Ahmad

TeliSyn AI

An enterprise-grade 24/7 AI voice receptionist engineered with LiveKit WebRTC and Gemini 2.5 Realtime for sub-second conversational latency. Handles inbound telephony via Twilio SIP trunking with dynamic multi-tenant prompt injection and automated call intelligence.

ClientFounder & Lead Architect (In-House SaaS)
IndustryAI & Deep Tech
Year2026
Services ProvidedFull-Stack Architecture, Real-Time VoIP Infra, Gemini Streaming, Multi-Tenant Cloud
TeliSyn AI
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Key Challenges®

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Building an enterprise telephony system powered by LLMs requires overcoming sub-second voice latency thresholds, real-time interruptions, and seamless multi-tenant isolation.

Sub-Second Audio Latency

Traditional STT-LLM-TTS pipelines add 2-3s delays. We engineered a native bidirectional audio streaming loop using Gemini 2.5 Realtime over WebRTC.

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Telephony SIP Integration

Bridging standard carrier phone lines (Twilio SIP Trunking) with self-hosted LiveKit WebRTC media servers on cloud VPS instances.

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Multi-Tenant Dynamic Routing

Injecting distinct business knowledge graphs, custom voices, and conversational rules dynamically per inbound phone number.

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Natural Barge-In & Interruption

Detecting human speech cues instantaneously to cancel active audio playback without jarring audio artifacts.

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Design Approach®

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We architected a distributed real-time audio pipeline combining cloud telephony with multimodal neural models for instantaneous voice interaction.

Bidirectional WebRTC Worker

Developed a low-footprint Node.js agent worker on @livekit/agents delivering raw PCM audio streams directly to Gemini Realtime API.

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Self-Hosted SIP Gateway

Deployed and tuned LiveKit Server and LiveKit SIP Server on Oracle Cloud VPS for deterministic routing and zero audio jitter.

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Call Intelligence & Analytics

Automated real-time call transcription, structured sentiment extraction, and CRM sync to MongoDB upon call termination.

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Final Outcome

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TeliSyn AI handles 100% of inbound customer calls autonomously with zero missed calls, sub-500ms response latency, and human-level natural conversation.

480ms

Average end-to-end conversational voice latency

100%

Autonomous call resolution without human intervention

24/7

Continuous uptime handling concurrent peak call volumes

94%

Customer satisfaction rating on voice naturalness

“TeliSyn AI completely transformed our front-desk operations. It sounds so natural and responds so fast that callers don't even realize they're speaking with an AI assistant.”

— Verified Beta Client Feedback
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