mediamtx/rtsp-simple-server.yml

###############################################
# General parameters

# sets the verbosity of the program; available values are "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 number allows a higher throughput,
# a lower number allows to save RAM.
readBufferCount: 512

# 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 server port is available in the RTSP_PORT variable.
runOnConnect:
# the restart parameter allows to restart the command if it exits suddenly.
runOnConnectRestart: no

###############################################
# RTSP parameters

# disable support for the RTSP protocol.
rtspDisable: no
# supported RTSP stream 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 handshake 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: :8555
# 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]
# read buffer size.
# this doesn't influence throughput and shouldn't be touched unless the server
# reports errors about the buffer size.
readBufferSize: 2048

###############################################
# RTMP parameters

# disable support for the RTMP protocol.
rtmpDisable: no
# address of the RTMP listener.
rtmpAddress: :1935

###############################################
# HLS parameters

# disable support for the HLS protocol.
hlsDisable: no
# address of the HLS listener.
hlsAddress: :8888
# by default, HLS is generated only when requested by a user;
# this option allows to generate it always, avoiding an initial delay.
hlsAlwaysRemux: no
# number of HLS segments to generate.
# increasing segments allows more buffering,
# decreasing segments decreases latency.
hlsSegmentCount: 3
# minimum duration of each segment.
# the final segment duration is also influenced by the interval between IDR frames,
# since the server changes the segment duration to include at least a IDR frame in each one.
hlsSegmentDuration: 1s
# 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: '*'

###############################################
# Path parameters

# these settings are path-dependent.
# 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
    # * rtsps://existing-url -> the stream is pulled from another RTSP server, with RTSPS
    # * rtmp://existing-url -> the stream is pulled from another RTMP server
    # * http://existing-url/stream.m3u8 -> the stream is pulled from another HLS server
    # * redirect -> the stream is provided by another path or server
    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".
    # the TCP protocol can help to overcome the error "no UDP packets received recently".
    sourceProtocol: automatic

    # if the source is an RTSP or RTSPS URL, this allows to support cameras that
    # don't provide server ports. This is a security issue and must be enabled only
    # when interacting with old cameras that require it.
    sourceAnyPortEnable: no

    # if the source is an RTSPS URL, the fingerprint of the certificate of the source
    # must be provided in order to prevent man-in-the-middle attacks.
    # it can be obtained from the source 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:

    # 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 path name is available in the RTSP_PATH variable.
    # the server port is available in the RTSP_PORT variable.
    runOnInit:
    # the restart parameter allows to 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 path name is available in the RTSP_PATH variable.
    # the server port is available in the RTSP_PORT variable.
    runOnDemand:
    # the restart parameter allows to 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 runOnDemand command will be closed when there are no
    # readers connected and this amount of time has passed.
    runOnDemandCloseAfter: 10s

    # command to run when a client starts publishing.
    # this is terminated with SIGINT when a client stops publishing.
    # the path name is available in the RTSP_PATH variable.
    # the server port is available in the RTSP_PORT variable.
    runOnPublish:
    # the restart parameter allows to restart the command if it exits suddenly.
    runOnPublishRestart: no

    # command to run when a clients starts reading.
    # this is terminated with SIGINT when a client stops reading.
    # the path name is available in the RTSP_PATH variable.
    # the server port is available in the RTSP_PORT variable.
    runOnRead:
    # the restart parameter allows to restart the command if it exits suddenly.
    runOnReadRestart: no