WebRTC-Echoes: Interop for C#, C++, Python, TypeScript, Go and Servers!

[se...@pion.ly]

I am so excited to share webrtc-echoes. This repository contains a test suite that asserts that many different WebRTC implementations can communicate with each other. It also asserts that these WebRTC implementations work with WebRTC servers. If you are interested in getting involved we would love to have you.

I think this is important for a few reasons
* Ensure that things connect, and prevents regressions. It makes all the implementations more robust.
* Great for learning. Developers see all the options available and can try different software.
* Builds community. With all of us working together it will help good ideas spread faster.

This is what is implemented today.
* aiortc: WebRTC and ORTC implementation for Python using asyncio. [0]
* libdatachannel: C/C++ WebRTC Data Channels and Media Transport standalone library (bindings for Rust, Node.js, and Unity)[1]
* Pion: Pure Go implementation of the WebRTC API. [2]
* SIPSorcery: A WebRTC, SIP and VoIP library for C# and .NET Core. Designed for real-time communications apps. [3]
* werift-webrtc: WebRTC Implementation for TypeScript (Node.js) [4]

Servers:
* GStreamer's webrtcbin [5]
* Janus [6]
* Kurento [7]
* Google WebRTC [8]

[0] https://github.com/aiortc/aiortc
[1] https://github.com/paullouisageneau/libdatachannel
[2] https://github.com/pion/webrtc
[3] https://github.com/sipsorcery-org/sipsorcery
[4] https://github.com/shinyoshiaki/werift-webrtc
[5] https://gstreamer.freedesktop.org/documentation/webrtc/index...
[6] https://janus.conf.meetecho.com/
[7] https://www.kurento.org/

[se...@pion.ly]

https://github.com/sipsorcery/webrtc-echoes

And I missed the most important part... a link to the repo itself. Thank you Tristan Matthews for telling me!