1. What is WebRTC?
WebRTC (Web Real-Time Communication) is an open-source standard created by the W3C and IETF that enables modern web browsers to establish peer-to-peer real-time audio, video, and data communication without third-party plugins or desktop software.
2. How WebRTC Transforms Modern Cloud Telephony
- Zero Desktop Installation: Agents simply log into the SanalPBX web dashboard and their extension registers instantaneously via secure WebSockets (WSS).
- Adaptive Opus Audio: Opus dynamically adjusts bitrates between 6 kbps and 510 kbps to deliver pristine studio-grade audio even over volatile Wi-Fi connections.
- Built-In End-to-End Encryption: Every media packet is encrypted using DTLS (Datagram Transport Layer Security) and SRTP.
3. NAT Traversal: STUN & TURN Infrastructure
Corporate firewalls and symmetric NAT gateways often block direct peer-to-peer media. SanalPBX solves this via globally distributed Coturn servers:
- STUN: Determines the endpoint's public-facing IP and UDP port mapping.
- TURN: Acts as a secure, low-latency relay when direct peer-to-peer streaming is restricted by strict corporate firewalls.