OpenAI Realtime 2
OpenAIRealtime2 is the engine marker for OpenAI’s GA Realtime API (the production endpoint that replaces the beta OpenAI-Beta: realtime=v1 channel). It targets gpt-realtime-2 by default and routes through OpenAIRealtime2Adapter — a dedicated adapter that speaks the GA session.update wire shape and performs bidirectional audio transcoding (mulaw 8 kHz ↔ PCM 24 kHz) required by the GA audio engine.
For the legacy beta endpoint and the lower-cost gpt-realtime-mini model, keep using OpenAIRealtime. The two engines coexist — pick OpenAIRealtime2 only when you specifically want the GA endpoint or the gpt-realtime-2 model.
The GA endpoint rejects the legacy
OpenAI-Beta: realtime=v1 header and expects output_modalities, nested audio.{input,output} blocks with MIME-type strings, and session.type = "realtime". These wire-shape differences are why GA needs its own adapter — the beta OpenAIRealtimeAdapter cannot reach gpt-realtime-2 reliably.When to use
Use OpenAIRealtime2 when… | Stick with OpenAIRealtime when… |
|---|---|
You want gpt-realtime-2 — strongest instruction following + 128K context + configurable reasoningEffort. | You’re on gpt-realtime-mini for cost / latency reasons. |
| You’re hitting the GA endpoint and the beta channel is being deprecated for your account. | You don’t need the GA wire shape and want to keep the existing adapter path. |
| You want the bidirectional PCM 24 kHz transcoding handled by the SDK rather than the model silently dropping mulaw frames. | Your audio is already PCM 24 kHz end-to-end and beta works for you. |
Quickstart
reasoningEffort: "low" is OpenAI’s recommended production tier for live voice — it gives the best instruction following without measurable per-turn latency.
Constructor
apiKey falls back to the OPENAI_API_KEY environment variable.
Reasoning effort
| Value | When to use |
|---|---|
"minimal" | Snappy turn-taking. Skips most reasoning. |
"low" | Recommended for production voice. Good instruction following without measurable per-turn latency. |
"medium" | Multi-step tool flows where the model should plan. Adds latency. |
"high" | Complex reasoning. Not recommended for live phone calls. |
session.reasoning = { effort: ... } into the GA session.update payload. When omitted, the field is not sent and OpenAI’s server default applies.
Streaming transcription
SetinputAudioTranscriptionModel to override audio.input.transcription.model. The same identifiers as the beta endpoint apply — see the streaming-transcription table on the OpenAI Realtime page for the full list (whisper-1, gpt-4o-mini-transcribe, gpt-4o-transcribe, gpt-realtime-whisper).
Tool-call preambles
gpt-realtime-2 treats preambles as first-class — the model speaks a short action sentence before a tool call when steered to. Set toolCallPreambles: true on phone.agent({ ... }) to prepend Patter’s built-in # Preambles guidance block to the session instructions, so a slow tool no longer leaves the caller in silence:
undefined / false (default) leaves the instructions byte-identical; a string overrides the block verbatim. See Tool-call preambles for the full behaviour and the per-tool sample-phrase nicety.
Audio path
The GA audio engine speaks PCM 24 kHz and silently drops mulaw frames. Patter handles the conversion transparently insideOpenAIRealtime2Adapter:
- Inbound (Twilio/Telnyx → model): mulaw 8 kHz → PCM 24 kHz
- Outbound (model → Twilio/Telnyx): PCM 24 kHz → mulaw 8 kHz
Speakerphone noise & false barge-in
On a speakerphone or in a noisy room, mouse clicks, the phone being picked up or set down, and background chatter can be mistaken for the caller speaking — the agent gets cut off mid-sentence. Because turn-taking is server-managed, you tune false barge-ins at the OpenAI VAD layer (no carrier-side change), not with a client gate.Input noise reduction
noiseReduction enables OpenAI’s native input noise reduction:
| Value | When to use |
|---|---|
"far_field" | Recommended for phone / speakerphone / conference audio. Filters room noise and distance. |
"near_field" | A handset held close to the mouth. |
undefined (default) | No reduction — today’s behaviour, field omitted entirely. |
session.audio.input.input_audio_noise_reduction.
Turn-detection tuning
turnDetection (RealtimeTurnDetection) is a readonly config. Each unset field falls back to the adapter default (server_vad, threshold 0.5, prefixPaddingMs 300, silenceDurationMs 300):
| Field | Applies to | Notes |
|---|---|---|
type | both | "server_vad" (default) or "semantic_vad". |
threshold | server_vad | 0..1; higher rejects more background noise. |
prefixPaddingMs | server_vad | Padding before detected speech. |
silenceDurationMs | server_vad | Trailing silence before end-of-turn. |
eagerness | semantic_vad | "low" lets the caller finish (least likely to interrupt), through "medium" / "high" / "auto". |
semantic_vad emits {type, eagerness} only — OpenAI rejects threshold / padding / silence on the semantic detector. Both knobs are also exposed directly on phone.agent({ openaiRealtimeNoiseReduction: ..., realtimeTurnDetection: ... }); an explicit agent() option wins over the engine marker value.
Direct adapter use
OpenAIRealtime2Adapter is exported and may be constructed directly when you need to share connection state across calls or override low-level fields. The constructor signature is positional (inherited from OpenAIRealtimeAdapter):
OpenAIRealtimeAdapter and overrides connect(), sendAudio(), receiveEvents(), and sendFirstMessage() for the GA wire shape.
Server-managed turn-taking
By default the GA adapter sets bothcreate_response: true and interrupt_response: true in session.update.turn_detection, so the OpenAI server owns turn-taking end to end: it runs VAD, decides end-of-turn, creates the response as soon as the caller stops speaking, and cancels its own response when the caller barges in. The input transcript (Whisper) is pure observability — it never gates or cancels the reply, so the transcription-model choice has no effect on reply latency.
On Patter’s WebSocket transport the client still does the bookkeeping the server cannot do for it: it clears the carrier playout buffer and sends conversation.item.truncate for the offset the caller actually heard (OpenAI auto-truncates only on WebRTC/SIP). It does not send a redundant response.cancel, run a client-side anti-flicker gate, or re-anchor turn metrics.
Tune false barge-ins (speakerphone / no-AEC PSTN where the agent’s own audio echoes into the input) with RealtimeTurnDetection — raise threshold or switch to semantic_vad with eagerness: "low" — rather than a client gate.
To restore the legacy client-managed path (Patter drives response.create / response.cancel and runs its own barge-in gate), set gateResponseOnTranscript: true on the engine marker or realtimeGateResponseOnTranscript: true on phone.agent(...). That emits create_response: false + interrupt_response: false and re-gates the reply on the transcript arriving — the escape hatch for no-AEC self-interruption scenarios.
Backward compatibility
- Existing
new OpenAIRealtime({...})callers are unaffected. The legacy engine continues to target the beta endpoint withgpt-realtime-minias the default. OpenAIRealtime2ships as an additive engine — no migration required. Pick it when you want the GA endpoint; otherwise stay where you are.- Pricing for
gpt-realtime-2is auto-resolved per model fromDEFAULT_PRICING.openai_realtime.models["gpt-realtime-2"]— see Metrics.
What’s Next
OpenAI Realtime (beta)
The legacy engine for
gpt-realtime-mini and earlier preview models.Engines
All engine classes side by side.
Agents
Configure system prompts, tools, and first messages.
Tools
Function calling inside a Realtime session.

