{"id":2527395,"url":"https://alion.io/job/practice-numbers-senior-network-engineer","title":"Senior Network Engineer","company":{"id":2763798,"name":"Practice Numbers","domain":"practicenumbers.com","url":"https://alion.io/company/practicenumbers","size_band":null,"is_staffing_agency":false,"employer_type":"direct","is_intermediary":false,"listed_via":null,"ats_vendor":"Gem","truth_index":null},"role":"Networking","role_family":"Networking","seniority":"senior","employment_type":"full_time","work_mode":"on_site","remote_scope":null,"remote_scope_basis":null,"remote_working_hours":null,"hiring_geo_confidence":"structured","locations":["Gurgaon, India"],"countries":["IN"],"hiring_countries":[],"hiring_countries_total":0,"salary":null,"salary_estimate":{"min_usd":17000,"max_usd":41000,"period":"year","method":"global_role_cell_scaled_by_country","sample_n":540},"experience_years_min":7,"visa_sponsorship":false,"relocation_package":false,"has_equity":false,"technologies":[{"name":"DHCP","optional":false},{"name":"DNS","optional":false},{"name":"Linux","optional":false},{"name":"TCP/IP","optional":false},{"name":"Tcpdump","optional":false},{"name":"WebRTC","optional":false},{"name":"WebSockets","optional":false},{"name":"Wi-Fi","optional":false},{"name":"Wireshark","optional":false},{"name":"BGP","optional":true},{"name":"Datadog","optional":true},{"name":"Grafana","optional":true},{"name":"OSPF","optional":true},{"name":"Prometheus","optional":true}],"status":"live","first_seen_at":"2026-09-02T14:18:12Z","employer_posted_date":"2026-09-22","last_verified_at":"2026-10-14T16:02:32Z","board_verified":true,"closed_at":null,"days_open":42,"trust":{"level":"ok","repost_count":null,"flags":[],"days_open":42},"description":"Senior Network Engineer - VoIP / Real-Time Communications\nPractice by Numbers (PBN) | Gurugram / Kolkata, India\nAbout Practice by Numbers\nPractice 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.\nAbout the Role\nThis 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.\nThe 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.\nThis 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.\nThe engineer will work closely with PBN's SIP/VoIP engineering team, customer IT teams, carriers, and\nsupport/operations teams to diagnose and resolve complex voice-quality, connectivity, registration,\nsignaling, and media-path issues.\nWork Hours:\n8:00 PM - 5:00 AM IST\nKey Responsibilities\nOwn end-to-end network troubleshooting for PBN's VoIP traffic between customer environments, PBN's SIP infrastructure, and upstream carriers/PSTN networks.\nDiagnose SIP signaling and RTP media issues across customer LANs, firewalls, routers, switches, Wi-Fi networks, VPNs, ISP connections, and internet paths.\nAnalyze SIP messages, SDP, RTP flows, packet captures, and network telemetry to identify the root cause of call failures and audio-quality issues.\nTroubleshoot 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.\nWork directly with customer IT/network teams to identify problems within their network and provide clear, technically accurate remediation guidance.\nGuide 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.\nValidate network readiness for PBN VoIP deployments, including connectivity, port accessibility, NAT behavior, SIP signaling, RTP media paths, and network quality.\nTroubleshoot SIP registration, call setup, one-way/no-way audio, dropped calls, delayed audio, call quality degradation, intermittent connectivity, and other real-time communications issues.\nInvestigate 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.\nWork with the SIP engineering team to distinguish network-layer problems from SIP application, routing, registrar, proxy, B2BUA, or media-server issues.\nTroubleshoot and validate SIP connectivity over UDP, TCP, and TLS, as well as media secured using SRTP.\nWork with RTP proxies and understand their role in resolving NAT and media-path challenges.\nTroubleshoot real-time communications involving STUN/TURN and WebRTC networking where applicable.\nImplement and troubleshoot network QoS, DSCP marking, traffic prioritization, and other mechanisms required for reliable voice communications.\nInvestigate voice-quality metrics including latency, jitter, packet loss, retransmissions, MOS, and RTP stream behavior.\nUse tools such as tcpdump, Wireshark, SIP tracing, network monitoring systems, and Linux networking utilities for packet-level troubleshooting.\nTroubleshoot connectivity and routing between PBN's SIP infrastructure and upstream SIP carriers/\nPSTN providers.\nAssist with network architecture and configuration changes required to support SIP servers, RTP/ media infrastructure, carrier connectivity, and high-availability VoIP services.\nDefine and maintain network monitoring, alerting, diagnostics, and operational tooling for VoIP connectivity and network health.\nParticipate in production incident response, escalation, root-cause analysis, and post-incident reviews.\nDevelop troubleshooting procedures, network diagnostic tools, runbooks, and documentation that reduce recurring VoIP incidents and improve MTTR.\nPerform capacity and performance analysis of network paths supporting real-time communications and proactively identify potential bottlenecks.\nCollaborate closely with SIP engineers to ensure that network architecture, SIP routing, media handling, and customer connectivity work reliably together.\nRequired Qualifications\n7-10 years of network engineering experience with at least 4-5 years working with VoIP, SIP, or real-time communications networks.\nStrong understanding of enterprise networking fundamentals including TCP/IP, UDP, routing, switching, VLANs, DHCP, DNS, NAT, firewalls, VPNs, and QoS.\nDeep practical understanding of VoIP networking and the behavior of SIP and RTP across real-world networks.\nStrong understanding of SIP signaling, SDP, RTP, SIP registration, call setup, media negotiation, and common SIP failure scenarios.\nHands-on experience troubleshooting SIP/RTP traffic using packet captures and tools such as Wireshark and tcpdump.\nStrong understanding of NAT traversal, symmetric RTP, SIP ALG, port mapping, firewall behavior, and asymmetric routing.\nExperience troubleshooting voice traffic across customer LANs, switches, routers, firewalls, Wi-Fi networks, VPNs, and ISP connections.\nAbility to analyze an end-to-end VoIP packet flow and determine where signaling or media is being blocked, modified, delayed, or lost.\nStrong Linux networking fundamentals and experience troubleshooting network connectivity from Linux systems.\nExperience working with SIP over UDP, TCP, and TLS and media using RTP/SRTP.\nUnderstanding of RTP proxies and their use in real-world SIP architectures.\nUnderstanding of STUN/TURN and NAT traversal techniques used by real-time communications systems.\nFamiliarity with WebRTC networking and SIP over WebSockets.\nStrong understanding of VoIP quality metrics including latency, jitter, packet loss, MOS, and RTP behavior.\nExperience with QoS, DSCP, traffic prioritization, and network performance optimization for voice traffic.\nExperience troubleshooting connectivity to SIP carriers, PSTN gateways, and external VoIP providers.\nStrong 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.\nExcellent communication skills and the ability to work directly with customer IT/network teams to explain technical issues and guide them through network changes.\nComfortable working on production incidents and participating in on-call/escalation responsibilities.\nNice to Have\nExperience working with Kamailio, OpenSIPS, FreeSWITCH, Asterisk, SBCs, or similar VoIP infrastructure.\nExperience troubleshooting RTP proxy deployments and complex NAT/media-path scenarios.\nExperience with SIP trunking and carrier interconnects.\nExperience with WebRTC, SIP over WebSockets, STUN/TURN, and browser-based real-time communications.\nExperience with network monitoring and observability platforms such as Prometheus, Grafana, Datadog, or similar tools.\nExperience building automated network diagnostics or VoIP troubleshooting tools.\nFamiliarity with BGP, OSPF, advanced routing, or carrier-grade network architectures.\nExperience troubleshooting enterprise firewall platforms and network equipment from vendors such as Cisco, Fortinet, Palo Alto, Juniper, Ubiquiti, or similar.\nExperience with network automation and infrastructure-as-code.\nExperience supporting geographically distributed VoIP or CPaaS platforms.\nExperience working with call centers, unified communications, SIP trunking, or large-scale telecommunications environments.\nSuccess Indicators\nVoIP 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.\nSignificant reduction in MTTR for network-related VoIP incidents through strong diagnostics, monitoring, automation, and well-defined runbooks.\nCustomer IT teams receive clear and actionable guidance for resolving issues involving firewalls, routers, switches, VLANs, Wi-Fi, VPNs, QoS, and ISP connectivity.\nReliable SIP signaling and RTP media connectivity across customer environments and PBN's VoIP infrastructure.\nNetwork-related call failures, one-way/no-way audio, dropped calls, registration issues, and voice-quality problems are proactively identified and reduced.\nStrong packet-level observability exists across critical VoIP network paths, enabling engineers to quickly identify signaling and media failures.\nNetwork architecture and operational practices support predictable latency, jitter, packet loss, and overall voice quality across geographies.\nClose collaboration between Network Engineering and SIP Engineering results in clear ownership boundaries and faster resolution of complex telephony incidents.\nNew customer VoIP deployments can be validated and brought online with repeatable network-readiness checks rather than manual trial-and-error 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