Senior Voice AI Engineer

Clera; United States; Remote; USD 96,000 - 110,000; Full time

Job summary

Clera is seeking a founding engineer to own the real-time voice layer, from speech input through AI reasoning to spoken output. You will help ensure end-to-end latency is minimized in production, delivering natural, responsive voice interactions.

You will build streaming speech-to-text, text-to-speech, and transport layers while collaborating with the founders on architectural decisions in a fast-moving, fully remote team with core overlap 13:00–17:00 UTC.

Qualifications

• 5+ years building production software.
• 2+ years shipping voice/real-time audio.
• Strong Python or TypeScript skills.
• Experience with audio stacks such as LiveKit, Pipecat, Vapi, Twilio Media Streams, Daily.
• Experience debugging audio at frame level (sample rates, codecs, jitter).
• Experience building LLM evaluation harnesses and optimizing latency.

Responsibilities

• Build and own streaming speech-to-text, LLM turn-taking, text-to-speech, and transport.
• Measure and reduce latency, targeting first audio under 800 ms on real calls.
• Address interruptions, barge-in, silence detection, overlapping speech, accents.
• Build an evaluation harness from recordings to detect regressions.
• Compare voice providers and models through evidence-based testing.
• Instrument production systems for turn latency and cost per minute.
• Work directly with founders and make technical decisions in a fast-moving team.

Skills

Real-time audio

Latency optimization

Python/TypeScript

Speech tech

Tools

LiveKit

Twilio Media Streams

WebRTC

Pipecat

Vapi

WebSocket

Daily

Job description

About the Role

As a founding engineer on a small conversational AI team, you will own the real-time voice layer, from incoming speech through AI reasoning to spoken responses. You will help make natural, responsive voice interactions work reliably in production, with a focus on end-to-end latency.

What You'll Do

• Build and own streaming speech-to-text, LLM turn-taking, text-to-speech, and telephony or WebRTC transport.

• Measure and reduce latency, targeting first audio under 800 milliseconds on real calls.

• Address interruptions, barge-in, silence detection, overlapping speech, poor audio, accents, and mid-sentence changes.

• Build an evaluation harness from recorded calls, transcripts, and scored turns to detect regressions and guide product decisions.

• Compare voice providers and models through evidence-based testing, and make changes based on results.

• Instrument production systems for turn latency, transcription confidence, drop-offs, and cost per minute.

• Work directly with founders and make technical decisions in a fast-moving team.

What We're Looking For

• At least 5 years building production software, including 2 or more years shipping voice, speech, or real-time audio systems.

• Experience building and shipping end-to-end real-time voice pipelines, including streaming speech recognition, LLM turn-taking, speech synthesis, and telephony or WebRTC.

• Strong Python or TypeScript skills and comfort working in both.

• Hands-on experience with an audio stack such as LiveKit, Pipecat, Vapi, Twilio Media Streams, Daily, or a custom WebSocket implementation.

• Experience debugging audio at the frame level, including sample rates, codecs, jitter, and voice activity detection thresholds.

• Experience building LLM evaluation harnesses, optimizing latency against real-world targets, and using evaluation results to make product decisions.

• Clear written English for asynchronous communication. Experience with speech model serving or fine-tuning, SIP, telephony, or LLM orchestration frameworks is a plus.

Compensation & Benefits

Compensation is $96,000 USD annually, regardless of location. Visa sponsorship is not available.

Location

Fully remote, anywhere in the world. Core team overlap is 13:00 to 17:00 UTC.