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The OpenAI Realtime API lets developers build interactive voice apps that exchange speech, text, images, and audio with a multimodal model in real time. Its documented connection options are WebRTC, WebSocket, and SIP. For a first browser-based voice agent, the key decisions are how to connect, what to put in the session configuration, and how the first audio response affects voice settings.

What the Realtime API enables

OpenAI describes the Realtime API as a way to “Communicate with a multimodal model in real time over low latency interfaces like WebRTC, WebSocket, and SIP.” The API supports speech-to-speech interaction as well as text, image, and audio input and output. That makes it a foundation for interactive voice apps and other multimodal applications, rather than only a transcription-and-text-response workflow. OpenAI Realtime API reference

Choose a connection path

The API reference names three interfaces. The documentation establishes that all three are supported; it does not establish that one is universally best or provide a complete deployment comparison. Choose according to the connection your application needs, then consult the corresponding current API instructions for its specific setup and operational requirements.

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Interface What the documentation establishes What to verify for your implementation
WebRTC Supported low-latency interface. The reference documents creating a call by posting an SDP offer to POST /v1/realtime/calls; the request may include session configuration, and the response supplies an SDP answer. OpenAI Realtime API reference How your client and server exchange the offer and answer, and which session settings to include.
WebSocket Supported low-latency interface for communicating with the Realtime API. OpenAI Realtime API reference Use the WebSocket-specific connection and event guidance in the API reference; the available sources here do not establish a full topology comparison.
SIP Supported low-latency interface. OpenAI Realtime API reference Consult the current SIP documentation for call integration details and requirements.

Understand the first session configuration

A Realtime session can specify audio input and output settings, instructions, the model, output modalities, and a voice, among other fields. Treat these as session-level choices: decide what your agent should receive and produce, and what instructions should govern its responses, before starting the interaction. Check the live reference for accepted fields and currently supported model options, which can change. OpenAI Realtime API reference

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Choose a voice before audio is generated

The API reference lists built-in voices including alloy, ash, ballad, coral, echo, sage, shimmer, verse, marin, and cedar; it recommends marin and cedar. Voice cannot be changed after the session has generated audio, so settle on a voice before the first spoken response. Confirm the available choices in the current reference. OpenAI Realtime API reference

Know what server-side voice activity detection does

The documentation describes server-side voice activity detection as activating when user speech is detected and stopping after a period of silence. This explains its basic role in a voice session; the reference should be consulted for current settings and event behavior rather than assuming particular thresholds or interruption handling. OpenAI Realtime API reference

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Set up the SDK and start from the quickstart

OpenAI’s developer quickstart identifies the official TypeScript/JavaScript SDK for server-side JavaScript environments including Node.js, Deno, and Bun, and presents Realtime as an option for interactive voice and multimodal apps. Use the quickstart for SDK and API setup, then follow the Realtime reference for connection-specific details, session configuration, and events; the quickstart is not a complete Realtime tutorial. OpenAI developer quickstart

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Implementation checklist

  • Select WebRTC, WebSocket, or SIP based on the connection your app requires; do not assume the same setup applies to every interface.
  • For WebRTC, create the call with an SDP offer at POST /v1/realtime/calls, optionally include session configuration, and use the returned SDP answer.
  • Decide on model, instructions, modalities, and audio settings using the current API reference.
  • Set the desired voice before the session produces audio.
  • Follow the current connection-specific documentation for details not established here, including event handling and operational requirements.

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