WebRTC: the magic of direct connections between browsers

What is WebRTC

WebRTC (Web Real-Time Communication) is a set of technologies that enables real-time audio, video, and data exchange directly between browsers and other devices, without the need for traditional servers to proxy media files. This is made possible by the use of Peer-to-Peer (P2P) protocols, which allow direct connections between users.

WebRTC was first introduced by Google in 2011. Over time, leading market players, including telecommunications companies and browser developers, joined in its development. In 2017, Apple also announced full support for WebRTC in Safari and on iOS devices, cementing the technology's status as a widely accepted standard.

The unique feature of WebRTC is that it allows users to create video chats without using a third-party server — all you need is a browser. WebRTC also doesn’t require the installation of additional plugins: to ensure smooth video streaming in your browser, all you need to do is write code in HTML or JavaScript. What's more, WebRTC isn't just for video chats: this technology can also be used to transfer files of any format and text messages.

How does WebRTC work?

WebRTC can operate on two «transmission paths» — TCP and UDP:

• TCP can be thought of as a postal service that delivers every letter in the correct order and without errors. This ensures that video is transmitted reliably, without loss. But there is a downside: this type of delivery takes longer and has volume limitations — for example, it is difficult to transmit Full HD video to several participants at the same time.
• UDP works faster — it is like an instant message without checking whether each character has been received. Here, you can easily transmit high-quality video to a large number of people, but sometimes individual «pieces» of the image may be lost. Because of this, the picture may jerk or become less clear.

Schematically, the principle of data transmission via TCP and UDP protocols looks like this:

In calls between a small number of people, everything works directly between browsers. But if there are many participants (hundreds of people in a large conference), additional servers are needed. They help distribute the load and make the video stable for everyone.

API WebRTC

In its work, the transformational WebRTC technology uses three main tools — Javascript API (Application Program Interface):

1. getUserMedia is a streaming media that receives audio and video.

   This tool allows the browser to work with your camera and microphone. You can choose which camera to use, which microphone to connect, and even adjust the video quality. For example, to prevent the connection from slowing down on a weak Internet connection, you can reduce the quality to 360p. And the built-in algorithm removes background noise and makes the voice clearer.

2. RTCPeerConnection is a peer-to-peer connection that transmits audio and video.

   This is the «heart» of WebRTC, which provides a direct connection between two browsers. Thanks to this, video and audio are transmitted directly from one person to another, without intermediate servers.

   But there is one problem: Internet service providers often use NAT (a technology that can block direct connections). To get around this limitation, WebRTC has built-in support for special servers (STUN/TURN). They help «find the way» between two users and avoid connection failures.

3. RTCDataChannel is a data channel that transmits arbitrary data.

   This tool allows you to send any data along with video and audio. This can be text (as in chat) or files (documents, images, etc.). Everything happens directly between users, without the need for an intermediate server. The only file size limitations are those of the browser itself — usually, several gigabytes can be transferred.

Simply put, WebRTC makes it possible not only to see and hear each other directly through the browser, but also to share files or chat in the built-in chat without any additional software.

Advantages of using WebRTC

WebRTC opens up a new level of online communication thanks to a number of advantages:

1. Security.

   WebRTC technology works through proven network protocols, which guarantees the security of data transmission. The system doesn’t use any additional servers or intermediary applications, and all corporate metadata remains on the company's server or directly on the participants' devices (thanks to P2P protocols).

   WebRTC also uses always-on encryption for voice and video. This is done using the Secure RTP (SRTP) protocol, which not only encrypts data but also guarantees its authenticity. This is especially useful when users connect via unsecured Wi-Fi networks, as it prevents the possibility of your call being listened to or recorded.

2. Reliability.

   Session reliability is particularly important in situations where the network has limitations, for example, due to the use of network address translators (NAT). NAT is a technology that allows multiple devices on a home or corporate network to have a single public IP address, but it can sometimes interfere with the proper functioning of Internet connections or even block certain protocols.

   WebRTC is able to bypass the need to transfer data through media servers, allowing users to connect directly with each other. This helps reduce latency and improve video and audio quality. For example, when you make a video call, data is transferred directly between you and your conversation partner without unnecessary intermediaries, which significantly reduces potential latency and improves connection clarity.

   In addition, this type of connection reduces the load on servers. Since most data doesn’t need to be transferred via third-party servers, this optimises resource usage and ensures more stable and efficient operation of the system as a whole.

3. Versatility.

   WebRTC works on virtually any device and in any browser, regardless of the operating system. You can use the technology for video calls or voice calls, even if you have different devices — one may run on Windows, while another runs on macOS or Android, and the connection will still be established without any problems.

   WebRTC allows you to establish real-time connections even if you are using different browsers (Google Chrome, Mozilla Firefox, Safari, or Microsoft Edge). This is made possible by the use of standard APIs from the W3C (World Wide Web Consortium) and protocols from the IETF (Internet Engineering Task Force), which define how these technologies should work to ensure compatibility between different systems.

   Thanks to this versatility, users can make video calls and transfer data without the need for specialised software or additional settings — all they need is a standard browser that supports WebRTC. This makes the technology accessible and convenient for a wide range of users, regardless of the device or operating system they use.

4. Audio and video quality.
   WebRTC technology uses special «smart» codecs:
   • Opus for audio, which makes voices sound natural, without metallic echoes or distortion, and can also adjust to the quality of the Internet so that speech remains clear even in difficult conditions.
   • VP8 for video, which compresses the image so that it is clear and doesn’t take up too much traffic, so the video looks smooth, without constant freezes.

   This gives the user high-quality sound and clear video without the need to install or configure anything.

5. The ability to use multiple media streams.

   When you are talking on a video call and suddenly the Internet starts to slow down, with regular services this can lead to freezes, distorted voice, or a black screen. WebRTC works differently: it adjusts itself to the quality of the Internet. If the connection weakens, the system reduces the video quality or compresses the sound slightly so that the conversation remains clear and uninterrupted. When the Internet connection becomes stable again, the quality automatically improves.

   This is possible thanks to the use of a special set of protocols — RTCP (for transmission control) and SAVPF (for secure audio and video transmission with feedback). With their help, the browser receiving the media constantly informs the sending browser about the state of the network: whether there are delays, packet loss, etc. The sender analyses this data and immediately adjusts the connection quality so that the conversation remains stable even when the network is unstable.

6. Adaptability.

   Let's imagine that one participant in a call is sitting at a laptop with fast Internet, while another is connected from a smartphone with a slow mobile connection. WebRTC can negotiate separately for each device to select the optimal sound and video quality. This means that the laptop can receive high-resolution images, while the smartphone receives simplified images so as not to overload the network. As a result, both users remain connected without interruptions, and each receives the best quality available for their conditions.

   The process that allows two devices to «get to know» each other is called «signalling». First, both devices connect to a special intermediary server, through which they exchange technical information: what video format they support, what image size is best to transmit, how best to compress sound, etc. Once everything has been agreed upon, the devices can establish a direct connection and begin communicating.

   

7. Compatibility with VoIP and video.

   As a rule, companies already have phones for Internet calls or video conferencing that operate according to older standards. New technologies are usually incompatible with older devices, which creates a compatibility problem.

   The biggest advantage of WebRTC is that this technology integrates easily with what you already have and doesn’t require a complete replacement of equipment or software. WebRTC can communicate with many existing systems, such as corporate phones (SIP), messengers (XMPP, Jingle) or even regular telephone networks (PSTN). If devices adhere to standard rules, they will almost certainly be able to work with WebRTC. And where additional format conversion is required, special gateways are available on the market.

Who benefits from WebRTC and how?

Now let's look at the benefits of WebRTC for different categories:

• make video and audio calls directly in their browser without installing additional software;
• enjoy high-quality sound and video even with an unstable Internet connection;
• use the open API easily and simply: just open a page or link, and the connection is already working.

• lower the entry threshold and simplify the development process, as it is part of the HTML5 specification, supported by international standards (IETF, W3C) and available directly in the browser via JavaScript;
• launch projects quickly and inexpensively thanks to its simple architecture: only two browsers and an SDP exchange mechanism are needed for a basic prototype;
• open up access to a large audience without rewriting for each client, as WebRTC is supported in Chrome, Firefox, Opera, and most modern mobile browsers;
• simplify the integration of video and audio communication into web applications thanks to a standardised API.

In general, WebRTC is ideal for creating browser-based video conferencing applications, effectively allowing you to implement Skype within a browser.

• connect specialists directly when users visit the website;
• offer additional conveniences, such as joint page viewing or file sharing;
• simplify the process of connecting customers to operators as much as possible (without installing programmes and plugins), thereby providing a better customer experience;
• reduce equipment and maintenance costs by using standard web technologies.

For the customer, it looks like this: one click and you are already connected to the company via video or voice, without any extra steps.

Today, we can confidently say that the open WebRTC technology has a promising future, as it opens up new opportunities for innovative real-time communication. With the growing demand for VoIP calls and OTT (Over-the-Top) programmes, WebRTC has every chance of gradually replacing traditional telephone numbers, enabling calls to be integrated directly through websites and applications.