WebRTC Encryption: How Secure Voice and Video Calls Really Work

When you make a video call using WebRTC, your conversation is protected by WebRTC encryption, a built-in security layer that automatically encrypts audio, video, and data streams between browsers and devices. Also known as end-to-end encrypted real-time communication, it’s the reason your Zoom, Google Meet, or Slack call doesn’t get intercepted by hackers or ISPs. Unlike older VoIP systems where encryption was optional, WebRTC makes it mandatory—no settings to toggle, no config files to edit. It just works.

This encryption relies on two key technologies: SRTP, Secure Real-time Transport Protocol, which encrypts the actual voice and video data, and DTLS, Datagram Transport Layer Security, which handles the key exchange to make sure only your device and the person you’re calling can decrypt the stream. Together, they prevent eavesdropping, replay attacks, and man-in-the-middle snooping—even on public Wi-Fi. You don’t need a VPN. You don’t need to install extra software. The browser does it all.

That’s why WebRTC encryption matters more than ever. Businesses use it for client calls. Doctors rely on it for HIPAA-compliant consultations. Travelers use it to avoid corporate spying abroad. And because it’s built into Chrome, Firefox, Edge, and Safari, there’s no way around it—if you’re using a modern browser for voice or video, you’re already encrypted. But not all implementations are equal. Some services add their own layers, others skip key validation. That’s where things get messy. The posts below show you exactly how it works under the hood, what can go wrong, and how to spot a fake secure connection.

From real-world SRTP overhead numbers to how DTLS handshake failures cause dropped calls, you’ll find practical fixes and clear comparisons. No theory. No fluff. Just what you need to know to keep your calls private, reliable, and truly secure.