mirror of
https://github.com/bluenviron/mediamtx
synced 2024-12-15 19:24:55 +00:00
509 lines
21 KiB
YAML
509 lines
21 KiB
YAML
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###############################################
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# General settings
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# Sets the verbosity of the program; available values are "error", "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: mediamtx.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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# Size of the queue of outgoing packets.
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# A higher value allows to increase throughput, a lower value allows to save RAM.
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writeQueueSize: 512
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# Maximum size of outgoing UDP packets.
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# This can be decreased to avoid fragmentation on networks with a low UDP MTU.
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udpMaxPayloadSize: 1472
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# HTTP URL to perform external authentication.
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# Every time a user wants to authenticate, the server calls this URL
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# with the POST method and a body containing:
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# {
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# "ip": "ip",
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# "user": "user",
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# "password": "password",
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# "path": "path",
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# "protocol": "rtsp|rtmp|hls|webrtc",
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# "id": "id",
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# "action": "read|publish",
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# "query": "query"
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# }
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# If the response code is 20x, authentication is accepted, otherwise
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# it is discarded.
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externalAuthenticationURL:
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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 following environment variables are available:
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# * RTSP_PORT: RTSP server port
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runOnConnect:
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# Restart the command if it exits.
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runOnConnectRestart: no
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# Command to run when a client disconnects from the server.
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# Environment variables are the same of runOnConnect.
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runOnDisconnect:
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###############################################
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# RTSP settings
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# Allow publishing and reading streams with the RTSP protocol.
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rtsp: yes
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# List of enabled RTSP transport 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 handshakes 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: :8322
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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. Available are "basic" and "digest".
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# "digest" doesn't provide any additional security and is available for compatibility reasons only.
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authMethods: [basic]
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###############################################
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# RTMP settings
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# Allow publishing and reading streams with the RTMP protocol.
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rtmp: yes
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# Address of the RTMP listener. This is needed only when encryption is "no" or "optional".
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rtmpAddress: :1935
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# Encrypt connections with TLS (RTMPS).
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# Available values are "no", "strict", "optional".
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rtmpEncryption: "no"
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# Address of the RTMPS listener. This is needed only when encryption is "strict" or "optional".
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rtmpsAddress: :1936
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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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rtmpServerKey: 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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rtmpServerCert: server.crt
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###############################################
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# HLS settings
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# Allow reading streams with the HLS protocol.
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hls: yes
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# Address of the HLS listener.
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hlsAddress: :8888
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# Enable TLS/HTTPS on the HLS server.
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# This is required for Low-Latency HLS.
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hlsEncryption: no
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# Path to the server key. This is needed only when encryption is yes.
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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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hlsServerKey: server.key
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# Path to the server certificate.
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hlsServerCert: server.crt
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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 the delay between request and generation.
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hlsAlwaysRemux: no
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# Variant of the HLS protocol to use. Available options are:
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# * mpegts - uses MPEG-TS segments, for maximum compatibility.
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# * fmp4 - uses fragmented MP4 segments, more efficient.
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# * lowLatency - uses Low-Latency HLS.
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hlsVariant: lowLatency
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# Number of HLS segments to keep on the server.
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# Segments allow to seek through the stream.
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# Their number doesn't influence latency.
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hlsSegmentCount: 7
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# Minimum duration of each segment.
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# A player usually puts 3 segments in a buffer before reproducing the stream.
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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 duration in order to include at least one IDR frame
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# in each segment.
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hlsSegmentDuration: 1s
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# Minimum duration of each part.
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# A player usually puts 3 parts in a buffer before reproducing the stream.
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# Parts are used in Low-Latency HLS in place of segments.
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# Part duration is influenced by the distance between video/audio samples
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# and is adjusted in order to produce segments with a similar duration.
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hlsPartDuration: 200ms
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# Maximum size of each segment.
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# This prevents RAM exhaustion.
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hlsSegmentMaxSize: 50M
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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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# List of IPs or CIDRs of proxies placed before the HLS server.
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# If the server receives a request from one of these entries, IP in logs
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# will be taken from the X-Forwarded-For header.
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hlsTrustedProxies: []
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# Directory in which to save segments, instead of keeping them in the RAM.
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# This decreases performance, since reading from disk is less performant than
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# reading from RAM, but allows to save RAM.
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hlsDirectory: ''
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###############################################
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# WebRTC settings
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# Allow publishing and reading streams with the WebRTC protocol.
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webrtc: yes
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# Address of the WebRTC listener.
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webrtcAddress: :8889
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# Enable TLS/HTTPS on the WebRTC server.
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webrtcEncryption: no
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# Path to the server key.
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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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webrtcServerKey: server.key
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# Path to the server certificate.
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webrtcServerCert: server.crt
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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 WebRTC stream from an external website.
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webrtcAllowOrigin: '*'
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# List of IPs or CIDRs of proxies placed before the WebRTC server.
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# If the server receives a request from one of these entries, IP in logs
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# will be taken from the X-Forwarded-For header.
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webrtcTrustedProxies: []
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# List of ICE servers.
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webrtcICEServers2:
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# URL can point to a STUN, TURN or TURNS server.
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# STUN servers are used to obtain the public IP of server and clients. They are
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# needed when server and clients are on different LANs.
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# TURN/TURNS servers are needed when a direct connection between server and
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# clients is not possible. All traffic is routed through them.
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- url: stun:stun.l.google.com:19302
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# if user is "AUTH_SECRET", then authentication is secret based.
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# the secret must be inserted into the password field.
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username: ''
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password: ''
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# List of public IP addresses that are to be used as a host.
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# This is used typically for servers that are behind 1:1 D-NAT.
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webrtcICEHostNAT1To1IPs: []
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# Address of a ICE UDP listener in format host:port.
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# If filled, ICE traffic will pass through a single UDP port,
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# allowing the deployment of the server inside a container or behind a NAT.
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webrtcICEUDPMuxAddress:
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# Address of a ICE TCP listener in format host:port.
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# If filled, ICE traffic will pass through a single TCP port,
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# allowing the deployment of the server inside a container or behind a NAT.
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# Using this setting forces usage of the TCP protocol, which is not
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# optimal for WebRTC.
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webrtcICETCPMuxAddress:
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###############################################
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# SRT settings
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# Allow publishing and reading streams with the SRT protocol.
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srt: yes
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# Address of the SRT listener.
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srtAddress: :8890
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###############################################
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# Recording settings
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# Record streams to disk.
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record: no
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# Path of recording segments.
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# Extension is added automatically.
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# Available variables are %path (path name), %Y %m %d %H %M %S %f (time in strftime format)
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recordPath: ./recordings/%path/%Y-%m-%d_%H-%M-%S-%f
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# Format of recorded segments.
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# Currently the only available format is fmp4 (fragmented MP4).
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recordFormat: fmp4
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# fMP4 files are concatenation of small MP4 files (parts), each with this duration.
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# When a system failure occurs, the last part gets lost.
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# Therefore, the part duration is equal to the RPO (recovery point objective).
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recordPartDuration: 100ms
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# Minimum duration of each segment.
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recordSegmentDuration: 1h
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# Delete segments after this timespan.
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# Set to 0 to disable automatic deletion.
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recordDeleteAfter: 24h
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###############################################
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# Path settings
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# These settings are path-dependent, and the map key is the name of the path.
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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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# 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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###############################################
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# General path settings
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# Source of the stream. This can be:
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# * publisher -> the stream is provided by a RTSP, RTMP, WebRTC or SRT client
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# * rtsp://existing-url -> the stream is pulled from another RTSP server / camera
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# * rtsps://existing-url -> the stream is pulled from another RTSP server / camera with RTSPS
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# * rtmp://existing-url -> the stream is pulled from another RTMP server / camera
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# * rtmps://existing-url -> the stream is pulled from another RTMP server / camera with RTMPS
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# * http://existing-url/stream.m3u8 -> the stream is pulled from another HLS server / camera
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# * https://existing-url/stream.m3u8 -> the stream is pulled from another HLS server / camera with HTTPS
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# * udp://ip:port -> the stream is pulled with UDP, by listening on the specified IP and port
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# * srt://existing-url -> the stream is pulled from another SRT server / camera
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# * whep://existing-url -> the stream is pulled from another WebRTC server / camera
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# * wheps://existing-url -> the stream is pulled from another WebRTC server / camera with HTTPS
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# * redirect -> the stream is provided by another path or server
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# * rpiCamera -> the stream is provided by a Raspberry Pi Camera
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source: publisher
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# If the source is a URL, and the source certificate is self-signed
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# or invalid, you can provide the fingerprint of the certificate in order to
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# validate it anyway. It can be obtained 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 a 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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# Maximum number of readers. Zero means no limit.
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maxReaders: 0
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# Record streams to disk (if global recording is enabled).
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record: yes
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###############################################
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# Authentication path settings
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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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###############################################
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# Publisher path settings (when source is "publisher")
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# allow another client to disconnect the current publisher and publish in its place.
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overridePublisher: yes
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# if no one is publishing, redirect readers to this path.
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# 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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###############################################
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# RTSP path settings (when source is a RTSP or a RTSPS URL)
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# protocol used to pull the stream. available values are "automatic", "udp", "multicast", "tcp".
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sourceProtocol: automatic
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# support sources that don't provide server ports or use random server ports. This is a security issue
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# and must be used only when interacting with sources that require it.
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sourceAnyPortEnable: no
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# range header to send to the source, in order to start streaming from the specified offset.
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# available values:
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# * clock: Absolute time
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# * npt: Normal Play Time
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# * smpte: SMPTE timestamps relative to the start of the recording
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rtspRangeType:
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# available values:
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# * clock: UTC ISO 8601 combined date and time string, e.g. 20230812T120000Z
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# * npt: duration such as "300ms", "1.5m" or "2h45m", valid time units are "ns", "us" (or "µs"), "ms", "s", "m", "h"
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# * smpte: duration such as "300ms", "1.5m" or "2h45m", valid time units are "ns", "us" (or "µs"), "ms", "s", "m", "h"
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rtspRangeStart:
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###############################################
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# Redirect path settings (when source is "redirect")
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# RTSP URL which clients will be redirected to.
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sourceRedirect:
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###############################################
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# Raspberry Pi Camera path settings (when source is "rpiCamera")
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# ID of the camera
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rpiCameraCamID: 0
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# width of frames
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rpiCameraWidth: 1920
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# height of frames
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rpiCameraHeight: 1080
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# flip horizontally
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rpiCameraHFlip: false
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# flip vertically
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rpiCameraVFlip: false
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# brightness [-1, 1]
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rpiCameraBrightness: 0
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# contrast [0, 16]
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rpiCameraContrast: 1
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# saturation [0, 16]
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rpiCameraSaturation: 1
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# sharpness [0, 16]
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rpiCameraSharpness: 1
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# exposure mode.
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# values: normal, short, long, custom
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rpiCameraExposure: normal
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# auto-white-balance mode.
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# values: auto, incandescent, tungsten, fluorescent, indoor, daylight, cloudy, custom
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rpiCameraAWB: auto
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# denoise operating mode.
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# values: off, cdn_off, cdn_fast, cdn_hq
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rpiCameraDenoise: "off"
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# fixed shutter speed, in microseconds.
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rpiCameraShutter: 0
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# metering mode of the AEC/AGC algorithm.
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# values: centre, spot, matrix, custom
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rpiCameraMetering: centre
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# fixed gain
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rpiCameraGain: 0
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# EV compensation of the image [-10, 10]
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rpiCameraEV: 0
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# Region of interest, in format x,y,width,height
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rpiCameraROI:
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# whether to enable HDR on Raspberry Camera 3.
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rpiCameraHDR: false
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# tuning file
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rpiCameraTuningFile:
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# sensor mode, in format [width]:[height]:[bit-depth]:[packing]
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# bit-depth and packing are optional.
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rpiCameraMode:
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# frames per second
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rpiCameraFPS: 30
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# period between IDR frames
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rpiCameraIDRPeriod: 60
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# bitrate
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rpiCameraBitrate: 1000000
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# H264 profile
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rpiCameraProfile: main
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# H264 level
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rpiCameraLevel: '4.1'
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# Autofocus mode
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# values: auto, manual, continuous
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rpiCameraAfMode: auto
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# Autofocus range
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# values: normal, macro, full
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rpiCameraAfRange: normal
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# Autofocus speed
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# values: normal, fast
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rpiCameraAfSpeed: normal
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# Lens position (for manual autofocus only), will be set to focus to a specific distance
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# calculated by the following formula: d = 1 / value
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# Examples: 0 moves the lens to infinity.
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# 0.5 moves the lens to focus on objects 2m away.
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# 2 moves the lens to focus on objects 50cm away.
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rpiCameraLensPosition: 0.0
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# Specifies the autofocus window, in the form x,y,width,height where the coordinates
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# are given as a proportion of the entire image.
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rpiCameraAfWindow:
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# enables printing text on each frame.
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rpiCameraTextOverlayEnable: false
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# text that is printed on each frame.
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# format is the one of the strftime() function.
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rpiCameraTextOverlay: '%Y-%m-%d %H:%M:%S - MediaMTX'
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###############################################
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# External commands path settings
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# Command to run when this path is initialized.
|
|
# This can be used to publish a stream and keep it always opened.
|
|
# This is terminated with SIGINT when the program closes.
|
|
# The following environment variables are available:
|
|
# * MTX_PATH: path name
|
|
# * RTSP_PORT: RTSP server port
|
|
# * G1, G2, ...: regular expression groups, if path name is
|
|
# a regular expression.
|
|
runOnInit:
|
|
# Restart the command if it exits.
|
|
runOnInitRestart: no
|
|
|
|
# Command to run when this path is requested.
|
|
# This can be used to publish a stream on demand.
|
|
# This is terminated with SIGINT when the path is not requested anymore.
|
|
# The following environment variables are available:
|
|
# * MTX_PATH: path name
|
|
# * RTSP_PORT: RTSP server port
|
|
# * G1, G2, ...: regular expression groups, if path name is
|
|
# a regular expression.
|
|
runOnDemand:
|
|
# Restart the command if it exits.
|
|
runOnDemandRestart: no
|
|
# Readers will be put on hold until the runOnDemand command starts publishing
|
|
# or until this amount of time has passed.
|
|
runOnDemandStartTimeout: 10s
|
|
# The command will be closed when there are no
|
|
# readers connected and this amount of time has passed.
|
|
runOnDemandCloseAfter: 10s
|
|
|
|
# Command to run when the stream is ready to be read, whenever it is
|
|
# published by a client or pulled from a server / camera.
|
|
# This is terminated with SIGINT when the stream is not ready anymore.
|
|
# The following environment variables are available:
|
|
# * MTX_PATH: path name
|
|
# * RTSP_PORT: RTSP server port
|
|
# * G1, G2, ...: regular expression groups, if path name is
|
|
# a regular expression.
|
|
runOnReady:
|
|
# Restart the command if it exits.
|
|
runOnReadyRestart: no
|
|
# Command to run when the stream is not available anymore.
|
|
# Environment variables are the same of runOnReady.
|
|
runOnNotReady:
|
|
|
|
# Command to run when a client starts reading.
|
|
# This is terminated with SIGINT when a client stops reading.
|
|
# The following environment variables are available:
|
|
# * MTX_PATH: path name
|
|
# * RTSP_PORT: RTSP server port
|
|
# * G1, G2, ...: regular expression groups, if path name is
|
|
# a regular expression.
|
|
runOnRead:
|
|
# Restart the command if it exits.
|
|
runOnReadRestart: no
|
|
# Command to run when a client stops reading.
|
|
# Environment variables are the same of runOnRead.
|
|
runOnUnread:
|