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
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