[SOLVED] Users are kicked out with message "you have been disconnected, rejoin" on Docker-jitis-meet with NGINX proxy

pide2000 · April 22, 2020, 10:54am

Hi Guys,

first of all thanks to all for their awesome work.

Facts:

Debian GNU/Linux 9.12 (stretch) with public IP
NGINX as proxy (takes care of ssl)

I can access the ui and I can create a new channel.

With Firefox: I keep getting disconnected with the rejoin message.
With Chromium: I get disonnected when someone else attempts to connect (everyone has the same experience, tested on Fedora, Adnroid client and MAC Chrome)
subdomain.mydomain.tld/http-bind?room=tester gives the following output.
It works! Now point your BOSH client to this URL to connect to Prosody.
For more information see Prosody: Setting up BOSH.
Port 10000/udp has been tested with nc in the following way:
On the server side LISTEN to 10000 with: : nc -ulk -p 10000 On the client side : nc -vu serverIP ServerPort
Everything I’ve typed in at the client sied will be transferred to the server and is shown in the cli
Other Ports are working as I can connect to the UI.
Config directories are created and are writable

My configs and at the end the error log of firefox.

Jitsi Config:

# Security
#
# Set these to strong passwords to avoid intruders from impersonating a service account
# The service(s) won't start unless these are specified
# Running ./gen-passwords.sh will update .env with strong passwords
# You may skip the Jigasi and Jibri passwords if you are not using those
# DO NOT reuse passwords
#

# XMPP component password for Jicofo
JICOFO_COMPONENT_SECRET=secret

# XMPP password for Jicofo client connections
JICOFO_AUTH_PASSWORD=secret

# XMPP password for JVB client connections
JVB_AUTH_PASSWORD=secret

# XMPP password for Jigasi MUC client connections
JIGASI_XMPP_PASSWORD=secret

# XMPP recorder password for Jibri client connections
JIBRI_RECORDER_PASSWORD=secret

# XMPP password for Jibri client connections
JIBRI_XMPP_PASSWORD=secret


#
# Basic configuration options
#

# Directory where all configuration will be stored
CONFIG=/srv/docker/jitsi/.jitsi-meet-cfg

# Exposed HTTP port
HTTP_PORT=127.0.0.1:8000

# Exposed HTTPS port
HTTPS_PORT=127.0.0.1:8443

# System time zone
TZ=Europe/Berlin

# Public URL for the web service
PUBLIC_URL=https://subdomain.mydomain.tld

# IP address of the Docker host
# See the "Running behind NAT or on a LAN environment" section in the README
DOCKER_HOST_ADDRESS=MYIP


#
# Let's Encrypt configuration
#

# Enable Let's Encrypt certificate generation
#ENABLE_LETSENCRYPT=1

# Domain for which to generate the certificate
#LETSENCRYPT_DOMAIN=meet.example.com

# E-Mail for receiving important account notifications (mandatory)
#LETSENCRYPT_EMAIL=alice@atlanta.net


#
# Etherpad integration (for document sharing)
#

# Set etherpad-lite URL (uncomment to enable)
#ETHERPAD_URL_BASE=http://etherpad.meet.jitsi:9001


#
# Basic Jigasi configuration options (needed for SIP gateway support)
#

# SIP URI for incoming / outgoing calls
#JIGASI_SIP_URI=test@sip2sip.info

# Password for the specified SIP account as a clear text
#JIGASI_SIP_PASSWORD=passw0rd

# SIP server (use the SIP account domain if in doubt)
#JIGASI_SIP_SERVER=sip2sip.info

# SIP server port
#JIGASI_SIP_PORT=5060

# SIP server transport
#JIGASI_SIP_TRANSPORT=UDP

#
# Authentication configuration (see README for details)
#

# Enable authentication
#ENABLE_AUTH=1

# Enable guest access
#ENABLE_GUESTS=1

# Select authentication type: internal, jwt or ldap
#AUTH_TYPE=internal

# JWT authentication
#

# Application identifier
#JWT_APP_ID=my_jitsi_app_id

# Application secret known only to your token
#JWT_APP_SECRET=my_jitsi_app_secret

# (Optional) Set asap_accepted_issuers as a comma separated list
#JWT_ACCEPTED_ISSUERS=my_web_client,my_app_client

# (Optional) Set asap_accepted_audiences as a comma separated list
#JWT_ACCEPTED_AUDIENCES=my_server1,my_server2


# LDAP authentication (for more information see the Cyrus SASL saslauthd.conf man page)
#

# LDAP url for connection
#LDAP_URL=ldaps://ldap.domain.com/

# LDAP base DN. Can be empty
#LDAP_BASE=DC=example,DC=domain,DC=com

# LDAP user DN. Do not specify this parameter for the anonymous bind
#LDAP_BINDDN=CN=binduser,OU=users,DC=example,DC=domain,DC=com

# LDAP user password. Do not specify this parameter for the anonymous bind
#LDAP_BINDPW=LdapUserPassw0rd

# LDAP filter. Tokens example:
# %1-9 - if the input key is user@mail.domain.com, then %1 is com, %2 is domain and %3 is mail
# %s - %s is replaced by the complete service string
# %r - %r is replaced by the complete realm string
#LDAP_FILTER=(sAMAccountName=%u)

# LDAP authentication method
#LDAP_AUTH_METHOD=bind

# LDAP version
#LDAP_VERSION=3

# LDAP TLS using
#LDAP_USE_TLS=1

# List of SSL/TLS ciphers to allow
#LDAP_TLS_CIPHERS=SECURE256:SECURE128:!AES-128-CBC:!ARCFOUR-128:!CAMELLIA-128-CBC:!3DES-CBC:!CAMELLIA-128-CBC

# Require and verify server certificate
#LDAP_TLS_CHECK_PEER=1

# Path to CA cert file. Used when server certificate verify is enabled
#LDAP_TLS_CACERT_FILE=/etc/ssl/certs/ca-certificates.crt

# Path to CA certs directory. Used when server certificate verify is enabled
#LDAP_TLS_CACERT_DIR=/etc/ssl/certs

# Wether to use starttls, implies LDAPv3 and requires ldap:// instead of ldaps://
# LDAP_START_TLS=1


#
# Advanced configuration options (you generally don't need to change these)
#

# Internal XMPP domain
XMPP_DOMAIN=meet.jitsi

# Internal XMPP server
XMPP_SERVER=xmpp.meet.jitsi

# Internal XMPP server URL
XMPP_BOSH_URL_BASE=http://xmpp.meet.jitsi:5280

# Internal XMPP domain for authenticated services
XMPP_AUTH_DOMAIN=auth.meet.jitsi

# XMPP domain for the MUC
XMPP_MUC_DOMAIN=muc.meet.jitsi

# XMPP domain for the internal MUC used for jibri, jigasi and jvb pools
XMPP_INTERNAL_MUC_DOMAIN=internal-muc.meet.jitsi

# XMPP domain for unauthenticated users
XMPP_GUEST_DOMAIN=guest.meet.jitsi

# Custom Prosody modules for XMPP_DOMAIN (comma separated)
XMPP_MODULES=

# Custom Prosody modules for MUC component (comma separated)
XMPP_MUC_MODULES=

# Custom Prosody modules for internal MUC component (comma separated)
XMPP_INTERNAL_MUC_MODULES=

# MUC for the JVB pool
JVB_BREWERY_MUC=jvbbrewery

# XMPP user for JVB client connections
JVB_AUTH_USER=jvb

# STUN servers used to discover the server's public IP
JVB_STUN_SERVERS=stun.adminforge.de:3478

# Media port for the Jitsi Videobridge
JVB_PORT=10000

# TCP Fallback for Jitsi Videobridge for when UDP isn't available
JVB_TCP_HARVESTER_DISABLED=true
JVB_TCP_PORT=4443
JVB_TCP_MAPPED_PORT=4443

# A comma separated list of APIs to enable when the JVB is started [default: none]
# See https://github.com/jitsi/jitsi-videobridge/blob/master/doc/rest.md for more information
#JVB_ENABLE_APIS=rest,colibri

# XMPP user for Jicofo client connections.
# NOTE: this option doesn't currently work due to a bug
JICOFO_AUTH_USER=focus

# Base URL of Jicofo's reservation REST API
#JICOFO_RESERVATION_REST_BASE_URL=http://reservation.example.com

# XMPP user for Jigasi MUC client connections
JIGASI_XMPP_USER=jigasi

# MUC name for the Jigasi pool
JIGASI_BREWERY_MUC=jigasibrewery

# Minimum port for media used by Jigasi
JIGASI_PORT_MIN=20000

# Maximum port for media used by Jigasi
JIGASI_PORT_MAX=20050

# Enable SDES srtp
#JIGASI_ENABLE_SDES_SRTP=1

# Keepalive method
#JIGASI_SIP_KEEP_ALIVE_METHOD=OPTIONS

# Health-check extension
#JIGASI_HEALTH_CHECK_SIP_URI=keepalive

# Health-check interval
#JIGASI_HEALTH_CHECK_INTERVAL=300000
#
# Enable Jigasi transcription
#ENABLE_TRANSCRIPTIONS=1

# Jigasi will record audio when transcriber is on [default: false]
#JIGASI_TRANSCRIBER_RECORD_AUDIO=true

# Jigasi will send transcribed text to the chat when transcriber is on [default: false]
#JIGASI_TRANSCRIBER_SEND_TXT=true

# Jigasi will post an url to the chat with transcription file [default: false]
#JIGASI_TRANSCRIBER_ADVERTISE_URL=true

# Credentials for connect to Cloud Google API from Jigasi
# Please read https://cloud.google.com/text-to-speech/docs/quickstart-protocol
# section "Before you begin" paragraph 1 to 5
# Copy the values from the json to the related env vars
#GC_PROJECT_ID=
#GC_PRIVATE_KEY_ID=
#GC_PRIVATE_KEY=
#GC_CLIENT_EMAIL=
#GC_CLIENT_ID=
#GC_CLIENT_CERT_URL=

# Enable recording
#ENABLE_RECORDING=1

# XMPP domain for the jibri recorder
XMPP_RECORDER_DOMAIN=recorder.meet.jitsi

# XMPP recorder user for Jibri client connections
JIBRI_RECORDER_USER=recorder

# Directory for recordings inside Jibri container
JIBRI_RECORDING_DIR=/config/recordings

# The finalizing script. Will run after recording is complete
JIBRI_FINALIZE_RECORDING_SCRIPT_PATH=/config/finalize.sh

# XMPP user for Jibri client connections
JIBRI_XMPP_USER=jibri

# MUC name for the Jibri pool
JIBRI_BREWERY_MUC=jibribrewery

# MUC connection timeout
JIBRI_PENDING_TIMEOUT=90

# When jibri gets a request to start a service for a room, the room
# jid will look like: roomName@optional.prefixes.subdomain.xmpp_domain
# We'll build the url for the call by transforming that into:
# https://xmpp_domain/subdomain/roomName
# So if there are any prefixes in the jid (like jitsi meet, which
# has its participants join a muc at conference.xmpp_domain) then
# list that prefix here so it can be stripped out to generate
# the call url correctly
JIBRI_STRIP_DOMAIN_JID=muc

# Directory for logs inside Jibri container
JIBRI_LOGS_DIR=/config/logs

# Disable HTTPS: handle TLS connections outside of this setup
DISABLE_HTTPS=1

# Redirect HTTP traffic to HTTPS
# Necessary for Let's Encrypt, relies on standard HTTPS port (443)
#ENABLE_HTTP_REDIRECT=1

# Container restart policy
# Defaults to unless-stopped
RESTART_POLICY=unless-stopped

NGINX config (for Information, ssl certificates are manged central and are not configured in every single server configuration, as I can connect to the UI with a valid certificate everything should be okay from this side):

upstream jitsi {
    server 127.0.0.1:8000;
}

server {
        listen 443 ssl http2;
        listen [::]:443 ssl http2;
        server_name subdomain.mydomain.tld;

        ssl_protocols               TLSv1 TLSv1.1 TLSv1.2;
        ssl_ciphers                  RC4:HIGH:HIGH:!aNULL:!MD5;
        ssl_prefer_server_ciphers   on;

        ssl_session_cache    shared:SSL:10m;
        ssl_session_timeout  10m;

        add_header Strict-Transport-Security "max-age=31536000";

        gzip on;
        gzip_types text/plain text/css application/javascript application/json;
        gzip_vary on;


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


        location / {
            proxy_pass         http://jitsi;
            proxy_redirect     off;
            proxy_set_header   Host $host;
            proxy_set_header   X-Real-IP $remote_addr;
            proxy_set_header   X-Forwarded-For $proxy_add_x_forwarded_for;
            proxy_set_header   X-Forwarded-Host $server_name;
        }
}

Firefox console log:

2020-04-22T10:07:56.076Z [modules/xmpp/xmpp.js] <value>:  (TIME) Strophe disconnecting:	 25688 Logger.js:154:22
2020-04-22T10:07:56.079Z [modules/xmpp/strophe.util.js] <t.a/i.Strophe.log>:  Strophe: Disconnect was called because: undefined Logger.js:154:22
2020-04-22T10:07:56.081Z [modules/xmpp/strophe.util.js] <t.a/i.Strophe.log>:  Strophe: Disconnect was called before Strophe connected to the server Logger.js:154:22
2020-04-22T10:07:56.085Z [modules/xmpp/xmpp.js] <value>:  (TIME) Strophe disconnected:	 25697 Logger.js:154:22
(TIME) index.html loaded:	 425 tester:22:17
2020-04-22T10:07:56.650Z [modules/browser/BrowserCapabilities.js] This appears to be firefox, ver: 75.0 Logger.js:154:22
2020-04-22T10:07:57.718Z [index.web] (TIME) document ready:	 1783 Logger.js:154:22
2020-04-22T10:07:57.825Z [features/base/lastn] There is no active conference, not updating last N Logger.js:154:22
2020-04-22T10:07:57.826Z [modules/UI/videolayout/VideoLayout.js] <changeUserAvatar>:  Missed avatar update - no small video yet for undefined Logger.js:154:22
2020-04-22T10:07:57.844Z [features/base/media] Start muted: Logger.js:154:22
2020-04-22T10:07:57.853Z [features/base/media] Start audio only set to false Logger.js:154:22
2020-04-22T10:07:57.862Z [features/analytics] <te/</<>:  Loaded 0 analytics handlers Logger.js:154:22
2020-04-22T10:07:57.863Z [modules/statistics/AnalyticsAdapter.js] <value>:  Disposing of analytics adapter. Logger.js:154:22
2020-04-22T10:07:57.864Z [features/analytics] <te/<>:  Error: "No analytics handlers created!"
    te functions.js:100
Logger.js:154:22
2020-04-22T10:07:58.067Z [features/base/devices] setAudioOutputDevice: undefined[default] Logger.js:154:22
2020-04-22T10:07:58.077Z [features/base/devices] <O/</</<>:  Failed to set audio output device.
                        Default audio output device will be used instead Error: Audio output device change is not supported Logger.js:154:22
2020-04-22T10:07:58.919Z [modules/RTC/RTCUtils.js] <value/<>:  Available devices:  
Array(7) [ MediaDeviceInfo, MediaDeviceInfo, MediaDeviceInfo, MediaDeviceInfo, MediaDeviceInfo, MediaDeviceInfo, MediaDeviceInfo ]
Logger.js:154:22
2020-04-22T10:07:58.922Z [features/base/devices] Device list updated:
audioInput:
		[TpXU9NyILKjV6ByOxizE4RODNEq0G7XMl4a0NtdNFkE=]
		[M51Dg71bB7DDRaKDZCQiwEpKSL8bXmbWp5BR6Rs54Zs=]
		[4ZdGehBvPeHC82HfztD4AolEQSlHLjraf/7vC+lbgog=]
		[5/hp2IBuzHRJ5TM/GUnWEYIPG1YBQ800kZkmSYGaii8=]
		[M/RiTR/cE279ODJ/frcsZ06K/nEJP9RdZYOVXgNSSzQ=]
audioOutput:

videoInput:
		[1eLz0+YEmckYHGqJdtbnl7wpglXxmBu4pHVm2LEDecw=]
		[1eLz0+YEmckYHGqJdtbnl7wpglXxmBu4pHVm2LEDecw=] Logger.js:154:22
2020-04-22T10:07:58.939Z [modules/xmpp/xmpp.js] <value>:  P2P STUN servers:  
Array [ {…} ]
Logger.js:154:22
2020-04-22T10:07:58.950Z [modules/xmpp/xmpp.js] <value>:  (TIME) Strophe connecting:	 3016 Logger.js:154:22
2020-04-22T10:07:58.956Z [modules/RTC/RTCUtils.js] <value>:  Using the new gUM flow Logger.js:154:22
2020-04-22T10:07:58.957Z [modules/RTC/RTCUtils.js] <value/i<>:  Got media constraints:  
Object { video: {…}, audio: {…} }
Logger.js:154:22
2020-04-22T10:07:59.774Z [features/base/storage] <persistState>:  redux state persisted. beb67e96e9425554e30a9042c7027898 -> d3befd57fe85847dcba2fead18c6a9e9 Logger.js:154:22
2020-04-22T10:07:59.819Z [modules/xmpp/XmppConnection.js] <value>:  Stream resume enabled, but WebSockets are not enabled Logger.js:154:22
2020-04-22T10:07:59.819Z [modules/xmpp/xmpp.js] <value>:  (TIME) Strophe connected:	 3885 Logger.js:154:22
2020-04-22T10:07:59.820Z [modules/xmpp/xmpp.js] <value>:  My Jabber ID: 2ydoneivrgnymiqd@meet.jitsi/3i1SXbtG Logger.js:154:22
2020-04-22T10:07:59.991Z [modules/xmpp/strophe.ping.js] <value>:  XMPP pings will be sent every 10000 ms Logger.js:154:22
2020-04-22T10:08:01.370Z [modules/RTC/RTCUtils.js] <value/</<>:  onUserMediaSuccess Logger.js:154:22
2020-04-22T10:08:01.376Z [modules/RTC/JitsiLocalTrack.js] <value>:  Setting new MSID: {7328684b-64e5-402e-8ccc-70d175e75877} {1fa0fac8-44ab-4573-90c5-6e3563371b98} on LocalTrack[undefined,audio] Logger.js:154:22
2020-04-22T10:08:01.379Z [modules/RTC/JitsiLocalTrack.js] <value>:  Setting new MSID: {7d7eabd8-62c5-4374-ab97-fa0595830788} {2ee2344a-f84d-4678-899d-719c8da951ed} on LocalTrack[undefined,video] Logger.js:154:22
2020-04-22T10:08:01.420Z [conference.js] <init/<>:  initialized with 2 local tracks Logger.js:154:22
2020-04-22T10:08:01.460Z [modules/xmpp/xmpp.js] <value>:  JID 2ydoneivrgnymiqd@meet.jitsi/3i1SXbtG using MUC nickname feaee37a Logger.js:154:22
2020-04-22T10:08:01.461Z [modules/xmpp/ChatRoom.js] Joined MUC as tester@muc.meet.jitsi/feaee37a Logger.js:154:22
2020-04-22T10:08:01.463Z [modules/e2eping/e2eping.js] Initializing e2e ping; pingInterval=10000, analyticsInterval=60000. Logger.js:154:22
2020-04-22T10:08:01.465Z [modules/connectivity/ParticipantConnectionStatus.js] RtcMuteTimeout set to: 10000 Logger.js:154:22
2020-04-22T10:08:01.469Z [modules/statistics/AvgRTPStatsReporter.js] Avg RTP stats will be calculated every 15 samples Logger.js:154:22
2020-04-22T10:08:01.471Z [JitsiConference.js] <oe>:  backToP2PDelay: 5 Logger.js:154:22
2020-04-22T10:08:01.472Z [modules/videosipgw/VideoSIPGW.js] creating VideoSIPGW Logger.js:154:22
2020-04-22T10:08:01.502Z [JitsiConference.js] <oe.prototype._doReplaceTrack>:  _doReplaceTrack - no JVB JingleSession Logger.js:154:22
2020-04-22T10:08:01.503Z [JitsiConference.js] <oe.prototype._doReplaceTrack>:  _doReplaceTrack - no P2P JingleSession Logger.js:154:22
2020-04-22T10:08:01.505Z [JitsiConference.js] <oe.prototype._doReplaceTrack>:  _doReplaceTrack - no JVB JingleSession Logger.js:154:22
2020-04-22T10:08:01.507Z [JitsiConference.js] <oe.prototype._doReplaceTrack>:  _doReplaceTrack - no P2P JingleSession Logger.js:154:22
2020-04-22T10:08:01.514Z [modules/remotecontrol/RemoteControl.js] <init>:  Initializing remote control. Logger.js:154:22
2020-04-22T10:08:01.522Z [modules/xmpp/moderator.js] <l.prototype.setFocusUserJid>:  Focus jid set to:  focus@auth.meet.jitsi Logger.js:154:22
2020-04-22T10:08:01.524Z [modules/xmpp/moderator.js] <l.prototype.createConferenceIq>:  Session ID: null machine UID: 0c7e6ff11349d3e0ef39bf96cdbd8a99 Logger.js:154:22
2020-04-22T10:08:01.537Z [features/base/tracks] <Y/</</<>:  Replace audio track - unmuted Logger.js:154:22
2020-04-22T10:08:01.541Z [features/base/tracks] <Y/</</<>:  Replace video track - unmuted Logger.js:154:22
2020-04-22T10:08:01.625Z [modules/UI/videolayout/VideoLayout.js] <updateLargeVideo>:  Enforcing large video update for stream change Logger.js:154:22
2020-04-22T10:08:01.634Z [modules/UI/videolayout/SmallVideo.js] <updateView>:  Displaying blackness-with-name for feaee37a, data: [{"isCurrentlyOnLargeVideo":true,"isHovered":false,"isAudioOnly":false,"tileViewEnabled":false,"isVideoPlayable":true,"hasVideo":true,"connectionStatus":null,"videoStream":true,"isVideoMuted":false,"videoStreamMuted":false}] Logger.js:154:22
2020-04-22T10:08:02.136Z [modules/UI/videolayout/LargeVideoManager.js] <scheduleLargeVideoUpdate/<>:  hover in feaee37a Logger.js:154:22
2020-04-22T10:08:02.476Z [modules/xmpp/moderator.js] <l.prototype.parseConfigOptions>:  Authentication enabled: false Logger.js:154:22
2020-04-22T10:08:02.477Z [modules/xmpp/moderator.js] <l.prototype.parseConfigOptions>:  External authentication enabled: false Logger.js:154:22
2020-04-22T10:08:02.482Z [modules/xmpp/moderator.js] <l.prototype.parseConfigOptions>:  Sip gateway enabled:  false Logger.js:154:22
2020-04-22T10:08:02.814Z [modules/UI/videolayout/LargeVideoManager.js] <scheduleLargeVideoUpdate/<>:  hover in feaee37a Logger.js:154:22
2020-04-22T10:08:03.681Z [modules/xmpp/ChatRoom.js] <value>:  entered tester@muc.meet.jitsi/focus 
Object { affiliation: "owner", role: "moderator", jid: "focus@auth.meet.jitsi/focus1298574085840080", isFocus: true, isHiddenDomain: false }
Logger.js:154:22
2020-04-22T10:08:03.683Z [modules/xmpp/ChatRoom.js] <value>:  Ignore focus: tester@muc.meet.jitsi/focus, real JID: focus@auth.meet.jitsi/focus1298574085840080 Logger.js:154:22
2020-04-22T10:08:03.687Z [modules/version/ComponentsVersions.js] <o.prototype.processVersions/<>:  Got xmpp version: Prosody(0.11.5,Linux) Logger.js:154:22
2020-04-22T10:08:03.687Z [modules/version/ComponentsVersions.js] <o.prototype.processVersions/<>:  Got focus version: JiCoFo(1.0.544,Linux) Logger.js:154:22
2020-04-22T10:08:03.688Z [modules/version/ComponentsVersions.js] <o.prototype.processVersions/<>:  Got videobridge version: 2.1.169-ga28eb88e Logger.js:154:22
2020-04-22T10:08:03.688Z [modules/xmpp/ChatRoom.js] <value>:  entered tester@muc.meet.jitsi/85d69d41 
Object { affiliation: "owner", role: "moderator", jid: "3ysffgdor3iwrlv9@meet.jitsi/4U2Y4jXk", isFocus: false, isHiddenDomain: false, statsID: "Johan-UUq", nick: undefined }
Logger.js:154:22
2020-04-22T10:08:03.720Z [modules/UI/videolayout/RemoteVideo.js] <updateConnectionStatusIndicator>:  85d69d41 thumbnail connection status: active Logger.js:154:22
2020-04-22T10:08:03.732Z [modules/UI/videolayout/SmallVideo.js] <updateView>:  Displaying avatar for 85d69d41, data: [{"isCurrentlyOnLargeVideo":false,"isHovered":false,"isAudioOnly":false,"tileViewEnabled":false,"isVideoPlayable":null,"hasVideo":false,"connectionStatus":"active","mutedWhileDisconnected":false,"wasVideoPlayed":false,"videoStream":false,"isVideoMuted":false,"videoStreamMuted":"no stream"}] Logger.js:154:22
2020-04-22T10:08:03.739Z [modules/UI/videolayout/RemoteVideo.js] <updateConnectionStatusIndicator>:  85d69d41 thumbnail connection status: active Logger.js:154:22
2020-04-22T10:08:03.753Z [features/base/lastn] There is no active conference, not updating last N Logger.js:154:22
2020-04-22T10:08:03.778Z [conference.js] <_setupListeners/<>:  USER 85d69d41 connnected: 
Object { _jid: "tester@muc.meet.jitsi/85d69d41", _id: "85d69d41", _conference: {…}, _displayName: undefined, _supportsDTMF: false, _tracks: [], _role: "moderator", _status: undefined, _hidden: false, _statsID: "Johan-UUq", … }
Logger.js:154:22
2020-04-22T10:08:03.782Z [JitsiConference.js] <oe.prototype._maybeStartOrStopP2P>:  Auto P2P disabled Logger.js:154:22
2020-04-22T10:08:03.801Z [conference.js] <_setupListeners/<>:  My role changed, new role: none Logger.js:154:22
2020-04-22T10:08:03.807Z [modules/xmpp/ChatRoom.js] <value>:  (TIME) MUC joined:	 7873 Logger.js:154:22
2020-04-22T10:08:03.852Z [modules/xmpp/ChatRoom.js] <value>:  Subject is changed to Logger.js:154:22
wasm streaming compile failed: TypeError: Response has unsupported MIME type index.js:8:9333
falling back to ArrayBuffer instantiation index.js:8:9375
2020-04-22T10:08:04.275Z [modules/UI/videolayout/LargeVideoManager.js] <scheduleLargeVideoUpdate/<>:  hover in 85d69d41 Logger.js:154:22
2020-04-22T10:08:04.842Z [features/base/devices] Device list updated:
audioInput:
		Built-in Audio Analog Stereo[TpXU9NyILKjV6ByOxizE4RODNEq0G7XMl4a0NtdNFkE=]
		Monitor of Built-in Audio Analog Stereo[M51Dg71bB7DDRaKDZCQiwEpKSL8bXmbWp5BR6Rs54Zs=]
		Monitor of USB Audio Line Out[4ZdGehBvPeHC82HfztD4AolEQSlHLjraf/7vC+lbgog=]
		Monitor of USB Audio Headphones[5/hp2IBuzHRJ5TM/GUnWEYIPG1YBQ800kZkmSYGaii8=]
		USB Audio Microphone[M/RiTR/cE279ODJ/frcsZ06K/nEJP9RdZYOVXgNSSzQ=]
audioOutput:

videoInput:
		Integrated_Webcam_HD: Integrate[1eLz0+YEmckYHGqJdtbnl7wpglXxmBu4pHVm2LEDecw=]
		Integrated_Webcam_HD: Integrate[1eLz0+YEmckYHGqJdtbnl7wpglXxmBu4pHVm2LEDecw=] Logger.js:154:22
2020-04-22T10:08:04.871Z [features/local-recording] <setMicDevice>:  Switch microphone to TpXU9NyILKjV6ByOxizE4RODNEq0G7XMl4a0NtdNFkE= Logger.js:154:22
2020-04-22T10:08:04.896Z [modules/UI/videolayout/RemoteVideo.js] <updateConnectionStatusIndicator>:  85d69d41 thumbnail connection status: active Logger.js:154:22
2020-04-22T10:08:04.899Z [modules/UI/videolayout/SmallVideo.js] <updateView>:  Displaying avatar-with-name for 85d69d41, data: [{"isCurrentlyOnLargeVideo":true,"isHovered":false,"isAudioOnly":false,"tileViewEnabled":false,"isVideoPlayable":null,"hasVideo":false,"connectionStatus":"active","mutedWhileDisconnected":false,"wasVideoPlayed":false,"videoStream":false,"isVideoMuted":true,"videoStreamMuted":"no stream"}] Logger.js:154:22
2020-04-22T10:08:04.900Z [modules/UI/videolayout/SmallVideo.js] <updateView>:  Displaying video for feaee37a, data: [{"isCurrentlyOnLargeVideo":false,"isHovered":false,"isAudioOnly":false,"tileViewEnabled":false,"isVideoPlayable":true,"hasVideo":true,"connectionStatus":null,"videoStream":true,"isVideoMuted":false,"videoStreamMuted":false}] Logger.js:154:22
2020-04-22T10:08:05.061Z [modules/RTC/RTCUtils.js] <value>:  Using the new gUM flow Logger.js:154:22
2020-04-22T10:08:05.063Z [modules/RTC/RTCUtils.js] <value/i<>:  Got media constraints:  
Object { video: false, audio: {…} }
Logger.js:154:22
2020-04-22T10:08:05.217Z [JitsiMeetJS.js] <getGlobalOnErrorHandler>:  UnhandledError: null Script: null Line: null Column: null StackTrace:  Error: "Strophe: request id 10.1 error 413 happened"
    log strophe.util.js:89
    error strophe.umd.js:1392
    _onRequestStateChange strophe.umd.js:5017
Logger.js:154:22
2020-04-22T10:08:05.221Z [modules/xmpp/strophe.util.js] <t.a/i.Strophe.log>:  Strophe: request id 10.1 error 413 happened Logger.js:154:22
2020-04-22T10:08:05.223Z [modules/xmpp/strophe.util.js] <t.a/i.Strophe.log>:  Strophe: request errored, status: 413, number of errors: 1 Logger.js:154:22
2020-04-22T10:08:05.225Z [modules/xmpp/xmpp.js] <value>:  (TIME) Strophe disconnecting:	 9291 Logger.js:154:22
2020-04-22T10:08:05.229Z [modules/xmpp/xmpp.js] <value>:  (TIME) Strophe disconnected:	 9294 Logger.js:154:22
2020-04-22T10:08:05.230Z [modules/xmpp/strophe.ping.js] <value>:  Ping interval cleared Logger.js:154:22
2020-04-22T10:08:05.232Z [modules/xmpp/xmpp.js] <value>:  XMPP connection dropped! Logger.js:154:22
2020-04-22T10:08:05.233Z [modules/statistics/statistics.js] <S.sendAnalyticsAndLog>:  {"type":"operational","action":"connection.failed","attributes":{"error_type":"connection.droppedError","error_message":"connection-dropped-error","suspend_time":0,"time_since_last_success":1555}} Logger.js:154:22
2020-04-22T10:08:05.242Z [features/overlay] <componentDidMount>:  The conference will be reloaded after 26 seconds. Logger.js:154:22
2020-04-22T10:08:05.272Z [modules/UI/videolayout/VideoLayout.js] <changeUserAvatar>:  Missed avatar update - no small video yet for local Logger.js:154:22
2020-04-22T10:08:05.295Z [modules/UI/videolayout/VideoLayout.js] <removeParticipantContainer>:  Removing remote video: 85d69d41 Logger.js:154:22
2020-04-22T10:08:05.298Z [modules/UI/videolayout/SmallVideo.js] <remove>:  Remove thumbnail 85d69d41 Logger.js:154:22
2020-04-22T10:08:05.310Z [features/base/lastn] There is no active conference, not updating last N Logger.js:154:22
2020-04-22T10:08:05.312Z [modules/UI/videolayout/SmallVideo.js] <remove>:  Remove thumbnail feaee37a Logger.js:154:22
2020-04-22T10:08:05.319Z [modules/rttmonitor/rttmonitor.js] <value>:  Stopping RttMonitor. Logger.js:154:22
2020-04-22T10:08:05.320Z [modules/e2eping/e2eping.js] <value>:  Stopping e2eping Logger.js:154:22
2020-04-22T10:08:05.323Z [modules/xmpp/ChatRoom.js] <value>:  do leave tester@muc.meet.jitsi/feaee37a Logger.js:154:22
2020-04-22T10:08:05.324Z [modules/RTC/RTC.js] <value>:  Removed remote tracks for 85d69d41 count: 0 Logger.js:154:22
2020-04-22T10:08:05.328Z [modules/UI/videolayout/VideoLayout.js] <removeParticipantContainer>:  No remote video for 85d69d41 Logger.js:154:22
2020-04-22T10:08:05.329Z [conference.js] <_setupListeners/<>:  USER 85d69d41 LEFT: 
Object { _jid: "tester@muc.meet.jitsi/85d69d41", _id: "85d69d41", _conference: {…}, _displayName: undefined, _supportsDTMF: false, _tracks: [], _role: "moderator", _status: undefined, _hidden: false, _statsID: "Johan-UUq", … }
Logger.js:154:22
2020-04-22T10:08:05.330Z [JitsiConference.js] <oe.prototype._maybeStartOrStopP2P>:  Auto P2P disabled Logger.js:154:22
XML Parsing Error: mismatched tag. Expected: </hr>.
Location: https://MYDOMAIN/http-bind?room=tester
Line Number 6, Column 3: http-bind:6:3
2020-04-22T10:08:05.473Z [JitsiMeetJS.js] <getGlobalOnErrorHandler>:  UnhandledError: null Script: null Line: null Column: null StackTrace:  Error: "Not connected"
    value XmppConnection.js:351
    value ChatRoom.js:261
    value ChatRoom.js:1495
    value ChatRoom.js:1472
    leave JitsiConference.js:585
    Z conference.js:428
    emit events.js:151
    value xmpp.js:330
    value XmppConnection.js:243
    _changeConnectStatus strophe.umd.js:3011
    _doDisconnect strophe.umd.js:3052
    _interceptDoDisconnect strophe.stream-management.js:207
    _onRequestStateChange strophe.umd.js:5026
Logger.js:154:22
Error: Not connected XmppConnection.js:351:18
2020-04-22T10:08:06.846Z [features/base/storage] <persistState>:  redux state persisted. d3befd57fe85847dcba2fead18c6a9e9 -> 7ec64dc0f386b53d29c9933efb529392 Logger.js:154:22
2020-04-22T10:08:09.368Z [modules/RTC/RTCUtils.js] <value/</<>:  onUserMediaSuccess Logger.js:154:22
2020-04-22T10:08:09.369Z [modules/RTC/JitsiLocalTrack.js] <value>:  Setting new MSID: {7bb95317-d5c2-415e-b845-d3454833a7d3} {92b002b1-10e8-425e-8129-39b9305aadeb} on LocalTrack[undefined,audio] Logger.js:154:22
Connecting AudioNodes from AudioContexts with different sample-rate is currently not supported. TrackVADEmitter.js:104:47
2020-04-22T10:08:09.406Z [JitsiMeetJS.js] <getGlobalOnErrorHandler>:  UnhandledError: null Script: null Line: null Column: null StackTrace:  DOMException: "AudioContext.createMediaStreamSource: Connecting AudioNodes from AudioContexts with different sample-rate is currently not supported." Logger.js:154:22
NotSupportedError: AudioContext.createMediaStreamSource: Connecting AudioNodes from AudioContexts with different sample-rate is currently not supported. lib-jitsi-meet.min.js:1

If there is any further information you need just let me know.

Thanks in advance and stay save

pide2000 · April 22, 2020, 7:18pm

So, figured it out on my own.

The important part is the message:
Error: "Strophe: request id 10.1 error 413 happened"
in the console log. I just not realized 413 is a http erro code. Later on I just come around the network tab in the dev tools and there it was again.
Little investigation brings up this lines in the nginx root conf:
client_body_buffer_size 1K;
client_max_body_size 1k;
Change it to a higher value for the jitsi domain fixed the problem.

hkhait · April 23, 2020, 9:21am

Hello,
Could you please tell me what values did you set for client_body_buffer_size and client_max_body_size ? is there a maximum value for each parameter ?

pide2000 · April 23, 2020, 4:19pm

I’ve set it to zero which corresponds to no limit.
I will track the size of the post request in the browser to the bosh url subdomain.mydomain.tld/http-bind?room=tester soon so I can adjust the size and I will inform you here.