Owns end-to-end network troubleshooting for VoIP and real-time communications across customer networks, PBN infrastructure, SIP carriers, and the public internet. Diagnoses SIP signaling, RTP media, NAT, firewall, routing, QoS, latency, jitter, and packet-loss issues using packet captures and network telemetry. Guides customer IT teams, validates VoIP network readiness, supports architecture and monitoring, participates in incident response, and develops diagnostics, runbooks, and automation to improve reliability and reduce MTTR.
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, andnetwork 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 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.
- 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.
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.
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