WebRTC

WebRTC is a browser and mobile-native protocol for real-time audio and video, built to handle the parts that make live audio hard over an ordinary network connection: adapting to changing network conditions, recovering from a dropped packet, reconnecting cleanly, all without a browser plugin. For a voice agent, it is what lets a web page or an app stream a caller's microphone audio directly, with no phone call involved at all.

An enterprise's own product, a customer portal, a mobile app, often needs a voice agent embedded directly in its own interface, not routed out through a phone number and back in through a carrier.

A customer clicks talk to support inside a web app and starts talking through their microphone immediately, with no phone number dialed and no call placed on any carrier.

In Dynamiq, web and mobile callers connect to a voice agent over WebRTC through a client SDK that captures the microphone, and it is what the platform's own browser test tab uses to let you talk to an agent before deploying it. A backend that already holds the audio itself, a SIP trunk or a telephony stack, uses the WebSocket transport instead.