Mediamtx binary compiled from source fails to launch after #5476

[Chase-Stewart-IS4S]

Which version are you using?

2b302e7

Which operating system are you using?

Linux arm64 standard

Describe the issue

Let me know if I can provide any further information as well to help this along!

What I expect to happen

• Pull down mediamtx main repo
• Compile from source using a script
• Use default mediamtx.yml file from a script
• mediamtx binary starts up and runs

What I am observing to happen instead

Mediamtx binary fails to start with the following:

Mar 17 22:50:37 ark-jetson-devicelab-gps mediamtx[41737]: ERR: json: unknown field "pathDefaults.rtspDemuxMpegts"

More Details

I have been using a script based on the instructions for compiling a mediamtx binary from source https://mediamtx.org/docs/misc/compile for a few weeks now, and it has been stable for my needs. Recently when re-installing, I found that mediamtx was failing to launch with the only hint being this line below:

Mar 17 22:50:37 ark-jetson-devicelab-gps mediamtx[41737]: ERR: json: unknown field "pathDefaults.rtspDemuxMpegts"

I can see that the configuration option in question was just recently added in #5476 - Without being over-prescriptive of the problem vs my observation, it strongly seems like some kind of binding or connection has not been made when compiling from source.

Describe how to replicate the issue

Step 1: build mediamtx binary from source:

# Substitute SRC_DIR and DEST_BIN as needed
rm -rf "$SRC_DIR"
wget -P "$SRC_DIR" https://github.com/bluenviron/mediamtx/archive/refs/heads/main.zip
pushd . || exit
cd "$SRC_DIR" || exit
unzip -q main.zip
cd mediamtx-main || exit
/snap/bin/go generate ./...
CGO_ENABLED=0 /snap/bin/go build -o mediamtx .

# move necessary files into XD
cp -f mediamtx mediamtx.yml "$DEST_BIN/"
chmod +x "$DEST_BIN/mediamtx"
popd || exit

echo "mediamtx installed to $DEST_BIN/mediamtx"

Step 3: run mediamtx binary

jetson@ark-jetson-devicelab-gps:~/.local/bin$ ./mediamtx mediamtx.yml 
ERR: json: unknown field "pathDefaults.rtspDemuxMpegts"

Workaround

Commenting out rtspDemuxMpegts: false fixes the issue

Here a workaround that I am using right now during my installation:

sed -i 's/rtspDemuxMpegts/# rtspDemuxMpegts/g' mediamtx.yml

MediaMTX configuration

###############################################
# Global settings

# Settings in this section are applied anywhere.

###############################################
# Global settings -> General

# Verbosity of the program; available values are "error", "warn", "info", "debug".
logLevel: warn
# Destinations of log messages; available values are "stdout", "file" and "syslog".
logDestinations: [stdout]
# When destination is "stdout" or "file", emit logs in structured format (JSONL).
logStructured: false
# When "file" is in logDestinations, this is the file which will receive logs.
logFile: mediamtx.log
# When "syslog" is in logDestinations, use prefix for logs.
sysLogPrefix: mediamtx
# Dump packets to disk. This is useful for debugging.
dumpPackets: false

# 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 payloads.
# It defaults to the maximum packet size on ethernet (1500) minus IPv6 and UDP headers (48).
# This can be decreased to avoid fragmentation on networks with a low MTU.
udpMaxPayloadSize: 1452
# Size of the read buffer of every UDP socket.
# This can be increased to decrease packet losses.
# It defaults to the default value of the operating system.
udpReadBufferSize: 0

# 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: false
# 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: credentials 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.
# Enabled 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: []
  # 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:
# If the HTTP authentication 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 auth_http_domain:443 </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 ':'
authHTTPFingerprint:
# 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 or as password.
# 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 2>/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 and will be disabled in the future.
# RTSP and RTMP always allow JWT in query even if disabled, since there is no alternative.
authJWTInHTTPQuery: true
# Expected issuer (iss) claim in the JWT. Leave empty to skip validation.
authJWTIssuer:
# Expected audience (aud) claim in the JWT. Leave empty to skip validation.
authJWTAudience:

###############################################
# Global settings -> Control API

# Enable controlling the server through the Control API.
api: false
# Address of the Control API listener.
apiAddress: :9997
# Enable HTTPS on the Control API server.
apiEncryption: false
# 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
# Allowed CORS origins.
# Supports wildcards: ['http://*.example.com']
apiAllowOrigins: ['*']
# IPs or CIDRs of proxies placed before the HTTP server.
# These proxies can use the X-Forwarded-For header to set the real IP of clients,
# and the X-Forwarded-Proto header to set the original protocol.
apiTrustedProxies: []

###############################################
# Global settings -> Metrics

# Enable Prometheus-compatible metrics.
metrics: false
# Address of the metrics HTTP listener.
metricsAddress: :9998
# Enable HTTPS on the Metrics server.
metricsEncryption: false
# 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
# Allowed CORS origins.
# Supports wildcards: ['http://*.example.com']
metricsAllowOrigins: ['*']
# IPs or CIDRs of proxies placed before the HTTP server.
# These proxies can use the X-Forwarded-For header to set the real IP of clients,
# and the X-Forwarded-Proto header to set the original protocol.
metricsTrustedProxies: []

###############################################
# Global settings -> PPROF

# Enable pprof-compatible endpoint to monitor performances.
pprof: false
# Address of the pprof listener.
pprofAddress: :9999
# Enable HTTPS on the pprof server.
pprofEncryption: false
# 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
# Allowed CORS origins.
# Supports wildcards: ['http://*.example.com']
pprofAllowOrigins: ['*']
# IPs or CIDRs of proxies placed before the HTTP server.
# These proxies can use the X-Forwarded-For header to set the real IP of clients,
# and the X-Forwarded-Proto header to set the original protocol.
pprofTrustedProxies: []

###############################################
# Global settings -> Playback server

# Enable downloading recordings from the playback server.
playback: false
# Address of the playback server listener.
playbackAddress: :9996
# Enable HTTPS on the playback server.
playbackEncryption: false
# 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
# Allowed CORS origins.
# Supports wildcards: ['http://*.example.com']
playbackAllowOrigins: ['*']
# IPs or CIDRs of proxies placed before the HTTP server.
# These proxies can use the X-Forwarded-For header to set the real IP of clients,
# and the X-Forwarded-Proto header to set the original protocol.
playbackTrustedProxies: []

###############################################
# Global settings -> RTSP server

# Enable publishing and reading streams with the RTSP protocol.
rtsp: true
# Enabled RTSP transport protocols. The handshake is always performed with TCP.
rtspTransports: [udp, multicast, tcp]
# Use secure protocol variants (RTSPS, SRTP, SRTCP).
# 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/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 and encryption is "no" or "optional".
rtpAddress: :8000
# Address of the UDP/RTCP listener. This is needed only when "udp" is in rtspTransports and encryption is "no" or "optional".
rtcpAddress: :8001
# IP range of all UDP-multicast listeners. This is needed only when "multicast" is in rtspTransports and encryption is "no" or "optional".
multicastIPRange: 224.1.0.0/16
# Port of all UDP-multicast/RTP listeners. This is needed only when "multicast" is in rtspTransports and encryption is "no" or "optional".
multicastRTPPort: 8002
# Port of all UDP-multicast/RTCP listeners. This is needed only when "multicast" is in rtspTransports and encryption is "no" or "optional".
multicastRTCPPort: 8003
# Address of the UDP/SRTP listener. This is needed only when "udp" is in rtspTransports and encryption is "strict" or "optional".
srtpAddress: :8004
# Address of the UDP/SRTCP listener. This is needed only when "udp" is in rtspTransports and encryption is "strict" or "optional".
srtcpAddress: :8005
# Port of all UDP-multicast/SRTP listeners. This is needed only when "multicast" is in rtspTransports and encryption is "strict" or "optional".
multicastSRTPPort: 8006
# Port of all UDP-multicast/SRTCP listeners. This is needed only when "multicast" is in rtspTransports and encryption is "strict" or "optional".
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]

###############################################
# Global settings -> RTMP server

# Enable publishing and reading streams with the RTMP protocol.
rtmp: true
# Use the secure protocol variant (RTMP).
# Available values are "no", "strict", "optional".
rtmpEncryption: "no"
# Address of the RTMP listener. This is needed only when encryption is "no" or "optional".
rtmpAddress: :1935
# 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: true
# Address of the HLS listener.
hlsAddress: :8888
# Enable HTTPS on the HLS server.
# This is required for Low-Latency HLS to function correctly on Apple devices.
hlsEncryption: false
# 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
# Allowed CORS origins.
# Supports wildcards: ['http://*.example.com']
hlsAllowOrigins: ['*']
# 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: false
# 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: true
# Address of the WebRTC HTTP listener.
webrtcAddress: :8889
# Enable HTTPS on the WebRTC server.
# This covers only the WebRTC handshake and does not influence the encryption of WebRTC streams
# which are always encrypted, with a key that is exchanged during the WebRTC handshake.
webrtcEncryption: false
# 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
# Allowed CORS origins.
# Supports wildcards: ['http://*.example.com']
webrtcAllowOrigins: ['*']
# 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: true
# Interfaces whose IPs will be sent to clients.
# An empty value means to use all available interfaces.
webrtcIPsFromInterfacesList: []
# Additional hosts or IPs to send to clients.
webrtcAdditionalHosts: []
# ICE servers. Needed only when local listeners can't be reached by clients.
# STUN servers allow to obtain and share the public IP of the server.
# TURN/TURNS servers force 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
# Maximum time to gather STUN candidates.
webrtcSTUNGatherTimeout: 5s
# Time to wait for the WebRTC handshake to complete.
webrtcHandshakeTimeout: 10s
# Maximum time to gather tracks.
webrtcTrackGatherTimeout: 2s

###############################################
# Global settings -> SRT server

# Enable publishing and reading streams with the SRT protocol.
srt: true
# 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
  # * rtsp+http://existing-url -> the stream is pulled from another RTSP server / camera, with HTTP tunneling
  # * rtsps+http://existing-url -> the stream is pulled from another RTSP server / camera, with HTTPS tunneling
  # * rtsp+ws://existing-url -> the stream is pulled from another RTSP server / camera, with WebSocket tunneling
  # * rtsps+ws://existing-url -> the stream is pulled from another RTSP server / camera, with secure WebSocket tunneling
  # * 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
  # * srt://existing-url -> the stream is pulled from another SRT server / camera
  # * whep://existing-url -> the stream is pulled from another WebRTC server / camera with HTTP+WHEP
  # * wheps://existing-url -> the stream is pulled from another WebRTC server / camera with HTTPS+WHEP
  # * 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 TLS 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: false
  # 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:
  # Use absolute timestamp of frames, instead of replacing them with the current time.
  useAbsoluteTimestamp: false

  ###############################################
  # Default path settings -> Always available

  # Enable always-available mode, in which a offline segment is played on repeat when the stream is not available.
  alwaysAvailable: false
  # Tracks of the default offline segment.
  alwaysAvailableTracks: []
    # Available values are: AV1, VP9, H265, H264, Opus, MPEG4Audio, G711, LPCM
    # - codec: H264
    #   # in case of MPEG4Audio, G711, LPCM, sampleRate and ChannelCount must be provided too.
    #   sampleRate: 48000
    #   channelCount: 2
    #   # in case of G711, muLaw must be provided too.
    #   muLaw: false
  # A MP4 file can be used instead of the default offline segment.
  alwaysAvailableFile: ''

  ###############################################
  # Default path settings -> Record

  # Record streams to disk.
  record: false
  # 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: true
  # SRT encryption passphrase required to publish to this path.
  srtPublishPassphrase:
  # Demux MPEG-TS over RTSP into elementary streams.
  # When enabled, RTSP publishers sending MP2T/90000 will be demultiplexed
  # and their elementary streams (H.264, H.265, AAC, etc.) exposed as native tracks.
  # This allows HLS, WebRTC, and other outputs to work transparently with MPEG-TS sources.
  rtspDemuxMpegts: false

  ###############################################
  # 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: false
  # 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:
  # Range of ports used as source port in outgoing UDP packets.
  rtspUDPSourcePortRange: [10000, 65535]

  ###############################################
  # Default path settings -> RTP source (when source is RTP)

  # session description protocol (SDP) of the RTP stream.
  rtpSDP:

  ###############################################
  # Default path settings -> WebRTC / WHEP source (when source is WHEP)

  # Token to insert in the Authorization: Bearer header.
  whepBearerToken: ''
  # Maximum time to gather STUN candidates.
  whepSTUNGatherTimeout: 5s
  # Time to wait for the WebRTC handshake to complete.
  whepHandshakeTimeout: 10s
  # Maximum time to gather tracks.
  whepTrackGatherTimeout: 2s

  ###############################################
  # 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: false

  # 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: false
  # 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: false
  # 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: false
  # 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:

MediaMTX logs

Mar 17 22:50:37 ark-jetson-devicelab-gps systemd[1765]: mediamtx.service: Scheduled restart job, restart counter is at 151.
Mar 17 22:50:37 ark-jetson-devicelab-gps systemd[1765]: Stopped (video-pipeline) MediaMTX media proxy user service.
Mar 17 22:50:37 ark-jetson-devicelab-gps systemd[1765]: Started (video-pipeline) MediaMTX media proxy user service.
Mar 17 22:50:37 ark-jetson-devicelab-gps mediamtx[41737]: ERR: json: unknown field "pathDefaults.rtspDemuxMpegts"
Mar 17 22:50:37 ark-jetson-devicelab-gps systemd[1765]: mediamtx.service: Main process exited, code=exited, status=1/FAILURE
Mar 17 22:50:37 ark-jetson-devicelab-gps systemd[1765]: mediamtx.service: Failed with result 'exit-code'.
Mar 17 22:50:42 ark-jetson-devicelab-gps systemd[1765]: mediamtx.service: Scheduled restart job, restart counter is at 152.
Mar 17 22:50:42 ark-jetson-devicelab-gps systemd[1765]: Stopped (video-pipeline) MediaMTX media proxy user service.
Mar 17 22:50:42 ark-jetson-devicelab-gps systemd[1765]: Started (video-pipeline) MediaMTX media proxy user service.
Mar 17 22:50:42 ark-jetson-devicelab-gps mediamtx[42105]: ERR: json: unknown field "pathDefaults.rtspDemuxMpegts"
Mar 17 22:50:42 ark-jetson-devicelab-gps systemd[1765]: mediamtx.service: Main process exited, code=exited, status=1/FAILURE
Mar 17 22:50:42 ark-jetson-devicelab-gps systemd[1765]: mediamtx.service: Failed with result 'exit-code'.

Packet dump

N/A as it is failing to run.

[Chase-Stewart-IS4S]

Looking at other changes that modified these configurations as a total outsider, I wonder if #5476 needed changes in internal/api?

Let me know if I can help my cause by making a PR with said changes- However I wonder if this would be trivial for a maintainer.

[kondas]

Hi, I have similar issue with another parameter since 1.6.3
ERR: json: unknown field "sourceondemand"