
Once again, WebRTC Live broadcasted direct from the RTC.ON conference in Kraków, Poland where WebRTC.ventures CTO Alberto Gonzalez interviewed two speakers: And one guest: WebRTC Live #117: Live from RTC.ON 2026 Episode highlights below the video. Episode Highlights Pratim Mallick, Staff Software Engineer at Stream Charles Duyk,

The most frustrating thing about speaking with a Voice AI agent is when it interrupts you, or it doesn’t understand the normal flow of human conversation. There are multiple ways to solve this depending on your use case, and coming up with the best architecture for your

When WebRTC quality degrades on your SFU (jitter spikes, frozen frames, choppy audio), your monitoring tells you something is wrong. It doesn’t tell you what is responsible. Is the network dropping packets? Or is your server too loaded to read them from the socket on time? The

SIP and RTP feed a call and its media into a voice AI pipeline. What happens next runs through the control plane: what the voice agent is told, and what it decides to do about it. In my talk at ClueCon this year, From SIP to Tokens: