# main TCP port for RoomService and RTC endpoint
# for production setups, this port should be placed behind a load balancer with TLS
port: 7880
log_level: info

# when redis is set, LiveKit will automatically operate in a fully distributed fashion
# clients could connect to any node and be routed to the same room
redis:
  address: 127.0.0.1:6379
  # To require TLS transport
  # use_tls: true
  # db: 0
  # username: myuser
  # password: mypassword
  # To use sentinel remove the address key above and add the following
  # sentinel_master_name: livekit
  # sentinel_addresses:
  # - livekit-redis-node-0.livekit-redis-headless:26379
  # - livekit-redis-node-1.livekit-redis-headless:26379
  # If you use a different set of credentials for sentinel add
  # sentinel_username: user
  # sentinel_password: pass

# WebRTC configuration
rtc:
  # UDP ports to use for client traffic.
  # this port range should be open for inbound traffic on the firewall
  # udp_port: 7882
  port_range_start: 50006
  port_range_end: 60000
  # when set, LiveKit enable WebRTC ICE over TCP when UDP isn't available
  # this port *cannot* be behind load balancer or TLS, and must be exposed on the node
  # WebRTC transports are encrypted and do not require additional encryption
  # only 80/443 on public IP are allowed if less than 1024
  tcp_port: 7881
  # when set to true, attempts to discover the host's public IP via STUN
  # this is useful for cloud environments such as AWS & Google where hosts have an internal IP
  # that maps to an external one
  use_external_ip: true
  # # when set, LiveKit will attempt to use a UDP mux so all UDP traffic goes through
  # # a single port. This simplifies deployment, but mux will become an overhead for
  # # highly trafficked deployments.
  # # port_range_start & end must not be set for this config to take effect
  # # when set to true, server will use a lite ice agent, that will speed up ice connection, but
  # # might cause connect issue if server running behind NAT.
  # use_ice_lite: true
  # # optional STUN servers for LiveKit clients to use. Clients will be configured to use these STUN servers automatically.
  # # by default LiveKit clients use Google's public STUN servers
  # stun_servers:
  #   - server1
  # # optional TURN servers for clients. This isn't necessary if using embedded TURN server (see below).
  # turn_servers:
  #   - host: <http://myhost.com|myhost.com>
      # port: 443
  #     # tls, tcp, or udp
  #     protocol: tls
  #     username: ""
  #     credential: ""
  # # allows LiveKit to monitor congestion when sending streams and automatically
  # # manage bandwidth utilization to avoid congestion/loss. Enabled by default
  # congestion_control:
    # enabled: true
  #   # in the unlikely event of highly congested networks, SFU may choose to pause some tracks
  #   # in order to allow others to stream smoothly. You can disable this behavior here
  #   allow_pause: true
  # # allows automatic connection fallback to TCP and TURN/TLS (if configured) when UDP has been unstable, default true
  # allow_tcp_fallback: true
  # # number of packets to buffer in the SFU, defaults to 500
  # packet_buffer_size: 500
  # # minimum amount of time between pli/fir rtcp packets being sent to an individual
  # # producer. Increasing these times can lead to longer black screens when new participants join,
  # # while reducing them can lead to higher stream bitrate.
  # pli_throttle:
  #   low_quality: 500ms
  #   mid_quality: 1s
  #   high_quality: 1s
  # # network interface filter. If the machine has more than one network interface and you'd like it to use or skip specific interfaces
  # # both inclusion and exclusion filters can be used together. If neither is defined (default), all interfaces on the machine will be used.
  # # If both of them are set, then only include takes effect.
  # interfaces:
  #   includes:
  #     - en0
  #   excludes:
  #     - docker0
  # # ip address filter. If the machine has more than one ip address and you'd like it to use or skip specific ips,
  # # both inclusion and exclusion CIDR filters can be used together. If neither is defined (default), all ip on the machine will be used.
  # # If both of them are set, then only include takes effect.
  # ips:
  #   includes:
  #     - 10.0.0.0/16
  #   excludes:
  #     - 192.168.1.0/24

# when enabled, LiveKit will expose prometheus metrics on :6789/metrics
# prometheus_port: 6789

# API key / secret pairs.
# Keys are used for JWT authentication, server APIs would require a keypair in order to generate access tokens
# and make calls to the server
keys:
  API8k8Yb21ZRzmKpz: AOt8guwcvpj32YQjIYiPWiiBiw2V8VacxqkXSrb2X6ehX
  # key2: AOt8guwcvpjYQjIYiPWiiBiw2V8VacxqkXSrb2X6ehX

# Logging config
# logging:
#   # log level, valid values: debug, info, warn, error
#   level: info
#   # log level for pion, default error
#   pion_level: error
#   # when set to true, emit json fields
#   json: false
#   # for production setups, enables sampling algorithm
#   # <https://github.com/uber-go/zap/blob/master/FAQ.md#why-sample-application-logs>
#   sample: false


# Default room config
# Each room created will inherit these settings. If rooms are created explicitly with CreateRoom, they will take
# precedence over defaults
# room:
#   # allow rooms to be automatically created when participants join, defaults to true
#   # auto_create: false
#   # number of seconds to leave a room open when it's empty
#   empty_timeout: 300
#   # limit number of participants that can be in a room, 0 for no limit
#   max_participants: 0
#   # only accept specific codecs for clients publishing to this room
#   # this is useful to standardize codecs across clients
#   # other supported codecs are video/h264
#   enabled_codecs:
#     - mime: audio/opus
#     - mime: video/vp8
#   # allow tracks to be unmuted remotely, defaults to false
#   # tracks can always be muted from the Room Service APIs
#   enable_remote_unmute: true
#   # limit size of room and participant's metadata, 0 for no limit
#   max_metadata_size: 0

# Webhooks
# when configured, LiveKit notifies your URL handler with room events
# webhook:
#   # the API key to use in order to sign the message
#   # this must match one of the keys LiveKit is configured with
#   api_key: <api_key>
#   # list of URLs to be notified of room events
#   urls:
#     - <https://your-host.com/handler>

# customize audio level sensitivity
# audio:
#   # minimum level to be considered active, 0-127, where 0 is loudest
#   # defaults to 30
#   active_level: 30
#   # percentile to measure, a participant is considered active if it has exceeded the
#   # ActiveLevel more than MinPercentile% of the time
#   # defaults to 40
#   min_percentile: 40
#   # frequency in ms to notify changes to clients, defaults to 500
#   update_interval: 500
#   # to prevent speaker updates from too jumpy, smooth out values over N samples
#   smooth_intervals: 4

<video _src_*=*{subscribedTracks[0]*?.*_track_*.*_mediaStream_}></video>;

useEffect(() *=>* {

const el *=* ref*.*_current_;

*if* (*!*el) {

*return*;

}

el*.*_muted_

*=* true;

track*?.*attach(el); // call the attach..

*return* () *=>* {

track*?.*detach(el);

};

}, [track, ref]);

*return* <video _ref_*=*{ref} />;

EntryPointNotFoundException: PushString assembly:<unknown assembly> type:<unknown type> member:(null)
LiveKit.Room.GetLocalDevices (System.Nullable`1[T] kind, System.Nullable`1[T] requestPermissions) (at Library/PackageCache/io.livekit.unity@20a7cdbbc3/Runtime/Scripts/Room/Room.cs:424)
JoinHandler+<StartPreviewCamera>d__29.MoveNext () (at Assets/Scripts/JoinHandler.cs:101)
UnityEngine.SetupCoroutine.InvokeMoveNext (System.Collections.IEnumerator enumerator, System.IntPtr returnValueAddress) (at /Users/bokken/buildslave/unity/build/Runtime/Export/Scripting/Coroutines.cs:17)
UnityEngine.MonoBehaviour:StartCoroutine(IEnumerator)
JoinHandler:Start() (at Assets/Scripts/JoinHandler.cs:60)