A real-time websocket transport implementation for interacting with Google's Gemini Multimodal Live API, supporting bidirectional audio and unidirectional text communication.
npm install \
@pipecat-ai/client-js \
@pipecat-ai/openai-realtime-webrtc-transport
The OpenAIRealTimeWebRTCTransport
is a fully functional RTVI Transport
. It provides a framework for implementing real-time communication directly with the OpenAI Realtime API using WebRTC voice-to-voice service. It handles media device management, audio/video streams, and state management for the connection.
- Real-time bidirectional communication with OpenAI Realtime API
- Input device management
- Audio streaming support
- Text message support
- Automatic reconnection handling
- Configurable generation parameters
- Support for initial conversation context
import { OpenAIRealTimeWebRTCTransport, OpenAIServiceOptions } from '@pipecat-ai/openai-realtime-webrtc-transport';
const options: OpenAIServiceOptions = {
api_key: 'YOUR_API_KEY',
session_config: {
instructions: 'you are a confused jellyfish',
}
};
const transport = new OpenAIRealTimeWebRTCTransport(options);
let RTVIConfig: RTVIClientOptions = {
transport,
...
};
interface OpenAIServiceOptions {
api_key: string; // Required: Your OpenAI API key
initial_messages?: Array<{ // Optional: Initial conversation context
content: string;
role: string;
}>;
session_config?: {
modailities?: string;
instructions?: string;
voice?:
| "alloy"
| "ash"
| "ballad"
| "coral"
| "echo"
| "sage"
| "shimmer"
| "verse";
input_audio_transcription?: {
model: "whisper-1";
};
temperature?: number;
max_tokens?: number | "inf";
};
}
// at setup time...
llmHelper = new LLMHelper({});
rtviClient.registerHelper("llm", llmHelper);
// the 'llm' name in this call above isn't used.
//that value is specific to working with a pipecat pipeline
// at time of sending message...
// Send text prompt message
llmHelper.appendToMessages({ role: "user", content: 'Hello OpenAI!' });
The transport implements the various RTVI event handlers. Check out the docs or samples for more info.
initialize()
: Set up the transport and establish connectionsendMessage(message)
: Send a text messagehandleUserAudioStream(data)
: Stream audio data to the modeldisconnectLLM()
: Close the connectionsendReadyMessage()
: Signal ready state
The transport can be in one of the following states:
- "disconnected"
- "initializing"
- "initialized"
- "connecting"
- "connected"
- "ready"
- "disconnecting
- "error"
The transport includes comprehensive error handling for:
- Connection failures
- WebRTC connection errors
- API key validation
- Message transmission errors
BSD-2 Clause