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انتهت صلاحية هذه الوظيفة في 19/09/2026. لم تعد تقبل الطلبات.

Senior Voice AI Engineer / Voice Platform Engineer

Parse AI

Senior 🇬🇧 English
Twilio Media Streams SIP trunking WebRTC LiveKit Telnyx Voice activity detection Telephony audio formats Voice latency measurement Production debugging Provider evaluation and tuning

وصف الوظيفة

About the role

Parse AI is seeking a Senior Voice AI Engineer to own the real‑time voice layer of its AI‑powered call‑center platform for hospitality operators. You will be responsible for building a reliable, low‑latency voice pipeline that handles telephony, speech recognition, LLM orchestration, text‑to‑speech, barge‑in, and call transfers.

Key responsibilities

  • Design and improve real‑time STT → LLM → TTS pipelines.
  • Fine‑tune voice providers such as Deepgram, OpenAI, ElevenLabs, Cartesia, Hamsa, etc.
  • Enhance VAD, endpoint detection, silence handling, turn‑taking, and barge‑in behavior.
  • Build pronunciation and recognition lexicons for multilingual hospitality vocabularies.
  • Work with Twilio Media Streams, SIP trunking, call transfers, and telephony reliability.
  • Measure and reduce voice latency across STT, LLM, TTS, network, and telephony layers.
  • Debug production voice issues like garbled audio, echo, false interruptions, and failed transfers.
  • Scale concurrent call capacity across multiple customers and environments.
  • Collaborate with AI and backend engineers to ensure a fast, safe, production‑ready agent experience.

Required profile

  • Proven experience building and operating production voice systems.
  • Strong background in real‑time voice agents or voice platforms.
  • Hands‑on work with STT, TTS, and speech‑to‑speech technologies.
  • Familiarity with Twilio Media Streams, SIP trunks, WebRTC, LiveKit, Telnyx, or similar telephony solutions.
  • Expertise in voice activity detection, endpointing, barge‑in, and interruption handling.
  • Experience handling multilingual or accent‑heavy voice environments.
  • Ability to evaluate and tune multiple STT/TTS providers.

Required skills

  • Real‑time voice agent development
  • Speech‑to‑text (STT) and text‑to‑speech (TTS) integration
  • Large language model (LLM) orchestration
  • Twilio Media Streams
  • SIP trunking
  • WebRTC / LiveKit / Telnyx
  • Voice activity detection (VAD)
  • Endpoint detection and barge‑in handling
  • Streaming audio processing and audio buffer management
  • Telephony audio formats
  • Voice latency measurement and optimization
  • Production debugging of phone‑call issues
  • Multilingual/accent‑heavy speech system support
  • Provider evaluation and tuning across STT/TTS vendors

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