Role Overview
As an AI Voice & LLM Integration Developer, you will play a pivotal role in building and optimizing company’s AI-driven voice platform. 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 & handoffs with AI agents.
- Ensuring high-quality, natural-sounding AI-driven voice interactions.
- Implementing https://livekit.io/ into Azure
- Integrating external trunking provider like Twilio/Bandwith.com, etc into livekit
- Working with front end development team to build voice into core AI agent platform
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 handoff 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
- Experience integrating GPT APIs (ChatGPT, Whisper, TTS engines) for voice applications.
- Understanding of real-time AI voice processing and conversational AI workflows.
- Voice & Audio Streaming
- Experience working with low-latency, real-time audio streaming.
- Familiarity with WebRTC, SIP/RTP, and VoIP streaming.
- Programming & Middleware Development
- Proficiency in Python, Node.js, or Go for API and middleware development.
- Experience developing RESTful APIs and real-time streaming solutions.
- VoIP & Telephony Knowledge
- Familiarity with SIP, FreeSWITCH, Asterisk, or other PBX platforms.
- Experience working with SIP trunking providers (Telnyx, Bandwidth, Flowroute, etc.).
- Performance & Optimization
- Ability to reduce latency in AI-driven conversations.
- 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).
- Background in signal processing or audio engineering.
- Familiarity with cloud-based voice solutions (AWS Connect, Twilio Voice, Dialogflow CX).
- Knowledge of containerization (Docker, Kubernetes) for scaling AI voice workloads.
- Experience with multi-tenant architectures for voice AI.