diff --git a/Jitsi/configmap.yaml b/Jitsi/configmap.yaml index 844c1cc..638c995 100644 --- a/Jitsi/configmap.yaml +++ b/Jitsi/configmap.yaml @@ -4,10 +4,10 @@ name: jvb-config data: sip-communicator.properties: | - org.jitsi.videobridge.NAT_HARVESTER_LOCAL_ADDRESS=home.jerxie.com + org.jitsi.videobridge.NAT_HARVESTER_LOCAL_ADDRESS=127.0.0.1 org.jitsi.videobridge.NAT_HARVESTER_PUBLIC_ADDRESS=home.jerxie.com org.ice4j.ice.harvest.DISABLE_AWS_HARVESTER=true - org.ice4j.ice.harvest.STUN_MAPPING_HARVESTER_ADDRESSES=meet-jit-si-turnrelay.jitsi.net:443 + #org.ice4j.ice.harvest.STUN_MAPPING_HARVESTER_ADDRESSES=meet-jit-si-turnrelay.jitsi.net:443 org.jitsi.videobridge.ENABLE_STATISTICS=true org.jitsi.videobridge.STATISTICS_TRANSPORT=muc org.jitsi.videobridge.xmpp.user.shard.HOSTNAME=localhost @@ -16,3 +16,1074 @@ org.jitsi.videobridge.xmpp.user.shard.PASSWORD=109ndhgh org.jitsi.videobridge.xmpp.user.shard.MUC_JIDS=JvbBrewery@internal.auth.localhost org.jitsi.videobridge.xmpp.user.shard.MUC_NICKNAME=e35c27c4-db13-4673-acbf-63848217b83f +--- +apiVersion: v1 +kind: ConfigMap +metadata: + name: web-config +data: + config.js: | + /* eslint-disable no-unused-vars, no-var */ + + var config = { + // Connection + // + + hosts: { + // XMPP domain. + domain: 'jitsi-meet.example.com', + + // When using authentication, domain for guest users. + // anonymousdomain: 'guest.example.com', + + // Domain for authenticated users. Defaults to . + // authdomain: 'jitsi-meet.example.com', + + // Focus component domain. Defaults to focus.. + // focus: 'focus.jitsi-meet.example.com', + + // XMPP MUC domain. FIXME: use XEP-0030 to discover it. + muc: 'conference.jitsi-meet.example.com' + }, + + // BOSH URL. FIXME: use XEP-0156 to discover it. + bosh: '//jitsi-meet.example.com/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. This setting has been deprecated. + // desktopSharingFrameRate.max now determines whether simulcast will be enabled + // or disabled for the screenshare. + // capScreenshareBitrate: 1 // 0 to disable - deprecated. + + // 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. + }, + + // Enables reactions feature. + // enableReactions: false, + + // Disables ICE/UDP by filtering out local and remote UDP candidates in + // signalling. + // webrtcIceUdpDisable: false, + + // Disables ICE/TCP by filtering out local and remote TCP candidates in + // signalling. + // webrtcIceTcpDisable: false, + + + // Media + // + + // Enable unified plan implementation support on Chromium based browsers. + // enableUnifiedOnChrome: false, + + // 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 hide the "Show More" link in the GSM popover that can be + // used to display more statistics about the connection (IP, Port, protocol, etc). + // disableShowMoreStats: true, + + // Enabling this will run the lib-jitsi-meet noise detection module which will + // notify the user if there is noise, other than voice, coming from the current + // selected microphone. The purpose it to let the user know that the input could + // be potentially unpleasant for other meeting participants. + enableNoisyMicDetection: true, + + // Start the conference in audio only mode (no video is being received nor + // sent). + // startAudioOnly: false, + + // Every participant after the Nth will start audio muted. + // startAudioMuted: 10, + + // Start calls with audio muted. Unlike the option above, this one is only + // applied locally. FIXME: having these 2 options is confusing. + // startWithAudioMuted: false, + + // Enabling it (with #params) will disable local audio output of remote + // participants and to enable it back a reload is needed. + // startSilent: false + + // Enables support for opus-red (redundancy for Opus). + // enableOpusRed: false, + + // Specify audio quality stereo and opusMaxAverageBitrate values in order to enable HD audio. + // Beware, by doing so, you are disabling echo cancellation, noise suppression and AGC. + // audioQuality: { + // stereo: false, + // opusMaxAverageBitrate: null // Value to fit the 6000 to 510000 range. + // }, + + // 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: '' // 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, + + // If true transcriber will use the application language. + // The application language is either explicitly set by participants in their settings or automatically + // detected based on the environment, e.g. if the app is opened in a chrome instance which is using french as its + // default language then transcriptions for that participant will be in french. + // Defaults to true. + // transcribeWithAppLanguage: true, + + // Transcriber language. This settings will only work if "transcribeWithAppLanguage" is explicitly set to false. + // Available languages can be found in lang/language.json. + // preferredTranscribeLanguage: 'en', + + // 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 for the lastN value to be controlled through the UI. + // When startLastN is present, conference starts with a last-n value of startLastN and channelLastN + // value will be used when the quality level is selected using "Manage Video Quality" slider. + // startLastN: 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 + // }, + + // Provides a way to translate the legacy bridge signaling messages, 'LastNChangedEvent', + // 'SelectedEndpointsChangedEvent' and 'ReceiverVideoConstraint' into the new 'ReceiverVideoConstraints' message + // that invokes the new bandwidth allocation algorithm in the bridge which is described here + // - https://github.com/jitsi/jitsi-videobridge/blob/master/doc/allocation.md. + // useNewBandwidthAllocationStrategy: false, + + // 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 enforce the preferred codec for the conference even when the conference has endpoints + // // that do not support the preferred codec. For example, older versions of Safari do not support VP9 yet. + // // This will result in Safari not being able to decode video from endpoints sending VP9 video. + // // When set to false, the conference falls back to VP8 whenever there is an endpoint that doesn't support the + // // preferred codec and goes back to the preferred codec when that endpoint leaves. + // // enforcePreferredCodec: false, + // + // // 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: { + // H264: { + // low: 200000, + // standard: 500000, + // high: 1500000 + // }, + // VP8 : { + // low: 200000, + // standard: 500000, + // high: 1500000 + // }, + // VP9: { + // low: 100000, + // standard: 300000, + // high: 1200000 + // } + // }, + // + // // 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, + + // Enables forced reload of the client when the call is migrated as a result of + // the bridge going down. + // enableForcedReload: true, + + // 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. + // Setting this to null, will also disable showing the remote videos + // when the toolbar is shown on mouse movements + // disable1On1Mode: null | false | true, + + // Default language for the user interface. + // defaultLanguage: 'en', + + // 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/', + + // App name to be displayed in the invitation email subject, as an alternative to + // interfaceConfig.APP_NAME. + // inviteAppName: null, + + // Moved from interfaceConfig(TOOLBAR_BUTTONS). + // The name of the toolbar buttons to display in the toolbar, including the + // "More actions" menu. If present, the button will display. Exceptions are + // "livestreaming" and "recording" which also require being a moderator and + // some other values in config.js to be enabled. Also, the "profile" button will + // not display for users with a JWT. + // Notes: + // - it's impossible to choose which buttons go in the "More actions" menu + // - it's impossible to control the placement of buttons + // - 'desktop' controls the "Share your screen" button + // - if `toolbarButtons` is undefined, we fallback to enabling all buttons on the UI + // toolbarButtons: [ + // 'camera', + // 'chat', + // 'closedcaptions', + // 'desktop', + // 'download', + // 'embedmeeting', + // 'etherpad', + // 'feedback', + // 'filmstrip', + // 'fullscreen', + // 'hangup', + // 'help', + // 'invite', + // 'livestreaming', + // 'microphone', + // 'mute-everyone', + // 'mute-video-everyone', + // 'participants-pane', + // 'profile', + // 'raisehand', + // 'recording', + // 'security', + // 'select-background', + // 'settings', + // 'shareaudio', + // 'sharedvideo', + // 'shortcuts', + // 'stats', + // 'tileview', + // 'toggle-camera', + // 'videoquality', + // '__end' + // ], + + // 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, + + // Controls the percentage of automatic feedback shown to participants when callstats is enabled. + // The default value is 100%. If set to 0, no automatic feedback will be requested + // feedbackPercentage: 100, + + // 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, + + // Enable unified plan implementation support on Chromium for p2p connection. + // enableUnifiedOnChrome: false, + + // 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, + + // 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' } + ] + }, + + 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: '' + + // 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, + + // Disables the sounds that play when other participants join or leave the + // conference (if set to true, these sounds will not be played). + // disableJoinLeaveSounds: false, + + // Disables the sounds that play when a chat message is received. + // disableIncomingMessageSound: 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 + // the `toolbarButtons`-array 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, + + // A property used to unset the default flip state of the local video. + // When it is set to 'true', the local(self) video will not be mirrored anymore. + // doNotFlipLocalVideo: 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 the 'Grant moderator' button will be disabled. + // disableGrantModerator: 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. + The config file should be in JSON. + None of the fields are mandatory and the response must have the shape: + { + // The domain url to apply (will replace the domain in the sharing conference link/embed section) + inviteDomain: 'example-company.org, + // 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: '', + + // Sets the background transparency level. '0' is fully transparent, '1' is opaque. + // backgroundAlpha: 1, + + // 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 recording label + // hideRecordingLabel: false, + + // Hides the conference timer. + // hideConferenceTimer: true, + + // Hides the participants stats + // hideParticipantsStats: true, + + // Sets the conference subject + // subject: 'Conference Subject', + + // This property is related to the use case when jitsi-meet is used via the IFrame API. When the property is true + // jitsi-meet will use the local storage of the host page instead of its own. This option is useful if the browser + // is not persisting the local storage inside the iframe. + // useHostPageLocalStorage: true, + + // etherpad ("shared document") integration. + // + + // If set, add a "Open shared document" link to the bottom right menu that + // will open an etherpad document. + // etherpad_base: 'https://your-etherpad-installati.on/p/', + + // If etherpad_base is set, and useRoomAsSharedDocumentName is set to true, + // open a pad with the name of the room (lowercased) instead of a pad with a + // random UUID. + // useRoomAsSharedDocumentName: true, + + // List of undocumented settings used in jitsi-meet + /** + _immediateReloadThreshold + debug + debugAudioLevels + deploymentInfo + dialInConfCodeUrl + dialInNumbersUrl + dialOutAuthUrl + dialOutCodesUrl + disableRemoteControl + displayJids + 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 + */ + + /** + * Default interval (milliseconds) for triggering mouseMoved iframe API event + */ + mouseMoveCallbackInterval: 1000, + + /** + 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.sessionRestarted', // show when a client reload is initiated because of bridge migration + // '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 + // ], + + // Prevent the filmstrip from autohiding when screen width is under a certain threshold + // disableFilmstripAutohiding: false, + + // 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 */ + + // Begin default config overrides. + + if (!config.hasOwnProperty('hosts')) config.hosts = {}; + + config.hosts.domain = 'meet.jitsi'; + config.focusUserJid = 'focus@auth.meet.jitsi'; + + config.hosts.muc = 'muc.meet.jitsi'; + config.bosh = '/http-bind'; + config.websocket = 'wss://meet.jerxie.com/xmpp-websocket'; + // Video configuration. + // + + if (!config.hasOwnProperty('constraints')) config.constraints = {}; + if (!config.constraints.hasOwnProperty('video')) config.constraints.video = {}; + + config.resolution = 720; + config.constraints.video.height = { ideal: 720, max: 720, min: 180 }; + config.constraints.video.width = { ideal: 1280, max: 1280, min: 320}; + config.disableSimulcast = true; + config.startVideoMuted = 10; + config.startWithVideoMuted = false; + + // ScreenShare Configuration. + // + config.desktopSharingFrameRate = { min: 5, max: 5 }; + + // Audio configuration. + // + + config.enableNoAudioDetection = false; + config.enableTalkWhileMuted = false; + config.disableAP = false; + config.stereo = false; + config.startAudioOnly = false; + config.startAudioMuted = 10; + config.startWithAudioMuted = false; + config.startSilent = false; + config.disableAudioLevels = false; + config.enableNoisyMicDetection = true; + + + // Peer-to-Peer options. + // + + if (!config.hasOwnProperty('p2p')) config.p2p = {}; + + config.p2p.enabled = true; + + + // Etherpad + // + + // Recording. + // + + // Analytics. + // + + if (!config.hasOwnProperty('analytics')) config.analytics = {}; + + // Enables callstatsUsername to be reported as statsId and used + // by callstats as repoted remote id. + config.enableStatsID = false; + + + // Dial in/out services. + // + + // Calendar service integration. + // + + config.enableCalendarIntegration = false; + + // Invitation service. + // + + // Miscellaneous. + // + + // Prejoin page. + config.prejoinPageEnabled = false; + + // Welcome page. + config.enableWelcomePage = true; + + // Close page. + config.enableClosePage = false; + + // Default language. + // Require users to always specify a display name. + config.requireDisplayName = false; + + // Chrome extension banner. + // Advanced. + // + + // Lipsync hack in jicofo, may not be safe. + config.enableLipSync = false; + + config.enableRemb = true; + config.enableTcc = true; + + // Enable IPv6 support. + config.useIPv6 = true; + + // Transcriptions (subtitles and buttons can be configured in interface_config) + config.transcribingEnabled = false; + + // Deployment information. + // + + if (!config.hasOwnProperty('deploymentInfo')) config.deploymentInfo = {}; + + // Testing + // + + if (!config.hasOwnProperty('testing')) config.testing = {}; + if (!config.testing.hasOwnProperty('octo')) config.testing.octo = {}; + + config.testing.capScreenshareBitrate = 1; + config.testing.octo.probability = 0; + + // Deep Linking + config.disableDeepLinking = false; diff --git a/Jitsi/configmap.yaml b/Jitsi/configmap.yaml index 844c1cc..638c995 100644 --- a/Jitsi/configmap.yaml +++ b/Jitsi/configmap.yaml @@ -4,10 +4,10 @@ name: jvb-config data: sip-communicator.properties: | - org.jitsi.videobridge.NAT_HARVESTER_LOCAL_ADDRESS=home.jerxie.com + org.jitsi.videobridge.NAT_HARVESTER_LOCAL_ADDRESS=127.0.0.1 org.jitsi.videobridge.NAT_HARVESTER_PUBLIC_ADDRESS=home.jerxie.com org.ice4j.ice.harvest.DISABLE_AWS_HARVESTER=true - org.ice4j.ice.harvest.STUN_MAPPING_HARVESTER_ADDRESSES=meet-jit-si-turnrelay.jitsi.net:443 + #org.ice4j.ice.harvest.STUN_MAPPING_HARVESTER_ADDRESSES=meet-jit-si-turnrelay.jitsi.net:443 org.jitsi.videobridge.ENABLE_STATISTICS=true org.jitsi.videobridge.STATISTICS_TRANSPORT=muc org.jitsi.videobridge.xmpp.user.shard.HOSTNAME=localhost @@ -16,3 +16,1074 @@ org.jitsi.videobridge.xmpp.user.shard.PASSWORD=109ndhgh org.jitsi.videobridge.xmpp.user.shard.MUC_JIDS=JvbBrewery@internal.auth.localhost org.jitsi.videobridge.xmpp.user.shard.MUC_NICKNAME=e35c27c4-db13-4673-acbf-63848217b83f +--- +apiVersion: v1 +kind: ConfigMap +metadata: + name: web-config +data: + config.js: | + /* eslint-disable no-unused-vars, no-var */ + + var config = { + // Connection + // + + hosts: { + // XMPP domain. + domain: 'jitsi-meet.example.com', + + // When using authentication, domain for guest users. + // anonymousdomain: 'guest.example.com', + + // Domain for authenticated users. Defaults to . + // authdomain: 'jitsi-meet.example.com', + + // Focus component domain. Defaults to focus.. + // focus: 'focus.jitsi-meet.example.com', + + // XMPP MUC domain. FIXME: use XEP-0030 to discover it. + muc: 'conference.jitsi-meet.example.com' + }, + + // BOSH URL. FIXME: use XEP-0156 to discover it. + bosh: '//jitsi-meet.example.com/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. This setting has been deprecated. + // desktopSharingFrameRate.max now determines whether simulcast will be enabled + // or disabled for the screenshare. + // capScreenshareBitrate: 1 // 0 to disable - deprecated. + + // 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. + }, + + // Enables reactions feature. + // enableReactions: false, + + // Disables ICE/UDP by filtering out local and remote UDP candidates in + // signalling. + // webrtcIceUdpDisable: false, + + // Disables ICE/TCP by filtering out local and remote TCP candidates in + // signalling. + // webrtcIceTcpDisable: false, + + + // Media + // + + // Enable unified plan implementation support on Chromium based browsers. + // enableUnifiedOnChrome: false, + + // 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 hide the "Show More" link in the GSM popover that can be + // used to display more statistics about the connection (IP, Port, protocol, etc). + // disableShowMoreStats: true, + + // Enabling this will run the lib-jitsi-meet noise detection module which will + // notify the user if there is noise, other than voice, coming from the current + // selected microphone. The purpose it to let the user know that the input could + // be potentially unpleasant for other meeting participants. + enableNoisyMicDetection: true, + + // Start the conference in audio only mode (no video is being received nor + // sent). + // startAudioOnly: false, + + // Every participant after the Nth will start audio muted. + // startAudioMuted: 10, + + // Start calls with audio muted. Unlike the option above, this one is only + // applied locally. FIXME: having these 2 options is confusing. + // startWithAudioMuted: false, + + // Enabling it (with #params) will disable local audio output of remote + // participants and to enable it back a reload is needed. + // startSilent: false + + // Enables support for opus-red (redundancy for Opus). + // enableOpusRed: false, + + // Specify audio quality stereo and opusMaxAverageBitrate values in order to enable HD audio. + // Beware, by doing so, you are disabling echo cancellation, noise suppression and AGC. + // audioQuality: { + // stereo: false, + // opusMaxAverageBitrate: null // Value to fit the 6000 to 510000 range. + // }, + + // 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: '' // 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, + + // If true transcriber will use the application language. + // The application language is either explicitly set by participants in their settings or automatically + // detected based on the environment, e.g. if the app is opened in a chrome instance which is using french as its + // default language then transcriptions for that participant will be in french. + // Defaults to true. + // transcribeWithAppLanguage: true, + + // Transcriber language. This settings will only work if "transcribeWithAppLanguage" is explicitly set to false. + // Available languages can be found in lang/language.json. + // preferredTranscribeLanguage: 'en', + + // 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 for the lastN value to be controlled through the UI. + // When startLastN is present, conference starts with a last-n value of startLastN and channelLastN + // value will be used when the quality level is selected using "Manage Video Quality" slider. + // startLastN: 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 + // }, + + // Provides a way to translate the legacy bridge signaling messages, 'LastNChangedEvent', + // 'SelectedEndpointsChangedEvent' and 'ReceiverVideoConstraint' into the new 'ReceiverVideoConstraints' message + // that invokes the new bandwidth allocation algorithm in the bridge which is described here + // - https://github.com/jitsi/jitsi-videobridge/blob/master/doc/allocation.md. + // useNewBandwidthAllocationStrategy: false, + + // 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 enforce the preferred codec for the conference even when the conference has endpoints + // // that do not support the preferred codec. For example, older versions of Safari do not support VP9 yet. + // // This will result in Safari not being able to decode video from endpoints sending VP9 video. + // // When set to false, the conference falls back to VP8 whenever there is an endpoint that doesn't support the + // // preferred codec and goes back to the preferred codec when that endpoint leaves. + // // enforcePreferredCodec: false, + // + // // 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: { + // H264: { + // low: 200000, + // standard: 500000, + // high: 1500000 + // }, + // VP8 : { + // low: 200000, + // standard: 500000, + // high: 1500000 + // }, + // VP9: { + // low: 100000, + // standard: 300000, + // high: 1200000 + // } + // }, + // + // // 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, + + // Enables forced reload of the client when the call is migrated as a result of + // the bridge going down. + // enableForcedReload: true, + + // 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. + // Setting this to null, will also disable showing the remote videos + // when the toolbar is shown on mouse movements + // disable1On1Mode: null | false | true, + + // Default language for the user interface. + // defaultLanguage: 'en', + + // 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/', + + // App name to be displayed in the invitation email subject, as an alternative to + // interfaceConfig.APP_NAME. + // inviteAppName: null, + + // Moved from interfaceConfig(TOOLBAR_BUTTONS). + // The name of the toolbar buttons to display in the toolbar, including the + // "More actions" menu. If present, the button will display. Exceptions are + // "livestreaming" and "recording" which also require being a moderator and + // some other values in config.js to be enabled. Also, the "profile" button will + // not display for users with a JWT. + // Notes: + // - it's impossible to choose which buttons go in the "More actions" menu + // - it's impossible to control the placement of buttons + // - 'desktop' controls the "Share your screen" button + // - if `toolbarButtons` is undefined, we fallback to enabling all buttons on the UI + // toolbarButtons: [ + // 'camera', + // 'chat', + // 'closedcaptions', + // 'desktop', + // 'download', + // 'embedmeeting', + // 'etherpad', + // 'feedback', + // 'filmstrip', + // 'fullscreen', + // 'hangup', + // 'help', + // 'invite', + // 'livestreaming', + // 'microphone', + // 'mute-everyone', + // 'mute-video-everyone', + // 'participants-pane', + // 'profile', + // 'raisehand', + // 'recording', + // 'security', + // 'select-background', + // 'settings', + // 'shareaudio', + // 'sharedvideo', + // 'shortcuts', + // 'stats', + // 'tileview', + // 'toggle-camera', + // 'videoquality', + // '__end' + // ], + + // 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, + + // Controls the percentage of automatic feedback shown to participants when callstats is enabled. + // The default value is 100%. If set to 0, no automatic feedback will be requested + // feedbackPercentage: 100, + + // 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, + + // Enable unified plan implementation support on Chromium for p2p connection. + // enableUnifiedOnChrome: false, + + // 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, + + // 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' } + ] + }, + + 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: '' + + // 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, + + // Disables the sounds that play when other participants join or leave the + // conference (if set to true, these sounds will not be played). + // disableJoinLeaveSounds: false, + + // Disables the sounds that play when a chat message is received. + // disableIncomingMessageSound: 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 + // the `toolbarButtons`-array 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, + + // A property used to unset the default flip state of the local video. + // When it is set to 'true', the local(self) video will not be mirrored anymore. + // doNotFlipLocalVideo: 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 the 'Grant moderator' button will be disabled. + // disableGrantModerator: 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. + The config file should be in JSON. + None of the fields are mandatory and the response must have the shape: + { + // The domain url to apply (will replace the domain in the sharing conference link/embed section) + inviteDomain: 'example-company.org, + // 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: '', + + // Sets the background transparency level. '0' is fully transparent, '1' is opaque. + // backgroundAlpha: 1, + + // 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 recording label + // hideRecordingLabel: false, + + // Hides the conference timer. + // hideConferenceTimer: true, + + // Hides the participants stats + // hideParticipantsStats: true, + + // Sets the conference subject + // subject: 'Conference Subject', + + // This property is related to the use case when jitsi-meet is used via the IFrame API. When the property is true + // jitsi-meet will use the local storage of the host page instead of its own. This option is useful if the browser + // is not persisting the local storage inside the iframe. + // useHostPageLocalStorage: true, + + // etherpad ("shared document") integration. + // + + // If set, add a "Open shared document" link to the bottom right menu that + // will open an etherpad document. + // etherpad_base: 'https://your-etherpad-installati.on/p/', + + // If etherpad_base is set, and useRoomAsSharedDocumentName is set to true, + // open a pad with the name of the room (lowercased) instead of a pad with a + // random UUID. + // useRoomAsSharedDocumentName: true, + + // List of undocumented settings used in jitsi-meet + /** + _immediateReloadThreshold + debug + debugAudioLevels + deploymentInfo + dialInConfCodeUrl + dialInNumbersUrl + dialOutAuthUrl + dialOutCodesUrl + disableRemoteControl + displayJids + 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 + */ + + /** + * Default interval (milliseconds) for triggering mouseMoved iframe API event + */ + mouseMoveCallbackInterval: 1000, + + /** + 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.sessionRestarted', // show when a client reload is initiated because of bridge migration + // '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 + // ], + + // Prevent the filmstrip from autohiding when screen width is under a certain threshold + // disableFilmstripAutohiding: false, + + // 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 */ + + // Begin default config overrides. + + if (!config.hasOwnProperty('hosts')) config.hosts = {}; + + config.hosts.domain = 'meet.jitsi'; + config.focusUserJid = 'focus@auth.meet.jitsi'; + + config.hosts.muc = 'muc.meet.jitsi'; + config.bosh = '/http-bind'; + config.websocket = 'wss://meet.jerxie.com/xmpp-websocket'; + // Video configuration. + // + + if (!config.hasOwnProperty('constraints')) config.constraints = {}; + if (!config.constraints.hasOwnProperty('video')) config.constraints.video = {}; + + config.resolution = 720; + config.constraints.video.height = { ideal: 720, max: 720, min: 180 }; + config.constraints.video.width = { ideal: 1280, max: 1280, min: 320}; + config.disableSimulcast = true; + config.startVideoMuted = 10; + config.startWithVideoMuted = false; + + // ScreenShare Configuration. + // + config.desktopSharingFrameRate = { min: 5, max: 5 }; + + // Audio configuration. + // + + config.enableNoAudioDetection = false; + config.enableTalkWhileMuted = false; + config.disableAP = false; + config.stereo = false; + config.startAudioOnly = false; + config.startAudioMuted = 10; + config.startWithAudioMuted = false; + config.startSilent = false; + config.disableAudioLevels = false; + config.enableNoisyMicDetection = true; + + + // Peer-to-Peer options. + // + + if (!config.hasOwnProperty('p2p')) config.p2p = {}; + + config.p2p.enabled = true; + + + // Etherpad + // + + // Recording. + // + + // Analytics. + // + + if (!config.hasOwnProperty('analytics')) config.analytics = {}; + + // Enables callstatsUsername to be reported as statsId and used + // by callstats as repoted remote id. + config.enableStatsID = false; + + + // Dial in/out services. + // + + // Calendar service integration. + // + + config.enableCalendarIntegration = false; + + // Invitation service. + // + + // Miscellaneous. + // + + // Prejoin page. + config.prejoinPageEnabled = false; + + // Welcome page. + config.enableWelcomePage = true; + + // Close page. + config.enableClosePage = false; + + // Default language. + // Require users to always specify a display name. + config.requireDisplayName = false; + + // Chrome extension banner. + // Advanced. + // + + // Lipsync hack in jicofo, may not be safe. + config.enableLipSync = false; + + config.enableRemb = true; + config.enableTcc = true; + + // Enable IPv6 support. + config.useIPv6 = true; + + // Transcriptions (subtitles and buttons can be configured in interface_config) + config.transcribingEnabled = false; + + // Deployment information. + // + + if (!config.hasOwnProperty('deploymentInfo')) config.deploymentInfo = {}; + + // Testing + // + + if (!config.hasOwnProperty('testing')) config.testing = {}; + if (!config.testing.hasOwnProperty('octo')) config.testing.octo = {}; + + config.testing.capScreenshareBitrate = 1; + config.testing.octo.probability = 0; + + // Deep Linking + config.disableDeepLinking = false; diff --git a/Jitsi/deployment.yaml b/Jitsi/deployment.yaml index 731a136..24a238e 100644 --- a/Jitsi/deployment.yaml +++ b/Jitsi/deployment.yaml @@ -93,10 +93,10 @@ - name: web image: jitsi/web:stable-5870 imagePullPolicy: IfNotPresent - #volumeMounts: - # - name: base-patch - # mountPath: /usr/share/jitsi-meet/base.html - # subPath: base.html + volumeMounts: + - name: web-patch + mountPath: /config/config.js + subPath: config.js env: - name: PUBLIC_URL value: "https://meet.jerxie.com" @@ -162,6 +162,9 @@ - name: TZ value: America/Los_Angeles volumes: + - name: web-patch + configMap: + name: web-config - name: jvb-patch configMap: name: jvb-config diff --git a/Jitsi/configmap.yaml b/Jitsi/configmap.yaml index 844c1cc..638c995 100644 --- a/Jitsi/configmap.yaml +++ b/Jitsi/configmap.yaml @@ -4,10 +4,10 @@ name: jvb-config data: sip-communicator.properties: | - org.jitsi.videobridge.NAT_HARVESTER_LOCAL_ADDRESS=home.jerxie.com + org.jitsi.videobridge.NAT_HARVESTER_LOCAL_ADDRESS=127.0.0.1 org.jitsi.videobridge.NAT_HARVESTER_PUBLIC_ADDRESS=home.jerxie.com org.ice4j.ice.harvest.DISABLE_AWS_HARVESTER=true - org.ice4j.ice.harvest.STUN_MAPPING_HARVESTER_ADDRESSES=meet-jit-si-turnrelay.jitsi.net:443 + #org.ice4j.ice.harvest.STUN_MAPPING_HARVESTER_ADDRESSES=meet-jit-si-turnrelay.jitsi.net:443 org.jitsi.videobridge.ENABLE_STATISTICS=true org.jitsi.videobridge.STATISTICS_TRANSPORT=muc org.jitsi.videobridge.xmpp.user.shard.HOSTNAME=localhost @@ -16,3 +16,1074 @@ org.jitsi.videobridge.xmpp.user.shard.PASSWORD=109ndhgh org.jitsi.videobridge.xmpp.user.shard.MUC_JIDS=JvbBrewery@internal.auth.localhost org.jitsi.videobridge.xmpp.user.shard.MUC_NICKNAME=e35c27c4-db13-4673-acbf-63848217b83f +--- +apiVersion: v1 +kind: ConfigMap +metadata: + name: web-config +data: + config.js: | + /* eslint-disable no-unused-vars, no-var */ + + var config = { + // Connection + // + + hosts: { + // XMPP domain. + domain: 'jitsi-meet.example.com', + + // When using authentication, domain for guest users. + // anonymousdomain: 'guest.example.com', + + // Domain for authenticated users. Defaults to . + // authdomain: 'jitsi-meet.example.com', + + // Focus component domain. Defaults to focus.. + // focus: 'focus.jitsi-meet.example.com', + + // XMPP MUC domain. FIXME: use XEP-0030 to discover it. + muc: 'conference.jitsi-meet.example.com' + }, + + // BOSH URL. FIXME: use XEP-0156 to discover it. + bosh: '//jitsi-meet.example.com/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. This setting has been deprecated. + // desktopSharingFrameRate.max now determines whether simulcast will be enabled + // or disabled for the screenshare. + // capScreenshareBitrate: 1 // 0 to disable - deprecated. + + // 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. + }, + + // Enables reactions feature. + // enableReactions: false, + + // Disables ICE/UDP by filtering out local and remote UDP candidates in + // signalling. + // webrtcIceUdpDisable: false, + + // Disables ICE/TCP by filtering out local and remote TCP candidates in + // signalling. + // webrtcIceTcpDisable: false, + + + // Media + // + + // Enable unified plan implementation support on Chromium based browsers. + // enableUnifiedOnChrome: false, + + // 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 hide the "Show More" link in the GSM popover that can be + // used to display more statistics about the connection (IP, Port, protocol, etc). + // disableShowMoreStats: true, + + // Enabling this will run the lib-jitsi-meet noise detection module which will + // notify the user if there is noise, other than voice, coming from the current + // selected microphone. The purpose it to let the user know that the input could + // be potentially unpleasant for other meeting participants. + enableNoisyMicDetection: true, + + // Start the conference in audio only mode (no video is being received nor + // sent). + // startAudioOnly: false, + + // Every participant after the Nth will start audio muted. + // startAudioMuted: 10, + + // Start calls with audio muted. Unlike the option above, this one is only + // applied locally. FIXME: having these 2 options is confusing. + // startWithAudioMuted: false, + + // Enabling it (with #params) will disable local audio output of remote + // participants and to enable it back a reload is needed. + // startSilent: false + + // Enables support for opus-red (redundancy for Opus). + // enableOpusRed: false, + + // Specify audio quality stereo and opusMaxAverageBitrate values in order to enable HD audio. + // Beware, by doing so, you are disabling echo cancellation, noise suppression and AGC. + // audioQuality: { + // stereo: false, + // opusMaxAverageBitrate: null // Value to fit the 6000 to 510000 range. + // }, + + // 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: '' // 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, + + // If true transcriber will use the application language. + // The application language is either explicitly set by participants in their settings or automatically + // detected based on the environment, e.g. if the app is opened in a chrome instance which is using french as its + // default language then transcriptions for that participant will be in french. + // Defaults to true. + // transcribeWithAppLanguage: true, + + // Transcriber language. This settings will only work if "transcribeWithAppLanguage" is explicitly set to false. + // Available languages can be found in lang/language.json. + // preferredTranscribeLanguage: 'en', + + // 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 for the lastN value to be controlled through the UI. + // When startLastN is present, conference starts with a last-n value of startLastN and channelLastN + // value will be used when the quality level is selected using "Manage Video Quality" slider. + // startLastN: 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 + // }, + + // Provides a way to translate the legacy bridge signaling messages, 'LastNChangedEvent', + // 'SelectedEndpointsChangedEvent' and 'ReceiverVideoConstraint' into the new 'ReceiverVideoConstraints' message + // that invokes the new bandwidth allocation algorithm in the bridge which is described here + // - https://github.com/jitsi/jitsi-videobridge/blob/master/doc/allocation.md. + // useNewBandwidthAllocationStrategy: false, + + // 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 enforce the preferred codec for the conference even when the conference has endpoints + // // that do not support the preferred codec. For example, older versions of Safari do not support VP9 yet. + // // This will result in Safari not being able to decode video from endpoints sending VP9 video. + // // When set to false, the conference falls back to VP8 whenever there is an endpoint that doesn't support the + // // preferred codec and goes back to the preferred codec when that endpoint leaves. + // // enforcePreferredCodec: false, + // + // // 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: { + // H264: { + // low: 200000, + // standard: 500000, + // high: 1500000 + // }, + // VP8 : { + // low: 200000, + // standard: 500000, + // high: 1500000 + // }, + // VP9: { + // low: 100000, + // standard: 300000, + // high: 1200000 + // } + // }, + // + // // 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, + + // Enables forced reload of the client when the call is migrated as a result of + // the bridge going down. + // enableForcedReload: true, + + // 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. + // Setting this to null, will also disable showing the remote videos + // when the toolbar is shown on mouse movements + // disable1On1Mode: null | false | true, + + // Default language for the user interface. + // defaultLanguage: 'en', + + // 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/', + + // App name to be displayed in the invitation email subject, as an alternative to + // interfaceConfig.APP_NAME. + // inviteAppName: null, + + // Moved from interfaceConfig(TOOLBAR_BUTTONS). + // The name of the toolbar buttons to display in the toolbar, including the + // "More actions" menu. If present, the button will display. Exceptions are + // "livestreaming" and "recording" which also require being a moderator and + // some other values in config.js to be enabled. Also, the "profile" button will + // not display for users with a JWT. + // Notes: + // - it's impossible to choose which buttons go in the "More actions" menu + // - it's impossible to control the placement of buttons + // - 'desktop' controls the "Share your screen" button + // - if `toolbarButtons` is undefined, we fallback to enabling all buttons on the UI + // toolbarButtons: [ + // 'camera', + // 'chat', + // 'closedcaptions', + // 'desktop', + // 'download', + // 'embedmeeting', + // 'etherpad', + // 'feedback', + // 'filmstrip', + // 'fullscreen', + // 'hangup', + // 'help', + // 'invite', + // 'livestreaming', + // 'microphone', + // 'mute-everyone', + // 'mute-video-everyone', + // 'participants-pane', + // 'profile', + // 'raisehand', + // 'recording', + // 'security', + // 'select-background', + // 'settings', + // 'shareaudio', + // 'sharedvideo', + // 'shortcuts', + // 'stats', + // 'tileview', + // 'toggle-camera', + // 'videoquality', + // '__end' + // ], + + // 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, + + // Controls the percentage of automatic feedback shown to participants when callstats is enabled. + // The default value is 100%. If set to 0, no automatic feedback will be requested + // feedbackPercentage: 100, + + // 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, + + // Enable unified plan implementation support on Chromium for p2p connection. + // enableUnifiedOnChrome: false, + + // 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, + + // 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' } + ] + }, + + 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: '' + + // 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, + + // Disables the sounds that play when other participants join or leave the + // conference (if set to true, these sounds will not be played). + // disableJoinLeaveSounds: false, + + // Disables the sounds that play when a chat message is received. + // disableIncomingMessageSound: 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 + // the `toolbarButtons`-array 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, + + // A property used to unset the default flip state of the local video. + // When it is set to 'true', the local(self) video will not be mirrored anymore. + // doNotFlipLocalVideo: 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 the 'Grant moderator' button will be disabled. + // disableGrantModerator: 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. + The config file should be in JSON. + None of the fields are mandatory and the response must have the shape: + { + // The domain url to apply (will replace the domain in the sharing conference link/embed section) + inviteDomain: 'example-company.org, + // 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: '', + + // Sets the background transparency level. '0' is fully transparent, '1' is opaque. + // backgroundAlpha: 1, + + // 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 recording label + // hideRecordingLabel: false, + + // Hides the conference timer. + // hideConferenceTimer: true, + + // Hides the participants stats + // hideParticipantsStats: true, + + // Sets the conference subject + // subject: 'Conference Subject', + + // This property is related to the use case when jitsi-meet is used via the IFrame API. When the property is true + // jitsi-meet will use the local storage of the host page instead of its own. This option is useful if the browser + // is not persisting the local storage inside the iframe. + // useHostPageLocalStorage: true, + + // etherpad ("shared document") integration. + // + + // If set, add a "Open shared document" link to the bottom right menu that + // will open an etherpad document. + // etherpad_base: 'https://your-etherpad-installati.on/p/', + + // If etherpad_base is set, and useRoomAsSharedDocumentName is set to true, + // open a pad with the name of the room (lowercased) instead of a pad with a + // random UUID. + // useRoomAsSharedDocumentName: true, + + // List of undocumented settings used in jitsi-meet + /** + _immediateReloadThreshold + debug + debugAudioLevels + deploymentInfo + dialInConfCodeUrl + dialInNumbersUrl + dialOutAuthUrl + dialOutCodesUrl + disableRemoteControl + displayJids + 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 + */ + + /** + * Default interval (milliseconds) for triggering mouseMoved iframe API event + */ + mouseMoveCallbackInterval: 1000, + + /** + 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.sessionRestarted', // show when a client reload is initiated because of bridge migration + // '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 + // ], + + // Prevent the filmstrip from autohiding when screen width is under a certain threshold + // disableFilmstripAutohiding: false, + + // 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 */ + + // Begin default config overrides. + + if (!config.hasOwnProperty('hosts')) config.hosts = {}; + + config.hosts.domain = 'meet.jitsi'; + config.focusUserJid = 'focus@auth.meet.jitsi'; + + config.hosts.muc = 'muc.meet.jitsi'; + config.bosh = '/http-bind'; + config.websocket = 'wss://meet.jerxie.com/xmpp-websocket'; + // Video configuration. + // + + if (!config.hasOwnProperty('constraints')) config.constraints = {}; + if (!config.constraints.hasOwnProperty('video')) config.constraints.video = {}; + + config.resolution = 720; + config.constraints.video.height = { ideal: 720, max: 720, min: 180 }; + config.constraints.video.width = { ideal: 1280, max: 1280, min: 320}; + config.disableSimulcast = true; + config.startVideoMuted = 10; + config.startWithVideoMuted = false; + + // ScreenShare Configuration. + // + config.desktopSharingFrameRate = { min: 5, max: 5 }; + + // Audio configuration. + // + + config.enableNoAudioDetection = false; + config.enableTalkWhileMuted = false; + config.disableAP = false; + config.stereo = false; + config.startAudioOnly = false; + config.startAudioMuted = 10; + config.startWithAudioMuted = false; + config.startSilent = false; + config.disableAudioLevels = false; + config.enableNoisyMicDetection = true; + + + // Peer-to-Peer options. + // + + if (!config.hasOwnProperty('p2p')) config.p2p = {}; + + config.p2p.enabled = true; + + + // Etherpad + // + + // Recording. + // + + // Analytics. + // + + if (!config.hasOwnProperty('analytics')) config.analytics = {}; + + // Enables callstatsUsername to be reported as statsId and used + // by callstats as repoted remote id. + config.enableStatsID = false; + + + // Dial in/out services. + // + + // Calendar service integration. + // + + config.enableCalendarIntegration = false; + + // Invitation service. + // + + // Miscellaneous. + // + + // Prejoin page. + config.prejoinPageEnabled = false; + + // Welcome page. + config.enableWelcomePage = true; + + // Close page. + config.enableClosePage = false; + + // Default language. + // Require users to always specify a display name. + config.requireDisplayName = false; + + // Chrome extension banner. + // Advanced. + // + + // Lipsync hack in jicofo, may not be safe. + config.enableLipSync = false; + + config.enableRemb = true; + config.enableTcc = true; + + // Enable IPv6 support. + config.useIPv6 = true; + + // Transcriptions (subtitles and buttons can be configured in interface_config) + config.transcribingEnabled = false; + + // Deployment information. + // + + if (!config.hasOwnProperty('deploymentInfo')) config.deploymentInfo = {}; + + // Testing + // + + if (!config.hasOwnProperty('testing')) config.testing = {}; + if (!config.testing.hasOwnProperty('octo')) config.testing.octo = {}; + + config.testing.capScreenshareBitrate = 1; + config.testing.octo.probability = 0; + + // Deep Linking + config.disableDeepLinking = false; diff --git a/Jitsi/deployment.yaml b/Jitsi/deployment.yaml index 731a136..24a238e 100644 --- a/Jitsi/deployment.yaml +++ b/Jitsi/deployment.yaml @@ -93,10 +93,10 @@ - name: web image: jitsi/web:stable-5870 imagePullPolicy: IfNotPresent - #volumeMounts: - # - name: base-patch - # mountPath: /usr/share/jitsi-meet/base.html - # subPath: base.html + volumeMounts: + - name: web-patch + mountPath: /config/config.js + subPath: config.js env: - name: PUBLIC_URL value: "https://meet.jerxie.com" @@ -162,6 +162,9 @@ - name: TZ value: America/Los_Angeles volumes: + - name: web-patch + configMap: + name: web-config - name: jvb-patch configMap: name: jvb-config diff --git a/Jitsi/kustomization.yaml b/Jitsi/kustomization.yaml index 0a2f59a..fd2e602 100644 --- a/Jitsi/kustomization.yaml +++ b/Jitsi/kustomization.yaml @@ -10,3 +10,19 @@ - deployment.yaml - web-service.yaml - ingress.yaml + +images: +- name: jitsi/jicofo + newTag: stable-6173 +- name: jitsi/prosody + newTag: stable-6173 +- name: jitsi/web + newTag: stable-6173 +- name: jitsi/jvb + newTag: stable-6173 + +resources: + - deployment.yaml +patchesStrategicMerge: + - set-resource.yaml + - set-node-selector.yaml diff --git a/Jitsi/configmap.yaml b/Jitsi/configmap.yaml index 844c1cc..638c995 100644 --- a/Jitsi/configmap.yaml +++ b/Jitsi/configmap.yaml @@ -4,10 +4,10 @@ name: jvb-config data: sip-communicator.properties: | - org.jitsi.videobridge.NAT_HARVESTER_LOCAL_ADDRESS=home.jerxie.com + org.jitsi.videobridge.NAT_HARVESTER_LOCAL_ADDRESS=127.0.0.1 org.jitsi.videobridge.NAT_HARVESTER_PUBLIC_ADDRESS=home.jerxie.com org.ice4j.ice.harvest.DISABLE_AWS_HARVESTER=true - org.ice4j.ice.harvest.STUN_MAPPING_HARVESTER_ADDRESSES=meet-jit-si-turnrelay.jitsi.net:443 + #org.ice4j.ice.harvest.STUN_MAPPING_HARVESTER_ADDRESSES=meet-jit-si-turnrelay.jitsi.net:443 org.jitsi.videobridge.ENABLE_STATISTICS=true org.jitsi.videobridge.STATISTICS_TRANSPORT=muc org.jitsi.videobridge.xmpp.user.shard.HOSTNAME=localhost @@ -16,3 +16,1074 @@ org.jitsi.videobridge.xmpp.user.shard.PASSWORD=109ndhgh org.jitsi.videobridge.xmpp.user.shard.MUC_JIDS=JvbBrewery@internal.auth.localhost org.jitsi.videobridge.xmpp.user.shard.MUC_NICKNAME=e35c27c4-db13-4673-acbf-63848217b83f +--- +apiVersion: v1 +kind: ConfigMap +metadata: + name: web-config +data: + config.js: | + /* eslint-disable no-unused-vars, no-var */ + + var config = { + // Connection + // + + hosts: { + // XMPP domain. + domain: 'jitsi-meet.example.com', + + // When using authentication, domain for guest users. + // anonymousdomain: 'guest.example.com', + + // Domain for authenticated users. Defaults to . + // authdomain: 'jitsi-meet.example.com', + + // Focus component domain. Defaults to focus.. + // focus: 'focus.jitsi-meet.example.com', + + // XMPP MUC domain. FIXME: use XEP-0030 to discover it. + muc: 'conference.jitsi-meet.example.com' + }, + + // BOSH URL. FIXME: use XEP-0156 to discover it. + bosh: '//jitsi-meet.example.com/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. This setting has been deprecated. + // desktopSharingFrameRate.max now determines whether simulcast will be enabled + // or disabled for the screenshare. + // capScreenshareBitrate: 1 // 0 to disable - deprecated. + + // 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. + }, + + // Enables reactions feature. + // enableReactions: false, + + // Disables ICE/UDP by filtering out local and remote UDP candidates in + // signalling. + // webrtcIceUdpDisable: false, + + // Disables ICE/TCP by filtering out local and remote TCP candidates in + // signalling. + // webrtcIceTcpDisable: false, + + + // Media + // + + // Enable unified plan implementation support on Chromium based browsers. + // enableUnifiedOnChrome: false, + + // 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 hide the "Show More" link in the GSM popover that can be + // used to display more statistics about the connection (IP, Port, protocol, etc). + // disableShowMoreStats: true, + + // Enabling this will run the lib-jitsi-meet noise detection module which will + // notify the user if there is noise, other than voice, coming from the current + // selected microphone. The purpose it to let the user know that the input could + // be potentially unpleasant for other meeting participants. + enableNoisyMicDetection: true, + + // Start the conference in audio only mode (no video is being received nor + // sent). + // startAudioOnly: false, + + // Every participant after the Nth will start audio muted. + // startAudioMuted: 10, + + // Start calls with audio muted. Unlike the option above, this one is only + // applied locally. FIXME: having these 2 options is confusing. + // startWithAudioMuted: false, + + // Enabling it (with #params) will disable local audio output of remote + // participants and to enable it back a reload is needed. + // startSilent: false + + // Enables support for opus-red (redundancy for Opus). + // enableOpusRed: false, + + // Specify audio quality stereo and opusMaxAverageBitrate values in order to enable HD audio. + // Beware, by doing so, you are disabling echo cancellation, noise suppression and AGC. + // audioQuality: { + // stereo: false, + // opusMaxAverageBitrate: null // Value to fit the 6000 to 510000 range. + // }, + + // 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: '' // 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, + + // If true transcriber will use the application language. + // The application language is either explicitly set by participants in their settings or automatically + // detected based on the environment, e.g. if the app is opened in a chrome instance which is using french as its + // default language then transcriptions for that participant will be in french. + // Defaults to true. + // transcribeWithAppLanguage: true, + + // Transcriber language. This settings will only work if "transcribeWithAppLanguage" is explicitly set to false. + // Available languages can be found in lang/language.json. + // preferredTranscribeLanguage: 'en', + + // 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 for the lastN value to be controlled through the UI. + // When startLastN is present, conference starts with a last-n value of startLastN and channelLastN + // value will be used when the quality level is selected using "Manage Video Quality" slider. + // startLastN: 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 + // }, + + // Provides a way to translate the legacy bridge signaling messages, 'LastNChangedEvent', + // 'SelectedEndpointsChangedEvent' and 'ReceiverVideoConstraint' into the new 'ReceiverVideoConstraints' message + // that invokes the new bandwidth allocation algorithm in the bridge which is described here + // - https://github.com/jitsi/jitsi-videobridge/blob/master/doc/allocation.md. + // useNewBandwidthAllocationStrategy: false, + + // 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 enforce the preferred codec for the conference even when the conference has endpoints + // // that do not support the preferred codec. For example, older versions of Safari do not support VP9 yet. + // // This will result in Safari not being able to decode video from endpoints sending VP9 video. + // // When set to false, the conference falls back to VP8 whenever there is an endpoint that doesn't support the + // // preferred codec and goes back to the preferred codec when that endpoint leaves. + // // enforcePreferredCodec: false, + // + // // 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: { + // H264: { + // low: 200000, + // standard: 500000, + // high: 1500000 + // }, + // VP8 : { + // low: 200000, + // standard: 500000, + // high: 1500000 + // }, + // VP9: { + // low: 100000, + // standard: 300000, + // high: 1200000 + // } + // }, + // + // // 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, + + // Enables forced reload of the client when the call is migrated as a result of + // the bridge going down. + // enableForcedReload: true, + + // 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. + // Setting this to null, will also disable showing the remote videos + // when the toolbar is shown on mouse movements + // disable1On1Mode: null | false | true, + + // Default language for the user interface. + // defaultLanguage: 'en', + + // 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/', + + // App name to be displayed in the invitation email subject, as an alternative to + // interfaceConfig.APP_NAME. + // inviteAppName: null, + + // Moved from interfaceConfig(TOOLBAR_BUTTONS). + // The name of the toolbar buttons to display in the toolbar, including the + // "More actions" menu. If present, the button will display. Exceptions are + // "livestreaming" and "recording" which also require being a moderator and + // some other values in config.js to be enabled. Also, the "profile" button will + // not display for users with a JWT. + // Notes: + // - it's impossible to choose which buttons go in the "More actions" menu + // - it's impossible to control the placement of buttons + // - 'desktop' controls the "Share your screen" button + // - if `toolbarButtons` is undefined, we fallback to enabling all buttons on the UI + // toolbarButtons: [ + // 'camera', + // 'chat', + // 'closedcaptions', + // 'desktop', + // 'download', + // 'embedmeeting', + // 'etherpad', + // 'feedback', + // 'filmstrip', + // 'fullscreen', + // 'hangup', + // 'help', + // 'invite', + // 'livestreaming', + // 'microphone', + // 'mute-everyone', + // 'mute-video-everyone', + // 'participants-pane', + // 'profile', + // 'raisehand', + // 'recording', + // 'security', + // 'select-background', + // 'settings', + // 'shareaudio', + // 'sharedvideo', + // 'shortcuts', + // 'stats', + // 'tileview', + // 'toggle-camera', + // 'videoquality', + // '__end' + // ], + + // 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, + + // Controls the percentage of automatic feedback shown to participants when callstats is enabled. + // The default value is 100%. If set to 0, no automatic feedback will be requested + // feedbackPercentage: 100, + + // 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, + + // Enable unified plan implementation support on Chromium for p2p connection. + // enableUnifiedOnChrome: false, + + // 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, + + // 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' } + ] + }, + + 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: '' + + // 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, + + // Disables the sounds that play when other participants join or leave the + // conference (if set to true, these sounds will not be played). + // disableJoinLeaveSounds: false, + + // Disables the sounds that play when a chat message is received. + // disableIncomingMessageSound: 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 + // the `toolbarButtons`-array 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, + + // A property used to unset the default flip state of the local video. + // When it is set to 'true', the local(self) video will not be mirrored anymore. + // doNotFlipLocalVideo: 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 the 'Grant moderator' button will be disabled. + // disableGrantModerator: 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. + The config file should be in JSON. + None of the fields are mandatory and the response must have the shape: + { + // The domain url to apply (will replace the domain in the sharing conference link/embed section) + inviteDomain: 'example-company.org, + // 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: '', + + // Sets the background transparency level. '0' is fully transparent, '1' is opaque. + // backgroundAlpha: 1, + + // 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 recording label + // hideRecordingLabel: false, + + // Hides the conference timer. + // hideConferenceTimer: true, + + // Hides the participants stats + // hideParticipantsStats: true, + + // Sets the conference subject + // subject: 'Conference Subject', + + // This property is related to the use case when jitsi-meet is used via the IFrame API. When the property is true + // jitsi-meet will use the local storage of the host page instead of its own. This option is useful if the browser + // is not persisting the local storage inside the iframe. + // useHostPageLocalStorage: true, + + // etherpad ("shared document") integration. + // + + // If set, add a "Open shared document" link to the bottom right menu that + // will open an etherpad document. + // etherpad_base: 'https://your-etherpad-installati.on/p/', + + // If etherpad_base is set, and useRoomAsSharedDocumentName is set to true, + // open a pad with the name of the room (lowercased) instead of a pad with a + // random UUID. + // useRoomAsSharedDocumentName: true, + + // List of undocumented settings used in jitsi-meet + /** + _immediateReloadThreshold + debug + debugAudioLevels + deploymentInfo + dialInConfCodeUrl + dialInNumbersUrl + dialOutAuthUrl + dialOutCodesUrl + disableRemoteControl + displayJids + 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 + */ + + /** + * Default interval (milliseconds) for triggering mouseMoved iframe API event + */ + mouseMoveCallbackInterval: 1000, + + /** + 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.sessionRestarted', // show when a client reload is initiated because of bridge migration + // '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 + // ], + + // Prevent the filmstrip from autohiding when screen width is under a certain threshold + // disableFilmstripAutohiding: false, + + // 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 */ + + // Begin default config overrides. + + if (!config.hasOwnProperty('hosts')) config.hosts = {}; + + config.hosts.domain = 'meet.jitsi'; + config.focusUserJid = 'focus@auth.meet.jitsi'; + + config.hosts.muc = 'muc.meet.jitsi'; + config.bosh = '/http-bind'; + config.websocket = 'wss://meet.jerxie.com/xmpp-websocket'; + // Video configuration. + // + + if (!config.hasOwnProperty('constraints')) config.constraints = {}; + if (!config.constraints.hasOwnProperty('video')) config.constraints.video = {}; + + config.resolution = 720; + config.constraints.video.height = { ideal: 720, max: 720, min: 180 }; + config.constraints.video.width = { ideal: 1280, max: 1280, min: 320}; + config.disableSimulcast = true; + config.startVideoMuted = 10; + config.startWithVideoMuted = false; + + // ScreenShare Configuration. + // + config.desktopSharingFrameRate = { min: 5, max: 5 }; + + // Audio configuration. + // + + config.enableNoAudioDetection = false; + config.enableTalkWhileMuted = false; + config.disableAP = false; + config.stereo = false; + config.startAudioOnly = false; + config.startAudioMuted = 10; + config.startWithAudioMuted = false; + config.startSilent = false; + config.disableAudioLevels = false; + config.enableNoisyMicDetection = true; + + + // Peer-to-Peer options. + // + + if (!config.hasOwnProperty('p2p')) config.p2p = {}; + + config.p2p.enabled = true; + + + // Etherpad + // + + // Recording. + // + + // Analytics. + // + + if (!config.hasOwnProperty('analytics')) config.analytics = {}; + + // Enables callstatsUsername to be reported as statsId and used + // by callstats as repoted remote id. + config.enableStatsID = false; + + + // Dial in/out services. + // + + // Calendar service integration. + // + + config.enableCalendarIntegration = false; + + // Invitation service. + // + + // Miscellaneous. + // + + // Prejoin page. + config.prejoinPageEnabled = false; + + // Welcome page. + config.enableWelcomePage = true; + + // Close page. + config.enableClosePage = false; + + // Default language. + // Require users to always specify a display name. + config.requireDisplayName = false; + + // Chrome extension banner. + // Advanced. + // + + // Lipsync hack in jicofo, may not be safe. + config.enableLipSync = false; + + config.enableRemb = true; + config.enableTcc = true; + + // Enable IPv6 support. + config.useIPv6 = true; + + // Transcriptions (subtitles and buttons can be configured in interface_config) + config.transcribingEnabled = false; + + // Deployment information. + // + + if (!config.hasOwnProperty('deploymentInfo')) config.deploymentInfo = {}; + + // Testing + // + + if (!config.hasOwnProperty('testing')) config.testing = {}; + if (!config.testing.hasOwnProperty('octo')) config.testing.octo = {}; + + config.testing.capScreenshareBitrate = 1; + config.testing.octo.probability = 0; + + // Deep Linking + config.disableDeepLinking = false; diff --git a/Jitsi/deployment.yaml b/Jitsi/deployment.yaml index 731a136..24a238e 100644 --- a/Jitsi/deployment.yaml +++ b/Jitsi/deployment.yaml @@ -93,10 +93,10 @@ - name: web image: jitsi/web:stable-5870 imagePullPolicy: IfNotPresent - #volumeMounts: - # - name: base-patch - # mountPath: /usr/share/jitsi-meet/base.html - # subPath: base.html + volumeMounts: + - name: web-patch + mountPath: /config/config.js + subPath: config.js env: - name: PUBLIC_URL value: "https://meet.jerxie.com" @@ -162,6 +162,9 @@ - name: TZ value: America/Los_Angeles volumes: + - name: web-patch + configMap: + name: web-config - name: jvb-patch configMap: name: jvb-config diff --git a/Jitsi/kustomization.yaml b/Jitsi/kustomization.yaml index 0a2f59a..fd2e602 100644 --- a/Jitsi/kustomization.yaml +++ b/Jitsi/kustomization.yaml @@ -10,3 +10,19 @@ - deployment.yaml - web-service.yaml - ingress.yaml + +images: +- name: jitsi/jicofo + newTag: stable-6173 +- name: jitsi/prosody + newTag: stable-6173 +- name: jitsi/web + newTag: stable-6173 +- name: jitsi/jvb + newTag: stable-6173 + +resources: + - deployment.yaml +patchesStrategicMerge: + - set-resource.yaml + - set-node-selector.yaml diff --git a/Jitsi/set-node-selector.yaml b/Jitsi/set-node-selector.yaml new file mode 100644 index 0000000..77305ef --- /dev/null +++ b/Jitsi/set-node-selector.yaml @@ -0,0 +1,9 @@ +apiVersion: apps/v1 +kind: Deployment +metadata: + name: jitsi +spec: + template: + spec: + nodeSelector: + name: k8sworker01 diff --git a/Jitsi/configmap.yaml b/Jitsi/configmap.yaml index 844c1cc..638c995 100644 --- a/Jitsi/configmap.yaml +++ b/Jitsi/configmap.yaml @@ -4,10 +4,10 @@ name: jvb-config data: sip-communicator.properties: | - org.jitsi.videobridge.NAT_HARVESTER_LOCAL_ADDRESS=home.jerxie.com + org.jitsi.videobridge.NAT_HARVESTER_LOCAL_ADDRESS=127.0.0.1 org.jitsi.videobridge.NAT_HARVESTER_PUBLIC_ADDRESS=home.jerxie.com org.ice4j.ice.harvest.DISABLE_AWS_HARVESTER=true - org.ice4j.ice.harvest.STUN_MAPPING_HARVESTER_ADDRESSES=meet-jit-si-turnrelay.jitsi.net:443 + #org.ice4j.ice.harvest.STUN_MAPPING_HARVESTER_ADDRESSES=meet-jit-si-turnrelay.jitsi.net:443 org.jitsi.videobridge.ENABLE_STATISTICS=true org.jitsi.videobridge.STATISTICS_TRANSPORT=muc org.jitsi.videobridge.xmpp.user.shard.HOSTNAME=localhost @@ -16,3 +16,1074 @@ org.jitsi.videobridge.xmpp.user.shard.PASSWORD=109ndhgh org.jitsi.videobridge.xmpp.user.shard.MUC_JIDS=JvbBrewery@internal.auth.localhost org.jitsi.videobridge.xmpp.user.shard.MUC_NICKNAME=e35c27c4-db13-4673-acbf-63848217b83f +--- +apiVersion: v1 +kind: ConfigMap +metadata: + name: web-config +data: + config.js: | + /* eslint-disable no-unused-vars, no-var */ + + var config = { + // Connection + // + + hosts: { + // XMPP domain. + domain: 'jitsi-meet.example.com', + + // When using authentication, domain for guest users. + // anonymousdomain: 'guest.example.com', + + // Domain for authenticated users. Defaults to . + // authdomain: 'jitsi-meet.example.com', + + // Focus component domain. Defaults to focus.. + // focus: 'focus.jitsi-meet.example.com', + + // XMPP MUC domain. FIXME: use XEP-0030 to discover it. + muc: 'conference.jitsi-meet.example.com' + }, + + // BOSH URL. FIXME: use XEP-0156 to discover it. + bosh: '//jitsi-meet.example.com/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. This setting has been deprecated. + // desktopSharingFrameRate.max now determines whether simulcast will be enabled + // or disabled for the screenshare. + // capScreenshareBitrate: 1 // 0 to disable - deprecated. + + // 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. + }, + + // Enables reactions feature. + // enableReactions: false, + + // Disables ICE/UDP by filtering out local and remote UDP candidates in + // signalling. + // webrtcIceUdpDisable: false, + + // Disables ICE/TCP by filtering out local and remote TCP candidates in + // signalling. + // webrtcIceTcpDisable: false, + + + // Media + // + + // Enable unified plan implementation support on Chromium based browsers. + // enableUnifiedOnChrome: false, + + // 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 hide the "Show More" link in the GSM popover that can be + // used to display more statistics about the connection (IP, Port, protocol, etc). + // disableShowMoreStats: true, + + // Enabling this will run the lib-jitsi-meet noise detection module which will + // notify the user if there is noise, other than voice, coming from the current + // selected microphone. The purpose it to let the user know that the input could + // be potentially unpleasant for other meeting participants. + enableNoisyMicDetection: true, + + // Start the conference in audio only mode (no video is being received nor + // sent). + // startAudioOnly: false, + + // Every participant after the Nth will start audio muted. + // startAudioMuted: 10, + + // Start calls with audio muted. Unlike the option above, this one is only + // applied locally. FIXME: having these 2 options is confusing. + // startWithAudioMuted: false, + + // Enabling it (with #params) will disable local audio output of remote + // participants and to enable it back a reload is needed. + // startSilent: false + + // Enables support for opus-red (redundancy for Opus). + // enableOpusRed: false, + + // Specify audio quality stereo and opusMaxAverageBitrate values in order to enable HD audio. + // Beware, by doing so, you are disabling echo cancellation, noise suppression and AGC. + // audioQuality: { + // stereo: false, + // opusMaxAverageBitrate: null // Value to fit the 6000 to 510000 range. + // }, + + // 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: '' // 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, + + // If true transcriber will use the application language. + // The application language is either explicitly set by participants in their settings or automatically + // detected based on the environment, e.g. if the app is opened in a chrome instance which is using french as its + // default language then transcriptions for that participant will be in french. + // Defaults to true. + // transcribeWithAppLanguage: true, + + // Transcriber language. This settings will only work if "transcribeWithAppLanguage" is explicitly set to false. + // Available languages can be found in lang/language.json. + // preferredTranscribeLanguage: 'en', + + // 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 for the lastN value to be controlled through the UI. + // When startLastN is present, conference starts with a last-n value of startLastN and channelLastN + // value will be used when the quality level is selected using "Manage Video Quality" slider. + // startLastN: 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 + // }, + + // Provides a way to translate the legacy bridge signaling messages, 'LastNChangedEvent', + // 'SelectedEndpointsChangedEvent' and 'ReceiverVideoConstraint' into the new 'ReceiverVideoConstraints' message + // that invokes the new bandwidth allocation algorithm in the bridge which is described here + // - https://github.com/jitsi/jitsi-videobridge/blob/master/doc/allocation.md. + // useNewBandwidthAllocationStrategy: false, + + // 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 enforce the preferred codec for the conference even when the conference has endpoints + // // that do not support the preferred codec. For example, older versions of Safari do not support VP9 yet. + // // This will result in Safari not being able to decode video from endpoints sending VP9 video. + // // When set to false, the conference falls back to VP8 whenever there is an endpoint that doesn't support the + // // preferred codec and goes back to the preferred codec when that endpoint leaves. + // // enforcePreferredCodec: false, + // + // // 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: { + // H264: { + // low: 200000, + // standard: 500000, + // high: 1500000 + // }, + // VP8 : { + // low: 200000, + // standard: 500000, + // high: 1500000 + // }, + // VP9: { + // low: 100000, + // standard: 300000, + // high: 1200000 + // } + // }, + // + // // 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, + + // Enables forced reload of the client when the call is migrated as a result of + // the bridge going down. + // enableForcedReload: true, + + // 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. + // Setting this to null, will also disable showing the remote videos + // when the toolbar is shown on mouse movements + // disable1On1Mode: null | false | true, + + // Default language for the user interface. + // defaultLanguage: 'en', + + // 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/', + + // App name to be displayed in the invitation email subject, as an alternative to + // interfaceConfig.APP_NAME. + // inviteAppName: null, + + // Moved from interfaceConfig(TOOLBAR_BUTTONS). + // The name of the toolbar buttons to display in the toolbar, including the + // "More actions" menu. If present, the button will display. Exceptions are + // "livestreaming" and "recording" which also require being a moderator and + // some other values in config.js to be enabled. Also, the "profile" button will + // not display for users with a JWT. + // Notes: + // - it's impossible to choose which buttons go in the "More actions" menu + // - it's impossible to control the placement of buttons + // - 'desktop' controls the "Share your screen" button + // - if `toolbarButtons` is undefined, we fallback to enabling all buttons on the UI + // toolbarButtons: [ + // 'camera', + // 'chat', + // 'closedcaptions', + // 'desktop', + // 'download', + // 'embedmeeting', + // 'etherpad', + // 'feedback', + // 'filmstrip', + // 'fullscreen', + // 'hangup', + // 'help', + // 'invite', + // 'livestreaming', + // 'microphone', + // 'mute-everyone', + // 'mute-video-everyone', + // 'participants-pane', + // 'profile', + // 'raisehand', + // 'recording', + // 'security', + // 'select-background', + // 'settings', + // 'shareaudio', + // 'sharedvideo', + // 'shortcuts', + // 'stats', + // 'tileview', + // 'toggle-camera', + // 'videoquality', + // '__end' + // ], + + // 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, + + // Controls the percentage of automatic feedback shown to participants when callstats is enabled. + // The default value is 100%. If set to 0, no automatic feedback will be requested + // feedbackPercentage: 100, + + // 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, + + // Enable unified plan implementation support on Chromium for p2p connection. + // enableUnifiedOnChrome: false, + + // 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, + + // 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' } + ] + }, + + 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: '' + + // 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, + + // Disables the sounds that play when other participants join or leave the + // conference (if set to true, these sounds will not be played). + // disableJoinLeaveSounds: false, + + // Disables the sounds that play when a chat message is received. + // disableIncomingMessageSound: 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 + // the `toolbarButtons`-array 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, + + // A property used to unset the default flip state of the local video. + // When it is set to 'true', the local(self) video will not be mirrored anymore. + // doNotFlipLocalVideo: 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 the 'Grant moderator' button will be disabled. + // disableGrantModerator: 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. + The config file should be in JSON. + None of the fields are mandatory and the response must have the shape: + { + // The domain url to apply (will replace the domain in the sharing conference link/embed section) + inviteDomain: 'example-company.org, + // 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: '', + + // Sets the background transparency level. '0' is fully transparent, '1' is opaque. + // backgroundAlpha: 1, + + // 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 recording label + // hideRecordingLabel: false, + + // Hides the conference timer. + // hideConferenceTimer: true, + + // Hides the participants stats + // hideParticipantsStats: true, + + // Sets the conference subject + // subject: 'Conference Subject', + + // This property is related to the use case when jitsi-meet is used via the IFrame API. When the property is true + // jitsi-meet will use the local storage of the host page instead of its own. This option is useful if the browser + // is not persisting the local storage inside the iframe. + // useHostPageLocalStorage: true, + + // etherpad ("shared document") integration. + // + + // If set, add a "Open shared document" link to the bottom right menu that + // will open an etherpad document. + // etherpad_base: 'https://your-etherpad-installati.on/p/', + + // If etherpad_base is set, and useRoomAsSharedDocumentName is set to true, + // open a pad with the name of the room (lowercased) instead of a pad with a + // random UUID. + // useRoomAsSharedDocumentName: true, + + // List of undocumented settings used in jitsi-meet + /** + _immediateReloadThreshold + debug + debugAudioLevels + deploymentInfo + dialInConfCodeUrl + dialInNumbersUrl + dialOutAuthUrl + dialOutCodesUrl + disableRemoteControl + displayJids + 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 + */ + + /** + * Default interval (milliseconds) for triggering mouseMoved iframe API event + */ + mouseMoveCallbackInterval: 1000, + + /** + 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.sessionRestarted', // show when a client reload is initiated because of bridge migration + // '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 + // ], + + // Prevent the filmstrip from autohiding when screen width is under a certain threshold + // disableFilmstripAutohiding: false, + + // 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 */ + + // Begin default config overrides. + + if (!config.hasOwnProperty('hosts')) config.hosts = {}; + + config.hosts.domain = 'meet.jitsi'; + config.focusUserJid = 'focus@auth.meet.jitsi'; + + config.hosts.muc = 'muc.meet.jitsi'; + config.bosh = '/http-bind'; + config.websocket = 'wss://meet.jerxie.com/xmpp-websocket'; + // Video configuration. + // + + if (!config.hasOwnProperty('constraints')) config.constraints = {}; + if (!config.constraints.hasOwnProperty('video')) config.constraints.video = {}; + + config.resolution = 720; + config.constraints.video.height = { ideal: 720, max: 720, min: 180 }; + config.constraints.video.width = { ideal: 1280, max: 1280, min: 320}; + config.disableSimulcast = true; + config.startVideoMuted = 10; + config.startWithVideoMuted = false; + + // ScreenShare Configuration. + // + config.desktopSharingFrameRate = { min: 5, max: 5 }; + + // Audio configuration. + // + + config.enableNoAudioDetection = false; + config.enableTalkWhileMuted = false; + config.disableAP = false; + config.stereo = false; + config.startAudioOnly = false; + config.startAudioMuted = 10; + config.startWithAudioMuted = false; + config.startSilent = false; + config.disableAudioLevels = false; + config.enableNoisyMicDetection = true; + + + // Peer-to-Peer options. + // + + if (!config.hasOwnProperty('p2p')) config.p2p = {}; + + config.p2p.enabled = true; + + + // Etherpad + // + + // Recording. + // + + // Analytics. + // + + if (!config.hasOwnProperty('analytics')) config.analytics = {}; + + // Enables callstatsUsername to be reported as statsId and used + // by callstats as repoted remote id. + config.enableStatsID = false; + + + // Dial in/out services. + // + + // Calendar service integration. + // + + config.enableCalendarIntegration = false; + + // Invitation service. + // + + // Miscellaneous. + // + + // Prejoin page. + config.prejoinPageEnabled = false; + + // Welcome page. + config.enableWelcomePage = true; + + // Close page. + config.enableClosePage = false; + + // Default language. + // Require users to always specify a display name. + config.requireDisplayName = false; + + // Chrome extension banner. + // Advanced. + // + + // Lipsync hack in jicofo, may not be safe. + config.enableLipSync = false; + + config.enableRemb = true; + config.enableTcc = true; + + // Enable IPv6 support. + config.useIPv6 = true; + + // Transcriptions (subtitles and buttons can be configured in interface_config) + config.transcribingEnabled = false; + + // Deployment information. + // + + if (!config.hasOwnProperty('deploymentInfo')) config.deploymentInfo = {}; + + // Testing + // + + if (!config.hasOwnProperty('testing')) config.testing = {}; + if (!config.testing.hasOwnProperty('octo')) config.testing.octo = {}; + + config.testing.capScreenshareBitrate = 1; + config.testing.octo.probability = 0; + + // Deep Linking + config.disableDeepLinking = false; diff --git a/Jitsi/deployment.yaml b/Jitsi/deployment.yaml index 731a136..24a238e 100644 --- a/Jitsi/deployment.yaml +++ b/Jitsi/deployment.yaml @@ -93,10 +93,10 @@ - name: web image: jitsi/web:stable-5870 imagePullPolicy: IfNotPresent - #volumeMounts: - # - name: base-patch - # mountPath: /usr/share/jitsi-meet/base.html - # subPath: base.html + volumeMounts: + - name: web-patch + mountPath: /config/config.js + subPath: config.js env: - name: PUBLIC_URL value: "https://meet.jerxie.com" @@ -162,6 +162,9 @@ - name: TZ value: America/Los_Angeles volumes: + - name: web-patch + configMap: + name: web-config - name: jvb-patch configMap: name: jvb-config diff --git a/Jitsi/kustomization.yaml b/Jitsi/kustomization.yaml index 0a2f59a..fd2e602 100644 --- a/Jitsi/kustomization.yaml +++ b/Jitsi/kustomization.yaml @@ -10,3 +10,19 @@ - deployment.yaml - web-service.yaml - ingress.yaml + +images: +- name: jitsi/jicofo + newTag: stable-6173 +- name: jitsi/prosody + newTag: stable-6173 +- name: jitsi/web + newTag: stable-6173 +- name: jitsi/jvb + newTag: stable-6173 + +resources: + - deployment.yaml +patchesStrategicMerge: + - set-resource.yaml + - set-node-selector.yaml diff --git a/Jitsi/set-node-selector.yaml b/Jitsi/set-node-selector.yaml new file mode 100644 index 0000000..77305ef --- /dev/null +++ b/Jitsi/set-node-selector.yaml @@ -0,0 +1,9 @@ +apiVersion: apps/v1 +kind: Deployment +metadata: + name: jitsi +spec: + template: + spec: + nodeSelector: + name: k8sworker01 diff --git a/Jitsi/set-resource.yaml b/Jitsi/set-resource.yaml new file mode 100644 index 0000000..8ceabfd --- /dev/null +++ b/Jitsi/set-resource.yaml @@ -0,0 +1,23 @@ +apiVersion: apps/v1 +kind: Deployment +metadata: + name: jitsi +spec: + template: + spec: + containers: + - name: jicofo + resources: + limits: + memory: 1Gi + #cpu: 1 + - name: prosody + resources: + limits: + memory: 1Gi + #cpu: 1 + - name: web + resources: + limits: + memory: 1Gi + #cpu: 1