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Voice call

Archived SDK documentation

This page preserves the previous SDK documentation and its existing URL. The generated API reference was produced from SDK 0.1.5, source commit 4db9fbe925bf07a4b95e936a5bb172d6ab5820bb. Handbook behavior and compatibility claims below are historical descriptions, not a newly verified support guarantee. For the reviewed version-specific reference, see the SDK 0.1.5 overview.

startCall opens a full WebRTC voice session. The SDK handles WebRTC gateway connection, SIP plugin attachment, ICE trickle, JSEP negotiation, and reconnect-after-reload.

This mirrors the WebRTCPhone.tsx example in the SDK's React examples app.

Prerequisites​

  • An endpoint ID that supports voice_conversation.
  • A <audio> element to render the remote audio (and optionally one for local mic monitoring).
  • Browser permission for getUserMedia({ audio: true }).

Minimal example​

const remoteAudioEl = document.querySelector<HTMLAudioElement>('#remote-audio')!;
const localAudioEl = document.querySelector<HTMLAudioElement>('#local-audio');

delphi.setRemoteAudioElement(remoteAudioEl);
delphi.setLocalAudioElement(localAudioEl); // optional

const session = await delphi.startCall({
  endpointId: '24599c70-1e79-4e52-9819-e2d97acf45a5',
  autoDial: true,
});

autoDial: true skips an intermediate "registered, ready to dial" state — the SDK places the call as soon as registration completes.

Mid-call operations​

session.sendReadAloud('Important: your appointment is tomorrow.');

await delphi.sendDtmf('5');

await delphi.endCall();

sendReadAloud over a voice session pushes a TTS line over the same SIP leg the caller is on. Useful for notifications and interjections without reopening a separate audio_playback session.

Observe call state​

const off = session.subscribe(() => {
  const s = session.getState();
  console.log(s.connected, s.lastMessage);
});

// Aggregated, client-level view:
delphi.subscribe(() => {
  const s = delphi.getState();
  console.log(s.voiceCall);
  // { inCall, calling, registered, telproDomain, ... }
});

Cleanup​

await delphi.endCall();
off();

endCall() tears down the SIP leg and the peer connection, and removes the voice_conversation session from the session map. The audio_playback session (if any) for the same endpoint is unaffected.

Reconnect after reload​

Voice-call state is persisted to sessionStorage for about 20 seconds after disconnect, so a page reload during a call can resume cleanly. See Reconnect and persistence.

Common gotchas​

  • Audio elements must be attached before startCall resolves. Otherwise the SDK's WebRTC track binding has nowhere to land.
  • Browsers will block getUserMedia outside a user gesture in some contexts; place startCall inside a click handler.
  • If iceServers are omitted, the SDK derives them from telproDomain returned in the session-token response. Check that your backend includes it.

Next step​

Browser actions — handle the AI-driven UI side-flows (navigate, show_alert, custom flows).