VoIP/Telephony Engineer (FreeSWITCH/Kamailio)
Swadesh Mobile Private Limited
Mumbai, Maharashtra, India · Full Time
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- Experience
- 2–12 yrs
- Salary
- INR 25,000 – INR 149,000 / month
- Openings
- 1
- Posted
- 5 days ago
- Work mode
- In office
- Resume
- Required to apply
Where you'll work
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Job description
About the Company
We operate a cloud-based phone system, a contact center, and a wholesale voice platform. We manage our own switches and carrier routes and do not act as resellers.
Role Overview
You will be responsible for full ownership of the call path. Any issue involving calls connecting, dropping, or audio quality falls within your scope.
Key Responsibilities
- Manage and maintain our FreeSWITCH and Kamailio infrastructure, including dialplan, routing, registration, and media handling.
- Troubleshoot and resolve real call failures such as one-way audio, call drops after fixed durations, failed transfers, codec conflicts, and calls that ring but do not connect.
- Develop and optimize contact center queues, dialers, and campaign call flows.
- Handle browser-based calling using WebRTC, including ICE, TURN, and SRTP protocols.
- Manage carrier interconnects, SIP trunks, IP allowlisting, and liaise with carriers to resolve faults.
- Monitor call quality metrics including ASR, ACD, PDD, MOS, jitter, and packet loss.
- Implement audio forking into AI systems for transcription and live agent assistance.
- Configure emergency call routing and location handling with compliance management.
Candidate Expectations
- Analyze packet captures and log lines to accurately diagnose call issues—no guesswork involved.
- Plan and schedule changes carefully, avoiding restarting live SIP profiles during traffic hours; prepare rollback plans.
- Validate fixes with real bidirectional test calls before confirming resolution.
- Document all dialplan and routing changes thoroughly with explanations and rollback instructions.
- Manage carrier interactions directly—open tickets with attached traces and pursue resolution.
- Within the first month, map our switch infrastructure and create comprehensive call-flow documentation.
- Within three months, take ownership of voice on-call rotation and eliminate recurring faults.
- Be prepared for on-call duties including night time response to live voice issues.
Mandatory Qualifications
- Minimum 2 years of hands-on production VoIP experience with live traffic; candidates with 8-12 years are highly valued.
- Expertise in FreeSWITCH, including dialplan, mod_callcenter, ESL, and Lua scripting.
- Proficiency with Kamailio focusing on routing logic, dialog management, and NAT traversal.
- Strong knowledge of SIP and RTP protocols at the packet level, capable of analyzing traces to detect faults.
- Daily practical experience with diagnostic tools like sngrep or Wireshark.
- Familiarity with codecs such as PCMU, PCMA, Opus, G.729, and understanding codec negotiation issues.
- Comfortable using Linux command line effectively.
Preferred Skills
- Experience with WebRTC and coturn servers, SRTP, TLS for SIP, and STIR/SHAKEN implementation.
- Understanding of toll fraud detection and prevention.
- Programming skills in Python or Node.js.
- Background working at a carrier or ITSP rather than solely in reseller roles.
Compensation
The salary range for this full-time role is between ₹25000 and ₹149000 per month, competitive depending on experience and expertise.
Application Process
Send your CV along with a detailed explanation of the toughest SIP or RTP issue you have troubleshot, including the symptom, diagnosis process, and verification method for resolution.