The web is no longer a stranger to real-time communication, as WebRTC (Web Real-Time Communication) comes into play. Although it was released in May 2011, it is still evolving and its standards are changing. The protocol set is standardized by the Working Group of WEB browsers in real-time, and the new API sets are standardized. With the advent of WebRTC, modern web applications can easily transmit audio and video content to millions of people.
The main scheme
WebRTC allows you to quickly and easily establish peer-to-peer connections with other web browsers. To create such an application from scratch, you will need a variety of frameworks and libraries that deal with common problems such as data loss, connection loss, and NAT bypass. With WebRTC, all of this is built into the browser out of the box. This technology does not require any plugins or third-party software. It has open-source code.
The WebRTC API includes media capture, audio, and video encoding and decoding, transport layer, and session management.
Media capture
We determine the type of available devices, obtain user permission to access these devices, and control the flow.
Coding and decoding of audio and video
Sending a stream of audio and video data over the Internet is not an easy task. This is where encoding and decoding are used. This is the process of dividing video frames and sound waves into smaller fragments and compressing them. This algorithm is called a codec. There are a huge number of different codecs that are supported by different companies with different business goals. There are also many codecs inside WebRTC, such as H.264, iSAC, Opus and VP8. When two browsers connect together, they choose the most optimal supported codec between the two users. Fortunately, WebRTC development does most of the coding behind the scenes.
Transport level
The transport layer controls the order of packets and handles packet loss and connection to other users. Again, the WebRTC API gives us easy access to events that tell us when there are connection problems.
Session management
Session management deals with the management, opening and organization of connections. This is usually called signaling. If you transmit audio and video streams to the user, it also makes sense to transmit the accompanying data. This is done using the RTCDataChannel API.
Engineers from such companies as Google, Mozilla, Opera and others have done a great job to bring this real-time experience to the Internet.
Browser compatibility
WebRTC standards are among the fastest growing on the web, so it doesn’t mean that every browser supports all the same features at the same time. Note that WebRTC is a set of APIs and protocols, not a single API. Support for each of them is developed in different browsers and operating systems at different levels. It tracks the adoption of modern APIs across multiple browsers. You can also find the latest information about browser support, as well as demos of WebRTC, which is supported by Mozilla, Google and Opera.
Cases of use
The network in real-time opens up access to a whole range of new applications, including text chat, screen and file sharing, games, video chat, and much more. Apart from communication, you can use WebRTC for other purposes such as:
- real-time marketing
- real-time advertising
- communication with the back office (CRM, ERP, SCM, FFM)
- personnel management
- social network
- dating services
- online medical consultations
- financial services
- observation
- multiplayer games
- live streaming
- Electronic training
Resume
Now you should also have an idea of what types of applications can be built with WebRTC since you’ve already tried it in your browser. In summary, WebRTC is a pretty useful technology.
