Models / OpenAI
Transcribe

Whisper Large v3 (Streaming)

Realtime speech transcription optimized for voice agents

About model

Whisper Large v3 (Streaming) has 1.55B parameters. On Together AI it costs $0.0035 per min.

Whisper Large v3 enables realtime voice agent applications with WebSocket streaming transcription on Together AI. Purpose-built infrastructure combines OpenAI's 1.55B parameter model with intelligent voice activity detection and turn detection, delivering complete transcripts 1.4 seconds faster than alternatives. Supports 99 languages with 10-20% error reduction compared to v2. Ideal for conversational AI, customer support bots, and any application requiring natural, low-latency voice interaction.

  • API usage

    • cURL
    • Python
    • Typescript

    Endpoint:

    openai/whisper-large-v3

    curl -X POST "https://api.together.ai/v1/audio/transcriptions" \
      -H "Authorization: Bearer $TOGETHER_API_KEY" \
      -F "model=openai/whisper-large-v3" \
      -F "language=en" \
      -F "response_format=json" \
      -F "timestamp_granularities=segment"
    
    from together import Together
    
    client = Together()
    response = client.audio.transcribe(
        model="openai/whisper-large-v3",
        language="en",
        response_format="json",
        timestamp_granularities="segment"
    )
    print(response.text)
    
    import Together from "together-ai";
    
    const together = new Together();
    
    const response = await together.audio.transcriptions.create(
      model: "openai/whisper-large-v3",
      language: "en",
      response_format: "json",
      timestamp_granularities: "segment"
    });
    console.log(response)
    
  • Model card

    Architecture Overview:  
    • Transformer-based encoder-decoder model with 1.55B parameters  
    • Uses 128 Mel frequency bins instead of 80 (key improvement over v2)  
    • Trained on 1M hours weakly labeled + 4M hours pseudo-labeled audio  
    • WebSocket-based streaming architecture eliminating connection overhead  
    • Supports 99 languages with strong zero-shot generalization  
    • Advanced voice activity detection (VAD) using carefully tuned detection thresholds  
     
    Streaming Methodology:  
    • Trained for 2.0 epochs on massive multilingual dataset  
    • Weakly supervised learning on large-scale noisy data  
    • Demonstrates 10-20% error reduction compared to Whisper Large v2  
    • Strong performance across diverse languages, accents, and domains  
    • Purpose-built infrastructure for realtime audio processing with minimal quality degradation  
     
    Streaming Performance Characteristics:  
    • Industry-leading 2488ms baseline response time (1.4 seconds faster than alternatives)  
    • Processes audio as it arrives rather than waiting for complete files  
    • Intelligent turn detection determining when users finish speaking  
    • Optimized for time-to-complete-transcript not just time-to-first-token  
    • 500ms silence threshold tuning for natural conversation pacing  
    • Minimal quality degradation compared to batch processing  
    • Consistent low-latency performance under production load  
    • WebSocket connection multiplexing for efficient resource utilization  
    • Geographic distribution ensuring low latency regardless of user location

  • Applications & use cases

    Voice Agent Applications:  
    • Customer support automation with natural conversation flow  
    • Interactive voice response (IVR) systems with realtime understanding  
    • Voice-enabled chatbots and virtual assistants  
    • Healthcare appointment scheduling and patient intake  
    • Financial services customer authentication and account management  
     
    Realtime Communication:  
    • Live meeting transcription with immediate accessibility  
    • Real-time translation services for multilingual conversations  
    • Voice-to-text note-taking during calls and meetings  
    • Live captioning for broadcasts and presentations  
     
    Enterprise Contact Centers:  
    • AI agent handling hundreds of simultaneous customer calls  
    • Call routing based on realtime speech understanding  
    • Agent assist tools providing realtime suggestions  
    • Quality monitoring with live transcription and analysis  
     
    Conversational AI Platforms:  
    • Voice interface for LLM-powered applications  
    • Multi-turn dialogue systems requiring low latency  
    • Voice-controlled smart home and IoT devices  
    • Hands-free voice interfaces for accessibility  
     
    High-Volume Deployments:  
    • SaaS platforms offering voice capabilities to customers  
    • Contact center software handling enterprise-scale call volumes  
    • Voice agent platforms requiring consistent sub-second performance  
    • Applications where response latency directly impacts user satisfaction

  • Released
    November 7, 2023
  • Last updated
    July 9, 2025
  • External link
  • Category
    Transcribe