VOICE AI DEVELOPMENT & INTEGRATION
Build voice workflows that capture the right details, connect with your business systems and bring in a person when the conversation needs one.
Share your goals. We’ll help define the next project.
WHEN THIS WORK HELPS
For engineering and operations leaders deploying voice agents for high-stakes intake, dispatch, appointment booking or customer support. We prioritize latency, reliability and human oversight.
SCOPE AND OUTPUTS
We agree on access, deliverables and checks for your environment.
Design real-time speech-to-speech pipelines using WebRTC, SIP trunking, fast STT streaming, speculative LLM inference and chunked TTS synthesis with latency benchmarking.
Deliverable: End-to-end pipeline blueprint, benchmarked roundtrip latency and failover topology.
Constrain model parameters with strict JSON schemas, idempotency keys and pre-execution validation before state mutations occur in your database or calendar.
Deliverable: Validated tool schemas, type-safe API bridges and simulated load tests.
Bridge existing phone infrastructure (Twilio, LiveKit, Telnyx, Asterisk) with SIP trunks, DTMF handling, call recording compliance and warm transfers to human agents.
Deliverable: Integrated SIP connectors, call routing rules and graceful transfer fallbacks.
Track turn-by-turn latency (TTFT, speech-to-speech), audio quality MOS, interruption rates and full audit trails of tool invocations without storing unconsented audio.
Deliverable: Real-time metrics dashboards, alert definitions and compliance audit logs.
HOW THE WORK PROGRESSES
Define the exact operational boundaries: allowed actions, mandatory disclosure, privacy constraints and human escalation triggers.
Measure end-to-end speech-to-speech latency across simulated network conditions, tuning chunk sizes and model inference parameters.
Wire backend APIs, CRM records and phone trunks with strict validation, retry backoffs and deduplication guarantees.
Run synthetic multi-caller stress tests, verify operator transfer under failures and roll out gradually behind feature flags.
BEFORE WE START
Price, timing, access and support follow the agreed project scope.
We set a target after testing the selected models, network path and telephony provider. End-to-end call latency must be measured with the actual workflow; no single target fits every system.
We define the escalation path and test the transfer with the selected provider. If a transfer fails, the call should have a documented fallback rather than an invented confirmation.
We review the provider's SIP, WebRTC or API capabilities and the access available on your plan before committing to an integration.
Retention, access, recording and redaction depend on the selected providers and project requirements. We define and test those controls before processing real calls.
No. Real-time voice models can still mishear accents or misinterpret ambiguous requests. We design explicit clarification loops and easy human escape hatches instead of pretending the system is infallible.
START WITH THE CONTEXT
Primary guidance to review for your implementation:
START WITH YOUR QUESTION
Tell us what you want to improve. We’ll review your goals and discuss the next step.
“Our support line handles appointment questions. Can we scope a voice flow with a clear transfer to a person?”
What happens after you send it?We review your description and reply with questions about your project. No calendar booking or newsletter signup.
No technical brief needed to start.