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.
Work Hours:
8:00 PM – 5:00 AM IST
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.