Services/whisper-call-center-ai
01 — Speech AI & Contact Center Intelligence

Fine-Tuned Whisper & Call Center Speech AI

Low-latency speech-to-text, real-time speaker diarization, and automated PII redaction for high-volume contact centers.

Standard foundation speech models struggle in noisy telephonic environments. We fine-tune OpenAI Whisper v3 models specifically for enterprise contact centers to deliver sub-20ms streaming latency, multi-speaker separation, and instant PCI-DSS/HIPAA PII masking.

Status
Sub-20ms Streaming STT
WER Reduction
Significant
Streaming Latency
<20ms
Noise Filtering
35dB SNR

Real-Time Telephony Acoustic & Diarization Pipeline

Production-tested execution flow designed for multi-region reliability and security.

01

Telephony RTP Ingestion

Consumes raw 8kHz G.711 / 16kHz PCM audio streams over WebSockets.

02

Acoustic Denoising

DeepFilterNet removes background office noise and crosstalk.

03

TensorRT-LLM Whisper Inference

Accelerated FP16/INT8 inference delivers transcript tokens continuously.

04

Diarization & Redaction

Identifies agent vs customer speaker channels and redacts sensitive PII.

Key Engineering Capabilities

Custom Acoustic & Vocabulary Fine-Tuning

Adapts Whisper v3 weights to specialized industry terms, medical jargon, product SKUs, and accents.

Real-Time Multi-Speaker Diarization

Tracks speaker embeddings continuously using PyAnnote 3.1 to attribute transcripts accurately.

Instant PII Masking & PCI Compliance

Redacts credit card numbers, Social Security Numbers, and addresses before text touches database logs.

Live Agent Scorecards & Sentiment Analytics

Feeds real-time sentiment signals into manager dashboards and agent co-pilot recommendations.

Production Code Blueprint

Streaming Telephony WebSocket Ingestion with TensorRT Whisper
import asyncio
import websockets
from whisper_tensorrt import WhisperTRTEngine

engine = WhisperTRTEngine("whisper-v3-callcenter-int8.engine")

async def handle_telephony_stream(websocket, path):
    async for audio_chunk in websocket:
        # 16-bit 16kHz PCM audio buffer
        denoised_pcm = engine.denoise(audio_chunk)
        tokens = engine.transcribe_chunk(denoised_pcm)
        redacted_text = engine.redact_pii(tokens)

        await websocket.send_json({
            "speaker": tokens.speaker_id,
            "transcript": redacted_text,
            "latency_ms": tokens.latency
        })

print("✓ Call Center Speech AI Server Online on port 8080")

Engineered With Core Technologies

Whisper v3TensorRT-LLMvLLMPyTorchWebSocketsPyAnnote.Audio

Frequently Asked Questions

How does fine-tuned Whisper handle noisy background environments?

We fine-tune Whisper v3 cross-attention layers on augmented contact center audio containing ambient chatter and phone compression codecs (G.711), dropping Word Error Rate (WER) dramatically compared to off-the-shelf APIs.

Is PII redacted in real time before data storage?

Yes. Our pipeline executes regular expression pattern matching and named entity recognition (NER) in memory prior to sending transcripts to database storage or WebSocket clients.

Deploy Fine-Tuned Whisper & Call Center Speech AI In Your Organization

Book an architecture review session with our senior AI engineers to assess your infrastructure and custom requirements.

Request Technical Consultation