WebRTC implements web-based video conferencing under the WHATWG protocol, which aims to provide real-time Communications (RTC) capabilities with simple javascript in a browser. EasyGBS has also been successfully connected to weBRTC video output.

WebRTC is a master of streaming media technology in the Internet industry, covering audio and video collection, media processing, coding, P2P, network transmission to network reception, decoding, widely used in live broadcasting, audio and video chat, video conference, so that people without audio and video development experience can also easily develop audio and video communication software. The traditional security video monitoring industry is also based on audio and video streaming media technology to do development, so we can guess that the future application of WEBRTC technology in the field of security is also one of the major trends.

Security industry is gradually developing to civil, with the arrival of mobile Internet network environment from LAN to narrowband public network. Security industry currently existing communication forwarding difficulties, echo, plug-in access and other pain points, can be solved through weBRTC.

P2P

P2p in WebRTC supports three network connection modes: direct LAN connection, public network penetration, and public network forwarding. For example, three Bridges are built on the same river at the same time. When the bridge with higher priority is completed, the bridge with lower priority will be dismantled (priority: direct connection > penetration > forward), and the bridge with higher priority will be used for traffic. In this way, even if the penetration fails, the map can be generated within 1 second. Each method will be tried for 15 seconds, if there is no connection within 15 seconds, it will automatically give up. WebRTC UDP transmission, WebRTC based on UDP P2P, has the advantages of fast, real-time, smooth graph.

2, echo cancellation

WebRTC’s predecessor is GIPS, which is the authority on echo cancellation.

3, Chrome browser plug-in free access to audio and video

WebRTC has the same code as Chrome, so it makes sense for Chrome to support WebRTC. Firefox, Edge, and Safari all support WebRTC and will get better and better. Webrtc provides interface calls to javascript. This ensures a solid environment for WebRTC applications under the B/S architecture.

WebRTC is mainly to achieve audio and video collection, encoding and decoding, these functions are not particularly important for security scenes, security industry also has its own relatively mature encoding and decoding scheme. However, in the application of WebRTC, mostly from p2p to start the operation, the client through its own hardware to decode, Chrome can first support from the server, edge end (NVR with strong performance, Haisi 3531, 3536) first support, IPC end also has its own collection scheme, According to the type of the client to carry out terminal identification, so as to carry out the choice of the scheme to achieve.

Traditional industries should be targeted and selectively absorbed when embracing the Internet, and the Internet should fully understand the background of the industry and combine with the reality of the industry when transforming traditional industries. Only in this way, can the two truly merge, collision sparks, the output of an innovative product and service. In the future, all series of Streaming media platforms of TSINGSEE Video cloud will fully support video output of WebRTC, including mainstream security platforms such as EasyDSS, EasyNVR and EasyCVR.