Senior Software Engineer – Core VoIP / SIP Backbone

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.

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.