My name is Octavio Alvarado and I study Software Engineering at the Technological University of Panama, currently in my fifth year of my degree. I was part of the UTP’s Professional Internship program, which allows students to join companies according to their skills and profiles. Through this
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
Voice AI agents have unique deployment needs. Operational complexity multiplies quickly. You’re not just deploying code; you’re orchestrating real-time audio pipelines that need to maintain call quality under load, coordinate between AI services that each have their own scaling characteristics, and handle the networking complexities of audio
If you’ve worked with WebRTC applications long enough, you’ve probably heard these complaints: “The video froze, but I could still hear them.” “The audio kept cutting in and out.” “The call just dropped for no reason.” You check your application performance monitoring (APM) dashboard and everything looks
