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54 lines
1.8 KiB
YAML
54 lines
1.8 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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# WebRTC configuration
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rtc:
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# when set, LiveKit will try to route traffic 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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tcp_port: 7881
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udp_port: 9000
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# UDP port range to transport WebRTC data
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port_range_start: 9000
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port_range_end: 11000
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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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# when using REMB, the max bitrate that the SFU would accept
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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 to use. by default LiveKit uses Google's public STUN servers
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# stun_servers:
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# - server1
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# API key / secret pairs.
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# Keys are used for JWT authentication
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keys:
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key1: secret1
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key2: secret2
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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 15
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# min_percentile: 15
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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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