Update Jitsi Meet from 48257 to 5390 (last stable)

Hi,

today I tried to update my jitsi meet installation and my video bridge installation from

to

It works, but the resolution of videos is much less after the update than on my old installation.
Do I need to change something in my old /etc/prosody/conf.d/domain.cfg.lua? Are there any new attributes available? I didn’t change the file after the update at all.

What’s the best way to update from one stable to a newer stable version? I just used apt-get update and apt-get upgrade to update the system.

I changed the /usr/share/jitsi-meet/interface_config.js a little. In my opinion this file has nothing to do with the video resolution. Is that right?

Thanks a lot for your help

Can you share your config.js?
Are there any errors in the console? Something about websocket connection to the bridge?
Do you have this in your nginx config: jitsi-meet/jitsi-meet.example at 2f7ff37472c00e8aeecd072f689c0acb1efdad62 · jitsi/jitsi-meet · GitHub

Hi,
Thanks a lot for the reply. I added the following lines in my nginx config. The video quality is still very bad (320x180).

# colibri (JVB) websockets for jvb1
    location ~ ^/colibri-ws/default-id/(.*) {
        proxy_pass http://127.0.0.1:9090/colibri-ws/default-id/$1$is_args$args;
        proxy_http_version 1.1;
        proxy_set_header Upgrade $http_upgrade;
        proxy_set_header Connection "upgrade";
        tcp_nodelay on;
    }

In each log and config file I replaced our domain with test.domain.at .

Here are the logs

In SYSLOG I see every minute the following lines

Jan 19 08:37:44 ip-172-31-33-26 turnserver: 300: IPv4. tcp or tls connected to: 127.0.0.1:34854
Jan 19 08:37:44 ip-172-31-33-26 turnserver: 300: HTTPS connection has been disabled due Vulnerability in the Web interface !!!
Jan 19 08:37:44 ip-172-31-33-26 turnserver: 300: session 001000000000000006: client socket to be closed in client handler: ss=0x7f2a6000c240
Jan 19 08:37:44 ip-172-31-33-26 turnserver: 300: session 001000000000000006: closed (2nd stage), user <> realm <test.domain.at> origin <>, local 127.0.0.1:5349, remote 127.0.0.1:34854, reason: general
Jan 19 08:37:44 ip-172-31-33-26 turnserver: 300: session 001000000000000006: SSL shutdown received, socket to be closed (local 127.0.0.1:5349, remote 127.0.0.1:34854)

The jicofo log seams to be okay (only info messages).

In prosody.err is see the following error. This could be the same as the one in syslog

Jan 19 08:32:46 portmanager	error	Error binding encrypted port for https: No certificate present in SSL/TLS configuration for https port 5281

The video bridge is installed on an extra machine. Here is the video bridge log:

2021-01-19 08:45:01.033 INFO: [19] VideobridgeExpireThread.expire#140: Running expire()
2021-01-19 08:45:01.059 INFO: [20] HealthChecker.run#170: Performed a successful health check in PT0S. Sticky failure: false
2021-01-19 08:45:11.059 INFO: [20] HealthChecker.run#170: Performed a successful health check in PT0S. Sticky failure: false
2021-01-19 08:45:21.059 INFO: [20] HealthChecker.run#170: Performed a successful health check in PT0S. Sticky failure: false
2021-01-19 08:45:31.059 INFO: [20] HealthChecker.run#170: Performed a successful health check in PT0S. Sticky failure: false
2021-01-19 08:45:41.059 INFO: [20] HealthChecker.run#170: Performed a successful health check in PT0S. Sticky failure: false
2021-01-19 08:45:51.059 INFO: [20] HealthChecker.run#170: Performed a successful health check in PT0S. Sticky failure: false

in the chrome console tab I see the following error (if I am connected with one other user that uses Safari)

Logger.js:154 2021-01-19T10:14:45.837Z [JitsiMeetJS.js] <Object.getGlobalOnErrorHandler>:  UnhandledError: Jingle error: {"reason":"timeout","session":"JingleSessionPC[p2p=true,initiator=false,sid=1e05e8912f9a]"} Script: null Line: null Column: null StackTrace:  Error: Jingle error: {"reason":"timeout","session":"JingleSessionPC[p2p=true,initiator=false,sid=1e05e8912f9a]"}
    at https://test.domain.at/libs/lib-jitsi-meet.min.js?v=4628:1:243678
    at I.TimedHandler.handler (https://test.domain.at/libs/lib-jitsi-meet.min.js?v=4628:1:31360)
    at I.TimedHandler.run (https://test.domain.at/libs/lib-jitsi-meet.min.js?v=4628:1:26901)
    at I.Connection._onIdle (https://test.domain.at/libs/lib-jitsi-meet.min.js?v=4628:1:42933)
    at https://test.domain.at/libs/lib-jitsi-meet.min.js?v=4628:1:43103

Here are my config files:

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

var config = {
    // Connection
    //

    hosts: {
        // XMPP domain.
        domain: 'test.domain.at',

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

        // Domain for authenticated users. Defaults to <domain>.
        // authdomain: 'test.domain.at',

        // Jirecon recording component domain.
        // jirecon: 'jirecon.test.domain.at',

        // Call control component (Jigasi).
        // call_control: 'callcontrol.test.domain.at',

        // Focus component domain. Defaults to focus.<domain>.
        // focus: 'focus.test.domain.at',

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

    // BOSH URL. FIXME: use XEP-0156 to discover it.
    bosh: '//test.domain.at/http-bind',

    // Websocket URL
    // websocket: 'wss://test.domain.at/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.test.domain.at',


    // Testing / experimental features.
    //

    testing: {
        // 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: true,
    // 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: 1960,

    // 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: 1080,
    //             max: 1080,
    //             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: 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

    // 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: 12,
         max: 24
    },

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

    // Recording

    // Whether to enable file recording or not.
    fileRecordingsEnabled: true,
    hiddenDomain: 'recorder.test.domain.at',

	// 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://test.domain.at/static/oauth.html'
    //     redirectURI:
    //          'https://test.domain.at/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: true,

    // 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: 8,

    // 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,

    // 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,

    // 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: 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.
    // 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: true,

    // 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,

    // When 'true', it shows an intermediate page before joining, where the user can  configure its devices.
    // prejoinPageEnabled: 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:test.domain.at:4446' },
            { urls: 'stun:meet-jit-si-turnrelay.jitsi.net:443' }
        ],

        // 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'

        // Matomo configuration:
        // matomoEndpoint: 'https://your-matomo-endpoint/',
        // matomoSiteID: '42',

        // 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"
    },

    // Decides whether the start/stop recording audio notifications should play on record.
    // disableRecordAudioNotification: 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
    // 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,

    // 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 all muting operations of remote participants will be disabled.
    // disableRemoteMute: true,

    // 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 */

In my interface config I removed all unnecessary elements:

/* eslint-disable no-unused-vars, no-var, max-len */
/* eslint sort-keys: ["error", "asc", {"caseSensitive": false}] */

var interfaceConfig = {
    APP_NAME: 'Jitsi Meet',
    AUDIO_LEVEL_PRIMARY_COLOR: 'rgba(255,255,255,0.4)',
    AUDIO_LEVEL_SECONDARY_COLOR: 'rgba(255,255,255,0.2)',

    /**
     * A UX mode where the last screen share participant is automatically
     * pinned. Valid values are the string "remote-only" so remote participants
     * get pinned but not local, otherwise any truthy value for all participants,
     * and any falsy value to disable the feature.
     *
     * Note: this mode is experimental and subject to breakage.
     */
    AUTO_PIN_LATEST_SCREEN_SHARE: 'remote-only',
    BRAND_WATERMARK_LINK: '',

    CLOSE_PAGE_GUEST_HINT: false, // A html text to be shown to guests on the close page, false disables it
    /**
     * Whether the connection indicator icon should hide itself based on
     * connection strength. If true, the connection indicator will remain
     * displayed while the participant has a weak connection and will hide
     * itself after the CONNECTION_INDICATOR_HIDE_TIMEOUT when the connection is
     * strong.
     *
     * @type {boolean}
     */
    CONNECTION_INDICATOR_AUTO_HIDE_ENABLED: true,

    /**
     * How long the connection indicator should remain displayed before hiding.
     * Used in conjunction with CONNECTION_INDICATOR_AUTOHIDE_ENABLED.
     *
     * @type {number}
     */
    CONNECTION_INDICATOR_AUTO_HIDE_TIMEOUT: 5000,

    /**
     * If true, hides the connection indicators completely.
     *
     * @type {boolean}
     */
    CONNECTION_INDICATOR_DISABLED: false,

    DEFAULT_BACKGROUND: '#474747',
    DEFAULT_LOCAL_DISPLAY_NAME: 'me',
    DEFAULT_LOGO_URL: 'images/watermark.svg',
    DEFAULT_REMOTE_DISPLAY_NAME: 'Fellow Jitster',
    DEFAULT_WELCOME_PAGE_LOGO_URL: 'images/watermark.svg',

    DISABLE_DOMINANT_SPEAKER_INDICATOR: true,

    DISABLE_FOCUS_INDICATOR: true,

    /**
     * If true, notifications regarding joining/leaving are no longer displayed.
     */
    DISABLE_JOIN_LEAVE_NOTIFICATIONS: true,

    /**
     * If true, presence status: busy, calling, connected etc. is not displayed.
     */
    DISABLE_PRESENCE_STATUS: true,

    /**
     * Whether the ringing sound in the call/ring overlay is disabled. If
     * {@code undefined}, defaults to {@code false}.
     *
     * @type {boolean}
     */
    DISABLE_RINGING: true,

    /**
     * Whether the speech to text transcription subtitles panel is disabled.
     * If {@code undefined}, defaults to {@code false}.
     *
     * @type {boolean}
     */
    DISABLE_TRANSCRIPTION_SUBTITLES: true,

    /**
     * Whether or not the blurred video background for large video should be
     * displayed on browsers that can support it.
     */
    DISABLE_VIDEO_BACKGROUND: false,

    DISPLAY_WELCOME_FOOTER: true,
    DISPLAY_WELCOME_PAGE_ADDITIONAL_CARD: false,
    DISPLAY_WELCOME_PAGE_CONTENT: false,
    DISPLAY_WELCOME_PAGE_TOOLBAR_ADDITIONAL_CONTENT: false,

    ENABLE_DIAL_OUT: true,

    ENABLE_FEEDBACK_ANIMATION: false, // Enables feedback star animation.

    FILM_STRIP_MAX_HEIGHT: 120,

    GENERATE_ROOMNAMES_ON_WELCOME_PAGE: true,

    /**
     * Hide the logo on the deep linking pages.
     */
    HIDE_DEEP_LINKING_LOGO: false,

    /**
     * Hide the invite prompt in the header when alone in the meeting.
     */
    HIDE_INVITE_MORE_HEADER: false,

    INITIAL_TOOLBAR_TIMEOUT: 20000,
    JITSI_WATERMARK_LINK: 'https://jitsi.org',

    LANG_DETECTION: false, // Allow i18n to detect the system language
    LIVE_STREAMING_HELP_LINK: 'https://jitsi.org/live', // Documentation reference for the live streaming feature.
    LOCAL_THUMBNAIL_RATIO: 16 / 9, // 16:9

    /**
     * Maximum coefficient of the ratio of the large video to the visible area
     * after the large video is scaled to fit the window.
     *
     * @type {number}
     */
    MAXIMUM_ZOOMING_COEFFICIENT: 1.3,

    /**
     * Whether the mobile app Jitsi Meet is to be promoted to participants
     * attempting to join a conference in a mobile Web browser. If
     * {@code undefined}, defaults to {@code true}.
     *
     * @type {boolean}
     */
    MOBILE_APP_PROMO: true,

    /**
     * Specify custom URL for downloading android mobile app.
     */
    MOBILE_DOWNLOAD_LINK_ANDROID: 'https://play.google.com/store/apps/details?id=org.jitsi.meet',

    /**
     * Specify custom URL for downloading f droid app.
     */
    MOBILE_DOWNLOAD_LINK_F_DROID: 'https://f-droid.org/en/packages/org.jitsi.meet/',

    /**
     * Specify URL for downloading ios mobile app.
     */
    MOBILE_DOWNLOAD_LINK_IOS: 'https://itunes.apple.com/us/app/jitsi-meet/id1165103905',

    NATIVE_APP_NAME: 'Jitsi Meet',

    // Names of browsers which should show a warning stating the current browser
    // has a suboptimal experience. Browsers which are not listed as optimal or
    // unsupported are considered suboptimal. Valid values are:
    // chrome, chromium, edge, electron, firefox, nwjs, opera, safari
    OPTIMAL_BROWSERS: [ 'chrome', 'chromium', 'firefox', 'nwjs', 'electron', 'safari' ],

    POLICY_LOGO: null,
    PROVIDER_NAME: 'Jitsi',

    /**
     * If true, will display recent list
     *
     * @type {boolean}
     */
    RECENT_LIST_ENABLED: false,
    REMOTE_THUMBNAIL_RATIO: 1, // 1:1

    SETTINGS_SECTIONS: [ 'devices', 'language', 'moderator',  'calendar' ],
    SHOW_BRAND_WATERMARK: false,

    /**
    * Decides whether the chrome extension banner should be rendered on the landing page and during the meeting.
    * If this is set to false, the banner will not be rendered at all. If set to true, the check for extension(s)
    * being already installed is done before rendering.
    */
    SHOW_CHROME_EXTENSION_BANNER: false,

    SHOW_DEEP_LINKING_IMAGE: false,
    SHOW_JITSI_WATERMARK: true,
    SHOW_POWERED_BY: false,
    SHOW_PROMOTIONAL_CLOSE_PAGE: false,

    /*
     * If indicated some of the error dialogs may point to the support URL for
     * help.
     */
    SUPPORT_URL: 'https://community.jitsi.org/',

    TOOLBAR_ALWAYS_VISIBLE: true,

    /**
     * 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 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
     */
    TOOLBAR_BUTTONS: [
        'microphone', 'camera',  'desktop', 'embedmeeting', 'fullscreen',
        'fodeviceselection', 'hangup',  'chat', 'recording',
        'livestreaming',  'sharedvideo', 'settings', 'raisehand',
        'videoquality', 'filmstrip',   'stats', 'shortcuts',
        'tileview',   'help', 'mute-everyone',  
    ],

    TOOLBAR_TIMEOUT: 4000,

    // Browsers, in addition to those which do not fully support WebRTC, that
    // are not supported and should show the unsupported browser page.
    UNSUPPORTED_BROWSERS: [],

    /**
     * Whether to show thumbnails in filmstrip as a column instead of as a row.
     */
    VERTICAL_FILMSTRIP: true,

    // Determines how the video would fit the screen. 'both' would fit the whole
    // screen, 'height' would fit the original video height to the height of the
    // screen, 'width' would fit the original video width to the width of the
    // screen respecting ratio.
    VIDEO_LAYOUT_FIT: 'both',

    /**
     * If true, hides the video quality label indicating the resolution status
     * of the current large video.
     *
     * @type {boolean}
     */
    VIDEO_QUALITY_LABEL_DISABLED: true,

    /**
     * How many columns the tile view can expand to. The respected range is
     * between 1 and 5.
     */
    // TILE_VIEW_MAX_COLUMNS: 5,

    /**
     * Specify Firebase dynamic link properties for the mobile apps.
     */
    // MOBILE_DYNAMIC_LINK: {
    //    APN: 'org.jitsi.meet',
    //    APP_CODE: 'w2atb',
    //    CUSTOM_DOMAIN: undefined,
    //    IBI: 'com.atlassian.JitsiMeet.ios',
    //    ISI: '1165103905'
    // },

    /**
     * Specify mobile app scheme for opening the app from the mobile browser.
     */
    // APP_SCHEME: 'org.jitsi.meet',

    /**
     * Specify the Android app package name.
     */
    // ANDROID_APP_PACKAGE: 'org.jitsi.meet',

    /**
     * Override the behavior of some notifications to remain displayed until
     * explicitly dismissed through a user action. The value is how long, in
     * milliseconds, those notifications should remain displayed.
     */
    // ENFORCE_NOTIFICATION_AUTO_DISMISS_TIMEOUT: 15000,

    // List of undocumented settings
    /**
     INDICATOR_FONT_SIZES
     PHONE_NUMBER_REGEX
    */

    // Allow all above example options to include a trailing comma and
    // prevent fear when commenting out the last value.
    // eslint-disable-next-line sort-keys
    makeJsonParserHappy: 'even if last key had a trailing comma'

    // No configuration value should follow this line.
};

/* eslint-enable no-unused-vars, no-var, max-len */

If you need any further informations, please let me know.
Thanks a lot for your help.

If you open 3 tabs from chrome is there audio and video?

Hi,

I just tested with 3 chrome browsers. It seams to work.

As you can see the above two videos has a very poor quality.
2021-01-19_13-56-53

Do you have any idea? Should I try to use the default settings of the config files? And change my adaptions afterwards?

This looks strange … Probably you want 1080 … And also you will need to change the constraints just below that …

Yes, please.

I will give it a try.
Can you provide me a link for the default interface and config file? I am not sure which one I should use.

Is the

resolution: 1080

equivalents to constraints.video.hieght.ideal?

constraints: {
video: {
height: {
ideal: 1080,
max: 1080,
min: 180
}
}
},

what is the diffrence?

I have also an issue with this last version, enableLayerSuspension: true doesn’t work and I don’t see any B/W decreasing when it’s true.

Thanks

How are you testing it? That should have an effect when for example all participants are in grid view and all participants will switch to 180p as no one is watching on stage participant.

thanks for the replay, in fact I compared B/W consumption between the two version the old and new one version for 4 participantes, for the new version, I got RX:4Mbps in both case enableLayerSuspension is true or false.

unlike the old version, I got 4Mbps if enableLayerSuspension is false and 2,7Mbps when it’s true

I just tested on meet.jit.si with 3 participants and they switched to 180p in tile view in a not so big window (the tiles were small enough to show just 180p):
image

I’ll do a test and let you know, here my config:

// Sets the preferred resolution (height) for local video. Defaults to 720.
// resolution: 720,
resolution:480,

// 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
//         }
//     }
// },
constraints: {
     video: {
         height: {
             ideal: 360,
             max: 480,
             min: 180
         }
     }
 },
// 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: true,

it seems donesn’t work for my new instance, i didn’t see “send max 180p”:
new fresh instance:

if I switch to preview one participante video the résolution dosent change normally shoud be switched from 180 (min) to ideal 360

unlike my old instance i have this:
old jitis:

@Yassine, Will you be able to post your browser console log with filter “SenderVideoConstraint” so that I can see what resolution is being requested by the bridge in the new instance where you are seeing the issue ?

@damencho
I used the config file from (with my adaptions) jitsi-meet/config.js at master · jitsi/jitsi-meet · GitHub
Now the videos are excellent. It seams to work. In the next couple of days I will make some further tests.

yes, I’ll do it. Big thanks!

voila, the output of the filter:

> 2021-01-19T17:15:55.239Z [modules/xmpp/JingleSessionPC.js] <w.setSenderVideoConstraint>:  JingleSessionPC[p2p=false,initiator=false,sid=91mvs3uqqodcl] setSenderVideoConstraint: 2160
> Logger.js:154 2021-01-19T17:15:55.239Z [modules/RTC/TraceablePeerConnection.js] <A.setSenderVideoConstraint>:  TPC[1,p2p:false] senderVideoMaxHeight: 2160
> Logger.js:154 2021-01-19T17:15:55.240Z [modules/RTC/TraceablePeerConnection.js] <A.setSenderVideoConstraint>:  TPC[1,p2p:false] setting max height of 2160, encodings: [{"active":true,"networkPriority":"low","priority":"low"},{"active":true,"networkPriority":"low","priority":"low"},{"active":true,"networkPriority":"low","priority":"low"}]

I’m seeing also some errors on the console:

BridgeChannel.js:86 WebSocket connection to 'wss://meet.mydomaine.com/colibri-ws/default-id/f0f608a0fafc2002/5044ede6?pwd=4or97t73130qvrg854gaebqp1l' failed: Error during WebSocket handshake: Unexpected response code: 405
    _initWebSocket @ BridgeChannel.js:86
    t @ BridgeChannel.js:105
    Logger.js:154 2021-01-19T17:17:58.072Z [modules/RTC/BridgeChannel.js] <WebSocket.e.onclose>:  Channel closed: 1006 
    o @ Logger.js:154
    e.onclose @ BridgeChannel.js:359
    Logger.js:154 2021-01-19T17:18:04.316Z [modules/RTC/BridgeChannel.js] <l._send>:  Bridge Channel send: no opened channel.
    o @ Logger.js:154
    _send @ BridgeChannel.js:383
    sendMessage @ BridgeChannel.js:190
    sendChannelMessage @ RTC.js:936
    ie.sendEndpointMessage @ JitsiConference.js:2519
    ie.broadcastEndpointMessage @ JitsiConference.js:2529
    _broadcastLocalStats @ ConnectionQuality.js:468
    _updateLocalStats @ ConnectionQuality.js:540
    a.emit @ events.js:157
    m._processAndEmitReport @ RTPStatsCollector.js:868
    m.processStatsReport @ RTPStatsCollector.js:726
    (anonymous) @ RTPStatsCollector.js:378
    Logger.js:154 2021-01-19T17:18:14.323Z [modules/RTC/BridgeChannel.js] <l._send>:  Bridge Channel send: no opened channel.
    o @ Logger.js:154
    _send @ BridgeChannel.js:383
    sendMessage @ BridgeChannel.js:190
    sendChannelMessage @ RTC.js:936
    ie.sendEndpointMessage @ JitsiConference.js:2519
    ie.broadcastEndpointMessage @ JitsiConference.js:2529
    _broadcastLocalStats @ ConnectionQuality.js:468
    _updateLocalStats @ ConnectionQuality.js:540
    a.emit @ events.js:157
    m._processAndEmitReport @ RTPStatsCollector.js:868
    m.processStatsReport @ RTPStatsCollector.js:726
    (anonymous) @ RTPStatsCollector.js:378
    Logger.js:154 2021-01-19T17:18:24.319Z [modules/RTC/BridgeChannel.js] <l._send>:  Bridge Channel send: no opened channel.
    o @ Logger.js:154
    _send @ BridgeChannel.js:383
    sendMessage @ BridgeChannel.js:190
    sendChannelMessage @ RTC.js:936
    ie.sendEndpointMessage @ JitsiConference.js:2519
    ie.broadcastEndpointMessage @ JitsiConference.js:2529
    _broadcastLocalStats @ ConnectionQuality.js:468
    _updateLocalStats @ ConnectionQuality.js:540
    a.emit @ events.js:157
    m._processAndEmitReport @ RTPStatsCollector.js:868
    m.processStatsReport @ RTPStatsCollector.js:726
    (anonymous) @ RTPStatsCollector.js:378
    Logger.js:154 2021-01-19T17:18:34.316Z [modules/RTC/BridgeChannel.js] <l._send>:  Bridge Channel send: no opened channel.
    o @ Logger.js:154
    _send @ BridgeChannel.js:383
    sendMessage @ BridgeChannel.js:190
    sendChannelMessage @ RTC.js:936
    ie.sendEndpointMessage @ JitsiConference.js:2519
    ie.broadcastEndpointMessage @ JitsiConference.js:2529
    _broadcastLocalStats @ ConnectionQuality.js:468
    _updateLocalStats @ ConnectionQuality.js:540
    a.emit @ events.js:157
    m._processAndEmitReport @ RTPStatsCollector.js:868
    m.processStatsReport @ RTPStatsCollector.js:726
    (anonymous) @ RTPStatsCollector.js:378
    Logger.js:154 2021-01-19T17:18:44.326Z [modules/RTC/BridgeChannel.js] <l._send>:  Bridge Channel send: no opened channel.
    o @ Logger.js:154
    _send @ BridgeChannel.js:383
    sendMessage @ BridgeChannel.js:190
    sendChannelMessage @ RTC.js:936
    ie.sendEndpointMessage @ JitsiConference.js:2519
    ie.broadcastEndpointMessage @ JitsiConference.js:2529
    _broadcastLocalStats @ ConnectionQuality.js:468
    _updateLocalStats @ ConnectionQuality.js:540
    a.emit @ events.js:157
    m._processAndEmitReport @ RTPStatsCollector.js:868
    m.processStatsReport @ RTPStatsCollector.js:726
    (anonymous) @ RTPStatsCollector.js:378
    Logger.js:154 2021-01-19T17:18:54.315Z [modules/RTC/BridgeChannel.js] <l._send>:  Bridge Channel send: no opened channel.

@Yassine you need to fix the connection from clients to the bridge, check messages above, probably you are missing nginx config.

here my nginx config, it’s included?

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;
}

colibri (JVB) websockets for jvb1

location ~ ^/colibri-ws/default-id/(.*) {
proxy_pass http://127.0.0.1:9090/colibri-ws/default-id/$1$is_args$args;
proxy_http_version 1.1;
proxy_set_header Upgrade $http_upgrade;
proxy_set_header Connection “upgrade”;
tcp_nodelay on;
}