These packages give you the full pipeline to turn microphone or phone audio into an AI agent response and back — speech-to-text, MCP tool calls, text-to-speech, and telephony or WebRTC transport, all orchestrated with per-stage latency budgets. You'd adopt them to build a production voice agent without writing the audio plumbing, provider switching, or session management yourself. The most distinctive thing is that the agent logic lives entirely in an external MCP server, so the pipeline is a pure transport layer that can be swapped between Twilio, WebRTC, or a local simulator without changing the agent.
A scaffolding CLI that generates a complete voice-agent-kit project with pipeline configuration, STT/TTS provider setup, telephony or WebRTC transport, an MCP client, and a ready-to-run server.
A Zod-validated configuration system and pipeline orchestrator for building voice-enabled AI agents, providing a `createPipeline()` function that coordinates STT, MCP, and TTS stages with latency enforcement, session management, and OpenTelemetry observability.
A JSON-RPC 2.0 client that connects to any MCP server endpoint, providing tool discovery, conversation history management, retry with backoff, and TTS-safe response sanitization. It exposes an `MCPClient` class with `connect()`, `sendRequest()`, `discoverTools()`, and `close()` methods, and requires only a `fetch`-compatible runtime.
A CLI and programmatic simulator that runs a voice agent pipeline (STT → MCP → TTS) locally from a WAV file or live microphone, reporting per-turn latency without requiring Twilio or cloud telephony. Exports `createSimulator`, `Simulator`, `renderLatencyWaterfall`, `readWavFile`, `writeWavFile`, `captureMicrophone`, and `playAudio`.
Provider-agnostic speech-to-text interface with a unified `STTProvider` class and seven adapter implementations (Deepgram, AWS Transcribe, Google Cloud Speech-to-Text, OpenAI Realtime, OpenAI Whisper, AssemblyAI, Groq Whisper), plus built-in audio format conversion between mulaw and linear16 with sample rate resampling.
A WebSocket handler for voice AI agents that normalizes bidirectional streaming protocols (start/media/stop/mark/DTMF) across Twilio, Telnyx, SignalWire, and Vonage, providing barge-in detection, audio buffering, and base64 encoding/decoding. Exports a `TwilioMediaStreamHandler` class (EventEmitter) with methods like `acceptConnection`, `sendAudio`, `clearAudio`, and `sendMark`, plus static `encodeForTwilio`/`decodeFromTwilio` utilities.
Provider-agnostic text-to-speech interface with five adapter implementations (Deepgram Aura, AWS Polly, Google Cloud TTS, ElevenLabs, Cartesia), returning streaming audio as `AsyncIterable<AudioChunk>` with cancelable synthesis and Twilio-ready audio formatting.
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.
