]+)$ {
+        rewrite ^/(.*)$ / break;
+    }
+    location / {
+        ssi on;
+    }
+
+    location /external_api.js {
+        alias /srv/jitsi-meet/libs/external_api.min.js;
+    }
+
+    location /http-bind {
+        proxy_pass      http://{{ env "NOMAD_ADDR_xmpp_port" }}/http-bind;
+        proxy_set_header X-Forwarded-For \$remote_addr;
+        proxy_set_header Host \$http_host;
+    }
+
+    
+  }
+}
diff --git a/app/jitsi/config/prosody.cfg.lua b/app/jitsi/config/prosody.cfg.lua
new file mode 100644
index 0000000..65dabf6
--- /dev/null
+++ b/app/jitsi/config/prosody.cfg.lua
@@ -0,0 +1,135 @@
+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 = { 5280 }
+c2s_ports = { 5222 }
+
+
+-- 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
new file mode 100644
index 0000000..9307945
--- /dev/null
+++ b/app/jitsi/config/videobridge.conf
@@ -0,0 +1,290 @@
+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/jitsi_secret_jvb_user" }}"
+          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
+  }
+}
+