Web

@aurix/web-sdk for browsers and Electron — WebTransport, WebRTC and WebSocket transports, Opus in a worklet, device handling, effects.

Install#

npm i @aurix/web-sdk

Ships as ESM with types. Requires a secure context (HTTPS or localhost) for microphone access.

Connect and join#

import { AurixClient } from '@aurix/web-sdk';

const t = await fetch('/voice/token', { method: 'POST' }).then((r) => r.json());   // your backend
const voice = new AurixClient({
  apiUrl: 'https://eu-central.cell.aurix.run',   // your project's cell endpoint (Keys & SDK)
  wsUrl: t.endpoint.ws_url,
  token: t.token,
  refreshToken: () => fetch('/voice/token', { method: 'POST' }).then((r) => r.json()).then((x) => x.token),
  transport: 'auto',      // webtransport → webrtc → websocket; or pin one
});

await voice.connect();
await voice.joinChannel(t.channels[0].id);

connect() resolves with the SessionInfo; the microphone is requested lazily on the first join unless you call voice.setInputDevice() earlier. Attach playback to an element you control (voice.attachAudioOutput(audioEl)) or let the SDK create one; call voice.resumeAudio() from a user gesture if autoplay is blocked.

Transports#

Read voice.mediaTransport after connecting to show it in diagnostics; voice.on('mediaTransport', …) fires when it changes.

Events#

voice.on('participantJoined', ({ channelId, participant }) => …);   // also participantLeft / participantUpdated
voice.on('speaking', ({ userId, speaking }) => …);
voice.on('stats', (s) => console.log(s));                             // RTT, loss, jitter, MOS
voice.on('networkQuality', (q) => …);
voice.on('sessionClosed', ({ reason }) => …);                         // connectionState for the full state machine

Positional audio#

voice.updatePosition(channelId, { x, y, z }, { yaw, pitch, roll });   // 10–20 Hz

The SDK renders remote participants with the Web Audio panner; voice.channelAudioPolicy(channelId) tells you what the server configured.

Devices, mute, levels#

await voice.setInputDevice(deviceId);  await voice.setOutputDevice(deviceId);
voice.setInputGain(1.2);  voice.setOutputVolume(0.8);  voice.setOutputMuted(false);
voice.setTransmission('ptt');  voice.transmitToChannel(teamChannelId);
voice.setParticipantVolume(userId, 0.5);  voice.setParticipantMuted(userId, true);

Effects (setVoiceEffects), visemes (setVisemes), audio injection (injectAudio for bots and tests) and server-side ducking are available on the same client — see the typed API in the package.

Electron and kiosks#

Works unchanged; WebTransport needs Chromium ≥ 97 and a certificate the node advertises (handled automatically with the hash-pinned certificates Aurix nodes issue).

Channel patterns#

The transmit channel is a client-side choice: join several, send to one, hear all with receive. Positional audio needs positions roughly 10 times a second; between updates the node interpolates.

Latency and quality#

Expect 30–60 ms mouth-to-ear on the same continent over QUIC, more over TURN. Opus runs at 32 kbps with DTX by default; the spending ladder may lower it to 16 kbps when your forecast exceeds the limit. Server-side noise suppression is on by default and costs nothing on the client.

Testing without a full build#

• Two players: the rooms demo in a browser plus your build — they share a cell, so a token from your backend joins the same channel by name.
• Mic path: an echo channel plays your own voice back through the real pipeline.
• Network: throttle UDP on one machine and watch the transport fall back to WebRTC / WebSocket in the SDK's connection state.

Troubleshooting#