WebRTC for Ultra-low Latency Video Transmission

What is WebRTC?

With WebRTC, you can add real-time communication capabilities to your application that works on top of an open standard. It supports video, voice, and generic data to be sent between peers, allowing developers to build powerful voice- and video-communication solutions. The technology is available on all modern browsers as well as on native clients for all major platforms.

What can a WebRTC Encoder do for your workflow?

As we have released the WebRTC for more than half a year, here are some scenarios that we have already explored about

1- Peer to Peer talk for Medical industry

Driven by the modern video solutions for the market, the WebRTC deliveres more possibility of using a low cost solutions for low latency video transmission application between the operation room and the demo room, in some applications, there is only one way transmission is needed, which generated by the operation room with audio and video output, and the receiver just need to talk back but without video callback, the WebRTC Encoder L2 became possible for this application with low cost and low latency (less than 300ms under local network).

2- Transcoding and processing

3- Restreaming

4- Drone and space video system

...

How it works?

Like the general Peer to Peer solution, the two way will be connected by a RTP connection, a handshake between two connections is needed. Two elementary parts for a WebRTC Encoder are the module to generate WebRTC and a host to build connection.

The WebRTC address will be generated by the encoder and it became directly accessible on any browser, the formart is like

https://webrtc-001.xyz:8443/videocall/XSWH-00-4S4A-2MWA-00000003
https://webrtc-001.xyz:8443/video/XSWH-00-4S4A-2MWA-00000003

Want to have a demo or want to know what we are going to do the next? Click here to email us about your concerning.