WebRTC was built for browsers, but many WebRTC connections now run through server programs, for example SFUs, gateways, cloud gaming, recording, or AI agents that join calls. Without a browser to handle the connection, each of these programs needs its own WebRTC implementation, usually a library built
If you run WebRTC at scale, a large share of your media goes through TURN relays. Restrictive NATs, enterprise firewalls, and clients that hide their IP addresses all push sessions onto relayed candidates, and when TURN server performance drops under load, call quality drops with it. This
On September 22–23, I was in Chicago for the IIT Real-Time Communications Conference at Illinois Tech, an annual IEEE conference where industry engineers and academic researchers present work on real-time communications. This year’s tracks covered programmable networks, WebRTC applications, emergency communications, and AI in real-time communications. I
A production voice agent has a latency budget of roughly 800 milliseconds before a conversation starts to feel slow. That budget is spent across eight sequential stages, typically split across three to five different vendors depending on how the stack is assembled. This post maps that voice
