WebRTC vs.
Traditional VoIP
Why modern businesses are moving from legacy SIP softphones to WebRTC. Lower latency, mandatory encryption, and HD audio—directly in the browser.
Security
Network Adaptability
Latency
Why WebRTC is More Secure
Traditional VoIP (SIP) was designed in the 90s without encryption as a mandatory requirement. This makes it susceptible to packet sniffing and eavesdropping.
WebRTC was built for the modern web. It requires DTLS (Datagram Transport Layer Security) for key exchange and SRTP (Secure Real-time Transport Protocol) for media. Without encryption, a WebRTC connection simply cannot be established.
- No VPN required for security
- End-to-End Encryption capability
- Sandboxed within the browser
The Opus Codec Advantage
G.711 (Legacy)
Used by traditional phone lines. Static bitrate. If your internet drops slightly, audio cuts out or becomes robotic.
Opus (WebRTC)
Dynamic bitrate. Adapts to network conditions in real-time. Maintains clarity even on 3G/4G connections.
Experience WebRTC Quality
Make a call right now from your browser and hear the difference.