Consider a support center with 50 representatives: average call duration is 6 minutes, expenses climb 20% yearly. Outsourcing becomes pricier, quality drops, employee turnover hits 40%. Our team, with over a decade in NLP and MLOps, builds AI customer service platforms that address these issues: 55–65% of calls are resolved by the bot without human help, customer satisfaction rises by 0.9 points. According to Gartner (2023), companies adopting such platforms cut handling costs by 40-60%. Our AI platform handles calls 1.6x faster and resolves issues 1.26x better than traditional call centers. With more than 30 deployments completed, we deliver measurable ROI.
How does an AI platform reduce operational costs?
Traditional call centers incur proportional staffing costs: each additional 1,000 calls per day requires 5 new hires, training, and licensing. An AI platform scales transparently—just boost model capacity. The result: containment rate jumps from 0% to 55–65%, average handle time drops from 360 to 220 seconds, and first-call resolution improves by 20%. On average, clients save $500k annually after deploying our solution.
How It Works
- Discovery – We map your processes and define top use cases.
- Model Training – Fine-tune speech-to-text and NLU models on your data.
- Integration – Connect to your CRM, telephony, and chat channels.
- Development – Build bot flows for key scenarios with fallback.
- Testing – Validate accuracy, performance, and security.
- Deployment – Roll out in phases with full support.
Core Capabilities
Our platform combines:
- Speech-to-Text: Deepgram or Whisper for real-time transcription with 95% accuracy.
- Natural Language Understanding: GPT-4o for intent classification and entity extraction.
- Dialogue Management: LangChain-based voice bot handling multi-turn conversations.
- CRM Integration: REST API connectors for Salesforce, Zendesk, HubSpot, and custom systems.
- Analytics Dashboard: Real-time dashboards for conversation insights, sentiment trends, and agent performance.
- Voice Assistant: Our voice assistant handles routine queries and escalates complex ones.
- Automated Customer Support: Automates responses across channels, reducing manual effort.
- System Integration: Seamlessly integrates with existing telephony and CRM infrastructure.
- Conversation Insights: Analyzes interactions to uncover trends and improvement opportunities.
We also offer agent augmentation tools that provide real-time suggestions.
What's Included in a Turnkey Deployment
| Service Component |
Deliverable |
Timeline |
| Discovery & Design |
Process map, use case specification, architecture document |
2 weeks |
| Speech & NLU Model Training |
Custom models fine-tuned on your data, accuracy validation report |
4–6 weeks |
| Integration & API Setup |
CRM, telephony, and chat channel connectors |
2–3 weeks |
| Bot Flow Development |
5 common scenarios with fallback handling |
3–4 weeks |
| Testing & QA |
UAT scripts, performance testing, security audit |
2 weeks |
| Deployment & Go-Live |
Staged rollout (10% → 50% → 100%), cutover plan |
1 week |
| Training & Documentation |
Operator manual, admin guide, 3-day on-site training |
1 week |
| Post-Launch Support |
6 months of 24/7 support, weekly optimization reports |
Ongoing |
Metrics & Track Record
With 10+ years of AI expertise and over 30 successful deployments, we have a 98% client retention rate. Our solutions deliver:
- 55–65% containment – bots resolve calls end-to-end, reducing agent load.
- 40% AHT reduction – from 360s to 220s, freeing agent time.
- 20% FCR improvement – issues resolved on first contact.
- 0.9 CSAT point rise – customer satisfaction scores climb.
Comparison: Traditional vs AI Platform
| Metric |
Traditional Call Center |
AI-Powered Platform |
Improvement |
| Containment Rate |
0% |
55–65% |
55x better |
| Average Handle Time |
360 sec |
220 sec |
1.6x faster |
| First Call Resolution |
72% |
91% |
1.26x better |
| Customer Satisfaction |
3.8/5 |
4.7/5 |
1.24x higher |
| Annual Cost per 50 Agents |
$2.5M |
$1.0M |
60% lower |
Security & Compliance Guaranteed
We deploy models on-premises or in your VPC. All transcripts are encrypted at rest and in transit. The system is GDPR compliant and can be configured for HIPAA or PCI-DSS. Your data never leaves your infrastructure. No third-party access. We provide a signed data protection agreement and regular penetration testing reports.
Why Choose Our Turnkey Solution?
- Experience: 10+ years in NLP, 30+ projects, team of PhDs and engineers.
- Speed: Prototype in 2 weeks, production in 3 months for core features.
- Transparency: No hidden costs, fixed-price contracts, milestone-based payments.
- Support: 6 months of round-the-clock support, free optimization sessions.
- Guarantee: 99.9% uptime SLA, performance-based refund clause.
Ready to transform your customer service? Contact us for a free, no-obligation cost estimate. We'll analyze your processes and deliver a tailored proposal within 5 business days.
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=True → pyannote 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.