First, WEBRTC related API
Reference: Https://github.com/ChenYilong/WebRTC/blob/master/WebRTC Getting Started tutorial/webrtc Getting Started tutorial. MD
1.1 Functional Divisions
- Get audio and video data
- Transmitting audio and video data
- Transfer arbitrary binary data
1.2 API partition: Three JS interface
- MediaStream (also called Getusermedia)
- Rtcpeerconnection (c + +)
- Rtcdatachannel
1.3 Stun and turn server functions
- STUN (Session traversal Utilities for NAT) can only UDP, tell me the address IP port that is exposed to the WAN, I use the mapped WAN address for peer data communication.
- TURN (Traversal Using Relays around for NAT) UDP or TCP, after hole failure, provide server transit data , both sides of the call data through the server, accounting for large server bandwidth- To ensure that calls work in most environments. Cross-network can only be used in the server relay (test found), using turn this situation in video calls accounted for 10%
- ICE Network Connection Service
Small knowledge of WEBRTC