WebRTC

Web Real-Time Communication protocol enabling peer-to-peer audio/video communication with minimal latency.

Definition by Hamming, the security layer for voice and chat agents. Drawing on 10M+ mins protected across 10K+ voice agents.

Expert-reviewed · 1 min read · Updated

Overview

Web Real-Time Communication protocol enabling peer-to-peer audio/video communication with minimal latency.

Why it matters

Essential for browser-based voice applications or when minimizing network latency is critical.

How it works

WebRTC functions through a series of API calls and event-driven processes. When triggered, it initiates a request-response cycle that processes data through defined protocols and interfaces.

Common issues

Organizations implementing WebRTC frequently encounter integration complexities, authentication issues, timeout configurations, and error handling scenarios. Common mistakes include inadequate retry logic, missing error boundaries, and insufficient logging for debugging. Automated testing and monitoring can help identify these issues before they impact production callers.

Implementation guide

Monitor WebRTC health: track connection establishment time, monitor packet loss and jitter, implement adaptive bitrate, and test across different network conditions.

Frequently asked
questions.

Browse every answer on the FAQ page