FreeSWITCH Implementation Engineer
Gurugram • Full-time
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.
Job Details & Requirements
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
- Develop dialplans using XML, Lua, JavaScript, or Python
- 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
- Implement browser-based calling solutions
- 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
- Strong Linux system administration skills
- Experience developing dialplans using XML, Lua, JavaScript, or Python
- 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
- Experience integrating AI voice bots using ASR/TTS
- Knowledge of CI/CD pipelines and DevOps practices
- Experience with load balancing and high-availability deployments
Benefits
- 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