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 positive, 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

Explicit written communication

📌 Voip Engineer Gurugram (India)
🏢 Mijas Consultants Jaipur
📍 India