What is WebRTC - How it works, advantages and examples of use

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Web Real-Time Communications is a relatively new technology that has taken online communications to a whole new level. The potential of WebRTC is far from exhausted and can become the basis for innovative solutions in the field of global communications. The technology has an open source code and is responsible for streaming data using the JavaScript API for peer-to-peer communication between browsers and applications.

The main advantage of WebRTC technology is that it does not require plugins or special software to establish communication between parties in real time. The standard, created in 2010, has become a revolution in the field of Internet communications. Now, to start a video conference or transfer files, a URL is enough. There is no need to switch between different interfaces.

The technology offers limitless possibilities for developers, businesses, individuals and all Internet users. WebRTC has a wide range of application scenarios in sectors such as education, personnel management and HR, banking, insurance, healthcare and many others.

Let's find out how WebRTC works and what are the advantages of the technology, what components it consists of and how it is used in practice.

How WebRTC Works​

WebRTC technology allows developers to create software for audio and video communication directly from the browser, without the need for additional resources. Real-time video communications allow data to be transmitted without intermediaries, which greatly simplifies communication and interaction between users around the world.

This level of approach ensures cross-platform and complete independence from hardware and operating systems, which opens up wide opportunities for developing VoIP (Voice over Internet Protocol) applications. Developers gain access to the technology through JavaScript API libraries. WebRTC is supported by Google, Firefox, Apple devices and many others.

The WebRTC standard is a product of combining two directions: interaction between a web browser and the computer's hardware multimedia capabilities (video camera, microphone) and the creation of a peer-to-peer network. The result is a direct exchange of information between browsers without the use of third-party servers.

If we look under the hood of WebRTC, we will find there a lot of interconnected APIs and protocols that work in sync and provide stable and robust communication between users, as well as the exchange of multimedia files.

Here are the main components of the technology:

• MediaStream API and GetUserMedia. The tool captures and manages data streams from audio and video (i.e. from users' microphones and cameras).
• RTCPeerConnection. A point-to-point client-to-client communication system that carries media and text data. RTC supports encryption methods and network firewall bypass tools.
• DataChannel API. Provides low-latency data exchange between browsers.

Since the WebRTC code is open to everyone, and the technology is available in any browser simply by clicking on the appropriate extension, users naturally have questions about data security.

Data in WebRTC is transmitted via the DTLS protocol, which encrypts user messages and eliminates the risk of their interception by third parties. That is, intruders will not be able to listen to, view, or read files. The protocol guarantees secure transmission and is implemented by default in all browsers with WebRTC support.

The SRTP protocol is also used, which also transmits data via encrypted channels. When using a microphone and video camera, programs that support WebRTC always ask the user for permission to turn on the devices using a flashing red dot on the browser tab.

WebRTC requires JavaScript and HTML5 support to work. These tools are used to create an interface for VoIP applications and to implement access to API libraries for WebRTC. Support for the technology is built into Chrome, Opera, Safari, and other browsers. Edge has its own implementation, called ORTC.

Network protocols ICE, STUN, TURN are used to overcome NAT and firewalls.

• ICE is a specialized protocol for communication between devices. It describes the optimal path for the nodes of the connection.
• STUN is a technology for determining the IP address of a router.
• TURN is a packet relay that participates in establishing a connection using the P2P protocol.
• NAT is a protocol for converting IP addresses to connect to a network and receive information.

All of the above protocols participate in the implementation of WebRTC communication.

Benefits of Using WebRTC​

At one time, the WebRTC mechanism transformed the possibilities of communication between users. The popularity of this technology is due to many advantages:

• Direct connection between users. Minimal communication delay, while intermediate servers and devices are not needed.
• Cross-platform. Supported by all top browsers and smartphone software.
• Security. Connections between points are encrypted by DTLS and SRTP protocols, third parties do not have access to the transferred data. At the same time, sites working via HTTPS are signed with security certificates.
• Conferences and meetings in the browser. The user does not have to install special applications to organize a video or audio meeting of a business or household nature.
• High-quality communication. Modern codecs are used, and the flow is adapted to the connection conditions. Echo and extraneous noise are automatically suppressed by the corresponding systems. There is a built-in microphone sensitivity adjustment.
• Open source. The availability of the technology allows developers to adapt WebRTC to their needs and create new products and services for private use.
• Accessibility of control. Users can connect any interface on HTML5 and JavaScript protocols.
• Resource saving. The software works equally effectively on any OS, in desktop and mobile versions. No additional resources are required.
• Speed. WebRTC is one of the fastest online communication technologies. The delay here is no more than 0.5 seconds, which is achieved through direct transmission without the participation of a server.

The combination of modern technologies turns WebRTC into an accessible communication environment that works here and now. Device owners can communicate in real time without using additional technical means.

Examples of WebRTC use in real-time applications​

In a relatively short period of time, the WebRTC stack has become a basic tool for modern communications for the largest players in the digital world, including Google and other platforms with a huge user base. Here are some of the most famous examples of the technology's use.

Voice calls​

Many platforms use VoIP services — telephony via the Internet. Voice calls are available in Google Meet and other messengers. Such communication reduces the costs of provider services and roaming. The location of the interlocutors does not matter — the cost of calls is included in the payment for the Internet package.

Collaboration​

Many companies with online presences work as a team via WebRTC services. Employees can be located in different regions, but still exchange files, share screens, and talk to each other in real time. Video conferences in Google Meet and other platforms take place with minimal signal delay.

Game applications​

Gamers around the world have found it easier to communicate in virtual space. WebRTC technology provides not only text, voice or visual interaction, but also data synchronization between participants.

E-commerce​

Companies selling goods and services via the Internet actively use the real-time communication mechanism. This makes it easier to tell about all the advantages of the product and demonstrate its benefits more clearly.

EnableX platform, based on WebRTC technologies, is embedded into any applications and websites, allowing exchange of audio and video in real time.

Online consultations​

Experts in various fields can consult clients via interactive chat, including video. Users can get answers to legal, medical and other questions at any time.

Telemedicine is becoming an increasingly popular service for those who do not have direct access to doctors. Private medical companies are actively developing this area in many countries.

Insurance​

Reviewing insurance claims in the field of car insurance takes a lot of time, and the services of consultants are usually required by clients immediately after an accident. Some companies provide a service for remote registration of an insurance situation via a mobile application.

Security services​

Banking institutions conduct online verification of clients without inviting them to the office. This saves time for users and employees of financial companies.

Basic steps for WebRTC integration​

Easy setup is one of the key advantages of the technology. Let's list the main steps for WebRTC integration:

1. Setting up access to devices. The first step in using WebRTC is capturing video and audio from the camera and microphone, respectively. This is done using the MediaStream API - the interface opens the browser to access devices.
2. Establishing a connection. Data exchange is only possible after the devices have "agreed" with each other on which protocol they will use. The process is called "signaling" and is implemented via HTTP or other external servers.
3. Connection data exchange. The ICE protocol provides an optimal way to connect devices by defining IP addresses, ports, and the connection method (in our case, Peer-to-peer).
4. Encryption and security protocols. All information transmitted via WebRTC is encrypted using the protocols we wrote about above.
5. Data transfer. The RTCPeerConnection mechanism is responsible for transferring the stream and media data between devices. All aspects of communication are controlled, including the quality of transfer - it is adapted in accordance with the network bandwidth. The DataChannel channel provides text and file exchange.
6. Bypass network obstacles. WebRTC successfully bypasses firewalls and NAT to ensure uninterrupted streaming.

The joint operation of the components eliminates long delays in transmission and guarantees the security of data exchange.

WebRTC Technological Limitations and Challenges​

Despite its many advantages, the WebRTC mechanism also has certain limitations:

• Protecting data from interception does not change the fact that WebRTC identifies real IP addresses of users. This means that anonymity of communication participants is not guaranteed. Only high-quality VPN services can provide full protection.
• Compatibility limitations. The standard is only responsible for the methods of transmitting video and audio, while the methods of implementing the communication process are left to the developers and clients who use these devices. That is, you cannot call from one WebRTC application to another (for example, log in to Google Meet via Jitsi Meet).
• Scaling issues. Communication between two devices is relatively easy to establish, but scaling in full-fledged multi-user systems is difficult. There is no adequate solution in WebRTC yet.
• Remote control. Some users believe that the technology lacks the ability to remotely control the desktop of another device. The user can broadcast what is happening on his screen, but only one-way. However, developers are limited by security protocols - the Javascript code only controls the current window.

There is no doubt that WebRTC is one of the most significant technologies of recent times, providing users and developers with wide opportunities for application. However, its active use requires an in-depth understanding of a number of issues.

Experts pay attention to recording data that is transmitted from device to device. By default, recording is not included in the WebRTC functionality. To enable this option, you need to consider which option to choose - recording on the client side or using the server? Both options have their pros and cons.

The WebRTC security architecture is quite robust, but like any open system, it has vulnerabilities. The ecosystem includes many client sides - hosts, servers, applications. Each of these nodes can become a potential source of cyber threats. Therefore, users cannot fully rely on the security of the protocol itself - they need to think about protection from external threats.