706 lines
28 KiB
YAML
706 lines
28 KiB
YAML
###############################################
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# Global settings
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# Settings in this section are applied anywhere.
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###############################################
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# Global settings -> General
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# Verbosity of the program; available values are "error", "warn", "info", "debug".
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logLevel: info
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# Destinations of log messages; available values are "stdout", "file" and "syslog".
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logDestinations: [stdout]
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# If "file" is in logDestinations, this is the file which will receive the logs.
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logFile: mediamtx.log
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# Timeout of read operations.
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readTimeout: 10s
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# Timeout of write operations.
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writeTimeout: 10s
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# Size of the queue of outgoing packets.
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# A higher value allows to increase throughput, a lower value allows to save RAM.
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writeQueueSize: 512
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# Maximum size of outgoing UDP packets.
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# This can be decreased to avoid fragmentation on networks with a low UDP MTU.
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udpMaxPayloadSize: 1472
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# Command to run when a client connects to the server.
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# This is terminated with SIGINT when a client disconnects from the server.
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# The following environment variables are available:
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# * RTSP_PORT: RTSP server port
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# * MTX_CONN_TYPE: connection type
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# * MTX_CONN_ID: connection ID
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runOnConnect:
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# Restart the command if it exits.
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runOnConnectRestart: no
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# Command to run when a client disconnects from the server.
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# Environment variables are the same of runOnConnect.
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runOnDisconnect:
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###############################################
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# Global settings -> Authentication
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# Authentication method. Available values are:
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# * internal: users are stored in the configuration file
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# * http: an external HTTP URL is contacted to perform authentication
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# * jwt: an external identity server provides authentication through JWTs
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authMethod: internal
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# Internal authentication.
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# list of users.
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authInternalUsers:
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# Default unprivileged user.
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# Username. 'any' means any user, including anonymous ones.
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- user: any
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# Password. Not used in case of 'any' user.
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pass:
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# IPs or networks allowed to use this user. An empty list means any IP.
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ips: []
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# List of permissions.
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permissions:
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# Available actions are: publish, read, playback, api, metrics, pprof.
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- action: publish
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# Paths can be set to further restrict access to a specific path.
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# An empty path means any path.
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# Regular expressions can be used by using a tilde as prefix.
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path:
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- action: read
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path:
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- action: playback
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path:
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# Default administrator.
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# This allows to use API, metrics and PPROF without authentication,
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# if the IP is localhost.
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- user: any
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pass:
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ips: ['127.0.0.1', '::1']
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permissions:
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- action: api
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- action: metrics
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- action: pprof
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# HTTP-based authentication.
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# URL called to perform authentication. Every time a user wants
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# to authenticate, the server calls this URL with the POST method
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# and a body containing:
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# {
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# "user": "user",
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# "password": "password",
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# "ip": "ip",
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# "action": "publish|read|playback|api|metrics|pprof",
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# "path": "path",
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# "protocol": "rtsp|rtmp|hls|webrtc|srt",
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# "id": "id",
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# "query": "query"
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# }
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# If the response code is 20x, authentication is accepted, otherwise
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# it is discarded.
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authHTTPAddress:
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# Actions to exclude from HTTP-based authentication.
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# Format is the same as the one of user permissions.
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authHTTPExclude:
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- action: api
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- action: metrics
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- action: pprof
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# JWT-based authentication.
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# Users have to login through an external identity server and obtain a JWT.
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# This JWT must contain the claim "mediamtx_permissions" with permissions,
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# for instance:
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# {
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# ...
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# "mediamtx_permissions": [
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# {
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# "action": "publish",
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# "path": "somepath"
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# }
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# ]
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# }
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# Users are expected to pass the JWT in the Authorization header or as a query parameter.
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# This is the JWKS URL that will be used to pull (once) the public key that allows
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# to validate JWTs.
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authJWTJWKS:
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# name of the claim that contains permissions.
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authJWTClaimKey: mediamtx_permissions
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###############################################
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# Global settings -> Control API
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# Enable controlling the server through the Control API.
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api: no
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# Address of the Control API listener.
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apiAddress: :9997
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# Enable TLS/HTTPS on the Control API server.
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apiEncryption: no
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# Path to the server key. This is needed only when encryption is yes.
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# This can be generated with:
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# openssl genrsa -out server.key 2048
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# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
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apiServerKey: server.key
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# Path to the server certificate.
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apiServerCert: server.crt
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# Value of the Access-Control-Allow-Origin header provided in every HTTP response.
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apiAllowOrigin: '*'
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# List of IPs or CIDRs of proxies placed before the HTTP server.
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# If the server receives a request from one of these entries, IP in logs
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# will be taken from the X-Forwarded-For header.
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apiTrustedProxies: []
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###############################################
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# Global settings -> Metrics
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# Enable Prometheus-compatible metrics.
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metrics: no
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# Address of the metrics HTTP listener.
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metricsAddress: :9998
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# Enable TLS/HTTPS on the Metrics server.
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metricsEncryption: no
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# Path to the server key. This is needed only when encryption is yes.
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# This can be generated with:
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# openssl genrsa -out server.key 2048
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# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
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metricsServerKey: server.key
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# Path to the server certificate.
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metricsServerCert: server.crt
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# Value of the Access-Control-Allow-Origin header provided in every HTTP response.
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metricsAllowOrigin: '*'
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# List of IPs or CIDRs of proxies placed before the HTTP server.
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# If the server receives a request from one of these entries, IP in logs
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# will be taken from the X-Forwarded-For header.
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metricsTrustedProxies: []
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###############################################
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# Global settings -> PPROF
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# Enable pprof-compatible endpoint to monitor performances.
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pprof: no
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# Address of the pprof listener.
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pprofAddress: :9999
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# Enable TLS/HTTPS on the pprof server.
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pprofEncryption: no
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# Path to the server key. This is needed only when encryption is yes.
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# This can be generated with:
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# openssl genrsa -out server.key 2048
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# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
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pprofServerKey: server.key
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# Path to the server certificate.
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pprofServerCert: server.crt
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# Value of the Access-Control-Allow-Origin header provided in every HTTP response.
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pprofAllowOrigin: '*'
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# List of IPs or CIDRs of proxies placed before the HTTP server.
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# If the server receives a request from one of these entries, IP in logs
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# will be taken from the X-Forwarded-For header.
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pprofTrustedProxies: []
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###############################################
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# Global settings -> Playback server
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# Enable downloading recordings from the playback server.
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playback: no
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# Address of the playback server listener.
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playbackAddress: :9996
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# Enable TLS/HTTPS on the playback server.
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playbackEncryption: no
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# Path to the server key. This is needed only when encryption is yes.
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# This can be generated with:
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# openssl genrsa -out server.key 2048
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# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
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playbackServerKey: server.key
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# Path to the server certificate.
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playbackServerCert: server.crt
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# Value of the Access-Control-Allow-Origin header provided in every HTTP response.
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playbackAllowOrigin: '*'
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# List of IPs or CIDRs of proxies placed before the HTTP server.
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# If the server receives a request from one of these entries, IP in logs
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# will be taken from the X-Forwarded-For header.
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playbackTrustedProxies: []
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###############################################
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# Global settings -> RTSP server
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# Enable publishing and reading streams with the RTSP protocol.
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rtsp: yes
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# List of enabled RTSP transport protocols.
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# UDP is the most performant, but doesn't work when there's a NAT/firewall between
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# server and clients, and doesn't support encryption.
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# UDP-multicast allows to save bandwidth when clients are all in the same LAN.
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# TCP is the most versatile, and does support encryption.
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# The handshake is always performed with TCP.
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protocols: [udp, multicast, tcp]
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# Encrypt handshakes and TCP streams with TLS (RTSPS).
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# Available values are "no", "strict", "optional".
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encryption: "no"
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# Address of the TCP/RTSP listener. This is needed only when encryption is "no" or "optional".
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rtspAddress: :8554
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# Address of the TCP/TLS/RTSPS listener. This is needed only when encryption is "strict" or "optional".
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rtspsAddress: :8322
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# Address of the UDP/RTP listener. This is needed only when "udp" is in protocols.
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rtpAddress: :8000
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# Address of the UDP/RTCP listener. This is needed only when "udp" is in protocols.
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rtcpAddress: :8001
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# IP range of all UDP-multicast listeners. This is needed only when "multicast" is in protocols.
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multicastIPRange: 224.1.0.0/16
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# Port of all UDP-multicast/RTP listeners. This is needed only when "multicast" is in protocols.
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multicastRTPPort: 8002
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# Port of all UDP-multicast/RTCP listeners. This is needed only when "multicast" is in protocols.
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multicastRTCPPort: 8003
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# Path to the server key. This is needed only when encryption is "strict" or "optional".
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# This can be generated with:
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# openssl genrsa -out server.key 2048
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# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
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serverKey: server.key
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# Path to the server certificate. This is needed only when encryption is "strict" or "optional".
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serverCert: server.crt
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# Authentication methods. Available are "basic" and "digest".
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# "digest" doesn't provide any additional security and is available for compatibility only.
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rtspAuthMethods: [basic]
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###############################################
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# Global settings -> RTMP server
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# Enable publishing and reading streams with the RTMP protocol.
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rtmp: yes
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# Address of the RTMP listener. This is needed only when encryption is "no" or "optional".
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rtmpAddress: :1935
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# Encrypt connections with TLS (RTMPS).
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# Available values are "no", "strict", "optional".
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rtmpEncryption: "no"
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# Address of the RTMPS listener. This is needed only when encryption is "strict" or "optional".
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rtmpsAddress: :1936
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# Path to the server key. This is needed only when encryption is "strict" or "optional".
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# This can be generated with:
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# openssl genrsa -out server.key 2048
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# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
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rtmpServerKey: server.key
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# Path to the server certificate. This is needed only when encryption is "strict" or "optional".
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rtmpServerCert: server.crt
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###############################################
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# Global settings -> HLS server
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# Enable reading streams with the HLS protocol.
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hls: yes
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# Address of the HLS listener.
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hlsAddress: :8888
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# Enable TLS/HTTPS on the HLS server.
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# This is required for Low-Latency HLS.
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hlsEncryption: no
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# Path to the server key. This is needed only when encryption is yes.
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# This can be generated with:
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# openssl genrsa -out server.key 2048
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# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
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hlsServerKey: server.key
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# Path to the server certificate.
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hlsServerCert: server.crt
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# Value of the Access-Control-Allow-Origin header provided in every HTTP response.
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# This allows to play the HLS stream from an external website.
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hlsAllowOrigin: '*'
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# List of IPs or CIDRs of proxies placed before the HLS server.
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# If the server receives a request from one of these entries, IP in logs
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# will be taken from the X-Forwarded-For header.
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hlsTrustedProxies: []
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# By default, HLS is generated only when requested by a user.
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# This option allows to generate it always, avoiding the delay between request and generation.
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hlsAlwaysRemux: no
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# Variant of the HLS protocol to use. Available options are:
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# * mpegts - uses MPEG-TS segments, for maximum compatibility.
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# * fmp4 - uses fragmented MP4 segments, more efficient.
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# * lowLatency - uses Low-Latency HLS.
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hlsVariant: lowLatency
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# Number of HLS segments to keep on the server.
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# Segments allow to seek through the stream.
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# Their number doesn't influence latency.
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hlsSegmentCount: 7
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# Minimum duration of each segment.
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# A player usually puts 3 segments in a buffer before reproducing the stream.
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# The final segment duration is also influenced by the interval between IDR frames,
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# since the server changes the duration in order to include at least one IDR frame
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# in each segment.
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hlsSegmentDuration: 1s
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# Minimum duration of each part.
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# A player usually puts 3 parts in a buffer before reproducing the stream.
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# Parts are used in Low-Latency HLS in place of segments.
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# Part duration is influenced by the distance between video/audio samples
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# and is adjusted in order to produce segments with a similar duration.
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hlsPartDuration: 200ms
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# Maximum size of each segment.
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# This prevents RAM exhaustion.
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hlsSegmentMaxSize: 50M
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# Directory in which to save segments, instead of keeping them in the RAM.
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# This decreases performance, since reading from disk is less performant than
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# reading from RAM, but allows to save RAM.
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hlsDirectory: ''
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# The muxer will be closed when there are no
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# reader requests and this amount of time has passed.
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hlsMuxerCloseAfter: 60s
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###############################################
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# Global settings -> WebRTC server
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# Enable publishing and reading streams with the WebRTC protocol.
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webrtc: yes
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# Address of the WebRTC HTTP listener.
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webrtcAddress: :8889
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# Enable TLS/HTTPS on the WebRTC server.
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webrtcEncryption: no
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# Path to the server key.
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# This can be generated with:
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# openssl genrsa -out server.key 2048
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# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
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webrtcServerKey: server.key
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# Path to the server certificate.
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webrtcServerCert: server.crt
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# Value of the Access-Control-Allow-Origin header provided in every HTTP response.
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# This allows to play the WebRTC stream from an external website.
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webrtcAllowOrigin: '*'
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# List of IPs or CIDRs of proxies placed before the WebRTC server.
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# If the server receives a request from one of these entries, IP in logs
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# will be taken from the X-Forwarded-For header.
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webrtcTrustedProxies: []
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# Address of a local UDP listener that will receive connections.
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# Use a blank string to disable.
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webrtcLocalUDPAddress: :8189
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# Address of a local TCP listener that will receive connections.
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# This is disabled by default since TCP is less efficient than UDP and
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# introduces a progressive delay when network is congested.
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webrtcLocalTCPAddress: ''
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# WebRTC clients need to know the IP of the server.
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# Gather IPs from interfaces and send them to clients.
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webrtcIPsFromInterfaces: yes
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# List of interfaces whose IPs will be sent to clients.
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# An empty value means to use all available interfaces.
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webrtcIPsFromInterfacesList: []
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# List of additional hosts or IPs to send to clients.
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webrtcAdditionalHosts: []
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# ICE servers. Needed only when local listeners can't be reached by clients.
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# STUN servers allows to obtain and share the public IP of the server.
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# TURN/TURNS servers forces all traffic through them.
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webrtcICEServers2: []
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# - url: stun:stun.l.google.com:19302
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# if user is "AUTH_SECRET", then authentication is secret based.
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# the secret must be inserted into the password field.
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# username: ''
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# password: ''
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# clientOnly: false
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# Time to wait for the WebRTC handshake to complete.
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webrtcHandshakeTimeout: 10s
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# Maximum time to gather video tracks.
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webrtcTrackGatherTimeout: 2s
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###############################################
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# Global settings -> SRT server
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# Enable publishing and reading streams with the SRT protocol.
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srt: yes
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# Address of the SRT listener.
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srtAddress: :8890
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###############################################
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# Default path settings
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# Settings in "pathDefaults" are applied anywhere,
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# unless they are overridden in "paths".
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pathDefaults:
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###############################################
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# Default path settings -> General
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# Source of the stream. This can be:
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# * publisher -> the stream is provided by a RTSP, RTMP, WebRTC or SRT client
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# * rtsp://existing-url -> the stream is pulled from another RTSP server / camera
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# * rtsps://existing-url -> the stream is pulled from another RTSP server / camera with RTSPS
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# * rtmp://existing-url -> the stream is pulled from another RTMP server / camera
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# * rtmps://existing-url -> the stream is pulled from another RTMP server / camera with RTMPS
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# * http://existing-url/stream.m3u8 -> the stream is pulled from another HLS server / camera
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# * https://existing-url/stream.m3u8 -> the stream is pulled from another HLS server / camera with HTTPS
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# * udp://ip:port -> the stream is pulled with UDP, by listening on the specified IP and port
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# * srt://existing-url -> the stream is pulled from another SRT server / camera
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# * whep://existing-url -> the stream is pulled from another WebRTC server / camera
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# * wheps://existing-url -> the stream is pulled from another WebRTC server / camera with HTTPS
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# * redirect -> the stream is provided by another path or server
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# * rpiCamera -> the stream is provided by a Raspberry Pi Camera
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# The following variables can be used in the source string:
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# * $MTX_QUERY: query parameters (passed by first reader)
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# * $G1, $G2, ...: regular expression groups, if path name is
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# a regular expression.
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source: publisher
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# If the source is a URL, and the source certificate is self-signed
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# or invalid, you can provide the fingerprint of the certificate in order to
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# validate it anyway. It can be obtained by running:
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# openssl s_client -connect source_ip:source_port </dev/null 2>/dev/null | sed -n '/BEGIN/,/END/p' > server.crt
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# openssl x509 -in server.crt -noout -fingerprint -sha256 | cut -d "=" -f2 | tr -d ':'
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sourceFingerprint:
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# If the source is a URL, it will be pulled only when at least
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# one reader is connected, saving bandwidth.
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sourceOnDemand: no
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# If sourceOnDemand is "yes", readers will be put on hold until the source is
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# ready or until this amount of time has passed.
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sourceOnDemandStartTimeout: 10s
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# If sourceOnDemand is "yes", the source will be closed when there are no
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# readers connected and this amount of time has passed.
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sourceOnDemandCloseAfter: 10s
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# Maximum number of readers. Zero means no limit.
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maxReaders: 0
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# SRT encryption passphrase require to read from this path
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srtReadPassphrase:
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# If the stream is not available, redirect readers to this path.
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# It can be can be a relative path (i.e. /otherstream) or an absolute RTSP URL.
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fallback:
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###############################################
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# Default path settings -> Record
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# Record streams to disk.
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record: no
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# Path of recording segments.
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# Extension is added automatically.
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|
# 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: 1s
|
|
# Minimum duration of each segment.
|
|
recordSegmentDuration: 1h
|
|
# Delete segments after this timespan.
|
|
# Set to 0s to disable automatic deletion.
|
|
recordDeleteAfter: 24h
|
|
|
|
###############################################
|
|
# Default path settings -> Publisher source (when source is "publisher")
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|
|
|
# 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)
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|
|
|
# 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
|
|
# Auto-white-balance fixed gains. This can be used in place of rpiCameraAWB.
|
|
# format: [red,blue]
|
|
rpiCameraAWBGains: [0, 0]
|
|
# 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
|
|
# 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:
|
|
# 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. Available values: auto, hardwareH264, softwareH264
|
|
rpiCameraCodec: auto
|
|
# Period between IDR frames
|
|
rpiCameraIDRPeriod: 60
|
|
# Bitrate
|
|
rpiCameraBitrate: 5000000
|
|
# H264 profile
|
|
rpiCameraProfile: main
|
|
# H264 level
|
|
rpiCameraLevel: '4.1'
|
|
|
|
###############################################
|
|
# 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
|
|
# * MTX_SEGMENT_DURATION: segment duration
|
|
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:
|