FreeSWITCH Implementation Engineer

Aurexion, Inc.; India; On-site; INR 1,200,000 - 2,400,000; Full time

Job summary

Aurexion, Inc. is seeking a FreeSWITCH Implementation Engineer to design, deploy, and maintain scalable VoIP platforms in production. You will own call routing, dialplans, IVRs, and multi-tenant architectures while ensuring high availability.

Responsibilities include configuring NAT/firewall rules, monitoring call quality, and integrating FreeSWITCH with CRM/ERP systems. Cloud deployments and DevOps collaboration are part of the role.

Qualifications

• Strong hands-on experience with FreeSWITCH.
• Solid understanding of SIP, RTP, WebRTC.
• Experience with ESL and event-driven call control.
• Experience working in production VoIP environments.

Responsibilities

• Install, configure, and maintain FreeSWITCH in production environments from scratch.
• Design and implement call flows, dialplans, IVRs, conferencing, and call routing with ownership.
• Configure SIP profiles, gateways, trunks, and extensions.
• Implement multi-tenant and high-availability FreeSWITCH architectures.
• Troubleshoot SIP signaling, RTP media, and one-way audio issues.
• Integrate FreeSWITCH with CRM, ERP, billing platforms, AI voice bots, WebSocket, and REST services.

Skills

FreeSWITCH expert

VoIP protocols

ESL experience

Tools

fs_cli

sngrep

tcpdump

siptrace

SQL for CDRs

Job description

We are looking for a FreeSWITCH Implementation Engineer who will be responsible for designing, deploying, configuring, integrating, and maintaining FreeSWITCH-based VoIP platforms. The role involves working closely with SIP, RTP, WebRTC, IVR, conferencing, call routing, and external integrations in production environments to deliver scalable and reliable communication solutions.

Responsibilities:

• Install, configure, and maintain FreeSWITCH in production environments from scratch
• Design and implement call flows, dialplans, IVRs, conferencing, and call routing with complete ownership
• Configure SIP profiles, gateways, trunks, and extensions
• Implement multi-tenant and high-availability FreeSWITCH architectures
• Troubleshoot SIP signaling, RTP media, SDP negotiation, call drops, registration failures, and one-way audio issues
• Configure NAT traversal, firewall rules, and RTP port ranges
• Analyze SIP logs and traces using tools such as siptrace, sngrep, and tcpdump
• Implement call queues, time-based routing, skill-based routing, and failover logic
• Integrate FreeSWITCH with CRM, ERP, billing platforms, AI voice bots, WebSocket, and REST services
• Develop real-time call control and monitoring using the Event Socket Library (ESL)
• Configure WebRTC, DTLS-SRTP, ICE, OPUS, mod_verto, and SIP over WSS
• Monitor system performance, call quality, logs, and CDRs
• Optimize codec usage, RTP performance, and resolve production incidents with RCA
• Deploy and manage FreeSWITCH on Linux and cloud platforms
• Configure high availability, load balancing, scaling, backups, and CI/CD automation with DevOps teams

Requirements:

• Strong hands-on experience with FreeSWITCH
• Solid understanding of VoIP protocols including SIP, RTP, RTCP, and WebRTC
• Hands-on experience with Event Socket Library (ESL) and event-driven call control
• Experience using debugging tools such as fs_cli, sngrep, and tcpdump
• Knowledge of SQL, MySQL, or PostgreSQL for CDRs and reporting
• Strong troubleshooting and problem-solving skills
• Experience working in production VoIP environments

Bonus Skills:

• Experience with AWS, Azure, or Google Cloud Platform
• Knowledge of multi-tenant VoIP architectures
• Knowledge of CI/CD pipelines and DevOps practices
• Experience with load balancing and high-availability deployments
• Opportunity to work on enterprise-scale VoIP and communication platforms
• Exposure to modern telephony technologies including WebRTC and FreeSWITCH
• Collaborative engineering environment
• Career growth with cutting-edge communication technologies
• Work on highly scalable and mission-critical production systems