Working on meet frontend

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diff --git a/app/jitsi/integration/meet/config.js b/app/jitsi/integration/meet/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: '//rayonx.machine.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: true,
+
+    // Start the conference in audio only mode (no video is being received nor
+    // sent).
+    // startAudioOnly: false,
+
+    // Every participant after the Nth will start audio muted.
+    // startAudioMuted: 10,
+
+    // Start calls with audio muted. Unlike the option above, this one is only
+    // applied locally. FIXME: having these 2 options is confusing.
+    // startWithAudioMuted: false,
+
+    // 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: 10,
+
+    // Start calls with video muted. Unlike the option above, this one is only
+    // applied locally. FIXME: having these 2 options is confusing.
+    // startWithVideoMuted: false,
+
+    // If set to true, prefer to use the H.264 video codec (if supported).
+    // Note that it's not recommended to do this because simulcast is not
+    // supported when  using H.264. For 1-to-1 calls this setting is enabled by
+    // default and can be toggled in the p2p section.
+    // 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: false,
+
+    // 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/integration/meet/nginx.conf b/app/jitsi/integration/meet/nginx.conf
new file mode 100644
index 0000000..0ee2135
--- /dev/null
+++ b/app/jitsi/integration/meet/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 0.0.0.0:443 ssl http2 default_server;
+    listen [::]:443 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://jitsi-xmpp:5280/http-bind;
+        proxy_set_header X-Forwarded-For \$remote_addr;
+        proxy_set_header Host \$http_host;
+    }
+
+    
+  }
+}