About the Company

The company is focused on building and managing advanced VoIP and telephony platforms, delivering scalable communication solutions for businesses. With a strong emphasis on real-time communication systems, the organization leverages modern cloud infrastructure and open-source technologies like FreeSWITCH to power high-performance voice applications.

Introduction

We are hiring a Telephony Engineer (FreeSWITCH) with strong expertise in VoIP systems and real-time communication platforms. This role involves designing, deploying, and maintaining FreeSWITCH-based solutions in production environments. It is a remote opportunity offering competitive compensation and the chance to work on large-scale, high-availability telephony systems.

Key Responsibilities

FreeSWITCH Implementation

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

VoIP & Networking

Manage SIP signaling, RTP media flow, and SDP configurations.
Troubleshoot call issues such as one-way audio, call drops, and registration failures.
Handle NAT traversal, firewall setup, and RTP port configurations.
Analyze SIP traces using tools like sngrep and tcpdump.

Dialplan & Scripting

Develop dialplans using XML, Lua, JavaScript, or Python.
Implement call queues, routing logic, and failover mechanisms.
Design time-based and skill-based call routing systems.

Integrations & Real-Time Systems

Integrate telephony systems with CRM/ERP platforms, billing systems, and AI voice bots.
Work with REST APIs and WebSocket services.
Use Event Socket Library (ESL) for real-time call control and automation.

WebRTC & Media Handling

Configure WebRTC components including DTLS-SRTP, ICE, and OPUS codecs.
Work with mod_verto and SIP over WSS.
Build browser-based calling solutions.

Monitoring & Deployment

Monitor system performance, call quality, and logs.
Analyze CDRs and optimize codecs for performance.
Deploy systems on Linux (CentOS/Ubuntu) and cloud platforms like AWS, GCP, or Azure.
Implement high availability, load balancing, and scaling strategies.
Collaborate with DevOps teams for CI/CD and automation processes.

Required Qualifications

5+ years of experience in telephony or VoIP engineering.
Strong hands-on experience with FreeSWITCH development and deployment.
Deep understanding of SIP, RTP, RTCP, and WebRTC protocols.
Proficiency in Linux system administration.
Experience with dialplan development using XML, Lua, JavaScript, or Python.
Knowledge of ESL and event-driven architectures.
Experience with debugging tools such as fs_cli, sngrep, and tcpdump.
Working knowledge of SQL databases (MySQL/PostgreSQL) for reporting and CDR analysis.

Preferred Qualifications

Experience designing high-availability and multi-tenant telephony systems.
Familiarity with cloud platforms (AWS, GCP, Azure).
Experience integrating AI-based voice systems or bots.
Knowledge of CI/CD pipelines and automation workflows.

Why Join Us / Benefits

Competitive salary package up to ₹25 LPA.
Fully remote work environment.
Opportunity to work on large-scale, real-time communication systems.
Exposure to modern VoIP technologies and cloud-based architectures.
High-impact role with ownership and technical growth opportunities.