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
