Quality, Testing & Rescue

A WebRTC rescue is what happens when a live video product is failing — calls drop, connections hang on “connecting,” audio goes one-way — and the team that owns the code can’t find the root cause. Usually the codebase was inherited or built by a team that has moved on, and the engineers left holding it are […]

You’ve built a real-time communication feature that includes video calls, screen sharing, and live collaboration. It worked in the demo. But now users are dropping calls, mobile devices fail to connect, and your engineering team has spent three sprints chasing ghosts in a codebase nobody fully understands anymore. This is the moment many CTOs reach […]

The worst way to find out your WebRTC platform has a quality problem is from users. By the time someone files a support ticket or leaves a negative review, dozens of other users have quietly given up, refreshed the page, or — in healthcare — abandoned the consultation entirely. Reactive monitoring in real-time media systems […]

Introduction WebRTC (Web Real-Time Communication) has revolutionized how we communicate online by enabling real-time audio, video, and data sharing directly through web browsers. However, achieving a smooth and seamless experience in WebRTC applications can be challenging due to various network issues, one of the most common being Jitter in WebRTC. Understanding and managing jitter is […]

The Problem Imagine you are a technological startup building a next generation video communication tool. Probably you have used modern cPaaS systems like Agora or Twilio to manage video streams. Or maybe you even went into the open-source world by using let’s say Janus server.  Finally, your platform is running on a stage server and […]