Senior Network Engineer in Gurugram, Haryāna at Practice By Numbers

Senior Network Engineer – VoIP / Real-Time

Communications

Practice by Numbers (PBN) | Gurugram / Kolkata, India

About Practice by Numbers

Practice by Numbers is a leading dental practice management SaaS platform serving over 1,500 dental

practices across North America, providing practice management, VoIP telephony, payments, and advanced

analytics. The platform is evolving into an AI-first product, adding conversational automation and intelligent

communication workflows for patients and front-office teams.

About the Role

This role owns the network connectivity and real-time communications infrastructure that enables PBN'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 PBN'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 should

be able to analyze SIP/RTP traffic at the packet level, determine where a signaling or media failure is

occurring, and clearly guide a customer's IT/network team toward the appropriate resolution.

The engineer will work closely with PBN'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.

Key Responsibilities

Own end-to-end network troubleshooting for PBN's VoIP traffic between customer environments,

PBN'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.

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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 PBN 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 PBN infrastructure and determine

whether failures originate in the customer network, internet path, PBN infrastructure, or 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, retransmissions, MOS, and RTP

stream behavior.

Use tools such as tcpdump, Wireshark, SIP tracing, network monitoring systems, and Linux

networking utilities for packet-level troubleshooting.

Troubleshoot connectivity and routing between PBN'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.

Required Qualifications

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.

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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 performance optimization for voice

traffic.

Experience troubleshooting connectivity to SIP carriers, PSTN gateways, 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.

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Success Indicators

VoIP connectivity and media issues are diagnosed quickly and accurately, with clear identification of

whether the root cause is within the customer network, internet path, PBN infrastructure, or carrier.

Significant reduction in MTTR for network-related VoIP incidents through strong diagnostics,

monitoring, automation, and well-defined runbooks.

Customer IT teams receive clear and actionable guidance for resolving issues involving firewalls,

routers, switches, VLANs, Wi-Fi, VPNs, QoS, and ISP connectivity.

Reliable SIP signaling and RTP media connectivity across customer environments and PBN's VoIP

infrastructure.

Network-related call failures, one-way/no-way audio, dropped calls, registration issues, and voice-

quality problems are proactively identified and reduced.

Strong packet-level observability exists across critical VoIP network paths, enabling engineers to

quickly identify signaling and media failures.

Network architecture and operational practices support predictable latency, jitter, packet loss, and

overall voice quality across geographies.

Close collaboration between Network Engineering and SIP Engineering results in clear ownership

boundaries and faster resolution of complex telephony incidents.

New customer VoIP deployments can be validated and brought online with repeatable network-

readiness checks rather than manual trial-and-error troubleshooting.