What is WebRTC?

What Is WebRTC?

WebRTC stands for Web Real-Time Communication. It is an open-source technology built into web browsers that enables voice calls, video calls, and data sharing directly between users without needing any plugins, downloads, or additional software. If you have ever made a video call through a web browser -- no app installed, no extension added -- there is a good chance WebRTC was the technology making it work.

The beauty of WebRTC is its simplicity for the end user. You click a link, your browser asks permission to use your microphone and camera, and you are connected. Behind the scenes, WebRTC handles everything: establishing the connection, encoding the audio and video, managing network conditions, and encrypting the data.

How WebRTC Works

WebRTC is a set of APIs (application programming interfaces) and protocols built into modern web browsers including Chrome, Firefox, Safari, and Edge. When two users want to communicate, the following process happens in the background:

Signalling

Before two browsers can communicate directly, they need to find each other and agree on how to connect. This initial handshake is called signalling. WebRTC itself does not define how signalling works -- that is left to the application developer -- but it typically involves exchanging connection details through a web server. Think of it as two people exchanging phone numbers before they can call each other.

Peer-to-Peer Connection

Once signalling is complete, WebRTC establishes a direct, peer-to-peer connection between the two browsers whenever possible. This means the audio and video data travels directly from one user to the other, without passing through a central server. This reduces latency (delay) and improves quality because there is no middleman slowing things down.

In practice, direct peer-to-peer connections are not always possible because of firewalls and network address translation (NAT). WebRTC uses a system called ICE (Interactive Connectivity Establishment) to find the best possible path between users, and if direct connection fails, it can relay traffic through a TURN server as a fallback.

Built-In Security

WebRTC encrypts all communications by default using DTLS (Datagram Transport Layer Security) and SRTP (Secure Real-Time Protocol). This encryption is mandatory -- it cannot be turned off. This means that voice and video data transmitted via WebRTC is protected from eavesdropping as standard, which is a significant advantage over some older communication technologies.

Why WebRTC Matters for Businesses

WebRTC has opened up communication possibilities that were previously complex and expensive to implement:

• Click-to-call buttons on websites that connect visitors directly to an agent through their browser, with no phone needed
• Video consultations for healthcare, financial advice, or customer support without requiring the customer to install anything
• Browser-based softphones that let contact centre agents take calls from any computer, anywhere
• In-app communication features added to web applications without building telephony infrastructure
• Screen sharing and co-browsing for real-time customer support

For contact centres, WebRTC is particularly significant because it enables fully browser-based agent desktops. Agents do not need traditional phones or specialised softphone software -- they can handle calls through their web browser. This makes remote working simpler, reduces IT overhead, and allows rapid scaling because setting up a new agent is as easy as giving them a login.

WebRTC and Telephone Payments

WebRTC creates interesting possibilities for telephone payments because it can carry both voice and data simultaneously. A customer connected to an agent via WebRTC could be presented with a secure payment form within the same browser session, eliminating the need to switch channels or enter card details over the phone entirely.

However, WebRTC-based payment flows need careful consideration from a security and compliance perspective. While WebRTC's built-in encryption protects data in transit, the browser environment itself can be vulnerable. If a customer enters card details into a form within a WebRTC session, the application needs to ensure that data is handled in a PCI-compliant manner -- that it is not stored locally, not exposed to the agent's view unless masked, and transmitted securely to the payment processor.

For traditional phone-based payments where the customer uses their telephone keypad, WebRTC can still play a role on the agent side. The agent's softphone may be WebRTC-based, and the customer's DTMF tones still need to be masked before reaching the agent's browser. The masking technology needs to work correctly regardless of whether the agent is using a traditional phone or a WebRTC-based softphone.

Practical Considerations

• WebRTC depends on the end user's browser and internet connection. While modern browsers all support it, older browsers and very slow connections can cause issues.
• Audio and video quality are affected by network conditions. WebRTC adapts dynamically, but on poor connections, quality will degrade.
• Not all WebRTC implementations are equal. The technology provides the building blocks, but the quality of the finished product depends heavily on how the application developer has implemented it.
• For payment applications, ensure that any WebRTC-based payment flow has been assessed for PCI DSS compliance by a qualified assessor.
• Test thoroughly across different browsers, devices, and network conditions before relying on WebRTC for customer-facing services.

WebRTC has fundamentally changed what is possible in browser-based communication. It has made real-time voice and video accessible to any web developer, removed the need for plugins and downloads, and enabled a new generation of communication-enabled applications. For businesses, it represents a flexible, cost-effective way to add real-time communication to their customer experience.