Senior Network Engineer

Role: Senior Network Engineer – VoIP / Real-Time Communications

Function: Network Engineering / VoIP Infrastructure

Location: Gurugram

Type: Full-time

Industry: AI, Medical SaaS, Dental Medical SaaS

About Company

The company builds AI-powered SaaS for the U.S. dental market.

Its all-in-one platform unifies analytics, scheduling, communications, payments, and marketing automation. Thousands of dental practices across North America rely on it daily.

It was founded in 2015 by a practicing dentist, a data strategist, and a PhD technologist. The team of 88 moves fast, ships real value, and works with high ownership and tight cross-functional collaboration.

Position Overview

This role owns the network connectivity and real-time communications infrastructure that enables the company's VoIP platform to reliably communicate with customer networks, SIP carriers, and the public internet. The engineer will be responsible for troubleshooting end-to-end VoIP connectivity and media issues across the entire network path — from customer phones and LAN infrastructure through switches, routers, firewalls, VLANs, Wi-Fi, VPNs, and ISP connections, to the company's SIP infrastructure and upstream carriers. This is a hands-on production troubleshooting role requiring strong networking fundamentals combined with a deep understanding of SIP, RTP, NAT traversal, and real-time communications. The engineer will work closely with the company's SIP/VoIP engineering team, customer IT teams, carriers, and support/operations teams to diagnose and resolve complex voice-quality, connectivity, registration, signaling, and media-path issues.

Role & Responsibilities

• Own end-to-end network troubleshooting for the company's VoIP traffic between customer environments, the company's SIP infrastructure, and upstream carriers/PSTN networks.
• Diagnose SIP signaling and RTP media issues across customer LANs, firewalls, routers, switches, Wi-Fi networks, VPNs, ISP connections, and internet paths.
• Analyze SIP messages, SDP, RTP flows, packet captures, and network telemetry to identify the root cause of call failures and audio-quality issues.
• Troubleshoot common VoIP networking problems including NAT traversal, SIP ALG, asymmetric routing, blocked ports, firewall policies, RTP path failures, DNS issues, packet loss, jitter, latency, and network instability.
• Work directly with customer IT/network teams to identify problems within their network and provide clear, technically accurate remediation guidance.
• Guide customers through configuration and troubleshooting of switches, routers, firewalls, VLANs, DHCP, Wi-Fi, VPNs, ISP connectivity, QoS, and related network infrastructure required for reliable VoIP service.
• Validate network readiness for the company's VoIP deployments, including connectivity, port accessibility, NAT behavior, SIP signaling, RTP media paths, and network quality.
• Troubleshoot SIP registration, call setup, one-way/no-way audio, dropped calls, delayed audio, call quality degradation, intermittent connectivity, and other real-time communications issues.
• Investigate SIP/RTP traffic between customer endpoints and the company's infrastructure and determine whether failures originate in the customer network, internet path, the company's infrastructure, or the carrier network.
• Work with the SIP engineering team to distinguish network-layer problems from SIP application, routing, registrar, proxy, B2BUA, or media-server issues.
• Troubleshoot and validate SIP connectivity over UDP, TCP, and TLS, as well as media secured using SRTP.
• Work with RTP proxies and understand their role in resolving NAT and media-path challenges.
• Troubleshoot real-time communications involving STUN/TURN and WebRTC networking where applicable.
• Implement and troubleshoot network QoS, DSCP marking, traffic prioritization, and other mechanisms required for reliable voice communications.
• Investigate voice-quality metrics including latency, jitter, packet loss, MOS, and RTP stream behavior.
• Use tools such as tcpdump, Wireshark, SIP trace, and other networking utilities for packet-level troubleshooting.
• Troubleshoot connectivity and routing between the company's SIP infrastructure and upstream SIP carriers/PSTN providers.
• Assist with network architecture and configuration changes required to support SIP servers, RTP/media infrastructure, carrier connectivity, and high-availability VoIP services.
• Define and maintain network monitoring, alerting, diagnostics, and operational tooling for VoIP connectivity and network health.
• Participate in production incident response, escalation, root-cause analysis, and post-incident reviews.
• Develop troubleshooting procedures, network diagnostic tools, runbooks, and documentation that reduce recurring VoIP incidents and improve MTTR.
• Perform capacity and performance analysis of network paths supporting real-time communications and proactively identify potential bottlenecks.
• Collaborate closely with SIP engineers to ensure that network architecture, SIP routing, media handling, and customer connectivity work reliably together.

Must Have Criteria

• 7–10 years of network engineering experience with at least 4–5 years working with VoIP, SIP, or real-time communications networks.
• Strong understanding of enterprise networking fundamentals including TCP/IP, UDP, routing, switching, VLANs, DHCP, DNS, NAT, firewalls, VPNs, and QoS.
• Deep practical understanding of VoIP networking and the behavior of SIP and RTP across real-world networks.
• Strong understanding of SIP signaling, SDP, RTP, SIP registration, call setup, media negotiation, and common SIP failure scenarios.
• Hands-on experience troubleshooting SIP/RTP traffic using packet captures and tools such as Wireshark and tcpdump.
• Strong understanding of NAT traversal, symmetric RTP, SIP ALG, port mapping, firewall behavior, and asymmetric routing.
• Experience troubleshooting voice traffic across customer LANs, switches, routers, firewalls, Wi-Fi networks, VPNs, and ISP connections.
• Ability to analyze an end-to-end VoIP packet flow and determine where signaling or media is being blocked, modified, delayed, or lost.
• Strong Linux networking fundamentals and experience troubleshooting network connectivity from Linux systems.
• Experience working with SIP over UDP, TCP, and TLS, and media using RTP/SRTP.
• Understanding of RTP proxies and their use in real-world SIP architectures.
• Understanding of STUN/TURN and NAT traversal techniques used by real-time communications systems.
• Familiarity with WebRTC networking and SIP over WebSockets.
• Strong understanding of VoIP quality metrics including latency, jitter, packet loss, MOS, and RTP behavior.
• Experience with QoS, DSCP, traffic prioritization, and network optimization for voice traffic.
• Experience troubleshooting connectivity to SIP carriers, PSTN gateways, interconnects, and external VoIP providers.
• Strong analytical and troubleshooting skills, with the ability to work from packet captures, SIP traces, logs, and network telemetry rather than relying only on application-level symptoms.
• Excellent communication skills and the ability to work directly with customer IT/network teams to explain technical issues and guide them through network changes.
• Comfortable working on production incidents and participating in on-call/escalation responsibilities.

Nice to Have

• Experience working with Kamailio, OpenSIPS, FreeSWITCH, Asterisk, SBCs, or similar VoIP infrastructure.
• Experience troubleshooting RTP proxy deployments and complex NAT/media-path scenarios.
• Experience with SIP trunking and carrier interconnects.
• Experience with WebRTC, SIP over WebSockets, STUN/TURN, and browser-based real-time communications.
• Experience with network monitoring and observability platforms such as Prometheus, Grafana, Datadog, or similar tools.
• Experience building automated network diagnostics or VoIP troubleshooting tools.
• Familiarity with BGP, OSPF, advanced routing, or carrier-grade network architectures.
• Experience troubleshooting enterprise firewall platforms and network equipment from vendors such as Cisco, Fortinet, Palo Alto, Juniper, Ubiquiti, or similar.
• Experience with network automation and infrastructure-as-code.
• Experience supporting geographically distributed VoIP or CPaaS platforms.
• Experience working with call centers, unified communications, SIP trunking, or large-scale telecommunications environments.

What We Offer

• End-to-end ownership of a critical, high-visibility infrastructure layer used by thousands of dental practices daily.
• Close collaboration with a focused SIP/VoIP engineering team working on an AI-first communications platform.
• Autonomy, flexible scheduling, and direct impact on measurable outcomes — call quality, uptime, and MTTR.
• Opportunity to build in a large, under-digitized healthcare vertical at a company with strong product-market fit and a clear growth trajectory.