Senior Software Engineer – Core VoIP / SIP Backbone (Gurugram)

Senior Software Engineer – Core VoIP / SIP Backbone (Gurugram)

17 Aug
|
Practice by Numbers
|
Gurugram

17 Aug

Practice by Numbers

Gurugram

Senior Software Engineer – Core VoIP / SIP BackbonePractice by Numbers (PBN) | Gurugram / Kolkata, India

About Practice by NumbersPractice by Numbers is a leading dental practice management SaaS platform serving over1,500 dental practices across North America, providing practice management, VOIP telephony,payments, and advanced analytics. The platform is evolving into an AI‐first product, addingconversational automation and intelligent communication workflows for patients and front-officeteams.

About the RoleThis role owns the design and implementation of the core VoIP/SIP telephony backbone forPBN, including SIP routing, media handling, scalability, and reliability. The engineer will architectand build Golang‐based SIP services and control planes on top of open‐source componentssuch as Kamailio (or similar SIP server), FreeSWITCH/OpenSIPS, and related media/RTPproxies.

Location: Gurugram / Kolkata, IndiaWork Mode: In-officeEmployment type: Full-time

Key Responsibilities- Design and implement a carrier‐grade SIP/VoIP core using components likeKamailio/OpenSIPS for SIP signaling and FreeSWITCH or similar for media andapplication 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‐5style 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 to translate business requirements (IVRs, callqueues, 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 oroperating 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 asKamailio/OpenSIPS and one media/application server such as FreeSWITCH (orAsterisk), including configuration, routing logic, and troubleshooting.





- Deep understanding of SIP, RTP, SDP, NAT traversal, registrar/registrations, B2BUA vsproxy behavior, and SBC concepts.

- Proven ability to design and run highly available telephony backbones: clustering, healthchecks, load balancing, and graceful failover.

- Solid Linux and networking fundamentals (iptables, firewalls, TCP/UDP, QoS), andcomfort debugging issues at packet‐level using tools like tcpdump/wireshark.

- Experience integrating VoIP platforms with RESTful backends, databases (PostgreSQLor MariaDB/MySQL), and message queues for control and billing workflows.

Nice to Have- Experience with WebRTC, SIP over WebSockets, and browser/mobile softphoneintegrations.- 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 clusterson 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 failurerates and predictable call quality across geographies.- Rapid rollout of new telephony features (IVRs, routing rules, AI agents) through cleanAPIs and configuration‐driven behavior rather than manual changes.- Demonstrated reduction in telephony incidents and MTTR through automation, strongmonitoring, and clear runbooks.

📌 Senior Software Engineer – Core VoIP / SIP Backbone (Gurugram)
🏢 Practice by Numbers
📍 Gurugram

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