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Ready-to-use SRT / WebRTC / RTSP / RTMP / LL-HLS media server and media proxy that allows to read, publish, proxy, record and playback video and audio streams.
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mediamtx/rtsp-simple-server.yml

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###############################################
# 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: rtsp-simple-server.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.
authMethods: [basic, digest]
###############################################
# 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:pass:host:port or type:host:port.
# type can be "stun", "turn" or "turns".
# STUN servers are used to get the public IP of both server and clients.
# TURN/TURNS servers are used as relay when a direct connection between server and clients is not possible.
# if user is "AUTH_SECRET", then authentication is secret based.
# the secret must be inserted into the pass 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 come 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 come through a single TCP port,
# allowing the deployment of the server inside a container or behind a NAT.
# At the moment, 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
# If the source is an RTSP or RTSPS URL, this is the protocol that will be used to
# pull the stream. available values are "automatic", "udp", "multicast", "tcp".
sourceProtocol: automatic
# Tf the source is an RTSP or RTSPS URL, this allows to 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
# If the source is a RTSPS, RTMPS or HTTPS 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 an RTSP or RTMP 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
# If the source is "redirect", this is the RTSP URL which clients will be
# redirected to.
sourceRedirect:
# If the source is "publisher" and a client is publishing, do not allow another
# client to disconnect the former and publish in its place.
disablePublisherOverride: no
# If the source is "publisher" and 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:
# If the source is "rpiCamera", these are the Raspberry Pi Camera parameters.
# 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'
# 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: []
# 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