What Is WebRTC? Web Real-Time Communication
The open standard that put a working phone inside every browser — no plugins, no downloads, no setup.
WebRTC (Web Real-Time Communication) is an open-source standard developed by Google and standardized by the IETF and W3C that enables browsers to make real-time voice and video calls without any plugins, downloads, or installed software. It handles everything a calling app needs — microphone capture, audio encoding, noise suppression, echo cancellation, and network traversal — all inside the browser sandbox.
How WebRTC works
WebRTC does three things simultaneously, all inside the browser:
- Media capture. It accesses your microphone (and camera, for video) through the browser's getUserMedia API. The audio is captured as a raw PCM stream.
- Encoding and processing. The raw audio is compressed with a codec — typically Opus, which delivers excellent voice quality at bitrates as low as 6–32 kbps. WebRTC also applies acoustic echo cancellation, noise suppression, and automatic gain control to clean up the audio before transmission.
- Network transport. WebRTC uses SRTP (Secure Real-Time Transport Protocol) to encrypt and transmit audio packets. It handles NAT traversal — the technical problem of connecting two devices behind different routers — via ICE, STUN, and TURN protocols. This is what makes browser-to-browser and browser-to-carrier connections work reliably across different networks.
Why WebRTC matters for business calling
Before WebRTC, making a VoIP call from a computer required either a downloaded softphone application (like the old Skype desktop client) or a physical VoIP desk phone connected to a PBX. WebRTC eliminated both dependencies:
- No installation. The browser IS the phone. Users go from "I need to make a call" to "I'm on the call" in under 30 seconds.
- No IT configuration. No SIP settings, no codec selection, no firewall rules. WebRTC handles network negotiation automatically.
- Cross-platform by default. Works on Chrome, Firefox, Safari, and Edge — on desktop and mobile — with identical call quality.
- Enterprise-grade security. All WebRTC media streams are mandatorily encrypted with DTLS-SRTP. There is no unencrypted mode in the standard.
WebRTC vs. traditional VoIP (SIP)
| Feature | WebRTC | Traditional VoIP (SIP) |
|---|---|---|
| Setup | None — works in browser | Requires softphone app or desk phone |
| Configuration | Automatic (ICE/STUN/TURN) | Manual (SIP proxy, codec list, NAT rules) |
| Encryption | Mandatory (DTLS-SRTP) | Optional (often unencrypted) |
| Codec | Opus (modern, adaptive) | G.711/G.729 (legacy) |
| Firewall traversal | Built-in (ICE) | Problematic; often needs configuration |
How AKITAKI uses WebRTC
AKITAKI is a WebRTC-powered dial pad. When you enter a number and press call, the browser's WebRTC engine captures your voice, encodes it with Opus, encrypts it with DTLS-SRTP, and sends it to our carrier (Telnyx). Telnyx bridges the WebRTC stream to the PSTN — the regular global phone network — completing the call to any mobile or landline number in 180+ countries. You hear the other party through the same encrypted WebRTC channel. Read more about the PSTN →