Role : 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. The engineer 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.

Location : Kolkata / Gurugram, India

Work Mode : In-office

Employment type : Full-time

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 PBNs 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 to translate business requirements (IVRs, call queues, routing rules, AI agents) into resilient VoIP and backend designs.

Required Qualifications :

- 7 to 10 years of software development experience with at least 4 to 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), and comfort debugging issues at packet-level using tools like 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.

Success Indicators :

- A robust, observable VoIP backbone that can sustain 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.