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Job Description
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Key Responsibilities
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- Integrate with SIP/VoIP providers: configure SIP trunking, channel provisioning, carrier-grade SIP-level integration
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- Deploy and manage VoIP platforms (Asterisk, Kamailio) and media relays (RTPengine); FreeSWITCH experience is a plus
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- Terminate and manage media flows in company data centres, supporting AI-based voice processing
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- Configure networking for call routing and NAT traversal in data centre and client environments
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- Handle Linux server administration, containerization, patching, and automation
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- Diagnose and debug real-time media issues using tools like Wireshark, sngrep, tcpdump
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- Architect call routes, dial plans, load balancing, failover strategies across deployments
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- Collaborate with engineering teams to integrate with platforms like Twilio and Genesys
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- Document systems, network designs, call flows, deployment SOPs
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- Travel to client on‑site locations to perform installations, network setups, and hands-on support
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• Required Technical Skills
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- Strong expertise with Asterisk: dial plans, ARI, AMI, voice codecs, IVR logic
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- Proficient in Kamailio: NAT handling, registrar, dispatcher and SIP proxy logic
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- Experience configuring RTPengine for media relay and NAT traversal
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- Deep understanding of SIP protocol, SDP negotiation, RTP, call flows and signaling
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- Advanced grid of computer networking: NAT, STUN/TURN, routing, subnetting, packet filtering, firewalls, load balancing
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- Experience integrating with SIP providers (DID/toll‑free channels through Twilio/Genesys or others)
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- Strong Linux administration: Debian/Ubuntu/CentOS/RHEL; scripting (bash, python) for automation; monitoring
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- Hands-on debugging experience with SIP/RTP diagnostics and trace tools
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- Experience with SIP load testing using tools like SIPp
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- Creation and management of observability dashboards using Prometheus and Grafana
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• Preferred/Bonus Skills
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- Experience with FreeSWITCH deployments
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- Docker and containerized deployment experience to manage VoIP stacks and services
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- Exposure to cloud platforms (AWS, Azure, or GCP) for hybrid or scalable VoIP deployments
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- Understanding of VoIP security (TLS, SRTP), SBC configuration, and QoS implementation
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- Experience with SBCs and secure SIP peering
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- Familiarity with traffic shaping, jitter buffering and SIP analytics platforms like Homer/SIPCAPTURE
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- REST API integration for Twilio, Genesys, or other CPaaS systems
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- Basic knowledge of databases like PostgreSQL or MongoDB
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- Version control with Git and basic CI/CD automation tools
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- Experience with Dialer or Call Center software such as Cisco or Genesys
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- Experience configuring agent desktops over CTI integrations
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• Soft Skills & Communication
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- Excellent verbal and written communication skills for interacting with clients and cross-functional teams
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- Proactive, detail‑oriented mindset with strong problem-solving and analytical skills
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- Ability to manage client expectations, lead on‑site technical discussions, and produce transparent documentation
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- Team player comfortable working in fast‑paced engineering culture

📌 DevOps Engineer (Mumbai)
🏢 gnani.ai
📍 Mumbai