WebRTC: Real-Time Communication For Your Startup
Yuriy Luchaninov, MobiDev

If your project involves audio and video communication in real time—regardless whether it is the main idea or just an auxiliary feature to enhance user experience—there is a technology that might become your optimal choice. It is called Web Real-Time Communication (WebRTC), and it is basically supported by any device that has a browser, spanning across multiple operating systems.

WebRTC is an open-source communication technology that was developed by Google and first release in 2011. It enables API-based communication, transmitting audio, video, and data, as well allowing developers to avoid the use of native plugins.

WebRTC is on the rise today. Facebook, Google, Apple, Microsoft and Amazon are now all on board with its implementation. Products employing WebRTC reached a global valuation of at least $10.7 billion. According to a Future Market Insights report from 2015, the market value of WebRTC products is expected to reach $23 billion by 2025. North American market share for WebRTC systems exceeded 50% in 2016, and it is expected to continue to perform well in regions where people have widespread access to high-speed internet and large numbers of mobile devices.

How WebRTC Works

The goal of WebRTC is enable two-way transmission of data, audio and video between participants in real time. Employing a combination of HTML5 and JavaScript APIs, WebRTC can be basically embedded in browsers to facilitate the following features:

• Sending and receiving streaming data
• Establishing and shutting connections
• Obtaining of network configuration information, including NATs, IP addresses, and firewalls
• Transmitting media data
• Reporting errors

There are three WebRTC APIs that enable this communication. RTCPeerConnection takes care of audio/video transmissions, bandwidth configuration, and encryption protocols. RTCDataChannel is built to handle generic data transmission. MediaStream provides access to components like cameras, webcams, shared desktops and microphones.

Keeping information private is a critical part of using WebRTC. The system has measures in place to avoid malicious activities, and it does not need additional plugins in order to maintain a secure connection. Standard encryption methods are used to preserve data integrity during transmits. All WebRTC components require encryption, and the JavaScript APIs utilize secure HTTPS connection.

WebRTC: Use Cases

The primary products that deploy WebRTC are ones that provide audio and video calls, such as Google Hangouts, Facebook Messenger and WhatsApp. There is also a lot of demand in the real-time video sector. Data sharing is another market segment where significant growth is anticipated.

Other spheres that can highly benefit from introduction of real-time communication within their processes are medicine, the Internet of Things (smart home products in particular), and the industrial sector.

The open source nature of WebRTC made it possible to create ready-made solutions that can be integrated with your product. One of such solutions is OpenTok, a Platform-as-a-Service product that saves time and eliminates the need for implementation from scratch. WebRTC provides a high level of flexibility, making it an appealing option for use in both public and internal communication systems, such as video-based conferencing in real time.

Tech companies are also exploring how WebRTC can be integrated with other emerging technologies. It's likely to form a foundation for apps that utilize augmented and virtual reality technologies,as well as IoT devices. Use cases that involve machine learning and AI systems are emerging continuously. An array of fields should see increased adoption of WebRTC over the next couple of years, especially the entertainment, surveillance, and field service industries.