Built for voice agents that can't wait 600ms

Qwen Alibaba Cloud: Qwen3 ASR Flash Realtime

qwen/qwen3-asr-flash-realtime
active stt 1 provider
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.

AuthorQwen Alibaba Cloud
Languages27 languages
Variant models3 IDs
Statusactive

Variant Models

These are native model IDs accepted by the author/provider for the same canonical model family.

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

Providers

Different providers can host the same model. Choose one provider when you need a fixed backend, or let routing select among them.

ProviderSlugPer hourLatencyiFor TTS, latency is measured as time to first byte of audio. For STT, latency is measured as time to first token.ModesUptime
Qwen Alibaba Cloud 7 params
qwen
$0.324
—
Streaming

Supported languages

27 languages
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

Features

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

Code

Client
Mode
Language

ElevenLabs SDK streaming STT

Streams microphone or file audio through the ElevenLabs-compatible allmodels route.

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.