Draft
This page is not complete.
The WebRTC API is designed to allow JavaScript applications to create real-time connections containing audio and video streams as well as data channels for arbitrary data. These connections are created to directly link two users' browsers, without necessarily requiring any intermediary servers which support the WebRTC protocols. WebRTC also leverages the getUserMedia() method to get access to microphone and camera data. In this article, we'll take a look at how peer-to-peer connections are created and managed using WebRTC and its RTCPeerConnection interface.
A high-level description of what happens in an RTCPeerConnection was shown in an Hacks article (see all WebRTC Hacks articles here):
Resources
- A good tutorial on basic features in WebRTC is at HTML5 Rocks. A collection of basic test pages to support development are at webrtc-landing.
- You can make simple person-to-person calls (including to people using Chrome) at apprtc.appspot.com.
- A high-level description of what happens in an
RTCPeerConnectionwas shown in the Mozilla Hacks blog article Embedding WebRTC video chat.
Specifications
| Specification | Status | Comment |
|---|---|---|
| Media Capture and Streams | Candidate Recommendation | Definition of the getUserMedia API |
| WebRTC 1.0: Real-time Communication Between Browsers | Candidate Recommendation | Initial definition |