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0.16.[1-3]

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
Adaptive Bitrate Streaming (ABR) for 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
Embedded Live Transcoder
- Video: VP8, H.264, Pass-through
- Audio: Opus, AAC, Pass-through
Clustering (Origin-Edge Structure)
Monitoring
Access Control
- AdmissionWebhooks
- SignedPolicy
File Recording
Push Publishing using RTMP and MPEG2-TS (Re-streaming)
Thumbnail
REST API
Experiment
- P2P Traffic Distribution (Only WebRTC)

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+
CentOS 7+
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:0.16.3

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 https://space.ovenplayer.com/. 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 Github 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+, and CentOS 7+.

curl -LOJ https://github.com/AirenSoft/OvenMediaEngine/archive/v0.16.3.tar.gz && \
tar xvfz OvenMediaEngine-0.16.3.tar.gz && \
OvenMediaEngine-0.16.3/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-0.16.3/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-0.16.3/src
make release
sudo make install
systemctl start ovenmediaengine
# If you want automatically start on boot
systemctl enable ovenmediaengine.service

sudo yum update
source scl_source enable devtoolset-7
cd OvenMediaEngine-0.16.3/src
make release
sudo make install
systemctl start ovenmediaengine
# If you want automatically start on boot
systemctl enable ovenmediaengine.service

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

$ echo 'source scl_source enable devtoolset-7' >> ~/.bashrc

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's Docker Hub (airensoft/ovenmediaengine) repository. After installing Docker, you can simply run the following command:

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:0.16.3

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 Complex Configuration 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

export OME_DOCKER_HOME=/opt/ovenmediaengine
sudo mkdir -p $OME_DOCKER_HOME/conf
sudo mkdir -p $OME_DOCKER_HOME/logs

# Set permissions for the created directory if necessary.
sudo chgrp -R docker $OME_DOCKER_HOME 
sudo chmod -R 775 $OME_DOCKER_HOME

# If you want to use OME_HOME permanently, add the following line to the ~/.profile file for bash, for other shells, you can do it accordingly.
echo "export OME_DOCKER_HOME=/opt/ovenmediaengine" >> ~/.profile

Copy the default configurations from Docker container

docker run -d --name tmp-ome airensoft/ovenmediaengine:0.16.3
docker cp tmp-ome:/opt/ovenmediaengine/bin/origin_conf/Server.xml $OME_DOCKER_HOME/conf
docker cp tmp-ome:/opt/ovenmediaengine/bin/origin_conf/Logger.xml $OME_DOCKER_HOME/conf
docker rm -f tmp-ome

Copy the certificate files to the directory

Copy your PEM certificate files to the path below if you need to enable TLS. The destination file names must match if using the default configuration. If you want to change the file name, you can do so by editing the Server.xml configuration file. See TLS Encryption for details.

cp /your/server_certificate_file.crt $OME_DOCKER_HOME/conf/cert.crt
cp /your/certificate_key_file.key $OME_DOCKER_HOME/conf/cert.key
cp /your/ca_bundle_file.ca-bundle $OME_DOCKER_HOME/conf/cert.ca-bundle

Modify Server.xml if necessary

vi $OME_DOCKER_HOME/conf/Server.xml

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.

docker run -d -it --name ome -e OME_HOST_IP=Your.HOST.IP.Address \
-v $OME_DOCKER_HOME/conf:/opt/ovenmediaengine/bin/origin_conf \
-v $OME_DOCKER_HOME/logs:/var/log/ovenmediaengine \
-p 1935:1935 -p 9999:9999/udp -p 9000:9000 -p 3333:3333 -p 3478:3478 \
-p 10000-10009:10000-10009/udp \
airensoft/ovenmediaengine:0.16.3

Check the log file

tail -f $OME_DOCKER_HOME/logs/ovenmediaengine.log

Restart Docker Container

docker restart ome

Stop and Remove Container

docker stop ome
docker rm ome

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:

setup
- 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.
start
- 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.
sh
- This command launches a bash shell inside the running OME Docker container, allowing users to execute commands and interact with the container.
status
- This command displays the status of the running OME Docker container, including information such as the container name, and running status.
stop
- This command stops the running OME Docker container.
restart
- 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.

curl -OL 'https://raw.githubusercontent.com/AirenSoft/OvenMediaEngine/master/misc/ome_docker_launcher.sh' && chmod +x ome_docker_launcher.sh

Below is an example of execution:

$ curl -OL 'https://raw.githubusercontent.com/AirenSoft/OvenMediaEngine/master/misc/ome_docker_launcher.sh' && chmod +x ome_docker_launcher.sh
$ ./ome_docker_launcher.sh -h

 ▄██████▀███▄
█████▀ ▄██████  OvenMediaEngine Launcher v0.1
███▄▄▄▄▀▀▀▀███
██████▀ ▄█████  https://github.com/AirenSoft/OvenMediaEngine
 ▀███▄██████▀

• Usage: ./ome_docker_launcher.sh [OPTIONS] COMMAND ...

• Options:
  -h, --help                        Show this help message and exit
  -v, --version                     Show the version and exit
  -d, --debug                       Show debug log
  -b, --hide_banner                 Hide the banner
  -m, --monochrome                  Disable colors

• Commands:
  setup       Download the latest Docker image and setup directories for the container
  start       Start a docker container
  sh          Run a shell in the docker container
  status      Show the status of the docker container
  stop        Stop the docker container
  restart     Restart the docker container

Commands

OME Docker Launcher can be executed in the following format:

• Usage: ./ome_docker_launcher.sh [OPTIONS] COMMAND ...

• Options:
  -h, --help                        Show this help message and exit
  -v, --version                     Show the version and exit
  -d, --debug                       Show debug log
  -b, --hide_banner                 Hide the banner
  -m, --monochrome                  Disable colors

• Commands:
  setup       Download the latest Docker image and setup directories for the container
  start       Start a docker container
  sh          Run a shell in the docker container
  status      Show the status of the docker container
  stop        Stop the docker container
  restart     Restart the docker container

`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.

$ ./ome_docker_launcher.sh setup

 ▄██████▀███▄ 
█████▀ ▄██████  OvenMediaEngine Launcher v0.1
███▄▄▄▄▀▀▀▀███
██████▀ ▄█████  https://github.com/AirenSoft/OvenMediaEngine
 ▀███▄██████▀ 

• Creating configuration directory /usr/share/ovenmediaengine/conf
• Copying configuration to /usr/share/ovenmediaengine/conf
• Copying logs directory
• Copying crash dump directory
• OvenMediaEngine is ready to start!

If you want to change the settings, please modify /usr/share/ovenmediaengine/conf/Server.xml
If you want to start OvenMediaEngine, please run ./ome_docker_launcher.sh start

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.

$ ./ome_docker_launcher.sh start

 ▄██████▀███▄ 
█████▀ ▄██████  OvenMediaEngine Launcher v0.1
███▄▄▄▄▀▀▀▀███
██████▀ ▄█████  https://github.com/AirenSoft/OvenMediaEngine
 ▀███▄██████▀ 

• Starting OvenMediaEngine...
• Obtaining the port list from /usr/share/ovenmediaengine/conf/Server.xml
  - RTMP Provider is configured to use 1935 (Port)
  - SRT Provider is configured to use 9999 (Port)
  - WebRTC Provider is configured to use 3333 (Port)
  - WebRTC Provider is configured to use 3334 (TLSPort)
  - WebRTC Provider is configured to use 10000-10004/UDP (IceCandidate)
  - WebRTC Provider is configured to use 3478 (TcpRelay)
  - OVT Publisher is configured to use 9000 (Port)
  - LLHLS Publisher is configured to use 3333 (Port)
  - LLHLS Publisher is configured to use 3334 (TLSPort)
  - WebRTC Publisher is configured to use 3333 (Port)
  - WebRTC Publisher is configured to use 3334 (TLSPort)
  - WebRTC Publisher is configured to use 10000-10004/UDP (IceCandidate)
  - WebRTC Publisher is configured to use 3478 (TcpRelay)
• Starting a container: ovenemediaengine
  docker> 7235ff9f80762b6e7b27ba3a9773f5584033d55c113340dabf0779e8f5cf53bb
• OvenMediaEngine is started successfully!

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.

$ OME_HOST_IP=1.2.3.4 ./ome_docker_launcher.sh start
...
• OvenMediaEngine is started successfully!

$ tail -f /usr/share/ovenmediaengine/logs/ovenmediaengine.log
...
[2023-11-01 00:00:00.000] I [OvenMediaEngine:1] ICE | ice_port_manager.cpp:305  | ICE candidate found: 1.2.3.4:40000
...

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.

OME_HOST_IP
OME_RTMP_PROV_PORT
OME_WEBRTC_CANDIDATE_IP
OME_WEBRTC_CANDIDATE_PORT
OME_WEBRTC_SIGNALLING_PORT
OME_WEBRTC_SIGNALLING_TLS_PORT
OME_WEBRTC_TCP_RELAY_PORT

`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.

$ ./ome_docker_launcher.sh sh

 ▄██████▀███▄ 
█████▀ ▄██████  OvenMediaEngine Launcher v0.1
███▄▄▄▄▀▀▀▀███
██████▀ ▄█████  https://github.com/AirenSoft/OvenMediaEngine
 ▀███▄██████▀ 

• Run a shell in the running container: ID: 7235ff9f8076
root@7235ff9f8076:/opt/ovenmediaengine/bin# ps -ef
UID          PID    PPID  C STIME TTY          TIME CMD
root           1       0  0 10:29 ?        00:00:01 /opt/ovenmediaengine/bin/OvenMediaEngine -c origin_conf
root          53       0  0 10:44 pts/0    00:00:00 /bin/bash
root          61      53  0 10:44 pts/0    00:00:00 ps -ef
root@7235ff9f8076:/opt/ovenmediaengine/bin# top -bn1
top - 10:44:44 up 333 days,  3:33,  0 users,  load average: 0.44, 0.78, 0.78
Tasks:   3 total,   1 running,   2 sleeping,   0 stopped,   0 zombie
%Cpu(s):  1.4 us,  0.3 sy,  0.0 ni, 98.3 id,  0.0 wa,  0.0 hi,  0.0 si,  0.0 st
MiB Mem : 128723.7 total,  10529.4 free,  31268.5 used,  86925.7 buff/cache
MiB Swap:  31250.0 total,  30345.8 free,    904.2 used.  96221.5 avail Mem 

    PID USER      PR  NI    VIRT    RES    SHR S  %CPU  %MEM     TIME+ COMMAND
      1 root      20   0  320136  21812  15772 S   0.0   0.0   0:01.48 OvenMediaEngine
     53 root      20   0    4116   3456   2896 S   0.0   0.0   0:00.01 bash
     62 root      20   0    5972   3160   2732 R   0.0   0.0   0:00.00 top
root@7235ff9f8076:/opt/ovenmediaengine/bin#

`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.

$ ./ome_docker_launcher.sh status

 ▄██████▀███▄ 
█████▀ ▄██████  OvenMediaEngine Launcher v0.1
███▄▄▄▄▀▀▀▀███
██████▀ ▄█████  https://github.com/AirenSoft/OvenMediaEngine
 ▀███▄██████▀ 

• Container is running: ID: 7235ff9f8076, name: ovenemediaengine

`stop`

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

$ ./ome_docker_launcher.sh stop

 ▄██████▀███▄ 
█████▀ ▄██████  OvenMediaEngine Launcher v0.1
███▄▄▄▄▀▀▀▀███
██████▀ ▄█████  https://github.com/AirenSoft/OvenMediaEngine
 ▀███▄██████▀ 

• Stopping a container: ovenemediaengine
  docker> ovenemediaengine
• Removing a container: ovenemediaengine
  docker> ovenemediaengine
• OvenMediaEngine is stopped successfully

`restart`

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

$ ./ome_docker_launcher.sh stop

 ▄██████▀███▄ 
█████▀ ▄██████  OvenMediaEngine Launcher v0.1
███▄▄▄▄▀▀▀▀███
██████▀ ▄█████  https://github.com/AirenSoft/OvenMediaEngine
 ▀███▄██████▀ 

• Restarting a container: ovenemediaengine
  docker> ovenemediaengine

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.

$ ./ome_docker_launcher.sh -d stop

 ▄██████▀███▄ 
█████▀ ▄██████  OvenMediaEngine Launcher v0.1
███▄▄▄▄▀▀▀▀███
██████▀ ▄█████  https://github.com/AirenSoft/OvenMediaEngine
 ▀███▄██████▀ 

• Stopping a container: ovenemediaengine
 ┌── /usr/bin/docker stop ovenemediaengine
   docker> ovenemediaengine
 └── Succeeded
• Removing a container: ovenemediaengine
 ┌── /usr/bin/docker rm ovenemediaengine
   docker> ovenemediaengine
 └── Succeeded
• OvenMediaEngine is stopped successfully

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.

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:

<Server>
    ...
    <Bind>
        <Providers>
            <RTMP>
              <Port>1935</Port>
            </RTMP>
        </Providers>
    </Bind>
    ...
    <VirtualHosts>
        <VirtualHost>
            <Application>
                <Providers>
                    <RTMP>
                    ...
                </RTMP>
                ...
                </Providers>
            <Application>
        </VirtualHost>
    </VirtualHosts>
</Server>

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:

<Application>
   <Providers>
      <RTMP>
         <BlockDuplicateStreamName>true</BlockDuplicateStreamName>
      </RTMP>
   </Providers>
<Application>

<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 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:

Application

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

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.

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

OBS Studio

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

SRT Socket Options

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

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.

ABR and Transcoding

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

Supported Video, Audio and Image Codecs

Decoders

Type

Codec

Encoders

Type

Codec

Codec of Configuration

OutputProfiles

OutputProfile

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

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

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

Encodes

Video

You can set the video profile as below:

The meaning of each property is as follows:

* 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 WebRTC is as follows.

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

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

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

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

Supported codecs by streaming protocol

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

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

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

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

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

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

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

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:

Low-Latency HLS

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

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

Title

Descriptions

Configuration

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

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

Adaptive Bitrates Streaming (ABR)

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

CrossDomain

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

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

Streaming

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

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

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

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

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

Live Rewind

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

ID3v2 Timed Metadata

Dump

TargetStreamName

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

Playlists

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

OutputPath

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

DRM (beta)

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

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

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

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

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

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

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

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

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

Low-Latency DASH and Legacy HLS streaming

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

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

For legacy HLS, DASH, and LLDASH, refer to the old version manual.

Access Control

Thumbnail

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

Configuration

Bind

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

<Bind>
    <Publishers>
      ...
        <Thumbnail>
            <Port>20080</Port>
            <!-- If you need TLS support, please uncomment below:
            <TLSPort>20081</TLSPort>
            -->
        </Thumbnail>
    </Publishers>
</Bind>

Encoding

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

<OutputProfiles>
    <OutputProfile>
        <Name>default_stream</Name>
        <OutputStreamName>${OriginStreamName}_preview</OutputStreamName>
        <Encodes>
            <Image>
                <Codec>jpeg</Codec>
                <Framerate>1</Framerate>
                <Width>1280</Width>
                <Height>720</Height>
            </Image>
            <Image>
                <Codec>png</Codec>
                <Framerate>1</Framerate>
                <Width>1280</Width>
                <Height>720</Height>
            </Image>
        </Encodes>
    </OutputProfile>
</OutputProfiles>

Publisher

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

<Publishers>
    ...
    <Thumbnail>
        <CrossDomains>
            <Url>*</Url>
        </CrossDomains>	
    </Thumbnail>
</Publishers>

Get thumbnails

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

Recording

OvenMediaEngine can record live streams. You can start and stop recording the output stream through REST API. When the recording is complete, a recording information file is created together with the recorded file so that the user can perform various post-recording processing.

Configuration

File Publisher

To enable recording, add the <FILE> publisher to the configuration file as shown below. <FilePath> and <InfoPath> are required and used as default values. <FilePath> is the setting for the file path and file name. <InfoPath>is the setting for the path and name of the XML file that contains information about the recorded files. If there is no file path value among parameters when requesting recording through API, recording is performed with the set default value. This may be necessary if for security reasons you do not want to specify the file path when calling the API to avoid exposing the server's internal path. <<RootPath> is an optional parameter. It is used when requesting with a relative path is required when requesting an API. also, it is applied to <FilePath> and <InfoPath> as in the example below.

You must specify .ts or .mp4 at the end of the FilePath string to select a container for the recording file. We recommend using .ts unless you have a special case. This is because vp8 and opus codecs are not recorded due to container limitations if you choose .mp4.

<Applications>
  <Application>
      ...
      <Publishers>
        <FILE>
          <RootPath>/mnt/shared_volumes</RootPath>
          <FilePath>/${VirtualHost}/${Application}/${Stream}/
             ${StartTime:YYYYMMDDhhmmss}_${EndTime:YYYYMMDDhhmmss}.ts</FilePath>
          <InfoPath>/${VirtualHost}/${Application}/${Stream}.xml</InfoPath>
        </FILE>
      </Publishers>
      ...

Various macro values are supported for file paths and names as shown below.

Macro Definition

Start & Stop Recording

For control of recording, use the REST API. Recording can be requested based on the output stream name (specified in the JSON body), and all/some tracks can be selectively recorded. And, it is possible to simultaneously record multiple files for the same stream. When recording is complete, an XML file is created at the path specified in InfoPath. For a sample of the recorded file information XML, refer to Appendix A.

For how to use the API, please refer to the link below.

Split Recording

Split recording methods provide interval and schedule. The interval method splits files based on the accumulated recording time. The Schedule method then splits files according to scheduling options based on system time. The scheduling option is the same as the pattern used in crontab. However, only three options are used: seconds/minutes/hour.

interval and schedule methods cannot be used simultaneously.

Appendix A. Recorded File Information Specification

The following is a sample of an XML file that expresses information on a recorded file.

<?xml version="1.0" encoding="utf-8"?>
<files>
  <file>
    <transactionId>bcUCyJeKuOGnsah3</transactionId>
    <id>CTS_ID001</id>
    <vhost>default</vhost>
    <app>app</app>
    <stream>stream_o</stream>
    <filePath><![CDATA[/home/dev/OvenMediaEngine/records/bcUCyJeKuOGnsah3_default_app_stream_o_20201204005351_20201204005405.ts]]></filePath>
    <recordBytes>8774737</recordBytes>
    <recordTime>60011</recordTime>
    <sequence>0</sequence>
    <interval>60000</interval>
    <lastSequence>true</lastSequence>
    <createdTime>2020-12-04T12:53:51.455+0900</createdTime>
    <startTime>2020-12-04T12:53:51.612+0900</startTime>
    <finishTime>2020-12-04T12:54:51.473+0900</finishTime>
  </file>
  <file>
    <transactionId>bcUCyJeKuOGnsah3</transactionId>
    <id>CTS_ID001</id>
    <vhost>default</vhost>
    <app>app</app>
    <stream>stream_o</stream>
    <filePath><![CDATA[/home/dev/OvenMediaEngine/records/bcUCyJeKuOGnsah3_default_app_stream_o_20201204005408_20201204005412.ts]]></filePath>
    <recordBytes>2285797</recordBytes>
    <recordTime>60012</recordTime>
    <sequence>0</sequence>
    <schedule>0 */1 *</schedule>
    <lastSequence>false</lastSequence>
    <createdTime>2020-12-04T12:53:00.000+0900</createdTime>
    <startTime>2020-12-04T12:53:00.000+0900</startTime>
    <finishTime>2020-12-04T12:54:00.000+0900</finishTime>
  </file>
  <file>
    <transactionId>bcUCyJeKuOGnsah3</transactionId>
    <id>CTS_ID001</id>
    <vhost>default</vhost>
    <app>app</app>
    <stream>stream_o</stream>
    <filePath><![CDATA[/home/dev/OvenMediaEngine/records/bcUCyJeKuOGnsah3_default_app_stream_o_20201204005415_20201204005422.ts]]></filePath>
    <recordBytes>4544626</recordBytes>
    <recordTime>60000</recordTime>
    <sequence>1</sequence>
    <schedule>0 */1 *</schedule>
    <lastSequence>true</lastSequence>
    <createdTime>2020-12-04T12:54:00.000+0900</createdTime>
    <startTime>2020-12-04T12:54:00.000+0900</startTime>
    <finishTime>2020-12-04T12:55:00.000+0900</finishTime>
  </file>
</files>

Push Publishing

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

Configuration

RTMPPush Publisher

To use RTMP Push Publishing, you need to declare the <RTMPPush> publisher in the configuration. There are no other detailed options.

<Applications>
  <Application>
     ...
    <Publishers>
      ... 
      <RTMPPush>
      </RTMPPush>
    </Publishers>
  </Application>
</Applications>

MPEGTSPush Publisher

To use MPEGTS Push Publishing, you need to declare the <MPEGTSPush> publisher in the configuration. There are no other detailed options.

<Applications>
  <Application>
     ...
    <Publishers>
      ...
      <MPEGTSPush>
      </MPEGTSPush>
    </Publishers>
  </Application>
</Applications>

Only H264 and AAC are supported codecs.

SRTPush Publisher

To use SRT Push Publishing, you need to declare the <SRTPush> publisher in the configuration. There are no other detailed options.

<Applications>
  <Application>
     ...
    <Publishers>
      ...
      <SRTPush>
      </SRTPush>
    </Publishers>
  </Application>
</Applications>

Only H264 and AAC are supported codecs.

Start & Stop Push

For control of push, use the REST API. SRT, RTMP, MPEGTS push can be requested based on the output stream name (specified in the JSON body), and you can selectively transfer all/some tracks. In addition, you must specify the URL and Stream Key of the external server to be transmitted. It can send multiple Pushes simultaneously for the same stream. If transmission is interrupted due to network or other problems, it automatically reconnects.

For how to use the API, please refer to the link below.

REST API

Overview

The REST APIs provided by OME allow you to query or change settings such as VirtualHost and Application/Stream.

There are some limitations/considerations.

If you add/change/delete the settings of the App/Output Profile by invoking the API, the app will be restarted. This means that all sessions connected to the app will be disconnected.
VirtualHost settings in Server.xml cannot be modified through API. This rule also applies to Application/OutputStream, etc. within that VirtualHost. So, if you call a POST/PUT/DELETE API for VirtualHost/Application/OutputProfile declared in Server.xml, it will not work with a 403 Forbidden error.

By default, OvenMediaEngine's API Server is disabled, so the following settings are required to use the API.

Setup API Server

Port Binding

The API server's port can be set in <Bind><Managers><API>. <Port> is an unsecured port and <TLSPort> is a secured port. To use TLSPort, TLS certificate must be set in the Managers.

<Server version="8">
	...
	<Bind>
		<Managers>
			<API>
				<Port>8081</Port>
				<TLSPort>8082</TLSPort>
			</API>
		</Managers>
		...
	</Bind>
	...
</Server>

Managers

In order to use the API server, you must configure <Managers> as well as port binding.

<Server version="8">
	<Bind>
		...
	</Bind>

	<Managers>
		<Host>
			<Names>
				<Name>*</Name>
			</Names>
			<TLS>
				<CertPath>airensoft_com.crt</CertPath>
				<KeyPath>airensoft_com.key</KeyPath>
				<ChainCertPath>airensoft_com_chain.crt</ChainCertPath>
			</TLS>
		</Host>
		<API>
			<AccessToken>your_access_token</AccessToken>
			<CrossDomains>
				<Url>*.airensoft.com</Url>
				<Url>http://*.sub-domain.airensoft.com</Url>
				<Url>http?://airensoft.*</Url>
			</CrossDomains>
		</API>
	</Managers>

	<VirtualHosts>
		...
	</VirtualHosts>
</Server>

Host

In <Names>, set the domain or IP that can access the API server. If * is set, any address is used. In order to access using the TLS Port, a certificate must be set in <TLS>.

AccessToken

API Server uses Basic HTTP Authentication Scheme to authenticate clients. An AccessToken is a plaintext credential string before base64 encoding. Setting the AccessToken to the form user-id:password per RFC7617 allows standard browsers to pass authentication, but it is not required.

For more information about HTTP Basic authentication, refer to the URL below.

https://developer.mozilla.org/en-US/docs/Web/HTTP/Authentication

CrossDomains

To enable CORS on your API Server, you can add a setting. You can add * to allow all domains. If contains a scheme, such as https://, only that scheme can be allowed, or if the scheme is omitted, such as *.airensoft.com, all schemes can be accepted.

API Request

API endpoints are provided in the following format.

Method http://API.Server.Address[:Port]/v1/Resource
Method https://API.Server.Address[:TLSPort]/v1/Resource

OvenMediaEngine supports GET, POST, and DELETE methods, and sometimes supports PATCH depending on the type of resource. For detailed API specifications, please check the subdirectory of this chapter.

Action

In OvenMediaEngine's REST API, action is provided in the following format.

POST http://host/v1/resource:{action name}

For example, an action to send an ID3 Timedmeta event to an LLHLS stream is provided by the endpoint below.

POST http://-/v1/vhosts/{vhost}/apps/{app}/streams/{stream}:sendEvent

Document format

In this API reference document, the API endpoint is described as follows. Note that scheme://Host[:Port] is omitted for all endpoints.

METHOD /v1/<API_PATH>

Header

Describe the required header values.

Header-Key: Value

# Header-Key
    Description

Body

Describe the request body content. The body of all APIs consists of Json content. Therefore, the Content-Type header value is always application/json, which can be omitted in the document.

{
    "requestId": "value"
}
    
# key (required)
    The description of the key/value of the body content is provided like this.

Responses from API endpoints are provided in the following format.

HTTP_Status_Code Code_Description

Header

Description of response headers

Header-Key: Value

Body

Description the response body content. The body of all response consists of Json content. Therefore, the Content-Type header value is always application/json, which can be omitted in the document.

{
	"statusCode": 200,
	"message": "OK",
	"response": {
	}
}

# statusCode
	Same as HTTP Status Code
# message
	A human-readable description of the response code
# response
	Response Contents

v1

VirtualHost

Get Virtual Host List

Request

GET /v1/vhosts

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

{
	"statusCode": 200,
	"message": "OK",
	"response": [
		"default",
		"service",
		"poc"
	]
}

# statusCode
	Same as HTTP Status Code
# message
	A human-readable description of the response code
# response
	Json array containing a list of virtual host names

401 Unauthorized

Authentication required

Header

WWW-Authenticate: Basic realm=”OvenMediaEngine”

Body

{
    "message": "[HTTP] Authorization header is required to call API (401)",
    "statusCode": 401
}

Create Virtual Host

Request

POST /v1/vhosts

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Body

Configure virtual hosts to be created in Json array format.

[
    {
        "name": "vhost",
        "host": {
            "names": [
                "ome-dev.airensoft.com",
                "prod.airensoft.com"
            ],
            "tls": {
                "certPath": "/etc/pki/airensoft.com/_airensoft_com.crt",
                "chainCertPath": "/etc/pki/airensoft.com/_airensoft_com.ca-bundle",
                "keyPath": "/etc/pki/airensoft.com/_airensoft_com.key"
            }
        },

        "signedPolicy": {
            "enables": {
                "providers": "rtmp,webrtc,srt",
                "publishers": "webrtc,llhls"
            },
            "policyQueryKeyName": "policy",
            "secretKey": "aKq#1kj",
            "signatureQueryKeyName": "signature"
        },

        "admissionWebhooks": {
            "controlServerUrl": "https://control.server/admission",
            "enables": {
                "providers": "rtmp,webrtc,srt",
                "publishers": "webrtc,llhls"
            },
            "secretKey": "",
            "timeout": 3000
        },
        
        "origins": {
            "origin": [
                {
                    "location": "/app/rtsp",
                    "pass": {
                        "scheme": "rtsp",
                        "urls": {
                            "url": [
                                "rtsp.server:8554/ca-01"
                            ]
                        }
                    }
                }
            ]
        },

        "originMapStore": {
            "originHostName": "ome-dev.airensoft.com",
            "redisServer": {
                "auth": "!@#ovenmediaengine",
                "host": "redis.server:6379"
            }
        }
    },
    {
        "name": "vhost2",
        "host": {
            "names": [
                "ovenmediaengine.com"
            ],
            "tls": {
                "certPath": "/etc/pki/ovenmediaengine.com/_ovenmediaengine_com.crt",
                "chainCertPath": "/etc/pki/ovenmediaengine.com/_ovenmediaengine_com.ca-bundle",
                "keyPath": "/etc/pki/ovenmediaengine.com/_ovenmediaengine_com.key"
            }
        }
    }
]

# name (required)
    The virtual host name. Cannot be duplicated.

# host (required)
    ## names (required)
        The addresses(IP or Domain)of the host. 
    ## tls (optional)
        The certificate file path. Required if using TLS. 
        
# signedPolicy (optional)
    The SignedPolicy setting. Please refer to the manual for details.
    
# admissionWebhooks (optional)
    The AdmissionWebhooks setting. Please refer to the manual for details.
    
# origins (optional)
    The Origins setting. Please refer to the manual for details.

# originMapStore (optional)
    The OriginMapStore setting. Please refer to the manual for details.

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

It responds with Json array for each request.

[
    {
        "message": "OK",
        "statusCode": 200,
        "response": {
            "name": "enterprise",

            "host": {
                "names": [
                    "ome-dev.airensoft.com",
                    "prod.airensoft.com"
                ],
                "tls": {
                    "certPath": "/etc/pki/airensoft.com/_airensoft_com.crt",
                    "chainCertPath": "/etc/pki/airensoft.com/_airensoft_com.ca-bundle",
                    "keyPath": "/etc/pki/airensoft.com/_airensoft_com.key"
                }
            },
            "signedPolicy": {
                "enables": {
                    "providers": "rtmp,webrtc,srt",
                    "publishers": "webrtc,llhls"
                },
                "policyQueryKeyName": "policy",
                "secretKey": "aKq#1kj",
                "signatureQueryKeyName": "signature"
            },
            "admissionWebhooks": {
                "controlServerUrl": "https://control.server/admission",
                "enables": {
                    "providers": "rtmp,webrtc,srt",
                    "publishers": "webrtc,llhls"
                },
                "secretKey": "",
                "timeout": 3000
            },
            "origins": {
                "origin": [
                    {
                        "location": "/app/rtsp",
                        "pass": {
                            "scheme": "rtsp",
                            "urls": {
                                "url": [
                                    "rtsp.server:8554/ca-01"
                                ]
                            }
                        }
                    }
                ]
            },
            "originMapStore": {
                "originHostName": "ome-dev.airensoft.com",
                "redisServer": {
                    "auth": "!@#ovenmediaengine",
                    "host": "redis.server:6379"
                }
            }
        }
    },
    {
        "message": "OK",
        "statusCode": 200,
        "response": {
            "name": "free",
            "host": {
                "names": [
                    "ovenmediaengine.com"
                ],
                "tls": {
                    "certPath": "/etc/pki/ovenmediaengine.com/_ovenmediaengine_com.crt",
                    "chainCertPath": "/etc/pki/ovenmediaengine.com/_ovenmediaengine_com.ca-bundle",
                    "keyPath": "/etc/pki/ovenmediaengine.com/_ovenmediaengine_com.key"
                }
            }
        }
    }
]

# statusCode
	Same as HTTP Status Code
# message
	A human-readable description of the response code
# response
	Created virtual host information

207 Multi-Status

There might be a mixture of responses.

Header

Content-Type: application/json

Body

It responds with Json array for each request.

[
    {
        "statusCode": 200,
        "message": "OK",
        "response": {
            "name": "enterprise",
            "host": {
                "names": [
            ...
    },
    {
        "statusCode": 409,
        "message": "Conflict",
        "response": {
            ...
        }
    }
}

# statusCode
    Same as HTTP Status Code
# message
    A human-readable description of the response code
# response
    Virtual host information created when statusCode is 200

400 Bad Request

Invalid request. Body is not a Json array or does not have a required value

401 Unauthorized

Authentication required

Header

WWW-Authenticate: Basic realm=”OvenMediaEngine”

Body

{
    "message": "[HTTP] Authorization header is required to call API (401)",
    "statusCode": 401
}

409 Conflict

A virtual host with that name already exists

Get Virtual Host Information

Request

GET /v1/vhosts/{vhost}

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body


    "message": "OK",
    "statusCode": 200
    "response": {
        "name": "default",
        "distribution": "ovenServer",

    "host": {
        "name": "default",
        "distribution": "ovenServer",

        "host": {
            "names": [
                "ome-dev.airensoft.com",
                "*"
            ],
            "tls": {
                "certPath": "/etc/pki/airensoft.com/_airensoft_com.crt",
                "chainCertPath": "/etc/pki/airensoft.com/_airensoft_com.ca-bundle",
                "keyPath": "/etc/pki/airensoft.com/_airensoft_com.key"
            }
        },
        
        "signedPolicy": {
            "enables": {
                "providers": "rtmp,webrtc,srt",
                "publishers": "webrtc,llhls"
            },
            "policyQueryKeyName": "policy",
            "secretKey": "aKq#1kj",
            "signatureQueryKeyName": "signature"
        },
        
        "admissionWebhooks": {
            "controlServerUrl": "https://control.server/admission",
            "enables": {
                "providers": "rtmp,webrtc,srt",
                "publishers": "webrtc,llhls"
            },
            "secretKey": "",
            "timeout": 3000
        },
        
        "origins": {
            "origin": [
                {
                    "location": "/app/rtsp",
                    "pass": {
                        "scheme": "rtsp",
                        "urls": {
                            "url": [
                                "rtsp.server:8554/ca-01"
                            ]
                        }
                    }
                }
            ]
        },

        "originMapStore": {
            "originHostName": "ome-dev.airensoft.com",
            "redisServer": {
                "auth": "!@#ovenmediaengine",
                "host": "redis.server:6379"
            }
        }
    }

# statusCode
	Same as HTTP Status Code
# message
	A human-readable description of the response code
# response
	Virtual host information

401 Unauthorized

Authentication required

Header

WWW-Authenticate: Basic realm=”OvenMediaEngine”

Body

{
    "message": "[HTTP] Authorization header is required to call API (401)",
    "statusCode": 401
}

404 Not Found

The given vhost name could not be found.

Body

{
    "message": "[HTTP] Could not find the virtual host: [default1] (404)",
    "statusCode": 404
}

Delete Virtual Host

Request

DELETE /v1/vhosts/{vhost}

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

{
    "message": "OK",
    "statusCode": 200
}

# statusCode
	Same as HTTP Status Code
# message
	A human-readable description of the response code

401 Unauthorized

Authentication required

Header

WWW-Authenticate: Basic realm=”OvenMediaEngine”

Body

{
    "message": "[HTTP] Authorization header is required to call API (401)",
    "statusCode": 401
}

404 Not Found

The given vhost name could not be found.

Body

{
    "message": "[HTTP] Could not find the virtual host: [default1] (404)",
    "statusCode": 404
}

500 Internal Server Error

The request failed due to an error on the server. Check the server log for the reason of the error.

Body

{
    "message": "[HTTP] Internal Server Error (500)",
    "statusCode": 500
}

Reload Certificate

Reload All Certificates

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

Request

POST /v1/vhosts:reloadAllCertificates

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

{
    "message": "OK",
    "statusCode": 200
}

# statusCode
	Same as HTTP Status Code
# message
	A human-readable description of the response code

500 Internal Server Error

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

Reload Certificate

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

Request

POST /v1/vhosts/{vhost}:reloadCertificate

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

{
    "message": "OK",
    "statusCode": 200
}

# statusCode
    Same as HTTP Status Code
# message
    A human-readable description of the response code

500 Internal Server Error

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

Output Profile

Get Output Profile List

Request

GET /v1/vhosts/{vhost}/apps/{app}/outputProfiles

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

{
	"statusCode": 200,
	"message": "OK",
	"response": [
		"default",
		"audio_only"
	]
}

# statusCode
	Same as HTTP Status Code
# message
	A human-readable description of the response code
# response
	Json array containing a list of output profile names

401 Unauthorized

Authentication required

Header

WWW-Authenticate: Basic realm=”OvenMediaEngine”

Body

{
    "message": "[HTTP] Authorization header is required to call API (401)",
    "statusCode": 401
}

404 Not Found

The given vhost or application name could not be found.

Body

{
    "message": "[HTTP] Could not find the application: [vhost/app1] (404)",
    "statusCode": 404
}

Create Output Profile

Add an Output Profile to the Application. If this request succeeds, the application will be restarted.

Request

POST /v1/vhosts/{vhost}/apps/{app}/outputProfiles

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Body

Configure output profiles to be created in Json array format.

[
  {
    "name": "bypass_profile",
    "outputStreamName": "${OriginStreamName}_bypass",
    "encodes": {
      "videos": [
        {
          "bypass": true
        }
      ],
      "audios": [
        {
          "bypass": true
        }
      ]
    }
  }
]

# name (required)
  The name of the output profile. cannot be duplicated

# outputStreamName (required)
  The name of the output stream to be created through this profile.
  
# encodes (required)
  encode profile list. It must have videos or audios, and must have at least one item.

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

It responds with Json array for each request.

[
    {
        "statusCode": 200,
        "message": "OK",
        "response": {
            "name": "bypass_profile",
            "outputStreamName": "${OriginStreamName}_bypass",
            "encodes": {
                "audios": [],
                "videos": [
                    {
                        "bypass": true
                    }
                ]
            }
    },
    {
        "statusCode": 200,
        "message": "OK",
        "response": {
            ...
        }
    }
}

# statusCode
    Same as HTTP Status Code
# message
    A human-readable description of the response code
# response
    Created output profile information

207 Multi-Status

There might be a mixture of responses.

Header

Content-Type: application/json

Body

It responds with Json array for each request.

[
    {
        "statusCode": 200,
        "message": "OK",
        "response": {
            "name": "app",
            "outputProfiles": {
            ...

            "providers": {
                "ovt": {},
                "rtmp": {},
            ...
    },
    {
        "statusCode": 409,
        "message": "Conflict",
        "response": {
            ...
        }
    }
}

# statusCode
    Same as HTTP Status Code
# message
    A human-readable description of the response code
# response
    Output profile information created when statusCode is 200

400 Bad Request

Invalid request. Body is not a Json array or does not have a required value

401 Unauthorized

Authentication required

Header

WWW-Authenticate: Basic realm=”OvenMediaEngine”

Body

{
    "message": "[HTTP] Authorization header is required to call API (401)",
    "statusCode": 401
}

404 Not Found

The given vhost or application name could not be found.

Body

{
    "message": "[HTTP] Could not find the application: [vhost/app1] (404)",
    "statusCode": 404
}

409 Conflict

A output profile name already exists

Get Output Profile Information

Request

GET /v1/vhosts/{vhost}/apps/{app}/outputProfiles/{profile}

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

{
    "statusCode": 200,
    "message": "OK",
    "response": {
        "name": "bypass_profile2",
        "outputStreamName": "${OriginStreamName}_bypass"
        "encodes": {
            "audios": [],
            "videos": [
                {
                    "bypass": true
                }
            ]
        }
    }
}

# statusCode
	Same as HTTP Status Code
# message
	A human-readable description of the response code
# response
	Output Profile information

401 Unauthorized

Authentication required

Header

WWW-Authenticate: Basic realm=”OvenMediaEngine”

Body

{
    "message": "[HTTP] Authorization header is required to call API (401)",
    "statusCode": 401
}

404 Not Found

The given vhost or application name could not be found.

Body

{
    "message": "[HTTP] Could not find the application: [vhost/app1] (404)",
    "statusCode": 404
}

Delete Output Profile

Delete output profile settings. If this request succeeds, the Application will be restarted.

Request

DELETE /v1/vhosts/{vhost}/apps/{app}/outputProfiles/{profile}

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

{
    "message": "OK",
    "statusCode": 200
}

# statusCode
	Same as HTTP Status Code
# message
	A human-readable description of the response code

401 Unauthorized

Authentication required

Header

WWW-Authenticate: Basic realm=”OvenMediaEngine”

Body

{
    "message": "[HTTP] Authorization header is required to call API (401)",
    "statusCode": 401
}

404 Not Found

The given vhost or application name could not be found.

Body

{
    "message": "[HTTP] Could not find the application: [vhost/app1] (404)",
    "statusCode": 404
}

Record

Start Recording

Start recording the stream. If the requested stream does not exist on the server, this recording task is reserved. And when the stream is created, it automatically starts recording.

Request

POST /v1/vhosts/{vhost}/apps/{app}:startRecord

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Body : Output the recorded video to a single file

{
    "id": "{unique_record_id}",
    "stream": {
        "name": "{output_stream_name}",
        "variantNames": []
    }
}

# id (required)
    unique ID to identify the recording task
    
# stream (required)
    ## name (required)
        output stream name
        
    ## variantNames (optional)
        Array of track names to record. If empty, all tracks will be 
        recorded. This value is Encodes.[Video|Audio|Data].Name in the
        OutputProfile setting.

Body : Output the recorded video to a file at intervals

{
  "id": "{unique_record_id}",
  "stream": {
    "name": "{output_stream_name}"
  },
  "interval": 60000,
  "segmentationRule": "discontinuity"
}

# id (required)
    unique ID to identify the recording task
    
# stream (required)
    ## name (required)
        output stream name
        
    ## variantNames (optional)
        Array of track names to record. If empty, all tracks will be 
        recorded. This value is Encodes.[Video|Audio|Data].Name in the
        OutputProfile setting.

# interval (optional)
    Recording time per file (milliseconds). Not allowed to use with schedule
    
# segmentationRule (optional)
    Define the policy for continuously or discontinuously timestamp 
    in divided recorded files.
    
    continuity : timestamp of recorded files is continuous
    discontinuity(default) : timestamp starts anew for each recorded file

Body : Output the recorded video to a file at the scheduled time

{
  "id": "{unique_record_id}",
  "stream": {
    "name": "{output_stream_name}"
  },
  "schedule" : "0 */1 *"
  "segmentationRule": "continuity"
}

# id (required)
    unique ID to identify the recording task
    
# stream (required)
    ## name (required)
        output stream name
        
    ## variantNames (optional)
        Array of track names to record. If empty, all tracks will be 
        recorded. This value is Encodes.[Video|Audio|Data].Name in the
        OutputProfile setting.

# schedule (optional)
    <Second Minute Hour> format, same as crontab syntax
    "10 */1 *" means to output the recorded file every 10 minutes of the hour
    Not allowed to use with schedule
    
# segmentationRule (optional)
    Define the policy for continuously or discontinuously timestamp 
    in divided recorded files.
    
    continuity : timestamp of recorded files is continuous
    discontinuity(default) : timestamp starts anew for each recorded file

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

Please note that responses are incorrectly returned in Json array format for version 0.15.3 and earlier.

{
    "statusCode": 200,
    "message": "OK",
    "response": {
        "id": "2",
        "state": "ready",
        "vhost": "default",
        "app": "app",
        "stream": {
            "name": "stream",
            "trackIds": [],
            "variantNames": []
        },
        "interval": 60000,
        "segmentationRule": "discontinuity",
        "createdTime": "2023-03-15T21:15:20.113+09:00",
    }
}

# statusCode
	Same as HTTP Status Code
# message
	A human-readable description of the response code
# response
	Created recording task information

400 Bad Request

Invalid request.

401 Unauthorized

Authentication required

Header

WWW-Authenticate: Basic realm=”OvenMediaEngine”

Body

{
    "message": "[HTTP] Authorization header is required to call API (401)",
    "statusCode": 401
}

404 Not Found

The given vhost or application name could not be found.

Body

{
    "message": "[HTTP] Could not find the application: [vhost/app1] (404)",
    "statusCode": 404
}

409 Conflict

duplicate ID

Stop Recording

Request

POST /v1/vhosts/{vhost}/apps/{app}:stopRecord

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Body

{
    "id": "{unique_record_id}"
}

# id (required)
    unique ID to identify the recording task

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

{
    "statusCode": 200,
    "message": "OK",
}

# statusCode
	Same as HTTP Status Code
# message
	A human-readable description of the response code

400 Bad Request

Invalid request.

401 Unauthorized

Authentication required

Header

WWW-Authenticate: Basic realm=”OvenMediaEngine”

Body

{
    "message": "[HTTP] Authorization header is required to call API (401)",
    "statusCode": 401
}

404 Not Found

The given vhost/application name or id of recording task could not be found.

Body

{
    "message": "[HTTP] Could not find the application: [vhost/app1] (404)",
    "statusCode": 404
}

Get Recording State

Request

POST /v1/vhosts/{vhost}/apps/{app}:records

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Body

{
    "id": "{unique_record_id}"
}

# id (optional)
    unique ID to identify the recording task. If no id is given in the request, the full list is returned.

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

The response is Json array format.

{
    "statusCode": 200,
    "message": "OK",
    "response": [
        {
            "id": "2",
            "state": "recording",
            "vhost": "default",
            "app": "app",
            "stream": {
                "name": "stream",
                "trackIds": [],
                "variantNames": []
            },
            "interval": 60000,
            "segmentationRule": "discontinuity",
            "createdTime": "2023-03-15T21:15:20.113+09:00",
        },
        {
            "id": "3",
            ...
        }
    ]
}

# statusCode
	Same as HTTP Status Code
# message
	A human-readable description of the response code
# response
	Information of recording tasks. If there is no recording task, 
	response with empty array ("response": [])

401 Unauthorized

Authentication required

Header

WWW-Authenticate: Basic realm=”OvenMediaEngine”

Body

{
    "message": "[HTTP] Authorization header is required to call API (401)",
    "statusCode": 401
}

404 Not Found

The given vhost or application name could not be found.

Body

{
    "message": "[HTTP] Could not find the application: [vhost/app1] (404)",
    "statusCode": 404
}

State of Recording

The Recording task has the state shown in the table below. You can get the state in the Start Recording and Get Recording State API response.

Push

Start Push Publishing

Start push publishing the stream with SRT, RTMP or MPEG2-TS. If the requested stream does not exist on the server, this task is reserved. And when the stream is created, it automatically starts push publishing.

Request

POST /v1/vhosts/{vhost}/apps/{app}:startPush

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Body : SRT

{
  "id": "{unique_push_id}",
  "stream": {
    "name": "{output_stream_name}",
    "variantNames": []
  },
  "protocol": "srt",
  "url": "srt://{host}[:port]?mode=caller&latency=120000&timeout=500000",
  "streamKey": ""
}

# id (required)
    unique ID to identify the task
    
# stream (required)
    ## name (required)
        output stream name
        
    ## variantNames (optional)
        Array of track names to publsh. 
        This value is Encodes.[Video|Audio|Data].Name in the OutputProfile
        setting.
        
        If empty, all tracks will be sent.

# protocol (required)
    srt
    
# url (required) 
    address of destination.
    options can be set in query-string format.
    
# streamKey (optional)
    not used with mpegts

In SRT Push Publisher, only the caller connection mode is supported.

Body : RTMP

{
  "id": "{unique_push_id}",
  "stream": {
    "name": "{output_stream_name}",
    "variantNames": [ "h264_fhd", "aac" ]
  },
  "protocol": "rtmp",
  "url":"rtmp://{host}[:port]/{app_name}",
  "streamKey":"{stream_name}"
}

# id (required)
    unique ID to identify the task
    
# stream (required)
    ## name (required)
        output stream name
        
    ## variantNames (optional)
        Array of track names to publsh. 
        This value is Encodes.[Video|Audio|Data].Name in the OutputProfile
        setting.
        
        If empty, The first track among video tracks (by ID) and the first 
        track among audio tracks (by ID) are selected automatically.

# protocol (required)
    rtmp
    
# url (required) 
    address of destination
    
# streamKey (required)
    RTMP stream key

Body : MPEG2-TS

{
  "id": "{unique_push_id}",
  "stream": {
    "name": "{output_stream_name}",
    "variantNames": []
  },
  "protocol": "mpegts",
  "url": "udp://{host}[:port]",
  "streamKey": ""
}

# id (required)
    unique ID to identify the task
    
# stream (required)
    ## name (required)
        output stream name
        
    ## variantNames (optional)
        Array of track names to publsh. 
        This value is Encodes.[Video|Audio|Data].Name in the OutputProfile
        setting.
        
        If empty, all tracks will be sent.

# protocol (required)
    mpegts
    
# url (required) 
    address of destination
    
# streamKey (optional)
    not used with mpegts

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

Please note that responses are incorrectly returned in Json array format for version 0.15.3 and earlier.

{
    "statusCode": 200,
    "message": "OK",
    "response": {
        "id": "{unique_push_id}",
        "state": "ready",
            
        "vhost": "default",
        "app": "app",
        "stream": {
            "name": "{output_stream_name}",
            "trackIds": [],
            "variantNames": []
        },
            
        "protocol": "rtmp",
        "url": "rtmp://{host}[:port]/{app_name}",
        "streamKey": "{stream_name}",
            
        "sentBytes": 0,
        "sentTime": 0,
        "sequence": 0,
        "totalsentBytes": 0,
        "totalsentTime": 0,
            
        "createdTime": "2023-03-15T23:02:34.371+09:00",
        "startTime": "1970-01-01T09:00:00.000+09:00",
        "finishTime": "1970-01-01T09:00:00.000+09:00"
    }
}

# statusCode
    Same as HTTP Status Code
# message
    A human-readable description of the response code
# response
    Created push publishing task information

400 Bad Request

Invalid request.

401 Unauthorized

Authentication required

Header

WWW-Authenticate: Basic realm=”OvenMediaEngine”

Body

{
    "message": "[HTTP] Authorization header is required to call API (401)",
    "statusCode": 401
}

404 Not Found

The given vhost or application name could not be found.

Body

{
    "message": "[HTTP] Could not find the application: [vhost/app1] (404)",
    "statusCode": 404
}

409 Conflict

duplicate ID

Stop Push Publishing

Request

POST /v1/vhosts/{vhost}/apps/{app}:stopPush

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Body

{
    "id": "{unique_push_id}"
}

# id (required)
    unique ID to identify the push publishing task

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

{
    "statusCode": 200,
    "message": "OK",
}

# statusCode
	Same as HTTP Status Code
# message
	A human-readable description of the response code

400 Bad Request

Invalid request.

401 Unauthorized

Authentication required

Header

WWW-Authenticate: Basic realm=”OvenMediaEngine”

Body

{
    "message": "[HTTP] Authorization header is required to call API (401)",
    "statusCode": 401
}

404 Not Found

The given vhost/application name or id of recording task could not be found.

Body

{
    "message": "[HTTP] Could not find the application: [vhost/app1] (404)",
    "statusCode": 404
}

Get Push Publishing State

Request

POST /v1/vhosts/{vhost}/apps/{app}:pushes

Header

Authorization: Basic {credentials}

# Authorization
    Credentials for HTTP Basic Authentication created with <AccessToken>

Body

{
    "id": "{unique_push_id}"
}

# id (optional)
    unique ID to identify the push publishing task. If no id is given in the request, the full list is returned.

Responses

200 Ok

The request has succeeded

Header

Content-Type: application/json

Body

The response is Json array format.

{
    "statusCode": 200,
    "message": "OK",
    "response": [
        {
            "id": "{unique_push_id}",
            "state": "started",
            
            "vhost": "default",
            "app": "app",
            "stream": {
                "name": "{output_stream_name}",
                "trackIds": [],
                "variantNames": []
            },
            
            "protocol": "rtmp",
            "url": "rtmp://{host}[:port]/{app_name}",
            "streamKey": "{stream_name}",
            
            "sentBytes": 0,
            "sentTime": 0,
            "sequence": 0,
            "totalsentBytes": 0,
            "totalsentTime": 0,
            
            "createdTime": "2023-03-15T23:02:34.371+09:00",
            "startTime": "1970-01-01T09:00:00.000+09:00",
            "finishTime": "1970-01-01T09:00:00.000+09:00"
        },
        {
            "id": "4",
            ...
        }
    ]
}

# statusCode
	Same as HTTP Status Code
# message
	A human-readable description of the response code
# response
	Information of recording tasks. If there is no recording task, 
	response with empty array ("response": [])

401 Unauthorized

Authentication required

Header

WWW-Authenticate: Basic realm=”OvenMediaEngine”

Body

{
    "message": "[HTTP] Authorization header is required to call API (401)",
    "statusCode": 401
}

404 Not Found

The given vhost or application name could not be found.

Body

{
    "message": "[HTTP] Could not find the application: [vhost/app1] (404)",
    "statusCode": 404
}

State of Push Publishing

The Push Publishing task has the state shown in the table below. You can get the state in the Start Push Publishing and Get Push Publishing State API response.

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.

/usr/share/ovenmediaengine/conf/Server.xml

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

/<OvenMediaEngine Binary Path>/conf/Server.xml

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

# For Origin mode
/opt/ovenmediaengine/bin/origin_conf/Server.xml
# For Edge mode
/opt/ovenmediaengine/bin/edge_conf/Server.xml

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.

<?xml version="1.0" encoding="UTF-8"?>
<Server version="8">
    <Name>OvenMediaEngine</Name>
    <IP>*</IP>
    <PrivacyProtection>false</PrivacyProtection>
    <StunServer>stun.l.google.com:19302</StunServer>
    <Bind>...</Bind>
    <VirtualHosts>...</VirtualHosts>
</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

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.

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

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.

<!-- 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

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.

<?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 SignedPolicy chapter for more information.

Origins

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

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

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

<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.

<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

<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>.

<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 Transcoding 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 Live Source 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 />
      <HLS />
      <DASH />
      <LLDASH />
      <WebRTC />
   </Publishers>
</Application>

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

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

Configuration Example

Finally, Server.xml is configured as follows:

<?xml version="1.0" encoding="UTF-8"?>

<Server version="8">
    <Name>OvenMediaEngine</Name>
    <!-- Host type (origin/edge) -->
    <Type>origin</Type>
    <!-- Specify IP address to bind (* means all IPs) -->
    <IP>*</IP>
    <PrivacyProtection>false</PrivacyProtection>

    <!-- 
    To get the public IP address(mapped address of stun) of the local server. 
    This is useful when OME cannot obtain a public IP from an interface, such as AWS or docker environment. 
    If this is successful, you can use ${PublicIP} in your settings.
    -->
    <StunServer>stun.l.google.com:19302</StunServer>

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

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

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

<!-- Settings for the ports to bind -->
<Bind>
    <!-- Enable this configuration if you want to use API Server -->
    <!--
    <Managers>
        <API>
            <Port>8081</Port>
            <TLSPort>8082</TLSPort>
            <WorkerCount>1</WorkerCount>
        </API>
    </Managers>
    -->

    <Providers>
        <!-- Pull providers -->
        <RTSPC>
            <WorkerCount>1</WorkerCount>
        </RTSPC>
        <OVT>
            <WorkerCount>1</WorkerCount>
        </OVT>
        <!-- Push providers -->
        <RTMP>
            <Port>1935</Port>
            <WorkerCount>1</WorkerCount>
        </RTMP>
        <SRT>
            <Port>9999</Port>
            <WorkerCount>1</WorkerCount>
        </SRT>
        <MPEGTS>
            <!--
                Listen on port 4000~4005 (<Port>4000-4004,4005/udp</Port>)
                This is just a demonstration to show that you can configure the port in several ways
            -->
            <Port>4000/udp</Port>
        </MPEGTS>
        <WebRTC>
            <Signalling>
                <Port>3333</Port>
                <TLSPort>3334</TLSPort>
                <WorkerCount>1</WorkerCount>
            </Signalling>

            <IceCandidates>
                <IceCandidate>*:10000/udp</IceCandidate>
                <!-- 
                    If you want to stream WebRTC over TCP, specify IP:Port for TURN server.
                    This uses the TURN protocol, which delivers the stream from the built-in TURN server to the player's TURN client over TCP. 
                    For detailed information, refer https://airensoft.gitbook.io/ovenmediaengine/streaming/webrtc-publishing#webrtc-over-tcp
                -->
                <TcpRelay>*:3478</TcpRelay>
                <!-- TcpForce is an option to force the use of TCP rather than UDP in WebRTC streaming. (You can omit ?transport=tcp accordingly.) If <TcpRelay> is not set, playback may fail. -->
                <TcpForce>true</TcpForce>
                <TcpRelayWorkerCount>1</TcpRelayWorkerCount>
            </IceCandidates>
        </WebRTC>
    </Providers>

    <Publishers>
        <OVT>
            <Port>9000</Port>
            <WorkerCount>1</WorkerCount>
        </OVT>
        <LLHLS>
            <!-- 
            OME only supports h2, so LLHLS works over HTTP/1.1 on non-TLS ports. 
            LLHLS works with higher performance over HTTP/2, 
            so it is recommended to use a TLS port.
            -->
            <Port>3333</Port>
            <!-- If you want to use TLS, specify the TLS port -->
            <TLSPort>3334</TLSPort>
            <WorkerCount>1</WorkerCount>
        </LLHLS>
        <WebRTC>
            <Signalling>
                <Port>3333</Port>
                <TLSPort>3334</TLSPort>
                <WorkerCount>1</WorkerCount>
            </Signalling>
            <IceCandidates>
                <IceCandidate>*:10000-10005/udp</IceCandidate>
                <!-- 
                    If you want to stream WebRTC over TCP, specify IP:Port for TURN server.
                    This uses the TURN protocol, which delivers the stream from the built-in TURN server to the player's TURN client over TCP. 
                    For detailed information, refer https://airensoft.gitbook.io/ovenmediaengine/streaming/webrtc-publishing#webrtc-over-tcp
                -->
                <TcpRelay>*:3478</TcpRelay>
                <!-- TcpForce is an option to force the use of TCP rather than UDP in WebRTC streaming. (You can omit ?transport=tcp accordingly.) If <TcpRelay> is not set, playback may fail. -->
                <TcpForce>true</TcpForce>
                <TcpRelayWorkerCount>1</TcpRelayWorkerCount>
            </IceCandidates>
        </WebRTC>
    </Publishers>
</Bind>

    <!--
        Enable this configuration if you want to use API Server
        
        <AccessToken> is a token for authentication, and when you invoke the API, you must put "Basic base64encode(<AccessToken>)" in the "Authorization" header of HTTP request.
        For example, if you set <AccessToken> to "ome-access-token", you must set "Basic b21lLWFjY2Vzcy10b2tlbg==" in the "Authorization" header.
    -->
    <!--
    <Managers>
        <Host>
            <Names>
                <Name>*</Name>
            </Names>
            <TLS>
                <CertPath>path/to/file.crt</CertPath>
                <KeyPath>path/to/file.key</KeyPath>
                <ChainCertPath>path/to/file.crt</ChainCertPath>
            </TLS>
        </Host>
        <API>
            <AccessToken>ome-access-token</AccessToken>

            <CrossDomains>
                <Url>*.airensoft.com</Url>
                <Url>http://*.sub-domain.airensoft.com</Url>
                <Url>http?://airensoft.*</Url>
            </CrossDomains>
        </API>
    </Managers>
    -->

    <VirtualHosts>
        <!-- You can use wildcard like this to include multiple XMLs -->
        <VirtualHost include="VHost*.xml" />
        <VirtualHost>
            <Name>default</Name>
            <!--Distribution is a value that can be used when grouping the same vhost distributed across multiple servers. This value is output to the events log, so you can use it to aggregate statistics. -->
            <Distribution>ovenmediaengine.com</Distribution>

            <!-- Settings for multi ip/domain and TLS -->
            <Host>
                <Names>
                    <!-- Host names
                        <Name>stream1.airensoft.com</Name>
                        <Name>stream2.airensoft.com</Name>
                        <Name>*.sub.airensoft.com</Name>
                        <Name>192.168.0.1</Name>
                    -->
                    <Name>*</Name>
                </Names>
                <!--
                <TLS>
                    <CertPath>path/to/file.crt</CertPath>
                    <KeyPath>path/to/file.key</KeyPath>
                    <ChainCertPath>path/to/file.crt</ChainCertPath>
                </TLS>
                -->
            </Host>

            <!-- 	
            Refer https://airensoft.gitbook.io/ovenmediaengine/signedpolicy
            <SignedPolicy>
                <PolicyQueryKeyName>policy</PolicyQueryKeyName>
                <SignatureQueryKeyName>signature</SignatureQueryKeyName>
                <SecretKey>aKq#1kj</SecretKey>

                <Enables>
                    <Providers>rtmp,webrtc,srt</Providers>
                    <Publishers>webrtc,hls,llhls,dash,lldash</Publishers>
                </Enables>
            </SignedPolicy>
            -->

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

            <!-- <Origins>
                <Properties>
                    <NoInputFailoverTimeout>3000</NoInputFailoverTimeout>
                    <UnusedStreamDeletionTimeout>60000</UnusedStreamDeletionTimeout>
                </Properties>
                <Origin>
                    <Location>/app/stream</Location>
                    <Pass>
                        <Scheme>ovt</Scheme>
                        <Urls><Url>origin.com:9000/app/stream_720p</Url></Urls>
                    </Pass>
                    <ForwardQueryParams>false</ForwardQueryParams>
                </Origin>
                <Origin>
                    <Location>/app/</Location>
                    <Pass>
                        <Scheme>ovt</Scheme>
                        <Urls><Url>origin.com:9000/app/</Url></Urls>
                    </Pass>
                </Origin>
                <Origin>
                    <Location>/edge/</Location>
                    <Pass>
                        <Scheme>ovt</Scheme>
                        <Urls><Url>origin.com:9000/app/</Url></Urls>
                    </Pass>
                </Origin>
            </Origins> -->
            
            <!-- Settings for applications -->
            <Applications>
                <Application>
                    <Name>app</Name>
                    <!-- Application type (live/vod) -->
                    <Type>live</Type>
                    <OutputProfiles>
                        <!-- Enable this configuration if you want to hardware acceleration using GPU -->
                        <HardwareAcceleration>false</HardwareAcceleration>
                        <OutputProfile>
                            <Name>bypass_stream</Name>
                            <OutputStreamName>${OriginStreamName}</OutputStreamName>
                            <Encodes>
                                <Audio>
                                    <Bypass>true</Bypass>
                                </Audio>
                                <Video>
                                    <Bypass>true</Bypass>
                                </Video>
                                <Audio>
                                    <Codec>opus</Codec>
                                    <Bitrate>128000</Bitrate>
                                    <Samplerate>48000</Samplerate>
                                    <Channel>2</Channel>
                                </Audio>
                                <!-- 							
                                <Video>
                                    <Codec>vp8</Codec>
                                    <Bitrate>1024000</Bitrate>
                                    <Framerate>30</Framerate>
                                    <Width>1280</Width>
                                    <Height>720</Height>
                                    <Preset>faster</Preset>
                                </Video>
                                -->
                            </Encodes>
                        </OutputProfile>
                    </OutputProfiles>
                    <Providers>
                        <OVT />
                        <WebRTC />
                        <RTMP />
                        <SRT />
                        <MPEGTS>
                            <StreamMap>
                                <!--
                                    Set the stream name of the client connected to the port to "stream_${Port}"
                                    For example, if a client connects to port 4000, OME creates a "stream_4000" stream
                                    <Stream>
                                        <Name>stream_${Port}</Name>
                                        <Port>4000,4001-4004</Port>
                                    </Stream>
                                    <Stream>
                                        <Name>stream_4005</Name>
                                        <Port>4005</Port>
                                    </Stream>
                                -->
                                <Stream>
                                    <Name>stream_${Port}</Name>
                                    <Port>4000</Port>
                                </Stream>
                            </StreamMap>
                        </MPEGTS>
                        <RTSPPull />
                        <WebRTC>
                            <Timeout>30000</Timeout>
                        </WebRTC>
                    </Providers>
                    <Publishers>
                        <AppWorkerCount>1</AppWorkerCount>
                        <StreamWorkerCount>8</StreamWorkerCount>
                        <OVT />
                        <WebRTC>
                            <Timeout>30000</Timeout>
                            <Rtx>false</Rtx>
                            <Ulpfec>false</Ulpfec>
                            <JitterBuffer>false</JitterBuffer>
                        </WebRTC>
                        <LLHLS>
                            <ChunkDuration>0.2</ChunkDuration>
                            <SegmentDuration>6</SegmentDuration>
                            <SegmentCount>10</SegmentCount>
                            <CrossDomains>
                                <Url>*</Url>
                            </CrossDomains>
                        </LLHLS>
                    </Publishers>
                </Application>
            </Applications>
        </VirtualHost>
    </VirtualHosts>
</Server>