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GRVYDEV / Lightspeed-webrtc

A RTP -> WebRTC broadcast server for Project Lightspeed.
MIT License
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Project Lightspeed WebRTC [Deprecated]

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NOTE: This repo has been deprecated in favor of a monorepo configuration. Please see
A RTP -> WebRTC server based on Pion written in Go. This server accepts RTP packets on port 65535 and broadcasts them via WebRTC

View Demo · Report Bug · Request Feature

Table of Contents
  1. About The Project
  2. Getting Started
  3. Usage
  4. Roadmap
  5. Contributing
  6. License
  7. Contact
  8. Acknowledgements

About The Project

This is one of three components required for Project Lightspeed. Project Lightspeed is a fully self contained live streaming server. With this you will be able to deploy your own sub-second latency live streaming platform. This particular repository takes RTP packets sent to the server and broadcasts them over WebRTC. In order for this to work the Project Lightspeed Ingest server is required to perfrom the FTL handshake with OBS. In order to view the live stream the Project Lightspeed viewer is required.

Built With

Dependencies

Getting Started

To get a local copy up and running follow these simple steps.

Prerequisites

In order to run this Golang is required. Installation instructions can be found here

Installation

Using go get

export GO111MODULE=on
go get github.com/GRVYDEV/lightspeed-webrtc

Using git

git clone https://github.com/GRVYDEV/Lightspeed-webrtc.git
cd Lightspeed-webrtc
go build

Usage

To run type the following command.

Using go get

lightspeed-webrtc --addr=XXX.XXX.XXX.XXX

Using git

cd Lightspeed-webrtc
go build
./lightspeed-webrtc --addr=XXX.XXX.XXX.XXX

Arguments

Argument Supported Values Defaults Notes
--addr A valid IP address localhost This is the local Ip address of your machine. It defaults to localhost but should be set to your local IP. For example 10.17.0.5 This is where the server will listen for UDP packets and where it will host the websocket endpoint for SDP negotiation
--ip A valid IP address none Sets the public IP address for WebRTC to use. This is especially useful in the context of Docker
--ports A valid UDP port range 20000-20500 This sets the UDP ports that WebRTC will use to connect with the client
--ws-port A valid port number 8080 This is the port on which the websocket will be hosted. If you change this value make sure that is reflected in the URL used by the react client
--rtp-port A valid port number 65535 This is the port on which the WebRTC service will listen for RTP packets. Ensure this is the same port that Lightspeed Ingest is negotiating with the client
--ssl-cert A valid ssl cert path This is the ssl cert that the websocket server will use. If omitted, the websocket will not be served over ssl.
--ssl-key A valid port number This is the ssl private key that the websocket server will use. If omitted, the websocket will not be served over ssl.

Roadmap

See the open issues for a list of proposed features (and known issues).

Contributing

Contributions are what make the open source community such an amazing place to be learn, inspire, and create. Any contributions you make are greatly appreciated.

  1. Fork the Project
  2. Create your Feature Branch (git checkout -b feature/AmazingFeature)
  3. Commit your Changes (git commit -m 'Add some AmazingFeature')
  4. Push to the Branch (git push origin feature/AmazingFeature)
  5. Open a Pull Request

License

Distributed under the MIT License. See LICENSE for more information.

Contact

Garrett Graves - @grvydev

Project Link: https://github.com/GRVYDEV/Lightspeed-webrtc

Acknowledgements