From e519e40a9956e97004afda1648034f0cbafa132a Mon Sep 17 00:00:00 2001 From: Rik Veenboer Date: Mon, 1 Sep 2025 09:23:25 +0200 Subject: [PATCH] configure mediamtx --- compose.mediamtx.yaml | 22 ++ mediamtx/mediamtx.yml | 769 ++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 791 insertions(+) create mode 100644 compose.mediamtx.yaml create mode 100644 mediamtx/mediamtx.yml diff --git a/compose.mediamtx.yaml b/compose.mediamtx.yaml new file mode 100644 index 0000000..7e909f5 --- /dev/null +++ b/compose.mediamtx.yaml @@ -0,0 +1,22 @@ +services: + mediamtx: + image: bluenviron/mediamtx:1.14.0-ffmpeg + container_name: mediamtx + environment: + MTX_RTSPTRANSPORTS: tcp + MTX_WEBRTCADDITIONALHOSTS: 192.168.2.200 + ports: + - "8554:8554" + - "1935:1935" + - "9888:8888" + - "9889:8889" + - "8890:8890/udp" + - "8189:8189/udp" + stdin_open: true + tty: true + volumes: + - /opt/mediamtx/mediamtx.yml:/mediamtx.yml:ro + - /mnt/yotta/radon/mediamtx:/recordings + restart: unless-stopped + +# vlc --network-caching=50 rtsp://192.168.2.200:8554/mystream diff --git a/mediamtx/mediamtx.yml b/mediamtx/mediamtx.yml new file mode 100644 index 0000000..007e0a9 --- /dev/null +++ b/mediamtx/mediamtx.yml @@ -0,0 +1,769 @@ +############################################### +# Global settings + +# Settings in this section are applied anywhere. + +############################################### +# Global settings -> General + +# Verbosity of the program; available values are "error", "warn", "info", "debug". +logLevel: info +# Destinations of log messages; available values are "stdout", "file" and "syslog". +logDestinations: [stdout] +# If "file" is in logDestinations, this is the file which will receive the logs. +logFile: mediamtx.log +# If "syslog" is in logDestinations, use prefix for logs. +sysLogPrefix: mediamtx + +# Timeout of read operations. +readTimeout: 10s +# Timeout of write operations. +writeTimeout: 10s +# Size of the queue of outgoing packets. +# A higher value allows to increase throughput, a lower value allows to save RAM. +writeQueueSize: 512 +# Maximum size of outgoing UDP packets. +# This can be decreased to avoid fragmentation on networks with a low UDP MTU. +udpMaxPayloadSize: 1472 + +# Command to run when a client connects to the server. +# This is terminated with SIGINT when a client disconnects from the server. +# The following environment variables are available: +# * MTX_CONN_TYPE: connection type +# * MTX_CONN_ID: connection ID +# * RTSP_PORT: RTSP server port +runOnConnect: +# Restart the command if it exits. +runOnConnectRestart: no +# Command to run when a client disconnects from the server. +# Environment variables are the same of runOnConnect. +runOnDisconnect: + +############################################### +# Global settings -> Authentication + +# Authentication method. Available values are: +# * internal: users are stored in the configuration file +# * http: an external HTTP URL is contacted to perform authentication +# * jwt: an external identity server provides authentication through JWTs +authMethod: internal + +# Internal authentication. +# list of users. +authInternalUsers: + # Default unprivileged user. + # Username. 'any' means any user, including anonymous ones. +- user: any + # Password. Not used in case of 'any' user. + pass: + # IPs or networks allowed to use this user. An empty list means any IP. + ips: [] + # List of permissions. + permissions: + # Available actions are: publish, read, playback, api, metrics, pprof. + - action: publish + # Paths can be set to further restrict access to a specific path. + # An empty path means any path. + # Regular expressions can be used by using a tilde as prefix. + path: + - action: read + path: + - action: playback + path: + + # Default administrator. + # This allows to use API, metrics and PPROF without authentication, + # if the IP is localhost. +- user: any + pass: + ips: ['127.0.0.1', '::1'] + permissions: + - action: api + - action: metrics + - action: pprof + +# HTTP-based authentication. +# URL called to perform authentication. Every time a user wants +# to authenticate, the server calls this URL with the POST method +# and a body containing: +# { +# "user": "user", +# "password": "password", +# "token": "token", +# "ip": "ip", +# "action": "publish|read|playback|api|metrics|pprof", +# "path": "path", +# "protocol": "rtsp|rtmp|hls|webrtc|srt", +# "id": "id", +# "query": "query" +# } +# If the response code is 20x, authentication is accepted, otherwise +# it is discarded. +authHTTPAddress: +# Actions to exclude from HTTP-based authentication. +# Format is the same as the one of user permissions. +authHTTPExclude: +- action: api +- action: metrics +- action: pprof + +# JWT-based authentication. +# Users have to login through an external identity server and obtain a JWT. +# This JWT must contain the claim "mediamtx_permissions" with permissions, +# for instance: +# { +# "mediamtx_permissions": [ +# { +# "action": "publish", +# "path": "somepath" +# } +# ] +# } +# Users are expected to pass the JWT in the Authorization header, password or query parameter. +# This is the JWKS URL that will be used to pull (once) the public key that allows +# to validate JWTs. +authJWTJWKS: +# If the JWKS URL has a self-signed or invalid certificate, +# you can provide the fingerprint of the certificate in order to +# validate it anyway. It can be obtained by running: +# openssl s_client -connect jwt_jwks_domain:443 /dev/null | sed -n '/BEGIN/,/END/p' > server.crt +# openssl x509 -in server.crt -noout -fingerprint -sha256 | cut -d "=" -f2 | tr -d ':' +authJWTJWKSFingerprint: +# name of the claim that contains permissions. +authJWTClaimKey: mediamtx_permissions +# Actions to exclude from JWT-based authentication. +# Format is the same as the one of user permissions. +authJWTExclude: [] +# allow passing the JWT through query parameters of HTTP requests (i.e. ?jwt=JWT). +# This is a security risk. +authJWTInHTTPQuery: true + +############################################### +# Global settings -> Control API + +# Enable controlling the server through the Control API. +api: no +# Address of the Control API listener. +apiAddress: :9997 +# Enable TLS/HTTPS on the Control API server. +apiEncryption: no +# Path to the server key. This is needed only when encryption is yes. +# This can be generated with: +# openssl genrsa -out server.key 2048 +# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 +apiServerKey: server.key +# Path to the server certificate. +apiServerCert: server.crt +# Value of the Access-Control-Allow-Origin header provided in every HTTP response. +apiAllowOrigin: '*' +# List of IPs or CIDRs of proxies placed before the HTTP server. +# If the server receives a request from one of these entries, IP in logs +# will be taken from the X-Forwarded-For header. +apiTrustedProxies: [] + +############################################### +# Global settings -> Metrics + +# Enable Prometheus-compatible metrics. +metrics: no +# Address of the metrics HTTP listener. +metricsAddress: :9998 +# Enable TLS/HTTPS on the Metrics server. +metricsEncryption: no +# Path to the server key. This is needed only when encryption is yes. +# This can be generated with: +# openssl genrsa -out server.key 2048 +# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 +metricsServerKey: server.key +# Path to the server certificate. +metricsServerCert: server.crt +# Value of the Access-Control-Allow-Origin header provided in every HTTP response. +metricsAllowOrigin: '*' +# List of IPs or CIDRs of proxies placed before the HTTP server. +# If the server receives a request from one of these entries, IP in logs +# will be taken from the X-Forwarded-For header. +metricsTrustedProxies: [] + +############################################### +# Global settings -> PPROF + +# Enable pprof-compatible endpoint to monitor performances. +pprof: no +# Address of the pprof listener. +pprofAddress: :9999 +# Enable TLS/HTTPS on the pprof server. +pprofEncryption: no +# Path to the server key. This is needed only when encryption is yes. +# This can be generated with: +# openssl genrsa -out server.key 2048 +# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 +pprofServerKey: server.key +# Path to the server certificate. +pprofServerCert: server.crt +# Value of the Access-Control-Allow-Origin header provided in every HTTP response. +pprofAllowOrigin: '*' +# List of IPs or CIDRs of proxies placed before the HTTP server. +# If the server receives a request from one of these entries, IP in logs +# will be taken from the X-Forwarded-For header. +pprofTrustedProxies: [] + +############################################### +# Global settings -> Playback server + +# Enable downloading recordings from the playback server. +playback: no +# Address of the playback server listener. +playbackAddress: :9996 +# Enable TLS/HTTPS on the playback server. +playbackEncryption: no +# Path to the server key. This is needed only when encryption is yes. +# This can be generated with: +# openssl genrsa -out server.key 2048 +# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 +playbackServerKey: server.key +# Path to the server certificate. +playbackServerCert: server.crt +# Value of the Access-Control-Allow-Origin header provided in every HTTP response. +playbackAllowOrigin: '*' +# List of IPs or CIDRs of proxies placed before the HTTP server. +# If the server receives a request from one of these entries, IP in logs +# will be taken from the X-Forwarded-For header. +playbackTrustedProxies: [] + +############################################### +# Global settings -> RTSP server + +# Enable publishing and reading streams with the RTSP protocol. +rtsp: yes +# List of enabled RTSP transport protocols. +# UDP is the most performant, but doesn't work when there's a NAT/firewall between +# server and clients. +# UDP-multicast allows to save bandwidth when clients are all in the same LAN. +# TCP is the most versatile. +# The handshake is always performed with TCP. +rtspTransports: [udp, multicast, tcp] +# Use secure protocol variants (RTSPS, TLS, SRTP). +# Available values are "no", "strict", "optional". +rtspEncryption: "no" +# Address of the TCP/RTSP listener. This is needed only when encryption is "no" or "optional". +rtspAddress: :8554 +# Address of the TCP/TLS/RTSPS listener. This is needed only when encryption is "strict" or "optional". +rtspsAddress: :8322 +# Address of the UDP/RTP listener. This is needed only when "udp" is in rtspTransports. +rtpAddress: :8000 +# Address of the UDP/RTCP listener. This is needed only when "udp" is in rtspTransports. +rtcpAddress: :8001 +# IP range of all UDP-multicast listeners. This is needed only when "multicast" is in rtspTransports. +multicastIPRange: 224.1.0.0/16 +# Port of all UDP-multicast/RTP listeners. This is needed only when "multicast" is in rtspTransports. +multicastRTPPort: 8002 +# Port of all UDP-multicast/RTCP listeners. This is needed only when "multicast" is in rtspTransports. +multicastRTCPPort: 8003 +# Address of the UDP/SRTP listener. This is needed only when "udp" is in rtspTransports and encryption is enabled. +srtpAddress: :8004 +# Address of the UDP/SRTCP listener. This is needed only when "udp" is in rtspTransports and encryption is enabled. +srtcpAddress: :8005 +# Port of all UDP-multicast/SRTP listeners. This is needed only when "multicast" is in rtspTransports and encryption is enabled. +multicastSRTPPort: 8006 +# Port of all UDP-multicast/SRTCP listeners. This is needed only when "multicast" is in rtspTransports and encryption is enabled. +multicastSRTCPPort: 8007 +# Path to the server key. This is needed only when encryption is "strict" or "optional". +# This can be generated with: +# openssl genrsa -out server.key 2048 +# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 +rtspServerKey: server.key +# Path to the server certificate. This is needed only when encryption is "strict" or "optional". +rtspServerCert: server.crt +# Authentication methods. Available are "basic" and "digest". +# "digest" doesn't provide any additional security and is available for compatibility only. +rtspAuthMethods: [basic] +# Size of the UDP buffer of the RTSP server. +# This can be increased to mitigate packet losses. +# It defaults to the default value of the operating system. +rtspUDPReadBufferSize: 0 + +############################################### +# Global settings -> RTMP server + +# Enable publishing and reading streams with the RTMP protocol. +rtmp: yes +# Address of the RTMP listener. This is needed only when encryption is "no" or "optional". +rtmpAddress: :1935 +# Encrypt connections with TLS (RTMPS). +# Available values are "no", "strict", "optional". +rtmpEncryption: "no" +# Address of the RTMPS listener. This is needed only when encryption is "strict" or "optional". +rtmpsAddress: :1936 +# Path to the server key. This is needed only when encryption is "strict" or "optional". +# This can be generated with: +# openssl genrsa -out server.key 2048 +# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 +rtmpServerKey: server.key +# Path to the server certificate. This is needed only when encryption is "strict" or "optional". +rtmpServerCert: server.crt + +############################################### +# Global settings -> HLS server + +# Enable reading streams with the HLS protocol. +hls: yes +# Address of the HLS listener. +hlsAddress: :8888 +# Enable TLS/HTTPS on the HLS server. +# This is required for Low-Latency HLS. +hlsEncryption: no +# Path to the server key. This is needed only when encryption is yes. +# This can be generated with: +# openssl genrsa -out server.key 2048 +# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 +hlsServerKey: server.key +# Path to the server certificate. +hlsServerCert: server.crt +# Value of the Access-Control-Allow-Origin header provided in every HTTP response. +# This allows to play the HLS stream from an external website. +hlsAllowOrigin: '*' +# List of IPs or CIDRs of proxies placed before the HLS server. +# If the server receives a request from one of these entries, IP in logs +# will be taken from the X-Forwarded-For header. +hlsTrustedProxies: [] +# By default, HLS is generated only when requested by a user. +# This option allows to generate it always, avoiding the delay between request and generation. +hlsAlwaysRemux: no +# Variant of the HLS protocol to use. Available options are: +# * mpegts - uses MPEG-TS segments, for maximum compatibility. +# * fmp4 - uses fragmented MP4 segments, more efficient. +# * lowLatency - uses Low-Latency HLS. +hlsVariant: lowLatency +# Number of HLS segments to keep on the server. +# Segments allow to seek through the stream. +# Their number doesn't influence latency. +hlsSegmentCount: 7 +# Minimum duration of each segment. +# A player usually puts 3 segments in a buffer before reproducing the stream. +# The final segment duration is also influenced by the interval between IDR frames, +# since the server changes the duration in order to include at least one IDR frame +# in each segment. +hlsSegmentDuration: 1s +# Minimum duration of each part. +# A player usually puts 3 parts in a buffer before reproducing the stream. +# Parts are used in Low-Latency HLS in place of segments. +# Part duration is influenced by the distance between video/audio samples +# and is adjusted in order to produce segments with a similar duration. +hlsPartDuration: 200ms +# Maximum size of each segment. +# This prevents RAM exhaustion. +hlsSegmentMaxSize: 50M +# Directory in which to save segments, instead of keeping them in the RAM. +# This decreases performance, since reading from disk is less performant than +# reading from RAM, but allows to save RAM. +hlsDirectory: '' +# The muxer will be closed when there are no +# reader requests and this amount of time has passed. +hlsMuxerCloseAfter: 60s + +############################################### +# Global settings -> WebRTC server + +# Enable publishing and reading streams with the WebRTC protocol. +webrtc: yes +# Address of the WebRTC HTTP listener. +webrtcAddress: :8889 +# Enable TLS/HTTPS on the WebRTC server. +webrtcEncryption: no +# Path to the server key. +# This can be generated with: +# openssl genrsa -out server.key 2048 +# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650 +webrtcServerKey: server.key +# Path to the server certificate. +webrtcServerCert: server.crt +# Value of the Access-Control-Allow-Origin header provided in every HTTP response. +# This allows to play the WebRTC stream from an external website. +webrtcAllowOrigin: '*' +# List of IPs or CIDRs of proxies placed before the WebRTC server. +# If the server receives a request from one of these entries, IP in logs +# will be taken from the X-Forwarded-For header. +webrtcTrustedProxies: [] +# Address of a local UDP listener that will receive connections. +# Use a blank string to disable. +webrtcLocalUDPAddress: :8189 +# Address of a local TCP listener that will receive connections. +# This is disabled by default since TCP is less efficient than UDP and +# introduces a progressive delay when network is congested. +webrtcLocalTCPAddress: '' +# WebRTC clients need to know the IP of the server. +# Gather IPs from interfaces and send them to clients. +webrtcIPsFromInterfaces: yes +# List of interfaces whose IPs will be sent to clients. +# An empty value means to use all available interfaces. +webrtcIPsFromInterfacesList: [] +# List of additional hosts or IPs to send to clients. +webrtcAdditionalHosts: [] +# ICE servers. Needed only when local listeners can't be reached by clients. +# STUN servers allows to obtain and share the public IP of the server. +# TURN/TURNS servers forces all traffic through them. +webrtcICEServers2: [] + # - url: stun:stun.l.google.com:19302 + # if user is "AUTH_SECRET", then authentication is secret based. + # the secret must be inserted into the password field. + # username: '' + # password: '' + # clientOnly: false +# Time to wait for the WebRTC handshake to complete. +webrtcHandshakeTimeout: 10s +# Maximum time to gather video tracks. +webrtcTrackGatherTimeout: 2s +# The maximum time to gather STUN candidates. +webrtcSTUNGatherTimeout: 5s + +############################################### +# Global settings -> SRT server + +# Enable publishing and reading streams with the SRT protocol. +srt: yes +# Address of the SRT listener. +srtAddress: :8890 + +############################################### +# Default path settings + +# Settings in "pathDefaults" are applied anywhere, +# unless they are overridden in "paths". +pathDefaults: + + ############################################### + # Default path settings -> General + + # Source of the stream. This can be: + # * publisher -> the stream is provided by a RTSP, RTMP, WebRTC or SRT client + # * rtsp://existing-url -> the stream is pulled from another RTSP server / camera + # * rtsps://existing-url -> the stream is pulled from another RTSP server / camera with RTSPS + # * rtmp://existing-url -> the stream is pulled from another RTMP server / camera + # * rtmps://existing-url -> the stream is pulled from another RTMP server / camera with RTMPS + # * http://existing-url/stream.m3u8 -> the stream is pulled from another HLS server / camera + # * https://existing-url/stream.m3u8 -> the stream is pulled from another HLS server / camera with HTTPS + # * udp+mpegts://ip:port -> the stream is pulled from MPEG-TS over UDP, by listening on the specified address + # * unix+mpegts://socket -> the stream is pulled from MPEG-TS over Unix socket, by using the socket + # * udp+rtp://ip:port -> the stream is pulled from RTP over UDP, by listening on the specified address + # * unix+rtp://socket -> the stream is pulled from RTP over Unix socket, by using the socket + # * srt://existing-url -> the stream is pulled from another SRT server / camera + # * whep://existing-url -> the stream is pulled from another WebRTC server / camera + # * wheps://existing-url -> the stream is pulled from another WebRTC server / camera with HTTPS + # * redirect -> the stream is provided by another path or server + # * rpiCamera -> the stream is provided by a Raspberry Pi Camera + # The following variables can be used in the source string: + # * $MTX_QUERY: query parameters (passed by first reader) + # * $G1, $G2, ...: regular expression groups, if path name is + # a regular expression. + source: publisher + # If the source is a URL, and the source certificate is self-signed + # or invalid, you can provide the fingerprint of the certificate in order to + # validate it anyway. It can be obtained by running: + # openssl s_client -connect source_ip:source_port /dev/null | sed -n '/BEGIN/,/END/p' > server.crt + # openssl x509 -in server.crt -noout -fingerprint -sha256 | cut -d "=" -f2 | tr -d ':' + sourceFingerprint: + # If the source is a URL, it will be pulled only when at least + # one reader is connected, saving bandwidth. + sourceOnDemand: no + # If sourceOnDemand is "yes", readers will be put on hold until the source is + # ready or until this amount of time has passed. + sourceOnDemandStartTimeout: 10s + # If sourceOnDemand is "yes", the source will be closed when there are no + # readers connected and this amount of time has passed. + sourceOnDemandCloseAfter: 10s + # Maximum number of readers. Zero means no limit. + maxReaders: 0 + # SRT encryption passphrase required to read from this path. + srtReadPassphrase: + # If the stream is not available, redirect readers to this path. + # It can be can be a relative path (i.e. /otherstream) or an absolute RTSP URL. + fallback: + # Route original absolute timestamps of RTSP and WebRTC frames, instead of replacing them. + useAbsoluteTimestamp: false + + ############################################### + # Default path settings -> Record + + # Record streams to disk. + record: yes + # Path of recording segments. + # Extension is added automatically. + # Available variables are %path (path name), %Y %m %d (year, month, day), + # %H %M %S (hours, minutes, seconds), %f (microseconds), %z (time zone), %s (unix epoch). + recordPath: ./recordings/%path/%Y-%m-%d_%H-%M-%S-%f + # Format of recorded segments. + # Available formats are "fmp4" (fragmented MP4) and "mpegts" (MPEG-TS). + recordFormat: fmp4 + # fMP4 segments are concatenation of small MP4 files (parts), each with this duration. + # MPEG-TS segments are concatenation of 188-bytes packets, flushed to disk with this period. + # When a system failure occurs, the last part gets lost. + # Therefore, the part duration is equal to the RPO (recovery point objective). + recordPartDuration: 1s + # This prevents RAM exhaustion. + recordMaxPartSize: 50M + # Minimum duration of each segment. + recordSegmentDuration: 1h + # Delete segments after this timespan. + # Set to 0s to disable automatic deletion. + recordDeleteAfter: 1d + + ############################################### + # Default path settings -> Publisher source (when source is "publisher") + + # Allow another client to disconnect the current publisher and publish in its place. + overridePublisher: yes + # SRT encryption passphrase required to publish to this path. + srtPublishPassphrase: + + ############################################### + # Default path settings -> RTSP source (when source is a RTSP or a RTSPS URL) + + # Transport protocol used to pull the stream. available values are "automatic", "udp", "multicast", "tcp". + rtspTransport: automatic + # Support sources that don't provide server ports or use random server ports. This is a security issue + # and must be used only when interacting with sources that require it. + rtspAnyPort: no + # Range header to send to the source, in order to start streaming from the specified offset. + # available values: + # * clock: Absolute time + # * npt: Normal Play Time + # * smpte: SMPTE timestamps relative to the start of the recording + rtspRangeType: + # Available values: + # * clock: UTC ISO 8601 combined date and time string, e.g. 20230812T120000Z + # * npt: duration such as "300ms", "1.5m" or "2h45m", valid time units are "ns", "us" (or "µs"), "ms", "s", "m", "h" + # * smpte: duration such as "300ms", "1.5m" or "2h45m", valid time units are "ns", "us" (or "µs"), "ms", "s", "m", "h" + rtspRangeStart: + # Size of the UDP buffer of the RTSP client. + # This can be increased to mitigate packet losses. + # It defaults to the default value of the operating system. + rtspUDPReadBufferSize: 0 + + ############################################### + # Default path settings -> MPEG-TS source (when source is MPEG-TS) + + # Size of the UDP buffer of the MPEG-TS client. + # This can be increased to mitigate packet losses. + # It defaults to the default value of the operating system. + mpegtsUDPReadBufferSize: 0 + + ############################################### + # Default path settings -> RTP source (when source is RTP) + + # session description protocol (SDP) of the RTP stream. + rtpSDP: + # Size of the UDP buffer of the RTP client. + # This can be increased to mitigate packet losses. + # It defaults to the default value of the operating system. + rtpUDPReadBufferSize: 0 + + ############################################### + # Default path settings -> Redirect source (when source is "redirect") + + # path which clients will be redirected to. + # It can be can be a relative path (i.e. /otherstream) or an absolute RTSP URL. + sourceRedirect: + + ############################################### + # Default path settings -> Raspberry Pi Camera source (when source is "rpiCamera") + + # ID of the camera. + rpiCameraCamID: 0 + # Whether this is a secondary stream. + rpiCameraSecondary: false + # Width of frames. + rpiCameraWidth: 1920 + # Height of frames. + rpiCameraHeight: 1080 + # Flip horizontally. + rpiCameraHFlip: false + # Flip vertically. + rpiCameraVFlip: false + # Brightness [-1, 1]. + rpiCameraBrightness: 0 + # Contrast [0, 16]. + rpiCameraContrast: 1 + # Saturation [0, 16]. + rpiCameraSaturation: 1 + # Sharpness [0, 16]. + rpiCameraSharpness: 1 + # Exposure mode. + # values: normal, short, long, custom. + rpiCameraExposure: normal + # Auto-white-balance mode. + # (auto, incandescent, tungsten, fluorescent, indoor, daylight, cloudy or custom). + rpiCameraAWB: auto + # Auto-white-balance fixed gains. This can be used in place of rpiCameraAWB. + # format: [red,blue]. + rpiCameraAWBGains: [0, 0] + # Denoise operating mode (off, cdn_off, cdn_fast, cdn_hq). + rpiCameraDenoise: "off" + # Fixed shutter speed, in microseconds. + rpiCameraShutter: 0 + # Metering mode of the AEC/AGC algorithm (centre, spot, matrix or custom). + rpiCameraMetering: centre + # Fixed gain. + rpiCameraGain: 0 + # EV compensation of the image in range [-10, 10]. + rpiCameraEV: 0 + # Region of interest, in format x,y,width,height (all normalized between 0 and 1). + rpiCameraROI: + # Whether to enable HDR on Raspberry Camera 3. + rpiCameraHDR: false + # Tuning file. + rpiCameraTuningFile: + # Sensor mode, in format [width]:[height]:[bit-depth]:[packing] + # bit-depth and packing are optional. + rpiCameraMode: + # frames per second. + rpiCameraFPS: 30 + # Autofocus mode (auto, manual or continuous). + rpiCameraAfMode: continuous + # Autofocus range (normal, macro or full). + rpiCameraAfRange: normal + # Autofocus speed (normal or fast). + rpiCameraAfSpeed: normal + # Lens position (for manual autofocus only), will be set to focus to a specific distance + # calculated by the following formula: d = 1 / value + # Examples: 0 moves the lens to infinity. + # 0.5 moves the lens to focus on objects 2m away. + # 2 moves the lens to focus on objects 50cm away. + rpiCameraLensPosition: 0.0 + # Autofocus window, in the form x,y,width,height where the coordinates + # are given as a proportion of the entire image. + rpiCameraAfWindow: + # Manual flicker correction period, in microseconds. + rpiCameraFlickerPeriod: 0 + # Enables printing text on each frame. + rpiCameraTextOverlayEnable: false + # Text that is printed on each frame. + # format is the one of the strftime() function. + rpiCameraTextOverlay: '%Y-%m-%d %H:%M:%S - MediaMTX' + # Codec (auto, hardwareH264, softwareH264 or mjpeg). + # When is "auto" and stream is primary, it defaults to hardwareH264 (if available) or softwareH264. + # When is "auto" and stream is secondary, it defaults to mjpeg. + rpiCameraCodec: auto + # Period between IDR frames (when codec is hardwareH264 or softwareH264). + rpiCameraIDRPeriod: 60 + # Bitrate (when codec is hardwareH264 or softwareH264). + rpiCameraBitrate: 5000000 + # Hardware H264 profile (baseline, main or high) (when codec is hardwareH264). + rpiCameraHardwareH264Profile: main + # Hardware H264 level (4.0, 4.1 or 4.2) (when codec is hardwareH264). + rpiCameraHardwareH264Level: '4.1' + # Software H264 profile (baseline, main or high) (when codec is softwareH264). + rpiCameraSoftwareH264Profile: baseline + # Software H264 level (4.0, 4.1 or 4.2) (when codec is softwareH264). + rpiCameraSoftwareH264Level: '4.1' + # M-JPEG JPEG quality (when codec is mjpeg). + rpiCameraMJPEGQuality: 60 + + ############################################### + # Default path settings -> Hooks + + # Command to run when this path is initialized. + # This can be used to publish a stream when the server is launched. + # 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 by a reader + # and no one is publishing to this path yet. + # This can be used to publish a stream on demand. + # This is terminated with SIGINT when there are no readers anymore. + # The following environment variables are available: + # * MTX_PATH: path name + # * MTX_QUERY: query parameters (passed by first reader) + # * 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 there are no readers anymore. + # Environment variables are the same of runOnDemand. + runOnUnDemand: + + # 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 + # * MTX_QUERY: query parameters (passed by publisher) + # * MTX_SOURCE_TYPE: source type + # * MTX_SOURCE_ID: source ID + # * 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 + # * MTX_QUERY: query parameters (passed by reader) + # * MTX_READER_TYPE: reader type + # * MTX_READER_ID: reader ID + # * 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: + + # Command to run when a recording segment is created. + # The following environment variables are available: + # * MTX_PATH: path name + # * MTX_SEGMENT_PATH: segment file path + # * RTSP_PORT: RTSP server port + # * G1, G2, ...: regular expression groups, if path name is + # a regular expression. + runOnRecordSegmentCreate: + + # Command to run when a recording segment is complete. + # The following environment variables are available: + # * MTX_PATH: path name + # * MTX_SEGMENT_PATH: segment file path + # * MTX_SEGMENT_DURATION: segment duration + # * RTSP_PORT: RTSP server port + # * G1, G2, ...: regular expression groups, if path name is + # a regular expression. + runOnRecordSegmentComplete: + +############################################### +# Path settings + +# Settings in "paths" are applied to specific paths, and the map key +# is the name of the path. +# Any setting in "pathDefaults" can be overridden here. +# It's possible to use regular expressions by using a tilde as prefix, +# for example "~^(test1|test2)$" will match both "test1" and "test2", +# for example "~^prefix" will match all paths that start with "prefix". +paths: + # example: + # my_camera: + # source: rtsp://my_camera + + # Settings under path "all_others" are applied to all paths that + # do not match another entry. + all_others: