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Required Technical Skills
- Strong expertise with Asterisk: dial plans, ARI, AMI, voice codecs, IVR logic
- Proficient in Kamailio: NAT handling, registrar, dispatcher and SIP proxy logic
- Experience configuring RTPengine for media relay and NAT traversal
- Deep understanding of SIP protocol, SDP negotiation, RTP, call flows and signaling
- Advanced grid of computer networking: NAT, STUN/TURN, routing, subnetting, packet filtering, firewalls, load balancing
- Experience integrating with SIP providers (DID/toll‑free channels through Twilio/Genesys or others)
- Strong Linux administration: Debian/Ubuntu/CentOS/RHEL; scripting (bash, python) for automation; monitoring
- Hands-on debugging experience with SIP/RTP diagnostics and trace tools
- Experience with SIP load testing using tools like SIPp
- Creation and management of observability dashboards using Prometheus and Grafana
Preferred/Bonus Skills
- Experience with FreeSWITCH deployments
- Docker and containerized deployment experience to manage VoIP stacks and services
- Exposure to cloud platforms (AWS, Azure, or GCP) for hybrid or scalable VoIP deployments
- Understanding of VoIP security (TLS, SRTP), SBC configuration, and QoS implementation
- Experience with SBCs and secure SIP peering
- Familiarity with traffic shaping, jitter buffering and SIP analytics platforms like Homer/SIPCAPTURE
- REST API integration for Twilio, Genesys, or other CPaaS systems
- Basic knowledge of databases like PostgreSQL or MongoDB
- Version control with Git and basic CI/CD automation tools
- Experience with Dialer or Call Center software such as Cisco or Genesys
- Experience configuring agent desktops over CTI integrations
Soft Skills & Communication
- Excellent verbal and written communication skills for interacting with clients and cross-functional teams
- Proactive, detail‑oriented mindset with strong problem-solving and analytical skills
- Ability to manage client expectations, lead on‑site technical discussions, and produce clear documentation
