Hecho para agentes de voz que no pueden esperar 600ms

Qwen Alibaba Cloud: Qwen3 ASR Flash Realtime

qwen/qwen3-asr-flash-realtime
activo stt 1 proveedor
Playground

Qwen streaming speech recognition model for live audio over an OpenAI-realtime-style WebSocket, returning incremental transcripts with a detected language and an always-on emotion label.

AutorQwen Alibaba Cloud
Idiomas27 idiomas
Modelos variantes3 IDs
Estadoactivo

Modelos variantes

Son IDs de modelo nativos que el autor o proveedor acepta para la misma familia de modelos canónica.

Qwen3 ASR Flash Realtime Default

Stable id, supplied as the model query parameter on the realtime WebSocket URL and currently equivalent to the qwen3-asr-flash-realtime-2025-10-27 snapshot.

qwen3-asr-flash-realtime
Qwen3 ASR Flash Realtime snapshot 2026-02-10 Variant

Latest dated snapshot of Qwen3 ASR Flash Realtime, listed separately from the stable id.

qwen3-asr-flash-realtime-2026-02-10
Qwen3 ASR Flash Realtime snapshot 2025-10-27 Variant

Earlier dated snapshot of Qwen3 ASR Flash Realtime; the stable qwen3-asr-flash-realtime id currently resolves to this snapshot.

qwen3-asr-flash-realtime-2025-10-27

Proveedores

Distintos proveedores pueden alojar el mismo modelo. Elige un proveedor cuando necesites un backend fijo, o deja que el enrutamiento elija entre ellos.

ProveedorSlugPor horaLatenciaiEn TTS, la latencia se mide como el tiempo hasta el primer byte de audio. En STT, se mide como el tiempo hasta el primer token.ModosDisponibilidad
Qwen Alibaba Cloud 7 parámetros
qwen
$0,324
—
Streaming

Idiomas admitidos

27 idiomas
Chinese zh
Cantonese yue
English en
Japanese ja
German de
Korean ko
Russian ru
French fr
Portuguese pt
Arabic ar
Italian it
Spanish es
Hindi hi
Indonesian id
Thai th
Turkish tr
Ukrainian uk
Vietnamese vi
Czech cs
Danish da
Filipino fil
Finnish fi
Icelandic is
Malay ms
Norwegian no
Polish pl
Swedish sv

Funciones

Speech to text

Transcribes spoken audio to text.

speech_to_text
Streaming

Realtime recognition over an OpenAI-realtime-style WebSocket, one session per connection, with the client appending base64 audio and the server returning incremental transcription events per turn.

streaming
Interim transcripts

Partial results arrive as transcription text events carrying a confirmed `text` prefix and a provisional `stash` suffix that concatenate to the sentence preview.

interim_transcripts
Turn detection

Server voice activity detection segments the stream into turns, or turn_detection null selects manual turns closed with input_audio_buffer.commit.

turn_detection
Automatic language detection

The language field is optional, and every transcription event reports the detected language alongside the text.

language_identification
Multilingual transcription

Supports the 27 language codes the realtime client-events reference enumerates, with Cantonese as the separate yue code.

multilingual_transcription
Context biasing

Accepts free-form context text in session.input_audio_transcription.corpus.text, up to 10,000 tokens, that biases recognition toward domain-specific terms.

context_biasing
Emotion recognition

Returns an always-on emotion label (surprised, neutral, happy, sad, disgusted, angry, fearful) on every transcription event.

emotion_recognition

Código

Cliente
Modo
Lenguaje

STT en streaming con el ElevenLabs SDK

Transmite audio del micrófono o de un archivo a través de la ruta de AllModels compatible con ElevenLabs.

import { ElevenLabsClient } from "elevenlabs";

const client = new ElevenLabsClient({
  apiKey: process.env.ALLMODELS_API_KEY,
  baseUrl: "https://api.allmodels.io/el"
});

const connection = await client.speechToText.realtime.connect({
  modelId: "qwen/qwen3-asr-flash-realtime",
  languageCode: "en"
}, {
  queryParams: {
    provider: {
      "only": [
        "qwen"
      ]
    }
  }
});

// Send audio frames to connection, then handle transcript events.