Sr. Network Engineer – VoIP

The Senior Network 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 client’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 client’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 client’s VoIP traffic between customer environments, client’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 client 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 client infrastructure and determine whether failures originate in the customer network, internet path, client 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 client’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

  • Working with Kamailio, OpenSIPS, FreeSWITCH, Asterisk, SBCs, or similar VoIP infrastructure.
  • Troubleshooting RTP proxy deployments and complex NAT/media-path scenarios.
  • SIP trunking and carrier interconnects.
  • 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.
  • Build automated network diagnostics or VoIP troubleshooting tools.
  • Familiarity with BGP, OSPF, advanced routing, or carrier-grade network architectures.
  • Troubleshooting enterprise firewall platforms and network equipment from vendors such as Cisco, Fortinet, Palo Alto, Juniper, Ubiquiti, or similar.
  • Network automation and infrastructure-as-code.
  • Supporting geographically distributed VoIP or CPaaS platforms.
  • Experience working with call centers, unified communications, SIP trunking, or large-scale telecommunications environments.

Work Hours:

8:00 PM – 5:00 AM IST
Job Category: IT
Job Type: Full Time
Job Location: Gurugram Kolkata

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