Team Lead – Voice AI & Telephony Engineering

Experience: 3–5 years

Role Overview

We are seeking a hands-on technical leader experienced in real-time voice, conversational AI, VoIP, and telephony systems.

The ideal candidate has experience with LiveKit, FreeSWITCH/Asterisk, SIP, GSM and SIP gateways, WebRTC, STT, TTS, LLM APIs, Python, and backend engineering.

Key Responsibilities
Lead the design, development, and operation of voice and telephony systems.
Design systems using LiveKit, FreeSWITCH/Asterisk, SIP/GSM gateways, WebRTC, and real-time audio.
Integrate STT, TTS, LLM, and other AI APIs.
Own call flows, routing, dialplans, session management, and gateway integrations.
Troubleshoot SIP, VoIP, audio quality, latency, reliability, and concurrency issues.
Analyze production data and implement permanent fixes.
Design for scalability, low latency, reliability, and fault tolerance.
Review code and architecture, mentor engineers, and establish best practices.
Plan work, coordinate delivery, and support deployments and incidents.
Establish monitoring,



logging, alerting, and performance metrics.

Technical Requirements

Must Have
3–5 years of backend/software engineering experience.
Strong Python, asyncio, concurrency, and event-driven systems knowledge.
Experience building voicebots, conversational AI, or real-time voice applications.
Solid SIP and VoIP knowledge.
Experience with FreeSWITCH, Asterisk, or similar platforms.
Experience with SIP/GSM gateways and LiveKit or similar frameworks.
Understanding of WebRTC and real-time audio streaming.
Experience integrating STT, TTS, and LLM APIs.
Knowledge of REST, WebSockets, streaming APIs, retries, timeouts, and rate limiting.
Solid Linux, Git, debugging, and production troubleshooting skills.
Working knowledge of MySQL, PostgreSQL, or SQLite.

Leadership & Engineering Skills
Ability to own systems from design through production support.
Robust system design, debugging,

📌 Voice Ai Engineer Bharatpur
🏢 Mediatechtemple
📍 Bharatpur