############################################### # 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 # 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 # 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: yes # 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: RTSP server port # * MTX_CONN_TYPE: connection type # * MTX_CONN_ID: connection ID 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 -> RTSP server # Allow 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, 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] ############################################### # Global settings -> RTMP server # Allow publishing and reading streams with the RTMP protocol. rtmp: 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 ############################################### # Global settings -> HLS server # Allow reading streams with the HLS protocol. hls: 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: '' ############################################### # Global settings -> WebRTC server # Allow publishing and reading streams with the WebRTC protocol. webrtc: no # 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: '' ############################################### # Global settings -> SRT server # Allow publishing and reading streams with the SRT protocol. srt: no # 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://ip:port -> the stream is pulled with UDP, by listening on the specified IP and port # * 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 # If path name is a regular expression, $G1, G2, etc will be replaced # with regular expression groups. 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 require 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: ############################################### # Default path settings -> Recording # Record streams to disk. record: no # Path of recording segments. # Extension is added automatically. # Available variables are %path (path name), %Y %m %d %H %M %S %f %s (time in strftime format) 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: 100ms # Minimum duration of each segment. recordSegmentDuration: 12h # Delete segments after this timespan. # Set to 0s to disable automatic deletion. recordDeleteAfter: 24h ############################################### # Default path settings -> Authentication # 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: [] ############################################### # 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: ############################################### # Default path settings -> Redirect source (when source is "redirect") # RTSP URL which clients will be redirected to. sourceRedirect: ############################################### # Default path settings -> Raspberry Pi Camera source (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: # 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 # 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: continuous # 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' ############################################### # 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) # * RTSP_PORT: RTSP server port # * G1, G2, ...: regular expression groups, if path name is # a regular expression. # * MTX_SOURCE_TYPE: source type # * MTX_SOURCE_ID: source ID 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) # * RTSP_PORT: RTSP server port # * G1, G2, ...: regular expression groups, if path name is # a regular expression. # * MTX_READER_TYPE: reader type # * MTX_READER_ID: reader ID 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 # * RTSP_PORT: RTSP server port # * G1, G2, ...: regular expression groups, if path name is # a regular expression. # * MTX_SEGMENT_PATH: segment file path runOnRecordSegmentCreate: # Command to run when a recording segment is complete. # 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. # * MTX_SEGMENT_PATH: segment file path 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: live: record: yes recordDeleteAfter: 0s recordPath: ./recordings/%path/%Y-%m-%d_%H-%M-%S-%f runOnRecordSegmentComplete: mv $MTX_SEGMENT_PATH ./recordings/vod # Settings under path "all_others" are applied to all paths that # do not match another entry. all_others: