WebRTC VoIP systems enable voice calling directly through web browsers and mobile apps without requiring any software downloads or plugins. This makes them ideal for customer support platforms, telehealth consultations, sales calls, and any application where you want to add voice communication without asking people to install
When WebRTC was first introduced by Google over a decade ago, it came with the promise of simplicity. “Just drop in a little JavaScript and you’ve got video chat in the browser with no downloads necessary!” While that vision helped kickstart a wave of innovation in real-time
WebRTC applications present unique operational challenges that traditional monitoring tools cannot address. Unlike conventional web applications, real-time communication systems operate with complex peer-to-peer connections, dynamic network conditions, and media processing pipelines that can fail silently or degrade gradually. The primary challenge lies in observability. WebRTC applications generate
Real-time video communication applications face unique scalability challenges that can make or break the user experience. When thousands of users simultaneously join virtual classrooms, video conferences or other streaming video experiences, traditional autoscaling approaches often fall short. The key to managing predictable traffic spikes in WebRTC applications
