WebRTC audio is the browser-native, standardized stack for real-time voice that any web page can use without plugins. It strings together the whole conversational pipeline: capturing the microphone via getUserMedia, running the '3A' processing chain — acoustic echo cancellation, automatic gain control, and noise suppression — encoding with Opus, transporting over encrypted SRTP with congestion control, and reconstructing playout through NetEQ's adaptive jitter buffer and concealment. Because it is an open standard with a high-quality reference implementation (libWebRTC) used by Chrome and most others, it gives every developer carrier-grade voice for free. Understanding its stages is the key to diagnosing echo, robotic audio, dropouts, and the other failures users report.
