OpenAI Splits Transcription Into Four Model IDs: The Routing Decision Every Audio Pipeline Needs
OpenAI launched gpt-transcribe and gpt-live-transcribe on July 28, replacing gpt-4o-transcribe. The two models split across endpoint, price, and capability — diarization, timestamps, and live stream each route to a different model ID.
Archive item produced with AI assistance from the cited source and published without individual review. Editor of record: Joe Werner.

OpenAI launched two new speech-to-text models on July 28 — gpt-transcribe and gpt-live-transcribe — and immediately made them the recommended defaults for all new transcription work. What looks like a model update is actually a routing architecture change: OpenAI has split transcription into a matrix of four model IDs, each covering a different capability set, and two of those capabilities no longer exist in the new defaults at all.
If your audio pipeline was built around gpt-4o-transcribe, your code still works today — but the routing table you're relying on has been replaced.
What changed and what each model ID now covers
Before July 28, gpt-4o-transcribe was the recommended model for all transcription work — file uploads, streaming, and realtime audio. It accepted the /v1/audio/transcriptions endpoint and covered most use cases from a single model name.
The new model matrix:
| Model ID | Endpoint | Price | Best for |
|---|---|---|---|
gpt-transcribe | v1/audio/transcriptions | $0.0045/min | Recorded files, streamed file transcripts, committed Realtime turns |
gpt-live-transcribe | v1/realtime/transcription_sessions | $0.017/min | Live audio streams, real-time transcript deltas |
gpt-4o-transcribe-diarize | v1/audio/transcriptions | (separate pricing) | Speaker-labeled transcripts |
whisper-1 | v1/audio/transcriptions | (token-based) | Word timestamps, SRT/VTT subtitles, audio translation |
The routing consequence: you now need to know the capability you want before you pick a model. The old gpt-4o-transcribe let you request features and the endpoint handled them. The new architecture puts each capability in a distinct model ID.
The endpoint split is the breaking change
gpt-live-transcribe does not support v1/audio/transcriptions. It only works through v1/realtime/transcription_sessions — OpenAI's WebSocket-based Realtime API. If your pipeline opens a file, uploads it, and expects a transcript back synchronously, gpt-live-transcribe will not serve that request.
Conversely, gpt-transcribe does not support the Realtime endpoint. It is a request-response model: send a file, get text back. It does work in the Realtime endpoint for one specific sub-case — transcribing committed turns (what the user already said in a voice session) — but live streaming transcript deltas require gpt-live-transcribe.
This matters for operators building voice agents. If you are running a WebSocket voice session and need partial transcripts while the user is still speaking, you need gpt-live-transcribe on the Realtime endpoint. If you only need to read back what the user said after each turn, gpt-transcribe on the Realtime transcription endpoint is cheaper and sufficient.
Pricing gap: 3.8x between file and live
gpt-transcribe costs $0.0045 per minute of audio. gpt-live-transcribe costs $0.017 per minute — roughly 3.8× more expensive for the same underlying speech content.
The premium is for latency: gpt-live-transcribe streams partial transcripts as they arrive, keeping connection state across the audio session. If your use case does not require partial transcripts before a speaker finishes — if you just need an accurate transcript of what was said, at the end — gpt-transcribe is the correct routing decision at a significant cost reduction.
For operators running call center analytics, meeting recorders, or podcast transcription pipelines: those workloads almost never need streaming partials. The work happens after the audio is captured. Every one of those pipelines should route to gpt-transcribe, not gpt-live-transcribe.
What whisper-1 still owns
Neither new model supports word-level timestamps, SRT output, VTT subtitles, or audio translation into English. Those capabilities remain exclusively on whisper-1. If you are building subtitle pipelines, captioning workflows, or any output that requires a timestamp per word, the new models are not a migration target.
This is the most common mistake to check in an existing integration: teams that were using gpt-4o-transcribe with timestamp_granularities: ["word"] or response_format: "srt" will need to keep whisper-1 for those calls, not upgrade to gpt-transcribe.
The context parameters are new and improve accuracy
gpt-transcribe and gpt-live-transcribe add two parameters that were not available on gpt-4o-transcribe:
keywords: an array of terms the model should prefer when the audio is ambiguous — model names, product SKUs, proper nouns, technical jargonlanguages: a list of language codes to hint expected input language; the model can handle code-switching within a clip
The prompt field continues to work as an unstructured context string. The new keywords parameter is structured and carries more weight for specific terms. For domain-specific audio — customer support calls mentioning product IDs, engineering meetings using API names — the keywords array is worth adding to existing integrations even before any other migration work.
Migration path from gpt-4o-transcribe
For most file transcription work:
# Before
transcription = client.audio.transcriptions.create(
model="gpt-4o-transcribe",
file=audio_file
)
# After
transcription = client.audio.transcriptions.create(
model="gpt-transcribe",
file=audio_file,
keywords=["your-domain-terms"] # new, improves accuracy
)
For live audio streams, the endpoint itself changes:
# gpt-live-transcribe requires v1/realtime/transcription_sessions
# Not v1/audio/transcriptions — these are distinct WebSocket sessions
Teams that hardcode the endpoint URL (not just the model name) need to verify that their endpoint matches the model. Sending gpt-live-transcribe to /v1/audio/transcriptions will fail; sending gpt-transcribe to /v1/realtime for streaming transcripts will return committed-turn results only, not partial deltas.
What routing teams should watch
The GPT Transcribe launch signals that OpenAI is treating audio as a capability cluster rather than a single model. Expect this matrix to grow — speaker diarization is already in a separate model (gpt-4o-transcribe-diarize), and translation remains in whisper-1. Planning a voice pipeline today means planning a routing policy across multiple model IDs, not a single alias.
For teams routing to OpenAI through an AI gateway, the model-to-endpoint mapping needs to be verified. A proxy that accepts any model name and forwards it to a fixed /v1/audio/transcriptions endpoint will silently mis-route gpt-live-transcribe requests. Operator-level routing rules for audio endpoints are now distinct from text completion routing.

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