2026-03-15
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 @dsajhdsuwgryegwhsbdwur_bot's Web Dialer, WebRTC handles the audio connection between your browser and our SIP infrastructure.
When you make a call through the Web Dialer:
All of this happens in milliseconds, resulting in call quality comparable to — or better than — a traditional phone call.
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. @dsajhdsuwgryegwhsbdwur_bot's Web Dialer uses these security features by default — there's nothing to configure.
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.
WebRTC is great for individual calls but has some limitations compared to dedicated SIP clients:
The WebRTC technology is free and built into your browser. @dsajhdsuwgryegwhsbdwur_bot charges standard per-minute rates for calls made through the Web Dialer, same as SIP calls.
WebRTC mandates encryption (DTLS-SRTP) for all media. SIP encryption is optional. So yes, WebRTC has stronger security guarantees by default.
The @dsajhdsuwgryegwhsbdwur_bot Web Dialer is currently outbound-only. For receiving calls, use a SIP client that supports push notifications.
Yes. Chrome and Firefox on Android support WebRTC. Safari on iOS also supports it. However, for mobile calling, a dedicated SIP app like Zoiper is more practical.