SIP vs WebRTC for AI voice agents

Every AI voice agent eventually has to answer a real phone. That means SIP — the protocol carriers, PBXs, and desk phones speak. But your web and mobile apps speak WebRTC. The question is not which one to pick; it is where to bridge them.

SIP: the phone network’s API

SIP carries calls between carriers, trunks, and devices. It is how a call to a phone number reaches your agent, and how your agent transfers a caller to a person. You get numbers, porting, STIR/SHAKEN, and E911 — and decades of interoperability.

WebRTC: the browser’s media stack

WebRTC gives browsers and apps encrypted, low-latency audio and video with built-in echo cancellation and NAT traversal. It is ideal for softphones, in-app calling, and conferences.

Bridge once, at the gateway

DialPress anchors media at the gateway and bridges WebRTC and SIP there, so the agent sees one call no matter where it came from. Your code handles a single INVITE; the gateway handles codecs, ICE, and DTLS-SRTP.

What that means for latency

Bridging at the edge — next to the speech engine and the agent — keeps the media path short. Every hop you remove is time back in the turn-taking budget, which is what makes an agent feel human.