From version 0.10.4, MPEG-2 TS input is supported. The supported codecs are H.264, AAC(ADTS). Supported codecs will continue to be added. And the current version only supports basic MPEG-2 TS with 188 bytes packet size. Since the information about the input stream is obtained using PAT and PMT, the client must send this table information as required.
To enable MPEG-2 TS, you must bind the ports fist and map the bound ports and streams.
To use multiple streams, it is necessary to bind multiple ports, so we provide a way to bind multiple ports as in the example below. You can use the dash to specify the port as a range, such as Start port-End port, and multiple ports using commas.
First, name the stream and map the port bound above. The macro ${Port} is provided to map multiple streams at once. Check out the example below.
This is an example of publishing using FFMPEG.
Giving the -pes_payload_size 0 option to the AAC codec is very important for AV synchronization and low latency. If this option is not given, FFMPEG bundles several ADTSs and is transmitted at once, which may cause high latency and AV synchronization errors.
OvenMediaEngine supports multiple protocols for input from various live sources, without compromising basic usability. This allows you to publish a variety of live sources with sub-second latency. See the sub-page for more information.
OvenMediaEngine can pull RTSP Stream in two ways. The first way is to use the Stream creation API, and the second way is to use OriginMap or OriginMapStore. The supported codecs are H.264, AAC(ADTS). Supported codecs will continue to be added.
You can create a stream by pulling an RTSP stream using the . For more information on using the , check out that chapter.
If OriginMapStore is configured and Redis Server provides an rtsp URL, OvenMediaEngine pulls the RTSP URL when a playback request comes in. Check out for more details.
RTSP Pull is provided through OriginMap configuration. OriginMap is the rule that the Edge server pulls the stream of the Origin server. Edge server can pull a stream of origin with RTSP and OVT (protocol defined by OvenMediaEngine for Origin-Edge) protocol. See the section for more information about OVT.
For example, in the above setup, when a player requests "ws://ome.com/app_name/rtsp_stream_name" to stream WebRTC, it pulls the stream from "rtsp://192.168.0.200:554" and publishes it to WebRTC.
If the app name set in Location isn't created, OvenMediaEngine creates the app with default settings. The default generated app doesn't have an OPUS encoding profile, so to use WebRTC streaming, you need to add the app to your configuration.
When a playback request comes in from the following URL, RTSP pull starts working according to Origins settings.
User can send video/audio from web browser to OvenMediaEngine via WebRTC without plug-in. Of course, you can use any encoder that supports WebRTC transmission as well as a browser.
OvenMediaEngine supports self-defined signaling protocol and for WebRTC ingest.
You can set the port to use for signaling in <Bind><Provider><WebRTC><Signaling>. <Port> is for setting an unsecured HTTP port, and <TLSPort> is for setting a secured HTTP port that is encrypted with TLS.
For WebRTC ingest, you must set the ICE candidates of the OvenMediaEnigne server to <IceCandidates>. The candidates set in <IceCandate> are delivered to the WebRTC peer, and the peer requests communication with this candidate. Therefore, you must set the IP that the peer can access. If the IP is specified as *, OvenMediaEngine gathers all IPs of the server and delivers them to the peer.
WebRTC input can be turned on/off for each application. As follows Setting enables the WebRTC input function of the application. The <CrossDomains> setting is used in WebRTC signaling.
OvenMediaEnigne supports self-defined signaling protocol and WHIP for WebRTC ingest.
The signaling URL for WebRTC ingest uses the query string ?direction=send as follows to distinguish it from the url for WebRTC playback. Since the self-defined WebRTC signaling protocol is based on WebSocket, you must specify ws[s] as the scheme.
ws[s]://<host>[:signaling port]/<app name>/<stream name>?direction=send
For ingest from the WHIP client, put ?direction=whip in the query string in the signaling URL as in the example below. Since WHIP is based on HTTP, you must specify http[s] as the scheme.
http[s]://<host>[:signaling port]/<app name>/<stream name>?direction=whip
ws[s]://<host>[:port]/<app name>/<stream name>?direction=send&transport=tcp
http[s]://<host>[:port]/<app name>/<stream name>?direction=whip&transport=tcp
To use WebRTC/tcp, <TcpRelay> must be turned on in <Bind> setting.
If <TcpForce> is set to true, it works over TCP even if you omit the ?transport=tcp query string from the URL.
We provide a demo page so you can easily test your WebRTC input. You can access the demo page at the URL below.
Providers ingests streams that come from a media source. OvenMediaEngine supports RTMP protocol. You can set it in the configuration as follows:
When a live source inputs to the <Application>, a stream is automatically created in the <Application>. The created stream is passed to Encoder and Publisher.
If you set up a live stream using an RTMP-based encoder, you need to set the following in Server.xml:
<BlockDuplicateStreamName> is a policy for streams that are inputted as overlaps.
<BlockDuplicateStreamName> works with the following rules:
To allow the duplicated stream name feature can cause several problems. When a new stream is an input the player may be disconnected. Most encoders have the ability to automatically reconnect when it is disconnected from the server. As a result, two encoders compete and disconnect each other, which can cause serious problems in playback.
If you want to publish the source stream, you need to set the following in the Encoder:
URL RTMP://<OvenMediaEngine IP>[:<RTMP Listen Port>]/<App Name]>
Stream Key Stream Name
If you use the default configuration, the <RTMP><ListenPort> is 1935, which is the default port for RTMP. So it can be omitted. Also, since the Application named app is created by default in the default configuration, you can enter app in the [App Name]. You can define a Stream Key and use it in the Encoder, and the Streaming URL will change according to the Stream Key.
Moreover, some encoders can include a stream key in the URL, and if you use these encoders, you need to set it as follows:
URL RTMP://<OvenMediaEngine IP>[:<RTMP Listen Port>/<App Name>/<Stream Name>
If you are using the default configuration, press the URL button in the top right corner of OvenStreamEnoder, and enter the URL as shown below:
Also, <App name> and <Stream name> can be changed and used as desired in the configuration.
If you use the default configuration, set the OBS as follows:
You can set the Stream Key to any name you like at any time.
Pulling type providers are activated by streaming requests from publishers. And by default, the provider is automatically disabled after 30 seconds of no client playback. If you want to change this setting, check out the chapter.
<TcpRelay> means OvenMediaEngine's built-in TURN Server. When this is enabled, the address of this turn server is passed to the peer via self-defined signaling protocol or WHIP, and the peer communicates with this turn server over TCP. This allows OvenMediaEngine to support WebRTC/TCP itself. For more information on URL settings, check out .
WebRTC transmission is sensitive to packet loss because it affects all players who access the stream. Therefore, it is recommended to provide WebRTC transmission over TCP. OvenMediaEngine has a built-in TURN server for WebRTC/TCP, and receives or transmits streams using the TCP session that the player's TURN client connects to the TURN server as it is. To use WebRTC/TCP, use transport=tcp query string as in WebRTC playback. See for more information.
The getUserMedia API to access the local device only works in a . So, the WebRTC Input demo page can only work on the https site **** . This means that due to you have to install the certificate in OvenMediaEngine and use the signaling URL as wss to test this. If you can't install the certificate in OvenMediaEngine, you can temporarily test it by allowing the insecure content of the demo.ovenplayer.com URL in your browser.
To create a custom WebRTC Producer, you need to implement OvenMediaEngine's Self-defined Signaling Protocol or WHIP. Self-defined protocol is structured in a simple format and uses the.
When the player connects to ws[s]://host:port/app/stream?direction=send through a web socket and sends a request offer command, the server responds to the offer sdp. If transport=tcp exists in the query string of the URL, information is included in offer sdp, which contains the information of OvenMediaEngine's built-in TURN server, so you need to set this in RTCPeerConnection to use WebRTC/TCP. The player then setsRemoteDescription and addIceCandidate offer sdp, generates an answer sdp, and responds to the server.
Protocol
URL
WebRTC
ws[s]:://host.com[:port]/app_name/rtsp_stream_name
LLHLS
http[s]://host.com[:port]/app_name/rtsp_stream_name/llhls.m3u8
true
Default Rejects the new stream inputted as overlap and maintains the existing stream.
false
Accepts a new stream inputted as overlap and disconnects the existing stream.
Secure Reliable Transport (or SRT in short) is an open source video transport protocol and technology stack that optimizes streaming performance across unpredictable networks with secure streams and easy firewall traversal, bringing the best quality live video over the worst networks. We consider SRT to be one of the great alternatives to RTMP, and OvenMediaEngine can receive video streaming over SRT. For more information on SRT, please visit the SRT Alliance website.
SRT uses the MPEG-TS format when transmitting live streams. This means that unlike RTMP, it can support many codecs. Currently, OvenMediaEngine supports H.264, H.265, and AAC codecs received by SRT.
Set the SRT listen port as follows:
SRT input can be turned on/off for each application. As follows Setting enables the SRT input function of the application.
There are various encoders that support SRT such as FFMPEG, OBS Studio, and srt-live-transmit. Please check the specifications of each encoder on how to transmit streams through SRT from the encoder. We describe an example using OBS Studio.
OvenMediaEngine classifies each stream using SRT's streamid. This means that unlike MEPG-TS/udp, OvenMediaEngine can receive multiple SRT streams through one port. For more information on streamid, see Haivision's official documentation.
Therefore, in order for the SRT encoder to transmit a stream to OvenMediaEngine, the following information must be included in the streamid as percent encoded.
streamid = percent_encoding("srt://{host}[:port]/{app name}/{stream name}[?query=value]")
The streamid contains the URL format, so it must be percent encoded****
OBS Studio 25.0 or later supports SRT. Please refer to the OBS official documentation for more information. Enter the address of OvenMediaEngine in OBS Studio's Server as follows: When using SRT in OBS, you can leave the Stream Key blank.
srt://ip:port?streamid=srt%3A%2F%2F{domain or IP address}[%3APort]%2F{App name}%2F{Stream name}
You can configure SRT's socket options using <Options>. This is particularly useful when setting the encryption for SRT, and you can specify a passphrase by configuring as follows:
For more information on SRT socket options, please refer to https://github.com/Haivision/srt/blob/master/docs/API/API-socket-options.md#list-of-options.
