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reaatechREAATECH

@reaatech/voice-agent-webrtc

npm v0.1.0

A WebSocket-based transport class (`WebRTCTransport`) for browser voice AI agents that handles Opus encode/decode, PCM resampling, and barge-in detection, paired with standalone PCM utility functions. Requires a WebSocket connection and optionally `@discordjs/opus` or `prism-media` for Opus support.

npm →source →

@reaatech/voice-agent-webrtc

npm version License: MIT CI

Status: Pre-1.0 — APIs may change in minor versions. Pin to a specific version in production.

Browser transport for voice AI agents over a WebSocket connection, with Opus encode/decode and PCM resampling helpers. Use this as the server-side counterpart to a browser client that streams microphone audio in, and plays agent audio out.

Installation

terminal
npm install @reaatech/voice-agent-webrtc
pnpm add @reaatech/voice-agent-webrtc

Opus codec (optional)

Opus encode/decode requires a native codec binding. Install one of the following only if you need Opus (raw PCM transport works without it):

terminal
pnpm add @discordjs/opus   # preferred — prebuilt native addon
# or
pnpm add prism-media

Both are optional and loaded lazily at runtime. isOpusAvailable() returns false when neither is installed, and encodeOpus / decodeOpus throw a clear error pointing you here.

Feature Overview

  • WebSocket transport — WebRTCTransport manages a browser audio session over ws
  • Barge-in detection — energy + confidence thresholds interrupt agent playback when the caller speaks
  • Opus codec — encodeOpus / decodeOpus with lazy-loaded native bindings
  • PCM utilities — resample, convertSampleFormat, interleaveToMono, monoToInterleave, changeVolume

Quick Start

typescript
import { WebRTCTransport } from '@reaatech/voice-agent-webrtc';
 
const transport = new WebRTCTransport(socket, {
  outputSampleRate: 48000,
  outputChannels: 1,
  bargeInEnabled: true,
  minSpeechDuration: 200,
  confidenceThreshold: 0.6,
  silenceThreshold: 500,
  frameDurationMs: 20,
});

API Reference

WebRTCTransport

typescript
class WebRTCTransport {
  constructor(socket: WebSocket, config: WebRTCTransportConfig);
}

Codec & PCM helpers

typescript
import {
  isOpusAvailable,
  encodeOpus,
  decodeOpus,
  resample,
  convertSampleFormat,
  interleaveToMono,
  monoToInterleave,
  changeVolume,
} from '@reaatech/voice-agent-webrtc';
FunctionDescription
isOpusAvailable()true if a native Opus binding is installed
encodeOpus(pcm, sampleRate, channels)PCM Int16 → Opus packet
decodeOpus(opus, sampleRate, channels)Opus packet → PCM Int16
resample(pcm, fromRate, toRate)Linear PCM resampling
convertSampleFormat(...)Convert between PCM sample formats
interleaveToMono / monoToInterleaveChannel layout conversion
changeVolume(pcm, gain)Apply a gain multiplier

Related Packages

License

MIT

All packagesvoice-agent-kit overview