Telephony Developer (Go or C++)
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, GCP 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