Senior Devops Engineer(Product & Platform)

Gnani Innovations Private Limited.; Bengaluru; On-site; INR 1,200,000 - 1,800,000; Full time

Job summary

Gnani Innovations Private Limited is seeking a VoIP/CTI engineer with 3-9 years of experience to design, deploy, and manage carrier-grade SIP/VoIP solutions. You will work across data centres and client sites, configuring SIP trunks, NAT traversal, and media relays while integrating with platforms like Twilio and Genesys.

Responsibilities include building call routes, dial plans, load balancing, and on-site installations; you will collaborate with engineering teams, document architectures, and

Qualifications

• Experience designing and deploying SIP/VoIP solutions.
• Strong knowledge of NAT traversal and media relays.
• Hands-on Linux server administration and automation.

Responsibilities

• Integrate with SIP/VoIP providers and configure SIP trunks.
• Deploy and manage VoIP platforms (Asterisk, Kamailio) and media relays.
• Terminate and manage media flows in data centres for AI-based voice processing.
• Configure networking for call routing and NAT traversal in data centres and client environments.
• Handle Linux server administration, containerization, patching, and automation.
• Diagnose and debug real-time media issues with Wireshark, sngrep, tcpdump.
• Architect call routes, dial plans, load balancing, failover strategies across deployments.
• Collaborate with engineering teams to integrate with Twilio and Genesys.
• Document systems, network designs, call flows, deployment SOPs.
• Travel to client on-site locations for installations and hands-on support.

Skills

Asterisk expertise

Kamailio expertise

RTPengine

SIP/RTP signaling

Linux administration

Scripting (bash, python)

Docker/Kubernetes

Git / CI/CD

Prometheus / Grafana

REST API integration with CPaaS

Tools

Wireshark

sngrep

tcpdump

SIPp

Job description

3 - 9 years experience

Responsibilities

• - Integrate with SIP/VoIP providers: configure SIP trunking, channel provisioning, carrier-grade SIP-level integration
• - 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

• - 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
• - Team player comfortable working in fast‑paced engineering culture

Skills Required

Primary Skills CI/CD Integration with Jenkins, Git, and Azure DevOps Redis Argo CD ELK Stack Basic Shell Scripting Helm Async Task Queues (Celery + Redis) Prometheus CI/CD Jenkins Docker/Kubernetes DevOps, Kubernetes, Infrastructure-as-Code Grafana Microsoft Azure