Senior VoIP Engineer

Tin Can is building a safer, simpler way for kids to connect — without smartphones. We’re creating screen-free, delightful devices and services that let families call the people who matter most, free from the noise of today’s digital world.

We’re building a bold, authentic, nostalgic, and kinda quirky brand that resonates with folks who want something simpler & better for their kids than the tech-infused lives we’re currently living (and who have a sense of humor about it). As we gear up to scale to thousands of families, we’re ready to bring on an engineer to help make it happen.

A call either connects or it doesn’t — there’s no partial credit when a kid picks up a Tin Can to reach Grandma, or to dial 911. We’re hiring a Senior VoIP Engineer to own the SIP and media plane that carries every one of those calls: how it routes, how it sounds, and how we know it works. It’s the first role dedicated entirely to that stack.

Our production stack is Kamailio (SIP proxy and registrar) and FreeSWITCH (media and voicemail), running on AWS across carrier trunks and our own device fleet. This is a deeply hands-on role: routing logic, dial plans, and packet captures every day. You’ll report to the Head of Engineering and own the call itself. You’ll set the direction for how the voice plane is designed and tuned and when a registration flaps at 3am, you’re the person who can go from a SIP trace to root cause and out the other side with a fix.

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  Own our production SIP and media plane end to end — Kamailio, FreeSWITCH, and the carrier trunking behind every call to the outside world

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  Be the reviewer of record for call-path changes, and author the SIP test scenarios that gate a voice change before it ships

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  Write and maintain the Kamailio routing logic behind every call — dispatcher, registrar and usrloc tuning, htable, DMQ replication, TLS

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  Build FreeSWITCH dialplans and integrations (ESL, mod_xml_curl, Lua, Sofia profiles), and keep call state correct across boxes, transfers, and NAT

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  Produce the packet-level truth about call quality — MOS, jitter, packet loss, setup time — and make it trustworthy enough to alarm on

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  Model and load-test call capacity — concurrent calls, calls per second, trunk headroom — against our holiday peak, using scenarios built on real call flows

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  Extend the SIP and media layer into new countries as we expand, where codec support and emergency-calling rules change market to market

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  Harden the signaling and media path: SIP over TLS, SRTP, rate limiting, toll-fraud prevention, and topology hiding

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  Raise the team’s voice fluency: the documentation, runbooks, and reviews that make this stack legible to every engineer here

• 5+ years building and operating real-time voice systems in production, with hands-on Kamailio and FreeSWITCH depth

• You can hold your own on SIP and RTP: location services, dialog state, transfers, Path routing, NAT traversal, and what happens to media when a packet arrives late

• You’ve debugged live production calls from a packet capture and a SIP trace to root cause

• You treat config as code and infrastructure as reproducible

• Fluency using AI as a multiplier across diagnosis, analysis, and testing

• A clear communicator who writes things down

• A collaborative, low-ego approach — everyone here sweeps the floor

• Bonus: rtpengine, Homer/HEP for capture and call-leg correlation, SIPp regression suites, Prometheus exporters for the voice plane, or STIR/SHAKEN and carrier interconnect

• Extra bonus: WebRTC internals (ICE, DTLS-SRTP, TURN) and bridging WebRTC to SIP