mirror of
https://github.com/bluenviron/mediamtx
synced 2024-12-16 03:34:45 +00:00
232 lines
9.6 KiB
YAML
232 lines
9.6 KiB
YAML
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###############################################
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# General parameters
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# sets the verbosity of the program; available values are "warn", "info", "debug".
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logLevel: info
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# destinations of log messages; available values are "stdout", "file" and "syslog".
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logDestinations: [stdout]
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# if "file" is in logDestinations, this is the file which will receive the logs.
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logFile: rtsp-simple-server.log
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# timeout of read operations.
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readTimeout: 10s
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# timeout of write operations.
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writeTimeout: 10s
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# number of read buffers.
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# a higher number allows a higher throughput,
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# a lower number allows to save RAM.
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readBufferCount: 512
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# enable the HTTP API.
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api: no
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# address of the API listener.
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apiAddress: 127.0.0.1:9997
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# enable Prometheus-compatible metrics.
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metrics: no
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# address of the metrics listener.
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metricsAddress: 127.0.0.1:9998
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# enable pprof-compatible endpoint to monitor performances.
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pprof: no
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# address of the pprof listener.
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pprofAddress: 127.0.0.1:9999
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# command to run when a client connects to the server.
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# this is terminated with SIGINT when a client disconnects from the server.
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# the server port is available in the RTSP_PORT variable.
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runOnConnect:
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# the restart parameter allows to restart the command if it exits suddenly.
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runOnConnectRestart: no
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###############################################
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# RTSP parameters
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# disable support for the RTSP protocol.
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rtspDisable: no
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# supported RTSP stream protocols.
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# UDP is the most performant, but doesn't work when there's a NAT/firewall between
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# server and clients, and doesn't support encryption.
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# UDP-multicast allows to save bandwidth when clients are all in the same LAN.
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# TCP is the most versatile, and does support encryption.
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# The handshake is always performed with TCP.
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protocols: [udp, multicast, tcp]
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# encrypt handshake and TCP streams with TLS (RTSPS).
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# available values are "no", "strict", "optional".
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encryption: "no"
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# address of the TCP/RTSP listener. This is needed only when encryption is "no" or "optional".
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rtspAddress: :8554
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# address of the TCP/TLS/RTSPS listener. This is needed only when encryption is "strict" or "optional".
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rtspsAddress: :8555
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# address of the UDP/RTP listener. This is needed only when "udp" is in protocols.
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rtpAddress: :8000
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# address of the UDP/RTCP listener. This is needed only when "udp" is in protocols.
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rtcpAddress: :8001
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# IP range of all UDP-multicast listeners. This is needed only when "multicast" is in protocols.
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multicastIPRange: 224.1.0.0/16
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# port of all UDP-multicast/RTP listeners. This is needed only when "multicast" is in protocols.
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multicastRTPPort: 8002
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# port of all UDP-multicast/RTCP listeners. This is needed only when "multicast" is in protocols.
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multicastRTCPPort: 8003
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# path to the server key. This is needed only when encryption is "strict" or "optional".
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# this can be generated with:
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# openssl genrsa -out server.key 2048
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# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
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serverKey: server.key
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# path to the server certificate. This is needed only when encryption is "strict" or "optional".
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serverCert: server.crt
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# authentication methods.
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authMethods: [basic, digest]
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# read buffer size.
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# this doesn't influence throughput and shouldn't be touched unless the server
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# reports errors about the buffer size.
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readBufferSize: 2048
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###############################################
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# RTMP parameters
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# disable support for the RTMP protocol.
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rtmpDisable: no
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# address of the RTMP listener.
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rtmpAddress: :1935
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###############################################
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# HLS parameters
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# disable support for the HLS protocol.
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hlsDisable: no
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# address of the HLS listener.
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hlsAddress: :8888
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# by default, HLS is generated only when requested by a user;
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# this option allows to generate it always, avoiding an initial delay.
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hlsAlwaysRemux: no
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# number of HLS segments to generate.
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# increasing segments allows more buffering,
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# decreasing segments decreases latency.
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hlsSegmentCount: 3
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# minimum duration of each segment.
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# the final segment duration is also influenced by the interval between IDR frames,
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# since the server changes the segment duration to include at least a IDR frame in each one.
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hlsSegmentDuration: 1s
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# value of the Access-Control-Allow-Origin header provided in every HTTP response.
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# This allows to play the HLS stream from an external website.
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hlsAllowOrigin: '*'
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###############################################
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# Path parameters
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# these settings are path-dependent.
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# it's possible to use regular expressions by using a tilde as prefix.
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# for example, "~^(test1|test2)$" will match both "test1" and "test2".
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# for example, "~^prefix" will match all paths that start with "prefix".
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# the settings under the path "all" are applied to all paths that do not match
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# another entry.
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paths:
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all:
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# source of the stream - this can be:
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# * publisher -> the stream is published by a RTSP or RTMP client
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# * rtsp://existing-url -> the stream is pulled from another RTSP server
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# * rtsps://existing-url -> the stream is pulled from another RTSP server, with RTSPS
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# * rtmp://existing-url -> the stream is pulled from another RTMP server
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# * http://existing-url/stream.m3u8 -> the stream is pulled from another HLS server
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# * redirect -> the stream is provided by another path or server
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source: publisher
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# if the source is an RTSP or RTSPS URL, this is the protocol that will be used to
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# pull the stream. available values are "automatic", "udp", "multicast", "tcp".
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# the TCP protocol can help to overcome the error "no UDP packets received recently".
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sourceProtocol: automatic
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# if the source is an RTSP or RTSPS URL, this allows to support cameras that
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# don't provide server ports. This is a security issue and must be enabled only
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# when interacting with old cameras that require it.
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sourceAnyPortEnable: no
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# if the source is an RTSPS URL, the fingerprint of the certificate of the source
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# must be provided in order to prevent man-in-the-middle attacks.
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# it can be obtained from the source by running:
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# openssl s_client -connect source_ip:source_port </dev/null 2>/dev/null | sed -n '/BEGIN/,/END/p' > server.crt
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# openssl x509 -in server.crt -noout -fingerprint -sha256 | cut -d "=" -f2 | tr -d ':'
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sourceFingerprint:
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# if the source is an RTSP or RTMP URL, it will be pulled only when at least
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# one reader is connected, saving bandwidth.
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sourceOnDemand: no
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# if sourceOnDemand is "yes", readers will be put on hold until the source is
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# ready or until this amount of time has passed.
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sourceOnDemandStartTimeout: 10s
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# if sourceOnDemand is "yes", the source will be closed when there are no
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# readers connected and this amount of time has passed.
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sourceOnDemandCloseAfter: 10s
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# if the source is "redirect", this is the RTSP URL which clients will be
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# redirected to.
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sourceRedirect:
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# if the source is "publisher" and a client is publishing, do not allow another
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# client to disconnect the former and publish in its place.
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disablePublisherOverride: no
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# if the source is "publisher" and no one is publishing, redirect readers to this
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# path. It can be can be a relative path (i.e. /otherstream) or an absolute RTSP URL.
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fallback:
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# username required to publish.
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# sha256-hashed values can be inserted with the "sha256:" prefix.
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publishUser:
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# password required to publish.
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# sha256-hashed values can be inserted with the "sha256:" prefix.
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publishPass:
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# ips or networks (x.x.x.x/24) allowed to publish.
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publishIPs: []
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# username required to read.
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# sha256-hashed values can be inserted with the "sha256:" prefix.
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readUser:
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# password required to read.
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# sha256-hashed values can be inserted with the "sha256:" prefix.
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readPass:
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# ips or networks (x.x.x.x/24) allowed to read.
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readIPs: []
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# command to run when this path is initialized.
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# this can be used to publish a stream and keep it always opened.
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# this is terminated with SIGINT when the program closes.
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# the path name is available in the RTSP_PATH variable.
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# the server port is available in the RTSP_PORT variable.
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runOnInit:
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# the restart parameter allows to restart the command if it exits suddenly.
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runOnInitRestart: no
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# command to run when this path is requested.
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# this can be used to publish a stream on demand.
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# this is terminated with SIGINT when the path is not requested anymore.
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# the path name is available in the RTSP_PATH variable.
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# the server port is available in the RTSP_PORT variable.
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runOnDemand:
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# the restart parameter allows to restart the command if it exits suddenly.
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runOnDemandRestart: no
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# readers will be put on hold until the runOnDemand command starts publishing
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# or until this amount of time has passed.
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runOnDemandStartTimeout: 10s
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# the runOnDemand command will be closed when there are no
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# readers connected and this amount of time has passed.
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runOnDemandCloseAfter: 10s
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# command to run when a client starts publishing.
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# this is terminated with SIGINT when a client stops publishing.
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# the path name is available in the RTSP_PATH variable.
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# the server port is available in the RTSP_PORT variable.
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runOnPublish:
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# the restart parameter allows to restart the command if it exits suddenly.
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runOnPublishRestart: no
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# command to run when a clients starts reading.
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# this is terminated with SIGINT when a client stops reading.
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# the path name is available in the RTSP_PATH variable.
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# the server port is available in the RTSP_PORT variable.
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runOnRead:
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# the restart parameter allows to restart the command if it exits suddenly.
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runOnReadRestart: no
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