Voice Engineer

DFX5 Corporation; Colombia; Presencial; COP 90.000.000 - 140.000.000; Jornada completa

Descripción de la vacante

DFX5 Corporation seeks a senior engineer to own and build the edge voice platform. You will implement routing, media relay, and deployment automation, and own SIP trunking from carrier to WebRTC sessions.

You will work on a live production system, tracing dropped calls, and ensuring robust SIP-TLS and SRTP in production. Strong Linux, AWS, and programming skills are required.

Formación

• 5+ years building and operating production real-time voice systems.
• Experience with Python, Go, TypeScript, or C and code review practices.
• Deep SIP knowledge including trunks, SDP, NAT traversal, and RTP/SRTP.
• Production experience with open-source SIP servers and media relays.

Responsabilidades

• Build and maintain voice platform services: call routing, media relay, provisioning and CI delivery.
• Own SIP trunk architecture, carrier interconnects, and NAT traversal details.
• Implement SIP-TLS and SRTP end-to-end in production environments.
• Automate provisioning with infrastructure-as-code and image pipelines.
• Develop observability: CAC, metrics, alerts, and failure tracing.
• Review documentation and code before production and mentor junior engineers.
• Participate in on-call rotation for production voice.

Conocimientos

Real-time voice systems

Python

Go

TypeScript

C

CI/CD

AWS

Linux fundamentals

SIP/SRTP

Educación

Bachelor's degree in Computer Science or related field

Herramientas

Cisco/Webex

Audiocodes

MS Teams

Descripción del empleo

At DFX5 we build edge technology that moves our clients' businesses forward - software and cloud solutions for some of the largest financial institutions, retailers, telcos, CPG and mining companies in Latin America and beyond. Voice is one of the places we do it: we build and run the platform that carries our clients' calls between carrier networks and the browser.

We're looking for a senior engineer to build and own that platform. You'll write the services that route calls and relay media, the automation that deploys them, and the instrumentation that proves they work. You'll own the SIP side end to end, from the carrier trunk to the moment a call becomes a WebRTC session.

This is an engineering role on a live production system. You will spend your days in code and in packet captures.

What You'll Do

• Build and maintain the services behind the voice platform: call routing logic, media relay, failover, provisioning and tenant configuration - as code, reviewed, tested and shipped through CI.
• Own SIP trunk architecture and carrier interconnects: dial plans, SDP and codec negotiation, port ranges, NAT traversal.
• Implement and troubleshoot SIP-TLS and SRTP in production, not as a checkbox - carrier contracts increasingly require both, and you'll be the one who makes that true end to end.
• Write the automation that provisions and configures the platform: infrastructure-as-code, image pipelines, environments that rebuild the same way every time.
• Build the observability that makes a dropped call explainable: call detail capture, metrics, alerting, and the tooling to trace a failure to its cause.
• Review technical documentation before it reaches a client. A wrong IP or an unimplemented control listed as "done" in a client-facing doc is a production incident waiting to happen.
• Review other engineers' code and configuration before it touches production, and mentor the engineers coming up behind you.
• Take part in the on-call rotation for production voice.

What You Bring

• 5+ years building and operating production real-time voice systems that carry real traffic.
• You write code as a daily habit - Python, Go, TypeScript, C or similar - and you version it, test it and put it through review. The platform is software, and you will be building it.
• Experience with technologies such as: Cisco/Webex, Audiocodes, MS Teams.
• Deep SIP: trunks, SBCs, dial plans, RFC 3261, SDP negotiation, DTMF (RFC 2833), NAT traversal, RTP and SRTP.
• Production experience with open-source SIP servers, proxies and media relays, extended with your own code where the stock behavior wasn't enough.
• SIP-TLS and SRTP implemented in a real deployment, and you can explain why a carrier would refuse plaintext SIP.
• Strong Linux fundamentals, and the discipline to take a packet capture and trace a dropped call to a root cause instead of guessing.
• Strong AWS fundamentals (EC2, VPC, security groups) and infrastructure-as-code.
• Git, code review, CI/CD and agile delivery as normal working habits.
• English good enough to run a technical call with a carrier or a client without a translator.

Education & certifications

• Bachelor's degree in Computer Science, Software/Systems Engineering, Telecommunications or a related field. Equivalent professional experience will be considered.
• Cloud credentials (AWS Solutions Architect, AWS Developer) are a strong plus.

How We Select

The process includes a coding exercise and a SIP troubleshooting exercise. Both are hands-on, and both are part of the interview.

The impact you'll have

The platform you build carries live customer calls for banks, retailers and telcos across Latin America. When it fails, a contact center stops answering, so the reliability work you do shows up in someone's business the same day. You'll also set how the rest of the team builds on it.

Why DFX5

• Build edge technology that matters. You'll create solutions for customers across financial institutions, retail, telcos, CPG and mining.
• Work with the best. You'll engage with top companies in Latin America and around the world.
• Big challenges, big rewards. You'll tackle genuinely complex problems - and be rewarded for solving them.