nginx warn: using uninitialized "subdomain" variable while sending response to client

solracsf commented 4 years ago

Nginx error log has many lines like these:

[warn] 32237#32237: *122 using uninitialized "subdomain" variable while sending response to client, client: 1.2.3.4, server: jitsi.example.com, request: "GET / HTTP/1.1", subrequest: "/config.js", host: "jitsi.example.com"

Seems like the $subdomain variable is called but not initialized in the vhost template.

saghul commented 4 years ago

Can you share your nginx config?

solracsf commented 4 years ago

Sure.

nginx.conf

worker_processes auto;
worker_cpu_affinity auto;
worker_priority -1;

pid /var/run/nginx.pid;

events {
    use epoll;
    worker_connections 2048;
}

http {
    server_names_hash_bucket_size 64;

    default_type application/octet-stream;

    sendfile on;
    send_timeout 3600;
    client_max_body_size 10240M;

    tcp_nopush on;
    tcp_nodelay on;

    keepalive_timeout 300s;
    keepalive_requests 200;
    reset_timedout_connection on;

    server_tokens off;

    # SSL
    ssl_certificate /etc/ssl/cert.pem;
    ssl_certificate_key /etc/ssl/cert.key;
    ssl_dhparam /etc/ssl/dhparam.pem;
    ssl_session_timeout 1d;
    ssl_session_cache shared:SSL:10m;
    ssl_session_tickets off;
    ssl_protocols TLSv1.2 TLSv1.3;
    ssl_ciphers EECDH+CHACHA20:EECDH+AES;
    ssl_ecdh_curve X25519;

    resolver 127.0.0.53;

    # Logging
    log_format  main_ext  '$remote_addr - $cookie_nc_username [$time_local] "$request" '
                          '$status $body_bytes_sent "$http_referer" '
                          '"$http_user_agent" "$http_x_forwarded_for" '
                          '"$host" sn="$server_name" '
                          'rt=$request_time ';

    access_log /var/log/nginx/access.log main_ext;
    error_log /var/log/nginx/error.log warn;

    # vHosts
    include /etc/nginx/vhosts/*.vhost;
}

jitsi.vhost

server {
    listen 80 default_server;
    listen [::]:80 default_server;
    server_name jitsi.example.com;

    return 301 https://$host$request_uri;
}

server {
    listen 443 default_server ssl http2;
    listen [::]:443 default_server ssl http2;
    server_name jitsi.example.com;

    root /usr/share/jitsi-meet;

    include includes/headers.conf;

    # ssi on with javascript for multidomain variables in config.js
    ssi on;
    ssi_types application/x-javascript application/javascript;

    index index.html index.htm;
    error_page 404 /static/404.html;

    location = /config.js {
        alias /etc/jitsi/meet/jitsi.example.com-config.js;
    }

    location = /external_api.js {
        alias /usr/share/jitsi-meet/libs/external_api.min.js;
    }

    #ensure all static content can always be found first
    location ~ ^/(libs|css|static|images|fonts|lang|sounds|connection_optimization|.well-known)/(.*)$ {
        add_header 'Access-Control-Allow-Origin' '*';
        alias /usr/share/jitsi-meet/$1/$2;
    }

    # BOSH
    location = /http-bind {
        proxy_pass        http://localhost:5280/http-bind;
        proxy_set_header  X-Forwarded-For $remote_addr;
        proxy_set_header  Host $http_host;
    }

    # xmpp websockets
    location = /xmpp-websocket {
        proxy_pass           http://127.0.0.1:5280/xmpp-websocket?prefix=$prefix&$args;
        proxy_http_version  1.1;
        proxy_set_header    Upgrade $http_upgrade;
        proxy_set_header    Connection "upgrade";
        proxy_set_header    Host $http_host;
        tcp_nodelay on;
    }

    location ~ ^/([^/?&:'"]+)$ {
        try_files $uri @root_path;
    }

    location @root_path {
        rewrite ^/(.*)$ / break;
    }

    location ~ ^/([^/?&:'"]+)/config.js$ {
       set $subdomain "$1.";
       set $subdir "$1/";

       alias /etc/jitsi/meet/jitsi.example.com-config.js;
    }

    #Anything that didn't match above, and isn't a real file, assume it's a room name and redirect to /
    location ~ ^/([^/?&:'"]+)/(.*)$ {
        set $subdomain "$1.";
        set $subdir "$1/";
        rewrite ^/([^/?&:'"]+)/(.*)$ /$2;
    }

    # BOSH for subdomains
    location ~ ^/([^/?&:'"]+)/http-bind {
        set $subdomain "$1.";
        set $subdir "$1/";
        set $prefix "$1";

        rewrite ^/(.*)$ /http-bind;
    }

    # websockets for subdomains
    location ~ ^/([^/?&:'"]+)/xmpp-websocket {
        set $subdomain "$1.";
        set $subdir "$1/";
        set $prefix "$1";

        rewrite ^/(.*)$ /xmpp-websocket;
    }
}

saghul commented 4 years ago

And your config.js please.

solracsf commented 4 years ago

@saghul here it is:

/* eslint-disable no-unused-vars, no-var */

var config = {
    // Connection
    //

    hosts: {
        // XMPP domain.
        domain: 'jitsi.example.com',

        // When using authentication, domain for guest users.
        // anonymousdomain: 'guest.jitsi.example.com',

        // Domain for authenticated users. Defaults to <domain>.
        // authdomain: 'jitsi.example.com',

        // Jirecon recording component domain.
        // jirecon: 'jirecon.jitsi.example.com',

        // Call control component (Jigasi).
        // call_control: 'callcontrol.jitsi.example.com',

        // Focus component domain. Defaults to focus.<domain>.
        // focus: 'focus.jitsi.example.com',

        // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
        muc: 'conference.<!--# echo var="subdomain" default="" -->jitsi.example.com'
    },

    // BOSH URL. FIXME: use XEP-0156 to discover it.
    bosh: '//jitsi.example.com/http-bind',

    // Websocket URL
    // websocket: 'wss://jitsi.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
    // focusUserJid: 'focus@auth.jitsi.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

        // 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
    },

    // Disables ICE/UDP by filtering out local and remote UDP candidates in
    // signalling.
    // webrtcIceUdpDisable: false,

    // Disables ICE/TCP by filtering out local and remote TCP candidates in
    // signalling.
    // webrtcIceTcpDisable: false,

    // Media
    //

    // Audio

    // Disable measuring of audio levels.
    // disableAudioLevels: false,
    // audioLevelsInterval: 200,

    // Enabling this will run the lib-jitsi-meet no audio detection module which
    // will notify the user if the current selected microphone has no audio
    // input and will suggest another valid device if one is present.
    enableNoAudioDetection: true,

    // Enabling this will 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

    // 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 independent from
    // this config's resolution value. Defaults to requesting an ideal aspect
    // ratio of 16:9 with an ideal resolution of 720.
    // constraints: {
    //     video: {
    //         aspectRatio: 16 / 9,
    //         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.
    // preferH264: true,

    // If set to true, disable H.264 video codec by stripping it out of the
    // SDP.
    // disableH264: false,

    // Desktop sharing

    // The ID of the jidesha extension for Chrome.
    desktopSharingChromeExtId: null,

    // Whether desktop sharing should be disabled on Chrome.
    // desktopSharingChromeDisabled: false,

    // 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',

    // Whether desktop sharing should be disabled on Firefox.
    // desktopSharingFirefoxDisabled: false,

    // Optional desktop sharing frame rate options. Default value: min:5, max:5.
    // desktopSharingFrameRate: {
    //     min: 5,
    //     max: 5
    // },

    // Try to start calls with screen-sharing instead of camera video.
    // startScreenSharing: false,

    // Recording

    // Whether to enable file recording or not.
    // fileRecordingsEnabled: false,
    // Enable the dropbox integration.
    // dropbox: {
    //     appKey: '<APP_KEY>' // Specify your app key here.
    //     // A URL to redirect the user to, after authenticating
    //     // by default uses:
    //     // 'https://jitsi.example.com/static/oauth.html'
    //     redirectURI:
    //          'https://jitsi.example.com/subfolder/static/oauth.html'
    // },
    // When integrations like dropbox are enabled only that will be shown,
    // by enabling fileRecordingsServiceEnabled, we show both the integrations
    // and the generic recording service (its configuration and storage type
    // depends on jibri configuration)
    // fileRecordingsServiceEnabled: false,
    // Whether to show the possibility to share file recording with other people
    // (e.g. meeting participants), based on the actual implementation
    // on the backend.
    // fileRecordingsServiceSharingEnabled: false,

    // Whether to enable live streaming or not.
    // liveStreamingEnabled: false,

    // Transcription (in interface_config,
    // subtitles and buttons can be configured)
    // transcribingEnabled: false,

    // Enables automatic turning on captions when recording is started
    // autoCaptionOnRecord: false,

    // Misc

    // Default value for the channel "last N" attribute. -1 for unlimited.
    channelLastN: -1,

    // Disables or enables RTX (RFC 4588) (defaults to false).
    // disableRtx: false,

    // Disables or enables TCC (the default is in Jicofo and set to true)
    // (draft-holmer-rmcat-transport-wide-cc-extensions-01). This setting
    // affects congestion control, it practically enables send-side bandwidth
    // estimations.
    // enableTcc: true,

    // Disables or enables REMB (the default is in Jicofo and set to false)
    // (draft-alvestrand-rmcat-remb-03). This setting affects congestion
    // control, it practically enables recv-side bandwidth estimations. When
    // both TCC and REMB are enabled, TCC takes precedence. When both are
    // disabled, then bandwidth estimations are disabled.
    // enableRemb: false,

    // Defines the minimum number of participants to start a call (the default
    // is set in Jicofo and set to 2).
    // minParticipants: 2,

    // 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,

    // Default language for the user interface.
    defaultLanguage: 'fr',

    // 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,

    // Whether or not some features are checked based on token.
    // enableFeaturesBasedOnToken: false,

    // Enable lock room for all moderators, even when userRolesBasedOnToken is enabled and participants are guests.
    // lockRoomGuestEnabled: 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,

    // 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 name to callstats
    // enableDisplayNameInStats: false,

    // enables sending participants email if available to callstats and other analytics
    // enableEmailInStats: false,

    // Privacy
    //

    // If third party requests are disabled, no other server will be contacted.
    // This means avatars will be locally generated and callstats integration
    // will not function.
    disableThirdPartyRequests: true,

    // 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:jitsi.example.com:443' },
            { urls: 'stun:global.stun.twilio.com:3478' }
        ],

        // 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
    },

    analytics: {
        // The Google Analytics Tracking ID:
        // googleAnalyticsTrackingId: 'your-tracking-id-UA-123456-1'

        // The Amplitude APP Key:
        // amplitudeAPPKey: '<APP_KEY>'

        // 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"
        // ],
    },

    // 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"
    },

    // Information for the chrome extension banner
    // chromeExtensionBanner: {
    //     // The chrome extension to be installed address
    //     url: 'https://chrome.google.com/webstore/detail/jitsi-meetings/kglhbbefdnlheedjiejgomgmfplipfeb',

    //     // Extensions info which allows checking if they are installed or not
    //     chromeExtensionsInfo: [
    //         {
    //             id: 'kglhbbefdnlheedjiejgomgmfplipfeb',
    //             path: 'jitsi-logo-48x48.png'
    //         }
    //     ]
    // },

    // Local Recording
    //

    // localRecording: {
    // Enables local recording.
    // Additionally, 'localrecording' (all lowercase) needs to be added to
    // TOOLBAR_BUTTONS in interface_config.js for the Local Recording
    // button to show up on the toolbar.
    //
    //     enabled: true,
    //

    // The recording format, can be one of 'ogg', 'flac' or 'wav'.
    //     format: 'flac'
    //

    // },

    // Options related to end-to-end (participant to participant) ping.
    // e2eping: {
    //   // The interval in milliseconds at which pings will be sent.
    //   // Defaults to 10000, set to <= 0 to disable.
    //   pingInterval: 10000,
    //
    //   // The interval in milliseconds at which analytics events
    //   // with the measured RTT will be sent. Defaults to 60000, set
    //   // to <= 0 to disable.
    //   analyticsInterval: 60000,
    //   },

    // If set, will attempt to use the provided video input device label when
    // triggering a screenshare, instead of proceeding through the normal flow
    // for obtaining a desktop stream.
    // NOTE: This option is experimental and is currently intended for internal
    // use only.
    // _desktopSharingSourceDevice: 'sample-id-or-label',

    // If true, any checks to handoff to another application will be prevented
    // and instead the app will continue to display in the current browser.
    // disableDeepLinking: false,

    // A property to disable the right click context menu for localVideo
    // the menu has option to flip the locally seen video for local presentations
    // disableLocalVideoFlip: false,

    // 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'
    // },

    // List of undocumented settings used in jitsi-meet
    /**
     _immediateReloadThreshold
     autoRecord
     autoRecordToken
     debug
     debugAudioLevels
     deploymentInfo
     dialInConfCodeUrl
     dialInNumbersUrl
     dialOutAuthUrl
     dialOutCodesUrl
     disableRemoteControl
     displayJids
     etherpad_base
     externalConnectUrl
     firefox_fake_device
     googleApiApplicationClientID
     iAmRecorder
     iAmSipGateway
     microsoftApiApplicationClientID
     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
     */

    // 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 */

saghul commented 4 years ago

    // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
    muc: 'conference.<!--# echo var="subdomain" default="" -->jitsi.example.com'

I don't get it, it's used there...

Can you try removing this part?

location = /config.js {
    alias /etc/jitsi/meet/jitsi.example.com-config.js;
}

solracsf commented 4 years ago

nginx:

subrequests cycle while processing "/config.js" while sending response to client, client: 2001:19f0:5:5c48:5400:1ff:fe89:f8b6, server: jitsi.example.com, request: "GET / HTTP/1.1", subrequest: "/index.html", host: "jitsi.example.com"

saghul commented 4 years ago

Nah, I think you need what I told you to remove. I know I have seen this problem, but I an't remember... @damencho any ideas?

damencho commented 4 years ago

For some reason this nginx is not doing ssi inside js files.

damencho commented 4 years ago

You can rewrite it to: muc: 'conference.jitsi.example.com' and just the subdomains will not work.

solracsf commented 4 years ago

Thanks for pointing me in the right direction. Maybe i'm wrong but shouldn't the $subdomain be set to an empty string here?

location = /config.js {
    set $subdomain "";
    alias /etc/jitsi/meet/jitsi.example.com-config.js;
}

because, if a subdomain is really used, it will be catch by the block bellow:

location ~ ^/([^/?&:'"]+)/config.js$ {
    set $subdomain "$1.";
    set $subdir "$1/";
    alias /etc/jitsi/meet/jitsi.example.com-config.js;
}

?

damencho commented 4 years ago

Well, it already has a default value:

solracsf commented 4 years ago

I know, but at least in my case, the above works, not the default as actually.

nginx 1.17.9

damencho commented 4 years ago

I see. Can you open a PR?

saghul commented 2 years ago

This has long ago been fixed.

jitsi / jitsi-meet