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realtime
Python WebSocket Client
import asyncio
import websockets
import json
import base64
import os

async def transcribe_audio():
    api_key = os.environ.get("TOGETHER_API_KEY")
    url = "wss://api.together.ai/v1/realtime?model=openai/whisper-large-v3&input_audio_format=pcm_s16le_16000"

    headers = {
        "Authorization": f"Bearer {api_key}"
    }

    async with websockets.connect(url, additional_headers=headers) as ws:
        # Read audio file
        with open("audio.wav", "rb") as f:
            audio_data = f.read()

        # Send audio in chunks with delay to simulate real-time
        chunk_size = 8192
        bytes_per_second = 16000 * 2  # 16kHz * 2 bytes (16-bit)
        delay_per_chunk = chunk_size / bytes_per_second

        for i in range(0, len(audio_data), chunk_size):
            chunk = audio_data[i:i+chunk_size]
            base64_chunk = base64.b64encode(chunk).decode('utf-8')
            await ws.send(json.dumps({
                "type": "input_audio_buffer.append",
                "audio": base64_chunk
            }))
            # Simulate real-time streaming
            if i + chunk_size < len(audio_data):
                await asyncio.sleep(delay_per_chunk)

        # Commit the audio buffer
        await ws.send(json.dumps({
            "type": "input_audio_buffer.commit"
        }))

        # Receive transcription results
        async for message in ws:
            data = json.loads(message)
            if data["type"] == "conversation.item.input_audio_transcription.delta":
                print(f"Partial: {data['delta']}")
            elif data["type"] == "conversation.item.input_audio_transcription.completed":
                print(f"Final: {data['transcript']}")
                break
            elif data["type"] == "conversation.item.input_audio_transcription.failed":
                error = data.get("error", {})
                print(f"Error: {error.get('message')}")
                break

asyncio.run(transcribe_audio())

Authorizations

​
Authorization
string
header
default:default
required

Bearer authentication header of the form Bearer <token>, where <token> is your auth token.

Query Parameters

​
model
string
required

The Whisper model to use for transcription

​
input_audio_format
enum<string>
default:pcm_s16le_16000
required

Audio format specification. Currently supports 16-bit PCM at 16kHz sample rate.

Available options:
pcm_s16le_16000

Response

101

Switching Protocols - WebSocket connection established successfully.

Error message format:

{
  "type": "conversation.item.input_audio_transcription.failed",
  "error": {
    "message": "Error description",
    "type": "invalid_request_error",
    "param": null,
    "code": "error_code"
  }
}