Posted on: 29/05/2026
Description :
Location & Travel Requirements :
- Based in Bengaluru, working from company office
- Travel to client sites for on- premise integrations and deployments
Company Overview :
We power GenAI voice bots, real-time agent assist systems, and telephony integrations for enterprises at scale.
Our infrastructure integrates with multiple SIP/VoIP providers and processes live call media through our proprietary ASR, TTS, and LLM engines.
We operate in hybrid environments cloud, on-premise, and private data centers and offer end-to-end solutions for outbound calling, call center automation, and voice analytics.
We also engage with CPaaS platforms like Twilio and Genesys to support a wide range of customer needs across telecom and enterprise domains.
Experience :
- Deploy and manage VoIP platforms (Asterisk, Kamailio) and media relays (RTPengine); FreeSWITCH experience is a plus
- Terminate and manage media flows in company data centres, supporting AI-based voice processing
- Configure networking for call routing and NAT traversal in data centre and client environments
- Handle Linux server administration, containerization, patching, and automation
- Diagnose and debug real-time media issues using tools like Wireshark, sngrep, tcpdump
- Architect call routes, dial plans, load balancing, failover strategies across deployments
- Collaborate with engineering teams to integrate with platforms like Twilio and Genesys
- Document systems, network designs, call flows, deployment SOPs
- Travel to client on- site locations to perform installations, network setups, and hands-on support
Required Technical Skills :
- 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 :
- 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 :
- 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
- Team player comfortable working in fast- paced engineering culture
What Gnani Offers :
- Hands?on exposure to diverse SIP integrations and cloud/edge voice deployments
- Professional growth in a high-tech, rapidly scaling organization
- Occasional travel to client sites, providing exposure to field deploy
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