README

This package provides a complete WebRTC implementation written entirely in PHP, including support for ICE, DTLS, SRTP, SCTP, RTP, and data channels. Designed for real-time audio, video, and data communication, it enables peer-to-peer connectivity without relying on external WebRTC libraries. Built with modular components and fully asynchronous using ReactPHP.

Features

Full peer-to-peer WebRTC stack in native PHP
Implements ICE for NAT traversal and candidate negotiation
Secure communication via DTLS and SRTP
Reliable data transfer with SCTP and support for data channels
RTP/RTCP handling for real-time audio and video streaming
Modular design with support for custom signaling implementations
Built on top of ReactPHP for asynchronous, non-blocking performance

Requirements

PHP ≥ 8.4 with FFI and GMP extension enabled
OpenSSL development libraries
Srtp development libraries
Linux environment (Windows/macOS support planned)
- Windows users: Use WSL or Docker Desktop. Note: Native Windows support is planned for an upcoming releases (in a few months.).
- macOS users: Use an emulator like UTM or run the project using Docker. Note: Native macOS support is coming in a few months.
FFmpeg/libav shared libraries (libavcodec, libavfilter, etc.)
- Compatible with FFmpeg version 7.1.1
libopus development libraries
libvpx development libraries
- Compatible with libvpx version 1.15.0

Documentation

This package is part of the PHP WebRTC library. For complete documentation, examples, and API reference, visit:

PHP WebRTC Documentation

Credits

Authors

Amin Yazdanpanah
- Website: aminyazdanpanah.com
- Email: github@aminyazdanpanah.com
Sana Moniri
- GtiHub: sanamoniri

Reporting Issues

Found a bug? Please report it on our issues.

Contributors

Your contribution is crucial to our success, regardless of its size. We appreciate your support and encourage you to read our CONTRIBUTING guide for detailed instructions on how to get involved. Together, we can make a significant impact.

Our Plans for the Future

We are not stopping here. We're actively continuing development on the PHP WebRTC packages.

Right now, we're working privately on a Selective Forwarding Unit (SFU) implementation and a Laravel package that bundles everything together, including this WebRTC package. Once that's ready, our goal is to build a minimal video conferencing web app using Laravel, and to continue maintaining this repository along with 24 other related packages.

If you're interested in building real-time communication tools in PHP, like a video conferencing app, you are more than welcome to join the project. Fork the repos, contribute to them, and help grow this community.

Please note: contributions should primarily focus on fixing bugs. If you'd like to add a new feature or function, it must match the original WebRTC API behavior. For example, if you want to add a method to the RTCPeerConnection class, it should exist in the original JavaScript WebRTC API and follow the same naming and purpose.

Want to Collaborate?

If you'd like to become a contributor or teammate, we’d love to hear from you. This is an open source, non-profit project, so we’re mainly looking for passionate developers.

You should have at least one project that demonstrates your knowledge and ability in PHP, Laravel, and WebRTC(only one of them is also enough). If that sounds like you, this could be a great place to get involved.

Feel free to reach out by email(via github@aminyazdanpanah.com) with your GitHub username (for example: github.com/your-username) and we’ll get back to you soon.

About Git History

We've been working on this repository and many other PHP WebRTC-related ones (such as ICE, RTP, RTCP, and more than 22 others) for a long time privately(in our git server) before making them open source(We released our packages only after they have been fully tested and thoroughly debugged). Originally, there was a long commit history that reflected all our work.

However, we decided to remove that history in the initial public commit to protect our privacy. The original commits included details like our working hours based on commit times and counts, as well as our personal email addresses, which we did not feel comfortable sharing publicly.

Removing the history helps us keep that information private and stay a bit safer from potential security risks.

License

BSD 3-Clause License. See LICENSE for details.

References

WebRTC Overview

quasarstream / webrtc

Maintainers

Details