Turn https version 2.0.4857-1

On version _all.deb jitsi-meet_2.0.4627-1_all.deb, everything works fine without any configuration to do.
It’s the first time I see a version with nothing to do.
Turn server to use jitsi only in https works fine and multiple parcipant from external works fine.

But I want to use the last version jitsi-meet_2.0.4857-1 and with this version Turn server to use only https doesn’t works. If I open port 10000 meeting works but I don’t want to use port 10000 I need only https exactly like the version 2.0.4627-1 do.
Do you know what has changed on the last version vs the 2.0.4627-1?
How to be able to use https only and not port 10000 like version 2.0.4627-1 please?

My config (usr/share/jitsi-meet-web-config):

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

var config = {
// Connection
//

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

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

    // Domain for authenticated users. Defaults to <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',

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

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

// Websocket URL
// websocket: 'wss://jitsi-meet.example.com/xmpp-websocket',

// The name of client node advertised in XEP-0115 'c' stanza
clientNode: 'http://jitsi.org/jitsimeet',

// The real JID of focus participant - can be overridden here
// focusUserJid: 'focus@auth.jitsi-meet.example.com',


// Testing / experimental features.
//

testing: {
    // Disables the End to End Encryption feature. Useful for debugging
    // issues related to insertable streams.
    // disableE2EE: false,

    // P2P test mode disables automatic switching to P2P when there are 2
    // participants in the conference.
    p2pTestMode: false

    // Enables the test specific features consumed by jitsi-meet-torture
    // testMode: false

    // Disables the auto-play behavior of *all* newly created video element.
    // This is useful when the client runs on a host with limited resources.
    // noAutoPlayVideo: false

    // Enable / disable 500 Kbps bitrate cap on desktop tracks. When enabled,
    // simulcast is turned off for the desktop share. If presenter is turned
    // on while screensharing is in progress, the max bitrate is automatically
    // adjusted to 2.5 Mbps. This takes a value between 0 and 1 which determines
    // the probability for this to be enabled.
    // capScreenshareBitrate: 1 // 0 to disable
},

// 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

// Sets the preferred target bitrate for the Opus audio codec by setting its
// 'maxaveragebitrate' parameter. Currently not available in p2p mode.
// Valid values are in the range 6000 to 510000
// opusMaxAvgBitrate: 20000,

// Video

// Sets the preferred resolution (height) for local video. Defaults to 720.
 resolution: 1080,

// 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: 360 
         }
     }
 },

// Enable / disable simulcast support.
 disableSimulcast: true,

// 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

// Optional desktop sharing frame rate options. Default value: min:5, max:5.
 desktopSharingFrameRate: {
     min: 25,
     max: 30 
 },

// 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-meet.example.com/static/oauth.html'
//     redirectURI:
//          'https://jitsi-meet.example.com/subfolder/static/oauth.html'
// },
// When integrations like dropbox are enabled only that will be shown,
// by enabling fileRecordingsServiceEnabled, we show both the integrations
// and the generic recording service (its configuration and storage type
// depends on jibri configuration)
// fileRecordingsServiceEnabled: false,
// Whether to show the possibility to share file recording with other people
// (e.g. meeting participants), based on the actual implementation
// on the backend.
// fileRecordingsServiceSharingEnabled: false,

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

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

// 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,

// // Options for the recording limit notification.
// recordingLimit: {
//
//    // The recording limit in minutes. Note: This number appears in the notification text
//    // but doesn't enforce the actual recording time limit. This should be configured in
//    // jibri!
//    limit: 60,
//
//    // The name of the app with unlimited recordings.
//    appName: 'Unlimited recordings APP',
//
//    // The URL of the app with unlimited recordings.
//    appURL: 'https://unlimited.recordings.app.com/'
// },

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

// Disables or enables TCC (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 the TURN servers discovered via XEP-0215 for the jitsi-videobridge
// connection
 useStunTurn: true,

// Use TURN/UDP servers for the jitsi-videobridge connection (by default
// we filter out TURN/UDP because it is usually not needed since the
// bridge itself is reachable via UDP)
// useTurnUdp: false

// 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
//

// 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,

// When 'true', it shows an intermediate page before joining, where the user can configure their devices.
// prejoinPageEnabled: false,

// If true, shows the unsafe room name warning label when a room name is
// deemed unsafe (due to the simplicity in the name) and a password is not
// set or the lobby is not enabled.
// enableInsecureRoomNameWarning: false,

// Stats
//

// Whether to enable stats collection or not in the TraceablePeerConnection.
// This can be useful for debugging purposes (post-processing/analysis of
// the webrtc stats) as it is done in the jitsi-meet-torture bandwidth
// estimation tests.
// gatherStats: false,

// The interval at which PeerConnection.getStats() is called. Defaults to 10000
// pcStatsInterval: 10000,

// To enable sending statistics to callstats.io you must provide the
// Application ID and Secret.
// callStatsID: '',
// callStatsSecret: '',

// Enables sending participants' display names to callstats
// enableDisplayNameInStats: false,

// Enables sending participants' emails (if available) to callstats and other analytics
// enableEmailInStats: false,

// 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:jitsi-meet.example.com:3478' },
        { 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>'

    // Configuration for the rtcstats server:
    // In order to enable rtcstats one needs to provide a endpoint url.
    // rtcstatsEndpoint: wss://rtcstats-server-pilot.jitsi.net/,

    // The interval at which rtcstats will poll getStats, defaults to 1000ms.
    // If the value is set to 0 getStats won't be polled and the rtcstats client
    // will only send data related to RTCPeerConnection events.
    // rtcstatsPolIInterval: 1000

    // Array of script URLs to load as lib-jitsi-meet "analytics handlers".
    // scriptURLs: [
    //      "libs/analytics-ga.min.js", // google-analytics
    //      "https://example.com/my-custom-analytics.js"
    // ],
},

// 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,

/**
 External API url used to receive branding specific information.
 If there is no url set or there are missing fields, the defaults are applied.
 None of the fields are mandatory and the response must have the shape:
 {
     // The hex value for the colour used as background
     backgroundColor: '#fff',
     // The url for the image used as background
     backgroundImageUrl: 'https://example.com/background-img.png',
     // The anchor url used when clicking the logo image
     logoClickUrl: 'https://example-company.org',
     // The url used for the image used as logo
     logoImageUrl: 'https://example.com/logo-img.png'
 }
*/
// brandingDataUrl: '',

// The URL of the moderated rooms microservice, if available. If it
// is present, a link to the service will be rendered on the welcome page,
// otherwise the app doesn't render it.
// moderatedRoomServiceUrl: 'https://moderated.jitsi-meet.example.com',

// 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 */

My config in /usr/share/jitsi-turner-server :

this is jitsi-meet nginx module configuration

this forward all http traffic to the nginx virtual host port

and the rest to the turn server

stream {
upstream web {
server 127.0.0.1:4444;
}
upstream turn {
server 127.0.0.1:5349;
}
# since 1.13.10
map $ssl_preread_alpn_protocols $upstream {
~\bh2\b web;
~\bhttp/1. web;
default turn;
}

server {
    listen 443;
    listen [::]:443;

    # since 1.11.5
    ssl_preread on;
    proxy_pass $upstream;

    # Increase buffer to serve video
    proxy_buffer_size 10m;
}

}
~

My config turnserver:

jitsi-meet coturn config. Do not modify this line

use-auth-secret
keep-address-family
static-auth-secret=turnSecret
realm=jitsi-meet.example.com
cert=/etc/jitsi/meet/jitsi-meet.example.com.crt
pkey=/etc/jitsi/meet/jitsi-meet.example.com.key
no-multicast-peers
no-cli
no-loopback-peers
no-tcp-relay
no-tcp
listening-port=3478
tls-listening-port=5349
external-ip=external_ip_address
no-tlsv1
no-tlsv1_1

https://ssl-config.mozilla.org/#server=haproxy&version=2.1&config=intermediate&openssl=1.1.0g&guideline=5.4

cipher-list=ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-ECDSA-CHACHA20-POLY1305:ECDHE-RSA-CHACHA20-POLY1305:DHE-RSA-AES128-GCM-SHA256:DHE-RSA-AES256-GCM-SHA384

jitsi-meet coturn relay disable config. Do not modify this line

denied-peer-ip=0.0.0.0-0.255.255.255
denied-peer-ip=10.0.0.0-10.255.255.255
denied-peer-ip=100.64.0.0-100.127.255.255
denied-peer-ip=127.0.0.0-127.255.255.255
denied-peer-ip=169.254.0.0-169.254.255.255
denied-peer-ip=127.0.0.0-127.255.255.255
denied-peer-ip=172.16.0.0-172.31.255.255
denied-peer-ip=192.0.0.0-192.0.0.255
denied-peer-ip=192.0.2.0-192.0.2.255
denied-peer-ip=192.88.99.0-192.88.99.255

Sip-communicator config:

/etc/jitsi/videobridge# vi sip-communicator.properties
org.ice4j.ice.harvest.NAT_HARVESTER_LOCAL_ADDRESS= 10.32.0.41
org.ice4j.ice.harvest.NAT_HARVESTER_PUBLIC_ADDRESS= 37.XXXXXX
org.ice4j.ice.harvest.DISABLE_AWS_HARVESTER=true
org.ice4j.ice.harvest.STUN_MAPPING_HARVESTER_ADDRESSES=meet-jit-si-turnrelay.jitsi.net:443
org.jitsi.videobridge.ENABLE_STATISTICS=true
org.jitsi.videobridge.STATISTICS_TRANSPORT=muc
org.jitsi.videobridge.xmpp.user.shard.HOSTNAME=localhost
org.jitsi.videobridge.xmpp.user.shard.DOMAIN=auth.meet5.mydomainXXXX
org.jitsi.videobridge.xmpp.user.shard.USERNAME=jvb
org.jitsi.videobridge.xmpp.user.shard.PASSWORD=qFhAMe9L
org.jitsi.videobridge.xmpp.user.shard.MUC_JIDS=JvbBrewery@internal.auth.meet5.mydomainXXXX
org.jitsi.videobridge.xmpp.user.shard.MUC_NICKNAME=85ba6a9d-fd61-4c7c-ac91-b7f3d71d63d3

/etc/jitsi/jicofo# vi sip-communicator.properties
org.jitsi.jicofo.BRIDGE_MUC=JvbBrewery@internal.auth.meet5.mydomainxxxxx
org.jitsi.jicofo.ENABLE_H264=false
org.jitsi.jicofo.ENABLE_VP8=false
org.jitsi.jicofo.ENABLE_VP9=true

I added my coturn IP with allowed-peer-ip=172.x.x.x into /etc/turnserver.conf because its IP block is already denied by a denied-peer-ip line. Nginx and coturn are on the same server in my case and 172.x.x.x is the local IP of this server.

But Nginx still can not differ turn and https traffic correctly. The system works if prosody publishes TCP/5349 as the turn port and if I allow accessing directly to this port but this is meaningless. This doesn’t solve the issue of the corporate clients.

Thanks for your reply, i did the test to allow my network in turnserver.conf but I still have the problem.

If I open port 10000 meetings works but I don’t want to use it.
In the previous version turn works perfectly doing nothing in the configuration but in the last version I don’t know what they changed and how to enable again turn.
I also tried the last nightly and same problem.