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0.16.5 Introduction

What is OvenMediaEngine?

OvenMediaEngine (OME) is a Sub-Second Latency Live Streaming Server with Large-Scale and High-Definition. With OME, you can create platforms/services/systems that transmit high-definition video to hundreds-thousand viewers with sub-second latency and be scalable, depending on the number of concurrent viewers.

OvenMediaEngine can receive a video/audio, video, or audio source from encoders and cameras such as , OBS, XSplit, and more, to WebRTC, SRT, RTMP, MPEG-2 TS, and RTSP as Input. Then, OME transmits this source using LLHLS (Low Latency HLS) and WebRTC as output. Also, we provide , an Open-Source and JavaScript-based WebRTC/LLHLS Player for OvenMediaEngine.

Our goal is to make it easier for you to build a stable broadcasting/streaming service with sub-second latency.

Features

Ingest
- Push: WebRTC, WHIP, SRT, RTMP, MPEG-2 TS
- Pull: RTSP

Supported Platforms

We have tested OvenMediaEngine on platforms, listed below. However, we think it can work with other Linux packages as well:

Docker ()
Ubuntu 18+
CentOS 7+

Getting Started

Please read chapter in the tutorials.

How to Contribute

Thank you so much for being so interested in OvenMediaEngine.

We need your help to keep and develop our open-source project, and we want to tell you that you can contribute in many ways. Please see our , , and .

We always hope that OvenMediaEngine will give you good inspiration.

For more information

License

OvenMediaEngine is licensed under the . However, if you need another license, please feel free to email us at .

Quick Start

This page provides the fastest way to check playback of WebRTC and LLHLS using OvenMediaEngine. For installation and detailed settings, please refer to other pages.

Run OvenMediaEngine

Run docker with the command below. OME_HOST_IP must be an IP address accessible by the player.

Check Status

You can check the docker container status with the following command:

You can view the log with the command below. This is important because you can check the version of OvenMediaEngine that is running.

Run OvenPlayer Demo

Check Status

You can access the OvenPlayerDemo docker container with a browser as shown below.

http://Your.Docker.Host.IP:8090/

Publishing

Publish your live stream to OvenMediaEngine using a live encoder like .

Ingest URLs

RTMP - rtmp://Your.Docker.Host.IP:1935/app/stream

SRT - srt://Your.Docker.Host.IP:9999?streamid=srt%3A%2F%2FYour.Docker.Host.IP%3A9999%2Fapp%2Fstream

WHIP - http://Your.Docker.Host.IP:3333/app/stream?direction=whip

The RTMP publishing address is :

Server rtmp://Your.Docker.Host.IP:1935/app

Stream Key stream

The settings below are recommended for ultra-low latency.

Setting

Value

Playback

Open the installed OvenPlayer Demo page in your browser.

http://Your.Docker.Host.IP:8090/

WebRTC Playback

Add ws://Your.Docker.Host.IP:3333/app/stream to the Playback URL and click the ADD SOURCE and LOAD PLAYER button to play the live stream with WebRTC.

LLHLS Playback

Add http://Your.Docker.Host.IP:3333/app/stream/llhls.m3u8 to the Playback URL and click the ADD SOURCE and LOAD PLAYER button to play the live stream with LLHLS.

Online Demo

Player & WebRTC Encoder

We provide online demos of OvenPlayer(WebRTC/LLHLS Player) and OvenLiveKit(WebRTC Live Encoder) so that users can easily test out OvenMediaEngine.

Getting Started

Getting Started with Docker Image

OvenMediaEngine provides Docker images from AirenSoft's Docker Hub (airensoft/ovenmediaengine) repository. You can easily use OvenMediaEngine server by using Docker image. See Getting Started with Docker for details.

Getting Started with Source Code

Installing dependencies

OvenMediaEngine can work with a variety of open-sources and libraries. First, install them on your clean Linux machine as described below. We think that OME can support most Linux packages, but the tested platforms we use are Ubuntu 18+, Fedora 28+, and CentOS 7+.

If the prerequisites.sh script fails, try to run sudo apt-get update and rerun it. If it's not enough proceed with the .

Building & Running

You can build the OvenMediaEngine source using the following command:

In addition, we recommend that you permanently set environment variables as follows.

if systemctl start ovenmediaengine fails in Fedora, SELinux may be the cause. See .

Ports used by default

The default configuration uses the following ports, so you need to open it in your firewall settings.

Port

Purpose

To use TLS, you must set up a certificate. See for more information.

You can open firewall ports as in the following example:

TLS Encryption

Most browsers can't load resources via HTTP and WS (WebSocket) from HTTPS web pages secured with TLS. Therefore, if the player is on an HTTPS page, the player must request streaming through "https" and "wss" URLs secured with TLS. In this case, you must apply the TLS certificate to the OvenMediaEngine.

You can set the port for TLS in TLSPort. Currently, LLHLS and WebRTC Signaling support TLS.

<Bind>
    <!-- For API Server -->
    <Managers>
	    <API>
	    <Port>8081</Port>
	    <TLSPort>8082</TLSPort>
            <WorkerCount>1</WorkerCount>
	</API>
    </Managers>
    <!-- For Providers -->
    <Providers>
        <WebRTC>
	    <Signalling>
		<Port>3333</Port>
		<TLSPort>3334</TLSPort>
		<WorkerCount>1</WorkerCount>
	    </Signalling>
            ...
        </WebRTC>
    </Providers>
    <!- For Publishers -->
    <Publishers>
        <LLHLS>
            <Port>80</Port>
            <TLSPort>443</TLSPort>
        </LLHLS>
        <WebRTC>
            <Signalling>
                <Port>3333</Port>
                <TLSPort>3334</TLSPort>
            </Signalling>
            ...
        </WebRTC>
    </Publishers>
</Bind>

Add your certificate files to as follows:

To enable HTTP for HLS and WebRTC signaling servers, you must enable the TLS element and install the certificate file in PEM format. This involves indicating a server certificate through the CertPath, as well as a private key file through the KeyPath. These paths can be specified as either absolute or relative paths from the executable. However, if the server certificate was issued using an intermediate certificate, some browsers may raise concerns about the certificate's authenticity. To address this, a bundle of chained certificates provided by a Certificate Authority can be set in the ChainCertPath.

Assuming the certificate settings are correctly configured, WebRTC streaming can then be played via the wss://url protocol, while LLHLS streaming can be accessed via .

Live Source

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.

RTMP

Configuration

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.

RTMP live stream

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:

Value

Description

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.

Publish

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>

Example with OvenLiveKit (OvenStreamEncoder)

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.

Example with OBS

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.

WebRTC / WHIP

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.

Configuration

MPEG-2 TS

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.

Configuration

To enable MPEG-2 TS, you must bind the ports fist and map the bound ports and streams.

RTSP Pull

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.

Pulling streams using the Stream Creation API

You can create a stream by pulling an RTSP stream using the Stream Creation API. For more information on using the REST API, check out that chapter.

Pulling streams using the OriginMapStore

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.

Pulling streams using the OriginMap

Configuration

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.

Event to trigger pulling

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.

When a playback request comes in from the following URL, RTSP pull starts working according to Origins settings.

Scheduled Channel

Scheduled Channel that allows you to create a live channel by scheduling pre-recorded files has been added to OvenMediaEngine. Other services or software call this Pre-recorded Live or File Live, but OvenMediaEngine plans to expand the function to organize live channels as a source, so we named it Scheduled Channel.

Getting Started

To use this feature, activate Schedule Provider as follows.

MediaRootDir

Multiplex Channel

Now with Multiplex Channel, you can configure ABR by combining multiple input streams into one, or duplicate external streams and send them to other applications.

Multiplex Channel takes tracks from other local streams and organizes them into its own tracks. This will pull in tracks that have already been encoded, which can be useful if you want to change codecs or adjust the quality once again. And the Multiplex Channel is sent to the publisher, unconditionally bypassing the encoder.

Configuration

To use this feature, enable Multiplex Provider in Server.xml.

Streaming

Low-Latency DASH and Legacy HLS streaming

From OvenMediaEngine v0.14.0, updates to legacy HLS, DASH, and LLDASH are now discontinued. These will be deprecated.

LLHLS, released from v0.14.0, is superior to Dash and LLDASH in all aspects of compatibility, performance and function, and also support legacy HLS players. Therefore, we decided not to update legacy HLS, DASH and LLDASH anymore. With the energy that was used to maintain these features, we will focus on more wonderful features in the future.

Access Control

Thumbnail

OvenMediaEngine can generate thumbnails from live streams. This allows you to organize a broadcast list on your website or monitor multiple streams at the same time.

Configuration

Bind

Thumbnails are published via HTTP(s). Set the port for thumbnails as follows. Thumbnail publisher can use the same port number as HLS and DASH.

Encoding

In order to publish thumbnails, an encoding profile for thumbnails must be set. JPG and PNG are supported as codec. And framerate and resolution can be adjusted. Framerate is the number of thumbnails extracted per second. We recommend 1 as the thumbnail framerate. Thumbnail encoding uses a lot of resources. Therefore, if you increase this value excessively, it can cause a failure due to excessive use of system resources. The resolution can be set as desired by the user, and if the ratio is different from the input image, it is stretched. We plan to support various ratio modes in the future.

Publisher

Declaring a thumbnail publisher. Cross-domain settings are available as a detailed option.

Get thumbnails

When the setting is made for the thumbnail and the stream is input, you can view the thumbnail through the following URL.

Push Publishing

OvenMediaEngine supports Push Publishing function that can retransmit live streams to other systems. The protocol supported for retransmission uses SRT, RTMP or MPEGTS. Because, most services and products support this protocol. also, one output stream can be transmitted to multiple destinations at the same time. You can start and stop pushing the output stream through REST API. Note that the only codecs that can be retransmitted in RTMP and MPEGTS protocol are H264 and AAC.

Configuration

v1

Reload Certificate

Reload All Certificates

Batch reload certificates of all Virtual Hosts. In case of failure, the existing certificate will continue to be used.

Request

POST /v1/vhosts:reloadAllCertificates

Header

Responses

200 Ok

The request has succeeded

Header

Body

500 Internal Server Error

Failed to reload certificate. Keep the existing certificate. The reason for the failure can be found in the server logs.

Reload Certificate

Reload the certificate of the specified Virtual Hosts. In case of failure, the existing certificate will continue to be used.

Request

POST /v1/vhosts/{vhost}:reloadCertificate

Header

Responses

200 Ok

The request has succeeded

Header

Body

500 Internal Server Error

Failed to reload certificate. Keep the existing certificate. The reason for the failure can be found in the server logs.

Statistics

P2P Delivery (Experiment)

OvenMediaEngine provides P2P Delivery to be able to distribute Edge Traffic to Player. This feature is currently the Preview version, and if you want to use it, you need only to use OvenPlayer. Moreover, we plan to perform more experiments in various real-world and then upgrade it to the full version in OvenMediaEngine.

First of all, we have rules. The peer that sends the Traffic in the P2P network is called a Host Peer, and the peer that receives the Traffic from the Host Peer is called a Client Peer. Also, P2P Delivery in OvenMediaEngine doesn't designate the Client Peer as the Host Peer again. In other words, it only operates as 1 Depth.

What are the benefits of using P2P Delivery?

According to our experiments so far, P2P Delivery provides the best performance and stability when using 1 Depth to connect between Players and connecting up to two Players to one Player.

In other words, P2P Delivery has distributed two-thirds of existing Traffic. So, this means that it can expand the Capacity of the Edge Network by three times and reduce Traffic costs by two-thirds.

How does it work?

You can use the P2P function of OvenMediaEngine by adding the <P2P> element as the following settings:

Also, If you want to use P2P Delivery when your OvenMediaEngine is running in Origin-Edge Cluster-Mode, you need to apply this setting to all the Edges. You can instantly test P2P Delivery with OvenPlayer.

<MaxClientPeersPerHostPeer> sets the number of Client Peers connecting to one Host Peer.

How does it classify Peers?

When OvenMediaEngine receives a WebRTC connection request from a new player, it determines the Host Peer or Client Peer according to the following rules:

Qualification for Host Peer

Qualification for Client Peer

When any Host Peer is disconnected, OvenMediaEngine detects this situation and immediately reconnects the Client Peer connected to that Host Peer to the Edge to ensure stability.

Also, we are preparing a smarter algorithm based on user location, platform performance, and network statistical information for classifying Host Peers or Client Peers.

If you have a better idea, we hope that you improve our code and contribute to our project. Please visit .

getroot@Jeheon-Main:/var/log/ovenmediaengine$ cat ovenmediaengine.log
[03-27 19:59:13.221] I 10996 Config | config_manager.cpp:144  | Trying to set logfile in directory... (/var/log/ovenmediaengine)
[03-27 19:59:13.221] I 10996 Config | config_manager.cpp:47   | Trying to load configurations... (origin_conf/Server.xml)
[03-27 19:59:13.235] I 10996 OvenMediaEngine | main.cpp:211  | OvenMediaEngine v0.9.5 (v0.9.1-422-g6e4b7ce) is started on [Jeheon-Main] (Linux x86_64 - 4.4.0-18362-Microsoft, #476-Microsoft Fri Nov 01 16:53:00 PST 2019)
[03-27 19:59:13.235] I 10996 OvenMediaEngine | main.cpp:213  | With modules:
[03-27 19:59:13.235] I 10996 OvenMediaEngine | main.cpp:214  |   FFmpeg 3.4.2
[03-27 19:59:13.235] I 10996 OvenMediaEngine | main.cpp:215  |     Configuration: --prefix=/opt/ovenmediaengine --enable-gpl --enable-nonfree --extra-cflags=-I/opt/ovenmediaengine/include --extra-ldflags='-L/opt/ovenmediaengine/lib -Wl,-rpath,/opt/ovenmediaengine/lib' --extra-libs=-ldl --enable-shared --disable-static --disable-debug --disable-doc --disable-programs --disable-avdevice --disable-dct --disable-dwt --disable-error-resilience --disable-lsp --disable-lzo --disable-rdft --disable-faan --disable-pixelutils --disable-everything --enable-zlib --enable-libopus --enable-libvpx --enable-libfdk_aac --enable-libx264 --enable-encoder='libvpx_vp8,libvpx_vp9,libopus,libfdk_aac,libx264' --enable-decoder='aac,aac_latm,aac_fixed,h264' --enable-parser='aac,aac_latm,aac_fixed,h264' --enable-network --enable-protocol=tcp --enable-protocol=udp --enable-protocol=rtp --enable-demuxer=rtsp --enable-filter='asetnsamples,aresample,aformat,channelmap,channelsplit,scale,transpose,fps,settb,asettb'
[03-27 19:59:13.235] I 10996 OvenMediaEngine | main.cpp:216  |     libavformat: 57.83.100
[03-27 19:59:13.235] I 10996 OvenMediaEngine | main.cpp:217  |     libavcodec: 57.107.100
[03-27 19:59:13.235] I 10996 OvenMediaEngine | main.cpp:218  |     libavutil: 55.78.100
[03-27 19:59:13.235] I 10996 OvenMediaEngine | main.cpp:219  |     libavfilter: 6.107.100
[03-27 19:59:13.235] I 10996 OvenMediaEngine | main.cpp:220  |     libswresample: 2.9.100
[03-27 19:59:13.235] I 10996 OvenMediaEngine | main.cpp:221  |     libswscale: 4.8.100
[03-27 19:59:13.235] I 10996 OvenMediaEngine | main.cpp:222  |   SRT: 1.3.3
[03-27 19:59:13.235] I 10996 OvenMediaEngine | main.cpp:223  |   SRTP: libsrtp2 2.2.0
[03-27 19:59:13.235] I 10996 OvenMediaEngine | main.cpp:224  |   OpenSSL: OpenSSL 1.1.0g  2 Nov 2017
[03-27 19:59:13.235] I 10996 OvenMediaEngine | main.cpp:225  |     Configuration: compiler: gcc -DDSO_DLFCN -DHAVE_DLFCN_H -DNDEBUG -DOPENSSL_THREADS -DOPENSSL_NO_STATIC_ENGINE -DOPENSSL_PIC -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DRC4_ASM -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DGHASH_ASM -DECP_NISTZ256_ASM -DPADLOCK_ASM -DPOLY1305_ASM -DOPENSSLDIR="\"/opt/ovenmediaengine\"" -DENGINESDIR="\"/opt/ovenmediaengine/lib/engines-1.1\""  -Wa,--noexecstack
[03-27 19:59:13.240] I 10996 Monitor | monitoring.cpp:35   | Create HostMetrics(default) for monitoring
[03-27 19:59:13.240] I 10996 OvenMediaEngine | main.cpp:148  | Trying to create a module MediaRouter for [default] host...
[03-27 19:59:13.240] I 10996 MediaRouter | media_router.cpp:40   | MediaRouter has been started.
[03-27 19:59:13.240] I 10996 OvenMediaEngine | main.cpp:151  | Trying to create a module RTMP Provider for [default] host...
[03-27 19:59:13.244] I 10996 RtmpProvider | rtmp_provider.cpp:63   | RTMP Server has started listening on 0.0.0.0:1935...
[03-27 19:59:13.246] I 10996 Provider | provider.cpp:40   | RtmpProvider has been started.
[03-27 19:59:13.246] I 10996 OvenMediaEngine | main.cpp:152  | Trying to create a module OVT Provider for [default] host...
[03-27 19:59:13.248] I 10996 Provider | provider.cpp:40   | OvtProvider has been started.
[03-27 19:59:13.248] I 10996 OvenMediaEngine | main.cpp:153  | Trying to create a module RTSPC Provider for [default] host...
[03-27 19:59:13.250] I 10996 Provider | provider.cpp:40   | RtspcProvider has been started.
[03-27 19:59:13.250] I 10996 OvenMediaEngine | main.cpp:154  | Trying to create a module RTSP Provider for [default] host...
[03-27 19:59:13.250] I 10996 RtspProvider | rtsp_provider.cpp:40   | RTSP is disabled in the configuration.
[03-27 19:59:13.251] I 10996 OvenMediaEngine | main.cpp:157  | Trying to create a module Transcoder for [default] host...
[03-27 19:59:13.251] I 10996 Transcoder | transcoder.cpp:38   | Transcoder has been started.
[03-27 19:59:13.251] I 10996 OvenMediaEngine | main.cpp:160  | Trying to create a module WebRTC Publisher for [default] host...
[03-27 19:59:13.251] I 10996 Signalling | rtc_signalling_server.cpp:74   | P2P is disabled in the configuration
[03-27 19:59:13.258] I 10996 Ice | ice_port.cpp:89   | ICE port is bound to 0.0.0.0:10000/UDP
[03-27 19:59:13.260] I 10996 Ice | ice_port.cpp:89   | ICE port is bound to 0.0.0.0:10001/UDP
[03-27 19:59:13.261] I 10996 Ice | ice_port.cpp:89   | ICE port is bound to 0.0.0.0:10002/UDP
[03-27 19:59:13.263] I 10996 Ice | ice_port.cpp:89   | ICE port is bound to 0.0.0.0:10003/UDP
[03-27 19:59:13.264] I 10996 Ice | ice_port.cpp:89   | ICE port is bound to 0.0.0.0:10004/UDP
[03-27 19:59:13.266] I 10996 Ice | ice_port.cpp:89   | ICE port is bound to 0.0.0.0:10005/UDP
[03-27 19:59:13.266] I 10996 Publisher | publisher.cpp:15   | WebRTC Publisher has been started.
[03-27 19:59:13.266] I 10996 WebRTC | webrtc_publisher.cpp:89   | WebRTC Publisher has started listening on 0.0.0.0:3333...
[03-27 19:59:13.266] I 10996 Publisher | publisher.cpp:15   | WebRTC Publisher has been started.
[03-27 19:59:13.266] I 10996 OvenMediaEngine | main.cpp:161  | Trying to create a module HLS Publisher for [default] host...
[03-27 19:59:13.273] I 10996 Publisher | segment_publisher.cpp:65   | HLS Publisher has started listening on 0.0.0.0:8080...
[03-27 19:59:13.273] I 10996 Publisher | publisher.cpp:15   | HLS Publisher has been started.
[03-27 19:59:13.275] I 10996 OvenMediaEngine | main.cpp:162  | Trying to create a module MPEG-DASH Publisher for [default] host...
[03-27 19:59:13.281] I 10996 Publisher | segment_publisher.cpp:65   | DASH Publisher has started listening on 0.0.0.0:8080...
[03-27 19:59:13.281] I 10996 Publisher | publisher.cpp:15   | DASH Publisher has been started.
[03-27 19:59:13.282] I 10996 OvenMediaEngine | main.cpp:163  | Trying to create a module Low-Latency MPEG-DASH Publisher for [default] host...
[03-27 19:59:13.289] I 10996 Publisher | segment_publisher.cpp:65   | LLDASH Publisher has started listening on 0.0.0.0:8080...
[03-27 19:59:13.289] I 10996 Publisher | publisher.cpp:15   | LLDASH Publisher has been started.
[03-27 19:59:13.291] I 10996 OvenMediaEngine | main.cpp:164  | Trying to create a module OVT Publisher for [default] host...
[03-27 19:59:13.294] I 10996 OVT | ovt_publisher.cpp:49   | Ovt Publisher has started listening on 0.0.0.0:9000
[03-27 19:59:13.294] I 10996 Publisher | publisher.cpp:15   | OVTPublisher has been started.
[03-27 19:59:13.294] I 10996 OvenMediaEngine | main.cpp:169  | All modules are initialized successfully
[03-27 19:59:13.294] I 10996 Orchestrator | orchestrator.cpp:856  | Trying to create an application: [#default#app]
[03-27 19:59:13.294] I 10996 Monitor | host_metrics.cpp:52   | Create ApplicationMetrics(#default#app) for monitoring
[03-27 19:59:13.297] I 10996 Provider | application.cpp:30   | [#default#app] RTMP Provider application has been started
[03-27 19:59:13.297] I 10996 Provider | application.cpp:30   | [#default#app] OVT Provider application has been started
[03-27 19:59:13.297] I 10996 Provider | application.cpp:30   | [#default#app] RTSP Pull Provider application has been started
[03-27 19:59:13.297] I 10996 Provider | application.cpp:30   | [#default#app] RTSP Provider application has been started
[03-27 19:59:13.298] I 10996 TranscodeApplication | transcode_application.cpp:36   | [#default#app] Transcoder Application has been started
[03-27 19:59:13.300] I 10996 Publisher | application.cpp:26   | [#default#app] WebRTC Publisher application has been started
[03-27 19:59:13.302] I 10996 Publisher | application.cpp:26   | [#default#app] HLS Publisher application has been started
[03-27 19:59:13.304] I 10996 Publisher | application.cpp:26   | [#default#app] DASH Publisher application has been started
[03-27 19:59:13.305] I 10996 Publisher | application.cpp:26   | [#default#app] LLDASH Publisher application has been started
[03-27 19:59:13.307] I 10996 Publisher | application.cpp:26   | [#default#app] OVT Publisher application has been started
[03-27 19:59:14.706] I 11002 RtmpProvider | rtmp_server.cpp:126  | A RTMP client has connected from <ClientSocket: 0x7fffd4000b70, #24, state: 4, TCP, 192.168.0.200:11031>
[03-27 19:59:14.835] I 11002 RtmpProvider | rtmp_server.cpp:226  | [#default#app/stream] RTMP Provider stream has been created: id(0/0) device(OBS) remote(<ClientSocket: 0x7fffd4000b70, #24, state: 4, TCP, 192.168.0.200:11031>)
[03-27 19:59:14.835] I 11002 MediaRouter.App | media_route_application.cpp:184  | Trying to create a stream: [#default#app/stream(2921228900)]
[03-27 19:59:14.836] I 11002 Monitor | stream.cpp:240  | 
[Stream Info]
id(2921228900), name(stream), SourceType(Rtmp), Created Time (Fri Mar 27 19:59:14 2020)

        Video Track #0: Bypass(false) Bitrate(2.50Mb) codec(1, avc) resolution(1280x720) framerate(30.00fps) timebase(1/90000)
        Audio Track #1: Bypass(false) Bitrate(160.00Kb) codec(5, aac) samplerate(44.1K) format(s16, 16) channel(stereo, 2) timebase(1/44100)
[03-27 19:59:14.836] I 11002 Monitor | application_metrics.cpp:56   | Create StreamMetrics(stream) for monitoring
[03-27 19:59:14.836] I 11002 TranscodeStream | transcode_stream.cpp:353  | [#default#app/stream(2921228900)] -> [#default#app/stream_medium_o(3169746412)] Transcoder output stream has been created.
[03-27 19:59:14.839] I 11002 FFmpeg | third_parties.cpp:115  |  [AVCodecContext] using SAR=1/1
[03-27 19:59:14.841] I 11002 FFmpeg | third_parties.cpp:115  |  [AVCodecContext] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[03-27 19:59:14.846] I 11002 FFmpeg | third_parties.cpp:115  |  [AVCodecContext] profile Constrained Baseline, level 3.0, 4:2:0, 8-bit
[03-27 19:59:14.849] I 11002 FFmpeg | third_parties.cpp:115  |  [AVCodecContext] v1.7.0
[03-27 19:59:14.864] I 11048 MediaRouter.App | media_route_application.cpp:184  | Trying to create a stream: [#default#app/stream_medium_o(3169746412)]
[03-27 19:59:14.864] I 11002 TranscodeStream | transcode_stream.cpp:108  | [#default#app/stream(2921228900)] Transcoder input stream has been started. Status : (2) Decoders, (4) Encoders
[03-27 19:59:14.865] I 11048 Monitor | stream.cpp:240  | 
[Stream Info]
id(3169746412), name(stream_medium_o), SourceType(Transcoder), Created Time (Fri Mar 27 19:59:14 2020)
        >> Origin Stream Info
        id(2921228900), name(stream), SourceType(Rtmp), Created Time (Fri Mar 27 19:59:14 2020)

        Video Track #0: Bypass(false) Bitrate(700.00Kb) codec(2, vp8) resolution(640x360) framerate(30.00fps) timebase(1/90000)
        Video Track #1: Bypass(false) Bitrate(700.00Kb) codec(1, avc) resolution(640x360) framerate(30.00fps) timebase(1/90000)
        Audio Track #2: Bypass(false) Bitrate(48.00Kb) codec(7, opus) samplerate(48.0K) format(s16, 16) channel(stereo, 2) timebase(1/48000)
        Audio Track #3: Bypass(false) Bitrate(48.00Kb) codec(5, aac) samplerate(48.0K) format(s16, 16) channel(stereo, 2) timebase(1/48000)
[03-27 19:59:14.865] I 11048 Monitor | application_metrics.cpp:56   | Create StreamMetrics(stream_medium_o) for monitoring
[03-27 19:59:14.865] I 11048 WebRTC | rtc_stream.cpp:181  | Unsupported codec(Audio/AAC) is being input from media track
[03-27 19:59:14.880] I 11048 Publisher | stream.cpp:192  | [stream_medium_o(3169746412)] WebRTC Publisher stream has been started
[03-27 19:59:14.881] I 11048 Publisher | stream.cpp:192  | [stream_medium_o(3169746412)] HLS Publisher stream has been started
[03-27 19:59:14.881] I 11048 Publisher | stream.cpp:192  | [stream_medium_o(3169746412)] DASH Publisher stream has been started
[03-27 19:59:14.881] I 11048 Publisher | stream.cpp:192  | [stream_medium_o(3169746412)] LLDASH Publisher stream has been started
[03-27 19:59:14.897] I 11048 Publisher | stream.cpp:192  | [stream_medium_o(3169746412)] OVT Publisher stream has been started
[03-27 19:59:14.898] I 11048 TranscodeCodec | transcode_codec_dec_aac.cpp:49   | [#default#app/stream(2921228900)] input stream information: [audio] aac (LC), 44100 Hz, stereo, fltp, 154 kbps, timebase: 1/44100, frame_size: 1024
[03-27 19:59:14.985] I 11048 TranscodeCodec | transcode_codec_dec_avc.cpp:48   | [#default#app/stream(2921228900)] input stream information: [video] h264 (Constrained Baseline 3.1), yuv420p, 1280x720 [SAR 0:1 DAR 16:9], 30 fps, 195 kbps, timebase: 1/60, frame_size: 0

Clustering

OvenMediaEngine supports clustering and ensures high availability (HA) and scalability. For this we provide the OriginMap and OriginMapStore features. OriginMap is a method of configuring Origin server information in each Edge server, and OriginMapStore is a method for Origin servers and Edge servers to dynamically share information through Redis Server.

OriginMap

The OvenMediaEngine running as edge pulls a stream from an external server when a user requests it. The external server could be another OvenMediaEngine with OVT enabled or another stream server that supports RTSP.

The OVT is a protocol defined by OvenMediaEngine to relay stream between Origin-Edge and OVT can be run over SRT and TCP. For more information on the SRT Protocol, please visit the site.

Origin

OvenMediaEngine provides OVT protocol for passing streams from the origin to the edge. To run OvenMediaEngine as Origin, OVT port, and OVT Publisher must be enabled as follows :

Edge

The role of the edge is to receive and distribute streams from an origin. You can configure hundreds of Edge to distribute traffic to your players. As a result of testing, a single edge can stream 4-5Gbps traffic by WebRTC based on AWS C5.2XLarge. If you need to stream to thousands of people, you can configure and use multiple edges.

The edge supports OVT and RTSP to pull stream from an origin. In the near future, we will support more protocols. The stream pulled through OVT is bypassed without being encoded.

To run OvenMediaEngine as Edge, you need to add Origins elements to the configuration file as follows:

The <Origin>is a rule about where to pull a stream from for what request.

The <Origin>has the ability to automatically create an application with that name if the application you set in <Location> doesn't exist on the server. If an application exists in the system, a stream will be created in the application.

<Properties>

NoInputFailoverTimeout (default 3000)

NoInputFailoverTimeout is the time (in milliseconds) to switch to the next URL if there is no input for the set time.

UnusedStreamDeletionTimeout (default 60000)

UnusedStreamDeletionTimeout is a function that deletes a stream created with OriginMap if there is no viewer for a set amount of time (milliseconds). This helps to save network traffic and system resources for Origin and Edge.

<Origin>

For a detailed description of Origin's elements, see:

Location

Origin is already filtered by domain because it belongs to VirtualHost. Therefore, in Location, set App, Stream, and File to match except domain area. If a request matches multiple Origins, the top of them runs.

Pass

Pass consists of Scheme and Url.

<Scheme> is the protocol that will use to pull from the Origin Stream. It currently can be configured as OVTor RTSP.

If the origin server is OvenMediaEngine, you have to set OVTinto the <Scheme>.

You can pull the stream from the RTSP server by setting RTSPinto the<Scheme>. In this case, the <RTSPPull> provider must be enabled. The application automatically generated by Origin doesn't need to worry because all providers are enabled.

Urls is the address of origin stream and can consist of multiple URLs.

ForwardQueryParams is an option to determine whether to pass the query string part to the server at the URL you requested to play.(Default : true) Some RTSP servers classify streams according to query strings, so you may want this option to be set to false. For example, if a user requests ws://host:port/app/stream?transport=tcp to play WebRTC, the ?transport=tcp may also be forwarded to the RTSP server, so the stream may not be found on the RTSP server. On the other hand, OVT does not affect anything, so you can use it as the default setting.

Rules for generating Origin URL

The final address to be requested by OvenMediaEngine is generated by combining the configured Url and user's request except for Location. For example, if the following is set

If a user requests , OvenMediaEngine makes an address to ovt: //origin.com: 9000/origin_app/stream.

OriginMapStore

OriginMapStore is designed to make it easier to support autoscaling within a cluster. All Origin Servers and Edge Servers in the cluster share stream information and origin OVT URLs through Redis. That is, when a stream is created on the Origin server, the Origin server sets the app/stream name and OVT url to access the stream to the Redis server. Edge gets the OVT url corresponding to the app/stream from the Redis server when the user's playback request comes in.

This means that existing settings do not need to be updated when extending Origin servers and Edge servers. Therefore, all Origins can be grouped into one domain, and all Edges can be bundled with one domain. OriginMapStore allows you to expand Origins or Edges within a cluster without any additional configuration.

OriginMapStore functionality has been tested with Redis Server 5.0.7. You can enable this feature by adding the following settings to Server.xml of Origin and Edge. Note that must be set in Server.xml of the Origin server. This is used when Origin registers its own OVT url, so you just need to set a domain name or IP address that can be accessed as an OVT publisher.

Dynamic Application

It is either impossible or very cumbersome for edge servers to pre-configure all applications. So OriginMap and OriginMapStore have the ability to dynamically create an application if the application does not exist when creating the stream. They create a new application by copying the application configuration with <Name>*</Name>. That is, the special application with the name * is a dynamic application template.

Load Balancer

When you are configuring Load Balancer, you need to use third-party solutions such as L4 Switch, LVS, or GSLB, but we recommend using DNS Round Robin. Also, services such as cloud-based , , or can be a good alternative.

Low-Latency HLS

Apple supports Low-Latency HLS (LLHLS), which enables low-latency video streaming while maintaining scalability. LLHLS enables broadcasting with an end-to-end latency of about 2 to 5 seconds. OvenMediaEngine officially supports LLHLS as of v0.14.0.

LLHLS is an extension of HLS, so legacy HLS players can play LLHLS streams. However, the legacy HLS player plays the stream without using the low-latency function.

Title

Descriptions

Delivery

HTTP/1.1 HTTP/2

Security

Configuration

To use LLHLS, you need to add the <LLHLS> elements to the <Publishers> in the configuration as shown in the following example.

Element

Decscription

HTTP/2 outperforms HTTP/1.1, especially with LLHLS. Since all current browsers only support h2, HTTP/2 is supported only on TLS port. Therefore, it is highly recommended to use LLHLS on the TLS port.

Adaptive Bitrates Streaming (ABR)

LLHLS can deliver adaptive bitrate streaming. OME encodes the same source with multiple renditions and delivers it to the players. And LLHLS Player, including OvenPlayer, selects the best quality rendition according to its network environment. Of course, these players also provide option for users to manually select rendition.

See the section for how to configure renditions.

CrossDomain

Most browsers and players prohibit accessing other domain resources in the currently running domain. You can control this situation through Cross-Origin Resource Sharing (CORS) or Cross-Domain (CrossDomain). You can set CORS and Cross-Domain as <CrossDomains> element.

You can set it using the <Url> element as shown above, and you can use the following values:

Url Value

Description

Streaming

LLHLS is ready when a live source is inputted and a stream is created. Viewers can stream using OvenPlayer or other players.

If your input stream is already h.264/aac, you can use the input stream as is like below. If not, or if you want to change the encoding quality, you can do .

When you create a stream, as shown above, you can play LLHLS with the following URL:

http[s]://domain[:port]/<app name>/<stream name>/llhls.m3u8

If you use the default configuration, you can start streaming with the following URL:

https://domain:3334/app/<stream name>/llhls.m3u8

We have prepared a test player that you can quickly see if OvenMediaEngine is working. Please refer to the for more information.

Live Rewind

You can create as long a playlist as you want by setting <DVR> to the LLHLS publisher as shown below. This allows the player to rewind the live stream and play older segments. OvenMediaEngine stores and uses old segments in a file in <DVR><TempStoragePath> to prevent excessive memory usage. It stores as much as <DVR><MaxDuration> and the unit is seconds.

ID3v2 Timed Metadata

ID3 Timed metadata can be sent to the LLHLS stream through the .

Dump

You can dump the LLHLS stream for VoD. You can enable it by setting the following in <Application><Publishers><LLHLS>. Dump function can also be controlled by .

TargetStreamName

The name of the stream to dump to. You can use * and ? to filter stream names.

Playlists

The name of the master playlist file to be dumped together.

OutputPath

The folder to output to. In the OutputPath you can use the macros shown in the table below. You must have write permission on the specified folder.

Macro

Description

DRM (beta)

OvenMediaEngine supports Widevine and Fairplay in LLHLS with simple setup since version 0.16.0.

Currently, DRM is only supported for H.264 and AAC codecs. Support for H.265 will be added soon.

To include DRM information in your LLHLS Publisher configuration, follow these steps. You can set the InfoFile path as either a relative path, starting from the directory where Server.xml is located, or as an absolute path.

The separation of the DRMInfoFile is designed to allow dynamic changes to the file. Any modifications to the DRMInfoFile will take effect when a new stream is generated.

Here's how you should structure your DRM Info File:

Multiple <DRM> can be set. Specify the VirtualHost, Application, and StreamName where DRM should be applied. StreamName supports wildcard regular expressions.

Currently, CencProtectScheme only supports "cbcs" since FairPlay also supports only cbcs. There may be limited prospects for adding other schemes in the near future.

KeyId, Key, Iv and Pssh values are essential and should be provided by your DRM provider. FairPlayKeyUrl is only need for FairPlay and if you want to enable FairPlay to your stream, it is required. It will be also provided by your DRM provider.

OvenPlayer now includes DRM-related options. Enable DRM and input the License URL. Your content is now securely protected.

Pallycon DRM

OvenMediaEngine integrates with , allowing you to more easily apply DRM to LLHLS streams.

To integrate Pallycon, configure the DRMInfo.xml file as follows.

Set DRMProviderto Pallycon. Then, set the necessary information as shown in the example. KMSUrl and KMSToken are values provided by the Pallycon console. ContentId can be created using VHostName, AppName, and StreamName macros.

Performance Tuning

Performance Test

OvenMediaEngine provides a tester for measuring WebRTC performance called OvenRtcTester. It is developed in Go language and uses the pion/webrtc/v3 and gorilla/websocket modules. Many thanks to the pion/webrtc and gorilla/websocket teams for contributing this wonderful project.

Getting Started OvenRtcTester

Install GO

Since OvenRtcTester is developed in Go language, Go must be installed on your system. Install Go from the following URL:

OvenRtcTester was tested with the latest version of go 1.17.

Run

You can simply run it like this: -url is required. If the -life option is not used, it will run indefinitely until the user presses ctrl+c.

You can also use go build or go install depending on your preference.

OvenRtcTester must test OvenMediaEngine 0.12.4 or higher as the target system. OvenMediaEngine versions below 0.12.4 have a problem with incorrectly calculating the RTP timestamp, so OvenRtcTester calculates the Video Delay value incorrectly.

Performance Tuning

Monitoring the usage of threads

Linux has various tools to monitor CPU usage per thread. We will check the simplest with the top command. If you issue the top -H -p [pid] command, you will see the following screen.

You can use OvenRtcTester to test the capacity of the server as shown below. When testing the maximum performance, OvenRtcTester also uses a lot of system resources, so test it separately from the system where OvenMediaEngine is running. Also, it is recommended to test OvenRtcTester with multiple servers. For example, simulate 500 players with -n 500 on one OvenRtcTester, and simulate 2000 players with four servers.

Building and running OvenMediaEngine in debug mode results in very poor performance. Be sure to test the maximum performance using the binary generated by make release && make install .

If the OvenMediaEngine's capacity is exceeded, you will notice it in OvenRtcTester's Summary report with Avg Video Delay and Avg Audio Delay or Packet loss.

On the right side of the above capture screen, we simulate 400 players with OvenRtcTester. <Summary> of OvenRtcTester shows that Avg Video Delay and Avg Audio Delay are very high, and Avg FPS is low.

And on the left, you can check the CPU usage by thread with the top -H -p command. This confirms that the StreamWorker threads are being used at 100%, and now you can scale the server by increasing the number of StreamWorker threads. If OvenMediaEngine is not using 100% of all cores of the server, you can improve performance by .

This is the result of tuning the number of StreamWorkerCount to 8 in config. This time, we simulated 1000 players with OvenRtcTester, and you can see that it works stably.

Tuning the number of threads

The WorkerCount in <Bind> can set the thread responsible for sending and receiving over the socket. Publisher's AppWorkerCount allows you to set the number of threads used for per-stream processing such as RTP packaging, and StreamWorkerCount allows you to set the number of threads for per-session processing such as SRTP encryption.

Scalable Threads and Configuration

AppWorkerCount

With AppWorkerCount, you can set the number of threads for distributed processing of streams when hundreds of streams are created in one application. When an application is requested to create a stream, the stream is evenly attached to one of created threads. The main role of Stream is to packetize raw media packets into the media format of the protocol to be transmitted. When there are thousands of streams, it is difficult to process them in one thread. Also, if StreamWorkerCount is set to 0, AppWorkerCount is responsible for sending media packets to the session.

It is recommended that this value does not exceed the number of CPU cores.

StreamWorkerCount

It may be impossible to send data to thousands of viewers in one thread. StreamWorkerCount allows sessions to be distributed across multiple threads and transmitted simultaneously. This means that resources required for SRTP encryption of WebRTC or TLS encryption of HLS/DASH can be distributed and processed by multiple threads. It is recommended that this value not exceed the number of CPU cores.

Use-Case

If a large number of streams are created and very few viewers connect to each stream, increase AppWorkerCount and lower StreamWorkerCount as follows.

If a small number of streams are created and a very large number of viewers are connected to each stream, lower AppWorkerCount and increase StreamWorkerCount as follows.

Troubleshooting

We will update this document as we gather troubleshooting examples. (Written in Nov 04, 2021)

`prerequisites.sh` Script Failed

If you have problems with the prerequisites.sh the script we have provided, please install it manually as follows.

Platform Specific Installation

Common Installation

`systemctl start ovenmediaengine` failed

Check SELinux

If SELinux is running on your system, SELinux can deny the execution of OvenMediaEngine.

You can choose between two methods of adding a policy to SELinux or setting SELinux to permissive mode. To add a policy, you must apply the SELinux policy file for the OvenMediaEngine service to your system as follows:

Setting SELinux to permissive mode is as simple as follows. But we don't recommend this method.

Streaming is not smooth

1. If you are using Transcoding as Bypass in OvenMediaEngine, and streaming does not work in all players

WebRTC does not support b-frame of H.264. But if your encoder sends b-frames the video will be stuttered in the player. In this case, you can solve the problem by disabling the b-frame function in your encoder. For OBS, you can set bframes=0 option as below.

Or by activating the encoding options in OvenMediaEngine.

Setting up Transcoding options in OvenMediaEngine:

2. When streaming does not work in some players

In this case, you are probably trying to stream with UDP in an environment where packet loss is high due to network performance, connection problems, etc., the interruption during stream playback may more and more worsen. This problem can be solved simply by playing with WebRTC/TCP.

If you want to monitor packet loss in your Chrome browser, you can access it by typing 'chrome://webrtc-internals' in the address bar.

Setting up WebRTC over TCP in OvenMediaEngine:

Also, if the device's network speed, which is running the player, isn't fast enough to accommodate the stream's BPS, the stuttering during streaming won't resolve and will eventually drop the connection. In this case, there is no other way than to speed up your network.

3. When streaming fails due to excessive CPU/Memory/Network usage of Origin in OvenMediaEngine

If the Origin server uses excessive CPU/Memory/Network, all players may experience stuttering during streaming.

When you see Origin is CPU intensive on your Origin-Edge structure, the transcoding options in the OvenMediaEngine may be the primary cause. That is, you may have set the quality of the input stream too high, or the output stream to exceed the capabilities of your hardware significantly. In this case, it can be solved by enabling the hardware encoder in OvenMediaEngine.

Setting up GPU Acceleration in OvenMediaEngine:

4. When streaming fails due to excessive CPU/Memory/Network usage of Edge in OvenMediaEngine

If the edge server excessively uses CPU/Memory/Network, the player connected to that Edge may experience stuttering during streaming. In this case, it can be solved by expanding Edge.

5. If you have enough CPU/Memory/Network, but streaming is not smooth

5-1. When a specific thread is using the CPU excessively

When you see a specific thread overuses the CPU, the video may not stream smoothly. Please refer to the manual below for more information on this.

Tuning OvenMediaEngine Performance:

5-2. Tuning your Linux kernel

The Linux kernel, which is set by default, cannot handle 1Gbps output, so put it as follows:

5-3. Congestion Control: Change to BBR

The mobile environment used by many people uses a wireless network. It has a high network speed but, conversely, can cause high packet loss.

Look, CUBIC, the Congestion Control set by default in your Linux, adjusts the TCP Window by packet loss, so it is not suitable to provide stable streaming in such an environment.

So our suggestion is to use Google's BBR. This setting is even more important if you mainly provide WebRTC services to mobile users who use a wireless network. Change the Congestion Control from CUBIC to BBR on your Linux.

Player connection fails

1. Due to the Mixed Contents

If you try to access OvenMediaEngine's WebRTC URL starting with ws:// (Non-TLS) from an HTTPS (HTTP/TLS) site, the connection may be rejected due to a mixed content problem depending on the browser.

In this case, you can solve this by installing a certificate in OvenMediaEngine and trying to connect with the wss:// (WebSocket/TLS) URL.

Setting up TLS Encryption in OvenMediaEngine:

As of October 2021, most browsers have enforced the , and CORS errors often occur when requesting access to other domains if it is not a TLS site. In this case, you can solve the problem by installing a certificate on the site that loads the player.

3. When the message "Too many open files" appears in the log, the player cannot connect

At some point, when the message "Too many open files" is output in your OvenMediaEngine log, it may not be able to handle any more player connections. In this case, you can solve the problem by setting it as follows:

Player takes a long time first to load while trying to stream

1. Due to the Keyframe Interval

If you use Transcoding as Bypass in OvenMediaEngine and set a long keyframe interval in the encoder, the WebRTC player cannot start streaming until a keyframe is an input.

In this case, you can solve this by setting the keyframe interval in the encoder to 1-2 seconds,

Or by enabling the encoding options in OvenMediaEngine.

Setting up Transcoding options in OvenMediaEngine:

A/V is out of sync

1. When the A/V sync does not match during initial streaming, and it gradually fits

A/V may not be input evenly from the encoder. There are some encoders with policies for reliable streaming that they decide, for example, sending audio first and video very later, or video first and audio very late.

OvenMediaEngine outputs the input received from the encoder as-is for sub-second latency streaming. The WebRTC player also streams the received input as-is, so the A/V sync may not match during the initial playback due to the policy of specific encoders.

However, this can be resolved naturally as the player will sync A/V while streaming based on Timestamp. Still, if this work looks like an error, you can also solve it by enabling JitterBuffer in OvenMediaEngine.

Also, suppose you are using a transcoder in OvenMediaEngine and trying to input with b-frames of H264. Audio is encoded fast, but a video is buffered at the decoder because of b-frames. Therefore, there is a time difference at the start of each encoding, which may cause the A/V to be out of sync. Even in this case, enabling JitterBuffer will solve this problem.

Setting up WebRTC JitterBuffer in OvenMediaEngine:

2. Time has passed, but A/V is out of sync

There may be cases where the A/V sync is not corrected even after a certain amount of time has elapsed after playback. This problem is caused by small internal buffers in some browsers such as Firefox, which causes the player to give up calibration if the A/V sync differs too much. But this can also be solved by enabling JitterBuffer.

Setting up WebRTC JitterBuffer in OvenMediaEngine:

Nevertheless, if the A/V sync is not corrected, you should suspect an error in the original video file, which can be checked by playing as HLS.

However, if A/V sync is well during streaming with HLS, this is OvenMediaEnigne's bug. If you find any bugs, please feel free to report them to .

No audio is output

1. When Opus is not set in the encoding options

WebRTC supports Opus, not AAC, as an audio codec. Because RTMP and other protocols mainly use and transmit AAC as the audio codec, you may not have set up Opus, but WebRTC cannot output audio without Opus. This can be solved by setting Opus in OvenMediaEnigne.

Setting up Opus Codec in OvenMediaEngine:

Not the video quality you want

1. When video encoding is enabled in OvenMediaEngine

If you are using video encoding in OME, the video bitrate may be set low. In this case, the video quality can be improved by increasing the unit of video bitrate.

However, since OvenMediaEngine has the default to the fastest encoding option for sub-second latency streaming, the video quality may not be as good as the set video bitrate. In this case, OvenMediaEngine provides an output profile preset that can control the quality, so you can choose to solve it.

Choosing an Encoding Preset in OvenMediaEngine:

2. If you are using Transcoding as Bypass in OvenMediaEngine

Since the encoder is transmitting video to OvenMediaEngine in low quality, you can solve it by increasing the input quality in the encoder settings.

WebRTC Streaming

OvenMediaEngine uses WebRTC to provide sub-second latency streaming. WebRTC uses RTP for media transmission and provides various extensions.

OvenMediaEngine provides the following features:

Title

Functions

Delivery

RTP / RTCP

Security

DTLS, SRTP

Configuration

If you want to use the WebRTC feature, you need to add <WebRTC> element to the <Publishers> and <Ports> in the Server.xml configuration file, as shown in the example below.

ICE

WebRTC uses ICE for connections and specifically NAT traversal. The web browser or player exchanges the Ice Candidate with each other in the Signalling phase. Therefore, OvenMediaEngine provides an ICE for WebRTC connectivity.

If you set IceCandidate to *: 10000-10005/udp, as in the example above, OvenMediaEngine automatically gets IP from the server and generates IceCandidate using UDP ports from 10000 to 10005. If you want to use a specific IP as IceCandidate, specify a specific IP. You can also use only one 10000 UDP Port, not a range, by setting it to *: 10000.

Signalling

OvenMediaEngine has embedded a WebSocket-based signalling server and provides our defined signalling protocol. Also, OvenPlayer supports our signalling protocol. WebRTC requires signalling to exchange Offer SDP and Answer SDP, but this part isn't standardized. If you want to use SDP, you need to create your exchange protocol yourself.

If you want to change the signaling port, change the value of <Ports><WebRTC><Signalling>.

Signalling Protocol

The Signalling protocol is defined in a simple way:

If you want to use a player other than OvenPlayer, you need to develop the signalling protocol as shown above and can integrate OvenMediaEngine.

Streaming

Publisher

Add WebRTC element to Publisher to provide streaming through WebRTC.

Option

Description

Default

WebRTC Publisher's <JitterBuffer> is a function that evenly outputs A/V (interleave) and is useful when A/V synchronization is no longer possible in the browser (player) as follows.

If the A/V sync is excessively out of sync, some browsers may not be able to handle this or it may take several seconds to synchronize.

Encoding

WebRTC Streaming starts when a live source is inputted and a stream is created. Viewers can stream using OvenPlayer or players that have developed or applied the OvenMediaEngine Signalling protocol.

Also, the codecs supported by each browser are different, so you need to set the Transcoding profile according to the browser you want to support. For example, Safari for iOS supports H.264 but not VP8. If you want to support all browsers, please set up VP8, H.264, and Opus codecs in all transcoders.

WebRTC doesn't support AAC, so when trying to bypass transcoding RTMP input, audio must be encoded as opus. See the settings below.

Some browsers support both H.264 and VP8 to send Answer SDP to OvenMediaEngine, but sometimes H.264 can't be played. In this situation, if you write the VP8 above the H.264 code line in the Transcoding profile setting, you can increase the priority of the VP8.

Using this manner so that some browsers, support H.264 but can't be played, can stream smoothly using VP8. This means that you can solve most problems with this method.

Playback

If you created a stream as shown in the table above, you can play WebRTC on OvenPlayer via the following URL:

Protocol

URL format

If you use the default configuration, you can stream to the following URL:

ws://[OvenMediaEngine IP]:3333/app/stream
wss://[OvenMediaEngine IP]:3333/app/stream

We have prepared a test player to make it easy to check if OvenMediaEngine is working. Please see the chapter for more information.

Adaptive Bitrates Streaming (ABR)

OvenMediaEnigne provides adaptive bitrates streaming over WebRTC. OvenPlayer can also play and display OvenMediaEngine's WebRTC ABR URL.

Create Playlist for WebRTC ABR

You can provide ABR by creating a playlist in <OutputProfile> as shown below. The URL to play the playlist is ws[s]://domain[:port]/<app name>/<stream name>/<playlist file name>

<Playlist><Rendition><Video> and <Playlist><Rendition><Audio> can connected using <Encodes><Video><Name> or <Encodes><Audio><Name>.

It is not recommended to use a <Bypass>true</Bypass> encode item if you want a seamless transition between renditions because there is a time difference between the transcoded track and bypassed track.

If <Options><WebRtcAutoAbr> is set to true, OvenMediaEngine will measure the bandwidth of the player session and automatically switch to the appropriate rendition.

Here is an example play URL for ABR in the playlist settings below. wss://domain:13334/app/stream/abr

Streaming starts from the top rendition of Playlist, and when Auto ABR is true, the server finds the best rendition and switches to it. Alternatively, the user can switch manually by selecting a rendition in the player.

See the section for more details on how to configure renditions.

Multiple codec support in Playlist

WebRTC can negotiate codecs with SDP to support more devices. Playlist can set rendition with different kinds of codec. And OvenMediaEngine includes only renditions corresponding to the negotiated codec in the playlist and provides it to the player.

If an unsupported codec is included in the Rendition, the Rendition is not used. For example, if the Rendition's Audio contains aac, WebRTC ignores the Rendition.

In the example below, it consists of renditions with H.264 and Opus codecs set and renditions with VP8 and Opus codecs set. If the player selects VP8 in the answer SDP, OvenMediaEngine creates a playlist with only renditions containing VP8 and Opus and passes it to the player.

WebRTC over TCP

There are environments where the network speed is fast but UDP packet loss is abnormally high. In such an environment, WebRTC may not play normally. WebRTC does not support streaming using TCP, but connections to the TURN () server support TCP. Based on these characteristics of WebRTC, OvenMediaEngine supports TCP connections from the player to OvenMediaEngine by embedding a TURN server.

Turn on TURN server

You can turn on the TURN server by setting <TcpRelay> in the WebRTC Bind.

Example : <TcpRelay>*:3478</TcpRelay>

OME may sometimes not be able to get the server's public IP to its local interface. (Environment like Docker or AWS) So, specify the public IP for Relay IP. If * is used, the public IP obtained from and all IPs obtained from the local interface are used. Port is the tcp port on which the TURN server is listening.

If * is used as the IP of TcpRelay and IceCandidate, all available candidates are generated and sent to the player, so the player tries to connect to all candidates until a connection is established. This can cause delay in initial playback. Therefore, specifying the ${PublicIP} macro or IP directly may be more beneficial to quality.

WebRTC over TCP with OvenPlayer

WebRTC players can configure the TURN server through the setting.

You can play the WebRTC stream over TCP by attaching the query transport=tcp to the existing WebRTC play URL as follows.

OvenPlayer automatically sets iceServers by obtaining TURN server information set in <TcpRelay> through signaling with OvenMediaEngine.

If <TcpForce> is set to true, it will force a TCP connection even if ?transport=tcp is not present. To use this, <TcpRelay> must be set.

Custom player

If you are using custom player, set iceServers like this:

When sending Request Offer in the phase with OvenMediaEngine, if you send the transport=tcp query string, ice_servers information is delivered as follows. You can use this information to set iceServers.

ABR and Transcoding

OvenMediaEngine has a built-in live transcoder. The live transcoder can decode the incoming live source and re-encode it with the set codec or adjust the quality to encode at multiple bitrates.

Supported Video, Audio and Image Codecs

Decoders

Type

Codec

Encoders

Type

Codec

Codec of Configuration

OutputProfiles

OutputProfile

The <OutputProfile> setting allows incoming streams to be re-encoded via the <Encodes> setting to create a new output stream. The name of the new output stream is determined by the rules set in <OutputStreamName>, and the newly created stream can be used according to the streaming URL format.

According to the above setting, if the incoming stream name is stream, the output stream becomes stream_bypassand the stream URL can be used as follows.

WebRTC ws://192.168.0.1:3333/app/stream_bypass
LLHLS http://192.168.0.1:8080/app/stream_bypass/llhls.m3u8

Encodes

Video

You can set the video profile as below:

The meaning of each property is as follows:

Property

Description

* required

Table of presets

A table in which presets provided for each codec library are mapped to OvenMediaEngine presets. Slow presets are of good quality and use a lot of resources, whereas Fast presets have lower quality and better performance. It can be set according to your own system environment and service purpose.

Presets

openh264

h264_nvenc

h264_qsv

vp8

References

https://trac.ffmpeg.org/wiki/Encode/VP8
https://docs.nvidia.com/video-technologies/video-codec-sdk/nvenc-preset-migration-guide/

Audio

You can set the audio profile as below:

The meaning of each property is as follows:

Property

Description

* required

It is possible to have an audio only output profile by specifying the Audio profile and omitting a Video one.

Image

You can set the Image profile as below:

The meaning of each property is as follows:

Property

Description

The image encoding profile is only used by thumbnail publishers. and, bypass option is not supported.

Bypass without transcoding

You can configure Video and Audio to bypass transcoding as follows:

You need to consider codec compatibility with some browsers. For example, chrome only supports OPUS codec for audio to play WebRTC stream. If you set to bypass incoming audio, it can't play on chrome.

WebRTC doesn't support AAC, so if video bypasses transcoding, audio must be encoded in OPUS.

Conditional transcoding

If the codec or quality of the input stream is the same as the profile to be encoded into the output stream. there is no need to perform re-transcoding while unnecessarily consuming a lot of system resources. If the quality of the input track matches all the conditions of BypassIfMatch, it will be Pass-through without encoding

Matching elements in video

Elements

Condition

Description

* eq: equal to / lte: less than or equal to / gte: greater than or equal to

Matching elements in audio

Elements

Condition

Description

* eq: equal to / lte: less than or equal to / gte: greater than or equal to

To support WebRTC and LLHLS, AAC and Opus codecs must be supported at the same time. Use the settings below to reduce unnecessary audio encoding.

If a video track with a lower quality than the encoding option is input, unnecessary upscaling can be prevented. SAR (Storage Aspect Ratio) is the ratio of original pixels. In the example below, even if the width and height of the original video are smaller than or equal to the width and height set in the encoding option, if the ratio is different, it means that encoding is performed without bypassing.

Keep the original with transcoding

If you want to transcode with the same quality as the original. See the sample below for possible parameters that OME supports to keep original. If you remove the Width, Height, Framerate, Samplerate, and Channel parameters. then, It is transcoded with the same options as the original.

Rescaling while keep the aspect ratio

To change the video resolution when transcoding, use the values of width and height in the Video encode option. If you don't know the resolution of the original, it will be difficult to keep the aspect ratio after transcoding. Please use the following methods to solve these problems. For example, if you input only the Width value in the Video encoding option, the Height value is automatically generated according to the ratio of the original video.

Adaptive Bitrate Streaming (ABR)

From version 0.14.0, OvenMediaEngine can encode same source with multiple bitrates renditions and deliver it to the player.

As shown in the example configuration below, you can provide ABR by adding <Playlists> to <OutputProfile>. There can be multiple playlists, and each playlist can be accessed with <FileName>.

The method to access the playlist set through LLHLS is as follows.

http[s]://<domain>[:port]/<app>/<stream>/<FileName>.m3u8

The method to access the Playlist set through WebRTC is as follows.

ws[s]://<domain>[:port]/<app>/<stream>/<FileName>

Note that <FileName> must never contain the playlist and chunklist keywords. This is a reserved word used inside the system.

To set up <Rendition>, you need to add <Name> to the elements of <Encodes>. Connect the set <Name> into <Rendition><Video> or <Rendition><Audio>.

In the example below, three quality renditions are provided and the URL to play the abr playlist as LLHLS is https://domain:port/app/stream/abr.m3u8 and The WebRTC playback URL is wss://domain:port/app/stream/abr

Supported codecs by streaming protocol

Even if you set up multiple codecs, there is a codec that matches each streaming protocol supported by OME, so it can automatically select and stream codecs that match the protocol. However, if you don't set a codec that matches the streaming protocol you want to use, it won't be streamed.

The following is a list of codecs that match each streaming protocol:

Protocol

Supported Codec

Therefore, you set it up as shown in the table. If you want to stream using LLHLS, you need to set up H.264 and AAC, and if you want to stream using WebRTC, you need to set up Opus.

Also, if you are going to use WebRTC on all platforms, you need to configure both VP8 and H.264. This is because different codecs are supported for each browser, for example, VP8 only, H264 only, or both.

However, don't worry. If you set the codecs correctly, OME automatically sends the stream of codecs requested by the browser.

Configuration

OvenMediaEngine has an XML configuration file. If you start OvenMediaEngine with systemctl start ovenmediaengine, the config file is loaded from the following path.

If you run it directly from the command line, it loads the configuration file from:

If you run it in Docker container, the path to the configuration file is:

Server

The Server is the root element of the configuration file. The versionattribute indicates the version of the configuration file. OvenMediaEngine uses this version information to check if the config file is a compatible version.

IP

The IP address is OvenMediaEngine will bind to. If you set *, all IP addresses of the system are used. If you enter a specific IP, the Host uses that IP only.

PrivacyProtection

PrivacyProtection is an option to comply with GDPR, PIPEDA, CCPA, LGPD, etc. by deleting the client's personal information (IP, Port) from all records. When this option is turned on, the client's IP and Port are converted to xxx.xxx.xxx.xxx:xxx in all logs and REST APIs.

StunServer

OvenMediaEngine needs to know its public IP in order to connect to the player through WebRTC. The server must inform the player of the IceCandidates and TURN server addresses when signaling, and this information must be the IP the player can connect to. However, in environments such as Docker or AWS, public IP cannot be obtained through a local interface, so a method of obtaining public IP using stun server is provided (available from version 0.11.1).

If OvenMediaEngine obtains the public IP through communication with the set stun server, you can set the public IP by using * or ${PublicIP} in IceCandidate and TcpRelay.

Bind

The Bind is the configuration for the server port that will be used. Bind consists of Providers and Publishers. The Providers are the server for stream input, and the Publishers are the server for streaming.

The meaning of each element is shown in the following table:

Element

Description

Virtual Host

VirtualHosts are a way to run more than one streaming server on a single machine. OvenMediaEngine supports IP-based virtual host and Domain-based virtual host. "IP-based" means that you can separate streaming servers into multiples by setting different IP addresses, and "Domain-based" means that even if the streaming servers use the same IP address, you can split the streaming servers into multiples by setting different domain names.

VirtualHostsconsist of Name, Host, Origins, SignedPolicy, and Applications.

Host

The Domain has Names and TLS. Names can be either a domain address or an IP address. Setting * means it allows all domains and IP addresses.

SignedPolicy

SignedPolicy is a module that limits the user's privileges and time. For example, operators can distribute RTMP URLs that can be accessed for 60 seconds to authorized users, and limit RTMP transmission to 1 hour. The provided URL will be destroyed after 60 seconds, and transmission will automatically stop after 1 hour. Users who are provided with a SingedPolicy URL cannot access resources other than the provided URL. This is because the SignedPolicy URL is authenticated. See the chapter for more information.

Origins

Origins (also we called OriginMap) are a feature to pull streams from external servers. It now supports OVT and RTSP for the pulling protocols. OVT is a protocol defined by OvenMediaEngine for Origin-Edge communication. It allows OvenMediaEngine to relay a stream from other OvenMediaEngines that have OVP Publisher turned on. Using RTSP, OvenMediaEngine pulls a stream from an RTSP server and creates a stream. RTSP stream from external servers can stream by WebRTC, HLS, and MPEG-DASH.

The Origin has Location and Pass elements. Location is a URI pattern for incoming requests. If the incoming URL request matches Location, OvenMediaEngine pulls the stream according to a Pass element. In the Pass element, you can set the origin stream's protocol and URLs.

To run the Edge server, Origin creates application and stream if there isn't those when user request. For more learn about Origin-Edge, see the chapter.

Application

<Application> consists of various elements that can define the operation of the stream, including Stream input, Encoding, and Stream output. In other words, you can create as many <Application> as you like and build various streaming environments.

<Application> needs to set <Name> and <Type> as follows:

<Name> is used to configure the Streaming URL.
<Type> defines the operation of <Application>. Currently, there is only a live type.

OutputProfiles

<OutputProfile> is a configuration that creates an output stream. Output stream name can be set with <OutputStreamName>, and transcoding properties can be set through <Encodes>. If you want to stream one input to multiple output streams, you can set multiple <OutputProfile>.

For more information about the OutputProfiles, please see the chapter.

Providers

Providers ingest streams that come from a media source.

If you want to get more information about the <Providers>, please refer to the chapter.

Publishers

You can configure the Output Stream operation in <Publishers>. <ThreadCount> is the number of threads used by each component responsible for the <Publishers> protocol.

You need many threads to transmit streams to a large number of users at the same time. So it's better to use a higher core CPU and set <ThreadCount> equal to the number of CPU cores.

OvenMediaEngine currently supports WebRTC, Low-Latency DASH, MEPG-DASH, and HLS. If you don't want to use any protocol then you can delete that protocol setting, the component for that protocol isn't initialized. As a result, you can save system resources by deleting the settings of unused protocol components.

If you want to learn more about WebRTC, visit the chapter. And if you want to get more information on Low-Latency DASH, MPEG-DASH, and HLS, refer to the chapter on .

Configuration Example

Finally, Server.xml is configured as follows:

<?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>

    <Modules>
        <!-- 
        Currently OME only supports h2 like all browsers do. Therefore, HTTP/2 only works on TLS ports.			
        -->
        <HTTP2>
            <Enable>true</Enable>
        </HTTP2>

        <LLHLS>
            <Enable>true</Enable>
        </LLHLS>

        <!-- P2P works only in WebRTC and is experiment feature -->
        <P2P>
            <!-- disabled by default -->
            <Enable>false</Enable>
            <MaxClientPeersPerHostPeer>2</MaxClientPeersPerHostPeer>
        </P2P>
    </Modules>

<!-- 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>
                <TLSPort>3334</TLSPort>
                <WorkerCount>1</WorkerCount>
            </Signalling>

            <IceCandidates>
                <IceCandidate>*:10000/udp</IceCandidate>
                <!-- 
                    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>
                <TcpRelayWorkerCount>1</TcpRelayWorkerCount>
            </IceCandidates>
        </WebRTC>
    </Providers>

    <Publishers>
        <OVT>
            <Port>9000</Port>
            <WorkerCount>1</WorkerCount>
        </OVT>
        <LLHLS>
            <!-- 
            OME only supports h2, so LLHLS works over HTTP/1.1 on non-TLS ports. 
            LLHLS works with higher performance over HTTP/2, 
            so it is recommended to use a TLS port.
            -->
            <Port>3333</Port>
            <!-- If you want to use TLS, specify the TLS port -->
            <TLSPort>3334</TLSPort>
            <WorkerCount>1</WorkerCount>
        </LLHLS>
        <WebRTC>
            <Signalling>
                <Port>3333</Port>
                <TLSPort>3334</TLSPort>
                <WorkerCount>1</WorkerCount>
            </Signalling>
            <IceCandidates>
                <IceCandidate>*:10000-10005/udp</IceCandidate>
                <!-- 
                    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>
                <TcpRelayWorkerCount>1</TcpRelayWorkerCount>
            </IceCandidates>
        </WebRTC>
    </Publishers>
</Bind>

    <!--
        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>*</Name>
            </Names>
            <TLS>
                <CertPath>path/to/file.crt</CertPath>
                <KeyPath>path/to/file.key</KeyPath>
                <ChainCertPath>path/to/file.crt</ChainCertPath>
            </TLS>
        </Host>
        <API>
            <AccessToken>ome-access-token</AccessToken>

            <CrossDomains>
                <Url>*.airensoft.com</Url>
                <Url>http://*.sub-domain.airensoft.com</Url>
                <Url>http?://airensoft.*</Url>
            </CrossDomains>
        </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>*</Name>
                </Names>
                <!--
                <TLS>
                    <CertPath>path/to/file.crt</CertPath>
                    <KeyPath>path/to/file.key</KeyPath>
                    <ChainCertPath>path/to/file.crt</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,llhls,dash,lldash</Publishers>
                </Enables>
            </SignedPolicy>
            -->

            <!--
            <AdmissionWebhooks>
                <ControlServerUrl></ControlServerUrl>
                <SecretKey></SecretKey>
                <Timeout>3000</Timeout>
                <Enables>
                    <Providers>rtmp,webrtc,srt</Providers>
                    <Publishers>webrtc,hls,llhls,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>
                    <ForwardQueryParams>false</ForwardQueryParams>
                </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 connects 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>
                        <LLHLS>
                            <ChunkDuration>0.2</ChunkDuration>
                            <SegmentDuration>6</SegmentDuration>
                            <SegmentCount>10</SegmentCount>
                            <CrossDomains>
                                <Url>*</Url>
                            </CrossDomains>
                        </LLHLS>
                    </Publishers>
                </Application>
            </Applications>
        </VirtualHost>
    </VirtualHosts>
</Server>

0.16.5

Introduction

hashtagWhat is OvenMediaEngine?

hashtagFeatures

hashtagSupported Platforms

hashtagGetting Started

hashtagHow to Contribute

hashtagFor more information

hashtagLicense

Quick Start

hashtagRun OvenMediaEngine

hashtagRun OvenPlayer Demo

hashtagPublishing

hashtagPlayback

hashtagWebRTC Playback

hashtagLLHLS Playback

Online Demo

hashtagPlayer & WebRTC Encoder

Getting Started

hashtagGetting Started with Docker Image

hashtagGetting Started with Source Code

hashtagInstalling dependencies

hashtagBuilding & Running

hashtagPorts used by default

TLS Encryption

Live Source

RTMP

hashtagConfiguration

hashtagRTMP live stream

hashtagPublish

hashtagExample with OvenLiveKit (OvenStreamEncoder)

hashtagExample with OBS

WebRTC / WHIP

hashtagConfiguration

hashtag

MPEG-2 TS

hashtagConfiguration

RTSP Pull

hashtagPulling streams using the Stream Creation API

hashtagPulling streams using the OriginMapStore

hashtagPulling streams using the OriginMap

hashtagConfiguration

hashtagEvent to trigger pulling

Scheduled Channel

hashtagGetting Started

Multiplex Channel

hashtagConfiguration

Streaming

Low-Latency DASH and Legacy HLS streaming

Access Control

Thumbnail

hashtagConfiguration

hashtagBind

hashtagEncoding

hashtagPublisher

hashtagGet thumbnails

Push Publishing

hashtagConfiguration

v1

Reload Certificate

hashtagReload All Certificates

hashtagRequest

hashtagHeader

hashtagResponses

hashtagReload Certificate

hashtagRequest

hashtagHeader

hashtagResponses

Statistics

P2P Delivery (Experiment)

hashtagWhat are the benefits of using P2P Delivery?

hashtagHow does it work?

hashtagHow does it classify Peers?

Getting Started

hashtagGetting Started with Docker Image

hashtagGetting Started with Source Code

hashtagInstalling dependencies

hashtagBuilding & Running

hashtagPorts used by default

Quick Start

What is OvenMediaEngine?

Features

Supported Platforms

Getting Started

How to Contribute

For more information

License

Run OvenMediaEngine

Run OvenPlayer Demo

Publishing

Playback

WebRTC Playback

LLHLS Playback

Player & WebRTC Encoder

Getting Started with Docker Image

Getting Started with Source Code

Installing dependencies

Building & Running

Ports used by default

Configuration

RTMP live stream

Publish

Example with OvenLiveKit (OvenStreamEncoder)

Example with OBS

Configuration

Configuration

Pulling streams using the Stream Creation API

Pulling streams using the OriginMapStore

Pulling streams using the OriginMap

Configuration

Event to trigger pulling

Getting Started

Configuration

Configuration

Bind

Encoding

Publisher

Get thumbnails

Configuration

Reload All Certificates

Request

Header

Responses

Reload Certificate

Request

Header

Responses

What are the benefits of using P2P Delivery?

How does it work?

How does it classify Peers?

Getting Started with Docker Image

Getting Started with Source Code

Installing dependencies

Building & Running

Ports used by default

Run OvenMediaEngine

Run OvenPlayer Demo

Publishing

Playback

WebRTC Playback

LLHLS Playback

Player & WebRTC Encoder

OvenSpace

What is OvenMediaEngine?

Features

Supported Platforms

Getting Started

How to Contribute

For more information

License

What are the benefits of using P2P Delivery?

How does it work?

How does it classify Peers?

Configuration

Bind

Encoding

Publisher

Get thumbnails

Configuration

MPEGTSPush Publisher