Problem to test jitsi install

OS: Ubuntu server 16.04
Web: meet.convoco.online
WebService: Apache

Hello!

I have installed everything following the quick steps guide that they have on github, putting in the DNS section the subdomain to which the videoconferences would have to be made (meet.convoco.online), but when the installation is finished, I open the browser and the jitsi page, but the default plesk page.

Did I do something wrong? Do I need extra configuration?

I have checked the subdomain directory and there is only the default index of the plesk.

Do I have to put something there?

Also attached is the config.js located in etc / jitsi / meet

Summary

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

var config = {
// Connection
//

hosts: {
    // XMPP domain.
    domain: 'meet.convoco.online',

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

    // Domain for authenticated users. Defaults to <domain>.
    // authdomain: 'meet.convoco.online',

    // Jirecon recording component domain.
    // jirecon: 'jirecon.meet.convoco.online',

    // Call control component (Jigasi).
    // call_control: 'callcontrol.meet.convoco.online',

    // Focus component domain. Defaults to focus.<domain>.
    // focus: 'focus.meet.convoco.online',

    // XMPP MUC domain. FIXME: use XEP-0030 to discover it.
    muc: 'conference.meet.convoco.online'
},

// BOSH URL. FIXME: use XEP-0156 to discover it.
bosh: '//meet.convoco.online/http-bind',

// Websocket URL
// websocket: 'wss://meet.convoco.online/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.meet.convoco.online',


// 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://meet.convoco.online/static/oauth.html'
//     redirectURI:
//          'https://meet.convoco.online/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: '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,

// 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: 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:meet.convoco.online:443' },
        { 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
},

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 */

I can’t tell from what you have posted, but if you have apache2 already installed and serving plesk, I guess you’ll need to configure a reverse proxy, to push the traffic back to jitsi.

Hi Neil!

Yes, I have apache 2 and reverse proxy nginx.

There is a guide to follow to configure it?

Thanks!

EDIT:

I see a “Install NGINX” part in doc/manual-install.md, I have to follow this steps to configure it?

If you have apache2 and nginx, I suspect you are causing problems for yourself.

I would pick one, and stick with it for your hosting and reverse proxying.

Typically, it’s easier to reverse proxy using nginx.

Thanks for the quick answer!

I can not “remove” apache2 (or should not) because I have several websites and is based to function properly.

I have reviewed the PLESK documentation and it indicates that both services can coexist without problem, what’s more, they even recommend it so that the server is not overloaded so much, I leave here the link where it indicates it ( https://docs.plesk.com/en-US/onyx/administrator-guide/web-servers/apache-and-nginx-web-servers-linux/apache-with-nginx.70837/ )

Any other ideas of what I should do?

You can run both, as long as they don’t try to bind on the same ports.

The doc you’ve linked suggests using nginx as a reverse proxy, passing traffic on to apache2z

Is there what you are doing? If so, configure an additional reverse proxy in nginx to push the right traffic to jitsi.

But if you’re not sure what you are doing, I’d have thought you’d likely cause yourself more problems in running both apache2 and nginx :slight_smile:

If I’m honest, right now I’m a little lost jajaja.

According to the quick guide, jitsi searches if it is NGINX or APACHE, and it is installed based on the one it finds, or if it doesn’t find any it uses NGINX.

So I understand, since they were both in the system from the beginning, that it will have been configured for NGINX, right?

I just saw the edit that you have made, I think that it would only be necessary to do the reverse proxy that you comment, so should I follow the “install nginx” section in the doc/manual-install.md?

Content of the manual-install.md:

Summary

Add a new file jitsi.example.com in /etc/nginx/sites-available (see also the example config file):

server_names_hash_bucket_size 64;

server {
    listen 0.0.0.0:443 ssl http2;
    listen [::]:443 ssl http2;
    # tls configuration that is not covered in this guide
    # we recommend the use of https://certbot.eff.org/
    server_name jitsi.example.com;
    # set the root
    root /srv/jitsi-meet;
    index index.html;
    location ~ ^/([a-zA-Z0-9=\?]+)$ {
        rewrite ^/(.*)$ / break;
    }
    location / {
        ssi on;
    }
    # BOSH, Bidirectional-streams Over Synchronous HTTP
    # https://en.wikipedia.org/wiki/BOSH_(protocol)
    location /http-bind {
        proxy_pass      http://localhost:5280/http-bind;
        proxy_set_header X-Forwarded-For $remote_addr;
        proxy_set_header Host $http_host;
    }
    # external_api.js must be accessible from the root of the
    # installation for the electron version of Jitsi Meet to work
    # https://github.com/jitsi/jitsi-meet-electron
    location /external_api.js {
        alias /srv/jitsi-meet/libs/external_api.min.js;
    }
}

Add link for the added configuration

cd /etc/nginx/sites-enabled ln -s …/sites-available/jitsi.example.com jitsi.example.com

New info!

I’ve revised /etc/apache2/sites-enabled and has a “meet.convoco.online.config” with this content:

Summary

<VirtualHost *:80>
ServerName meet.convoco.online
Redirect permanent / https://meet.convoco.online/
RewriteEngine On
RewriteCond %{HTTPS} off
RewriteRule ^ https://%{HTTP_HOST}%{REQUEST_URI} [R=301,L]

<VirtualHost *:443>

ServerName meet.convoco.online

SSLProtocol TLSv1 TLSv1.1 TLSv1.2
SSLEngine on
SSLProxyEngine on
SSLCertificateFile /etc/jitsi/meet/meet.convoco.online.crt
SSLCertificateKeyFile /etc/jitsi/meet/meet.convoco.online.key
SSLCipherSuite <>
SSLHonorCipherOrder on
Header set Strict-Transport-Security “max-age=31536000”

DocumentRoot “/usr/share/jitsi-meet”
<Directory “/usr/share/jitsi-meet”>
Options Indexes MultiViews Includes FollowSymLinks
AddOutputFilter Includes html
AllowOverride All
Order allow,deny
Allow from all

ErrorDocument 404 /static/404.html

Alias “/config.js” “/etc/jitsi/meet/meet.convoco.online-config.js”
<Location /config.js>
Require all granted

Alias “/external_api.js” “/usr/share/jitsi-meet/libs/external_api.min.js”
<Location /external_api.js>
Require all granted

ProxyPreserveHost on
ProxyPass /http-bind http://localhost:5280/http-bind/
ProxyPassReverse /http-bind http://localhost:5280/http-bind/

RewriteEngine on
RewriteRule ^/([a-zA-Z0-9]+)$ /index.html

and etc/apache2/sites-available there is another “meet.convoco.online.config” with the same content.