WebRTC stands for Network real-time communication. It’s a very exciting, powerful and highly disruptive cutting-edge technology and standard. Since its inception, WebRTC has been supported by 80% of browsers. Data show that WebRTC market will grow at a compound annual growth rate of 34.37% from 2017 to 2021.

What is the WebRTC

As we all know, browsers themselves do not support direct channel communication between each other, are through the server for transit. For example, if you have two clients, A and B, and they want to communicate, you need to establish a channel between A and the server, and between B and the server. When user A sends a message to User B, User A first sends the message to the server. The server forwards the message to User B and vice versa. In this way, a message between A and B passes through two channels, and the communication efficiency is limited by the bandwidth of both channels. At the same time, such channels are not suitable for data stream transmission, and how to establish point-to-point transmission between browsers has been puzzling developers. WebRTC came into being.

WebRTC is a real-time communication solution initiated by Google, which contains video and audio collection, encoding and decoding, data transmission, audio and video display and other functions, we can quickly build an audio and video communication application through technology. Although its name is WebRTC, it actually supports not only audio and video communication between the Web, but also Android and IOS. In addition, because the project is open source, we can also compile C++ code, so as to achieve the interoperability of the whole platform.

What problem does WebRTC solve?

Before WebRTC, the difficulties of RTC communication for developers mainly come from the complexity of Internet network, sensitive delay, low real-time audio and video fluency and clarity, and high operating costs. But these problems have been better solved after the emergence of WebRTC:

1. The complex and different NAT and firewall of Internet network have brought great challenges to the establishment of media P2P. The advent of WebRTC provides direct end-to-end communication for browsers, making it easy for developers to implement such connections. At the same time, WebRTC has the open source project Libjingle, which supports STUN, TURN and other protocols.

2, delay sensitive in the early RTC technology, TCP (Transmission Control Protocol- Transmission Control Protocol) due to its own mechanism defects, can only use UDP Transmission, but this requires developers to solve retransmission, disorder and other problems. WebRTC provides NACK, FEC technology that eliminates the need for routing through the server, reducing latency and bandwidth consumption. Direct communication improves the speed of data transfer and file sharing.

3, the fluency of the Internet network is unstable, especially some small operators, in the peak traffic use often cannot ensure enough bandwidth. An adaptive algorithm is needed to deal with network congestion and smooth transmission. WebRTC provides TCC + SVC + PACER + JitterBuffer technology support.

4, clear voice due to the complex terminal equipment and environment, there will be noise, echo interference, at this time WebRTC provides 3A algorithm + NetEQ, so that the real-time environment of sound processing and interactive experience has been greatly improved.

For developers or enterprises, the process of using WebRTC simply needs to download a WeBRTC-compatible browser and use it, without the need for additional software, plug-ins or continuous server involvement, audio and video applications can be easily embedded into any website and connected through the Internet. Greatly saving the development time and cost.

Mainstream browsers such as Microsoft Edge, Google Chrome, Mozilla Firefox, Safari, Safari, Opera and Vivaldi all support WebRTC.

Current situation and future of WebRTC

Since 2020, browser development and compatibility have changed, WebRTC’s latency and security have been improved and protected, especially after the outbreak of the virus, the demand for real-time video has increased 30 times than before, which further stimulated the rapid development of WebRTC products.

And weBRTC-based products are not limited to traditional Internet applications or browser terminal operating environment. In fact, WebRTC can communicate with each other no matter whether the terminal running environment is PC, Android, iOS or device, as long as it meets WebRTC specifications. Therefore, products developed based on WebRTC are basically compatible in extensibility, application scenarios and terminals, which makes great progress in application scenarios such as online education, video conferencing, telemedicine and online live broadcasting.

The biggest advantage of weBRTC-based products is standardization, which provides unified, open standards for real-time communication capability description and connection establishment for all terminals that need real-time communication. The disadvantage is that there are some problems in the interoperability of current audio and video products due to the different compatibility of browsers from different manufacturers and the different SDK in the APP that needs to be integrated.

So in the future, as a terminal technical specification, WebRTC is only one part of the real-time communication solution, but it is the part closest to the user, and perhaps the most important part. Standardization of terminal technical specifications is a good start. Even Apple, which is known for its closed technology ecosystem, has started to embrace WebRTC, which will promote the development and popularity of WebRTC technology, and more and more Internet applications will build real-time communication services based on WebRTC.

The emergence of new application scenarios such as VR, AR and automatic driving will also bring new demands and power to WebRTC technology, and the commercial success of application scenarios will also continuously inject vitality and material resources into the development of technology. In recent years, the fierce development and fervour of Internet-based video applications have driven the development of internet-based real-time audio and video communication technology, calling for the maturity and implementation of uniform, open and transparent standards such as WebRTC.

In the future, we can imagine. In a world built based on webRTC, the connection process of all terminals is unified. As long as channels are open between terminals, real-time communication can be established.

Wechat and WhatsApp, for example, can set up video calls, much like you would use a cell phone in China to call a friend’s landline in the United States. You can even use wechat to connect to your car’s screen to play music and turn on the air conditioner in advance.

In the scene of real-time audio and video communication, anyRTC can provide one-stop audio and video solutions according to different needs, help enterprises reduce costs and increase efficiency, and make video create value.

AnyRTC will continue to study in the field of audio and video, provide customers in different fields with innovative and high-quality solutions, and contribute to the development of the industry.