Job
Description
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 o Experience working with low-latency, real-time audio streaming. o Familiarity with WebRTC, SIP/RTP, and VoIP streaming. Programming Middleware Development o 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 o Familiarity with SIP, FreeSWITCH, Asterisk, or other PBX platforms. o Experience working with SIP trunking providers (Telnyx, Bandwidth, Flowroute, etc.). Performance Optimization o 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: o 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.
