Add jitsi

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diff --git a/app/jitsi/integration/README.md b/app/jitsi/integration/README.md
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+## About Jitsi
+
+Several server components:
+  - prosody XMPP (ext)
+  - jitsi videobridge aka JVB
+  - jitsi conference focus aka jicofo aka focus
+  - jitsi meet
+  - octo
+  - jigasi
+  - jibri
+  - etc.
+
+Some libs:
+  - libjitsi seems deprecated
+  - jicoco contains some parent classes to handle Jitsi's Configuration
+  - [jitsi-utils](https://github.com/jitsi/jitsi-utils) contains the Logger definition for example
+  - [ice4j](https://github.com/jitsi/ice4j) contains jitsi's implementation of WebRTC
+  - etc.
+
+Client components:
+  - jitsi meet electron
+  - jitsi android/ios
+  - etc.
+
+## Conf
+
+Base conf:
+
+  - [videobridge.conf](https://github.com/jitsi/jitsi-videobridge/blob/master/jvb/src/main/resources/reference.conf)
+  - [jicofo.conf](https://github.com/jitsi/jicofo/blob/master/src/main/resources/reference.conf)
+
+the following is used in videobridge.conf:
+[jicoco/MucClientConfiguration](https://github.com/jitsi/jicoco/blob/master/jicoco/src/main/java/org/jitsi/xmpp/mucclient/MucClientConfiguration.java)
+
+How the new configuration is read in jicoco:
+https://github.com/jitsi/jicoco/blob/master/jicoco-config/src/main/kotlin/org/jitsi/config/JitsiConfig.kt#L83-L91
+They use this library: https://github.com/lightbend/config
+We are particularly interested by: https://github.com/lightbend/config#standard-behavior
+Using 'application.conf' with classpath does not seem to work.
+But, specifying the file path as `-Dconfig.file=/etc/jitsi/jicofo.conf` works!
+
+Some parameters are also set independently of lightbend hocon config.
+They are seen in jicofo entrypoint:
+https://github.com/jitsi/jicofo/blob/master/src/main/java/org/jitsi/jicofo/Main.java
+Many of these parameters can be in fact read from the HOCON file except one: the `--secret` parameter or the `JICOFO_SECRET` env variable.
+But we can see this is a deprecated thing, it has been already removed from master: https://github.com/jitsi/jicofo/commit/c9e5b50a8b4e77f8b8cb8831a4a044a53edfcf48
+For now (as per v5390) we will keep `JICOFO_SECRET` environment variable but will assume no other environment variable is set
+But maybe this value is deprecated: the check is still here but it is not used anymore?!
+
+## Generate certs with prosody
+```
+prosodyctl cert generate auth.jitsi
+prosodyctl cert generate jitsi
+```
+
+## An example prosody configuration file
+
+https://github.com/jitsi/jitsi-meet/blob/master/doc/example-config-files/prosody.cfg.lua.example
+
+but this one is not the one used by the [debian postinst script](https://github.com/jitsi/jitsi-meet/blob/master/debian/jitsi-meet-prosody.postinst)
+instead, we should look at this one: https://github.com/jitsi/jitsi-meet/blob/master/doc/debian/jitsi-meet-prosody/prosody.cfg.lua-jvb.example
+
+Jitsi can be configured to authenticated through tokens,
+the postinst file is here: https://github.com/jitsi/jitsi-meet/blob/master/debian/jitsi-meet-tokens.postinst
+
+## Remote debug
+
+Add this parameter to the java process you want to debug (either jicofo or jvb). It must be added by modifying the entrypoint script, next to the respective Dockerfile of each container.
+
+```
+-agentlib:jdwp=transport=dt_socket,server=y,suspend=y,address=*:5005
+```
+
+## Be careful
+
+jiti-videobridge (jvb) does not start to listen on ICE ports (both TCP and UDP) at boot.
+Instead, listening is triggered on the creation of the first conference (a 2 people P2P conference is enough).
+A nice entrypoint to check with your debugger is:
+  - [Videobridge.java#XmppConnectionEventHandle.colibriConferenceIqReceived](https://github.com/jitsi/jitsi-videobridge/blob/256dc7acb7ee10440502a6073a498329eaf1e819/jvb/src/main/java/org/jitsi/videobridge/Videobridge.java#L627)
+  - [VideobridgeShim.java#VideobridgeShim.handleColibriConferenceIQ](https://github.com/jitsi/jitsi-videobridge/blob/256dc7acb7ee10440502a6073a498329eaf1e819/jvb/src/main/java/org/jitsi/videobridge/shim/VideobridgeShim.java#L251)
+  - [ConferenceShim.java#ConferenceShim.initializeSignaledEndpoints](https://github.com/jitsi/jitsi-videobridge/blob/256dc7acb7ee10440502a6073a498329eaf1e819/jvb/src/main/java/org/jitsi/videobridge/shim/ConferenceShim.java#L274)
+  - [ConferenceShim.java#ConferenceShim.ensureEndpointCreated](https://github.com/jitsi/jitsi-videobridge/blob/256dc7acb7ee10440502a6073a498329eaf1e819/jvb/src/main/java/org/jitsi/videobridge/shim/ConferenceShim.java#L312)
+  - [Conference.java#Conference.createLocalEndpoint](https://github.com/jitsi/jitsi-videobridge/blob/256dc7acb7ee10440502a6073a498329eaf1e819/jvb/src/main/java/org/jitsi/videobridge/Conference.java#L602)
+  - [Endpoint.java#Endpoint.new](https://github.com/jitsi/jitsi-videobridge/blob/256dc7acb7ee10440502a6073a498329eaf1e819/jvb/src/main/java/org/jitsi/videobridge/Endpoint.java#L254)
+  - [IceTransport.kt#IceTransport.iceAgent(init)](https://github.com/jitsi/jitsi-videobridge/blob/0c2ac250ec6b518eaf75fbc83f7936ec01e7b5f6/jvb/src/main/kotlin/org/jitsi/videobridge/transport/ice/IceTransport.kt#L99)
+  - [IceTransport.kt#companionObject.appendHarvesters](https://github.com/jitsi/jitsi-videobridge/blob/0c2ac250ec6b518eaf75fbc83f7936ec01e7b5f6/jvb/src/main/kotlin/org/jitsi/videobridge/transport/ice/IceTransport.kt#L350)
+
+## Resources to understand jitsi
+
+  - [jicofo/debian/postinst](https://github.com/jitsi/jicofo/blob/master/debian/postinst)
+  - [videobridge/debian/postinst](https://github.com/jitsi/jitsi-videobridge/blob/master/debian/postinst)
diff --git a/app/jitsi/integration/docker-compose.yml b/app/jitsi/integration/docker-compose.yml
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+version: '3.4'
+services:
+  jitsi-xmpp:
+    image: superboum/amd64_jitsi_xmpp:v9
+    volumes:
+      - "./prosody/prosody.cfg.lua:/etc/prosody/prosody.cfg.lua:ro"
+      - "./prosody/certs/jitsi.crt:/var/lib/prosody/jitsi.crt:ro"
+      - "./prosody/certs/jitsi.key:/var/lib/prosody/jitsi.key:ro"
+      - "./prosody/certs/auth.jitsi.crt:/var/lib/prosody/auth.jitsi.crt:ro"
+      - "./prosody/certs/auth.jitsi.key:/var/lib/prosody/auth.jitsi.key:ro"
+    environment:
+      - JICOFO_AUTH_PASSWORD=jicofopass
+      - JVB_AUTH_PASSWORD=jvbpass
+ 
+  jitsi-conference-focus:
+    image: superboum/amd64_jitsi_conference_focus:v7
+    volumes:
+      - "./prosody/certs/jitsi.crt:/usr/local/share/ca-certificates/jitsi.crt:ro"
+      - "./prosody/certs/auth.jitsi.crt:/usr/local/share/ca-certificates/auth.jitsi.crt:ro"
+      - "./jicofo/jicofo.conf:/etc/jitsi/jicofo.conf:ro"
+
+  jitsi-videobridge:
+    image: superboum/amd64_jitsi_videobridge:v17
+    volumes:
+      - "./prosody/certs/jitsi.crt:/usr/local/share/ca-certificates/jitsi.crt:ro"
+      - "./prosody/certs/auth.jitsi.crt:/usr/local/share/ca-certificates/auth.jitsi.crt:ro"
+      - "./jvb/videobridge.conf:/etc/jitsi/videobridge.conf:ro"
+      - "./jvb/logging.properties:/usr/share/jvb/lib/logging.properties:ro"
+    ports:
+      - "8089:8089/tcp"
+      - "10000:10000/udp"
+
+  jitsi-meet:
+    image: superboum/amd64_jitsi_meet:v4
+    volumes:
+      - "./prosody/certs/jitsi.crt:/etc/nginx/jitsi.crt:ro"
+      - "./prosody/certs/jitsi.key:/etc/nginx/jitsi.key:ro"
+      - "./meet/config.js:/srv/jitsi-meet/config.js:ro"
+      - "./meet/nginx.conf:/etc/nginx/nginx.conf:ro"
+    ports:
+      - "443:443"
+
diff --git a/app/jitsi/integration/jicofo/jicofo.conf b/app/jitsi/integration/jicofo/jicofo.conf
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+++ b/app/jitsi/integration/jicofo/jicofo.conf
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+jicofo {
+  // Authentication with external services
+  authentication {
+    enabled = false
+    // The type of authentication. Supported values are XMPP, JWT or SHIBBOLETH (default).
+    type = SHIBBOLETH
+
+    // The pattern of authentication URL. See ShibbolethAuthAuthority for more information.
+    # login-url =
+
+    # logout-url =
+
+    authentication-lifetime = 24 hours
+    enable-auto-login = true
+  }
+  // Configuration related to jitsi-videobridge
+  bridge {
+    // The maximum number of participants in a single conference to put on one bridge (use -1 for no maximum).
+    max-bridge-participants = -1
+    // The assumed maximum packet rate that a bridge can handle.
+    max-bridge-packet-rate = 50000
+    // The assumed average packet rate per participant.
+    average-participant-packet-rate-pps = 500
+    // The assumed average stress per participant.
+    average-participant-stress = 0.01
+    // The assumed time that an endpoint takes to start contributing fully to the load on a bridge. To avoid allocating
+    // a burst of endpoints to the same bridge, the bridge stress is adjusted by adding the number of new endpoints
+    // in the last [participant-rampup-time] multiplied by [average-participant-stress].
+    participant-rampup-interval = 20 seconds
+    // The stress level above which a bridge is considered overstressed.
+    stress-threshold = 0.8
+    // The amount of to wait before retrying using a failed bridge.
+    failure-reset-threshold = 1 minute
+    // The bridge selection strategy. The built-in strategies are:
+    // SingleBridgeSelectionStrategy: Use the least loaded bridge, do not split a conference between bridges (Octo).
+    // SplitBridgeSelectionStrategy: Use a separate bridge for each participant (for testing).
+    // RegionBasedBridgeSelectionStrategy: Attempt to put each participant in a bridge in their local region (i.e. use
+    //    Octo for geo-location).
+    // IntraRegionBridgeSelectionStrategy: Use additional bridges when a bridge becomes overloaded (i.e. use Octo for
+    //    load balancing).
+    //
+    // Additionally, you can use the fully qualified class name for custom BridgeSelectionStrategy implementations.
+    selection-strategy = SingleBridgeSelectionStrategy
+    health-checks {
+      // Whether jicofo should perform periodic health checks to the connected bridges.
+      enabled = true
+      // The interval at which to perform health checks.
+      interval = 10 seconds
+      // When a health checks times out, jicofo will retry and only consider it fail after the retry fails. This
+      // configures the delay between the original health check timing out and the second health check being sent.
+      // It is a duration and defaults to half the [interval].
+      # retry-delay = 5 seconds
+    }
+
+    // The JID of the MUC to be used as a brewery for bridge instances.
+    brewery-jid = "jvbbrewery@internal.auth.jitsi"
+  }
+  // Configure the codecs and RTP extensions to be used in the offer sent to clients.
+  codec {
+    video {
+      vp8 {
+        enabled = true
+        pt = 100
+        // Payload type for the associated RTX stream. Set to -1 to disable RTX.
+        rtx-pt = 96
+      }
+      vp9 {
+        enabled = true
+        pt = 101
+        // Payload type for the associated RTX stream. Set to -1 to disable RTX.
+        rtx-pt = 97
+      }
+      h264 {
+        enabled = true
+        pt = 107
+        // Payload type for the associated RTX stream. Set to -1 to disable RTX.
+        rtx-pt = 99
+      }
+    }
+
+    audio {
+      isac-16000 {
+        enabled = true
+        pt = 103
+      }
+      isac-32000 {
+        enabled = true
+        pt = 104
+      }
+      opus {
+        enabled = true
+        pt = 111
+        minptime = 10
+        use-inband-fec = true
+        red {
+          enabled = false
+          pt = 112
+        }
+      }
+      telephone-event {
+        enabled = true
+        pt = 126
+      }
+    }
+
+    // RTP header extensions
+    rtp-extensions {
+      audio-level {
+        enabled = true
+        id = 1
+      }
+      tof {
+        // TOF is currently disabled, because we don't support it in the bridge
+        // (and currently clients seem to not use it when abs-send-time is
+        // available).
+        enabled = false
+        id = 2
+      }
+      abs-send-time {
+        enabled = true
+        id = 3
+      }
+      rid {
+        enabled = false
+        id = 4
+      }
+      tcc {
+        enabled = true
+        id = 5
+      }
+      video-content-type {
+        enabled = false
+        id = 7
+      }
+      framemarking {
+        enabled = false
+        id = 9
+      }
+    }
+  }
+
+  conference {
+    // Whether to automatically grant the 'owner' role to the first participant in the conference (and subsequently to
+    // the next in line when the current owner leaves).
+    enable-auto-owner = true
+
+    // How long to wait for the initial participant in a conference.
+    initial-timeout = 15 seconds
+
+    // Whether jicofo should inject a random SSRC for endpoints which don't advertise any SSRCs. This is a temporary
+    // workaround for an issue with signaling endpoints for Octo.
+    inject-ssrc-for-recv-only-endpoints = false
+
+    max-ssrcs-per-user = 20
+
+    // How long a participant's media session will be kept alive once it remains the only participant in the room.
+    single-participant-timeout = 20 seconds
+
+    // The minimum number of participants required for the conference to be started.
+    min-participants = 2
+
+    // Experimental.
+    enable-lip-sync = false
+
+    shared-document {
+      // If `true` the shared document uses a random name. Otherwise, it uses the conference name.
+      use-random-name = false
+    }
+  }
+
+  // Configuration for the internal health checks performed by jicofo.
+  health {
+    // Whether to perform health checks.
+    enabled = false
+
+    // The interval between health checks. If set to 0, periodic health checks will not be performed.
+    interval = 10 seconds
+
+    # The timeout for a health check
+    timeout = 30 seconds
+
+    # If performing a health check takes longer than this, it is considered unsuccessful.
+    max-check-duration = 20 seconds
+
+    # The prefix to use when creating MUC rooms for the purpose of health checks.
+    room-name-prefix = "__jicofo-health-check"
+  }
+
+  jibri {
+    // The JID of the MUC to be used as a brewery for jibri instances for streaming.
+    # brewery-jid = "jibribrewery@example.com"
+
+    // How many times to retry a given Jibri request before giving up. Set to -1 to allow infinite retries.
+    num-retries = 5
+
+    // How long to wait for Jibri to start recording from the time it accepts a START request.
+    pending-timeout = 90 seconds
+  }
+
+  jibri-sip {
+    // The JID of the MUC to be used as a brewery for jibri instances for SIP.
+    # brewery-jid = "jibrisipbrewery@example.com"
+  }
+
+  jigasi {
+    // The JID of the MUC to be used as a brewery for jigasi instances.
+    # brewery-jid = "jigasibrewery@example.com"
+  }
+
+  // The region in which the machine is running.
+  #local-region="us-east-1"
+
+  octo {
+    // Whether or not to use Octo. Note that when enabled, its use will be determined by
+    // $jicofo.bridge.selection-strategy.
+    enabled = false
+
+    // An identifier of the Jicofo instance, used for the purpose of generating conference IDs unique across a set of
+    // Jicofo instances. Valid values are [1, 65535]. The value 0 is used when none is explicitly configured.
+    id = 1
+  }
+
+  rest {
+    port = 8888
+    tls-port = 8843
+  }
+
+  sctp {
+    // Whether to allocate SCTP channels on the bridge (only when the client advertises support, and SCTP is
+    // enabled in the per-conference configuration).
+    enabled = true
+  }
+
+  task-pools {
+    shared-pool-max-threads = 1500
+  }
+
+  xmpp {
+    // The separate XMPP connection used for communication with clients (endpoints).
+    client {
+      enabled = true
+      hostname = "jitsi-xmpp"
+      port = 5222
+      domain = "auth.jitsi"
+      username = "focus"
+      password = "jicofopass"
+
+      // How long to wait for a response to a stanza before giving up.
+      reply-timeout = 15 seconds
+
+      // The JID/domain of the MUC service used for conferencing.
+      conference-muc-jid = conference.jitsi
+
+      // A flag to suppress the TLS certificate verification.
+      disable-certificate-verification = false
+    }
+    // The separate XMPP connection used for internal services (currently only jitsi-videobridge).
+    service {
+      enabled = false
+      hostname = "jitsi-xmpp"
+      port = 5222
+      domain = "auth.jitsi"
+      username = "focus"
+      password = "jicofopass"
+
+      // How long to wait for a response to a stanza before giving up.
+      reply-timeout = 15 seconds
+
+      // A flag to suppress the TLS certificate verification.
+      disable-certificate-verification = false
+    }
+  }
+}
diff --git a/app/jitsi/integration/jvb/logging.properties b/app/jitsi/integration/jvb/logging.properties
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+++ b/app/jitsi/integration/jvb/logging.properties
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+handlers= java.util.logging.ConsoleHandler
+#handlers= java.util.logging.ConsoleHandler, com.agafua.syslog.SyslogHandler
+#handlers= java.util.logging.ConsoleHandler, io.sentry.jul.SentryHandler
+
+java.util.logging.ConsoleHandler.level = ALL
+java.util.logging.ConsoleHandler.formatter = org.jitsi.utils.logging2.JitsiLogFormatter
+
+net.java.sip.communicator.util.ScLogFormatter.programname=JVB
+
+# default
+.level=INFO
+# for debug 
+#.level=FINE
+
+org.jitsi.videobridge.xmpp.ComponentImpl.level=FINE
+
+# All of the INFO level logs from MediaStreamImpl are unnecessary in the context of jitsi-videobridge.
+org.jitsi.impl.neomedia.MediaStreamImpl.level=WARNING
+
+# Syslog (uncomment handler to use)
+com.agafua.syslog.SyslogHandler.transport = udp
+com.agafua.syslog.SyslogHandler.facility = local0
+com.agafua.syslog.SyslogHandler.port = 514
+com.agafua.syslog.SyslogHandler.hostname = localhost
+com.agafua.syslog.SyslogHandler.formatter = org.jitsi.utils.logging2.JitsiLogFormatter
+com.agafua.syslog.SyslogHandler.escapeNewlines = false
+
+# Sentry (uncomment handler to use)
+io.sentry.jul.SentryHandler.level=WARNING
+
+# to disable double timestamps in syslog uncomment next line
+#net.java.sip.communicator.util.ScLogFormatter.disableTimestamp=true
+
+# time series logging
+java.util.logging.SimpleFormatter.format= %5$s%n
+java.util.logging.FileHandler.level = ALL
+java.util.logging.FileHandler.formatter = java.util.logging.SimpleFormatter
+java.util.logging.FileHandler.pattern = /tmp/jvb-series.log
+java.util.logging.FileHandler.limit = 200000000
+java.util.logging.FileHandler.count = 1
+java.util.logging.FileHandler.append = false
+
+timeseries.level=OFF
+timeseries.org.jitsi.videobridge.cc.allocation.BitrateAllocator.level=ALL
+timeseries.useParentHandlers = false
+# time series logging is disabled by default. Uncomment the line below to enable it.
+#timeseries.handlers = java.util.logging.FileHandler
diff --git a/app/jitsi/integration/jvb/videobridge.conf b/app/jitsi/integration/jvb/videobridge.conf
new file mode 100644
index 0000000..a11edc6
--- /dev/null
+++ b/app/jitsi/integration/jvb/videobridge.conf
@@ -0,0 +1,290 @@
+videobridge {
+  entity-expiration {
+    # If an entity has no activity after this timeout, it is expired
+    timeout=1 minute
+
+    # The interval at which the videobridge will check for expired entities
+    check-interval=${videobridge.entity-expiration.timeout}
+  }
+  health {
+    # The interval between health checks
+    interval=10 seconds
+
+    # The timeout for a health check
+    timeout=30 seconds
+
+    # If performing a health check takes longer than this, it is considered unsuccessful.
+    max-check-duration=3 seconds
+
+    # Whether or not health check failures should be 'sticky'
+    # (i.e. once the bridge becomes unhealthy, it will never
+    # go back to a healthy state)
+    sticky-failures=false
+  }
+  ep-connection-status {
+    # How long we'll wait for an endpoint to *start* sending
+    # data before we consider it 'inactive'
+    first-transfer-timeout=15 seconds
+
+    # How long an endpoint can be 'inactive' before it will
+    # be considered disconnected
+    max-inactivity-limit=3 seconds
+
+    # How often we check endpoint's connectivity status
+    check-interval=500 milliseconds
+  }
+  cc {
+    bwe-change-threshold=0.15
+    thumbnail-max-height-px=180
+    onstage-ideal-height-px=1080
+    onstage-preferred-height-px=360
+    onstage-preferred-framerate=30
+    enable-onstage-video-suspend=false
+    trust-bwe=true
+
+    # How often we check to send probing data
+    padding-period=15ms
+
+    # How often we'll force recalculations of forwarded
+    # streams
+    max-time-between-calculations = 15 seconds
+
+    # A JVB-wide last-n value, observed by all endpoints.  Endpoints
+    # will take the minimum of their setting and this one (-1 implies
+    # no last-n limit)
+    jvb-last-n = -1
+  }
+  # The APIs by which the JVB can be controlled
+  apis {
+    xmpp-client {
+      # The interval at which presence is published in the configured MUCs.
+      presence-interval = ${videobridge.stats.interval}
+
+      configs {
+        unique-xmpp-server {
+          hostname="jitsi-xmpp"
+          domain = "auth.jitsi"
+          username = "jvb"
+          password = "jvbpass"
+          port = 5222
+          muc_jids = "jvbbrewery@internal.auth.jitsi"
+          # The muc_nickname must be unique across all jitsi-videobridge instances
+          muc_nickname = "unique-jvb-server"
+          disable_certificate_verification = false
+        }
+        # example-connection-id {
+        #   For the properties which should be
+        #   filled out here, see MucClientConfiguration
+        # }
+      }
+    }
+    # The COLIBRI REST API
+    rest {
+      enabled = true
+    }
+    jvb-api {
+      enabled = true
+    }
+  }
+  # Configuration of the different REST APIs.
+  # Note that the COLIBRI REST API is configured under videobridge.apis.rest instead.
+  rest {
+    debug {
+      enabled = true
+    }
+    health {
+      enabled = true
+    }
+    shutdown {
+      # Note that the shutdown API requires the COLIBRI API to also be enabled.
+      enabled = false
+    }
+    version {
+      enabled = true
+    }
+  }
+  http-servers {
+    # The HTTP server which hosts services intended for 'public' use
+    # (e.g. websockets for the bridge channel connection)
+    public {
+      # See JettyBundleActivatorConfig in Jicoco for values
+      port = -1
+      tls-port = -1
+    }
+    # The HTTP server which hosts services intended for 'private' use
+    # (e.g. health or debug stats)
+    private {
+      # See JettyBundleActivatorConfig in Jicoco for values
+      host = 127.0.0.1
+    }
+  }
+  octo {
+    # Whether or not Octo is enabled
+    enabled=false
+
+    # A string denoting the 'region' of this JVB.  This region
+    # will be used by Jicofo in the selection of a bridge for
+    # a client by comparing it to the client's region.
+    # Must be set when 'enabled' is true
+    #region="us-west-1"
+
+    # The address on which the Octo relay should bind
+    # Must be set when 'enabled' is true
+    #bind-address=198.51.100.1
+
+    # The port to which the Octo relay should bind
+    bind-port=4096
+
+    # The address which controls the public address which
+    # will be part of the Octo relayId
+    #public-address=198.51.100.1
+
+    # The size of the incoming octo queue. This queue is per-remote-endpoint,
+    # so it matches what we use for local endpoints
+    recv-queue-size=1024
+
+    # The size of the outgoing octo queue.  This is a per-originating-endpoint
+    # queue, so assuming all packets are routed (as they currently are for Octo)
+    # it should be the same size as the transceiver recv queue in
+    # jitsi-media-transform.  Repeating the description from there:
+    #   Assuming 300pps for high-definition, 200pps for standard-definition,
+    #   100pps for low-definition and 50pps for audio, this queue is fed
+    #   650pps, so its size in terms of millis is 1024/650*1000 ~= 1575ms.
+    send-queue-size=1024
+  }
+  load-management {
+    # Whether or not the reducer will be enabled to take actions to mitigate load
+    reducer-enabled = false
+    load-measurements {
+      packet-rate {
+        # The packet rate at which we'll consider the bridge overloaded
+        load-threshold = 50000
+        # The packet rate at which we'll consider the bridge 'underloaded' enough
+        # to start recovery
+        recovery-threshold = 40000
+      }
+    }
+    load-reducers {
+      last-n {
+        # The factor by which we'll reduce the current last-n when trying to reduce load
+        reduction-scale = .75
+        # The factor by which we'll increase the current last-n when trying to recover
+        recover-scale = 1.25
+        # The minimum time in between runs of the last-n reducer to reduce or recover from
+        # load
+        impact-time = 1 minute
+        # The lowest value we'll set for last-n
+        minimum-last-n-value = 0
+        # The highest last-n value we'll enforce.  Once the enforced last-n exceeds this value
+        # we'll remove the limit entirely
+        maximum-enforced-last-n-value = 40
+      }
+    }
+  }
+  sctp {
+    # Whether SCTP data channels are enabled.
+    enabled=true
+  }
+  stats {
+    # Whether periodic collection of statistics is enabled or not. When enabled they are accessible through the REST
+    # API (at `/colibri/stats`), and are available to other modules (e.g. to be pushed to callstats or in a MUC).
+    enabled = true
+
+    # The interval at which stats are gathered.
+    interval = 5 seconds
+
+    # Configuration related to pushing statistics to callstats.io.
+    callstats {
+      # An integer application ID (use 0 to disable pushing stats to callstats).
+      app-id = 0
+
+      # The shared secred to authentication with callstats.io.
+      //app-secret = "s3cret"
+
+      # ID of the key that was used to generate token.
+      //key-id = "abcd"
+
+      # The path to private key file.
+      //key-path = "/etc/jitsi/videobridge/ecpriv.jwk"
+
+      # The ID of the server instance to be used when reporting to callstats.
+      bridge-id = "jitsi"
+
+      # TODO: document
+      //conference-id-prefix = "abcd"
+
+      # The interval at which statististics will be published to callstats. This affects both per-conference and global
+      # statistics.
+      # Note that this value will be overriden if a "callstatsio" transport is defined in the parent "stats" section.
+      interval = ${videobridge.stats.interval}
+    }
+  }
+  websockets {
+    enabled=false
+    server-id="default-id"
+
+    # Optional, even when 'enabled' is set to true
+    # tls=true
+    # Must be set when enabled = true
+    #domain="some-domain"
+  }
+  ice {
+    tcp {
+      # Whether ICE/TCP is enabled.
+      enabled = true
+
+      # The port to bind to for ICE/TCP.
+      port = 8089
+
+      # An optional additional port to advertise.
+      # mapped-port = 8443
+      # Whether to use "ssltcp" or plain "tcp".
+      ssltcp = true
+    }
+
+    udp {
+        # The port for ICE/UDP.
+        port = 10000
+    }
+
+    # An optional prefix to include in STUN username fragments generated by the bridge.
+    #ufrag-prefix = "jvb-123:"
+
+    # Which candidate pairs to keep alive. The accepted values are defined in ice4j's KeepAliveStrategy:
+    # "selected_and_tcp", "selected_only", or "all_succeeded".
+    keep-alive-strategy = "selected_and_tcp"
+
+    # Whether to use the "component socket" feature of ice4j.
+    use-component-socket = true
+
+    # Whether to attempt DNS resolution for remote candidates that contain a non-literal address. When set to 'false'
+    # such candidates will be ignored.
+    resolve-remote-candidates = false
+
+    # The nomination strategy to use for ICE.  THe accepted values are defined in ice4j's NominationStrategy:
+    # "NominateFirstValid", "NominateHighestPriority", "NominateFirstHostOrReflexiveValid", or "NominateBestRTT".
+    nomination-strategy = "NominateFirstValid"
+  }
+
+  transport {
+    send {
+      # The size of the dtls-transport outgoing queue. This is a per-participant
+      # queue. Packets from the egress end-up in this queue right before
+      # transmission by the outgoing srtp pipeline (which mainly consists of the
+      # packet sender).
+      #
+      # Its size needs to be of the same order of magnitude as the rtp sender
+      # queue. In a 100 participant call, assuming 300pps for the on-stage and
+      # 100pps for low-definition, last-n 20 and 2 participants talking, so
+      # 2*50pps for audio, this queue is fed 300+19*100+2*50 = 2300pps, so its
+      # size in terms of millis is 1024/2300*1000 ~= 445ms.
+      queue-size=1024
+    }
+  }
+
+  version {
+    // Wheather to announe the jitsi-videobridge version to clients in the ServerHello message.
+    announce = false
+  }
+}
+
diff --git a/app/jitsi/integration/meet/config.js b/app/jitsi/integration/meet/config.js
new file mode 100644
index 0000000..04414c3
--- /dev/null
+++ b/app/jitsi/integration/meet/config.js
@@ -0,0 +1,773 @@
+/* eslint-disable no-unused-vars, no-var */
+
+var config = {
+    // Connection
+    //
+
+    hosts: {
+        // XMPP domain.
+        domain: 'jitsi',
+
+        // When using authentication, domain for guest users.
+        // anonymousdomain: 'guest.example.com',
+
+        // Domain for authenticated users. Defaults to <domain>.
+        // authdomain: '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'
+    },
+
+    // BOSH URL. FIXME: use XEP-0156 to discover it.
+    bosh: '//rayonx.machine.deuxfleurs.fr/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
+    // Do not change username - FIXME: Make focus username configurable
+    // https://github.com/jitsi/jitsi-meet/issues/7376
+    // 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
+
+        // Enable callstats only for a percentage of users.
+        // This takes a value between 0 and 100 which determines the probability for
+        // the callstats to be enabled.
+        // callStatsThreshold: 5 // enable callstats for 5% of the users.
+    },
+
+    // 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 show a "Save Logs" link in the GSM popover that can be
+    // used to collect debug information (XMPP IQs, SDP offer/answer cycles)
+    // about the call.
+    // enableSaveLogs: false,
+
+    // 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
+    // opusMaxAverageBitrate: 20000,
+
+    // Enables support for opus-red (redundancy for Opus).
+    // enableOpusRed: false
+
+    // Video
+
+    // Sets the preferred resolution (height) for local video. Defaults to 720.
+    // resolution: 720,
+
+    // 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
+    //         }
+    //     }
+    // },
+
+    // Enable / disable simulcast support.
+    // disableSimulcast: false,
+
+    // Enable / disable layer suspension.  If enabled, endpoints whose HD
+    // layers are not in use will be suspended (no longer sent) until they
+    // are requested again.
+    // enableLayerSuspension: false,
+
+    // Every participant after the Nth will start video muted.
+    // startVideoMuted: 10,
+
+    // Start calls with video muted. Unlike the option above, this one is only
+    // applied locally. FIXME: having these 2 options is confusing.
+    // startWithVideoMuted: false,
+
+    // If set to true, prefer to use the H.264 video codec (if supported).
+    // Note that it's not recommended to do this because simulcast is not
+    // supported when  using H.264. For 1-to-1 calls this setting is enabled by
+    // default and can be toggled in the p2p section.
+    // This option has been deprecated, use preferredCodec under videoQuality section instead.
+    // 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: 5,
+    //     max: 5
+    // },
+
+    // Try to start calls with screen-sharing instead of camera video.
+    // startScreenSharing: false,
+
+    // Recording
+
+    // Whether to enable file recording or not.
+    // fileRecordingsEnabled: false,
+    // Enable the dropbox integration.
+    // dropbox: {
+    //     appKey: '<APP_KEY>' // Specify your app key here.
+    //     // A URL to redirect the user to, after authenticating
+    //     // by default uses:
+    //     // 'https://jitsi-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,
+
+    // Provides a way to use different "last N" values based on the number of participants in the conference.
+    // The keys in an Object represent number of participants and the values are "last N" to be used when number of
+    // participants gets to or above the number.
+    //
+    // For the given example mapping, "last N" will be set to 20 as long as there are at least 5, but less than
+    // 29 participants in the call and it will be lowered to 15 when the 30th participant joins. The 'channelLastN'
+    // will be used as default until the first threshold is reached.
+    //
+    // lastNLimits: {
+    //     5: 20,
+    //     30: 15,
+    //     50: 10,
+    //     70: 5,
+    //     90: 2
+    // },
+
+    // Specify the settings for video quality optimizations on the client.
+    // videoQuality: {
+    //    // Provides a way to prevent a video codec from being negotiated on the JVB connection. The codec specified
+    //    // here will be removed from the list of codecs present in the SDP answer generated by the client. If the
+    //    // same codec is specified for both the disabled and preferred option, the disable settings will prevail.
+    //    // Note that 'VP8' cannot be disabled since it's a mandatory codec, the setting will be ignored in this case.
+    //    disabledCodec: 'H264',
+    //
+    //    // Provides a way to set a preferred video codec for the JVB connection. If 'H264' is specified here,
+    //    // simulcast will be automatically disabled since JVB doesn't support H264 simulcast yet. This will only
+    //    // rearrange the the preference order of the codecs in the SDP answer generated by the browser only if the
+    //    // preferred codec specified here is present. Please ensure that the JVB offers the specified codec for this
+    //    // to take effect.
+    //    preferredCodec: 'VP8',
+    //
+    //    // Provides a way to configure the maximum bitrates that will be enforced on the simulcast streams for
+    //    // video tracks. The keys in the object represent the type of the stream (LD, SD or HD) and the values
+    //    // are the max.bitrates to be set on that particular type of stream. The actual send may vary based on
+    //    // the available bandwidth calculated by the browser, but it will be capped by the values specified here.
+    //    // This is currently not implemented on app based clients on mobile.
+    //    maxBitratesVideo: {
+    //        low: 200000,
+    //        standard: 500000,
+    //        high: 1500000
+    //    },
+    //
+    //    // The options can be used to override default thresholds of video thumbnail heights corresponding to
+    //    // the video quality levels used in the application. At the time of this writing the allowed levels are:
+    //    //     'low' - for the low quality level (180p at the time of this writing)
+    //    //     'standard' - for the medium quality level (360p)
+    //    //     'high' - for the high quality level (720p)
+    //    // The keys should be positive numbers which represent the minimal thumbnail height for the quality level.
+    //    //
+    //    // With the default config value below the application will use 'low' quality until the thumbnails are
+    //    // at least 360 pixels tall. If the thumbnail height reaches 720 pixels then the application will switch to
+    //    // the high quality.
+    //    minHeightForQualityLvl: {
+    //        360: 'standard',
+    //        720: 'high'
+    //    },
+    //
+    //    // Provides a way to resize the desktop track to 720p (if it is greater than 720p) before creating a canvas
+    //    // for the presenter mode (camera picture-in-picture mode with screenshare).
+    //    resizeDesktopForPresenter: false
+    // },
+
+    // // 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 support in this client (default: enabled).
+    // enableTcc: true,
+
+    // Disables or enables REMB support in this client (default: enabled).
+    // enableRemb: true,
+
+    // 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,
+
+    // 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
+
+    // UI
+    //
+
+    // Disables responsive tiles.
+    // disableResponsiveTiles: false,
+
+    // Hides lobby button
+    // hideLobbyButton: false,
+
+    // Require users to always specify a display name.
+    // requireDisplayName: true,
+
+    // Whether to use a welcome page or not. In case it's false a random room
+    // will be joined when no room is specified.
+    enableWelcomePage: true,
+
+    // Disable app shortcuts that are registered upon joining a conference
+    // disableShortcuts: false,
+
+    // Disable initial browser getUserMedia requests.
+    // This is useful for scenarios where users might want to start a conference for screensharing only
+    // disableInitialGUM: 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: 'fr',
+
+    // Disables profile and the edit of all fields from the profile settings (display name and email)
+    // disableProfile: false,
+
+    // Whether or not some features are checked based on token.
+    // enableFeaturesBasedOnToken: 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 etherpad integration is enabled, setting this to true will
+    // automatically open the etherpad when a participant joins.  This
+    // does not affect the mobile app since opening an etherpad
+    // obscures the conference controls -- it's better to let users
+    // choose to open the pad on their own in that case.
+    // openSharedDocumentOnJoin: 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,
+
+    // Whether to automatically copy invitation URL after creating a room.
+    // Document should be focused for this option to work
+    // enableAutomaticUrlCopy: false,
+
+    // Base URL for a Gravatar-compatible service. Defaults to libravatar.
+    // gravatarBaseURL: 'https://seccdn.libravatar.org/avatar/';
+
+    // 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,
+
+        // 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). This setting is deprecated, use preferredCodec instead.
+        // preferH264: true
+
+        // Provides a way to set the video codec preference on the p2p connection. Acceptable
+        // codec values are 'VP8', 'VP9' and 'H264'.
+        // preferredCodec: 'H264',
+
+        // If set to true, disable H.264 video codec by stripping it out of the
+        // SDP. This setting is deprecated, use disabledCodec instead.
+        // disableH264: false,
+
+        // Provides a way to prevent a video codec from being negotiated on the p2p connection.
+        // disabledCodec: '',
+
+        // 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:
+        // By enabling rtcstats server every time a conference is joined the rtcstats
+        // module connects to the provided rtcstatsEndpoint and sends statistics regarding
+        // PeerConnection states along with getStats metrics polled at the specified
+        // interval.
+        // rtcstatsEnabled: true,
+
+        // 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"
+        // ],
+    },
+
+    // Logs that should go be passed through the 'log' event if a handler is defined for it
+    // apiLogLevels: ['warn', 'log', 'error', 'info', 'debug'],
+
+    // 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,
+
+    // Enables support for lip-sync for this client (if the browser supports it).
+    // enableLipSync: false
+
+    /**
+     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'
+     }
+    */
+    // dynamicBrandingUrl: '',
+
+    // 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',
+
+    // If true, tile view will not be enabled automatically when the participants count threshold is reached.
+    // disableTileView: true,
+
+    // Hides the conference subject
+    // hideConferenceSubject: true
+
+    // Hides the conference timer.
+    // hideConferenceTimer: true,
+
+    // Hides the participants stats
+    // hideParticipantsStats: true
+
+    // Sets the conference subject
+    // subject: 'Conference Subject',
+
+    // List of undocumented settings used in jitsi-meet
+    /**
+     _immediateReloadThreshold
+     debug
+     debugAudioLevels
+     deploymentInfo
+     dialInConfCodeUrl
+     dialInNumbersUrl
+     dialOutAuthUrl
+     dialOutCodesUrl
+     disableRemoteControl
+     displayJids
+     etherpad_base
+     externalConnectUrl
+     firefox_fake_device
+     googleApiApplicationClientID
+     iAmRecorder
+     iAmSipGateway
+     microsoftApiApplicationClientID
+     peopleSearchQueryTypes
+     peopleSearchUrl
+     requireDisplayName
+     tokenAuthUrl
+     */
+
+    /**
+     * This property can be used to alter the generated meeting invite links (in combination with a branding domain
+     * which is retrieved internally by jitsi meet) (e.g. https://meet.jit.si/someMeeting
+     * can become https://brandedDomain/roomAlias)
+     */
+    // brandingRoomAlias: null,
+
+    // List of undocumented settings used in lib-jitsi-meet
+    /**
+     _peerConnStatusOutOfLastNTimeout
+     _peerConnStatusRtcMuteTimeout
+     abTesting
+     avgRtpStatsN
+     callStatsConfIDNamespace
+     callStatsCustomScriptUrl
+     desktopSharingSources
+     disableAEC
+     disableAGC
+     disableAP
+     disableHPF
+     disableNS
+     enableTalkWhileMuted
+     forceJVB121Ratio
+     forceTurnRelay
+     hiddenDomain
+     ignoreStartMuted
+     websocketKeepAlive
+     websocketKeepAliveUrl
+     */
+
+    /**
+        Use this array to configure which notifications will be shown to the user
+        The items correspond to the title or description key of that notification
+        Some of these notifications also depend on some other internal logic to be displayed or not,
+        so adding them here will not ensure they will always be displayed
+
+        A falsy value for this prop will result in having all notifications enabled (e.g null, undefined, false)
+    */
+    // notifications: [
+    //     'connection.CONNFAIL', // shown when the connection fails,
+    //     'dialog.cameraNotSendingData', // shown when there's no feed from user's camera
+    //     'dialog.kickTitle', // shown when user has been kicked
+    //     'dialog.liveStreaming', // livestreaming notifications (pending, on, off, limits)
+    //     'dialog.lockTitle', // shown when setting conference password fails
+    //     'dialog.maxUsersLimitReached', // shown when maximmum users limit has been reached
+    //     'dialog.micNotSendingData', // shown when user's mic is not sending any audio
+    //     'dialog.passwordNotSupportedTitle', // shown when setting conference password fails due to password format
+    //     'dialog.recording', // recording notifications (pending, on, off, limits)
+    //     'dialog.remoteControlTitle', // remote control notifications (allowed, denied, start, stop, error)
+    //     'dialog.reservationError',
+    //     'dialog.serviceUnavailable', // shown when server is not reachable
+    //     'dialog.sessTerminated', // shown when there is a failed conference session
+    //     'dialog.tokenAuthFailed', // show when an invalid jwt is used
+    //     'dialog.transcribing', // transcribing notifications (pending, off)
+    //     'dialOut.statusMessage', // shown when dial out status is updated.
+    //     'liveStreaming.busy', // shown when livestreaming service is busy
+    //     'liveStreaming.failedToStart', // shown when livestreaming fails to start
+    //     'liveStreaming.unavailableTitle', // shown when livestreaming service is not reachable
+    //     'lobby.joinRejectedMessage', // shown when while in a lobby, user's request to join is rejected
+    //     'lobby.notificationTitle', // shown when lobby is toggled and when join requests are allowed / denied
+    //     'localRecording.localRecording', // shown when a local recording is started
+    //     'notify.disconnected', // shown when a participant has left
+    //     'notify.grantedTo', // shown when moderator rights were granted to a participant
+    //     'notify.invitedOneMember', // shown when 1 participant has been invited
+    //     'notify.invitedThreePlusMembers', // shown when 3+ participants have been invited
+    //     'notify.invitedTwoMembers', // shown when 2 participants have been invited
+    //     'notify.kickParticipant', // shown when a participant is kicked
+    //     'notify.mutedRemotelyTitle', // shown when user is muted by a remote party
+    //     'notify.mutedTitle', // shown when user has been muted upon joining,
+    //     'notify.newDeviceAudioTitle', // prompts the user to use a newly detected audio device
+    //     'notify.newDeviceCameraTitle', // prompts the user to use a newly detected camera
+    //     'notify.passwordRemovedRemotely', // shown when a password has been removed remotely
+    //     'notify.passwordSetRemotely', // shown when a password has been set remotely
+    //     'notify.raisedHand', // shown when a partcipant used raise hand,
+    //     'notify.startSilentTitle', // shown when user joined with no audio
+    //     'prejoin.errorDialOut',
+    //     'prejoin.errorDialOutDisconnected',
+    //     'prejoin.errorDialOutFailed',
+    //     'prejoin.errorDialOutStatus',
+    //     'prejoin.errorStatusCode',
+    //     'prejoin.errorValidation',
+    //     'recording.busy', // shown when recording service is busy
+    //     'recording.failedToStart', // shown when recording fails to start
+    //     'recording.unavailableTitle', // shown when recording service is not reachable
+    //     'toolbar.noAudioSignalTitle', // shown when a broken mic is detected
+    //     'toolbar.noisyAudioInputTitle', // shown when noise is detected for the current microphone
+    //     'toolbar.talkWhileMutedPopup', // shown when user tries to speak while muted
+    //     'transcribing.failedToStart' // shown when transcribing fails to start
+    // ]
+
+    // 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 */
diff --git a/app/jitsi/integration/meet/nginx.conf b/app/jitsi/integration/meet/nginx.conf
new file mode 100644
index 0000000..16a63f9
--- /dev/null
+++ b/app/jitsi/integration/meet/nginx.conf
@@ -0,0 +1,72 @@
+# some doc: https://www.nginx.com/resources/wiki/start/topics/examples/full/
+error_log  /dev/stderr;
+
+events {}
+
+http {
+  ##
+  # Basic Settings
+  ##
+  sendfile on;
+  tcp_nopush on;
+  tcp_nodelay on;
+  keepalive_timeout 65;
+  types_hash_max_size 2048;
+
+
+  # mimetypes, required by jitsi!
+  include /etc/nginx/mime.types;
+  default_type application/octet-stream;
+
+  types {
+    application/wasm wasm;
+  }
+
+  ##
+  # SSL Settings
+  ##
+
+  ssl_protocols TLSv1 TLSv1.1 TLSv1.2; # Dropping SSLv3, ref: POODLE
+  ssl_prefer_server_ciphers on;
+       
+  ##
+  # Gzip Settings
+  ##
+  gzip on;
+
+  access_log /dev/stdout;
+  server_names_hash_bucket_size 64;
+
+  server {
+    listen 0.0.0.0:443 ssl http2 default_server;
+    listen [::]:443 ssl http2 default_server;
+    server_name _;
+    ssl_certificate /etc/nginx/jitsi.crt;
+    ssl_certificate_key /etc/nginx/jitsi.key;
+    root /srv/jitsi-meet;
+    index index.html;
+
+    # lot of work would be needed to improve location rules
+    #  - in order to allow - and _ in the URL, even space
+    #  - while not shadowing other files (.js and following locations)
+    #  - passed some times twice on the problem, not as easy as it seems
+    location ~ ^/([a-zA-Z0-9=\?]+)$ {
+        rewrite ^/(.*)$ / break;
+    }
+    location / {
+        ssi on;
+    }
+
+    location /external_api.js {
+        alias /srv/jitsi-meet/libs/external_api.min.js;
+    }
+
+    location /http-bind {
+        proxy_pass      http://jitsi-xmpp:5280/http-bind;
+        proxy_set_header X-Forwarded-For \$remote_addr;
+        proxy_set_header Host \$http_host;
+    }
+
+    
+  }
+}
diff --git a/app/jitsi/integration/prosody/prosody.cfg.lua b/app/jitsi/integration/prosody/prosody.cfg.lua
new file mode 100644
index 0000000..b5bc0b9
--- /dev/null
+++ b/app/jitsi/integration/prosody/prosody.cfg.lua
@@ -0,0 +1,137 @@
+modules_enabled = {
+  "roster"; -- Allow users to have a roster. Recommended ;)
+  "saslauth"; -- Authentication for clients and servers. Recommended if you want to log in.
+  "tls"; -- Add support for secure TLS on c2s/s2s connections
+  "dialback"; -- s2s dialback support
+  "disco"; -- Service discovery
+  "posix"; -- POSIX functionality, sends server to background, enables syslog, etc.
+  "version"; -- Replies to server version requests
+  "uptime"; -- Report how long server has been running
+  "time"; -- Let others know the time here on this server
+  "ping"; -- Replies to XMPP pings with pongs
+  "pep"; -- Enables users to publish their mood, activity, playing music and more
+  -- jitsi
+  --"smacks"; -- not shipped with prosody
+  "carbons";
+  "mam";
+  "lastactivity";
+  "offline";
+  "pubsub";
+  "adhoc";
+  "websocket";
+  --"http_altconnect"; -- not shipped with prosody
+}
+modules_disabled = { "s2s" }
+
+plugin_paths = { "/usr/share/jitsi-meet/prosody-plugins/" }
+
+log = {
+  --log less on console with warn="*console"; or err="*console" or more with debug="*console"
+  info="*console";
+}
+daemonize = false
+use_libevent = true
+
+-- domain mapper options, must at least have domain base set to use the mapper
+muc_mapper_domain_base = "jitsi.deuxfleurs.fr";
+
+--@FIXME would be great to configure it
+--turncredentials_secret = "__turnSecret__";
+
+--turncredentials = {
+--    { type = "stun", host = "jitmeet.example.com", port = "3478" },
+--    { type = "turn", host = "jitmeet.example.com", port = "3478", transport = "udp" },
+--    { type = "turns", host = "jitmeet.example.com", port = "5349", transport = "tcp" }
+--};
+
+cross_domain_bosh = false;
+consider_bosh_secure = true;
+--component_ports = { 5347 }
+component_ports = { } -- it seems we don't need external components for now...
+https_ports = { } -- we don't need http
+http_ports = { 5280 }
+c2s_ports = { 5222 }
+s2s_ports = { }
+
+
+-- https://ssl-config.mozilla.org/#server=haproxy&version=2.1&config=intermediate&openssl=1.1.0g&guideline=5.4
+ssl = {
+    protocol = "tlsv1_2+";
+    ciphers = "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"
+}
+
+VirtualHost "jitsi"
+    enabled = true -- Remove this line to enable this host
+    authentication = "anonymous"
+    -- Properties below are modified by jitsi-meet-tokens package config
+    -- and authentication above is switched to "token"
+    --app_id="example_app_id"
+    --app_secret="example_app_secret"
+    -- Assign this host a certificate for TLS, otherwise it would use the one
+    -- set in the global section (if any).
+    -- Note that old-style SSL on port 5223 only supports one certificate, and will always
+    -- use the global one.
+    ssl = {
+        key = "/var/lib/prosody/jitsi.key";
+        certificate = "/var/lib/prosody/jitsi.crt";
+    }
+    speakerstats_component = "speakerstats.jitsi"
+    conference_duration_component = "conferenceduration.jitsi"
+    -- we need bosh
+    modules_enabled = {
+        "bosh";
+        "pubsub";
+        "ping"; -- Enable mod_ping
+        "speakerstats";
+        --"turncredentials"; not supported yet
+        "conference_duration";
+        "muc_lobby_rooms";
+    }
+    c2s_require_encryption = false
+    lobby_muc = "lobby.jitsi"
+    main_muc = "conference.jitsi"
+    -- muc_lobby_whitelist = { "recorder.jitmeet.example.com" } -- Here we can whitelist jibri to enter lobby enabled rooms
+
+Component "conference.jitsi" "muc"
+    storage = "memory"
+    modules_enabled = {
+        "muc_meeting_id";
+        "muc_domain_mapper";
+        --"token_verification";
+    }
+    admins = { "focus@auth.jitsi" }
+    muc_room_locking = false
+    muc_room_default_public_jids = true
+
+-- internal muc component
+Component "internal.auth.jitsi" "muc"
+    storage = "memory"
+    modules_enabled = {
+        "ping";
+    }
+    admins = { "focus@auth.jitsi", "jvb@auth.jitsi" }
+    muc_room_locking = false
+    muc_room_default_public_jids = true
+
+VirtualHost "auth.jitsi"
+    ssl = {
+        key = "/var/lib/prosody/auth.jitsi.key";
+        certificate = "/var/lib/prosody/auth.jitsi.crt";
+    }
+    authentication = "internal_plain"
+
+Component "focus.jitsi" "client_proxy"
+    target_address = "focus@auth.jitsi"
+
+Component "speakerstats.jitsi" "speakerstats_component"
+    muc_component = "conference.jitsi"
+
+Component "conferenceduration.jitsi" "conference_duration_component"
+    muc_component = "conference.jitsi"
+
+Component "lobby.jitsi" "muc"
+    storage = "memory"
+    restrict_room_creation = true
+    muc_room_locking = false
+    muc_room_default_public_jids = true
+
diff --git a/app/jitsi/integration/prosody/prosody.cfg.lua.back b/app/jitsi/integration/prosody/prosody.cfg.lua.back
new file mode 100644
index 0000000..d03d7c9
--- /dev/null
+++ b/app/jitsi/integration/prosody/prosody.cfg.lua.back
@@ -0,0 +1,64 @@
+daemonize = false
+allow_registration = false
+use_libevent = true
+component_interface = "0.0.0.0"
+component_ports = { 5347 }
+http_ports = { 5280 }
+https_ports = {}
+
+-- Not sure all modules are required
+modules_enabled = {
+  "roster"; -- Allow users to have a roster. Recommended ;)
+  "saslauth"; -- Authentication for clients and servers. Recommended if you want to log in.
+  "tls"; -- Add support for secure TLS on c2s/s2s connections
+  "dialback"; -- s2s dialback support
+  "disco"; -- Service discovery
+  "posix"; -- POSIX functionality, sends server to background, enables syslog, etc.
+  "version"; -- Replies to server version requests
+  "uptime"; -- Report how long server has been running
+  "time"; -- Let others know the time here on this server
+  "ping"; -- Replies to XMPP pings with pongs
+  "pep"; -- Enables users to publish their mood, activity, playing music and more
+  -- jitsi
+  --"smacks"; -- not shipped with prosody
+  "carbons";
+  "mam";
+  "lastactivity";
+  "offline";
+  "pubsub";
+  "adhoc";
+  "websocket";
+  --"http_altconnect"; -- not shipped with prosody
+}
+
+log = {
+  --log less on console with warn="*console"; or err="*console" or more with debug="*console"
+  debug="*console";
+}
+
+VirtualHost "jitsi"
+  authentication = "anonymous"
+  ssl = {
+    key = "/var/lib/prosody/jitsi.key";
+    certificate = "/var/lib/prosody/jitsi.crt";
+  }
+  modules_enabled = {
+    "bosh";
+    "pubsub";
+  }
+  c2s_require_encryption = false
+
+VirtualHost "auth.jitsi"
+  ssl = {
+    key = "/var/lib/prosody/auth.jitsi.key";
+    certificate = "/var/lib/prosody/auth.jitsi.crt";
+  }
+  authentication = "internal_plain"
+  admins = { "focus@auth.jitsi"}
+
+Component "conference.jitsi" "muc"
+Component "internal.auth.jitsi" "muc"
+  storage = "memory"
+  modules_enabled = { "ping"; }
+  admins = { "focus@auth.jitsi", "jvb@auth.jitsi" }
+