About the role

We're looking for a VoIP engineer to build, run, and debug our telephony infrastructure. You'll work hands-on with our SIP servers, call flows, and the cloud infrastructure they run on from provisioning a current SIP proxy to tracing a one-way-audio complaint down to a specific RTP stream. Calls either connect and sound good, or they don't, and you'll own that outcome end to end.

What you'll do

- Build and operate the telephony stack
- Set up and maintain VoIP nodes SIP proxies, PBX, media servers — and wire them into a resilient stack
- Configure dialplans, routing logic, and call flows (IVR, transfers, forking, recording, conferencing)
- Manage carrier and SIP trunk connectivity: registrations, DIDs, number routing, failover between providers
- Configure supporting services — SSH, web servers, reverse proxies, process managers

Debug and improve call quality

- Investigate call failures and quality issues: SIP signaling errors, registration problems, NAT traversal, codec negotiation, one-way audio, jitter, packet loss
- Take and read packet captures (tcpdump, Wireshark, sngrep) and correlate them with logs to trace a call end to end
- Track quality metrics (MOS, ASR, ACD, latency) and drive them in the right direction




- Write up root-cause analyses so the same failure doesn't recur

Own the cloud infrastructure

- Deploy and manage AWS infrastructure: VPCs, subnets, route tables, security groups, EC2, Elastic IPs
- Handle firewall rules, port management, and DNS for SIP and RTP traffic
- Set up monitoring, alerting, and logging so problems are caught before customers report them
- Contribute to upgrades, capacity planning, and incident response

Work with the team

- Partner with backend engineers on telephony APIs, webhooks, and call-event handling
- Support customer-facing teams on escalated call issues with evidence-backed explanations
- Document configurations, runbooks, and troubleshooting guides

Must-haves

- Familiarity with any telephony/VoIP stack (FreeSWITCH, Kamailio, OpenSIPS, Asterisk, or similar)
- Working understanding of SIP and RTP: call setup, registration, common response codes, media negotiation
- Comfortable in a Linux terminal (systemd, logs, permissions, basic shell scripting)
- Basic networking knowledge — NAT, firewalls, ports, DNS, TCP/UDP
- Experience with AWS or a comparable cloud/VPC platform
- Ability to procure and read packet captures and logs to trace a call end to end

Clear written communication

📌 VoIP Engineer (Gurugram)
🏢 Mijas Consultants Jaipur
📍 Gurugram