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############################################### |
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# General parameters |
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# sets the 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: /tmp/rtsp-simple-server.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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# number of read buffers. |
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# a higher number allows a higher throughput, |
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# a lower number allows to save RAM. |
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readBufferCount: 512 |
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# enable the HTTP API. |
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api: yes |
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# address of the API listener. |
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apiAddress: 0.0.0.0:9997 |
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# enable Prometheus-compatible metrics. |
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metrics: no |
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# address of the metrics listener. |
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metricsAddress: 127.0.0.1:9998 |
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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: 127.0.0.1:9999 |
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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 server port is available in the RTSP_PORT variable. |
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runOnConnect: |
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# the restart parameter allows to restart the command if it exits suddenly. |
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runOnConnectRestart: no |
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############################################### |
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# RTSP parameters |
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# disable support for the RTSP protocol. |
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rtspDisable: no |
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# supported 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: [tcp, udp] |
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# encrypt handshake 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: :554 |
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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: :8555 |
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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. |
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authMethods: [basic, digest] |
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# read buffer size. |
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# this doesn't influence throughput and shouldn't be touched unless the server |
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# reports errors about the buffer size. |
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readBufferSize: 2048 |
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############################################### |
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# RTMP parameters |
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# disable support for the RTMP protocol. |
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rtmpDisable: no |
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# address of the RTMP listener. |
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rtmpAddress: :1935 |
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############################################### |
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# HLS parameters |
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# disable support for the HLS protocol. |
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hlsDisable: no |
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# address of the HLS listener. |
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hlsAddress: :8888 |
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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 an initial delay. |
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hlsAlwaysRemux: no |
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# number of HLS segments to generate. |
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# increasing segments allows more buffering, |
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# decreasing segments decreases latency. |
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hlsSegmentCount: 3 |
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# minimum duration of each segment. |
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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 segment duration to include at least a IDR frame in each one. |
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hlsSegmentDuration: 1s |
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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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############################################### |
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# Path parameters |
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# these settings are path-dependent. |
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# it's possible to use regular expressions by using a tilde as prefix. |
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# for example, "~^(test1|test2)$" will match both "test1" and "test2". |
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# for example, "~^prefix" will match all paths that start with "prefix". |
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# the settings under the path "all" are applied to all paths that do not match |
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# another entry. |
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paths: |
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all: |
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# source of the stream - this can be: |
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# * publisher -> the stream is published by a RTSP or RTMP client |
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# * rtsp://existing-url -> the stream is pulled from another RTSP server |
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# * rtsps://existing-url -> the stream is pulled from another RTSP server with RTSPS |
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# * rtmp://existing-url -> the stream is pulled from another RTMP server |
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# * http://existing-url/stream.m3u8 -> the stream is pulled from another HLS server |
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# * https://existing-url/stream.m3u8 -> the stream is pulled from another HLS server with HTTPS |
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# * redirect -> the stream is provided by another path or server |
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source: publisher |
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# if the source is an RTSP or RTSPS URL, this is the protocol that will be used to |
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# pull the stream. available values are "automatic", "udp", "multicast", "tcp". |
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# the TCP protocol can help to overcome the error "no UDP packets received recently". |
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sourceProtocol: automatic |
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# if the source is an RTSP or RTSPS URL, this allows to support sources that |
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# don't provide server ports or use random server ports. This is a security issue |
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# and must be used only when interacting with sources that require it. |
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sourceAnyPortEnable: no |
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# if the source is a RTSPS or HTTPS 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. |
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# the fingerprint 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 an RTSP or RTMP 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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# if the source is "redirect", this is the RTSP URL which clients will be |
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# redirected to. |
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sourceRedirect: |
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# if the source is "publisher" and a client is publishing, do not allow another |
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# client to disconnect the former and publish in its place. |
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disablePublisherOverride: no |
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# if the source is "publisher" and no one is publishing, redirect readers to this |
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# path. 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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# username required to publish. |
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# sha256-hashed values can be inserted with the "sha256:" prefix. |
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publishUser: |
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# password required to publish. |
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# sha256-hashed values can be inserted with the "sha256:" prefix. |
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publishPass: |
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# ips or networks (x.x.x.x/24) allowed to publish. |
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publishIPs: [] |
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# username required to read. |
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# sha256-hashed values can be inserted with the "sha256:" prefix. |
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readUser: |
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# password required to read. |
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# sha256-hashed values can be inserted with the "sha256:" prefix. |
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readPass: |
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# ips or networks (x.x.x.x/24) allowed to read. |
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readIPs: [] |
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# command to run when this path is initialized. |
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# this can be used to publish a stream and keep it always opened. |
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# this is terminated with SIGINT when the program closes. |
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# the path name is available in the RTSP_PATH variable. |
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# the server port is available in the RTSP_PORT variable. |
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runOnInit: |
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# the restart parameter allows to restart the command if it exits suddenly. |
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runOnInitRestart: no |
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# command to run when this path is requested. |
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# this can be used to publish a stream on demand. |
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# this is terminated with SIGINT when the path is not requested anymore. |
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# the path name is available in the RTSP_PATH variable. |
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# the server port is available in the RTSP_PORT variable. |
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runOnDemand: |
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# the restart parameter allows to restart the command if it exits suddenly. |
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runOnDemandRestart: no |
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# readers will be put on hold until the runOnDemand command starts publishing |
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# or until this amount of time has passed. |
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runOnDemandStartTimeout: 10s |
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# the runOnDemand command will be closed when there are no |
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# readers connected and this amount of time has passed. |
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runOnDemandCloseAfter: 10s |
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# command to run when a client starts publishing. |
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# this is terminated with SIGINT when a client stops publishing. |
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# the path name is available in the RTSP_PATH variable. |
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# the server port is available in the RTSP_PORT variable. |
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runOnPublish: |
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# the restart parameter allows to restart the command if it exits suddenly. |
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runOnPublishRestart: no |
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# command to run when a clients starts reading. |
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# this is terminated with SIGINT when a client stops reading. |
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# the path name is available in the RTSP_PATH variable. |
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# the server port is available in the RTSP_PORT variable. |
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runOnRead: |
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# the restart parameter allows to restart the command if it exits suddenly. |
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runOnReadRestart: no |
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