rtp2webrtc

A low latency WebRTC server in the form of a simple API for broadcasting RTP streams

About • Features • Build • Configuration • Run • Usage • License

About

rtp2webrtc is a low latency WebRTC server in the form of a simple API for broadcasting RTP streams to any browser.

Features

✅ A single statically compiled binary for each OS/architecture
✅ Easy configuration of the IP address and port of the server
✅ Easily configure the RTP source(s)
✅ Can receive a video and/or audio RTP stream
✅ Define track names without difficulty
✅ Effortlessly define your WebRTC stream codecs (must be the same as the RTP source)
✅ Easy configurable ICE server(s)
✅ Choose the address you want for the API endpoint
✅ Option to allow cross origin requests
✅ An easily configurable tool
✅ Can operate effortlessly as a daemon

Build

First check that you have Golang installed on your machine.
Then, run:

make

Quite simply!

Configuration

Here is an example of a configuration (YAML):

# IP on which rtp2webrtc should be reachable
serve-ip: 127.0.0.1
# Port on which rtp2webrtc should be reachable
serve-port: 8080
# IP of the device sending the RTP streams
rtp-host: 127.0.0.1
# If you want to retrieve a video stream (true/false)
enable-video: true
# If you want to retrieve an audio stream (true/false)
enable-audio: true
# Port on which the video stream is to be received
rtp-video-port: 9903
# Port on which the audio stream is to be received
rtp-audio-port: 9904
# Video track name
video-track-name: default
# Video track name
audio-track-name: default
# Video codec (must be the same as the video RTP stream, either video/H264 or video/VP8)
video-codec: video/H264
# Audio codec (must be the same as the audio RTP stream, audio/opus)
audio-codec: audio/opus
# A list of ICE servers that can be used for the server
ice-servers:
  - ice-server: stun:stun.l.google.com:19302
# Customisable API endpoint
api-location: /rtp2webrtc/api/v1/offer
# To enable support for cross origin requests (for testing purposes for example)
allow-cross-origin: false

Run

Direct use

To be able to use rtp2webrtc directly, you must set the environment variable RTP2WEBRTC_CONFIG_FILE_PATH as the path to the configuration file.
For example :

export RTP2WEBRTC_CONFIG_FILE_PATH=./config.yaml

Then you can run the program:

./rtp2webrtc

As a systemd service

It is possible to easily use rtp2webrtc as a daemon with the provided systemd service.
The systemd service provided can be adapted to your needs.

Steps to install rtp2webrtc as a systemd service:

Create a group:

groupadd rtp2webrtc

Create an user:

useradd -r -s /sbin/nologin -g rtp2webrtc rtp2webrtc

Copy the rtp2webrtc binary to /usr/bin/

cp rtp2webrtc /usr/bin/

Create a rtp2webrtc folder in /etc/ for the configuration file

mkdir /etc/rtp2webrtc

Copy the config.yaml configuration file to /etc/rtp2webrtc/

cp config.yaml /etc/rtp2webrtc/

Copy the rtp2webrtc.service systemd service file to /etc/systemd/system/

cp rtp2webrtc.service /etc/systemd/system/

Start the systemd service

systemctl start rtp2webrtc

Usage

In a browser

Here is an example of an HTML page containing the elements needed to display WebRTC streams in a browser:

<!DOCTYPE html>
<html>

<head>
    <meta charset="utf-8">
    <meta http-equiv="X-UA-Compatible" content="IE=edge">
    <title>WebRTC streaming</title>
    <meta name="description" content="">
    <meta name="viewport" content="width=device-width, initial-scale=1">
</head>

<body>
    <video id="feed"></video>

    <script>
        const defaultRtcConfig = new RTCPeerConnection({
            // ICE servers
            iceServers: [{
                urls: "stun:stun.l.google.com:19302"
            }]
        })

        function makeLivestream(videoElement, rtcConfig, endpoint, replyHandler, errorHandler) {
            let pc = new RTCPeerConnection(rtcConfig)

            pc.ontrack = function(event) {
                var el = videoElement
                el.srcObject = event.streams[0]
                el.autoplay = true
                el.controls = true
            }
            // If you want to receive video
            pc.addTransceiver("video", {
                "direction": "sendrecv"
            })
            // If you want to receive audio 
            pc.addTransceiver("audio", {
                "direction": "sendrecv"
            })
            pc.createDataChannel("noop")
            pc.createOffer()
                .then(offer => {
                    pc.setLocalDescription(offer)
                    return fetch(endpoint, {
                        method: "post",
                        headers: {
                            "Accept": "application/json, text/plain, */*",
                            "Content-Type": "application/json"
                        },
                        body: JSON.stringify(offer)
                    })
                })
                .then(res => res.json())
                .then(res => pc.setRemoteDescription(new RTCSessionDescription(res)))
                .then(replyHandler)
                .catch(err => console.log(err))
        }

        const videoElement = document.getElementById("feed")
        // Please update the URL according to your configuration
        makeLivestream(videoElement, defaultRtcConfig, "http://localhost:8080/rtp2webrtc/api/v1/offer", null, null)
    </script>
</body>

</html>

Behind a reverse proxy

We recommend that the server runs behind a reverse proxy.
Here is an example of a configuration for NGINX:

events {}

http {
    sendfile on;
    sendfile_max_chunk 512k;
    tcp_nopush on;
    tcp_nodelay on;

    server {
        server_name streaming.domain.tld;
        listen 80;

        location / {
            root /var/www/html;
            index index.html;
        }

        location /reverse {
            rewrite /reverse/(.*) /$1  break;
            proxy_http_version 1.1;
            proxy_set_header Upgrade $http_upgrade;
            proxy_set_header Connection upgrade;
            proxy_redirect off;
            proxy_buffering off;
            proxy_request_buffering off;
            proxy_set_header Host $host;
            proxy_set_header X-Real-IP $remote_addr;
            proxy_set_header X-Forwarded-For $proxy_add_x_forwarded_for;
            proxy_pass http://127.0.0.1:8080;
        }

        server_tokens off;
    }

}

License

This program is free software: you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation, either version 3 of the License, or (at your option) any later version.

This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details.

You should have received a copy of the GNU General Public License along with this program. If not, see http://www.gnu.org/licenses/.

gaetanlhf / rtp2webrtc

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