// Summary
VoIP engineer with deep expertise in architecting, securing, and scaling cloud-based telecom systems. I specialize in SIP signaling, RTP media flows, TLS/SRTP encryption, NAT traversal, SBC logic, SIP trunking, multi-tenant Cloud PBX, and advanced VoIP endpoint provisioning.
Highly skilled in PCAP analysis, Wireshark, tshark, sngrep, Homer/HEP, SIP ladder debugging, and end-to-end network failure investigation — from packet-level tracing to protocol-level troubleshooting across distributed, high-availability environments.
Experienced building and optimizing FreeSWITCH and Asterisk platforms, integrating REST APIs, managing CDRs with database backends, and engineering call flows for enterprise and carrier-grade deployments. Strong foundation in C programming, performance tuning, and telecom automation.
// How I Work
Diagnose at the packet level
I don't guess at call quality issues — I capture, read the SIP ladder, and find the actual root cause before touching config.
Defense in depth
Every system I build has at least two independent layers protecting it — a fast path and a slow path, never a single point of failure.
Document the why, not just the what
Config without reasoning rots. Every repo I ship explains the trade-off behind each decision, not just the syntax.
// Skills
Protocols & Signaling
- SIP / RTP
- TLS / SRTP
- NAT Traversal
- SBC Logic
- SIP Trunking
Platforms
- FreeSWITCH
- Asterisk
- Kamailio
- pjsip
- 2600Hz / KAZOO
- FusionPBX
- Bicom Systems (PBXware / gloCOM)
- Multi-tenant Cloud PBX
Diagnostics
- Wireshark / tshark
- sngrep
- Homer (HEP)
- PCAP analysis
- SIP ladder debugging
Security & Network
- Fortinet / FortiGate
- Cloudflare Tunnel / Zero Trust
- fail2ban
- Fraud/ghost-call detection
Cloud & Infra
- AWS (API Gateway, Lambda, DynamoDB)
- Docker
- Linux administration
- PostgreSQL
Engineering
- C programming
- Python
- REST APIs
- Telecom automation
// Platforms I've Deployed
// Featured Projects
kamailio-sbc-router
Multi-tenant SIP Session Border Controller: NAT traversal via rtpengine, TLS/SRTP termination, carrier failover via dispatcher.
KamailioSBCfreeswitch-cloud-pbx
Multi-tenant FreeSWITCH dialplan, ESL-driven REST automation API, and a real-time CDR pipeline into PostgreSQL.
FreeSWITCHREST APIpjsip-sip-diagnostics
A pjsua2-based SIP diagnostic client: registration latency, call setup time, and live RTP jitter/loss stats.
pjsipDiagnosticsaws-mobile-call-api
Serverless API Gateway + Lambda + DynamoDB backend for a mobile Cloud PBX app's click-to-call and call history.
AWSServerlesstelecom-stack-docker
Containerized telecom stack: Kamailio + FreeSWITCH + PostgreSQL + Homer (HEP) call-flow monitoring.
DockerInfralinux-voip-ops-toolkit
Trunk health checks, rotating SIP packet capture, system health reports, and automated CDR backups.
LinuxOpskazoo-crossbar-client
A Python client for 2600Hz's KAZOO platform Crossbar REST API — accounts, devices, and call origination.
2600Hz / KazooAPI Clientghost-call-defense
Detection and mitigation for nuisance/ghost SIP calls and scanning traffic at the SBC layer.
SecurityKamailiofortinet-voip-firewall
FortiGate config for SIP/RTP: the SIP ALG problem, firewall policy, and voice traffic QoS prioritization.
FortinetNetwork Securitycloudflare-tunnel-voip-access
Secure Cloud PBX admin access via a named Cloudflare Tunnel + Access (Zero Trust) policy — no exposed admin port.
CloudflareZero Trustvoip-phone-provisioning-guides
Auto-provisioning guides + example configs for Yealink, Cisco, Grandstream, Avaya, and Snom desk phones.
ProvisioningMulti-vendorvoip-call-parking-announce
FreeSWITCH call parking with an automatic spoken slot announcement over overhead paging.
FreeSWITCHLuamulticast-paging-system
SIP multicast overhead paging across Yealink, Snom, Grandstream, and Cisco desk phones — one page, every phone, instantly.
MulticastMulti-vendorvoip-phone-unlock-guide
Vendor-lock removal guide for repurposing owned/decommissioned desk phones across major PBX platforms.
ITADRefurbishai-voice-agent
Real-time conversational AI agent that auto-answers FreeSWITCH calls: speech-to-text, LLM, text-to-speech, streamed live.
AIFreeSWITCHteams-sip-bridge
Bridges Microsoft Teams Direct Routing to an on-prem SIP/FreeSWITCH Cloud PBX core.
Microsoft TeamsKamailioyealink-remote-phonebook
Live, server-generated company directory for Yealink desk phones via Remote Phone Book.
YealinkProvisioningsip-pcap-analyzer
Parses SIP pcap captures into a per-call summary: Call-ID, status, setup time, duration.
WiresharkPythonvoip-troubleshooting-playbook
A structured runbook for diagnosing one-way audio, jitter, registration failures, and dropped calls.
RunbookDiagnostics// Experience
VoIP Engineer / Telecommunications Technician
Nov 2021 — PresentCloud Telecom & UCaaS Services Provider · Canada (On-site)
- SIP trunking and multi-tenant Cloud PBX support for enterprise and carrier-grade voice traffic
- FreeSWITCH/Asterisk platform engineering and call-flow design
- Packet-level troubleshooting (Wireshark, sngrep, Homer/HEP) for distributed, high-availability voice infrastructure
IT Support Specialist
Oct 2020 — Apr 2021BECCOS India · New Delhi, India
- C programming and TCP/IP networking fundamentals
Electrical Engineer
Sep 2018 — Dec 2019Energy Efficiency Services Limited · Noida, India
// Certifications
// Education
B.E., Electrical and Electronics Engineering — Sant Longowal Institute of Engineering and Technology (2014 – 2018)