Initial import

web/Dockerfile

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+FROM jitsi/base
+
+RUN \
+	apt-dpkg-wrap apt-get update && \
+	apt-dpkg-wrap apt-get install -y nginx-extras jitsi-meet-web && \
+	apt-cleanup && \
+	rm -f /etc/nginx/conf.d/default.conf
+
+COPY rootfs/ /
+
+EXPOSE 80 443
+
+VOLUME /config
+

web/Makefile

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+build:
+	docker build -t jitsi/web .
+
+.PHONY: build

web/rootfs/defaults/config.js

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+/* eslint-disable no-unused-vars, no-var */
+
+var config = {
+    // Configuration
+    //
+
+    // Alternative location for the configuration.
+    // configLocation: './config.json',
+
+    // Custom function which given the URL path should return a room name.
+    // getroomnode: function (path) { return 'someprefixpossiblybasedonpath'; },
+
+
+    // Connection
+    //
+
+    hosts: {
+        // XMPP domain.
+        domain: 'jitsi-meet.example.com',
+
+        // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
+        muc: 'conference.jitsi-meet.example.com'
+
+        // When using authentication, domain for guest users.
+        // anonymousdomain: 'guest.example.com',
+
+        // Domain for authenticated users. Defaults to <domain>.
+        // authdomain: 'jitsi-meet.example.com',
+
+        // Jirecon recording component domain.
+        // jirecon: 'jirecon.jitsi-meet.example.com',
+
+        // Call control component (Jigasi).
+        // call_control: 'callcontrol.jitsi-meet.example.com',
+
+        // Focus component domain. Defaults to focus.<domain>.
+        // focus: 'focus.jitsi-meet.example.com',
+    },
+
+    // BOSH URL. FIXME: use XEP-0156 to discover it.
+    bosh: '/http-bind',
+
+    // 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
+    // focusUserJid: 'focus@auth.jitsi-meet.example.com',
+
+
+    // Testing / experimental features.
+    //
+
+    testing: {
+        // Enables experimental simulcast support on Firefox.
+        enableFirefoxSimulcast: false,
+
+        // P2P test mode disables automatic switching to P2P when there are 2
+        // participants in the conference.
+        p2pTestMode: 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
+    //
+
+    // Audio
+
+    // Disable measuring of audio levels.
+    // disableAudioLevels: false,
+
+    // Start the conference in audio only mode (no video is being received nor
+    // sent).
+    // startAudioOnly: false,
+
+    // Every participant after the Nth will start audio muted.
+    // startAudioMuted: 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,
+
+    // Video
+
+    // Sets the preferred resolution (height) for local video. Defaults to 720.
+    // resolution: 720,
+
+    // 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 independency from
+    // this config's resolution value. Defaults to requesting an ideal aspect
+    // ratio of 16:9 with an ideal resolution of 1080p.
+    // constraints: {
+    //     video: {
+    //         aspectRatio: 16 / 9,
+    //         height: {
+    //             ideal: 1080,
+    //             max: 1080,
+    //             min: 240
+    //         }
+    //     }
+    // },
+
+    // Enable / disable simulcast support.
+    // disableSimulcast: false,
+
+    // Suspend sending video if bandwidth estimation is too low. This may cause
+    // problems with audio playback. Disabled until these are fixed.
+    disableSuspendVideo: true,
+
+    // 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.
+    // preferH264: true,
+
+    // If set to true, disable H.264 video codec by stripping it out of the
+    // SDP.
+    // disableH264: false,
+
+    // Desktop sharing
+
+    // Enable / disable desktop sharing
+    // disableDesktopSharing: false,
+
+    // The ID of the jidesha extension for Chrome.
+    desktopSharingChromeExtId: null,
+
+    // Whether desktop sharing should be disabled on Chrome.
+    desktopSharingChromeDisabled: true,
+
+    // The media sources to use when using screen sharing with the Chrome
+    // extension.
+    desktopSharingChromeSources: [ 'screen', 'window', 'tab' ],
+
+    // Required version of Chrome extension
+    desktopSharingChromeMinExtVersion: '0.1',
+
+    // The ID of the jidesha extension for Firefox. If null, we assume that no
+    // extension is required.
+    desktopSharingFirefoxExtId: null,
+
+    // Whether desktop sharing should be disabled on Firefox.
+    desktopSharingFirefoxDisabled: false,
+
+    // The maximum version of Firefox which requires a jidesha extension.
+    // Example: if set to 41, we will require the extension for Firefox versions
+    // up to and including 41. On Firefox 42 and higher, we will run without the
+    // extension.
+    // If set to -1, an extension will be required for all versions of Firefox.
+    desktopSharingFirefoxMaxVersionExtRequired: 51,
+
+    // The URL to the Firefox extension for desktop sharing.
+    desktopSharingFirefoxExtensionURL: null,
+
+    // 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 recording or not.
+    // enableRecording: false,
+
+    // Type for recording: one of jibri or jirecon.
+    // recordingType: 'jibri',
+
+    // Misc
+
+    // Default value for the channel "last N" attribute. -1 for unlimited.
+    channelLastN: -1,
+
+    // Disables or enables RTX (RFC 4588) (defaults to false).
+    // disableRtx: false,
+
+    // Use XEP-0215 to fetch STUN and TURN servers.
+    // useStunTurn: true,
+
+    // Enable IPv6 support.
+    // useIPv6: true,
+
+    // Enables / disables a data communication channel with the Videobridge.
+    // Values can be 'datachannel', 'websocket', true (treat it as
+    // 'datachannel'), undefined (treat it as 'datachannel') and false (don't
+    // open any channel).
+    // openBridgeChannel: true,
+
+
+    // UI
+    //
+
+    // Use display name as XMPP nickname.
+    // useNicks: 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,
+
+    // Enabling the close page will ignore the welcome page redirection when
+    // a call is hangup.
+    // enableClosePage: false,
+
+    // Disable hiding of remote thumbnails when in a 1-on-1 conference call.
+    // disable1On1Mode: false,
+
+    // The minimum value a video's height (or width, whichever is smaller) needs
+    // to be in order to be considered high-definition.
+    minHDHeight: 540,
+
+    // Default language for the user interface.
+    // defaultLanguage: 'en',
+
+    // If true all users without a token will be considered guests and all users
+    // with token will be considered non-guests. Only guests will be allowed to
+    // edit their profile.
+    enableUserRolesBasedOnToken: false,
+
+    // Message to show the users. Example: 'The service will be down for
+    // maintenance at 01:00 AM GMT,
+    // noticeMessage: '',
+
+
+    // 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,
+
+    // To enable sending statistics to callstats.io you must provide the
+    // Application ID and Secret.
+    // callStatsID: '',
+    // callStatsSecret: '',
+
+    // enables callstatsUsername to be reported as statsId and used
+    // by callstats as repoted remote id
+    // enableStatsID: false
+
+    // enables sending participants display name to callstats
+    // enableDisplayNameInStats: false
+
+
+    // Privacy
+    //
+
+    // If third party requests are disabled, no other server will be contacted.
+    // This means avatars will be locally generated and callstats integration
+    // will not function.
+    // disableThirdPartyRequests: false,
+
+
+    // Peer-To-Peer mode: used (if enabled) when there are just 2 participants.
+    //
+
+    p2p: {
+        // Enables peer to peer mode. When enabled the system will try to
+        // establish a direct connection when there are exactly 2 participants
+        // in the room. If that succeeds the conference will stop sending data
+        // through the JVB and use the peer to peer connection instead. When a
+        // 3rd participant joins the conference will be moved back to the JVB
+        // connection.
+        enabled: true,
+
+        // Use XEP-0215 to fetch STUN and TURN servers.
+        // useStunTurn: true,
+
+        // The STUN servers that will be used in the peer to peer connections
+        stunServers: [
+            { urls: 'stun:stun.l.google.com:19302' },
+            { urls: 'stun:stun1.l.google.com:19302' },
+            { urls: 'stun:stun2.l.google.com:19302' }
+        ],
+
+        // 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).
+        preferH264: true
+
+        // If set to true, disable H.264 video codec by stripping it out of the
+        // SDP.
+        // disableH264: false,
+
+        // 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
+    },
+
+    // A list of scripts to load as lib-jitsi-meet "analytics handlers".
+    // analyticsScriptUrls: [
+    //      "libs/analytics-ga.js", // google-analytics
+    //      "https://example.com/my-custom-analytics.js"
+    // ],
+
+    // The Google Analytics Tracking ID
+    // googleAnalyticsTrackingId = 'your-tracking-id-here-UA-123456-1',
+
+    // 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"
+    }
+
+
+    // List of undocumented settings used in jitsi-meet
+    /**
+     alwaysVisibleToolbar
+     autoEnableDesktopSharing
+     autoRecord
+     autoRecordToken
+     debug
+     debugAudioLevels
+     deploymentInfo
+     dialInConfCodeUrl
+     dialInNumbersUrl
+     dialOutAuthUrl
+     dialOutCodesUrl
+     disableRemoteControl
+     displayJids
+     enableLocalVideoFlip
+     etherpad_base
+     externalConnectUrl
+     firefox_fake_device
+     iAmRecorder
+     iAmSipGateway
+     peopleSearchQueryTypes
+     peopleSearchUrl
+     requireDisplayName
+     tokenAuthUrl
+     */
+
+    // List of undocumented settings used in lib-jitsi-meet
+    /**
+     _peerConnStatusOutOfLastNTimeout
+     _peerConnStatusRtcMuteTimeout
+     abTesting
+     avgRtpStatsN
+     callStatsConfIDNamespace
+     callStatsCustomScriptUrl
+     desktopSharingSources
+     disableAEC
+     disableAGC
+     disableAP
+     disableHPF
+     disableNS
+     enableLipSync
+     enableTalkWhileMuted
+     forceJVB121Ratio
+     hiddenDomain
+     ignoreStartMuted
+     nick
+     startBitrate
+     */
+};
+
+/* eslint-enable no-unused-vars, no-var */

web/rootfs/defaults/default

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+server {
+	listen 80 default_server;
+	listen 443 ssl;
+
+	ssl_certificate /config/keys/cert.crt;
+	ssl_certificate_key /config/keys/cert.key;
+
+	server_name _;
+
+	client_max_body_size 0;
+
+	root /usr/share/jitsi-meet;
+	index index.html
+	error_page 404 /static/404.html;
+
+	location ~ ^/([a-zA-Z0-9=\?]+)$ {
+		rewrite ^/(.*)$ / break;
+	}
+
+   	location /config.js {
+        	alias /config/config.js;
+	}
+
+	location / {
+		ssi on;
+	}
+
+	# BOSH
+	location /http-bind {
+		proxy_pass ${XMPP_BOSH_URL_BASE}/http-bind;
+		proxy_set_header X-Forwarded-For $remote_addr;
+		proxy_set_header Host ${XMPP_DOMAIN};
+	}
+}

web/rootfs/defaults/nginx.conf

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+user www-data;
+worker_processes 4;
+pid /run/nginx.pid;
+include /etc/nginx/modules-enabled/*.conf;
+
+events {
+	worker_connections 768;
+	# multi_accept on;
+}
+
+http {
+
+	##
+	# Basic Settings
+	##
+
+	sendfile on;
+	tcp_nopush on;
+	tcp_nodelay on;
+	keepalive_timeout 65;
+	types_hash_max_size 2048;
+	# server_tokens off;
+
+	# server_names_hash_bucket_size 64;
+	# server_name_in_redirect off;
+
+	client_max_body_size 0;
+
+	include /etc/nginx/mime.types;
+	default_type application/octet-stream;
+
+	##
+	# Logging Settings
+	##
+
+	access_log /dev/stdout;
+	error_log /dev/stderr;
+
+	##
+	# Gzip Settings
+	##
+
+	gzip on;
+	gzip_disable "msie6";
+
+	# gzip_vary on;
+	# gzip_proxied any;
+	# gzip_comp_level 6;
+	# gzip_buffers 16 8k;
+	# gzip_http_version 1.1;
+	# gzip_types text/plain text/css application/json application/x-javascript text/xml application/xml application/xml+rss text/javascript;
+
+	##
+	# Virtual Host Configs
+	##
+        include /config/nginx/site-confs/*;
+}
+
+
+daemon off;

web/rootfs/etc/cont-init.d/10-config

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+#!/usr/bin/with-contenv bash
+
+# make our folders
+mkdir -p \
+	/config/{nginx/site-confs,keys} \
+	/run \
+	/var/lib/nginx/tmp/client_body \
+	/var/tmp/nginx
+
+# copy config files
+if [[ ! -f /config/nginx/nginx.conf ]]; then
+	cp /defaults/nginx.conf /config/nginx/nginx.conf
+fi
+
+if [[ ! -f /config/nginx/site-confs/default ]]; then
+	cp /defaults/default /config/nginx/site-confs/default
+	sed -i \
+	    -e "s,\${XMPP_DOMAIN},$XMPP_DOMAIN,g" \
+	    -e "s,\${XMPP_BOSH_URL_BASE},$XMPP_BOSH_URL_BASE,g" \
+	    /config/nginx/site-confs/default
+fi
+
+if [[ ! -f /config/config.js ]]; then
+	cp /defaults/config.js /config/config.js
+	sed -i "s/jitsi-meet.example.com/$XMPP_DOMAIN/g" /config/config.js
+fi

web/rootfs/etc/cont-init.d/20-keygen

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+#!/usr/bin/with-contenv bash
+
+if [[ -f /config/keys/cert.key && -f /config/keys/cert.crt ]]; then
+	echo "using keys found in /config/keys"
+else
+	echo "generating self-signed keys in /config/keys, you can replace these with your own keys if required"
+	SUBJECT="/C=US/ST=TX/L=Austin/O=jitsi.org/OU=Jitsi Server/CN=*"
+	openssl req -new -x509 -days 3650 -nodes -out /config/keys/cert.crt -keyout /config/keys/cert.key -subj "$SUBJECT"
+fi
+

web/rootfs/etc/services.d/nginx/run

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+#!/usr/bin/with-contenv bash
+nginx -c /config/nginx/nginx.conf
+