SIP Software Engineer

Join a fast-growing engineering team building reliable, real-time communication software used at scale. This role offers the opportunity to work deep in the backend on systems where performance, uptime, and quality truly matter.

Responsibilities

Design, build, and maintain backend services that power a modern telephony platform with a focus on reliability, scalability, and performance

Develop production-quality code primarily in Go (90%), with some exposure to C or C++ and occasional Python

Spend approximately 50 60% coding, 20% peer code reviews, 15 25% architecture and system design, and 5% miscellaneous work

Build and enhance SIP-based telephony services and real-time communication components

Collaborate closely with engineers on system design, feature delivery, and platform improvements

Participate in code reviews to maintain high standards for code quality, performance, and maintainability

Diagnose and resolve issues across distributed systems, networking, and telephony services

Write clean, well-documented code supported by automated tests at multiple levels

This is a full-time, direct-hire role. Fully remote within the US or Canada. Benefits include 401(k) match and unlimited PTO.

Required Skills

Three or more years of professional software development experience

Strong experience with Go or C or C++, with willingness to work primarily in Go

Experience building networked or distributed systems

Familiarity with real-time or low-latency systems

Experience contributing to production SaaS platforms

Experience participating in code reviews and writing automated tests

Strong communication skills and a collaborative mindset

Ability to start the workday by 10:00 am Eastern Time and travel twice a year to company meetings

Nice to Have

Experience with telephony engines such as Kamailio, RTP Engine, or Asterisk

Knowledge of SIP-related protocols, including SDP, RTP, or RTCP

Cloud Native Development Experience, Google Cloud Platform Preferred

Experience with Kubernetes or containerized microservices

Understanding of audio codecs such as G.711, Opus, or G.729

Exposure to WebRTC and NAT traversal techniques including STUN, TURN, or ICE