mirror of
https://github.com/bluenviron/mediamtx
synced 2024-12-11 17:28:55 +00:00
6b8c65f5d7
this fixes a regression introduced in v0.23.0.
459 lines
19 KiB
YAML
459 lines
19 KiB
YAML
|
|
###############################################
|
|
# General parameters
|
|
|
|
# Sets the 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
|
|
|
|
# Timeout of read operations.
|
|
readTimeout: 10s
|
|
# Timeout of write operations.
|
|
writeTimeout: 10s
|
|
# Number of read buffers.
|
|
# A higher value allows a wider throughput, a lower value allows to save RAM.
|
|
readBufferCount: 512
|
|
# Maximum size of payload of outgoing UDP packets.
|
|
# This can be decreased to avoid fragmentation on networks with a low UDP MTU.
|
|
udpMaxPayloadSize: 1472
|
|
|
|
# HTTP URL to perform external authentication.
|
|
# Every time a user wants to authenticate, the server calls this URL
|
|
# with the POST method and a body containing:
|
|
# {
|
|
# "ip": "ip",
|
|
# "user": "user",
|
|
# "password": "password",
|
|
# "path": "path",
|
|
# "protocol": "rtsp|rtmp|hls|webrtc",
|
|
# "id": "id",
|
|
# "action": "read|publish",
|
|
# "query": "query"
|
|
# }
|
|
# If the response code is 20x, authentication is accepted, otherwise
|
|
# it is discarded.
|
|
externalAuthenticationURL:
|
|
|
|
# Enable the HTTP API.
|
|
api: no
|
|
# Address of the API listener.
|
|
apiAddress: 127.0.0.1:9997
|
|
|
|
# Enable Prometheus-compatible metrics.
|
|
metrics: no
|
|
# Address of the metrics listener.
|
|
metricsAddress: 127.0.0.1:9998
|
|
|
|
# Enable pprof-compatible endpoint to monitor performances.
|
|
pprof: no
|
|
# Address of the pprof listener.
|
|
pprofAddress: 127.0.0.1:9999
|
|
|
|
# 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:
|
|
# * RTSP_PORT: server port
|
|
runOnConnect:
|
|
# Restart the command if it exits suddenly.
|
|
runOnConnectRestart: no
|
|
|
|
###############################################
|
|
# RTSP parameters
|
|
|
|
# Disable support for the RTSP protocol.
|
|
rtspDisable: no
|
|
# 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, and doesn't support encryption.
|
|
# UDP-multicast allows to save bandwidth when clients are all in the same LAN.
|
|
# TCP is the most versatile, and does support encryption.
|
|
# The handshake is always performed with TCP.
|
|
protocols: [udp, multicast, tcp]
|
|
# Encrypt handshakes and TCP streams with TLS (RTSPS).
|
|
# Available values are "no", "strict", "optional".
|
|
encryption: "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 protocols.
|
|
rtpAddress: :8000
|
|
# Address of the UDP/RTCP listener. This is needed only when "udp" is in protocols.
|
|
rtcpAddress: :8001
|
|
# IP range of all UDP-multicast listeners. This is needed only when "multicast" is in protocols.
|
|
multicastIPRange: 224.1.0.0/16
|
|
# Port of all UDP-multicast/RTP listeners. This is needed only when "multicast" is in protocols.
|
|
multicastRTPPort: 8002
|
|
# Port of all UDP-multicast/RTCP listeners. This is needed only when "multicast" is in protocols.
|
|
multicastRTCPPort: 8003
|
|
# 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
|
|
serverKey: server.key
|
|
# Path to the server certificate. This is needed only when encryption is "strict" or "optional".
|
|
serverCert: server.crt
|
|
# Authentication methods. Available are "basic" and "digest".
|
|
# "digest" doesn't provide any additional security and is available for compatibility reasons only.
|
|
authMethods: [basic]
|
|
|
|
###############################################
|
|
# RTMP parameters
|
|
|
|
# Disable support for the RTMP protocol.
|
|
rtmpDisable: no
|
|
# 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
|
|
|
|
###############################################
|
|
# HLS parameters
|
|
|
|
# Disable support for the HLS protocol.
|
|
hlsDisable: no
|
|
# 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
|
|
# 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
|
|
# 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: []
|
|
# 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: ''
|
|
|
|
###############################################
|
|
# WebRTC parameters
|
|
|
|
# Disable support for the WebRTC protocol.
|
|
webrtcDisable: no
|
|
# Address of the WebRTC 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: []
|
|
# List of ICE servers, in format type:user:password:host:port or type:host:port.
|
|
# type can be "stun", "turn" or "turns".
|
|
# STUN servers are used to obtain the public IP of server and clients. They are
|
|
# needed when server and clients are on different LANs.
|
|
# TURN servers are needed when a direct connection between server and clients
|
|
# is not possible. All traffic is routed through the chosen TURN server.
|
|
# if user is "AUTH_SECRET", then authentication is secret based.
|
|
# the secret must be inserted into the password field.
|
|
webrtcICEServers: [stun:stun.l.google.com:19302]
|
|
# List of public IP addresses that are to be used as a host.
|
|
# This is used typically for servers that are behind 1:1 D-NAT.
|
|
webrtcICEHostNAT1To1IPs: []
|
|
# Address of a ICE UDP listener in format host:port.
|
|
# If filled, ICE traffic will pass through a single UDP port,
|
|
# allowing the deployment of the server inside a container or behind a NAT.
|
|
webrtcICEUDPMuxAddress:
|
|
# Address of a ICE TCP listener in format host:port.
|
|
# If filled, ICE traffic will pass through a single TCP port,
|
|
# allowing the deployment of the server inside a container or behind a NAT.
|
|
# Setting this parameter forces usage of the TCP protocol, which is not
|
|
# optimal for WebRTC.
|
|
webrtcICETCPMuxAddress:
|
|
|
|
###############################################
|
|
# Path parameters
|
|
|
|
# These settings are path-dependent, and the map key is the name of the path.
|
|
# 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".
|
|
# The settings under the path "all" are applied to all paths that do not match
|
|
# another entry.
|
|
paths:
|
|
all:
|
|
# Source of the stream. This can be:
|
|
# * publisher -> the stream is published by a RTSP or RTMP 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
|
|
# * https://existing-url/stream.m3u8 -> the stream is pulled from another HLS server with HTTPS
|
|
# * udp://ip:port -> the stream is pulled from UDP, by listening on the specified IP and port
|
|
# * redirect -> the stream is provided by another path or server
|
|
# * rpiCamera -> the stream is provided by a Raspberry Pi Camera
|
|
source: publisher
|
|
|
|
###############################################
|
|
# General path parameters
|
|
|
|
# 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 2>/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
|
|
|
|
###############################################
|
|
# Authentication path parameters
|
|
|
|
# Username required to publish.
|
|
# SHA256-hashed values can be inserted with the "sha256:" prefix.
|
|
publishUser:
|
|
# Password required to publish.
|
|
# SHA256-hashed values can be inserted with the "sha256:" prefix.
|
|
publishPass:
|
|
# IPs or networks (x.x.x.x/24) allowed to publish.
|
|
publishIPs: []
|
|
|
|
# Username required to read.
|
|
# SHA256-hashed values can be inserted with the "sha256:" prefix.
|
|
readUser:
|
|
# password required to read.
|
|
# SHA256-hashed values can be inserted with the "sha256:" prefix.
|
|
readPass:
|
|
# IPs or networks (x.x.x.x/24) allowed to read.
|
|
readIPs: []
|
|
|
|
###############################################
|
|
# Publisher path parameters (when source is "publisher")
|
|
|
|
# do not allow another client to disconnect the current publisher and publish in its place.
|
|
disablePublisherOverride: no
|
|
# if no one is publishing, redirect readers to this path.
|
|
# It can be can be a relative path (i.e. /otherstream) or an absolute RTSP URL.
|
|
fallback:
|
|
|
|
###############################################
|
|
# RTSP path parameters (when source is a RTSP or a RTSPS URL)
|
|
|
|
# protocol used to pull the stream. available values are "automatic", "udp", "multicast", "tcp".
|
|
sourceProtocol: 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.
|
|
sourceAnyPortEnable: 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:
|
|
|
|
###############################################
|
|
# Redirect path parameters (when source is "redirect")
|
|
|
|
# RTSP URL which clients will be redirected to.
|
|
sourceRedirect:
|
|
|
|
###############################################
|
|
# Raspberry Pi Camera path parameters (when source is "rpiCamera")
|
|
|
|
# ID of the camera
|
|
rpiCameraCamID: 0
|
|
# 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.
|
|
# values: auto, incandescent, tungsten, fluorescent, indoor, daylight, cloudy, custom
|
|
rpiCameraAWB: auto
|
|
# denoise operating mode.
|
|
# values: off, cdn_off, cdn_fast, cdn_hq
|
|
rpiCameraDenoise: "off"
|
|
# fixed shutter speed, in microseconds.
|
|
rpiCameraShutter: 0
|
|
# metering mode of the AEC/AGC algorithm.
|
|
# values: centre, spot, matrix, custom
|
|
rpiCameraMetering: centre
|
|
# fixed gain
|
|
rpiCameraGain: 0
|
|
# EV compensation of the image [-10, 10]
|
|
rpiCameraEV: 0
|
|
# Region of interest, in format x,y,width,height
|
|
rpiCameraROI:
|
|
# tuning file
|
|
rpiCameraTuningFile:
|
|
# sensor mode, in format [width]:[height]:[bit-depth]:[packing]
|
|
# bit-depth and packing are optional.
|
|
rpiCameraMode:
|
|
# frames per second
|
|
rpiCameraFPS: 30
|
|
# period between IDR frames
|
|
rpiCameraIDRPeriod: 60
|
|
# bitrate
|
|
rpiCameraBitrate: 1000000
|
|
# H264 profile
|
|
rpiCameraProfile: main
|
|
# H264 level
|
|
rpiCameraLevel: '4.1'
|
|
# Autofocus mode
|
|
# values: auto, manual, continuous
|
|
rpiCameraAfMode: auto
|
|
# Autofocus range
|
|
# values: normal, macro, full
|
|
rpiCameraAfRange: normal
|
|
# Autofocus speed
|
|
# values: normal, 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
|
|
# Specifies the autofocus window, in the form x,y,width,height where the coordinates
|
|
# are given as a proportion of the entire image.
|
|
rpiCameraAfWindow:
|
|
# 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'
|
|
|
|
###############################################
|
|
# external commands path parameters
|
|
|
|
# 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:
|
|
# * RTSP_PATH: path name
|
|
# * RTSP_PORT: server port
|
|
# * G1, G2, ...: regular expression groups, if path name is
|
|
# a regular expression.
|
|
runOnInit:
|
|
# Restart the command if it exits suddenly.
|
|
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:
|
|
# * RTSP_PATH: path name
|
|
# * RTSP_PORT: server port
|
|
# * G1, G2, ...: regular expression groups, if path name is
|
|
# a regular expression.
|
|
runOnDemand:
|
|
# Restart the command if it exits suddenly.
|
|
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, whether 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:
|
|
# * RTSP_PATH: path name
|
|
# * RTSP_PORT: server port
|
|
# * G1, G2, ...: regular expression groups, if path name is
|
|
# a regular expression.
|
|
runOnReady:
|
|
# Restart the command if it exits suddenly.
|
|
runOnReadyRestart: no
|
|
|
|
# Command to run when a clients starts reading.
|
|
# This is terminated with SIGINT when a client stops reading.
|
|
# The following environment variables are available:
|
|
# * RTSP_PATH: path name
|
|
# * RTSP_PORT: server port
|
|
# * G1, G2, ...: regular expression groups, if path name is
|
|
# a regular expression.
|
|
runOnRead:
|
|
# Restart the command if it exits suddenly.
|
|
runOnReadRestart: no
|