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6751361187
In order to ensure clients are publishing with compatible codecs, the server can enforce codec restrictions.
99 lines
3.7 KiB
YAML
99 lines
3.7 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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# log level, valid values: debug, info, warning, error
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log_level: info
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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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# username: myuser
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# password: mypassword
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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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# WebRTC configuration
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rtc:
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# the main UDP port to transport WebRTC data
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udp_port: 9000
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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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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, 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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# optional settings
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# # when using REMB, the max bitrate that the SFU would accept, defaults to 3Mbps
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# max_bitrate: 3145728
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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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# # 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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# # 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 participants join,
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# # while reducing them can lead to higher producer bitrates.
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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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# 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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# 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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# # 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, video/vp9
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# enabled_codecs:
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# - mime: audio/opus
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# - mime: video/vp8
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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 40
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# active_level: 40
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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_samples: 8
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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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# needs to match tls cert domain
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domain: turn.myhost.com
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# defaults to 3478 - if not using a load balancer, this must be set to 443
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tls_port: 3478
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# optional
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# cert_file: /path/to/cert.pem
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# key_file: /path/to/key.pem
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