Realtime transport

Use this page to decide how realtime agents fit into your Python application.

!!! note "Python SDK boundary"

The Python SDK does **not** include a browser WebRTC transport. This page is only about Python SDK transport choices: server-side WebSockets and SIP attach flows. Browser WebRTC is a separate platform topic, documented in the official [Realtime API with WebRTC](https://developers.openai.com/api/docs/guides/realtime-webrtc/) guide.

Decision guide

Server-side WebSocket is the default Python path

RealtimeRunner uses OpenAIRealtimeWebSocketModel unless you pass a custom RealtimeModel.

That means the standard Python topology looks like this:

1. Your Python service creates a RealtimeRunner.
2. await runner.run() returns a RealtimeSession.
3. Enter the session and send text, structured messages, or audio.
4. Consume RealtimeSessionEvent items and forward audio or transcripts to your application.

This is the topology used by the core demo app, the CLI example, and the Twilio Media Streams example:

• examples/realtime/app
• examples/realtime/cli
• examples/realtime/twilio

Use this path when your server owns the audio pipeline, tool execution, approval flow, and history handling.

SIP attach is the telephony path

For the telephony flow documented in this repository, the Python SDK attaches to an existing realtime call via call_id.

This topology looks like:

1. OpenAI sends your service a webhook such as realtime.call.incoming.
2. Your service accepts the call through the Realtime Calls API.
3. Your Python service starts a RealtimeRunner(..., model=OpenAIRealtimeSIPModel()).
4. The session connects with model_config={"call_id": ...} and then processes events like any other realtime session.

This is the topology shown in examples/realtime/twilio_sip.

The broader Realtime API also uses call_id for some server-side control patterns, but this repository's shipped attach example is SIP.

Browser WebRTC is outside this SDK

If your app's primary client is a browser using Realtime WebRTC:

• Treat it as outside the scope of the Python SDK docs in this repository.
• Use the official Realtime API with WebRTC and Realtime conversations docs for the client-side flow and event model.
• Use the official Realtime server-side controls guide if you need a sideband server connection on top of a browser WebRTC client.
• Do not expect this repository to provide a browser-side RTCPeerConnection abstraction or a ready-made browser WebRTC sample.

This repository also does not currently ship a browser WebRTC plus Python sideband example.

Custom endpoints and attach points

The transport configuration surface in [RealtimeModelConfig][agents.realtime.model.RealtimeModelConfig] lets you adapt the default paths:

• url: Override the WebSocket endpoint
• headers: Provide explicit headers such as Azure auth headers
• api_key: Pass an API key directly or via callback
• call_id: Attach to an existing realtime call. In this repository, the documented example is SIP.
• playback_tracker: Report actual playback progress for interruption handling

See the Realtime agents guide for the detailed lifecycle and capability surface once you've chosen a topology.