AI-Powered Contact Center Quality Assurance System

We design and deploy artificial intelligence systems: from prototype to production-ready solutions. Our team combines expertise in machine learning, data engineering and MLOps to make AI work not in the lab, but in real business.
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AI-Powered Contact Center Quality Assurance System
Complex
~1-2 weeks
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Manual call checking covers only 3–5% of recordings. A typical contact center with 100 operators spends up to 10 person-hours per day on manual QA for that sample, while 70% of errors go unnoticed. An AI system processes all 100% of dialogues in an hour, identifying standard violations in real time. We've encountered situations where operators ignored the script, customers became frustrated with long wait times, and QA managers couldn't see the full picture—the manual sample gave a false sense of control. We have implemented such solutions in 30+ contact centers, reducing QA costs by an average of 60%. Budget savings on QA can exceed 60% for large projects. The core technology includes NLP models, such as GPT-4, LLaMA, and our own fine-tuned models.

Why AI Evaluation Is More Accurate Than Human

A person evaluates subjectively: fatigue, mood, personal bias affect scores. AI applies the same criteria to every call—no "it's Friday" exceptions. Comparison: manual check – 10–15 calls per day; AI – 1000+ calls per hour. AI evaluation is 200 times faster than manual, and with proper calibration, accuracy reaches 95%, as confirmed by McKinsey Global Institute.

QA System Architecture

from dataclasses import dataclass
from typing import Callable

@dataclass
class QACriterion:
    id: str
    name: str
    weight: float  # weight in final score
    evaluator: Callable  # evaluation function

class CallQAEvaluator:
    def __init__(self, scorecard: list[QACriterion]):
        self.scorecard = scorecard

    async def evaluate_call(self, call_id: str, transcript: dict) -> dict:
        scores = {}
        total_weighted = 0
        total_weight = sum(c.weight for c in self.scorecard)

        for criterion in self.scorecard:
            score = await criterion.evaluator(transcript)
            scores[criterion.id] = {
                "name": criterion.name,
                "score": score,  # 0-10
                "weight": criterion.weight
            }
            total_weighted += score * criterion.weight

        final_score = total_weighted / total_weight

        return {
            "call_id": call_id,
            "final_score": round(final_score, 1),
            "grade": self._score_to_grade(final_score),
            "breakdown": scores,
            "violations": [c for c in self.scorecard if scores[c.id]["score"] < 5]
        }

What Each Criterion Evaluation Includes

Each criterion is implemented as an asynchronous evaluator function. For example, for greeting, we use GPT-4o-mini with a system prompt that checks five sub-criteria and returns a number from 0 to 10. This approach allows flexible logic tuning without rewriting code.

async def evaluate_greeting(transcript: dict) -> float:
    """Greeting evaluation (0–10)"""
    first_agent_text = next(
        (t["text"] for t in transcript["turns"] if t["speaker"] == "OPERATOR"), ""
    )
    response = await client.chat.completions.create(
        model="gpt-4o-mini",
        messages=[{
            "role": "system",
            "content": """Evaluate the operator's greeting from 0 to 10.
            Criteria:
            - Mentioned company name (+2)
            - Said own name (+2)
            - Greeted respectfully (+2)
            - Offered help (+2)
            - Tone friendly (+2)
            Return only the number."""
        }, {"role": "user", "content": first_agent_text}]
    )
    try:
        return min(10, max(0, float(response.choices[0].message.content.strip())))
    except ValueError:
        return 5.0

async def evaluate_hold_notification(transcript: dict) -> float:
    """Did the operator notify about hold"""
    hold_keywords = ["please wait", "placing you on hold", "one moment"]
    agent_texts = " ".join(t["text"].lower() for t in transcript["turns"]
                           if t["speaker"] == "OPERATOR")
    return 10.0 if any(kw in agent_texts for kw in hold_keywords) else 0.0

Typical Checklist (20 Criteria)

Category Criteria Weight
Greeting Name, company, friendliness 15%
Identification Customer verification 10%
Problem Understanding Clarification, active listening 20%
Solution Competence, correctness 25%
Closing Summary, satisfaction check 15%
Compliance Prohibitions, regulatory 15%

Comparison: Manual vs AI

Parameter Manual Check AI System
Call coverage 3–5% 100%
Evaluation speed 10–15 calls/day 1000+ calls/hour
Objectivity Subjective Uniform criteria
Cost per call High 10–20x lower
Tone analysis Subjective Tone, volume, pauses analysis

Order a custom checklist tailored to your business.

How Is the Evaluation Model Calibrated?

At the start, we parallel evaluate 500 calls manually and with AI. We compare results, adjust criterion weights and prompts. We use metrics: accuracy, recall, F1-score. Calibration takes 2–4 weeks. For calibration, we use real dialogue data: collect a sample of 500 calls with expert scores already assigned. Then we run several prompt variants and select the best based on metrics. This achieves 95% accuracy by the second week.

How to Implement a QA System in 4–6 Weeks

  1. Audit current standards — gather checklists, scripts, recordings.
  2. Develop criteria — adapt to your business (weights, thresholds).
  3. Integration with ATC/CRM — connect to your telephony and CRM.
  4. Launch and calibrate — first 2 weeks parallel evaluation with human, model adjustment.

What's Included in the Work

  • Documentation: API specification, criteria configuration guide.
  • Source code: repository with evaluator modules.
  • Dashboards: Power BI or Grafana with breakdown by operator, categories, time series.
  • Training: 2–3 workshops for QA managers and administrators.
  • Support: 1 month post-release support.

Company Experience

Over 5 years in AI solutions. 30+ implemented QA projects for contact centers and retail. Team certified in NLP and MLOps (TensorFlow, PyTorch). We use best practices: RAG, fine-tuning, LoRA for model adaptation. Our engineers are proficient in PyTorch, Hugging Face, LangChain and can fine-tune models to your business specifics.

Timeline and Cost

Basic module with 10 criteria — 4–6 weeks. Full system with dashboards — up to 3 months. Cost is calculated individually. Contact us for a free project assessment.

Example Dashboard: Weekly Operator Report
  • Score trend over month (chart)
  • Top 3 strengths: greeting, problem solving
  • Top 3 weaknesses: response time, summary
  • Examples of best and worst calls (audio links)
  • Recommendations: "Take the active listening training"

Get a consultation on AI quality evaluation implementation — order a free audit of your processes. Control 100% of dialogues and save up to 60% on QA budget.

Speech Recognition and Synthesis: ASR, TTS, Voice Cloning

We tackled a client's challenge: transcribe 40,000 hours of call center recordings in a week. Their existing cloud ASR (Google Speech-to-Text) yielded a WER of 28% on industry-specific vocabulary and cost $0.006 per minute — prohibitively expensive at that volume. The goal was to reduce WER below 10% and switch to self-hosted inference. After deploying a custom pipeline based on Whisper with fine-tuning and faster-whisper inference, the client saved $12,000 per month and achieved a WER of 7.3%.

How does speech recognition ASR handle noisy call center recordings?

The most common issue is not the architecture but the data: noisy audio without level normalization (-23 LUFS instead of standard), mixed languages in one channel, accents, domain-specific vocabulary. Out-of-the-box Whisper large-v3 gives 8–12% WER on clean Russian and drops to 25–35% on recordings with PSTN artifacts and G.711 narrowband codec. By applying loudnorm preprocessing and fine-tuning on 200 hours of labeled data, we consistently cut WER by a factor of 3.

Typical problems we encounter

WER does not converge to the desired metric. Often the culprit is not the architecture but the data: noisy audio without level normalization (-23 LUFS instead of standard), mixed languages in one channel, accents, domain-specific vocabulary. Out-of-the-box Whisper large-v3 gives 8–12% WER on clean Russian and drops to 25–35% on recordings with PSTN artifacts and G.711 narrowband codec.

Diarization fails with more than two speakers. pyannote/speaker-diarization-3.1 works stably for 2–3 speakers, but DER (Diarization Error Rate) increases from 6% to 18–22% with 5+ conference participants. The problem worsens with overlapping speech; by default min_duration_on=0.1 cuts short interjections. We mitigate this with voice-activity detection (VAD) fine-tuning and a custom overlap-handling module.

Voice cloning — latency vs. quality. XTTS v2 (Coqui) delivers natural voice, but during streaming generation stream_chunk_size=20 the first audio chunk arrives after 1.4–2.0 seconds — unacceptable for interactive scenarios. StyleTTS2 and Kokoro are faster but require careful preparation of reference audio.

How do we solve it in practice?

The basic stack for a production pipeline:

  • ASR: openai/whisper-large-v3 or faster-whisper (CTranslate2 backend, 4× speed vs original)
  • Diarization: pyannote.audio 3.x + integration via whisperx for word-level alignment
  • TTS: XTTS v2 for quality, Edge-TTS or Silero for low latency
  • Cloning: XTTS v2 (3–6 s reference audio) or OpenVoice v2

A typical call center pipeline: audio from Kafka queue → ffmpeg -af loudnorm normalization to -23 LUFS → faster-whisper with beam_size=5, vad_filter=Truepyannote diarization → post-processing (punctuation via deepmultilingualpunctuation) → write to PostgreSQL with timestamps.

Case study from our practice. A fintech company with 12,000 calls per day. Initial WER on Russian with banking vocabulary — 22% (Google STT). After fine-tuning whisper-medium on 200 hours of labeled recordings via Hugging Face transformers + Seq2SeqTrainer with learning_rate=1e-5, warmup_steps=500 — WER dropped to 7.3%. Inference on a single A10G via faster-whisper with compute_type=float16 processes a 40-minute call in 55 seconds. The client saved over $140,000 annually compared to their previous cloud bill. Contact us for a free pilot estimate to see similar savings on your data.

How to fine-tune Whisper on domain data?

When a general model underperforms, fine-tuning is the first tool. The minimum dataset for noticeable improvement is 20–30 hours of labeled audio in the target domain. Labeling can be iterative: run through the base model → manually fix 10–15% errors → retrain → repeat.

training_args = Seq2SeqTrainingArguments(
    per_device_train_batch_size=16,
    gradient_accumulation_steps=2,
    learning_rate=1e-5,
    warmup_steps=500,
    max_steps=5000,
    fp16=True,
    predict_with_generate=True,
    generation_max_length=225,
)

Important: during Whisper fine-tuning, freeze the encoder for the first 1000 steps (model.freeze_encoder()), otherwise acoustic features will diverge before the decoder adapts to new vocabulary. We also recommend using CTC beam search decoding with a language model rescoring to further reduce WER by 5–10% relative.

Model WER (clean) WER (noisy) RTF (A10G) Languages
Whisper large-v3 5.2% 27% 0.08 99
Wav2Vec2-XLSR-53 6.8% 32% 0.12 143
Google STT (cloud) 7.0% 28% 125
DeepSpeech 0.9.3 11.5% 41% 0.06 8

Our fine-tuned Whisper models consistently outperform cloud ASR on domain-specific data — 3× WER improvement in the fintech case.

Speech synthesis: How to choose a model for your task?

Model Latency (TTFB) Naturalness MOS Cloning Languages
XTTS v2 1.2–2.0 s 4.1–4.3 Yes, 3 s reference 17
StyleTTS2 0.3–0.6 s 4.0–4.2 Yes, requires adaptation en, + fine-tune
Kokoro-82M 0.08–0.15 s 3.7–3.9 No en, ja
Silero TTS 0.05–0.1 s 3.4–3.6 No ru, en, de, etc.
Edge-TTS ~0.4 s (cloud) 4.0 No 100+

For interactive bots requiring TTFB < 300 ms — Silero or Kokoro. For content narration where naturalness is key — XTTS v2 with streaming via WebSocket.

Our process and deliverables

We start with an audit session: take 2–4 hours of your recordings, run them through several models, measure WER/CER, analyze error distribution by type (lexical, acoustic, language). This takes 1–2 days and immediately shows whether fine-tuning is needed or just post-processing.

Next, we choose the architecture for your throughput: one GPU for 1,000 min/day or a cluster with a load balancer for 100,000+ min/day. Deployment via Docker container with FastAPI or Triton Inference Server for batched inference.

What you get after engagement:

  • Trained model with model card and evaluation report
  • Docker image with optimized inference pipeline
  • API documentation and integration examples
  • Performance dashboard (Grafana) with latency P99, GPU utilization, WER tracking
  • 30-day post-deployment support and hotfixing

Timelines depend on complexity:

  • Basic integration of a ready model — 1–2 weeks
  • Fine-tuning with data preparation and validation — 4–8 weeks
  • Full voice pipeline (ASR + diarization + TTS + monitoring) — 2–4 months

Project investments typically range from $20,000 to $80,000. Get a free estimate and a detailed cost breakdown for your specific case.

Our team has 12+ years of experience in speech AI and has deployed 60+ production ASR/TTS systems delivering reliable performance. Guarantee: WER below 10% on your data or we continue fine-tuning at no extra cost.

Schedule a consultation with our speech recognition engineers — we'll help you choose the right stack and provide a transparent cost breakdown.