mirror of
https://github.com/livekit/livekit.git
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09af509edb
* Add subscription limits * Add limit to ParticipantParams * Don't change desired of subscription when reaching limits * Add subscription limits config * Revert comment * solve comments
270 lines
12 KiB
YAML
270 lines
12 KiB
YAML
# main TCP port for RoomService and RTC endpoint
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# for production setups, this port should be placed behind a load balancer with TLS
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port: 7880
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# when redis is set, LiveKit will automatically operate in a fully distributed fashion
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# clients could connect to any node and be routed to the same room
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redis:
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address: redis.host:6379
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# To require TLS transport
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# use_tls: true
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# db: 0
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# username: myuser
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# password: mypassword
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# To use sentinel remove the address key above and add the following
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# sentinel_master_name: livekit
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# sentinel_addresses:
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# - livekit-redis-node-0.livekit-redis-headless:26379
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# - livekit-redis-node-1.livekit-redis-headless:26379
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# If you use a different set of credentials for sentinel add
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# sentinel_username: user
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# sentinel_password: pass
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#
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# To use cluster remove the address key above and add the following
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# cluster_addresses:
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# - livekit-redis-node-0.livekit-redis-headless:6379
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# - livekit-redis-node-1.livekit-redis-headless:6380
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# And it will use the password key above as cluster password
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# And the db key will not be used due to cluster mode not support it.
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# WebRTC configuration
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rtc:
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# UDP ports to use for client traffic.
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# this port range should be open for inbound traffic on the firewall
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port_range_start: 50000
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port_range_end: 60000
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# when set, LiveKit enable WebRTC ICE over TCP when UDP isn't available
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# this port *cannot* be behind load balancer or TLS, and must be exposed on the node
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# WebRTC transports are encrypted and do not require additional encryption
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# only 80/443 on public IP are allowed if less than 1024
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tcp_port: 7881
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# when set to true, attempts to discover the host's public IP via STUN
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# this is useful for cloud environments such as AWS & Google where hosts have an internal IP
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# that maps to an external one
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use_external_ip: true
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# # when set, LiveKit will attempt to use a UDP mux so all UDP traffic goes through
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# # a single port. This simplifies deployment, but mux will become an overhead for
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# # highly trafficked deployments.
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# # port_range_start & end must not be set for this config to take effect
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# udp_port: 7882
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# # when set to true, server will use a lite ice agent, that will speed up ice connection, but
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# # might cause connect issue if server running behind NAT.
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# use_ice_lite: true
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# # optional STUN servers for LiveKit clients to use. Clients will be configured to use these STUN servers automatically.
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# # by default LiveKit clients use Google's public STUN servers
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# stun_servers:
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# - server1
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# # optional TURN servers for clients. This isn't necessary if using embedded TURN server (see below).
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# turn_servers:
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# - host: myhost.com
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# port: 443
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# # tls, tcp, or udp
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# protocol: tls
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# username: ""
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# credential: ""
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# # allows LiveKit to monitor congestion when sending streams and automatically
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# # manage bandwidth utilization to avoid congestion/loss. Enabled by default
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# congestion_control:
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# enabled: true
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# # in the unlikely event of highly congested networks, SFU may choose to pause some tracks
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# # in order to allow others to stream smoothly. You can disable this behavior here
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# allow_pause: true
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# # allows automatic connection fallback to TCP and TURN/TLS (if configured) when UDP has been unstable, default true
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# allow_tcp_fallback: true
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# # number of packets to buffer in the SFU, defaults to 500
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# packet_buffer_size: 500
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# # minimum amount of time between pli/fir rtcp packets being sent to an individual
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# # producer. Increasing these times can lead to longer black screens when new participants join,
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# # while reducing them can lead to higher stream bitrate.
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# pli_throttle:
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# low_quality: 500ms
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# mid_quality: 1s
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# high_quality: 1s
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# # when set, Livekit will collect loopback candidates, it is useful for some VM have public address mapped to its loopback interface.
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# enable_loopback_candidate: true
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# # network interface filter. If the machine has more than one network interface and you'd like it to use or skip specific interfaces
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# # both inclusion and exclusion filters can be used together. If neither is defined (default), all interfaces on the machine will be used.
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# # If both of them are set, then only include takes effect.
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# interfaces:
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# includes:
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# - en0
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# excludes:
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# - docker0
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# # ip address filter. If the machine has more than one ip address and you'd like it to use or skip specific ips,
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# # both inclusion and exclusion CIDR filters can be used together. If neither is defined (default), all ip on the machine will be used.
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# # If both of them are set, then only include takes effect.
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# ips:
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# includes:
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# - 10.0.0.0/16
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# excludes:
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# - 192.168.1.0/24
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# # Set to true to enable mDNS name candidate. This should be left disabled for most users.
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# # when enabled, it will impact performance since each PeerConnection will process the same mDNS message independently
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# use_mdns: true
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# # Set to false to disable strict ACKs for peer connections where LiveKit is the dialing side,
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# # ie. subscriber peer connections. Disabling strict ACKs will prevent clients that do not ACK
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# # peer connections from getting kicked out of rooms by the monitor. Note that if strict ACKs
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# # are disabled and clients don't ACK opened peer connections, only reliable, ordered delivery
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# # will be available.
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# strict_acks: true
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# when enabled, LiveKit will expose prometheus metrics on :6789/metrics
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# prometheus_port: 6789
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# set a custom environment variable. prometheus metrics will be labeled with this value. defaults to an empty string
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# environment: custom-value
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# API key / secret pairs.
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# Keys are used for JWT authentication, server APIs would require a keypair in order to generate access tokens
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# and make calls to the server
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keys:
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key1: secret1
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key2: secret2
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# Logging config
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# logging:
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# # log level, valid values: debug, info, warn, error
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# level: info
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# # log level for pion, default error
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# pion_level: error
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# # when set to true, emit json fields
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# json: false
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# # for production setups, enables sampling algorithm
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# # https://github.com/uber-go/zap/blob/master/FAQ.md#why-sample-application-logs
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# sample: false
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# Default room config
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# Each room created will inherit these settings. If rooms are created explicitly with CreateRoom, they will take
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# precedence over defaults
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# room:
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# # allow rooms to be automatically created when participants join, defaults to true
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# # auto_create: false
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# # number of seconds to leave a room open when it's empty
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# empty_timeout: 300
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# # limit number of participants that can be in a room, 0 for no limit
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# max_participants: 0
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# # only accept specific codecs for clients publishing to this room
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# # this is useful to standardize codecs across clients
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# # other supported codecs are video/h264
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# enabled_codecs:
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# - mime: audio/opus
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# - mime: video/vp8
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# # allow tracks to be unmuted remotely, defaults to false
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# # tracks can always be muted from the Room Service APIs
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# enable_remote_unmute: true
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# # limit size of room and participant's metadata, 0 for no limit
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# max_metadata_size: 0
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# Webhooks
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# when configured, LiveKit notifies your URL handler with room events
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# webhook:
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# # the API key to use in order to sign the message
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# # this must match one of the keys LiveKit is configured with
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# api_key: <api_key>
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# # list of URLs to be notified of room events
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# urls:
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# - https://your-host.com/handler
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# Signal Relay
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# since v1.4.0, a more reliable, psrpc based signal relay is available
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# this gives us the ability to reliably proxy messages between a signal server and RTC node
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# signal_relay:
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# # disabled by default. will be enabled by default in future versions
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# enabled: true
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# # amount of time a message delivery is tried before giving up
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# retry_timeout: 30s
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# # minimum amount of time to wait for RTC node to ack,
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# # retries use exponentially increasing wait on every subsequent try
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# # with an upper bound of max_retry_interval
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# min_retry_interval: 500ms
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# # maximum amount of time to wait for RTC node to ack
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# max_retry_interval: 5s
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# # number of messages to buffer before dropping
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# stream_buffer_size: 1000
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# customize audio level sensitivity
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# audio:
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# # minimum level to be considered active, 0-127, where 0 is loudest
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# # defaults to 30
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# active_level: 30
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# # percentile to measure, a participant is considered active if it has exceeded the
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# # ActiveLevel more than MinPercentile% of the time
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# # defaults to 40
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# min_percentile: 40
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# # frequency in ms to notify changes to clients, defaults to 500
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# update_interval: 500
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# # to prevent speaker updates from too jumpy, smooth out values over N samples
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# smooth_intervals: 4
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# # enable red encoding downtrack for opus only audio up track
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# active_red_encoding: true
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# turn server
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# turn:
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# # Uses TLS. Requires cert and key pem files by either:
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# # - using turn.secretName if deploying with our helm chart, or
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# # - setting LIVEKIT_TURN_CERT and LIVEKIT_TURN_KEY env vars with file locations, or
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# # - using cert_file and key_file below
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# # defaults to false
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# enabled: false
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# # defaults to 3478 - recommended to 443 if not running HTTP3/QUIC server
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# # only 53/80/443 are allowed if less than 1024
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# udp_port: 3478
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# # defaults to 5349 - if not using a load balancer, this must be set to 443
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# tls_port: 5349
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# # set UDP port range for TURN relay to connect to LiveKit SFU, by default it uses a any available port
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# relay_range_start: 1024
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# relay_range_end: 30000
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# # set external_tl to true if using a L4 load balancer to terminate TLS. when enabled,
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# # LiveKit expects unencrypted traffic on tls_port, and still advertise tls_port as a TURN/TLS candidate.
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# external_tls: true
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# # needs to match tls cert domain
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# domain: turn.myhost.com
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# # optional (set only if not using external TLS termination)
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# # cert_file: /path/to/cert.pem
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# # key_file: /path/to/key.pem
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# ingress server
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# ingress:
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# # Prefix used to generate RTMP URLs for RTMP ingress.
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# # The stream_key will be appended to this base and returned as part of the
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# # ingress info
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# rtmp_base_url: "rtmp://my.domain.com/live"
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# egress server
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# egress:
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# # Whether to use the PSRPC enabled RPC implementation. This requires livekit egress version >=1.5.4
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# # The legacy, non PSRPC RPC implementation will be removed eventually
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# use_psrpc: false
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# Region of the current node. Required if using regionaware node selector
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# region: us-west-2
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# # node selector
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# node_selector:
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# # default: any. valid values: any, sysload, cpuload, regionaware
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# kind: sysload
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# # priority used for selection of node when multiple are available
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# # default: random. valid values: random, sysload, cpuload, rooms, clients, tracks, bytespersec
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# sort_by: sysload
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# # used in sysload and regionaware
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# # do not assign room to node if load per CPU exceeds sysload_limit
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# sysload_limit: 0.7
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# # used in regionaware
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# # list of regions and their lat/lon coordinates
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# regions:
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# - name: us-west-2
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# lat: 44.19434095976287
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# lon: -123.0674908379146
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# # node limits
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# # set to -1 to disable a limit
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# limit:
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# # defaults to 400 tracks in & out per CPU, up to 8000
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# num_tracks: -1
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# # defaults to 1 GB/s, or just under 10 Gbps
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# bytes_per_sec: 1_000_000_000
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# # how many tracks (audio / video) that a single participant can subscribe at same time.
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# # if the limit is exceeded, subscriptions will be pending until any subscribed track has been unsubscribed.
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# # value less or equal than 0 means no limit.
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# subscription_limit_video: 0
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# subscription_limit_audio: 0
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