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jitsi / jitsi-videobridge

Jitsi Videobridge is a WebRTC compatible video router or SFU that lets build highly scalable video conferencing infrastructure (i.e., up to hundreds of conferences per server).
https://jitsi.org/jitsi-videobridge
Apache License 2.0
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Jigasi is not working after Updating from jitsi-videobridge2=2.2-45* to jitsi-videobridge2_2.2-61* #1973

Closed holzi1005 closed 1 year ago

holzi1005 commented 1 year ago

We run a jitsi shard with two videobridges and one jicofo,prosody server together with a separat jigasi server. Before updating the videobridge to the new version, jigasi was able to call into a conference and the caller could talk and hear others.

After the update, after the caller joint the conference, jigasi stoped the stream and hang up after appr. 5 seconds, hence the caller is leving the meeting.

Steps to reproduce

Upgrade all services from Jitsi to the latest version. Keep the old configuration and restart all services.

Environment details

Server 1: jitsi-meet-web 1.0.6776-1 jitsi-videobridge2_2.2-61-g98c9f868-1 / jitsi-videobridge2 2.2-45-ge8b20f06-1 prosody 0.11.9-2+deb11u2 jicofo 1.0-954-1

Server 2: jigasi 1.1-231-gd792d0f-1 asterisk 1:18.10.0~dfsg+~cs6.10.40431411-2

Configuration

Config Jigasi

#Sample config with one XMPP and one SIP account configured
# Replace {sip-pass-hash} with SIP user password hash
# as well as other account properties

# Name of default JVB room that will be joined if no special header is included
# in SIP invite
# org.jitsi.jigasi.DEFAULT_JVB_ROOM_NAME=siptest@conference.xxxxxxxxxxxxxxx

org.jitsi.jigasi.MUC_SERVICE_ADDRESS=conference.xxxxxxxxxxxxxxx
net.java.sip.communicator.impl.protocol.SingleCallInProgressPolicy.enabled=false
org.jitsi.impl.neomedia.transform.csrc.CsrcTransformEngine.DISCARD_CONTRIBUTING_SOURCES=true
# Should be enabled when using translator mode
#net.java.sip.communicator.impl.neomedia.audioSystem.audiosilence.captureDevice_list=["AudioSilenceCaptureDevice:noTransferData"]

net.java.sip.communicator.plugin.reconnectplugin.ATLEAST_ONE_SUCCESSFUL_CONNECTION.acc1=true
net.java.sip.communicator.impl.protocol.sip.SKIP_REINVITE_ON_FOCUS_CHANGE_PROP=true

# Adjust opus encoder complexity
net.java.sip.communicator.impl.neomedia.codec.audio.opus.encoder.COMPLEXITY=10

# Disables packet logging
net.java.sip.communicator.packetlogging.PACKET_LOGGING_ENABLED=false

net.java.sip.communicator.impl.protocol.sip.acc1=acc1
net.java.sip.communicator.impl.protocol.sip.acc1.ACCOUNT_UID=xxxxxxxxxxxxxxx
net.java.sip.communicator.impl.protocol.sip.acc1.PASSWORD=xxxxxxxxxxxxxxx
net.java.sip.communicator.impl.protocol.sip.acc1.PROTOCOL_NAME=SIP
net.java.sip.communicator.impl.protocol.sip.acc1.SERVER_ADDRESS=xxxxxxxxxxxxxxx
net.java.sip.communicator.impl.protocol.sip.acc1.USER_ID=xxxxxxxxxxxxxxx
net.java.sip.communicator.impl.protocol.sip.acc1.KEEP_ALIVE_INTERVAL=25
net.java.sip.communicator.impl.protocol.sip.acc1.KEEP_ALIVE_METHOD=OPTIONS
net.java.sip.communicator.impl.protocol.sip.acc1.VOICEMAIL_ENABLED=false
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.AMR-WB/16000=750
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.G722/16000=710
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.G722/8000=700
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.ALAW/8000=690
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.ALAW/16000=680
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.ULAW/8000=670
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.ULAW/16000=660
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.GSM/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.H263-1998/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.H264/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.PCMA/8000=600
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.PCMU/8000=650
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.SILK/12000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.SILK/16000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.SILK/24000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.SILK/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.VP8/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.iLBC/8000=10
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.opus/48000=1000
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.red/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.speex/16000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.speex/32000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.speex/8000=0
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.telephone-event/8000=1
net.java.sip.communicator.impl.protocol.sip.acc1.Encodings.ulpfec/90000=0
net.java.sip.communicator.impl.protocol.sip.acc1.OVERRIDE_ENCODINGS=true
net.java.sip.communicator.impl.protocol.sip.acc1.DEFAULT_ENCRYPTION=false

# If an authenticated (hidden) domain is used to connect to a conference,
# PREVENT_AUTH_LOGIN will prevent the SIP participant from being seen as a
# hidden participant in the conference
#net.java.sip.communicator.impl.protocol.sip.acc1.PREVENT_AUTH_LOGIN=FALSE

# Used when incoming calls are used in multidomain environment, used to detect subdomains
# used for constructing callResource and eventually contacting jicofo
net.java.sip.communicator.impl.protocol.sip.acc1.DOMAIN_BASE=xxxxxxxxxxxxxxx
net.java.sip.communicator.impl.protocol.sip.acc1.JITSI_AUTH_TOKEN_HEADER_NAME=Jitsi-Auth-Token
net.java.sip.communicator.impl.protocol.sip.acc1.JITSI_MEET_ROOM_HEADER_NAME=Jitsi-Conference-Room
net.java.sip.communicator.impl.protocol.sip.acc1.JITSI_MEET_DOMAIN_BASE_HEADER_NAME=Jitsi-Domain-Header

# the pattern to be used as bosh url when using bosh in multidomain environment
net.java.sip.communicator.impl.protocol.sip.acc1.BOSH_URL_PATTERN=xxxxxxxxxxxxxxx

# can be enabled to disable audio mixing and use translator, jigasi will act as jvb, just forward every ssrc stream it receives.
#net.java.sip.communicator.impl.protocol.sip.acc1.USE_TRANSLATOR_IN_CONFERENCE=true

# we can receive dial/hangup only from the control muc
org.jitsi.jigasi.ALLOWED_JID=JigasiBrewery@internal.auth.xxxxxxxxxxxxxxx

org.jitsi.jigasi.BREWERY_ENABLED=true

# We can use the prefix org.jitsi.jigasi.xmpp.acc to override any of the
# properties that will be used for creating xmpp account for communication.

# The following two props assume we are using jigasi on the same machine as
# the xmpp server.
org.jitsi.jigasi.xmpp.acc.IS_SERVER_OVERRIDDEN=true
org.jitsi.jigasi.xmpp.acc.SERVER_ADDRESS=xxxxxxxxxxxxxxx
org.jitsi.jigasi.xmpp.acc.VIDEO_CALLING_DISABLED=true
org.jitsi.jigasi.xmpp.acc.JINGLE_NODES_ENABLED=false
org.jitsi.jigasi.xmpp.acc.AUTO_DISCOVER_STUN=false
org.jitsi.jigasi.xmpp.acc.IM_DISABLED=true
org.jitsi.jigasi.xmpp.acc.SERVER_STORED_INFO_DISABLED=true
org.jitsi.jigasi.xmpp.acc.IS_FILE_TRANSFER_DISABLED=true
org.jitsi.jigasi.xmpp.acc.KEEP_ALIVE_METHOD=XEP-0199
org.jitsi.jigasi.xmpp.acc.KEEP_ALIVE_INTERVAL=30
org.jitsi.jigasi.xmpp.acc.USE_DEFAULT_STUN_SERVER=false

# Or you can use bosh for the connection establishment by specifing the URL to use.
# Used when outgoing calls are used in multidomain environment, used to detect subdomains
# org.jitsi.jigasi.xmpp.acc.DOMAIN_BASE=xxxxxxxxxxxxxxx
org.jitsi.jigasi.xmpp.acc.BOSH_URL_PATTERN=xxxxxxxxxxxxxxx

# can be enabled to disable audio mixing and use translator, jigasi will act as jvb, just forward every ssrc stream it receives.
# org.jitsi.jigasi.xmpp.acc.USE_TRANSLATOR_IN_CONFERENCE=true

# If you want jigasi to perform authenticated login instead of anonymous login
# to the XMPP server, you can set the following properties.
# org.jitsi.jigasi.xmpp.acc.USER_ID=jigasi@auth.xxxxxxxxxxxxxxx
# org.jitsi.jigasi.xmpp.acc.PASS=xxxxxxxxxxxxxxx
# org.jitsi.jigasi.xmpp.acc.ANONYMOUS_AUTH=false

# To fix SSL/TLS required by client but not supported by server
org.jitsi.jigasi.xmpp.acc.ALLOW_NON_SECURE=true

# If you want to disconnect jigasi calls automatically when all web users have
# left, you can set the following property to false.
# org.jitsi.jigasi.ALLOW_ONLY_JIGASIS_IN_ROOM=true

# If you want to use the SIP user part of the incoming/outgoing call SIP URI
# you can set the following property to true.
# org.jitsi.jigasi.USE_SIP_USER_AS_XMPP_RESOURCE=true

# Activate this property if you are using self-signed certificates or other
# type of non-trusted certicates. In this mode your service trust in the
# remote certificates always.
net.java.sip.communicator.service.gui.ALWAYS_TRUST_MODE_ENABLED=true

# Enable this property to be able to shutdown gracefully jigasi using
# a rest command
# org.jitsi.jigasi.ENABLE_REST_SHUTDOWN=true

# Options regarding Transcription. Read the README for a detailed description
# about each property

#org.jitsi.jigasi.ENABLE_TRANSCRIPTION=false
#org.jitsi.jigasi.ENABLE_SIP=true

# whether to use the more expensive, but better performing
# "video" model when doing transcription
# org.jitsi.jigasi.transcription.USE_VIDEO_MODEL = false

# delivering final transcript
# org.jitsi.jigasi.transcription.DIRECTORY=/var/lib/jigasi/transcripts
# org.jitsi.jigasi.transcription.BASE_URL=http://localhost/
# org.jitsi.jigasi.transcription.jetty.port=-1
# org.jitsi.jigasi.transcription.ADVERTISE_URL=false

# save formats
# org.jitsi.jigasi.transcription.SAVE_JSON=false
# org.jitsi.jigasi.transcription.SAVE_TXT=true

# send formats
# org.jitsi.jigasi.transcription.SEND_JSON=true
# org.jitsi.jigasi.transcription.SEND_TXT=false

# Vosk server
# org.jitsi.jigasi.transcription.customService=org.jitsi.jigasi.transcription.VoskTranscriptionService
# org.jitsi.jigasi.transcription.vosk.websocket_url=ws://localhost:2700

# translation
# org.jitsi.jigasi.transcription.ENABLE_TRANSLATION=false

# record audio. Currently only wav format is supported
# org.jitsi.jigasi.transcription.RECORD_AUDIO=false
# org.jitsi.jigasi.transcription.RECORD_AUDIO_FORMAT=wav

# execute one or more scripts when a transcript or recording is saved
# org.jitsi.jigasi.transcription.EXECUTE_SCRIPTS=true
# org.jitsi.jigasi.transcription.SCRIPTS_TO_EXECUTE_LIST_SEPARATOR=","
# org.jitsi.jigasi.transcription.SCRIPTS_TO_EXECUTE_LIST=script/example_handle_transcript_directory.sh

# filter out silent audio
#org.jitsi.jigasi.transcription.FILTER_SILENCE = false

# properties for optionally sending statistics to a DataDog server
#org.jitsi.ddclient.prefix=jitsi.jigasi
#org.jitsi.ddclient.host=localhost
#org.jitsi.ddclient.port=8125

# sip health checking
# Enables sip health checking by specifying a number/uri to call
# the target just needs to auto-connect the call play some audio,
# the call must be established for less than 10 seconds
# org.jitsi.jigasi.HEALTH_CHECK_SIP_URI=healthcheck
#
# The interval between healthcheck calls, by default is 5 minutes
# org.jitsi.jigasi.HEALTH_CHECK_INTERVAL=300000
#
# The timeout of healthcheck, if there was no successful health check for
# 10 minutes (default value) we consider jigasi unhealthy
# org.jitsi.jigasi.HEALTH_CHECK_TIMEOUT=600000

# Enabled or disable the notification when max occupants limit is reached
# org.jitsi.jigasi.NOTIFY_MAX_OCCUPANTS=false

org.jitsi.jigasi.rest.jetty.port=9001

Config Videobridge

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=1 minute

    # The timeout for a health check. This needs to be higher than [interval], otherwise health checks timeout because
    # none were scheduled.
    timeout=90 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
    // Whether the bridge is allowed to oversend (send the lowest layer regardless of BWE) for on-stage endpoints. If
    // allowed, it's only used when an endpoint is screensharing.
    allow-oversend-onstage=true
    // The maximum bitrate by which the estimation will be exceeded when oversending (if oversending is allowed).
    max-oversend-bitrate=500 kbps
    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}

      # Controls which statistics are sent.
      stats-filter {
        # Whether to filter the statistics.
        # If true, send whitelisted keys only. If false, send all statistics.
        enabled = false

        # Which statistics to send, when filter is enabled.
        # Ignored if filter is disabled.
        whitelist = ["average_participant_stress", "colibri2", "current_timestamp", "drain", "graceful_shutdown",
          "healthy", "region", "relay_id", "release", "shutting_down", "stress_level", "version"]
      }

      # The size of the Smack JID cache
      jid-cache-size = 1000

    configs {
                shard-local {
                    hostname = "localhost"
                    domain = "auth.xxxxxxxxxxxxxx"
                    username = "jvb"
                    password = "xxxxxxxxxxxxxx"
                    muc_jids = "JvbBrewery@internal.auth.xxxxxxxxxxxxxx"
                    muc_nickname = "xxxxxxxxxxxxxx"
                    disable_certificate_verification = true
                }
      }
    }
    # 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 = true
    }
    drain {
      enabled = true
    }
    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 = xxxxxxxxxxxxxx
    }
    # The HTTP server which hosts services intended for 'private' use
    # (e.g. health or debug stats)
    private {
      # See JettyBundleActivatorConfig in Jicoco for values
      host = xxxxxxxxxxxxxx
      port = xxxxxxxxxxxxxx
    }
  }
  relay {
    # Whether or not relays (octo) are enabled
    enabled=true

    # 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="xxxxxxxxxxxxxx"

    # The unique identifier of the jitsi-videobridge instance as a relay.
    # Must be set when 'enabled' is true.
    relay-id="xxxxxxxxxxxxxx"
  }
  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 = 1
        # 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
      }
    }
    // LastN limits based on conference size. Maps a conference size to the maximum number of streams that will be
    // forwarded to endpoints in a conference of that size or larger (up until the next entry).
    conference-last-n-limits {
      // With these example values conferences with size<20 endpoints can have arbitrary lastN, conferences with
      // 20 < size <= 29 can have lastN at most 20, conferences 30 < size <= 50 can have lastN at most 15.
      #20 = 20,
      #30 = 15,
      #50 = 10,
      #90 = 2
    }
    // An estimation of the stress an endpoint adds to the bridge when ramped up. This is communicated to jicofo
    // together with the stress level, where it is used to calculate an augmented stress value for the bridge, which
    // takes into account recently added endpoints:
    // augmented_stress = jvb_reported_stress + number_recently_added_participants * average-participant-stress
    average-participant-stress = 0.01
  }
  sctp {
    # Whether SCTP data channels are enabled.
    enabled=true
  }
  stats {
    # The interval at which stats are gathered.
    interval = 5 seconds
    transports = [
      { type = "muc" }
    ]
  }
  websockets {
    enabled=true
    server-id="xxxxxxxxxxxxxx"
    # Whether to negotiate WebSocket compression (permessage-deflate)
    enable-compression = true

    # Optional, even when 'enabled' is set to true
    tls=true
    # Must be set when enabled = true
    domain="xxxxxxxxxxxxxx:443"
  }
  ice {
    tcp {
      # Whether ICE/TCP is enabled.
      enabled = true

      # The port to bind to for ICE/TCP.
      # port = 443

      # 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 = "xxxxxxxxxxxxxx:"

    # 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"

    # Whether to advertise private ICE candidates, i.e. RFC 1918 IPv4 addresses and fec0::/10 and fc00::/7 IPv6 addresses.
    advertise-private-candidates = true
  }

  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
    }
  }

  # The experimental multiple streams per endpoint support
  multi-stream {
    enabled = true
  }

  speech-activity {
    # The number of speakers to include in the list of recent speakers sent with dominant speaker change
    # notifications.
    recent-speakers-count = 10
  }

  loudest {
      # Whether to route only the loudest speakers. If false, all audio is forwarded.
      route-loudest-only = false

      # The number of current loudest speakers to route audio packets for.
      # Ignored if route-loudest-only = false.
      num-loudest = 3

      # Whether to route dominant speaker when it is not among the current loudest speakers.
      # Ignored if route-loudest-only = false.
      always-route-dominant = true

      # Time after which speaker is removed from loudest list if
      # no new audio packets have been received from that speaker.
      energy-expire-time = 150 milliseconds

      # Alpha factor for exponential smoothing of energy values, multiplied by 100.
      energy-alpha-pct = 50
  }

  version {
    # Whether to announce the jitsi-videobridge version to clients in the ServerHello message.
    announce = false

    # A release identifier. When set, this will be appended to the version string
    # that is associated with this bridge when selecting bridges for a conference.
    # This allows finer-grained control of the selection process.
    # Note that multiple bridges must have the same version and release identifier
    # in order to share conferences between them.
    #release = "123"
  }

  shutdown {
    # The maximum amount of time to stay in GRACEFUL_SHUTDOWN before going into SHUTTING_DOWN. See ShutdownManager.
    graceful-shutdown-max-duration = 24 hours

    # The minimum number of participants required to stay in GRACEFUL_SHUTDOWN. If the number of participants falls
    # at or below this threshold the bridge will transition to SHUTTING_DOWN. See ShutdownManager.
    graceful-shutdown-min-participants = 0

    # The amount of time to stay in SHUTTING_DOWN before acutally shutting down. Gives jicofo time to actively move
    # participants off this bridge.
    shutting-down-delay = 1 minutes
  }

  # Whether to start the videobridge in drain mode.
  initial-drain-mode = false

  # In small conferences we disable the stats filter and always forward stats. This sets the minimum conference
  # size for which the filter is enabled.
  stats-filter-threshold = 20
}

Logs from the Videobridge and Jigasi

Link to Logs in a Gist

damencho commented 1 year ago

Please, when you have questions or problems use the community forum before opening new issues, thank you.

This had been fixed https://github.com/jitsi/jicofo/commit/c4bf5604f32673c97be8b20b60f5b36cffc979bb