WebRTC Calling Explained

WHAT IS WebRTC?

WebRTC (Web Real-Time Communication) is a technology built into modern browsers that enables voice, video, and data communication without plugins or external software. When you use SpoofGlobal's Web Dialer, WebRTC handles the audio connection between your browser and our SIP infrastructure.

HOW IT WORKS

When you make a call through the Web Dialer:

1. Your browser captures audio from your microphone
2. WebRTC encrypts the audio using DTLS-SRTP
3. The encrypted audio stream is sent to SpoofGlobal's media server
4. Our server bridges the WebRTC stream to a standard SIP call
5. The call is routed to the destination through our SIP trunks

All of this happens in milliseconds, resulting in call quality comparable to — or better than — a traditional phone call.

WebRTC vs TRADITIONAL SIP

• No installation — WebRTC runs in the browser, SIP requires a softphone app
• Built-in encryption — WebRTC mandates DTLS-SRTP encryption. SIP encryption is optional.
• NAT traversal — WebRTC uses ICE/STUN/TURN to handle NAT automatically. SIP clients often struggle with NAT.
• Codec support — WebRTC uses Opus by default (superior quality). SIP commonly uses G.711 or G.729.

SECURITY

WebRTC is inherently secure. All media streams are encrypted with DTLS-SRTP, and signaling goes over WSS (WebSocket Secure). Your audio cannot be intercepted in transit. SpoofGlobal's Web Dialer uses these security features by default — there's nothing to configure.

BROWSER SUPPORT

WebRTC is supported in all major browsers: Chrome, Firefox, Edge, Safari, Brave, Opera. Chrome and Edge generally provide the best WebRTC performance and lowest latency.

LIMITATIONS

WebRTC is great for individual calls but has some limitations compared to dedicated SIP clients:

• No push notifications — the browser tab must be open to receive calls
• Browser tabs can be throttled by the OS, affecting audio quality in the background
• For high-volume or PBX integration, SIP or IP2IP is better suited