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VoIP Infrastructure Engineer

ViH Labs and Innovation

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# 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.

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Listed on wellfound · posted 2026-09-25. ApplySarthi collects openings and links to application pages; the role is advertised by ViH Labs and Innovation, not by us.