SmallWebRTCTransport enables peer-to-peer WebRTC connections between clients and your Pipecat application. It implements bidirectional audio and video streaming using WebRTC for real-time communication. This transport is intended for lightweight implementations, particularly for local development and testing. It expects your Pipecat server to include the corresponding SmallWebRTCTransport server-side implementation.

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Usage

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Basic Setup

import { PipecatClient } from "@pipecat-ai/client-js";
import { SmallWebRTCTransport } from "@pipecat-ai/small-webrtc-transport";

const pcClient = new PipecatClient({
  transport: new SmallWebRTCTransport({
    // Optional configuration for the transport
    iceServers: ["stun:stun.l.google.com:19302"],
  }),
  enableCam: false, // Default camera off
  enableMic: true, // Default microphone on
  callbacks: {
    // Event handlers
  },
});

await pcClient.connect({
  webrtcUrl: "/api/offer", // Your WebRTC signaling server endpoint
});

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API Reference

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Constructor Options

interface SmallWebRTCTransportConstructorOptions {
  iceServers?: RTCIceServer[];
  waitForICEGathering?: boolean;
  webrtcUrl?: string;
  audioCodec?: string;
  videoCodec?: string;
  mediaManager?: MediaManager;
}

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Properties

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TransportConnectionParams

export type SmallWebRTCTransportConnectionOptions = {
  webrtcUrl?: string;
};
On connect(), the SmallWebRTCTransport optionally takes a set of connection parameters. This can be provided directly to the PipecatClient’s connect() method or via a starting endpoint passed to the PipecatClient’s startBotAndConnect() method. If using an endpoint, your endpoint should return a JSON object matching the SmallWebRTCTransportConnectionOptions type, which currently expects a single webrtcUrl property. 
pcClient.startBotAndConnect({
  endpoint: '/api/start', // Your server endpoint to start the bot and return the webrtcUrl
});
// OR...
pcClient.connect({
  webrtcUrl: '/api/offer', // Your WebRTC offer/answer endpoint
});

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Methods

For most operations, you will not interact with the transport directly. Most methods have an equivalent in the PipecatClient and should be called from the PipecatClient. However, there are a few transport-specific methods that you may need to call directly. When doing so, be sure to access your transport via the transport property of the PipecatClient instance.
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setAudioCodec
method
Sets the preferred audio codec.
transport.setAudioCodec("opus");
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setVideoCodec
method
Sets the preferred video codec.
transport.setVideoCodec("VP8");

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Events

The transport implements the various PipecatClient event handlers.

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Connection Process

The connection process follows these steps:
  1. The transport negotiates a WebRTC connection with the corresponding pipecat transport, complete with transceivers for the media and a data channel for messaging.
  2. The transport sends a message to the pipecat transport to let it know it’s ready.
  3. The Pipecat transport sends a message letting the client know it is ready.

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Reconnection Handling

The transport includes automatic reconnection logic:
  • Up to 3 reconnection attempts after connection failures
  • Detection of ICE connection state changes
  • Graceful recovery from temporary disconnections
  • Graceful disconnect when reconnection attempts fail

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More Information

Video Transform Demo

Real-time video transformation example

Package

@pipecat-ai/small-webrtc-transport