VoIP Engineer

Mijas Consultants Jaipur

Hyderabad

On-site

INR 1,000,000 - 2,000,000

Full time

14 days+
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Job summary

We are seeking a skilled VoIP engineer to build, run, and debug our telephony infrastructure. You will own end-to-end call flow, SIP servers, and cloud deployment from provisioning a SIP proxy to tracing one-way audio issues.

You will configure dialplans, manage SIP trunk connectivity, and ensure resilient operation across AWS-based infrastructure. You will work with backend engineers and support teams to diagnose and resolve call quality problems quickly.

Qualifications

  • Familiarity with 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.

Responsibilities

  • 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.

Skills

VoIP stack familiarity
SIP/RTP knowledge
Linux terminal
Networking fundamentals
AWS/cloud experience
Packet capture analysis

Tools

FreeSWITCH
Kamailio
OpenSIPS
Asterisk

Job description

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 new 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

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