AI Voice & LLM Integration Engineer

Job Description

Role & responsibilities ole Overview As an AI Voice & LLM Integration Developer, you will play a pivotal role in building and optimizing Your work will focus on: • Integrating GPT's ASR (speech-to-text) and TTS (text-to-speech) APIs into our platform. • Handling real-time voice streaming from VoIP/SIP systems. • Optimizing low-latency communication between AI models and telephony platforms (e.g., FreeSWITCH, WebRTC). • Supporting PBX call routing & handoWs with AI agents. • Ensuring high-quality, natural-sounding AI-driven voice interactions. Key Responsibilities Design and develop real-time voice processing pipelines for AI-driven conversations. • Integrate OpenAI's Whisper (ASR) and TTS APIs with VoIP and WebRTC systems. • Develop low-latency audio streaming middleware between telephony providers(Telnyx, Bandwidth) and GPT. • Work with SIP and WebRTC to ensure seamless audio transmission. • Implement PBX call routing and AI handoW mechanisms (e.g., transferring AI-handled calls to human agents). • Optimize AI voice latency and response times for a real-time, natural experience. • Collaborate with VoIP engineers to integrate AI capabilities into SIP-based systems. • Design APIs to manage AI-driven IVR workflows and customer interactions. • Ensure compliance with telephony regulations (e.g., STIR/SHAKEN, call recording laws). • Monitor system performance and implement scalability strategies for voice interactions. Required Qualifications AI & LLM Integration o Experience integrating GPT APIs (ChatGPT, Whisper, TTS engines) for voice applications. o Understanding of real-time AI voice processing and conversational AI workflows. Voice & Audio Streaming Experience working with low-latency, real-time audio streaming. o Familiarity with WebRTC, SIP/RTP, and VoIP streaming. Programming & Middleware Development Proficiency in Python, Node.js, or Go for API and middleware development. o Experience developing RESTful APIs and real-time streaming solutions. VoIP & Telephony Knowledge Familiarity with SIP, FreeSWITCH, Asterisk, or other PBX platforms. o Experience working with SIP trunking providers (Telnyx, Bandwidth, Flowroute, etc.). Performance & Optimization Ability to reduce latency in AI-driven conversations. o Experience with audio codec optimization (G.711, Opus, etc.). One or more of the following additional qualifications: Experience with STT/ASR models beyond GPT (Google Speech-to- Text, Deepgram, Kaldi). o Background in signal processing or audio engineering. o Familiarity with cloud-based voice solutions (AWS Connect, Twilio Voice, Dialogflow CX). o Knowledge of containerization (Docker, Kubernetes) for scaling AI voice workloads. o Experience with multi-tenant architectures for voice AI.