MetroLeads – Network & VoIP Engineer (Open-Source SIP Stack)
Role Description
MetroGuild is seeking a Network Engineer with VoIP exposure who is excited to work on modern, open-source telephony technologies. You will play a central role in helping us manage and evolve our SIP-based infrastructure using platforms like FreeSWITCH, Kamailio, and related systems — with mentoring and guidance provided as needed.
This is a hybrid network + telephony + cloud infrastructure role. Prior deep experience in FreeSWITCH or Kamailio is not expected — what we need is strong networking fundamentals, solid Linux experience, familiarity with cloud platforms, and a proactive, learning mindset.
You’ll be joining a distributed team that builds and supports voice solutions for our customers across the US, EU, and India. Our stack includes Python, ElasticSearch, ReactJS, React Native, Cassandra, AWS, and modern open-source VoIP components.
What You’ll Work On
Our platform enables:
- Inbound & outbound SIP/WebRTC calling
- Campaign management (SMS, email, voice)
- Multi-channel lead capturing
- Automated lead nurturing workflows
- Team and agent dashboards
- Role-based controls and secure communication
Responsibilities
- Configure and maintain SIP-based VoIP infrastructure using FreeSWITCH, Kamailio, and RTP relays
- Monitor and troubleshoot SIP signaling, RTP flows, NAT traversal, and WebRTC performance
- Work closely with React Native and backend developers integrating SIP.js and WebRTC flows
- Integrate and maintain SIP trunks (Twilio, VoIP.ms, Airtel, BSNL, etc.)
- Manage internal office LAN/WAN, DHCP/DNS, and network security across teams
- Administer AWS resources: Route 53 (DNS), S3 (object storage), EC2 instances
- Ensure high uptime, secure networking, and proper infrastructure documentation
- Extend call routing logic and debug live systems under production load.
Minimum Qualifications
- 3–5 years of experience in Networking, DevOps, or Infrastructure Engineering
- Strong understanding of networking protocols: TCP/IP, NAT, DNS, SIP, RTP
- Hands-on experience managing AWS services, especially Route 53, S3, and EC2
- Prior experience managing or configuring internal corporate networks and routers
- Linux system administration and troubleshooting proficiency
- Familiar with one scripting/backend language (Python, Java, or Node.js)
- Basic exposure to VoIP/SIP concepts and desire to grow into the domain
- Good communication and collaboration skills
- Bachelor’s degree in computer science, IT, or a related field
Nice to Have (Optional but Valued)
- Exposure to FreeSWITCH, Kamailio, or OpenSIPS
- Familiarity with WebRTC, SIP.js, and SIP over WebSocket
- Experience working with trunk providers (Twilio, Airtel, BSNL, TATA)
- Experience with CI/CD, monitoring tools, and Docker-based deployment
Work Location
This is an on-site role (Pune) as the telephony infrastructure and internal networking are managed locally. Occasional client site visits may be required based on project needs.