Role: Senior VoIP/SIP Engineer

Function: Engineering

Location: Gurugram, India

Type: Full-time

Industry: SaaS, Healthcare Technology, Telecommunications

About Company

A dental practice management SaaS platform serving 1,500+ practices across North America. Founded by a practicing dentist who experienced operational challenges firsthand.

The platform combines practice management, VoIP telephony, payments, and advanced analytics. The company is evolving into an AI-first product with conversational automation and intelligent communication workflows.

Position Overview

You will own the design and implementation of the core VoIP/SIP telephony backbone that powers communication for thousands of dental practices. You'll architect and build carrier-grade Golang-based SIP services on top of open-source components like Kamailio and FreeSWITCH. This role offers direct impact on a critical platform component serving over 1,500 practices while working closely with the Engineering India Head.

Role & Responsibilities

Design and implement a carrier-grade SIP/VoIP core using components like Kamailio/OpenSIPS for SIP signaling and FreeSWITCH for media and application services
Build Golang-based SIP services including 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 with multi-node SIP proxies, distributed media servers, and 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 including TLS/SRTP, authentication/authorization, rate limiting, fraud detection hooks, and abuse controls
Integrate the telephony backbone with the company's SaaS platform via well-defined internal APIs and webhooks for user accounts, billing, analytics, and AI/automation flows
Define monitoring, alerting, logging, and capacity planning for SIP signaling, RTP/media, and VoIP quality metrics including MOS, jitter, and packet loss
Collaborate with product and operations teams to translate business requirements for IVRs, call queues, routing rules, and AI agents into resilient VoIP and backend designs

Must Have Criteria

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
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 including clustering, health checks, load balancing, and graceful failover
Strong Linux and networking fundamentals including 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 including containerized Kamailio/FreeSWITCH clusters on AWS/GCP, Kubernetes, and service meshes
Prior work building call center or CPaaS-style platforms including programmable IVRs, queues, and analytics

What We Offer

In-office work environment in Gurugram
Opportunity to build telephony infrastructure serving 1,500+ dental practices
Direct reporting to Engineering India Head
Work on AI-first product evolution with conversational automation

Gloroots AI

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