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0.16.8 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 OvenLiveKit, 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 OvenPlayer, 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
- Scheduled Channel (Pre-recorded Live)
- Multiplex Channel (Duplicate stream / Mux tracks)
Adaptive Bitrate Streaming (ABR) for HLS, LLHLS and WebRTC
Low-Latency Streaming using LLHLS
- DVR (Live Rewind)
- Dump for VoD
- ID3v2 timed metadata
- DRM (Widevine, Fairplay)
Sub-Second Latency Streaming using WebRTC
- WebRTC over TCP (with embedded TURN server)
- Embedded WebRTC Signaling Server (WebSocket based)
- Retransmission with NACK
- ULPFEC (Uneven Level Protection Forward Error Correction)
  - VP8, H.264
- In-band FEC (Forward Error Correction)
  - Opus
HLS (version 3) Streaming support for legacy devices
- MPEG-2 TS Container
- Audio/Video Muxed
- DVR (Live Rewind)
Embedded Live Transcoder
- Video: VP8, H.264, H.265(Hardware encoder only), Pass-through
- Audio: Opus, AAC, Pass-through
Clustering (Origin-Edge Structure)
Monitoring
Access Control
- AdmissionWebhooks
- SignedPolicy
File Recording
Push Publishing using SRT, RTMP and MPEG2-TS (Re-streaming)
Thumbnail
REST API

Supported Platforms

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

Docker (https://hub.docker.com/r/airensoft/ovenmediaengine)
Ubuntu 18+
Rocky Linux 9+
AlmaLinux 9+
Fedora 28+

Getting Started

Please read Getting Started 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 Guidelines, Rules, and Contribute.

We always hope that OvenMediaEngine will give you good inspiration.

For more information

OvenMediaEngine GitHub
OvenMediaEngine Website
OvenMediaEngine Tutorial Source
Test Player
- Without TLS: http://demo.ovenplayer.com
- With TLS: https://demo.ovenplayer.com
OvenPlayer Github
AirenSoft Website

License

OvenMediaEngine is licensed under the AGPL-3.0-only. However, if you need another license, please feel free to email us at contact@airensoft.com.

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.

$ docker run --name ome -d -e OME_HOST_IP=Your.HOST.IP.Address \
-p 1935:1935 -p 9999:9999/udp -p 9000:9000 -p 3333:3333 -p 3478:3478 -p 10000-10009:10000-10009/udp \
airensoft/ovenmediaengine:0.16.8

Check Status

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

$ docker ps -f name=ome
CONTAINER ID   IMAGE                              COMMAND                  CREATED              STATUS              PORTS                                                                                                                                                                                                                                                                                                           NAMES
c9dd9e56d7a0   airensoft/ovenmediaengine:latest   "/opt/ovenmediaengin…"   About a minute ago   Up About a minute   0.0.0.0:1935->1935/tcp, :::1935->1935/tcp, 80/tcp, 0.0.0.0:3333->3333/tcp, :::3333->3333/tcp, 3334/tcp, 8080/tcp, 0.0.0.0:3478->3478/tcp, :::3478->3478/tcp, 4000-4005/udp, 8090/tcp, 0.0.0.0:9000->9000/tcp, :::9000->9000/tcp, 10010/udp, 0.0.0.0:9999-10009->9999-10009/udp, :::9999-10009->9999-10009/udp   ome

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

$ docker logs ome -f
[2023-03-06 08:01:24.810] I [OvenMediaEngine:1] Config | config_manager.cpp:239  | Trying to set logfile in directory... (/var/log/ovenmediaengine)
[2023-03-06 08:01:24.810] I [OvenMediaEngine:1] Config | config_manager.cpp:261  | Trying to load configurations... (origin_conf/Server.xml)
[2023-03-06 08:01:24.816] I [OvenMediaEngine:1] OvenMediaEngine | banner.cpp:23   | OvenMediaEngine v0.15.1 () is started on [ab3995acafd4] (Linux x86_64 - 5.13.0-44-generic, #49~20.04.1-Ubuntu SMP Wed May 18 18:44:28 UTC 2022)
...

Run OvenPlayer Demo

$ docker run -d -p 8090:80 airensoft/ovenplayerdemo:latest

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 OBS.

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.

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.

To connect to your OvenMediaEngine in the online demo, you will need to install a certificate and use either the HTTPS or WSS protocol. Unsecured HTTP or WS protocols could not work in online demos due to browser security policies.

OvenSpace

OvenSpace offers a fast and easy way to experience the powerful tools of OvenMediaEngine, OvenPlayer, and OvenLiveKit in action.

With OvenSpace, you can quickly and easily stream content with sub-second latency using WebRTC technology, or take advantage of Apple's LLHLS specification to deliver low-latency live streaming. The platform allows you to stream from various sources, including your webcam, microphone, screen, or an external live encoder that supports RTMP and SRT.

OvenSpace is available online, so you can try it out for yourself at . You'll get a hands-on experience of how OvenMediaEngine, OvenPlayer, and OvenLiveKit work together seamlessly to deliver top-quality streaming, whether you're a developer looking to build a media service or someone who wants to experience sub-second or low-latency streaming firsthand.

OvenSpace is also available on as open source. It will be a good reference when developing media services using OvenMediaEngine, OvenPlayer and OvenLiveKit.

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+, Rocky Linux 9+, and AlmaLinux 9+.

curl -LOJ https://github.com/AirenSoft/OvenMediaEngine/archive/v0.16.8.tar.gz && \
tar xvfz OvenMediaEngine-0.16.8.tar.gz && \
OvenMediaEngine-0.16.8/misc/prerequisites.sh

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

Building & Running

You can build the OvenMediaEngine source using the following command:

sudo apt-get update
cd OvenMediaEngine-0.16.8/src
make release
sudo make install
systemctl start ovenmediaengine
# If you want automatically start on boot
systemctl enable ovenmediaengine.service

sudo dnf update
cd OvenMediaEngine-0.16.8/src
make release
sudo make install
systemctl start ovenmediaengine
# If you want automatically start on boot
systemctl enable ovenmediaengine.service

sudo dnf update
cd OvenMediaEngine-0.16.8/src
make release
sudo make install
systemctl start ovenmediaengine
# If you want automatically start on boot
systemctl enable ovenmediaengine.service

sudo dnf update
cd OvenMediaEngine-0.16.8/src
make release
sudo make install
systemctl start ovenmediaengine
# If you want automatically start on boot
systemctl enable ovenmediaengine.service

if systemctl start ovenmediaengine fails in Fedora, SELinux may be the cause. See Check SELinux section of Troubleshooting.

Ports used by default

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

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

You can open firewall ports as in the following example:

$ sudo firewall-cmd --add-port=3333/tcp
$ sudo firewall-cmd --add-port=3334/tcp
$ sudo firewall-cmd --add-port=1935/tcp
$ sudo firewall-cmd --add-port=9999/udp
$ sudo firewall-cmd --add-port=4000/udp
$ sudo firewall-cmd --add-port=3478/tcp
$ sudo firewall-cmd --add-port=9000/tcp
$ sudo firewall-cmd --add-port=10000-10009/udp

Getting Started with Docker

Getting Started with default settings

OvenMediaEngine provides the Docker image from (airensoft/ovenmediaengine) repository. After installing , you can simply run the following command:

If a certificate is not installed in OvenMediaEngine, some functions (WebRTC Ingest, LLHLS playback) may not work due to the browser's security policy. Please refer to section to install the certificate.

You can set the following environment variables.

Environment Variables

Env

Default Value

Getting Started with Complex Configuration

When you need to install a certificate in OME or apply a complex configuration, you can do it by following the procedure below to modify Server.xml inside Docker.

OvenMediaEngine docker container loads configuration files from the following path.

There are many ways to change files inside a Docker container, but this document describes how to change them using Docker's bind mounts.

Setup

Create the directories

Copy the default configurations from Docker container

Copy the certificate files to the directory

Modify Server.xml if necessary

Running

Run Docker Container

The command below will make your OvenMediaEngine docker container run with $OME_DOCKER_HOME/conf/Server.xml and $OME_DOCKER_HOME/conf/Logger.xml files on your host. It will also create $OME_DOCKER_HOME/logs/ovenmediaengine.log file.

Check the log file

Restart Docker Container

Stop and Remove Container

Getting Started with OME Docker Launcher

Overview and Features

The OME Docker Launcher is a tool that simplifies the process of deploying and managing the OvenMediaEngine (OME) application using Docker containers. This tool can be used by developers and system administrators who want to quickly deploy and test the OME application in a Docker environment.

The OME Docker Launcher provides a set of commands that allow users to easily manage the OME Docker container. These commands include:

- This command pulls the OME Docker image(airensoft/ovenmediaengine:latest) from the Docker registry and copies the necessary configuration files to a specified location. This command needs to be run before starting the OME Docker container.
- This command creates and starts the Docker container. Once the container is started, the OME application can be accessed through a web browser using the container's IP address.
- This command launches a bash shell inside the running OME Docker container, allowing users to execute commands and interact with the container.
- This command displays the status of the running OME Docker container, including information such as the container name, and running status.
- This command stops the running OME Docker container.
- This command stops and then starts the OME Docker container.

Using the OME Docker Launcher, you can easily set up and manage an OME Docker container, without having to manually configure and manage the Docker container. This can save time and effort, especially for users who are not familiar with Docker or who do not want to spend time manually setting up and configuring the OME application.

OME Docker Launcher has not been tested in various environments yet. Therefore, sharing any issues that occur while using it is always welcome.

Installation

Run the following command in your Linux shell.

Below is an example of execution:

Commands

OME Docker Launcher can be executed in the following format:

`setup`

The setup command pulls the OME Docker image from the Docker registry and copies the necessary configuration files to the host's /usr/share/ovenmediaengine directory. Additionally, it initializes the log path and crash dump path that will be mounted into the container when it is run.

This command prepares the host environment for running the OME Docker container and sets up the necessary directories and configurations for the container to run correctly.

If you run the "setup" command, the following files and directories will be created:

/usr/share/ovenmediaengine/conf
- This directory contains the OME configuration files and is mounted into the container when it is run.
/usr/share/ovenmediaengine/logs
- This directory is the log path for OME and is mounted into the container when it is run. Log files generated by OME will be stored in this directory.
/usr/share/ovenmediaengine/dumps
- This directory is the crash dump path for OME and is mounted into the container when it is run. Crash dumps generated by OME will be stored in this directory.

If you want to change the configuration of OME, you can edit the /usr/share/ovenmediaengine/conf/Server.xml file. This file contains the server configuration settings for OME, such as the server's IP address, port, and SSL settings. Once you have made changes to this file, you will need to restart the OME Docker container for the changes to take effect. You can do this by running the restart command provided by the OME Docker Launcher.

Certificate Installation

To install a certificate in OvenMediaEngine, copy the certificate files to /usr/share/ovenmediaengine/conf with the following names:

If you want to change the file names, you can modify Server.xml.

`start`

Once the setup phase is complete, you can use the start command to run the OME Docker container. The start command creates and starts the Docker container, enabling the OME application to receive stream packets using protocols such as RTMP and SRT. Before running the start command, ensure that the necessary configuration files have been copied to the host's /usr/share/ovenmediaengine directory by running the setup command.

When running the OME Docker Launcher, you can specify the IP to be used as an ICE Candidate by using the OME_HOST_IP environment variable. For instance, specifying the OME_HOST_IP as shown below will propagate the ICE Candidate to that particular address.

The OME Docker Launcher automatically detects a list of port numbers specified in the Server.xml file and passes them to the Docker -p option. However, if you use the include attribute inside the <Providers> or <Publishers> element, the launcher may not detect them correctly.

If you have declared the following environment variable in the shell where you run the OME Docker Launcher, this value will be used to bind the port and passed into the Docker container. This enables you to dynamically set configuration values using environment variables.

`sh`

The sh command allows you to enter into the shell of the running container. You can use this command for troubleshooting purpose. Once you enter into the container's shell, you can execute any commands just like you do in a normal Linux shell. This allows you to inspect the container's internal state and debug any issues that you might be facing with the container or the application running inside it.

`status`

The status command shows the current execution status of the container. If the container is running, it displays the ID and name of the container. This command helps you to verify whether the container is up and running or not. If the container is not running, you can use the start command to start the container.

`stop`

The stop command stops the running container and removes it from the list of Docker containers.

`restart`

The restart command restarts the container. This is useful when you need to apply changes to the Server.xml.

Troubleshootings

Enable debug log

If you encounter any problems during the execution, try using the -d option in the [OPTIONS] to view detailed logs. This option shows the command sets and their results that are executed internally.

Get the crash dump of OME in the container

If OME terminates abnormally, providing the crash dump to the OME team can be helpful. The crash dump is stored in the /usr/share/ovenmediaengine/dumps directory, which is created during the setup phase. You can find the dump file named crash_<yyyymmdd>.dump in this directory.

Sharing those log and dump file would be greatly appreciated and helpful for the development of OME.

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:

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

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.

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>.

Providers

Providers ingest streams that come from a media source.

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.

Configuration Example

Finally, Server.xml is configured as follows:

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.

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.

IPv6

Starting from version OME v0.15.1, IPv6 is supported.

To use IPv6, you need to change the settings of the Server.xml file as follows:

1. Configuration for listening

You can use /Server/IP to support IPv6. In versions prior to v0.15.0, only one /Server/IP setting could be specified, but in versions after v0.15.1, multiple settings can be specified. That is, if you add an /Server/IP element for IPv6 to the existing configuration as follows, you can accept IPv6 requests from clients:

* means 0.0.0.0(INADDR_ANY) in IPv4, and :: means ::0(in6addr_any) in IPv6.

Of course, you can also specify a specific IP address of an interface instead of ::.

Example 1) IPv4 Only

OME listens to the 1935 port for RTMP as follows:

Example 2) IPv6 Only

OME listens to the 1935 port for RTMP as follows:

Example 3) IPv4 + IPv6

OME listens to the 1935 port for RTMP as follows:

2. Configuration for `IceCandidates` (for WebRTC)

When you specify IPv6 interface /Server/IP, most Providers/Publishers will work with IPv6, but WebRTC will not. While the WebSocket server used as the WebRTC Signalling server works well with the above setting, but more setting is required for ICE Candidates that actually transmit/receive data.

To use IPv6 ICE Candidate, you need to add an IPv6 IceCandidate to /Server/Bind/(Providers|Publishers)/WebRTC/IceCandidates.

To support IPv6 in URL format settings, use [::] instead of ::

The IceCandidate settings for Providers and Publishers are the same.

By setting up as above, OME is ready to use ICE Candidates for IPv6 as well as IPv4. The ICE Candidate generated here can be viewed in the signaling step of the web browser.

3. Configuration for `<Origin>`

Now you can set up the OME edge to look at an origin with an IPv6 IP address. To do this, you can set /Server/VirtualHosts/VirtualHost/Origins/Origin/Pass/Urls/Url as follows:

This configuration creates a stream that refers an RTSP source provided on port 1234 of an origin which has an IPv6 address of 1:2:3:4:5:6:7:8.

4. Configuration for `<AdmissionWebhooks>`

You can also specify an IPv6 address for the server that AdmissionWebhooks is using. To do this, set the value of /Server/VirtualHosts/VirtualHost/AdmissionWebhooks/ControlServerUrl as follows:

The above configuration asks whether the client has the permission to publish or playback using http://[1:2:3:4:5:6:7:8]:7000/a/b/c.

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:

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

Bind

OvenMediaEngine supports self-defined signaling protocol and WHIP for WebRTC ingest.

<Bind>
    <Providers>
        ...
        <WebRTC>
            <Signalling>
                <Port>3333</Port>
                <TLSPort>3334</TLSPort>
            </Signalling>
            <IceCandidates>
                <TcpRelay>*:3478</TcpRelay>
                <TcpForce>false</TcpForce>
                <IceCandidate>*:10000-10005/udp</IceCandidate>
            </IceCandidates>
        </WebRTC>
    </Providers>

You can set the port to use for signaling in <Bind><Provider><WebRTC><Signaling>. <Port> is for setting an unsecured HTTP port, and <TLSPort> is for setting a secured HTTP port that is encrypted with TLS.

For WebRTC ingest, you must set the ICE candidates of the OvenMediaEnigne server to <IceCandidates>. The candidates set in <IceCandate> are delivered to the WebRTC peer, and the peer requests communication with this candidate. Therefore, you must set the IP that the peer can access. If the IP is specified as *, OvenMediaEngine gathers all IPs of the server and delivers them to the peer.

<TcpRelay> means OvenMediaEngine's built-in TURN Server. When this is enabled, the address of this turn server is passed to the peer via self-defined signaling protocol or WHIP, and the peer communicates with this turn server over TCP. This allows OvenMediaEngine to support WebRTC/TCP itself. For more information on URL settings, check out WebRTC over TCP.

Application

WebRTC input can be turned on/off for each application. As follows Setting enables the WebRTC input function of the application. The <CrossDomains> setting is used in WebRTC signaling.

<Applications>
    <Application>
        <Name>app</Name>
        <Providers>
            <WebRTC>
                <Timeout>30000</Timeout>
                <CrossDomains>
                    <Url>*</Url>
                </CrossDomains>
            </WebRTC>

URL Pattern

OvenMediaEnigne supports self-defined signaling protocol and WHIP for WebRTC ingest.

Self-defined Signaling URL

The signaling URL for WebRTC ingest uses the query string ?direction=send as follows to distinguish it from the url for WebRTC playback. Since the self-defined WebRTC signaling protocol is based on WebSocket, you must specify ws[s] as the scheme.

ws[s]://<host>[:signaling port]/<app name>/<stream name>?direction=send

WHIP URL

For ingest from the WHIP client, put ?direction=whip in the query string in the signaling URL as in the example below. Since WHIP is based on HTTP, you must specify http[s] as the scheme.

http[s]://<host>[:signaling port]/<app name>/<stream name>?direction=whip

WebRTC over TCP

WebRTC transmission is sensitive to packet loss because it affects all players who access the stream. Therefore, it is recommended to provide WebRTC transmission over TCP. OvenMediaEngine has a built-in TURN server for WebRTC/TCP, and receives or transmits streams using the TCP session that the player's TURN client connects to the TURN server as it is. To use WebRTC/TCP, use transport=tcp query string as in WebRTC playback. See WebRTC/tcp playback for more information.

ws[s]://<host>[:port]/<app name>/<stream name>?direction=send&transport=tcp
http[s]://<host>[:port]/<app name>/<stream name>?direction=whip&transport=tcp

To use WebRTC/tcp, <TcpRelay> must be turned on in <Bind> setting.

If <TcpForce> is set to true, it works over TCP even if you omit the ?transport=tcp query string from the URL.

WebRTC Producer

We provide a demo page so you can easily test your WebRTC input. You can access the demo page at the URL below.

The getUserMedia API to access the local device only works in a secure context. So, the WebRTC Input demo page can only work on the https site **** https://demo.ovenplayer.com/demo_input.html. This means that due to mixed content you have to install the certificate in OvenMediaEngine and use the signaling URL as wss to test this. If you can't install the certificate in OvenMediaEngine, you can temporarily test it by allowing the insecure content of the demo.ovenplayer.com URL in your browser.

Self-defined WebRTC Ingest Signaling Protocol

To create a custom WebRTC Producer, you need to implement OvenMediaEngine's Self-defined Signaling Protocol or WHIP. Self-defined protocol is structured in a simple format and uses the same method as WebRTC Streaming.

When the player connects to ws[s]://host:port/app/stream?direction=send through a web socket and sends a request offer command, the server responds to the offer sdp. If transport=tcp exists in the query string of the URL, iceServers information is included in offer sdp, which contains the information of OvenMediaEngine's built-in TURN server, so you need to set this in RTCPeerConnection to use WebRTC/TCP. The player then setsRemoteDescription and addIceCandidate offer sdp, generates an answer sdp, and responds to the server.

SRT

Secure Reliable Transport (or SRT in short) is an open source video transport protocol and technology stack that optimizes streaming performance across unpredictable networks with secure streams and easy firewall traversal, bringing the best quality live video over the worst networks. We consider SRT to be one of the great alternatives to RTMP, and OvenMediaEngine can receive video streaming over SRT. For more information on SRT, please visit the SRT Alliance website.

SRT uses the MPEG-TS format when transmitting live streams. This means that unlike RTMP, it can support many codecs. Currently, OvenMediaEngine supports H.264, H.265, and AAC codecs received by SRT.

Configuration

Bind

Set the SRT listen port as follows:

<Bind>
    <Providers>
        ...
        <SRT>
            <Port>9999</Port>
            <!-- <WorkerCount>1</WorkerCount> -->
        </SRT>
    </Providers>

Application

SRT input can be turned on/off for each application. As follows Setting enables the SRT input function of the application.

<Applications>
    <Application>
        <Name>app</Name>
        <Providers>
            <SRT/>

Encoders and streamid

There are various encoders that support SRT such as FFMPEG, OBS Studio, and srt-live-transmit. Please check the specifications of each encoder on how to transmit streams through SRT from the encoder. We describe an example using OBS Studio.

OvenMediaEngine classifies each stream using SRT's streamid. This means that unlike MEPG-TS/udp, OvenMediaEngine can receive multiple SRT streams through one port. For more information on streamid, see Haivision's official documentation.

Therefore, in order for the SRT encoder to transmit a stream to OvenMediaEngine, the following information must be included in the streamid as percent encoded.

streamid = percent_encoding("srt://{host}[:port]/{app name}/{stream name}[?query=value]")

The streamid contains the URL format, so it must be percent encoded****

OBS Studio

OBS Studio 25.0 or later supports SRT. Please refer to the OBS official documentation for more information. Enter the address of OvenMediaEngine in OBS Studio's Server as follows: When using SRT in OBS, leave the Stream Key blank.

srt://{full domain or IP address}:port?streamid=srt%3A%2F%2F{full domain or IP address}[%3APort]%2F{App name}%2F{Stream name}&latency=2000000

The streamid has to be the urlencoded address of the server name as specified in the ome server configuration plus the app name and the stream name, each separated by /. The latency configures the size of the server-side recive buffer and the time limit for SRT in nanoseconds. Typical value for latency are 150000 (150ms) for stremaing to a server in the local network, 600000 (600ms) for streaming to a server over the internet in the local region, and 2000000 (2 seconds) when stremaing over long distance.

Blackmagic Web Presenter

For configuring a Blackmagic Web Presenter, ATEM Mini Pro or similar device to stream to OvenMediaEngine over SRT, choose the "Custom URL H264/H265" platform option with the following syntax:

Server: srt://your-server-domain.com:<SRT_PORT>
Key: <PERCENT_ENCODED_STREAM_ID>

The default streaming profiles work well, and there are more advanced configuration options available if you edit the streaming.xml settings file

SRT Socket Options

You can configure SRT's socket options of the OvenMediaEngine server using <Options>. This is particularly useful when setting the encryption for SRT, and you can specify a passphrase by configuring as follows:

<Server>
    <Bind>
        <Providers>
            <SRT>
                ...
                <Options>
                    <Option>
                        <Key>SRTO_PBKEYLEN</Key>
                        <Value>16</Value>
                    </Option>
                    <Option>
                        <Key>SRTO_PASSPHRASE</Key>
                        <Value>thisismypassphrase</Value>
                    </Option>
                </Options>
            </SRT>
...

For more information on SRT socket options, please refer to https://github.com/Haivision/srt/blob/master/docs/API/API-socket-options.md#list-of-options.

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.

Bind

To use multiple streams, it is necessary to bind multiple ports, so we provide a way to bind multiple ports as in the example below. You can use the dash to specify the port as a range, such as Start port-End port, and multiple ports using commas.

Stream mapping

First, name the stream and map the port bound above. The macro ${Port} is provided to map multiple streams at once. Check out the example below.

Publish

This is an example of publishing using FFMPEG.

Giving the -pes_payload_size 0 option to the AAC codec is very important for AV synchronization and low latency. If this option is not given, FFMPEG bundles several ADTSs and is transmitted at once, which may cause high latency and AV synchronization errors.

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 . For more information on using the , 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

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.

<VirtualHost>
    <Applications>
        <Providers>
            <Schedule>
                <MediaRootDir>/opt/ovenmediaengine/media</MediaRootDir>
                <ScheduleFilesDir>/opt/ovenmediaengine/media</ScheduleFilesDir>
            </Schedule>
            ...

MediaRootDir Root path where media files are located. If you specify a relative path, the directory where the config file is located is root.

ScheduleFileDir Root path where the schedule file is located. If you specify a relative path, the directory where the config file is located is root.

Schedule Files

Scheduled Channel creates/updates/deletes streams by creating/editing/deleting files with the .sch extension in the ScheduleFileDir path. Schedule files (.sch) use the following XML format. When a <Stream Name>.sch file is created in ScheduleFileDir, OvenMediaEngine analyzes the file and creates a Schedule Channel with <Stream Name>. If the contents of <Stream Name>.sch are changed, the Schedule Channel is updated, and if the file is deleted, the stream is deleted.

<?xml version="1.0" encoding="UTF-8"?>
<Schedule>
    <Stream>
        <Name>tv1</Name> <!-- optional, using filename without ext -->
        <BypassTranscoder>false</BypassTranscoder>
        <VideoTrack>true</VideoTrack>
        <AudioTrack>true</AudioTrack>
    </Stream>
    
    <FallbackProgram> <!-- Not yet supported -->
        <Item url="file://sample.mp4" start="0" duration="60000"/>
    </FallbackProgram>

    <Program name="1" scheduled="2023-09-27T13:21:15.123+09:00" repeat="true">
        <Item url="stream://default/app/stream1" duration="60000"/>
    </Program>
    <Program name="2" scheduled="2022-03-14T15:10:0.0+09:00" repeat="true">
        <Item url="file://sample.mp4" start="0" duration="60000"/>
        <Item url="stream://default/app/stream1" duration="60000"/> <!-- Not yet supported -->
        <Item url="file://sample.mp4" start="60000" duration="120000"/>
    </Program>
</Schedule>

Stream (required) This is the stream information that the Channel needs to create.

Stream.Name (optional) It's the stream's name. This is a reference value extracted from the file name for usage. It's recommended to set it same for consistency, although it's for reference purposes.

Stream.BypassTranscoder (optional, default: false) Set to true if transcoding is not desired.

Stream.VideoTrack (optional, default: true) Determines whether to use the video track. If VideoTrack is set to true and there's no video track in the Item, an error will occur.

Stream.AudioTrack (optional, default: true) Determines whether to use the audio track. If AudioTrack is set to true and there's no audio track in the Item, an error will occur.

FallbackProgram (optional) It is a program that switches automatically when there is no program scheduled at the current time or an error occurs in an item. If the program is updated at the current time or the item returns to normal, it will fail back to the original program. Both files and live can be used for items in FallbackProgram. However, it is recommended to use a stable file.

Program (optional) Schedules a program. The name is an optional reference value. If not set, a random name will be assigned. Set the start time in ISO8601 format in the scheduled attribute. Decide whether to repeat the Items when its playback ends.

Program.Item (optional) Configures the media source to broadcast.

The url points to the location of the media source. If it starts with file://, it refers to a file within the MediaRootDir directory. If it starts with stream://, it refers to another stream within the same OvenMediaEngine. stream:// has the following format: stream://vhost_name/app_name/stream_name

For 'file' cases, the start attribute can be set in milliseconds to indicate where in the file playback should start. duration indicates the playback time of that item in milliseconds. After the duration ends, it moves to the next item. Both 'start' and 'duration' are optional. If not set, start defaults to 0, and duration defaults to the file's duration; if not specified, the media file will be played until its full duration.

Application : Persistent Live Channel

This function is a scheduling channel, but it can be used for applications such as creating a permanent stream as follows.

<?xml version="1.0"?>
<Schedule>
        <Stream>
                <Name>stream</Name>
                <BypassTranscoder>false</BypassTranscoder>
                <VideoTrack>true</VideoTrack>
                <AudioTrack>true</AudioTrack>
        </Stream>
        <FallbackProgram>
                <Item url="file://hevc.mov"/>
                <Item url="file://avc.mov"/>
        </FallbackProgram>

        <Program name="origin" scheduled="2000-01-01T20:57:00.000+09" repeat="true">
                <Item url="stream://default/app/input" duration="-1" />
        </Program>
</Schedule>

This channel normally plays default/app/input, but when live input is stopped, it plays the file in FallbackProgram. This will last forever until the .sch file is deleted. One trick was to set the origin program's schedule time to year 2000 so that this stream would play unconditionally.

You may experience some buffering when going from file to live. This is unavoidable due to the nature of the function and low latency. If this is inconvenient, buffering issues can disappear if you add a little delay in advance by setting PartHoldBack in LLHLS to 5 or more. It is a choice between delay and buffering.

REST API

ScheduledChannel can also be controlled via API. Please refer to the page below.

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.

<VirtualHost>
    <Applications>
	<Providers>
		...
		<Multiplex>
			<MuxFilesDir>mux_files</MuxFilesDir>
		</Multiplex>
	</Providers>

Multiplex Channels are created through .mux files or API. MuxFilesDir is the path where the .mux files are located and can be set to an absolute system path or relative to the path where the Server.xml configuration is located.

The Multiplex Provider monitors the MuxFilesDir path, and when a mux file is created, it parses the file and creates a multiplex channel. When the mux file is modified, the channel is deleted and created again, and when the mux file is deleted, the channel is deleted.

Mux file format

mux files can be created or deleted while the system is running. This works dynamically. The mux file has the format below.

<?xml version="1.0" encoding="UTF-8"?>
<Multiplex>
    
    <OutputStream>
        <Name>stream</Name>
    </OutputStream>

    <SourceStreams>
        <SourceStream>
            <Name>tv1</Name>
            <Url>stream://default/app/tv1</Url>
            <TrackMap>
                <Track>
                    <SourceTrackName>bypass_video</SourceTrackName>
                    <NewTrackName>tv1_video</NewTrackName>
                </Track>
                <Track>
                    <SourceTrackName>bypass_audio</SourceTrackName>
                    <NewTrackName>tv1_audio</NewTrackName>
                </Track>
                <Track>
                    <SourceTrackName>opus</SourceTrackName>
                    <NewTrackName>tv1_opus</NewTrackName>
                </Track>
            </TrackMap>
        </SourceStream>
        <SourceStream>
            <Name>tv2</Name>
            <Url>stream://default/app/tv2</Url>

            <TrackMap>
                <Track>
                    <SourceTrackName>bypass_video</SourceTrackName>
                    <NewTrackName>tv2_video</NewTrackName>
                </Track>
                <Track>
                    <SourceTrackName>bypass_audio</SourceTrackName>
                    <NewTrackName>tv2_audio</NewTrackName>
                </Track>
                <Track>
                    <SourceTrackName>opus</SourceTrackName>
                    <NewTrackName>tv2_opus</NewTrackName>
                </Track>
            </TrackMap>

        </SourceStream>
    </SourceStreams>
    
    <Playlists>
        <Playlist>
            <Name>LLHLS ABR</Name>
            <FileName>abr</FileName>
            
            <Rendition>
                <Name>1080p</Name>
                <Video>tv1_video</Video>
                <Audio>tv1_audio</Audio>
            </Rendition>
            <Rendition>
                <Name>720p</Name>
                <Video>tv2_video</Video>
                <Audio>tv2_audio</Audio>
            </Rendition>
        </Playlist>
    </Playlists>
    
</Multiplex>

OutputStream This is information about the stream to be newly created. It must be the same as the file name. <stream name>.mux

SourceStreams Specifies the internal stream to be muxed. You can also load streams from other VHosts or applications in the format stream://<vhost name>/<app name>/<stream name>. Because multiple streams are muxed into one stream, track names may be duplicated. Therefore, it is necessary to change the Track name for each SourceStream through <TrackMap>. <SourceTrackName> is either <OutputProfile><Encodes><Video><Name> or <OutputProfile><Encodes><Audio><Name>.

Playlist The same format as <OutputProfile> must be used, and the Playlist must be constructed using the newly mapped Track name in SourceStreams' TrackMap. The Playlist configured here exists only in this stream. The Playlist's FileName must be unique throughout the application.

REST API

MultiplexChannel can also be controlled via API. Please refer to the page below.

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
HLS http://192.168.0.1:8080/app/stream_bypass/ts:playlist.m3u8

Encodes

Video

You can set the video profile as below:

The meaning of each property is as follows:

* 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.

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:

* 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:

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

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

Matching elements in audio

* 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 HLS is as follows.

http[s]://<domain>[:port]/<app>/<stream>/<FileName>.m3u8?format=ts

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

TS files used in HLS must have A/V pre-muxed, so the EnableTsPackaging option must be set in the Playlist.

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:

Therefore, you set it up as shown in the table. If you want to stream using LLHLS, you need to set up H.264, H.265 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.

Enable GPU Acceleration

OvenMediaEngine supports GPU-based hardware decoding and encoding. Currently supported GPU acceleration devices are Intel's QuickSync and NVIDIA. This article explains how to install the drivers for OvenMediaEngine and set up the configuration to use your GPU.

1. Install Drivers

1. Install NVIDIA GPU Driver

If you are using an NVIDIA graphics card, please refer to the following guide to install the driver. The OS that supports installation with the provided script are CentOS 7/8 and Ubuntu 18/20 versions. If you want to install the driver in another OS, please refer to the manual installation guide document.

CentOS environment requires the process of uninstalling the nouveau driver. After uninstalling the driver, the first reboot is required, and a new NVIDIA driver must be installed and rebooted. Therefore, two install scripts must be executed.

How to check driver installation

After the driver installation is complete, check whether the driver is operating normally with the nvidia-smi command.

2 . Prerequisites

If you have finished installing the driver to use the GPU, you need to reinstall the open source library using Prerequisites.sh . The purpose is to allow external libraries to use the installed graphics driver.

1. Install NVIDIA GPU Driver

Please refer to the NVIDIA Driver installation guide written previously.

2. Install NVIDIA Container Toolkit

To use GPU acceleration in Docker, you need to install NVIDIA drivers on your host OS and install the NVIDIA Container Toolkit. This toolkit includes container runtime libraries and utilities for using NVIDIA GPUs in Docker containers.

3 . Build Image

A Docker Image build script that supports NVIDIA GPU is provided separately. Please refer to the previous guide for how to build

1. Install Intel QuickSync Driver

If you are using an Intel CPU that supports QuickSync, please refer to the following guide to install the driver. The OSes that support installation using the provided scripts are CentOS 7/8 and Ubuntu 18/20 versions. If you want to install the driver on a different OS, please refer to the Manual Installation Guide document.

When the Intel QuickSync driver installation is complete, the OS must be rebooted for normal operation.

How to check driver installation

After the driver installation is complete, check whether the driver operates normally with the Matrix Monitor program.

2. Prerequisites

Using Intel QuickSync GPU

1. Install XCODER

Please refer to the Netint documentation to install XCODER.

How to check driver installation

After the driver installation is complete, check if the libxcoder exist: the CLI must return something like libxcoder_logan.so (libc6,x86-64) => /usr/local/lib/libxcoder_logan.so

2. Prerequisites

If you have finished installing the driver to use the VPU, you need to reinstall the open source library using Prerequisites.sh . The purpose is to allow external libraries to use the installed graphics driver. You also have to unzip the ffmpeg patch provide by netint in a specfic path

Using Netint VPU

2. Build & Run

Please refer to the link for how to build and run.

Intructions on running Docker

you must include the --gpus all option when running Docker

3. Configuration

To use hardware acceleration, set the HardwareAcceleration option to true under OutputProfiles. If this option is enabled, a hardware codec is automatically used when creating a stream, and if it is unavailable due to insufficient hardware resources, it is replaced with a software codec.

Appendix. Support Format

The codecs available using hardware accelerators in OvenMediaEngine are as shown in the table below. Different GPUs support different codecs. If the hardware codec is not available, you should check if your GPU device supports the codec.

D : Decoding, E : Encoding

Reference

TranscodeWebhook

TranscodeWebhook allows OvenMediaEngine to use OutputProfiles from the Control Server's response instead of the OutputProfiles in the local configuration (Server.xml). OvenMediaEngine requests OutputProfiles from the Control Server when streams are created, enabling the specification of different profiles for each individual stream.

Configuration

Enable (required) You can enable or disable TranscodeWebhook settings.

ControlServerUrl (required) It's the URL of the Control Server, and it supports both HTTP and HTTPS.

SecretKey (optional) This is the Secret Key used to pass authentication for the Control Server. To pass security authentication, an HMAC-SHA1 encrypted value of the HTTP Payload is added to the HTTP Header's X-OME-Signature. This Key is used for generating this value.

Timeout (optional, default: 1500) This is the Timeout value used when connecting to the Control Server.

UseLocalProfilesOnConnectionFailure(optional, default: true) This determines whether to use the OutputProfiles from Local settings in case of communication failure with the Control Server. If it's set to "false," a communication failure with the Control Server will result in a failure to create the Output stream.

UseLocalProfilesOnServerDisallow (optional, default: false) When the Control Server responds with a 200 OK, but "allowed" is set to "false," this policy is followed.

UseLocalProfilesOnErrorResponse (optional, default: false) When the Control Server responds with error status codes such as 400 Bad Request, 404 Not Found, 500 Internal Error, OvenMediaEngine follows this policy.

Protocol

Request (OME → Control Server)

OvenMediaEngine sends requests to the Control Server in the following format.

Response (Control Server → OME)

The Control Server responds in the following format to specify OutputProfiles for the respective stream.

The "outputProfiles" in the JSON is identical to the configuration in Server.xml, and the format is as follows.

Streaming

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

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.

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.
Players that do not support RTCP also cannot A/V sync.

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:

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

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.

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

Turn on TURN server

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

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

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

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:

HLS

Beta

HLS is still in development and some features such as SignedPolicy and AdmissionWebhooks are not supported.

HLS based on MPEG-2 TS containers is still useful because it provides high compatibility, including support for older devices. Therefore, OvenMediaEngine decided to officially support HLS version 7+ based on fragmented MP4 containers, called LL-HLS, as well as HLS version 3+ based on MPEG-2 TS containers.

Title

Descriptions

To differentiate between LLHLS and HLS playback addresses, we created the following rule for HLS playback addresses:

http[s]://domain[:port]/<app name>/<stream name>/ts:<playlist file name>.m3u8

http[s]://domain[:port]/<app name>/<stream name>/<playlist file name>.m3u8?format=ts

Configuration

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

<?xml version="1.0" encoding="UTF-8"?>
<Server version="8">
  <Bind>
    <Publishers>
      <HLS>
        <Port>13333</Port>
        <TLSPort>13334</TLSPort>
        <WorkerCount>1</WorkerCount>
      </HLS>
    </Publishers>
  </Bind>
  <VirtualHosts>
    <VirtualHost>
      <Applications>
        <Application>
          <Publishers>
            <HLS>
              <SegmentCount>5</SegmentCount>
              <SegmentDuration>4</SegmentDuration>
              <DVR>
                <Enable>true</Enable>
                <EventPlaylistType>false</EventPlaylistType>
                <TempStoragePath>/tmp/ome_dvr/</TempStoragePath>
                <MaxDuration>600</MaxDuration>
              </DVR>
              <CrossDomains>
                <Url>*</Url>
              </CrossDomains>
            </HLS>
          </Publishers>
        </Application>
      </Applications>
    </VirtualHost>
  </VirtualHosts>
</Server>

Safari Native Player only provides the Seek UI if #EXT-X-PLAYLIST-TYPE: EVENT is present. Since it is specified that nothing can be removed from the playlist when it is of type EVENT, you must call the concludeHlsLive API to switch to VoD or terminate the stream before MaxDuration is exceeded if you use this option. Otherwise, unexpected behavior may occur in the Safari Player.

Playback

HLS 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 Transcoding.

<OutputProfiles>
    <OutputProfile>
        <Name>bypass_stream</Name>
        <OutputStreamName>${OriginStreamName}</OutputStreamName>
        <Encodes>
            <Audio>
                <Bypass>true</Bypass>
            </Audio>
            <Video>
                <Bypass>true</Bypass>
            </Video>
        </Encodes>
    </OutputProfile>
</OutputProfiles>

HLS Publisher basically creates a playlist.m3u8 Playlist using the first video track and the first audio track. When you create a stream, as shown above, you can play HLS with the following URL:

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

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

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

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

Adaptive Bitrates Streaming (ABR)

HLS can deliver adaptive bitrate streaming. OME encodes the same source with multiple renditions and delivers it to the players. And HLS 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 Adaptive Bitrates Streaming section for how to configure renditions.

HLS Publisher basically creates a playlist.m3u8 Playlist using the first video track and the first audio track. If you want to create a new playlist for ABR, you can add it to Server.xml as follows:

Since TS files used in HLS must have A/V pre-muxed, the Playlist must have the EnableTsPackaging option set.

<?xml version="1.0" encoding="UTF-8"?>
<OutputProfile>
  <Name>abr_stream</Name>
  <OutputStreamName>${OriginStreamName}</OutputStreamName>
  <Playlist>
    <Name>abr</Name>
    <FileName>abr</FileName>
    <Options>
      <HLSChunklistPathDepth>0</HLSChunklistPathDepth>
      <EnableTsPackaging>true</EnableTsPackaging>
    </Options>
    <Rendition>
      <Name>SD</Name>
      <Video>video_360</Video>
      <Audio>aac_audio</Audio>
    </Rendition>
    <Rendition>
      <Name>HD</Name>
      <Video>video_720</Video>
      <Audio>aac_audio</Audio>
    </Rendition>
    <Rendition>
      <Name>FHD</Name>
      <Video>video_1080</Video>
      <Audio>aac_audio</Audio>
    </Rendition>
  </Playlist>
  <Encodes>
    <Audio>
      <Name>aac_audio</Name>
      <Codec>aac</Codec>
      <Bitrate>128000</Bitrate>
      <Samplerate>48000</Samplerate>
      <Channel>2</Channel>
      <BypassIfMatch>
        <Codec>eq</Codec>
      </BypassIfMatch>
    </Audio>
    <Video>
      <Name>video_360</Name>
      <Codec>h264</Codec>
      <Bitrate>365000</Bitrate>
      <Framerate>30</Framerate>
      <Width>640</Width>
      <Height>360</Height>
      <KeyFrameInterval>30</KeyFrameInterval>
      <ThreadCount>2</ThreadCount>
      <Preset>medium</Preset>
      <BFrames>0</BFrames>
      <ThreadCount>1</ThreadCount>
    </Video>
    <Video>
      <Name>video_720</Name>
      <Codec>h264</Codec>
      <Profile>high</Profile>
      <Bitrate>1500000</Bitrate>
      <Framerate>30</Framerate>
      <Width>1280</Width>
      <Height>720</Height>
      <KeyFrameInterval>30</KeyFrameInterval>
      <Preset>medium</Preset>
      <BFrames>2</BFrames>
      <ThreadCount>4</ThreadCount>
    </Video>
    <Video>
      <Name>video_1080</Name>
      <Codec>h264</Codec>
      <Bitrate>6000000</Bitrate>
      <Framerate>30</Framerate>
      <Width>1920</Width>
      <Height>1080</Height>
      <KeyFrameInterval>30</KeyFrameInterval>
      <ThreadCount>8</ThreadCount>
      <Preset>medium</Preset>
      <BFrames>0</BFrames>
    </Video>
  </Encodes>
</OutputProfile>

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.

Server.xml

<CrossDomains>
    <Url>*</Url>
    <Url>*.airensoft.com</Url>
    <Url>http://*.ovenplayer.com</Url>
    <Url>https://demo.ovenplayer.com</Url>
</CrossDomains>

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

Live Rewind

You can create as long a playlist as you want by setting <DVR> to the HLS 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.

<HLS>
    ...
    <DVR>
        <Enable>true</Enable>
        <TempStoragePath>/tmp/ome_dvr/</TempStoragePath>
        <MaxDuration>3600</MaxDuration>
    </DVR>
    ...
</HLS>

Access Control

Statistics

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

IP

    <IP>*</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

StunServer

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.

<StunServer>stun.l.google.com:19302</StunServer>

Bind

<!-- Settings for the ports to bind -->
<Bind>
    <!-- Enable this configuration if you want to use API Server -->
    <!--
    <Managers>
        <API>
            <Port>8081</Port>
            <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>

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

Element

Description

Virtual Host

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

<?xml version="1.0" encoding="UTF-8"?>
<Server version="8">
    <Name>OvenMediaEngine</Name>
    <VirtualHosts>
        <VirtualHost>
            <Name>default</Name>
            <Host>
            ...
            </Host>

            <Origins>
            ...
            </Origins>

            <SignedPolicy>
            ...
            </SignedPolicy>

            <Applications>
            ...
            </Applications>
        </Host>
    </Hosts>
</Server>

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.

<Host>
        <Names>
            <!-- Domain names
            <Name>stream1.airensoft.com</Name>
            <Name>stream2.airensoft.com</Name>
            <Name>*.sub.airensoft.com</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>

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

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.

<Origins>
    <Origin>
        <Location>/app/stream</Location>
        <Pass>
            <Scheme>ovt</Scheme>
            <Urls><Url>origin.com:9000/app/stream_720p</Url></Urls>
        </Pass>
    </Origin>
    <Origin>
        <Location>/app/</Location>
        <Pass>
            <Scheme>ovt</Scheme>
            <Urls><Url>origin.com:9000/app/</Url></Urls>
        </Pass>
    </Origin>
    <Origin>
        <Location>/rtsp/stream</Location>
        <Pass>
            <Scheme>rtsp</Scheme>
            <Urls><Url>rtsp-server.com:554/</Url></Urls>
        </Pass>
    </Origin>
    <Origin>
        <Location>/</Location>
        <Pass>
            <Scheme>ovt</Scheme>
            <Urls><Url>origin2.com:9000/</Url></Urls>
        </Pass>
    </Origin>
</Origins>

Application

<VirtualHost>
    ...
    <Applications>
        <Application>
            ...
        </Application>
        <Application>
            ...
        </Application>
    </Applications>
</VirtualHost>

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

<Application>
    <Name>app</Name>
    <Type>live</Type>
    <OutputProfiles> ... </OutputProfiles>
    <Providers> ... </Providers>
    <Publishers> ... </Publishers>
</Application>

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

OutputProfiles

<Application>
    <OutputProfiles>
            <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>
                    </Video>                                
                    -->
                </Encodes>
            </OutputProfile>
        </OutputProfiles>
</Application>

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

Providers

Providers ingest streams that come from a media source.

<Application>
   <Providers>
      <RTMP/>
      <WebRTC/>
      <SRT/>
      <RTSPPull/>
      <OVT/>
      <MPEGTS>
         <StreamMap>
            ...
         </StreamMap>
      </MPEGTS>
   </Providers>
</Application>

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.

<Application>
   <Publishers>
      <OVT />
	  <LLHLS />
      <WebRTC />
   </Publishers>
</Application>

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,llhls</Publishers>
                </Enables>
            </SignedPolicy>
            -->

            <!--
            <AdmissionWebhooks>
                <ControlServerUrl></ControlServerUrl>
                <SecretKey></SecretKey>
                <Timeout>3000</Timeout>
                <Enables>
                    <Providers>rtmp,webrtc,srt</Providers>
                    <Publishers>webrtc,llhls</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>

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

sudo apt install -y build-essential nasm autoconf libtool zlib1g-dev tclsh cmake curl

sudo yum install -y gcc-c++ make nasm autoconf libtool zlib-devel tcl cmake

sudo dnf install -y bc gcc-c++ autoconf libtool tcl bzip2 zlib-devel cmake libuuid-devel
sudo dnf install -y perl-IPC-Cmd perl-FindBin

Common Installation

Install OpenSSL

PREFIX=/opt/ovenmediaengine && \
OPENSSL_VERSION=1.1.0g && \
DIR=/tmp/openssl && \
mkdir -p ${DIR} && \
cd ${DIR} && \
curl -sLf https://www.openssl.org/source/openssl-${OPENSSL_VERSION}.tar.gz | tar -xz --strip-components=1 && \
./config --prefix="${PREFIX}" --openssldir="${PREFIX}" -Wl,-rpath="${PREFIX}/lib" shared no-idea no-mdc2 no-rc5 no-ec2m no-ecdh no-ecdsa no-async && \
make -j 4 && \
sudo make install_sw && \
rm -rf ${DIR} && \
sudo rm -rf ${PREFIX}/bin

Install SRTP

PREFIX=/opt/ovenmediaengine && \
SRTP_VERSION=2.2.0 && \
DIR=/tmp/srtp && \
mkdir -p ${DIR} && \
cd ${DIR} && \
curl -sLf https://github.com/cisco/libsrtp/archive/v${SRTP_VERSION}.tar.gz | tar -xz --strip-components=1 && \
./configure --prefix="${PREFIX}" --enable-shared --disable-static --enable-openssl --with-openssl-dir="${PREFIX}" && \
make shared_library -j 4 && \
sudo make install && \
rm -rf ${DIR}

Install SRT

PREFIX=/opt/ovenmediaengine && \
SRT_VERSION=1.3.3 && \
DIR=/tmp/srt && \
mkdir -p ${DIR} && \
cd ${DIR} && \
curl -sLf https://github.com/Haivision/srt/archive/v${SRT_VERSION}.tar.gz | tar -xz --strip-components=1 && \
PKG_CONFIG_PATH=${PREFIX}/lib/pkgconfig:${PKG_CONFIG_PATH} ./configure --prefix="${PREFIX}" --enable-shared --disable-static && \
make -j 4 && \
sudo make install && \
rm -rf ${DIR} && \
sudo rm -rf ${PREFIX}/bin

Install Opus

PREFIX=/opt/ovenmediaengine && \
OPUS_VERSION=1.1.3 && \
DIR=/tmp/opus && \
mkdir -p ${DIR} && \
cd ${DIR} && \
curl -sLf https://archive.mozilla.org/pub/opus/opus-${OPUS_VERSION}.tar.gz | tar -xz --strip-components=1 && \
autoreconf -fiv && \
./configure --prefix="${PREFIX}" --enable-shared --disable-static && \
make -j 4&& \
sudo make install && \
sudo rm -rf ${PREFIX}/share && \
rm -rf ${DIR}

Install x264

PREFIX=/opt/ovenmediaengine && \
X264_VERSION=20190513-2245-stable && \
DIR=/tmp/x264 && \
mkdir -p ${DIR} && \
cd ${DIR} && \
curl -sLf https://download.videolan.org/pub/videolan/x264/snapshots/x264-snapshot-${X264_VERSION}.tar.bz2 | tar -jx --strip-components=1 && \
./configure --prefix="${PREFIX}" --enable-shared --enable-pic --disable-cli && \
make -j 4&& \
sudo make install && \
rm -rf ${DIR}

Install VPX

PREFIX=/opt/ovenmediaengine && \
VPX_VERSION=1.7.0 && \
DIR=/tmp/vpx && \
mkdir -p ${DIR} && \
cd ${DIR} && \
curl -sLf https://codeload.github.com/webmproject/libvpx/tar.gz/v${VPX_VERSION} | tar -xz --strip-components=1 && \
./configure --prefix="${PREFIX}" --enable-vp8 --enable-pic --enable-shared --disable-static --disable-vp9 --disable-debug --disable-examples --disable-docs --disable-install-bins && \
make -j 4 && \
sudo make install && \
rm -rf ${DIR}

Install FDK-AAC

PREFIX=/opt/ovenmediaengine && \
FDKAAC_VERSION=0.1.5 && \
DIR=/tmp/aac && \
mkdir -p ${DIR} && \
cd ${DIR} && \
curl -sLf https://github.com/mstorsjo/fdk-aac/archive/v${FDKAAC_VERSION}.tar.gz | tar -xz --strip-components=1 && \
autoreconf -fiv && \
./configure --prefix="${PREFIX}" --enable-shared --disable-static --datadir=/tmp/aac && \
make -j 4&& \
sudo make install && \
rm -rf ${DIR}

Install FFMPEG

PREFIX=/opt/ovenmediaengine && \
FFMPEG_VERSION=3.4 && \
DIR=/tmp/ffmpeg && \
mkdir -p ${DIR} && \
cd ${DIR} && \
curl -sLf https://github.com/AirenSoft/FFmpeg/archive/ome/${FFMPEG_VERSION}.tar.gz | tar -xz --strip-components=1 && \
PKG_CONFIG_PATH=${PREFIX}/lib/pkgconfig:${PKG_CONFIG_PATH} ./configure \
--prefix="${PREFIX}" \
--enable-gpl \
--enable-nonfree \
--extra-cflags="-I${PREFIX}/include"  \
--extra-ldflags="-L${PREFIX}/lib -Wl,-rpath,${PREFIX}/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 && \
make && \
sudo make install && \
sudo rm -rf ${PREFIX}/share && \
rm -rf ${DIR}

Install JEMALLOC

PREFIX=/opt/ovenmediaengine && \
JEMALLOC_VERSION=5.2.1 && \
DIR=${TEMP_PATH}/jemalloc && \
mkdir -p ${DIR} && \
cd ${DIR} && \
curl -sLf https://github.com/jemalloc/jemalloc/releases/download/${JEMALLOC_VERSION}/jemalloc-${JEMALLOC_VERSION}.tar.bz2 | tar -jx --strip-components=1 && \
./configure --prefix="${PREFIX}" && \
make && \
sudo make install_include install_lib && \
rm -rf ${DIR}

`systemctl start ovenmediaengine` failed

Check SELinux

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

# Example of SELinux disallow OvenMediaEngine execution
$ systemctl start ovenmediaengine
==== AUTHENTICATING FOR org.freedesktop.systemd1.manage-units ====
Authentication is required to start 'ovenmediaengine.service'.
Authenticating as: Jeheon Han (getroot)
Password:
==== AUTHENTICATION COMPLETE ====
Failed to start ovenmediaengine.service: Unit ovenmediaengine. service not found.
# Check if SELinux is enabled
$ sestatus
SELinux status:                 enabled
SELinuxfs mount:                /sys/fs/selinux
SELinux root directory:         /etc/selinux
Loaded policy name:             targeted
Current mode:                   enforcing
Mode from config file:          enforcing
Policy MLS status:              enabled
Policy deny_unknown status:     allowed
Memory protection checking:     actual (secure)
Max kernel policy version:      31
# Check if SELinux denies execution
$ sudo tail /var/log/messages
...
May 17 12:44:24 localhost audit[1]: AVC avc:  denied  { read } for  pid=1 comm="systemd" name="ovenmediaengine.service" dev="dm-0" ino=16836708 scontext=system_u:system_r:init_t:s0 tcontext=system_u:object_r:default_t:s0 tclass=file permissive=0
May 17 12:44:24 localhost audit[1]: AVC avc:  denied  { read } for  pid=1 comm="systemd" name="ovenmediaengine.service" dev="dm-0" ino=16836708 scontext=system_u:system_r:init_t:s0 tcontext=system_u:object_r:default_t:s0 tclass=file permissive=0

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:

$ cd <OvenMediaEngine Git Clone Root Path>
$ sudo semodule -i misc/ovenmediaengine.pp
$ sudo touch /.autorelabel
# If you add a policy to SELinux, you must reboot the system.
$ sudo reboot

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

$ sudo setenforce 0

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: https://airensoft.gitbook.io/ovenmediaengine/transcoding#encodes

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: https://airensoft.gitbook.io/ovenmediaengine/streaming/webrtc-publishing#webrtc-over-tcp

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: https://airensoft.gitbook.io/ovenmediaengine/transcoding/gpu-usage

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: https://airensoft.gitbook.io/ovenmediaengine/performance-tuning#performance-tuning

5-2. Tuning your Linux kernel

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

[ec2-user@ip-172-31-56-213 ~]$ cat /etc/sysctl.conf
fs.file-max = 100000
net.core.somaxconn = 65535
net.ipv4.tcp_max_tw_buckets = 1440000
net.ipv4.ip_local_port_range = 1024 65000
net.ipv4.tcp_fin_timeout = 15
net.ipv4.tcp_window_scaling = 1
net.ipv4.tcp_max_syn_backlog = 324000
net.core.rmem_max = 16777216
net.core.wmem_max = 16777216
net.core.rmem_default = 16777216
net.core.wmem_default = 16777216
net.core.optmem_max = 40960
net.ipv4.tcp_rmem = 4096 87380 16777216
net.ipv4.tcp_wmem = 4096 65536 16777216
net.core.netdev_max_backlog = 50000
net.ipv4.tcp_max_tw_buckets = 2000000
net.ipv4.tcp_tw_reuse = 1
net.ipv4.tcp_fin_timeout = 10
net.ipv4.tcp_slow_start_after_idle = 0

[ec2-user@ip-172-31-56-213 ~]$ cat /etc/security/limits.conf
* soft nofile 1048576
* hard nofile 1048576

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: https://airensoft.gitbook.io/ovenmediaengine/streaming/tls-encryption

As of October 2021, most browsers have enforced the CORS policy, 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:

[ec2-user@ip-172-31-56-213 ~]$ cat /etc/security/limits.conf
* soft nofile 1048576
* hard nofile 1048576

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: https://airensoft.gitbook.io/ovenmediaengine/transcoding#encodes

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: https://airensoft.gitbook.io/ovenmediaengine/streaming/webrtc-publishing#publisher

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: https://airensoft.gitbook.io/ovenmediaengine/streaming/webrtc-publishing#publisher

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 OvenMediaEngine GitHub Issues.

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: https://airensoft.gitbook.io/ovenmediaengine/transcoding#audio

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: https://airensoft.gitbook.io/ovenmediaengine/transcoding#video