Telephony Engineer (FreeSWITCH, VoIP )

Strong FreeSWITCH Implementation Engineer Profiles

2

Mandatory (Experience 1) – Must have minimum 4+ years experience, with at least 3+ years of hands-on experience implementing, configuring, and supporting FreeSWITCH in production environments.

3

Mandatory (Experience 2) – Must have strong experience in designing and implementing FreeSWITCH call flows, including dialplans, IVRs, call routing, conferencing, SIP profiles, gateways, trunks, and extensions with end-to-end ownership.

4

Mandatory (Experience 3) – Must have strong hands-on knowledge of VoIP technologies, including SIP, RTP, RTCP, SDP, WebRTC, and should be capable of troubleshooting call drops, one-way audio, registration failures, NAT traversal, and SIP/RTP-related issues.

5

Mandatory (Experience 4) – Must have hands-on experience developing FreeSWITCH dialplans using XML and at least one scripting language such as Lua, JavaScript, or Python, including implementation of call queues, time-based routing, skill-based routing, and failover logic.

6

Mandatory (Experience 5) – Must have 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) for real-time call control.

7

Mandatory (Experience 6) – Must have strong Linux administration experience and should be comfortable deploying and managing FreeSWITCH on Ubuntu/CentOS, along with debugging using fs_cli, sngrep, tcpdump, and working with SQL (MySQL/PostgreSQL) for CDRs and reporting.

8

Mandatory (Notice Period) – Immediate Joiners or candidates who can join within 15 days will be highly preferred.

9

Mandatory (Location) - Mandatory (Location) – Candidate must be currently based in Delhi NCR only

10

Preferred (Experience 1) – Experience with AWS, Azure, or GCP, High Availability (HA), Load Balancing, CI/CD pipelines, automation, backups, and scalable FreeSWITCH deployments.

11

Preferred (Experience 2) – Experience implementing WebRTC-based calling solutions, including mod_verto, SIP over WSS, DTLS-SRTP, ICE, and OPUS codec configuration.