VoIP Infrastructure Engineer

VoIP Voice Infrastructure Engineer

About the Role

We are looking for a VoIP / SIP & Voice Infrastructure Engineer with 2–3+ years of hands-on experience in SIP, Asterisk, VoIP, RTP and telephony infrastructure to help us build and integrate reliable telephony infrastructure for our AI Voicebot platform.

You will work closely with our AI, backend and infrastructure teams to integrate SIP lines, telecom providers and voice infrastructure with our real-time Voice AI architecture.

The ideal candidate should understand how a phone call travels end-to-end — from SIP signalling and call establishment to RTP/audio streaming, call routing, transfers and termination — and be comfortable troubleshooting issues across the entire stack.

Key Responsibilities

• Design, configure and integrate SIP-based telephony infrastructure with our Voice AI platform.
• Work with Asterisk, FreeSWITCH or similar VoIP platforms for call routing and telephony integration.
• Integrate SIP trunks / SIP lines from telecom providers with our backend and Voicebot infrastructure.
• Configure and troubleshoot SIP signalling, RTP media streams, codecs and call flows.
• Build reliable inbound and outbound calling flows for AI Voicebots.
• Integrate telephony systems with WebSocket-based real-time audio streaming and Voice AI services.
• Work with SIP registration, authentication, routing, dial plans and number management.
• Handle call bridging, transfers, DTMF, caller ID, IVR and call termination.
• Diagnose issues related to one-way audio, no audio, call drops, latency, SIP registration failures, codec mismatches and RTP connectivity.
• Work with NAT, firewalls, STUN/TURN and network-level VoIP configurations where required.
• Monitor call quality and infrastructure performance and help improve latency, reliability and concurrent call capacity.
• Develop scripts/tools for automation, monitoring and troubleshooting of telephony infrastructure.
• Collaborate with backend engineers to integrate call events, call IDs, customer metadata and call status into our platform.
• Work with DevOps teams on deploying and scaling VoIP infrastructure using Linux, Docker and Kubernetes where applicable.
• Maintain technical documentation for SIP integrations, call flows and telephony configurations.

Required Skills

VoIP / Telephony

• 2–3+ years of hands-on experience in VoIP / SIP / telephony systems.
• Strong understanding of SIP protocol and SIP call flows.
• Hands-on experience with Asterisk or FreeSWITCH.
• Good understanding of RTP and real-time audio streaming.
• Experience with SIP Trunks, SIP registration and telecom/VoIP providers.
• Understanding of codecs such as G.711 (PCMU/PCMA), Opus and G.729.
• Experience troubleshooting SIP signalling and RTP/media issues.

Networking & Infrastructure

• Strong Linux fundamentals.
• Understanding of TCP/IP, UDP, DNS, NAT, firewalls and network routing.
• Experience troubleshooting real-time communication/networking issues.
• Familiarity with Docker and preferably Kubernetes.
• Experience working with cloud infrastructure such as AWS/GCP/Azure is a plus.

Programming / Automation

• Comfortable with Python, Bash or similar scripting languages.
• Ability to work with REST APIs and WebSockets.
• Familiarity with integrating telephony systems with backend applications.

Good to Have

• Experience working with AI Voicebots / Conversational AI / Speech AI.
• Experience integrating SIP with WebSocket or WebRTC-based audio systems.
• Experience with Exotel, Twilio, Plivo, Vonage, Knowlarity or similar telecom platforms.
• Experience with real-time ASR/TTS pipelines.
• Experience handling high-concurrency voice traffic.
• Knowledge of SIPREC, WebRTC or SRTP.
• Experience with telecom compliance, call recording and call analytics.
• Familiarity with tools such as sngrep, tcpdump and Wireshark for SIP/RTP debugging.

What You Will Work On

You will be part of the team building our real-time Voice AI infrastructure.

A typical call flow may involve:

Customer → Telecom/SIP Provider → SIP Infrastructure → Voicebot Gateway → WebSocket Audio Stream → AI/ASR/LLM/TTS Pipeline → Audio Stream → Customer

You will be responsible for making the telephony layer reliable, scalable and tightly integrated with the Voice AI stack.

You will also work on integrating new telecom providers and SIP lines while ensuring that inbound and outbound calls can be provisioned and operated with minimal manual intervention.

Ideal Candidate

We are looking for someone who:

• Understands VoIP beyond configuration-level knowledge.
• Can independently troubleshoot a SIP call using logs, SIP traces and packet captures.
• Is comfortable working close to the infrastructure/networking layer.
• Can understand both telephony systems and software engineering.
• Enjoys solving real-time communication and latency-sensitive problems.
• Can work closely with AI, backend and DevOps engineers.
• Is comfortable taking ownership of a telephony integration from provider → SIP infrastructure → Voicebot → production deployment.

Why Join Us?

• Work on production-grade Voice AI infrastructure.
• Solve challenging problems involving SIP, real-time audio, AI and distributed systems.
• Work directly with AI researchers, backend engineers and infrastructure teams.
• Opportunity to work on large-scale conversational AI and voice automation use cases.
• High ownership and opportunity to influence the architecture of our Voice AI platform.