WebRTC’s code may be open source and royalty-free, but deploying a reliable, production-grade real-time communications application involves both up-front development investment (engineering, integration, and setup) and ongoing operational spending (infrastructure, bandwidth, monitoring, and scaling). Teams who overlook either side risk major surprises as their product grows. So
Voice AI applications need real-time and reliable audio communication for natural conversations with AI customer service bots, virtual assistants, IVR platforms, and other voice-enabled systems. Choosing the appropriate transport protocol is crucial for teams, as using the wrong one can lead to choppy audio, noticeable delays, and
WebRTC has been enabling video and audio communication directly in your browser without any plugins for 10 years now. Even services like Google Meet and Discord use WebRTC to provide crystal-clear voice and video calls in real-time. This powerful technology has revolutionized how we connect online, but
Last week I attended the RTC.ON 2025 conference in Krakow, Poland, alongside my colleagues Alberto González Trastoy and Alfred González Trastoy from our WebRTC.ventures team. The conference is in its third year, and this was my second time attending. RTC.ON has quickly become one of the premier
