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Speech AI & Telephony Engineering

Enterprise AI Voice Agent Development & Speech AI

Reviewed by Umar Abbas • Founder & Principal AI Architect

Last reviewed: 19 August 2026

Building enterprise voice agents requires sub-500ms end-to-end latency across speech recognition, state machine reasoning, and speech synthesis. We engineer streaming voice pipelines with conversational turn-taking, barge-in detection, and legacy telephony integration.

Latency SLA285ms p95
TelephonySIP & WebRTC
InterruptionReal-Time VAD
Resolution Rate94.2% FCR
Capabilities Grid

Real-time voice engineering for enterprise calls

We replace rigid IVR push-button trees with natural, human-like voice agents that converse naturally, understand complex intent, and take system actions.

Streaming STT & VAD Pipeline

Sub-80ms speech recognition using Deepgram Nova-2 or Whisper, paired with continuous frame-level voice activity detection.

Full-Duplex Interruption Gateway

Instant barge-in controller that clears audio playback buffers the moment a customer starts speaking.

Ultra-Low Latency TTS

Cartesia Sonic and ElevenLabs Turbo streaming synthesis generating natural human inflection under 95ms.

Warm SIP & Softphone Transfer

SIP REFER handoffs transferring live calls and complete context summaries to human call center agents upon escalation.

SIP Telephony / WebRTCTwilio · Plivo · G.711 AudioDeepgram STT + VAD (<80ms)Real-time speech to textvLLM Dialogue Router (<110ms)Streaming dialogue tokensCartesia Sonic TTS (<95ms)Real-time voice synthesisTurn Latency: 285ms p95
Voice Stack

Voice & Speech Technologies

Deepgram Nova-2 Whisper Cartesia Sonic LiveKit WebRTC Twilio SIP vLLM FastAPI
Technical FAQ

Frequently Asked Questions

How do your voice agents achieve sub-300ms turn-taking latency?↓

We pipeline speech-to-text (Deepgram Nova-2 under 80ms), streaming LLM token generation (vLLM under 110ms), and streaming text-to-speech synthesis (Cartesia Sonic under 95ms) over full-duplex WebSockets, bypassing HTTP request-response overhead.

How does the system handle caller interruption (barge-in) naturally?↓

Our voice activity detection (VAD) nodes continuously listen on the incoming audio stream. When user speech is detected during agent playback, an instant cancellation event interrupts the active TTS output buffer, preventing voice collision.

Can the voice agent connect to existing telephony providers like Twilio, Avaya, or Genesys?↓

Yes. We interface with legacy PBX and call center infrastructures via SIP trunks, WebRTC gateways, and Twilio Media Streams, allowing seamless inbound/outbound call routing and warm transfers to human agents.

Are caller transcripts and audio streams secured for PCI-DSS and HIPAA compliance?↓

Yes. Audio streams travel encrypted via Secure RTP (SRTP) and TLS 1.3. We implement automatic real-time DTMF and audio masking for credit cards or SSNs, with optional zero data retention policies.

Can the voice agent execute backend database actions while speaking?↓

Yes. Our agent orchestrators call external REST/gRPC tools in parallel with token generation, allowing the agent to check inventory, verify account credentials, or issue refunds mid-call without pausing conversation.

What speech recognition and synthesis models do you support for multilingual callers?↓

We deploy multilingual Whisper v3 and Deepgram for transcription across 30+ languages, coupled with ElevenLabs and Cartesia multilingual models that preserve natural prosody, regional accents, and inflection.

How are warm transfers to human contact center agents orchestrated?↓

When intent classifiers detect high escalation risk or explicit customer requests, the voice agent executes a SIP REFER transfer, passing the full structured conversation summary and intent payload to the human agent's CRM console.

Can voice agents be deployed entirely on private sovereign infrastructure?↓

Yes. We deploy open-weights speech pipelines (Whisper v3, Llama 3.3, and XTTS-v2) on private client GPU clusters for government, defense, and healthcare organizations requiring total on-premise data isolation.

Deploy Sub-300ms Voice AI Agents

Book a voice architecture discovery session to evaluate speech latency, SIP trunking, and call center ROI.

Architecture Pipeline

Turn-taking audio execution flow

Full-duplex WebSocket stream linking caller audio, transcript parsing, model inference, and voice output.

1. Audio IngestPCM 8kHz / WebRTC
2. STT + VADSpeech frame processing
3. Token StreamDialogue context & tools
4. TTS SynthesisInstant audio stream
Business Impact

Call Center ROI & Cost Reduction

Deploying speech AI voice agents cuts customer handle time while delivering instantaneous phone resolution at scale.

95% Cost Reduction per Call

Lower handling costs from $6.80/call (human agent) to $0.34/call (voice AI agent) on routine inquiries.

75% Reduction in Average Handle Time

Instant access to account databases and zero queue waiting cuts call duration from 8.5 minutes to 2.1 minutes.

88.6% First Call Resolution (FCR)

Deep knowledge retrieval and real-time backend tool calls allow agents to resolve complex issues without transfers.