> ## Documentation Index
> Fetch the complete documentation index at: https://docs.inworld.ai/llms.txt
> Use this file to discover all available pages before exploring further.
# Intro to Realtime STT
> Transcribe audio to text using leading STT providers through a single API.
The Realtime Speech-to-Text (STT) API provides a unified integration point for industry-leading transcription providers. You get consistent authentication, request formatting, and response handling across providers — without managing multiple SDKs or credentials.
The API supports both synchronous transcription for complete audio files and real-time bidirectional streaming over WebSocket for live audio.
Make your first STT API call and get a transcript.
View the complete API specification.
Browse ready-to-use GitHub samples for sync and real-time STT.
## Supported Providers
### Inworld (first-party) — Experimental
| **Model ID** | **Endpoints** | **Best for** |
| :---------------------- | :------------------- | :---------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| `inworld/inworld-stt-1` | Sync API + WebSocket | Voice agents and character-driven apps that benefit from transcription plus Voice Profile (age, pitch, emotion, vocal style, accent) and configurable turn-taking |
The Inworld first-party model is currently Experimental. Features and pricing are subject to change.
Supports English plus 29 additional languages in experimental mode. See [Supported Languages](#supported-languages) for the full list.
### Groq
| **Model ID** | **Endpoints** | **Best for** |
| :---------------------- | :------------ | :----------------------------------------------- |
| `groq/whisper-large-v3` | Sync API only | General-purpose transcription for recorded audio |
### AssemblyAI
| **Model ID** | **Endpoints** | **Best for** |
| :-------------------------------------------- | :------------- | :--------------------------------------------------------------------------------------------------------------- |
| `assemblyai/universal-streaming-multilingual` | WebSocket only | Multilingual streaming (English, Spanish, French, German, Italian, Portuguese) |
| `assemblyai/universal-streaming-english` | WebSocket only | English-optimized streaming |
| `assemblyai/u3-rt-pro` | WebSocket only | High-accuracy, sub-300ms latency, multilingual streaming (English, Spanish, French, German, Italian, Portuguese) |
| `assemblyai/whisper-rt` | WebSocket only | Real-time Whisper transcription |
AssemblyAI models currently support the WebSocket streaming endpoint only. Sync HTTP support is coming soon.
### Soniox
| **Model ID** | **Endpoints** | **Best for** |
| :----------------- | :------------- | :--------------------------------------------------------------------------------------------- |
| `soniox/stt-rt-v4` | WebSocket only | High-accuracy real-time streaming with semantic end-of-turn detection and multilingual support |
Soniox models currently support the WebSocket streaming endpoint only.
For pricing details, see [Billing](/stt/resources/billing) or [inworld.ai/pricing](https://inworld.ai/pricing).
## Model comparison
| **Feature** | **inworld/inworld-stt-1** | **groq/whisper-large-v3** | **assemblyai/universal-streaming-multilingual** | **assemblyai/universal-streaming-english** | **assemblyai/u3-rt-pro** | **assemblyai/whisper-rt** | **soniox/stt-rt-v4** |
| :------------------ | :----------------------------------------------------------- | :----------------------------------------------- | :----------------------------------------------------------------------------- | :----------------------------------------- | :------------------------------------------------------------------------------------------------------ | :---------------------------------------- | :--------------------------------------------------------------------------------------------- |
| Pricing | [See pricing](https://inworld.ai/pricing) | [See pricing](https://inworld.ai/pricing) | [See pricing](https://inworld.ai/pricing) | [See pricing](https://inworld.ai/pricing) | [See pricing](https://inworld.ai/pricing) | [See pricing](https://inworld.ai/pricing) | [See pricing](https://inworld.ai/pricing) |
| Endpoint | Sync API + WebSocket | Sync API only | WebSocket only | WebSocket only | WebSocket only | WebSocket only | WebSocket only |
| Real-time streaming | | | | | | | |
| Best for | Voice agents with Voice Profile and configurable turn-taking | General-purpose transcription for recorded audio | Multilingual streaming (English, Spanish, French, German, Italian, Portuguese) | English-optimized streaming | High-accuracy, sub-300ms multilingual streaming (English, Spanish, French, German, Italian, Portuguese) | Real-time Whisper transcription | High-accuracy real-time streaming with semantic end-of-turn detection and multilingual support |
| Languages | English; 29 Experimental ([see below](#supported-languages)) | 100+ (Whisper) | 6 languages | English | 6 languages | 100+ (Whisper) | Multilingual |
## Supported Audio Formats
| **Format** | **Sync API** | **WebSocket Streaming** |
| :--------------- | :------------------------------ | :------------------------------ |
| `LINEAR16` (PCM) | | |
| `MP3` | | |
| `OGG_OPUS` | | |
| `FLAC` | | |
| `AUTO_DETECT` | | |
Recommended defaults: 16,000 Hz sample rate, 16-bit depth, mono. For container formats (MP3, FLAC, OGG\_OPUS, WAV), `sampleRateHertz` is optional — the API auto-detects it from the file header.
STT performs best with 16 kHz audio. Lower sample rates (such as 8 kHz telephony audio) contain fewer data points for the model to interpret, which reduces transcription accuracy. Upsampling low-sample-rate audio does not improve quality — it only interpolates between existing samples without adding new information.
## Endpoints
| **Endpoint** | **Method** | **Description** |
| :--------------------------------------------------------------------------------------------------------- | :--------- | :------------------------------------------------------------------------------- |
| [`/stt/v1/transcribe`](/api-reference/sttAPI/speechtotext/transcribe) | POST | Send complete audio, receive full transcript |
| [`/stt/v1/transcribe:streamBidirectional`](/api-reference/sttAPI/speechtotext/transcribe-stream-websocket) | WebSocket | Stream audio in real time, receive transcription chunks as they become available |
## Supported Languages
Language support depends on the STT provider. See [Model comparison](#model-comparison) above for more details.
### Inworld first-party model (`inworld/inworld-stt-1`)
**Available:**
* English (en)
**Experimental:**
* Spanish (es)
* French (fr)
* German (de)
* Italian (it)
* Portuguese (pt)
* Dutch (nl)
* Russian (ru)
* Chinese (zh)
* Japanese (ja)
* Korean (ko)
* Arabic (ar)
* Hindi (hi)
* Turkish (tr)
* Polish (pl)
* Swedish (sv)
* Cantonese (yue)
* Indonesian (id)
* Thai (th)
* Vietnamese (vi)
* Malay (ms)
* Danish (da)
* Finnish (fi)
* Czech (cs)
* Filipino (fil)
* Persian (fa)
* Greek (el)
* Hungarian (hu)
* Macedonian (mk)
* Romanian (ro)
Use `language` when you want to force recognition for a known language. Omit `language` to allow auto-detection when supported.
## Error Handling
Errors follow the standard [gRPC status](https://grpc.io/docs/guides/status-codes/) format.
**Authentication error**
```json theme={"system"}
{
"code": 16,
"message": "Unauthenticated: invalid or missing API key.",
"details": []
}
```
**Invalid request**
```json theme={"system"}
{
"code": 3,
"message": "Unsupported audio encoding.",
"details": []
}
```
**Common gRPC status codes**
| **Code** | **Name** | **Description** |
| :------- | :------------------- | :---------------------------------------------------------------- |
| `3` | `INVALID_ARGUMENT` | Invalid or missing request field (encoding, model ID, audio data) |
| `8` | `RESOURCE_EXHAUSTED` | Too many concurrent requests (rate limit) |
| `16` | `UNAUTHENTICATED` | Invalid or missing API key |
## Best Practices
* **Model choice** — Use `inworld/inworld-stt-1` when you want Voice Profile or Inworld-optimized turn-taking; use Groq/AssemblyAI/Soniox for specific latency/accuracy needs.
* **Audio** — Use MP3/OGG\_OPUS for file uploads to reduce size; use LINEAR16 for streaming (required) and when you need highest quality.
* **Streaming** — For Inworld model with manual turn-taking, send `endTurn` at each turn boundary and `closeStream` when done.
* **Speech events** — Listen for `speechStarted` and `speechStopped` events in the streaming response to detect when a speaker begins and stops talking. Use these to build custom turn-taking logic or visualize voice activity.
* **Voice Profile** — Set `voiceProfileConfig.enableVoiceProfile` to `true` and optionally adjust `topN` (default: 10) to control how many labels per category are returned.
* Test with sample audio and your target language before production.
## Troubleshooting
| **Issue** | **What to check** |
| :------------------ | :------------------------------------------------------------------------------------------------------------------------------------------------------------ |
| No transcript | API key, audio encoding matches request, valid audio file |
| `UNAUTHENTICATED` | `INWORLD_API_KEY` set correctly and not expired in Portal |
| `INVALID_ARGUMENT` | `audioEncoding` matches the actual format (LINEAR16 for raw PCM, MP3 for MP3, etc.) |
| Poor quality | Try a higher-accuracy model; use 16 kHz sample rate (8 kHz telephony audio has fewer data points and will produce lower-quality results); ensure clear speech |
| Large file failures | Split or compress (e.g. MP3/OGG\_OPUS); respect upload size limits |
| No Voice Profile | Ensure `voiceProfileConfig.enableVoiceProfile` is set to `true` in your request; response may also omit it if the selected model does not support it |
For more help, see the [Inworld Discord community](https://discord.gg/inworld).