VoIP/Telephony Engineer (free SWITCH/Kamailio)

VoIP / Telephony Engineer (FreeSWITCH + Kamailio)

 |  Full-time  |  ₹6 – 30 lakh  |  2+ years, 8–12 preferred

This is the deepest technical role in the company and the hardest to fill. If your VoIP experience is genuinely deep, we are flexible on everything else and you will sit at the top of the band.

About us

We run a cloud phone system, a contact centre and a wholesale voice platform. We own our switches and our carrier routes. We are not a reseller.

The role

You own the call path. Every call that connects, drops, or sounds wrong is your territory.

What you will do

• Run our FreeSWITCH and Kamailio estate — dialplan, routing, registration, media

• Debug real call failures: one-way audio, calls dropping at a fixed number of seconds, failed transfers, codec mismatches, calls that ring and never answer
• Build and tune contact centre queues, dialers and campaign call flows
• Browser calling over WebRTC — ICE, TURN, SRTP
• Carrier interconnects, SIP trunks, IP allowlisting, and chasing carriers when the fault is theirs
• Call quality: ASR, ACD, PDD, MOS, jitter and packet loss
• Fork call audio into our AI systems for transcription and live agent assist
• Emergency calling routing and location handling, with our compliance manager

What we expect from you

• You read traces, you do not guess. Every statement about a call is backed by a packet capture or a log line
• You never restart a live SIP profile during traffic hours to test a theory. Plan the change, pick the window, have the rollback ready
• You prove a fix with a real test call, in both directions, before it is called fixed
• Every dialplan or routing change is written down — what changed, why, and how to put it back
• You handle carriers yourself. Open the ticket, attach the trace, chase it until it is resolved
• First 30 days: map our switch estate and produce the call-flow document we do not have today
• First 90 days: own the voice on-call rota, and cut the repeat faults
• Ongoing: voice breaks in real time, and some of that will be at night. We do not pretend otherwise

Must you have

• 2+ years in production VoIP with real traffic. 8–12 years and we are very interested
• FreeSWITCH — dialplan, mod_callcenter, ESL, Lua
• Kamailio — routing logic, dialog handling, NAT traversal
• SIP and RTP at packet level. You can open a trace and find the fault
• sngrep or Wireshark, used daily, not once
• Codecs: PCMU, PCMA, Opus, G.729, and what happens when two sides disagree
• Linux command line, properly
• 
  

Preferred

WebRTC and coturn, SRTP and TLS for SIP, STIR/SHAKEN, toll fraud patterns, Python or Node, and time at a carrier or ITSP rather than only at a reseller.

To apply

CV to [email], plus one answer: describe the hardest SIP or RTP problem you have debugged. What was the symptom, how did you find the cause, and how did you prove it was fixed?