We are seeking a skilled Telephony Engineer with expertise in FreeSWITCH to design, deploy, configure, integrate, and maintain VoIP platforms. This role involves working with SIP, RTP, WebRTC, IVR, conferencing, call routing, and external integrations in a production environment.

Key Responsibilities

FreeSWITCH Core

- Install, configure, and maintain FreeSWITCH in production environments.
- Design and implement call flows, dialplans, IVRs, conferencing, and call routing.
- Configure SIP profiles, gateways, trunks, and extensions.
- Implement multi-tenant and high-availability FreeSWITCH architectures.

SIP, RTP & Networking
- Understand SIP signaling, RTP media, and SDP negotiation.
- Troubleshoot audio issues, call drops, and registration failures.
- Configure NAT traversal, firewall rules, and RTP port ranges.
- Analyze SIP logs and traces (siptrace, sngrep, tcpdump).

Dialplan & Scripting
- Develop dialplans using XML, Lua, JavaScript, or Python.
- Implement logic for call queues, time-based routing, skill-based routing, and failover handling.

Integrations & APIs
- Integrate FreeSWITCH with CRM/ERP systems, billing platforms, AI voice bots (ASR/TTS), and external WebSocket or REST services.
- Utilize Event Socket Library (ESL) for inbound/outbound connections, real-time call control, and monitoring.

WebRTC & Media
- Configure WebRTC support (DTLS-SRTP, ICE, OPUS).
- Work with mod_verto and SIP over WSS.
- Implement browser-based calling solutions.

Monitoring & Debugging
- Monitor system performance and call quality.
- Analyze logs and CDRs.
- Optimize codec usage and RTP performance.
- Handle live production incidents and conduct root cause analysis (RCA).

Deployment & Operations
- Deploy FreeSWITCH on Linux (CentOS/Ubuntu)



and cloud platforms (AWS/GCP/Azure).
- Configure High Availability (HA), load balancing, and scaling.
- Collaborate with DevOps teams for CI/CD, backups, and automation.

Ideal Candidate Profile

Mandatory Requirements:
- Minimum 4 years of experience, with at least 3 years of hands-on experience implementing, configuring, and supporting FreeSWITCH in production.
- Robust experience designing and implementing FreeSWITCH call flows, including dialplans, IVRs, call routing, conferencing, SIP profiles, gateways, trunks, and extensions with end-to-end ownership.
- Hands-on knowledge of VoIP technologies (SIP, RTP, RTCP, SDP, WebRTC) and troubleshooting related issues (call drops, one-way audio, registration failures, NAT traversal).
- Hands-on experience developing FreeSWITCH dialplans using XML and at least one scripting language (Lua, JavaScript, Python), including implementing call queues, time-based routing, skill-based routing, and failover logic.
- Experience integrating FreeSWITCH with CRM/ERP systems, REST APIs, WebSockets, AI Voice Bots (ASR/TTS), or Billing Platforms, along with hands-on experience using Event Socket Library (ESL).
- Strong Linux administration experience (Ubuntu/CentOS) and comfort with debugging tools (fs_cli, sngrep, tcpdump). Experience with SQL (MySQL/PostgreSQL) for CDRs and reporting.
- Ability to join immediately or within 15 days.
- Must be currently based in Delhi NCR.

Preferred Qualifications:
- Experience with AWS, Azure, or GCP, High Availability (HA), Load Balancing, CI/CD pipelines, automation, backups, and scalable FreeSWITCH deployments.
- Experience implementing WebRTC-based calling solutions, including mod_verto, SIP over WSS, DTLS-SRTP, ICE, and OPUS codec configuration.

📌 Telephony Engineer (FreeSWITCH, VoIP) (Haryana)
🏢 TheStaffingCircle
📍 Haryana