Real-Time Voice AI Platform

🎙️ Voice Module

A production voice AI system with real-time speech-to-text, text-to-speech, and conversational AI — built for call centers, IVR systems, and voice-first applications.

<200ms STT Latency
40+ Languages
99.5% Word Accuracy
24/7 Availability
Capabilities

What Voice Module Does

🎤

Real-Time STT

Sub-200ms speech-to-text with streaming transcription. Handles accents, background noise, and domain-specific vocabulary.

STTStreamingLow Latency
🔊

Neural TTS

Natural-sounding text-to-speech with multiple voices, emotion control, and SSML support. Streaming audio output for instant playback.

TTSNeuralSSML
🧠

Conversational AI

Multi-turn voice conversations with context retention, intent classification, and dynamic response generation powered by LLMs.

ConversationIntentLLM
📞

Call Center Integration

Drop-in integration for Twilio, SIP trunks, and WebRTC. Handle inbound and outbound calls with AI agents.

TwilioSIPWebRTC
🏭

Factory Pattern Deployment

Per-tenant voice AI instances deployed via factory pattern. Each tenant gets isolated models, vocabularies, and call routing.

Multi-TenantFactoryIsolation
📊

Voice Analytics

Call sentiment analysis, keyword spotting, compliance monitoring, and conversation summarization in real-time.

SentimentComplianceAnalytics
Why Voice Module

What Sets It Apart

1. Factory Pattern Isolation

Same factory pattern as Demo Generator. Each tenant gets their own voice AI instance with custom vocabulary and models.

2. True Streaming

Audio streams in and text streams out simultaneously. No batch processing, no waiting for utterance completion.

3. On-Prem Option

Run entirely on your infrastructure for regulatory compliance. No audio data leaves your network.

Technology

Built With

Backend
Go, WebSocket, gRPC
AI/ML
Whisper, custom TTS models, Claude for conversation
Audio
WebRTC, Opus codec, RTP streaming
Integration
Twilio, SIP, PSTN gateway
FAQ

Common Questions

What's the latency for real-time transcription?

Under 200ms from audio input to text output in streaming mode. Suitable for real-time conversation applications.

Can it handle multiple languages?

Yes. 40+ languages for STT, 20+ for TTS. Language detection is automatic — no need to specify upfront.

Does it work with existing phone systems?

Yes. Integrates with Twilio for cloud telephony, SIP trunks for on-prem PBX, and WebRTC for browser-based calling.

Can I run it on-premises?

Absolutely. The entire stack runs on local hardware for organizations with strict data residency requirements.

Ready to Get Started?

Add production-grade voice AI to your application. Real-time STT, TTS, and conversational intelligence.