CloudUCaaS Team
May 28, 2026 · 11 min read
WebRTC enables browser-based voice and video calling without plugins. Building a production softphone requires SIP registration, ICE/STUN/TURN for NAT traversal, codec negotiation, and mobile push notifications — plus cross-browser testing on Chrome, Safari, Firefox, and mobile.
WebRTC Architecture Overview
A WebRTC softphone connects browser media streams to a SIP backend via WebSocket (WSS) or a SIP gateway. Signaling handles registration, call setup, hold, transfer, and teardown. Media flows peer-to-peer when possible, or through TURN relays when NAT prevents direct connection.
SIP Integration Options
Choose based on your existing PBX or contact center stack.
- ✓Direct SIP over WebSocket to FreeSWITCH or Asterisk
- ✓SIP gateway translating WebRTC to traditional SIP trunk
- ✓CPaaS API for programmatic call control without managing SIP directly
Production Checklist
Before launch, verify every item.
- ✓TLS/SRTP encryption for all media and signaling
- ✓STUN/TURN server configuration for corporate firewall users
- ✓Opus codec for HD voice with G.711 fallback
- ✓Push notifications for incoming calls on mobile
- ✓Cross-browser QA on desktop and mobile Safari
Conclusion
WebRTC softphones eliminate the friction of desk phones and downloads for remote agents and customer click-to-call. CloudUCaaS develops white-label WebRTC softphones for web and mobile with CRM integration and enterprise security.
