Connect with LiveKit WebRTC or SIP

Connect a browser with WebRTC#

Request WebRTC on the server, then pass only the returned descriptor to the browser. The adapter uses the official LiveKit client and enables microphone publishing by default.

// Trusted server
const session = await ox.realtime.sessions.create({
  model: "ox/voice-agent",
  transport: "webrtc",
  identity: signedInUser.id,
});

// Browser, called from a user gesture
import { connectLiveKitRealtime } from "@ox/sdk/browser";

const room = await connectLiveKitRealtime(session);

Import the same adapter from @ox/sdk/livekit when a dedicated subpath is preferable. Private-preview applications without package installation can import it from https://www.amerint.co/voice-sdk/ox-livekit.js. Pass publishMicrophone: false if the application will publish its own audio track. The participant token is scoped to one room and microphone publishing.

Place an outbound SIP call#

Outbound telephony uses the same room and bridge. The server validates E.164 destinations and dispatches the model participant before dialing.

const call = await ox.realtime.calls.create({ to: "+14155550100" });