ammar1798-server.xml
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<?xml version="1.0" encoding="UTF-8"?>
<Server version="8">
<Name>OvenMediaEngine</Name>
<!-- Host type (origin/edge) -->
<Type>origin</Type>
<!-- Specify IP address to bind (* means all IPs) -->
<IP>*</IP>
<PrivacyProtection>false</PrivacyProtection>
<!--
To get the public IP address(mapped address of stun) of the local server.
This is useful when OME cannot obtain a public IP from an interface, such as AWS or docker environment.
If this is successful, you can use ${PublicIP} in your settings.
-->
<StunServer>stun.l.google.com:19302</StunServer>
<!-- Settings for the ports to bind -->
<Bind>
<!-- Enable this configuration if you want to use API Server -->
<Managers>
<API>
<Port>8081</Port>
<TLSPort>8082</TLSPort>
<WorkerCount>1</WorkerCount>
</API>
</Managers>
<Providers>
<!-- Pull providers -->
<RTSPC>
<WorkerCount>1</WorkerCount>
</RTSPC>
<OVT>
<WorkerCount>1</WorkerCount>
</OVT>
<!-- Push providers -->
<RTMP>
<Port>1935</Port>
<WorkerCount>1</WorkerCount>
</RTMP>
<SRT>
<Port>9999</Port>
<WorkerCount>1</WorkerCount>
</SRT>
<MPEGTS>
<!--
Listen on port 4000~4005 (<Port>4000-4004,4005/udp</Port>)
This is just a demonstration to show that you can configure the port in several ways
-->
<Port>4000/udp</Port>
</MPEGTS>
<WebRTC>
<Signalling>
<Port>3333</Port>
<!-- If you want to use TLS, specify the TLS port -->
<TLSPort>3334</TLSPort>
<WorkerCount>1</WorkerCount>
</Signalling>
<IceCandidates>
<!--
If you want to stream WebRTC over TCP, specify IP:Port for TURN server.
This uses the TURN protocol, which delivers the stream from the built-in TURN server to the player's TURN client over TCP.
For detailed information, refer https://airensoft.gitbook.io/ovenmediaengine/streaming/webrtc-publishing#webrtc-over-tcp
-->
<TcpRelay>*:3478</TcpRelay>
<!-- TcpForce is an option to force the use of TCP rather than UDP in WebRTC streaming. (You can omit ?transport=tcp accordingly.) If <TcpRelay> is not set, playback may fail. -->
<TcpForce>true</TcpForce>
<IceCandidate>*:10000/udp</IceCandidate>
<TcpRelayWorkerCount>1</TcpRelayWorkerCount>
</IceCandidates>
</WebRTC>
</Providers>
<Publishers>
<OVT>
<Port>9000</Port>
<WorkerCount>1</WorkerCount>
</OVT>
<HLS>
<Port>90</Port>
<!-- If you want to use TLS, specify the TLS port -->
<!-- <TLSPort>8443</TLSPort> -->
<WorkerCount>1</WorkerCount>
</HLS>
<DASH>
<Port>90</Port>
<!-- If you want to use TLS, specify the TLS port -->
<!-- <TLSPort>8443</TLSPort> -->
<WorkerCount>1</WorkerCount>
</DASH>
<WebRTC>
<Signalling>
<Port>3333</Port>
<!-- If you want to use TLS, specify the TLS port -->
<TLSPort>3334</TLSPort>
<WorkerCount>1</WorkerCount>
</Signalling>
<IceCandidates>
<!--
If you want to stream WebRTC over TCP, specify IP:Port for TURN server.
This uses the TURN protocol, which delivers the stream from the built-in TURN server to the player's TURN client over TCP.
For detailed information, refer https://airensoft.gitbook.io/ovenmediaengine/streaming/webrtc-publishing#webrtc-over-tcp
-->
<TcpRelay>*:3478</TcpRelay>
<!-- TcpForce is an option to force the use of TCP rather than UDP in WebRTC streaming. (You can omit ?transport=tcp accordingly.) If <TcpRelay> is not set, playback may fail. -->
<TcpForce>true</TcpForce>
<IceCandidate>*:10000/udp</IceCandidate>
<TcpRelayWorkerCount>1</TcpRelayWorkerCount>
</IceCandidates>
</WebRTC>
</Publishers>
</Bind>
<!-- P2P works only in WebRTC -->
<!--
<P2P>
<MaxClientPeersPerHostPeer>2</MaxClientPeersPerHostPeer>
</P2P>
-->
<!--
Enable this configuration if you want to use API Server
<AccessToken> is a token for authentication, and when you invoke the API, you must put "Basic base64encode(<AccessToken>)" in the "Authorization" header of HTTP request.
For example, if you set <AccessToken> to "ome-access-token", you must set "Basic b21lLWFjY2Vzcy10b2tlbg==" in the "Authorization" header.
-->
<Managers>
<Host>
<Names>
<Name>oven.domain.com</Name>
</Names>
<TLS>
<CertPath>/etc/letsencrypt/live/domain.com/fullchain.pem</CertPath>
<KeyPath>/etc/letsencrypt/live/domain.com/privkey.pem</KeyPath>
<ChainCertPath>/etc/letsencrypt/live/domain.com/fullchain.pem</ChainCertPath>
</TLS>
</Host>
<API>
<AccessToken>key123</AccessToken>
</API>
</Managers>
<VirtualHosts>
<!-- You can use wildcard like this to include multiple XMLs -->
<VirtualHost include="VHost*.xml" />
<VirtualHost>
<Name>default</Name>
<!--Distribution is a value that can be used when grouping the same vhost distributed across multiple servers. This value is output to the events log, so you can use it to aggregate statistics. -->
<Distribution>ovenmediaengine.com</Distribution>
<!-- Settings for multi ip/domain and TLS -->
<Host>
<Names>
<!-- Host names
<Name>stream1.airensoft.com</Name>
<Name>stream2.airensoft.com</Name>
<Name>*.sub.airensoft.com</Name>
<Name>192.168.0.1</Name>
-->
<Name>oven.domain.com</Name>
</Names>
<TLS>
<CertPath>/etc/letsencrypt/live/domain.com/fullchain.pem</CertPath>
<KeyPath>/etc/letsencrypt/live/domain.com/privkey.pem</KeyPath>
<ChainCertPath>/etc/letsencrypt/live/domain.com/fullchain.pem</ChainCertPath>
</TLS>
</Host>
<!--
Refer https://airensoft.gitbook.io/ovenmediaengine/signedpolicy
<SignedPolicy>
<PolicyQueryKeyName>policy</PolicyQueryKeyName>
<SignatureQueryKeyName>signature</SignatureQueryKeyName>
<SecretKey>aKq#1kj</SecretKey>
<Enables>
<Providers>rtmp,webrtc,srt</Providers>
<Publishers>webrtc,hls,dash,lldash</Publishers>
</Enables>
</SignedPolicy>
-->
<!--
<AdmissionWebhooks>
<TargetUrl></TargetUrl>
<SecretKey></SecretKey>
<Timeout>3000</Timeout>
<Enables>
<Providers>rtmp,webrtc,srt</Providers>
<Publishers>webrtc,hls,dash,lldash</Publishers>
</Enables>
</AdmissionWebhooks>
-->
<!-- <Origins>
<Properties>
<NoInputFailoverTimeout>3000</NoInputFailoverTimeout>
<UnusedStreamDeletionTimeout>60000</UnusedStreamDeletionTimeout>
</Properties>
<Origin>
<Location>/app/stream</Location>
<Pass>
<Scheme>ovt</Scheme>
<Urls><Url>origin.com:9000/app/stream_720p</Url></Urls>
</Pass>
</Origin>
<Origin>
<Location>/app/</Location>
<Pass>
<Scheme>ovt</Scheme>
<Urls><Url>origin.com:9000/app/</Url></Urls>
</Pass>
</Origin>
<Origin>
<Location>/edge/</Location>
<Pass>
<Scheme>ovt</Scheme>
<Urls><Url>origin.com:9000/app/</Url></Urls>
</Pass>
</Origin>
</Origins> -->
<!-- Settings for applications -->
<Applications>
<Application>
<Name>app</Name>
<!-- Application type (live/vod) -->
<Type>live</Type>
<OutputProfiles>
<!-- Enable this configuration if you want to hardware acceleration using GPU -->
<HardwareAcceleration>false</HardwareAcceleration>
<OutputProfile>
<Name>bypass_stream</Name>
<OutputStreamName>${OriginStreamName}</OutputStreamName>
<Encodes>
<Audio>
<Bypass>true</Bypass>
</Audio>
<Video>
<Bypass>true</Bypass>
</Video>
<Audio>
<Codec>opus</Codec>
<Bitrate>128000</Bitrate>
<Samplerate>48000</Samplerate>
<Channel>2</Channel>
</Audio>
<!--
<Video>
<Codec>vp8</Codec>
<Bitrate>1024000</Bitrate>
<Framerate>30</Framerate>
<Width>1280</Width>
<Height>720</Height>
<Preset>faster</Preset>
</Video>
-->
</Encodes>
</OutputProfile>
</OutputProfiles>
<Providers>
<OVT />
<WebRTC />
<RTMP />
<SRT />
<MPEGTS>
<StreamMap>
<!--
Set the stream name of the client connected to the port to "stream_${Port}"
For example, if a client connets to port 4000, OME creates a "stream_4000" stream
<Stream>
<Name>stream_${Port}</Name>
<Port>4000,4001-4004</Port>
</Stream>
<Stream>
<Name>stream_4005</Name>
<Port>4005</Port>
</Stream>
-->
<Stream>
<Name>stream_${Port}</Name>
<Port>4000</Port>
</Stream>
</StreamMap>
</MPEGTS>
<RTSPPull />
<WebRTC>
<Timeout>30000</Timeout>
</WebRTC>
</Providers>
<Publishers>
<AppWorkerCount>1</AppWorkerCount>
<StreamWorkerCount>8</StreamWorkerCount>
<OVT />
<WebRTC>
<Timeout>30000</Timeout>
<Rtx>false</Rtx>
<Ulpfec>false</Ulpfec>
<JitterBuffer>false</JitterBuffer>
</WebRTC>
<HLS>
<SegmentDuration>5</SegmentDuration>
<SegmentCount>3</SegmentCount>
<CrossDomains>
<Url>*</Url>
</CrossDomains>
</HLS>
<DASH>
<SegmentDuration>5</SegmentDuration>
<SegmentCount>3</SegmentCount>
<CrossDomains>
<Url>*</Url>
</CrossDomains>
<!--
Enable DASH player to obtain UTCTiming from OME using /time?iso&ms API
-->
<UTCTiming>
<Scheme>urn:mpeg:dash:utc:http-xsdate:2014</Scheme>
<Value>/time?iso&ms</Value>
</UTCTiming>
</DASH>
<LLDASH>
<SegmentDuration>5</SegmentDuration>
<CrossDomains>
<Url>*</Url>
</CrossDomains>
<!--
Use default options for UTCTiming
- scheme: urn:mpeg:dash:utc:http-xsdate:2014
- value: /time?iso&ms
-->
<UTCTiming />
</LLDASH>
</Publishers>
</Application>
</Applications>
</VirtualHost>
</VirtualHosts>
</Server>
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