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Job Description

What You'll Do :


- Develop real-time voice bots using WebRTC, LiveKit, WebSockets, and telephony integrations.


- Build ASR pipelines using Whisper, VAD, turn detection, and inference systems.


- Fine-tune and deploy Generative AI models using Ollama, vLLM, and HuggingFace.


- Optimize TTS streaming systems using Orpheus, Spark, Fish Audio, or similar open-source models.


- Implement Python-based inference microservices and MLOps pipelines for scalable deployments.


- Ensure low-latency, high-performance, and secure streaming voice solutions.


You Should Have :


- 3+ years of relevant experience in real-time AI and voice streaming development, primarily working with open-source models without using OpenAI, Azure, or Amazon Lex.


- Strong Python programming skills with hands-on experience in ASR, TTS, WebRTC, LiveKit, and WebSockets.


- Knowledge of Generative AI, model fine-tuning, quantization techniques, and HuggingFace hosting.


- Experience with MLOps, scalable microservices architecture, and low-latency streaming systems.


- Familiarity with telephony integrations and real-time conversational AI systems.

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