Senior Software Engineer – Core VoIP / SIP Backbone at Practice By Numbers

Bellevue, Washington, United States -

Full Time

Start Date

Immediate

Expiry Date

03 Aug, 26

Salary

180000.0

Posted On

05 May, 26

Experience

5 year(s) or above

Remote Job

Yes

Telecommute

Yes

Sponsor Visa

Skills

Golang, SIP, VoIP, Kamailio, FreeSWITCH, OpenSIPS, RTP, SDP, Linux, TCP/UDP, Wireshark, PostgreSQL, WebRTC, RESTful APIs, SBC, Network Security

Industry

Software Development

Description

Senior Software Engineer – Core VoIP / SIP Backbone Location: Remote (US) | Open to candidates authorized to work in the United States Work Mode: Remote-first Employment Type: Full-Time Compensation: $150K- 180K base salary, depending on experience + equity + benefits About Practice by Numbers Practice by Numbers (PBN) is a leading dental practice management SaaS platform serving over 1,500 dental practices across North America. Our platform delivers practice management, VoIP telephony, payments, and advanced analytics. We are evolving into an AI-first product, adding conversational automation and intelligent communication workflows for patients and front-office teams. About the Role This role owns the design and implementation of the core VoIP/SIP telephony backbone for PBN, including SIP routing, media handling, scalability, and reliability. You will architect and build Golang-based SIP services and control planes on top of open-source components such as Kamailio (or similar SIP server), FreeSWITCH/OpenSIPS, and related media/RTP proxies. This is a senior individual contributor role with high ownership and direct impact on a product used by thousands of dental practices daily. Key Responsibilities * Design and implement a carrier-grade SIP/VoIP core using components like Kamailio/OpenSIPS for SIP signaling and FreeSWITCH or similar for media and application services. * Build Golang-based SIP services (registrar, SBC-like components, routing logic, monitoring daemons) and internal APIs to control routing, policies, and provisioning. * Configure and operate SIP load balancing, failover, and high-availability setups (multi-node SIP proxies, distributed media servers, RTP proxies). * Implement and maintain dial plans, least-cost routing, DID management, class-4/class-5 style switching logic, and integration with upstream carriers and PSTN gateways. * Own security and robustness of the VoIP stack: TLS/SRTP, authentication/authorization, rate limiting, fraud detection hooks, and abuse controls. * Integrate the telephony backbone with PBN's SaaS platform (user accounts, billing, analytics, AI/automation flows) via well-defined internal APIs and webhooks. * Define monitoring, alerting, logging, and capacity planning for SIP signaling, RTP/media, and VoIP quality (MOS, jitter, packet loss). * Collaborate with product and operations teams to translate business requirements (IVRs, call queues, routing rules, AI agents) into resilient VoIP and backend designs. Required Qualifications * 7–10 years of software development experience with at least 4–5 years building or operating large-scale VoIP/SIP systems. * Strong Golang skills, including building high-performance networked services, concurrent processing, and production-grade APIs. * Hands-on experience with at least one open-source SIP server such as Kamailio/OpenSIPS and one media/application server such as FreeSWITCH or Asterisk, including configuration, routing logic, and troubleshooting. * Deep understanding of SIP, RTP, SDP, NAT traversal, registrar/registrations, B2BUA vs. proxy behavior, and SBC concepts. * Proven ability to design and run highly available telephony backbones: clustering, health checks, load balancing, and graceful failover. * Strong Linux and networking fundamentals (iptables, firewalls, TCP/UDP, QoS), comfortable debugging at packet level using tcpdump/Wireshark. * Experience integrating VoIP platforms with RESTful backends, databases (PostgreSQL or MariaDB/MySQL), and message queues for control and billing workflows. Nice to Have * Experience with WebRTC, SIP over WebSockets, and browser/mobile softphone integrations. * Familiarity with VoIP billing, rating engines, CDR processing, and reseller hierarchies (class-4/class-5 softswitch products or similar). * Cloud-native deployment of VoIP stacks (containerized Kamailio/FreeSWITCH clusters on AWS/GCP, Kubernetes, service meshes). * Prior work building call center or CPaaS-style platforms, including programmable IVRs, queues, and analytics. What Success Looks Like * A robust, observable VoIP backbone that sustains high call volumes with low failure rates and predictable call quality across geographies. * Rapid rollout of new telephony features (IVRs, routing rules, AI agents) through clean APIs and configuration-driven behavior rather than manual changes. * Demonstrated reduction in telephony incidents and MTTR through automation, strong monitoring, and clear runbooks. Benefits & Perks * Competitive base salary (Salary range) + equity * Comprehensive medical, dental, and vision insurance * 401(k) with company match * Flexible PTO and paid holidays * Remote-first culture with team offsites

Responsibilities

Design and implement a carrier-grade VoIP/SIP core using Golang and open-source components like Kamailio and FreeSWITCH. Own the scalability, security, and integration of the telephony backbone with the company's SaaS platform.