What is WebRTC?
Web Real Time Communications, or "WebRTC", is a powerful new standard for real time video, audio and data communications.
The W3C and IETF define, implement, and support the WebRTC standard. Many companies (including Temasys) and individuals actively contribute to that effort. The vision they share for WebRTC is real time video and audio communication should "just work" in web browsers without plugins. That means with no Flash, no Silverlight, no Active-X controls, and no Java Applets.
In real life, WebRTC is implemented in an open source software library and a low-level API which enables "peer to peer" (P2P) video, audio, and data transfer between the Google Chrome, Mozilla Firefox and Opera web browsers (Temasys also enables WebRTC in Microsoft Internet Explorer and Apple Safari).
And, WebRTC does essentially three (3) basic but important things:
- It enables access to the camera and microphone of a user's computer or mobile device;
- It renders video and audio in web browsers; and
- It defines how video, audio, data and chat should be transmitted between end points, or peers.
All of that is pretty fantastic. These three things used to be extremely difficult to build and support without spending enormous amounts of money on people, code, and infrastructure. If it wasn't impossible, it took a long time and a lot of patience to make it all work, reliably.
And yet, for the average developer or business that would love to embed Real Time Communications in their apps or websites, there is A LOT MORE that needs to be done to make a real time communications service reliable, stable and scalable.
That's why we built the Temasys Platform for Real Time Communications. We help make WebRTC work for you,
in any App, on any Device, at any Scale.