mahmudkuet

Jitsi config

Aug 5th, 2021
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  1. /* eslint-disable no-unused-vars, no-var */
  2.  
  3. var jitsiConnectionOptions = {
  4.     // Connection
  5.     //
  6.     serviceUrl: `https://${process.env.MIX_JITSI_DEPLOYMENT_DOMAIN}/http-bind`,
  7.  
  8.     hosts: {
  9.         // XMPP domain.
  10.         domain: process.env.MIX_JITSI_DEPLOYMENT_DOMAIN,
  11.  
  12.         // When using authentication, domain for guest users.
  13.         // anonymousdomain: 'guest.example.com',
  14.  
  15.         // Domain for authenticated users. Defaults to <domain>.
  16.         // authdomain: 'meet.mazegeek.com',
  17.  
  18.         // Call control component (Jigasi).
  19.         // call_control: 'callcontrol.meet.mazegeek.com',
  20.  
  21.         // Focus component domain. Defaults to focus.<domain>.
  22.         // focus: 'focus.meet.mazegeek.com',
  23.  
  24.         // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
  25.         muc: `conference.${process.env.MIX_JITSI_DEPLOYMENT_DOMAIN}`
  26.     },
  27.  
  28.     // BOSH URL. FIXME: use XEP-0156 to discover it.
  29.     bosh: `https://${process.env.MIX_JITSI_DEPLOYMENT_DOMAIN}/http-bind`,
  30.  
  31.     // Websocket URL
  32.     // websocket: 'wss://meet.mazegeek.com/xmpp-websocket',
  33.  
  34.     // The name of client node advertised in XEP-0115 'c' stanza
  35.     clientNode: 'https://jitsi.org/jitsimeet',
  36.  
  37.     // The real JID of focus participant - can be overridden here
  38.     // Do not change username - FIXME: Make focus username configurable
  39.     // https://github.com/jitsi/jitsi-meet/issues/7376
  40.     // focusUserJid: 'focus@auth.meet.mazegeek.com',
  41.  
  42.  
  43.     // Testing / experimental features.
  44.     //
  45.  
  46.     testing: {
  47.         // Disables the End to End Encryption feature. Useful for debugging
  48.         // issues related to insertable streams.
  49.         // disableE2EE: false,
  50.  
  51.         // P2P test mode disables automatic switching to P2P when there are 2
  52.         // participants in the conference.
  53.         p2pTestMode: false
  54.  
  55.         // Enables the test specific features consumed by jitsi-meet-torture
  56.         // testMode: false
  57.  
  58.         // Disables the auto-play behavior of *all* newly created video element.
  59.         // This is useful when the client runs on a host with limited resources.
  60.         // noAutoPlayVideo: false
  61.  
  62.         // Enable / disable 500 Kbps bitrate cap on desktop tracks. When enabled,
  63.         // simulcast is turned off for the desktop share. If presenter is turned
  64.         // on while screensharing is in progress, the max bitrate is automatically
  65.         // adjusted to 2.5 Mbps. This takes a value between 0 and 1 which determines
  66.         // the probability for this to be enabled.
  67.         // capScreenshareBitrate: 1 // 0 to disable
  68.  
  69.         // Enable callstats only for a percentage of users.
  70.         // This takes a value between 0 and 100. A hash value is calculated using the
  71.         // conference name and callstats is enabled if the hash value is less than the
  72.         // the threshold specified here.
  73.         // callStatsThreshold: 5 // enable callstats for 5% of the conferences.
  74.     },
  75.  
  76.     // Disables ICE/UDP by filtering out local and remote UDP candidates in
  77.     // signalling.
  78.     // webrtcIceUdpDisable: false,
  79.  
  80.     // Disables ICE/TCP by filtering out local and remote TCP candidates in
  81.     // signalling.
  82.     // webrtcIceTcpDisable: false,
  83.  
  84.  
  85.     // Media
  86.     //
  87.  
  88.     // Audio
  89.  
  90.     // Disable measuring of audio levels.
  91.     // disableAudioLevels: false,
  92.     // audioLevelsInterval: 200,
  93.  
  94.     // Enabling this will run the lib-jitsi-meet no audio detection module which
  95.     // will notify the user if the current selected microphone has no audio
  96.     // input and will suggest another valid device if one is present.
  97.     enableNoAudioDetection: true,
  98.  
  99.     // Enabling this will run the lib-jitsi-meet noise detection module which will
  100.     // notify the user if there is noise, other than voice, coming from the current
  101.     // selected microphone. The purpose it to let the user know that the input could
  102.     // be potentially unpleasant for other meeting participants.
  103.     enableNoisyMicDetection: true,
  104.  
  105.     // Start the conference in audio only mode (no video is being received nor
  106.     // sent).
  107.     // startAudioOnly: false,
  108.  
  109.     // Every participant after the Nth will start audio muted.
  110.     // startAudioMuted: 10,
  111.  
  112.     // Start calls with audio muted. Unlike the option above, this one is only
  113.     // applied locally. FIXME: having these 2 options is confusing.
  114.     // startWithAudioMuted: false,
  115.  
  116.     // Enabling it (with #params) will disable local audio output of remote
  117.     // participants and to enable it back a reload is needed.
  118.     // startSilent: false
  119.  
  120.     // Sets the preferred target bitrate for the Opus audio codec by setting its
  121.     // 'maxaveragebitrate' parameter. Currently not available in p2p mode.
  122.     // Valid values are in the range 6000 to 510000
  123.     // opusMaxAverageBitrate: 20000,
  124.  
  125.     // Enables redundancy for Opus
  126.     // enableOpusRed: false
  127.  
  128.     // Video
  129.  
  130.     // Sets the preferred resolution (height) for local video. Defaults to 720.
  131.     // resolution: 720,
  132.  
  133.     // How many participants while in the tile view mode, before the receiving video quality is reduced from HD to SD.
  134.     // Use -1 to disable.
  135.     // maxFullResolutionParticipants: 2,
  136.  
  137.     // w3c spec-compliant video constraints to use for video capture. Currently
  138.     // used by browsers that return true from lib-jitsi-meet's
  139.     // util#browser#usesNewGumFlow. The constraints are independent from
  140.     // this config's resolution value. Defaults to requesting an ideal
  141.     // resolution of 720p.
  142.     // constraints: {
  143.     //     video: {
  144.     //         height: {
  145.     //             ideal: 720,
  146.     //             max: 720,
  147.     //             min: 240
  148.     //         }
  149.     //     }
  150.     // },
  151.  
  152.     // Enable / disable simulcast support.
  153.     // disableSimulcast: false,
  154.  
  155.     // Enable / disable layer suspension.  If enabled, endpoints whose HD
  156.     // layers are not in use will be suspended (no longer sent) until they
  157.     // are requested again.
  158.     // enableLayerSuspension: false,
  159.  
  160.     // Every participant after the Nth will start video muted.
  161.     // startVideoMuted: 10,
  162.  
  163.     // Start calls with video muted. Unlike the option above, this one is only
  164.     // applied locally. FIXME: having these 2 options is confusing.
  165.     // startWithVideoMuted: false,
  166.  
  167.     // If set to true, prefer to use the H.264 video codec (if supported).
  168.     // Note that it's not recommended to do this because simulcast is not
  169.     // supported when  using H.264. For 1-to-1 calls this setting is enabled by
  170.     // default and can be toggled in the p2p section.
  171.     // This option has been deprecated, use preferredCodec under videoQuality section instead.
  172.     // preferH264: true,
  173.  
  174.     // If set to true, disable H.264 video codec by stripping it out of the
  175.     // SDP.
  176.     // disableH264: false,
  177.  
  178.     // Desktop sharing
  179.  
  180.     // Optional desktop sharing frame rate options. Default value: min:5, max:5.
  181.     // desktopSharingFrameRate: {
  182.     //     min: 5,
  183.     //     max: 5
  184.     // },
  185.  
  186.     // Try to start calls with screen-sharing instead of camera video.
  187.     // startScreenSharing: false,
  188.  
  189.     // Recording
  190.  
  191.     // Whether to enable file recording or not.
  192.     // fileRecordingsEnabled: false,
  193.     // Enable the dropbox integration.
  194.     // dropbox: {
  195.     //     appKey: '<APP_KEY>' // Specify your app key here.
  196.     //     // A URL to redirect the user to, after authenticating
  197.     //     // by default uses:
  198.     //     // 'https://meet.mazegeek.com/static/oauth.html'
  199.     //     redirectURI:
  200.     //          'https://meet.mazegeek.com/subfolder/static/oauth.html'
  201.     // },
  202.     // When integrations like dropbox are enabled only that will be shown,
  203.     // by enabling fileRecordingsServiceEnabled, we show both the integrations
  204.     // and the generic recording service (its configuration and storage type
  205.     // depends on jibri configuration)
  206.     // fileRecordingsServiceEnabled: false,
  207.     // Whether to show the possibility to share file recording with other people
  208.     // (e.g. meeting participants), based on the actual implementation
  209.     // on the backend.
  210.     // fileRecordingsServiceSharingEnabled: false,
  211.  
  212.     // Whether to enable live streaming or not.
  213.     // liveStreamingEnabled: false,
  214.  
  215.     // Transcription (in interface_config,
  216.     // subtitles and buttons can be configured)
  217.     // transcribingEnabled: false,
  218.  
  219.     // Enables automatic turning on captions when recording is started
  220.     // autoCaptionOnRecord: false,
  221.  
  222.     // Misc
  223.  
  224.     // Default value for the channel "last N" attribute. -1 for unlimited.
  225.     channelLastN: -1,
  226.  
  227.     // Provides a way to use different "last N" values based on the number of participants in the conference.
  228.     // The keys in an Object represent number of participants and the values are "last N" to be used when number of
  229.     // participants gets to or above the number.
  230.     //
  231.     // For the given example mapping, "last N" will be set to 20 as long as there are at least 5, but less than
  232.     // 29 participants in the call and it will be lowered to 15 when the 30th participant joins. The 'channelLastN'
  233.     // will be used as default until the first threshold is reached.
  234.     //
  235.     // lastNLimits: {
  236.     //     5: 20,
  237.     //     30: 15,
  238.     //     50: 10,
  239.     //     70: 5,
  240.     //     90: 2
  241.     // },
  242.  
  243.     // Specify the settings for video quality optimizations on the client.
  244.     // videoQuality: {
  245.     //    // Provides a way to prevent a video codec from being negotiated on the JVB connection. The codec specified
  246.     //    // here will be removed from the list of codecs present in the SDP answer generated by the client. If the
  247.     //    // same codec is specified for both the disabled and preferred option, the disable settings will prevail.
  248.     //    // Note that 'VP8' cannot be disabled since it's a mandatory codec, the setting will be ignored in this case.
  249.     //    disabledCodec: 'H264',
  250.     //
  251.     //    // Provides a way to set a preferred video codec for the JVB connection. If 'H264' is specified here,
  252.     //    // simulcast will be automatically disabled since JVB doesn't support H264 simulcast yet. This will only
  253.     //    // rearrange the the preference order of the codecs in the SDP answer generated by the browser only if the
  254.     //    // preferred codec specified here is present. Please ensure that the JVB offers the specified codec for this
  255.     //    // to take effect.
  256.     //    preferredCodec: 'VP8',
  257.     //
  258.     //    // Provides a way to configure the maximum bitrates that will be enforced on the simulcast streams for
  259.     //    // video tracks. The keys in the object represent the type of the stream (LD, SD or HD) and the values
  260.     //    // are the max.bitrates to be set on that particular type of stream. The actual send may vary based on
  261.     //    // the available bandwidth calculated by the browser, but it will be capped by the values specified here.
  262.     //    // This is currently not implemented on app based clients on mobile.
  263.     //    maxBitratesVideo: {
  264.     //        low: 200000,
  265.     //        standard: 500000,
  266.     //        high: 1500000
  267.     //    },
  268.     //
  269.     //    // The options can be used to override default thresholds of video thumbnail heights corresponding to
  270.     //    // the video quality levels used in the application. At the time of this writing the allowed levels are:
  271.     //    //     'low' - for the low quality level (180p at the time of this writing)
  272.     //    //     'standard' - for the medium quality level (360p)
  273.     //    //     'high' - for the high quality level (720p)
  274.     //    // The keys should be positive numbers which represent the minimal thumbnail height for the quality level.
  275.     //    //
  276.     //    // With the default config value below the application will use 'low' quality until the thumbnails are
  277.     //    // at least 360 pixels tall. If the thumbnail height reaches 720 pixels then the application will switch to
  278.     //    // the high quality.
  279.     //    minHeightForQualityLvl: {
  280.     //        360: 'standard,
  281.     //        720: 'high'
  282.     //    }
  283.     // },
  284.  
  285.     // // Options for the recording limit notification.
  286.     // recordingLimit: {
  287.     //
  288.     //    // The recording limit in minutes. Note: This number appears in the notification text
  289.     //    // but doesn't enforce the actual recording time limit. This should be configured in
  290.     //    // jibri!
  291.     //    limit: 60,
  292.     //
  293.     //    // The name of the app with unlimited recordings.
  294.     //    appName: 'Unlimited recordings APP',
  295.     //
  296.     //    // The URL of the app with unlimited recordings.
  297.     //    appURL: 'https://unlimited.recordings.app.com/'
  298.     // },
  299.  
  300.     // Disables or enables RTX (RFC 4588) (defaults to false).
  301.     // disableRtx: false,
  302.  
  303.     // Disables or enables TCC (the default is in Jicofo and set to true)
  304.     // (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting
  305.     // affects congestion control, it practically enables send-side bandwidth
  306.     // estimations.
  307.     // enableTcc: true,
  308.  
  309.     // Disables or enables REMB (the default is in Jicofo and set to false)
  310.     // (draft-alvestrand-rmcat-remb-03). This setting affects congestion
  311.     // control, it practically enables recv-side bandwidth estimations. When
  312.     // both TCC and REMB are enabled, TCC takes precedence. When both are
  313.     // disabled, then bandwidth estimations are disabled.
  314.     // enableRemb: false,
  315.  
  316.     // Enables ICE restart logic in LJM and displays the page reload overlay on
  317.     // ICE failure. Current disabled by default because it's causing issues with
  318.     // signaling when Octo is enabled. Also when we do an "ICE restart"(which is
  319.     // not a real ICE restart), the client maintains the TCC sequence number
  320.     // counter, but the bridge resets it. The bridge sends media packets with
  321.     // TCC sequence numbers starting from 0.
  322.     // enableIceRestart: false,
  323.  
  324.     // Defines the minimum number of participants to start a call (the default
  325.     // is set in Jicofo and set to 2).
  326.     // minParticipants: 2,
  327.  
  328.     // Use the TURN servers discovered via XEP-0215 for the jitsi-videobridge
  329.     // connection
  330.     useStunTurn: true,
  331.  
  332.     // Use TURN/UDP servers for the jitsi-videobridge connection (by default
  333.     // we filter out TURN/UDP because it is usually not needed since the
  334.     // bridge itself is reachable via UDP)
  335.     // useTurnUdp: false
  336.  
  337.     // Enables / disables a data communication channel with the Videobridge.
  338.     // Values can be 'datachannel', 'websocket', true (treat it as
  339.     // 'datachannel'), undefined (treat it as 'datachannel') and false (don't
  340.     // open any channel).
  341.     // openBridgeChannel: true,
  342.  
  343.  
  344.     // UI
  345.     //
  346.  
  347.     // Hides lobby button
  348.     // hideLobbyButton: false,
  349.  
  350.     // Require users to always specify a display name.
  351.     // requireDisplayName: true,
  352.  
  353.     // Whether to use a welcome page or not. In case it's false a random room
  354.     // will be joined when no room is specified.
  355.     enableWelcomePage: true,
  356.  
  357.     // Enabling the close page will ignore the welcome page redirection when
  358.     // a call is hangup.
  359.     // enableClosePage: false,
  360.  
  361.     // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
  362.     // disable1On1Mode: false,
  363.  
  364.     // Default language for the user interface.
  365.     // defaultLanguage: 'en',
  366.  
  367.     // If true all users without a token will be considered guests and all users
  368.     // with token will be considered non-guests. Only guests will be allowed to
  369.     // edit their profile.
  370.     enableUserRolesBasedOnToken: false,
  371.  
  372.     // Whether or not some features are checked based on token.
  373.     // enableFeaturesBasedOnToken: false,
  374.  
  375.     // Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests.
  376.     // lockRoomGuestEnabled: false,
  377.  
  378.     // When enabled the password used for locking a room is restricted to up to the number of digits specified
  379.     // roomPasswordNumberOfDigits: 10,
  380.     // default: roomPasswordNumberOfDigits: false,
  381.  
  382.     // Message to show the users. Example: 'The service will be down for
  383.     // maintenance at 01:00 AM GMT,
  384.     // noticeMessage: '',
  385.  
  386.     // Enables calendar integration, depends on googleApiApplicationClientID
  387.     // and microsoftApiApplicationClientID
  388.     // enableCalendarIntegration: false,
  389.  
  390.     // When 'true', it shows an intermediate page before joining, where the user can configure their devices.
  391.     // prejoinPageEnabled: false,
  392.  
  393.     // If true, shows the unsafe room name warning label when a room name is
  394.     // deemed unsafe (due to the simplicity in the name) and a password is not
  395.     // set or the lobby is not enabled.
  396.     // enableInsecureRoomNameWarning: false,
  397.  
  398.     // Whether to automatically copy invitation URL after creating a room.
  399.     // Document should be focused for this option to work
  400.     // enableAutomaticUrlCopy: false,
  401.  
  402.     // Stats
  403.     //
  404.  
  405.     // Whether to enable stats collection or not in the TraceablePeerConnection.
  406.     // This can be useful for debugging purposes (post-processing/analysis of
  407.     // the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
  408.     // estimation tests.
  409.     // gatherStats: false,
  410.  
  411.     // The interval at which PeerConnection.getStats() is called. Defaults to 10000
  412.     // pcStatsInterval: 10000,
  413.  
  414.     // To enable sending statistics to callstats.io you must provide the
  415.     // Application ID and Secret.
  416.     // callStatsID: '',
  417.     // callStatsSecret: '',
  418.  
  419.     // Enables sending participants' display names to callstats
  420.     // enableDisplayNameInStats: false,
  421.  
  422.     // Enables sending participants' emails (if available) to callstats and other analytics
  423.     // enableEmailInStats: false,
  424.  
  425.     // Privacy
  426.     //
  427.  
  428.     // If third party requests are disabled, no other server will be contacted.
  429.     // This means avatars will be locally generated and callstats integration
  430.     // will not function.
  431.     // disableThirdPartyRequests: false,
  432.  
  433.  
  434.     // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
  435.     //
  436.  
  437.     p2p: {
  438.         // Enables peer to peer mode. When enabled the system will try to
  439.         // establish a direct connection when there are exactly 2 participants
  440.         // in the room. If that succeeds the conference will stop sending data
  441.         // through the JVB and use the peer to peer connection instead. When a
  442.         // 3rd participant joins the conference will be moved back to the JVB
  443.         // connection.
  444.         enabled: true,
  445.  
  446.         // Use XEP-0215 to fetch STUN and TURN servers.
  447.         useStunTurn: true,
  448.  
  449.         // The STUN servers that will be used in the peer to peer connections
  450.         stunServers: [
  451.  
  452.             // { urls: 'stun:meet.mazegeek.com:3478' },
  453.             { urls: 'stun:meet-jit-si-turnrelay.jitsi.net:443' }
  454.         ]
  455.  
  456.         // Sets the ICE transport policy for the p2p connection. At the time
  457.         // of this writing the list of possible values are 'all' and 'relay',
  458.         // but that is subject to change in the future. The enum is defined in
  459.         // the WebRTC standard:
  460.         // https://www.w3.org/TR/webrtc/#rtcicetransportpolicy-enum.
  461.         // If not set, the effective value is 'all'.
  462.         // iceTransportPolicy: 'all',
  463.  
  464.         // If set to true, it will prefer to use H.264 for P2P calls (if H.264
  465.         // is supported). This setting is deprecated, use preferredCodec instead.
  466.         // preferH264: true
  467.  
  468.         // Provides a way to set the video codec preference on the p2p connection. Acceptable
  469.         // codec values are 'VP8', 'VP9' and 'H264'.
  470.         // preferredCodec: 'H264',
  471.  
  472.         // If set to true, disable H.264 video codec by stripping it out of the
  473.         // SDP. This setting is deprecated, use disabledCodec instead.
  474.         // disableH264: false,
  475.  
  476.         // Provides a way to prevent a video codec from being negotiated on the p2p connection.
  477.         // disabledCodec: '',
  478.  
  479.         // How long we're going to wait, before going back to P2P after the 3rd
  480.         // participant has left the conference (to filter out page reload).
  481.         // backToP2PDelay: 5
  482.     },
  483.  
  484.     analytics: {
  485.         // The Google Analytics Tracking ID:
  486.         // googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'
  487.  
  488.         // Matomo configuration:
  489.         // matomoEndpoint: 'https://your-matomo-endpoint/',
  490.         // matomoSiteID: '42',
  491.  
  492.         // The Amplitude APP Key:
  493.         // amplitudeAPPKey: '<APP_KEY>'
  494.  
  495.         // Configuration for the rtcstats server:
  496.         // By enabling rtcstats server every time a conference is joined the rtcstats
  497.         // module connects to the provided rtcstatsEndpoint and sends statistics regarding
  498.         // PeerConnection states along with getStats metrics polled at the specified
  499.         // interval.
  500.         // rtcstatsEnabled: true,
  501.  
  502.         // In order to enable rtcstats one needs to provide a endpoint url.
  503.         // rtcstatsEndpoint: wss://rtcstats-server-pilot.jitsi.net/,
  504.  
  505.         // The interval at which rtcstats will poll getStats, defaults to 1000ms.
  506.         // If the value is set to 0 getStats won't be polled and the rtcstats client
  507.         // will only send data related to RTCPeerConnection events.
  508.         // rtcstatsPolIInterval: 1000
  509.  
  510.         // Array of script URLs to load as lib-jitsi-meet "analytics handlers".
  511.         // scriptURLs: [
  512.         //      "libs/analytics-ga.min.js", // google-analytics
  513.         //      "https://example.com/my-custom-analytics.js"
  514.         // ],
  515.     },
  516.  
  517.     // Logs that should go be passed through the 'log' event if a handler is defined for it
  518.     // apiLogLevels: ['warn', 'log', 'error', 'info', 'debug'],
  519.  
  520.     // Information about the jitsi-meet instance we are connecting to, including
  521.     // the user region as seen by the server.
  522.     deploymentInfo: {
  523.         // shard: "shard1",
  524.         // region: "europe",
  525.         // userRegion: "asia"
  526.     },
  527.  
  528.     // Decides whether the start/stop recording audio notifications should play on record.
  529.     // disableRecordAudioNotification: false,
  530.  
  531.     // Information for the chrome extension banner
  532.     // chromeExtensionBanner: {
  533.     //     // The chrome extension to be installed address
  534.     //     url: 'https://chrome.google.com/webstore/detail/jitsi-meetings/kglhbbefdnlheedjiejgomgmfplipfeb',
  535.  
  536.     //     // Extensions info which allows checking if they are installed or not
  537.     //     chromeExtensionsInfo: [
  538.     //         {
  539.     //             id: 'kglhbbefdnlheedjiejgomgmfplipfeb',
  540.     //             path: 'jitsi-logo-48x48.png'
  541.     //         }
  542.     //     ]
  543.     // },
  544.  
  545.     // Local Recording
  546.     //
  547.  
  548.     // localRecording: {
  549.     // Enables local recording.
  550.     // Additionally, 'localrecording' (all lowercase) needs to be added to
  551.     // TOOLBAR_BUTTONS in interface_config.js for the Local Recording
  552.     // button to show up on the toolbar.
  553.     //
  554.     //     enabled: true,
  555.     //
  556.  
  557.     // The recording format, can be one of 'ogg', 'flac' or 'wav'.
  558.     //     format: 'flac'
  559.     //
  560.  
  561.     // },
  562.  
  563.     // Options related to end-to-end (participant to participant) ping.
  564.     // e2eping: {
  565.     //   // The interval in milliseconds at which pings will be sent.
  566.     //   // Defaults to 10000, set to <= 0 to disable.
  567.     //   pingInterval: 10000,
  568.     //
  569.     //   // The interval in milliseconds at which analytics events
  570.     //   // with the measured RTT will be sent. Defaults to 60000, set
  571.     //   // to <= 0 to disable.
  572.     //   analyticsInterval: 60000,
  573.     //   },
  574.  
  575.     // If set, will attempt to use the provided video input device label when
  576.     // triggering a screenshare, instead of proceeding through the normal flow
  577.     // for obtaining a desktop stream.
  578.     // NOTE: This option is experimental and is currently intended for internal
  579.     // use only.
  580.     // _desktopSharingSourceDevice: 'sample-id-or-label',
  581.  
  582.     // If true, any checks to handoff to another application will be prevented
  583.     // and instead the app will continue to display in the current browser.
  584.     // disableDeepLinking: false,
  585.  
  586.     // A property to disable the right click context menu for localVideo
  587.     // the menu has option to flip the locally seen video for local presentations
  588.     // disableLocalVideoFlip: false,
  589.  
  590.     // Mainly privacy related settings
  591.  
  592.     // Disables all invite functions from the app (share, invite, dial out...etc)
  593.     // disableInviteFunctions: true,
  594.  
  595.     // Disables storing the room name to the recents list
  596.     // doNotStoreRoom: true,
  597.  
  598.     // Deployment specific URLs.
  599.     // deploymentUrls: {
  600.     //    // If specified a 'Help' button will be displayed in the overflow menu with a link to the specified URL for
  601.     //    // user documentation.
  602.     //    userDocumentationURL: 'https://docs.example.com/video-meetings.html',
  603.     //    // If specified a 'Download our apps' button will be displayed in the overflow menu with a link
  604.     //    // to the specified URL for an app download page.
  605.     //    downloadAppsUrl: 'https://docs.example.com/our-apps.html'
  606.     // },
  607.  
  608.     // Options related to the remote participant menu.
  609.     // remoteVideoMenu: {
  610.     //     // If set to true the 'Kick out' button will be disabled.
  611.     //     disableKick: true
  612.     // },
  613.  
  614.     // If set to true all muting operations of remote participants will be disabled.
  615.     // disableRemoteMute: true,
  616.  
  617.     /**
  618.      External API url used to receive branding specific information.
  619.      If there is no url set or there are missing fields, the defaults are applied.
  620.      None of the fields are mandatory and the response must have the shape:
  621.      {
  622.          // The hex value for the colour used as background
  623.          backgroundColor: '#fff',
  624.          // The url for the image used as background
  625.          backgroundImageUrl: 'https://example.com/background-img.png',
  626.          // The anchor url used when clicking the logo image
  627.          logoClickUrl: 'https://example-company.org',
  628.          // The url used for the image used as logo
  629.          logoImageUrl: 'https://example.com/logo-img.png'
  630.      }
  631.     */
  632.     // brandingDataUrl: '',
  633.  
  634.     // The URL of the moderated rooms microservice, if available. If it
  635.     // is present, a link to the service will be rendered on the welcome page,
  636.     // otherwise the app doesn't render it.
  637.     // moderatedRoomServiceUrl: 'https://moderated.meet.mazegeek.com',
  638.  
  639.     // List of undocumented settings used in jitsi-meet
  640.     /**
  641.      _immediateReloadThreshold
  642.      debug
  643.      debugAudioLevels
  644.      deploymentInfo
  645.      dialInConfCodeUrl
  646.      dialInNumbersUrl
  647.      dialOutAuthUrl
  648.      dialOutCodesUrl
  649.      disableRemoteControl
  650.      displayJids
  651.      etherpad_base
  652.      externalConnectUrl
  653.      firefox_fake_device
  654.      googleApiApplicationClientID
  655.      iAmRecorder
  656.      iAmSipGateway
  657.      microsoftApiApplicationClientID
  658.      peopleSearchQueryTypes
  659.      peopleSearchUrl
  660.      requireDisplayName
  661.      tokenAuthUrl
  662.      */
  663.  
  664.     /**
  665.      * This property can be used to alter the generated meeting invite links (in combination with a branding domain
  666.      * which is retrieved internally by jitsi meet) (e.g. https://meet.mazegeek.com/someMeeting
  667.      * can become https://brandedDomain/roomAlias)
  668.      */
  669.     // brandingRoomAlias: null,
  670.  
  671.     // List of undocumented settings used in lib-jitsi-meet
  672.     /**
  673.      _peerConnStatusOutOfLastNTimeout
  674.      _peerConnStatusRtcMuteTimeout
  675.      abTesting
  676.      avgRtpStatsN
  677.      callStatsConfIDNamespace
  678.      callStatsCustomScriptUrl
  679.      desktopSharingSources
  680.      disableAEC
  681.      disableAGC
  682.      disableAP
  683.      disableHPF
  684.      disableNS
  685.      enableLipSync
  686.      enableTalkWhileMuted
  687.      forceJVB121Ratio
  688.      hiddenDomain
  689.      ignoreStartMuted
  690.      nick
  691.      startBitrate
  692.      */
  693.  
  694.  
  695.     // Allow all above example options to include a trailing comma and
  696.     // prevent fear when commenting out the last value.
  697.     makeJsonParserHappy: 'even if last key had a trailing comma'
  698.  
  699.     // no configuration value should follow this line.
  700. };
  701.  
  702. /* eslint-enable no-unused-vars, no-var */
  703.  
  704. export default jitsiConnectionOptions;
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