412 lines
12 KiB
XML
412 lines
12 KiB
XML
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<?xml version="1.0" encoding="UTF-8" ?>
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<!--
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This is a stripped down config meant for streamers looking to self-host, rebroadcast
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and maybe offer transcoding for their content. OvenMediaEngine supports extra features
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not covered here like VOD storage, scheduled contents and multiplexing, check the website
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https://airensoft.gitbook.io/ovenmediaengine for a more comprehesive feature list.
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-->
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<Server version="8">
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<!-- Not sure this is used anywhere, just call it whatever -->
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<Name>Your stream "service" name</Name>
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<!-- Single node setup -->
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<Type>origin</Type>
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<!-- Specify IP address to bind ("*" means all IPv4 IPs, "::" means all IPv6 IPs) -->
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<!-- Uncomment the line below to enable IPv6 -->
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<!-- <IP>::</IP> -->
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<IP>*</IP>
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<!-- Anonymize IPs in logs for compliance with privacy laws -->
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<PrivacyProtection>true</PrivacyProtection>
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<!--
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To get the public IP address(mapped address of stun) of the local server.
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This is useful when OME cannot obtain a public IP from an interface, such as AWS or docker environment.
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If this is successful, you can use ${PublicIP} in your settings.
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-->
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<StunServer>stun.ovenmediaengine.com:13478</StunServer>
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<Modules>
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<!--
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Currently OME only supports h2 like all browsers do. Therefore, HTTP/2 only works on TLS ports.
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-->
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<HTTP2>
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<Enable>true</Enable>
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</HTTP2>
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<!--
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Low Latency HLS is a Apple-specific thing, don't expect it to work elsewhere!
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Other clients will treat is as HLS with its normal latency of a few seconds.
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-->
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<LLHLS>
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<Enable>true</Enable>
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</LLHLS>
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<!-- P2P works only in WebRTC and is experiment feature -->
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<P2P>
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<!-- disabled by default -->
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<Enable>false</Enable>
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<MaxClientPeersPerHostPeer>2</MaxClientPeersPerHostPeer>
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</P2P>
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</Modules>
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<!-- Fuck with this to do your performance tuning and port shuffling -->
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<Bind>
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<Managers>
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<API>
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<Port>8081</Port>
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<WorkerCount>1</WorkerCount>
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</API>
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</Managers>
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<Providers>
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<RTSPC>
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<WorkerCount>1</WorkerCount>
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</RTSPC>
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<OVT>
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<WorkerCount>1</WorkerCount>
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</OVT>
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<RTMP>
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<Port>1935</Port>
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<WorkerCount>1</WorkerCount>
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</RTMP>
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<SRT>
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<Port>9999</Port>
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<WorkerCount>1</WorkerCount>
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</SRT>
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<WebRTC>
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<Signalling>
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<Port>3333</Port>
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<TLSPort>3334</TLSPort>
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<WorkerCount>1</WorkerCount>
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</Signalling>
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<IceCandidates>
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<!-- Uncomment the line below to use IPv6 ICE Candidate -->
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<!-- <IceCandidate>[::]:10000-10004/udp</IceCandidate> -->
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<IceCandidate>*:10000/udp</IceCandidate>
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<TcpRelay>*:3478</TcpRelay>
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<TcpForce>false</TcpForce>
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<TcpRelayWorkerCount>1</TcpRelayWorkerCount>
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</IceCandidates>
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</WebRTC>
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</Providers>
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<Publishers>
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<!-- The OVT is protocol for ORIGIN-EDGE -->
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<OVT>
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<Port>9000</Port>
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<WorkerCount>1</WorkerCount>
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</OVT>
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<LLHLS>
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<!--
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OME only supports h2, so LLHLS works over HTTP/1.1 on non-TLS ports.
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LLHLS works with higher performance over HTTP/2,
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so it is recommended to use a TLS port.
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-->
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<Port>3333</Port>
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<TLSPort>3334</TLSPort>
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<WorkerCount>1</WorkerCount>
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</LLHLS>
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<WebRTC>
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<Signalling>
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<Port>3333</Port>
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<TLSPort>3334</TLSPort>
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<WorkerCount>1</WorkerCount>
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</Signalling>
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<IceCandidates>
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<!-- Uncomment the line below to use IPv6 ICE Candidate -->
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<!-- <IceCandidate>[::]:10000-10004/udp</IceCandidate> -->
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<IceCandidate>*:10001-10004/udp</IceCandidate>
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<!--
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If you want to stream WebRTC over TCP, specify IP:Port for TURN server.
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This uses the TURN protocol, which delivers the stream from the built-in TURN server to the player's TURN client over TCP.
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-->
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<TcpRelay>*:3478</TcpRelay>
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<!--
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TcpForce is an option to force the use of TCP rather than UDP in WebRTC streaming.
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(You can omit ?transport=tcp accordingly.) If <TcpRelay> is not set, playback may fail.
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-->
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<TcpForce>false</TcpForce>
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<TcpRelayWorkerCount>1</TcpRelayWorkerCount>
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</IceCandidates>
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</WebRTC>
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</Publishers>
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</Bind>
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<Managers>
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<Host>
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<!--
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Add your TLS-covered names here, if your certificate has multiple names
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(e.g. wildcard cert) add more <Name> entries like the commented one
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-->
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<Names>
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<Name>localhost</Name>
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<!-- <Name>*.localhost</Name> -->
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</Names>
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<!--
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Bring the path to your cert and key file, if your certificate contains the whole chain (likely)
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use that for the last parameter. These files are relative to the config folder
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-->
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<TLS>
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<CertPath>cert.crt</CertPath>
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<KeyPath>cert.key</KeyPath>
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<ChainCertPath>cert.crt</ChainCertPath>
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</TLS>
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</Host>
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<API>
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<!-- IMPORTANT: Change this to some username:password of choice -->
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<AccessToken>admin:changeme</AccessToken>
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<CrossDomains>
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<Url>*</Url>
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</CrossDomains>
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</API>
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</Managers>
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<VirtualHosts>
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<VirtualHost>
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<Name>default</Name>
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<!--
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Distribution is a value that can be used when grouping the same vhost distributed across multiple servers.
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This value is output to the events log, so you can use it to aggregate statistics.
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-->
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<Distribution>localhost</Distribution>
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<!--
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Bring the path to your cert and key file, if your certificate contains the whole chain (likely)
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use that for the last parameter. These files are relative to the config folder
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-->
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<Host>
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<!--
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Add your TLS-covered names here, if your certificate has multiple names
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(e.g. wildcard cert) add more <Name> entries like the commented one
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-->
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<Names>
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<Name>localhost</Name>
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<!-- <Name>*.localhost</Name> -->
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</Names>
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<TLS>
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<CertPath>cert.crt</CertPath>
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<KeyPath>cert.key</KeyPath>
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<ChainCertPath>cert.crt</ChainCertPath>
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</TLS>
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</Host>
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<!--
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This is an important bit, it prevents just about anyone from streaming on this instance.
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Set the secret key to some string of choice, then use https://nebula.cafe/ome.html
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to sign URLs for both streaming and playback.
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-->
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<SignedPolicy>
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<PolicyQueryKeyName>policy</PolicyQueryKeyName>
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<SignatureQueryKeyName>signature</SignatureQueryKeyName>
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<!-- IMPORTANT: Change this to some alphanumeric string of choice -->
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<SecretKey>CHANGEME</SecretKey>
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<Enables>
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<Publishers>webrtc,hls,llhls,dash,lldash</Publishers>
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<Providers>rtmp,webrtc,srt</Providers>
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</Enables>
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</SignedPolicy>
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<CrossDomains>
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<Url>*</Url>
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</CrossDomains>
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<!--
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Every application is a streaming endpoint, this example just uses one but you
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could theoretically host as many streams as you want with their own settings.
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-->
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<Applications>
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<Application>
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<Name>app</Name>
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<!-- Application type (live/vod) -->
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<Type>live</Type>
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<OutputProfiles>
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<!-- Enable this configuration if you want to hardware acceleration using GPU -->
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<HWAccels>
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<Decoder>
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<Enable>false</Enable>
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<!--
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Setting for Hardware Modules.
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- xma :Xilinx Media Accelerator
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- qsv :Intel Quick Sync Video
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- nv : Nvidia Video Codec SDK
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- nilogan: Netint VPU
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You can use multiple modules by separating them with commas.
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For example, if you want to use xma and nv, you can set it as follows.
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<Modules>[ModuleName]:[DeviceId],[ModuleName]:[DeviceId],...</Modules>
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<Modules>xma:0,nv:0</Modules>
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-->
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<!-- <Modules>nv</Modules> -->
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</Decoder>
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<Encoder>
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<Enable>false</Enable>
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<!-- <Modules>nv</Modules> -->
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</Encoder>
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</HWAccels>
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<OutputProfile>
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<Name>stream</Name>
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<OutputStreamName>stream</OutputStreamName>
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<!-- Transcoding options go here -->
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<Encodes>
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<!--
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Create bypass_video/audio for exposing the ingest streams
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and serve them as "Source" options. This has basically no overhead.
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-->
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<Video>
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<Name>bypass_video</Name>
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<Bypass>true</Bypass>
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</Video>
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<Audio>
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<Name>bypass_audio</Name>
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<Bypass>True</Bypass>
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</Audio>
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<!--
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Encode the audio as AAC and OPUS. The "BypassIfMatch" will replace the
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encoding with the source audio stream if settings match.e
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This has pretty negligible CPU usage.
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-->
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<Audio>
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<Name>aac</Name>
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<Codec>aac</Codec>
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<Bitrate>128000</Bitrate>
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<Samplerate>48000</Samplerate>
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<Channel>2</Channel>
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<BypassIfMatch>
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<Codec>eq</Codec>
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</BypassIfMatch>
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</Audio>
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<Audio>
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<Name>opus</Name>
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<Codec>opus</Codec>
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<Bitrate>128000</Bitrate>
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<Samplerate>48000</Samplerate>
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<Channel>2</Channel>
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<BypassIfMatch>
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<Codec>eq</Codec>
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</BypassIfMatch>
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</Audio>
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<!--
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The following enables video transcoding.
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You can have as many as you want, but they will cause a lot of CPU usage
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(unless using hardware acceleration)
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-->
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<!--
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<Video>
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<Name>720p</Name>
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<Codec>h264</Codec>
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<Bitrate>2024000</Bitrate>
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<Width>1280</Width>
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<Height>720</Height>
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<Framerate>30</Framerate>
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<KeyFrameInterval>30</KeyFrameInterval>
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<BFrames>0</BFrames>
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<Preset>faster</Preset>
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</Video>
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-->
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</Encodes>
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<!--
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You can provide ABR with Playlist. Currently, ABR is supported in LLHLS and WebRTC.
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You can play this playlist with
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LLHLS : http[s]://<domain>[:port]/<app>/<stream>/<FileName>.m3u8
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WebRTC : ws[s]://<domain>[:port]/<app>/<stream>/<FileName>
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Note that the keywords "playlist" and "chunklist" MUST NOT be included in FileName.
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-->
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<!--
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The WebRTC playlist, there is no reason for using different playlists for LLHLS vs WebRTC
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other than being able to provide different transcodes (if that's what you want) but mostly
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it's for using OPUS instead of AAC for audio (HLS is stinky)
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-->
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<Playlist>
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<Name>webrtc</Name>
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<FileName>webrtc</FileName>
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<Options>
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<WebRtcAutoAbr>true</WebRtcAutoAbr>
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</Options>
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<!-- Each rendition is a quality option made of a video+audio encode, eg. 480p with AAC audio -->
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<Rendition>
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<Name>Source</Name>
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<Video>bypass_video</Video>
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<Audio>bypass_audio</Audio>
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</Rendition>
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<!-- If you enabled transcoding, you can uncomment the following and fiddle with it -->
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|||
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<!--
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<Rendition>
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<Name>720p</Name>
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<Video>720p</Video>
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<Audio>opus</Audio>
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</Rendition>
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-->
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</Playlist>
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<Playlist>
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<Name>llhls</Name>
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<FileName>llhls</FileName>
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<Rendition>
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<Name>Source</Name>
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<Video>bypass_video</Video>
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<Audio>aac</Audio> <!-- Make sure we're using a supported audio format for HLS -->
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</Rendition>
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<!-- If you enabled transcoding, you can uncomment the following and fiddle with it -->
|
|||
|
|
<!--
|
|||
|
|
<Rendition>
|
|||
|
|
<Name>720p</Name>
|
|||
|
|
<Video>720p</Video>
|
|||
|
|
<Audio>opus</Audio>
|
|||
|
|
</Rendition>
|
|||
|
|
-->
|
|||
|
|
</Playlist>
|
|||
|
|
|
|||
|
|
</OutputProfile>
|
|||
|
|
</OutputProfiles>
|
|||
|
|
<Providers>
|
|||
|
|
<WebRTC />
|
|||
|
|
<RTMP />
|
|||
|
|
<SRT />
|
|||
|
|
<RTSPPull />
|
|||
|
|
</Providers>
|
|||
|
|
<Publishers>
|
|||
|
|
<AppWorkerCount>1</AppWorkerCount>
|
|||
|
|
<StreamWorkerCount>8</StreamWorkerCount>
|
|||
|
|
<WebRTC>
|
|||
|
|
<Timeout>30000</Timeout>
|
|||
|
|
<Rtx>true</Rtx>
|
|||
|
|
<Ulpfec>true</Ulpfec>
|
|||
|
|
<JitterBuffer>false</JitterBuffer>
|
|||
|
|
</WebRTC>
|
|||
|
|
<LLHLS>
|
|||
|
|
<OriginMode>false</OriginMode>
|
|||
|
|
<CacheControl>
|
|||
|
|
<MasterPlaylistMaxAge>0</MasterPlaylistMaxAge>
|
|||
|
|
<ChunklistMaxAge>0</ChunklistMaxAge>
|
|||
|
|
<ChunklistWithDirectivesMaxAge>60</ChunklistWithDirectivesMaxAge>
|
|||
|
|
<SegmentMaxAge>-1</SegmentMaxAge>
|
|||
|
|
<PartialSegmentMaxAge>-1</PartialSegmentMaxAge>
|
|||
|
|
</CacheControl>
|
|||
|
|
<ChunkDuration>0.5</ChunkDuration>
|
|||
|
|
<PartHoldBack>1.5</PartHoldBack>
|
|||
|
|
<SegmentDuration>6</SegmentDuration>
|
|||
|
|
<SegmentCount>10</SegmentCount>
|
|||
|
|
<CrossDomains>
|
|||
|
|
<Url>*</Url>
|
|||
|
|
</CrossDomains>
|
|||
|
|
</LLHLS>
|
|||
|
|
<RTMPPush>
|
|||
|
|
</RTMPPush>
|
|||
|
|
</Publishers>
|
|||
|
|
</Application>
|
|||
|
|
</Applications>
|
|||
|
|
</VirtualHost>
|
|||
|
|
</VirtualHosts>
|
|||
|
|
</Server>
|