Voice AI Agent Development — Real-Time Agents That Take Interruptions, Phone Calls, and a Compliance Review.
Trembit is a real-time voice AI agent development company. We build the voice agents and voice assistants that talk to your customers on the phone, in your app, and inside video calls: the speech pipeline, the latency budget, barge-in and turn-taking, and the SIP and WebRTC transport underneath. When a voice agent feels broken, the model is rarely the cause; the real-time path around it usually is. Over 15 years we’ve delivered 50+ video and voice implementations.
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Voice AI Agent Development, In Brief
Trembit is a voice AI agent development company that builds real-time voice agents and voice assistants for phone lines, apps, and video calls, including the telephony, latency, and compliance engineering a platform demo skips. A typical engagement runs as a free scoping call, then discovery and a latency budget (weeks 1–2), then a proof of concept on real calls (4–8 weeks in total), then a production build (typically 3–5 months). The main cost drivers are call volume and peak concurrency, cascaded or speech-to-speech architecture, telephony integration, the number of languages, and compliance and data-residency scope.
Who works on a voice agent build:
- A tech lead who owns the latency budget end to end
- Real-time engineers for WebRTC, SIP, and media handling
- AI and backend engineers for STT, LLM, TTS, and tool calls
- A conversation designer for flows, fallbacks, and human handoff
- QA who test on phone-line audio, bad networks, and interruptions
Sound Familiar?
Your voice agent demos well, and then callers talk over it.
It keeps speaking after the caller interrupts, or cuts them off mid-sentence. Barge-in and turn-taking were left at the framework defaults, and real callers don’t pause the way test scripts do.
Responses are too slow, and nobody can say which layer is eating the time.
Speech recognition, the LLM, speech synthesis, the jitter buffer, an extra network hop: every stage adds delay. Without a budget for each one, teams swap models and the lag stays.
It works in the browser, but it has to answer real phone numbers.
Your customers call from mobile phones and desk lines, through SIP trunks and an existing PBX or contact-centre stack. Bridging phone audio into an AI pipeline is its own engineering project.
Legal won't let patient or payment audio go to a third-party API.
Voice carries regulated data the moment someone speaks. A vendor that’s fine for text may not cover audio under its BAA, and your data may have to stay in the EU or on your own servers.
Your IVR makes callers press 0, and your first bot wasn't much better.
Menu trees frustrate callers. Keyword bots fail on your industry’s vocabulary, on accents, and on narrowband phone audio, so the calls end up with a human anyway.
Per-minute platform pricing that looked cheap in the pilot now dominates the bill.
Managed voice platforms charge per minute. That’s often the right trade at pilot scale. At production volume, the line item grows with every call.
A Voice Agent Is a Real-Time System, Not a Prompt
Connecting an LLM to a microphone takes an afternoon. What decides whether callers stay on the line is everything around the model: how quickly the agent knows the caller has finished, how fast it stops when interrupted, how audio gets from a phone network or browser into the pipeline, and where that audio is allowed to go. That’s real-time systems engineering, the layer we’ve worked in for 15+ years.
These are the problems teams bring us, usually after a platform prototype that worked in the demo.
Voice AI Agent Development Services
AI Phone Agents & IVR Replacement
For teams whose customers call a phone number.
Inbound and outbound voice agents on your SIP trunks or phone numbers, replacing menu-tree IVR incrementally, starting with the highest-volume call types.
- SIP trunk, PSTN, and existing PBX integration
- Human handoff with the conversation context attached
- AI self-identification, consent capture, and opt-out handling
- Integration with CRM, ATS, EHR, and ticketing systems
Voice Agents in Apps and Video Calls
For products where the agent talks inside your web or mobile app.
Voice agents over WebRTC with barge-in and turn-taking tuned against a real latency budget, including multi-party calls where the agent joins humans.
For AI across the whole live media path (moderation, captions, analytics), see WebRTC AI
- Cascaded (STT → LLM → TTS) or speech-to-speech architecture, chosen per use case
- Barge-in that flushes queued audio, not only the generator
- Endpointing tested on real mobile networks
Compliant & Self-Hosted Voice AI
For healthcare, finance, and any team whose audio can't leave its boundary.
Voice pipelines designed so regulated audio stays inside a compliant path, from the BAA chain to on-premises speech models.
- HIPAA and GDPR architecture, with a BAA or DPA for every vendor that touches audio
- EU or on-premises hosting for speech recognition and synthesis
- Zero-persistence audio processing where the regime requires it
- Audit logs that record what happened, not what was said
Voice Agent Rescue & Latency Tuning
For agents already live that are too slow, too robotic, or blocked by compliance.
We audit the latency budget stage by stage, fix barge-in and endpointing, and migrate from a managed platform to your own stack when volume or compliance calls for it.
- Stage-by-stage latency measurement, from STT and LLM to TTS and transport
- Jitter-buffer, codec, and topology fixes the model layer can't reach
- Platform-to-custom migration without a hard cutover
- Take-over of voice agent codebases another team left unfinished
Voice AI Platform or Custom Build?
Most teams should start with the question, not a vendor. Often the right answer is a managed platform. A custom build pays off when compliance, telephony depth, cost at volume, or latency control are what your product depends on. Here’s how we decide with clients.
Managed voice agent platform (e.g. Vapi, Retell)
Right when you’re validating a use case, volume is modest, and the call flow is standard. You get to production fastest, with telephony, orchestration, and hosting included. Both let you bring your own SIP trunk. Check the details that matter to you: on Vapi, for example, HIPAA mode needs a signed BAA and an Enterprise plan or add-on, and EU hosting is a separate region.
Open-source framework (e.g. LiveKit Agents)
Right when you want to own the agent code and choose your own models and hosting, and you have engineers to run it. LiveKit Agents is Apache 2.0 open source, handles the STT-LLM-TTS pipeline, turn detection, and interruptions, and connects to SIP. You can self-host it or run it on LiveKit Cloud, which offers BAAs on higher tiers and region pinning.
Model API direct (e.g. OpenAI Realtime API)
Right when a single-user voice experience needs the most natural speech-to-speech feel with the fewest moving parts. It connects over WebRTC, WebSocket, or SIP. For healthcare, check before building: when we last checked (July 2026), the audio modality wasn’t on Azure’s HIPAA-eligible service list. Read the production guide.
Custom pipeline (what we build)
Right when audio must stay in the EU or on your servers, the agent must fit an existing PBX or contact centre, volume makes per-minute pricing the biggest cost line, or you need control over latency the platform doesn’t expose. We build on the frameworks and APIs above where they fit and replace them where they don’t. For the full cost model, see the cost of running a voice AI platform.
| You need… | Managed platform | Open framework | Model API direct | Custom pipeline |
|---|---|---|---|---|
| Fastest route to a working agent | ✅ Best | Good | Good for single-user | Slowest start |
| Modest volume, standard call flow | ✅ Best | Good | Good | Usually overkill |
| Audio stays in the EU or on your servers | Check the vendor’s region and plan | ✅ Self-host | Check per endpoint | ✅ Designed in |
| HIPAA with audio | Check BAA scope and plan | Self-host, or a cloud BAA on higher tiers | Verify audio modality coverage | ✅ BAA chain designed per vendor |
| Existing PBX / contact-centre integration | SIP trunk only | SIP, with your engineers | SIP | ✅ Full PBX and dialplan work |
| Cost at high call volume | Per-minute | Infra + model APIs | Per audio token | Infra + model APIs, optimisable per stage |
| Control over jitter buffer, endpointing, topology | Limited | Partial | Limited | ✅ Full |
Connecting Voice Agents to Real Phone Lines
A voice agent that only works in a browser tab isn’t a phone agent. Most production voice AI still has to answer a phone number, which means SIP, the public phone network, and whatever PBX the business already runs.
SIP trunks and the phone network
Inbound and outbound calls reach the agent through your SIP provider or PSTN gateway. We handle signalling, codecs, and narrowband audio (8 kHz phone audio recognises differently from a browser mic), so recognition holds up on real phone calls.
Your existing PBX or contact centre
Agents often have to live beside a FreeSWITCH or Asterisk deployment, with dialplans, queues, and human agents. The architecture decision that matters most: keep AI inference off the switch’s media threads, so one slow LLM call doesn’t stall audio for every other caller.
One agent for phone and app callers
The same agent can take calls from a phone line and from a browser or mobile app. That means bridging WebRTC and SIP correctly, including TURN and encryption on the browser side.
Compliant call paths
Encrypted signalling and media on every leg, internal ones included; recording consent; and call logs that don’t turn into an unplanned store of patient data.
We run FreeSWITCH in production as a SIP proxy and PSTN bridge on a HIPAA/GDPR-compliant healthcare communications platform, connecting WebRTC and mobile clients with phone callers and with hospitals’ own SIP conferencing infrastructure. More in FreeSWITCH in production.
Latency: What We Target, and What We've Measured
Latency claims in voice AI are mostly marketing. Here are the two things we’ll show you, kept separate: the engineering targets we design against, and results from systems we’ve delivered.
Engineering targets (industry-reported ranges, not our measured results)
- First response after the caller stops speaking: roughly 500–1,200 ms on the first turn and 300–600 ms on later turns feels human; past about 2 seconds, callers assume the line dropped.
- A well-chosen cascaded pipeline can land in roughly 450–950 ms; production medians reported across the industry sit nearer 1.4–1.7 s, which is the gap tuning closes.
- Barge-in: the agent should stop within tens of milliseconds of the caller starting to speak, with queued audio flushed.
→ Where every millisecond goes: Voice AI agents on WebRTC: latency, barge-in, and turn-taking
Measured on systems we delivered
| System | Result | Case study |
|---|---|---|
| AI phone-screening agent (staffing) | 10× screening throughput, same-day turnaround, 80%+ fewer recruiter screening hours | Voice recruiting bot |
| In-call medical translation (HIPAA/GDPR) | Sub-second from speaker’s utterance to dubbed audio, zero audio persistence | Healthcare video translation |
| Real-time speech-to-text (media) | Partial results in 1–2 s, on-premises, no audio sent to external services | Real-time speech-to-text |
Our Voice AI Development Process
Voice UX Research
Conversation design, intent mapping, user journey analysis, persona definition. We design how the voice assistant should think and talk before building anything.
Architecture & Model Selection
Cascaded or speech-to-speech; STT, LLM, and TTS choice; the telephony path (SIP, PSTN, PBX); where audio is allowed to go; and a latency budget for every stage. You approve the architecture before we build on it.
Development & Training
Voice pipeline development, conversation flows, domain-specific training, integration with your systems. You hear working voice demos every 2 weeks.
User Testing & Tuning
Real callers, diverse accents, and narrowband phone audio, plus simulated packet loss and jitter. We tune barge-in, endpointing, and fallbacks under the network conditions your users actually have, not on a clean office connection.
Deployment & Integration
Production deployment, telephony or app integration, monitoring setup. We ensure the voice AI works in your real environment with real users.
Learning & Optimization
Conversation analysis, failure-pattern review, prompt, vocabulary, and flow tuning, and latency monitoring in production. We fine-tune models only where that measurably beats a better prompt or vendor.
Voice AI Technology Stack
We work across the full voice AI pipeline — from speech recognition to response generation.
Speech Recognition
Language Models
Speech-to-Speech
Text-to-Speech
Voice Infrastructure
Conversation Design
Agent Frameworks
Monitoring & Analytics
Voice & Speech AI We've Delivered
Production voice and speech systems, each linked to its full case study.
AI Phone-Screening Voice Agent
Staffing Company
Challenge: An autonomous voice agent calls candidates, runs the screening conversation, handles interruptions and follow-ups, then scores and summarises every answer for recruiters. A multi-signal turn-taking model (audio energy, cadence, syntactic completeness, prosody) kept candidates on the call. 10× screening throughput, same-day turnaround, 80%+ fewer initial screening hours.
Read Full Case Study →Real-Time Translation Inside HIPAA-Compliant Video Calls
Healthcare
Challenge: Clinician speech is recognised, translated, and delivered as dubbed audio plus subtitles inside the same encrypted call, with audio intercepted at RTP level in a custom mediasoup SFU. Sub-second translation latency, zero audio persistence, HIPAA and GDPR compliant.
Read Full Case Study →Context-Aware Real-Time Speech-to-Text
Media Production
Challenge: Streaming recognition with speaker diarization and live, context-aware error correction, accurate enough for live captions. Partial results in 1–2 seconds; deployed on-premises on GPU, with no audio sent to external services.
Read Full Case Study →What Our Clients Say
“They know the inner workings of the tech and were able to inherit our semi-functional code and get it to work where multiple prior teams couldn’t.
“Trembit built us a voice bot that screens candidates the way our best recruiters do — it asks the right questions, listens to the answers, and tells us who to talk to next. We went from a three-day screening backlog to same-day turnaround, and we are not losing top candidates to competitors anymore.
“Trembit built us a transcription system that actually knows what we are talking about. […] For live broadcasts, the captions are reliable enough that we stopped keeping a human captioner on standby.
Why Choose Trembit for Voice AI Agent Development?
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15+ Years at the transport layer
We find the latency a model benchmark can't
When an agent feels slow, we measure every stage, from speech recognition, the LLM, and synthesis to the jitter buffer, codec, and network hop, before touching the model. Protocol-level WebRTC work across 50+ video and voice projects is why we look below the API.
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SIP + WebRTC in production
Phone callers and app users, one agent
Voice agents have to answer real phone lines. We run FreeSWITCH in production as a SIP proxy and PSTN bridge on a compliant healthcare platform, and we build WebRTC voice for browser and mobile, so both sides of the bridge are familiar ground.
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HIPAA · GDPR · KBV
Compliance for audio, not just text
We’ve shipped real-time speech AI inside a HIPAA/GDPR call path with zero audio persistence, on-premises speech-to-text where audio couldn’t leave the building, and a KBV-certified psychotherapy video platform in Germany. Compliance is designed in from the first sprint.
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Platform-neutral
We'll tell you when a platform is enough
If Vapi, Retell, or LiveKit Cloud fits your volume and compliance scope, we’ll say so on the scoping call. We build custom when your product depends on control the platforms don’t give you, not because it’s the bigger contract.
Voice AI Engagement Models
Voice AI Proof of Concept
A focused 4-8 week sprint building a working voice assistant prototype with your domain data and real conversation flows.
Best for: Teams exploring voice AI for the first time or validating a specific voice use case.
Dedicated Voice AI Team
Speech engineers, NLU specialists, and conversation designers embedded in your product team for ongoing voice AI development.
Best for: Companies building voice-first products or replacing legacy IVR systems.
Voice AI Consulting
Expert assessment of your voice AI opportunity — technology selection, architecture design, and implementation roadmap.
Best for: Teams evaluating voice AI platforms against a custom build, or with an agent already live that is too slow or blocked by compliance.
Frequently Asked Questions
Planning a Voice AI Agent?
Tell us who the agent talks to, over which channel, and what data it touches. A real-time engineer will map the latency budget and the compliance path, and tell you honestly if a platform would serve you better than a custom build.