For the third year in a row, I had the pleasure of attending the RTC.ON conference where our company WebRTC.ventures was a sponsor. As usual, the panel was stacked with great talks and speakers. The topics ranged from AI, computer vision, voice, audio, telephony, broadcast and transport
WebRTC and SIP both carry live voice to an AI agent. Voice AI platforms list them side by side, so teams building an AI receptionist often expect to pick one or the other for connecting callers to the agent. In practice, each protocol serves a different way
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
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
