VOIP Developer

Keka Technologies Private Limited; Ahmedabad District; On-site; INR 1,200,000 - 1,800,000; Full time

Job summary

Elision Technologies Pvt. Ltd. is seeking an experienced VoIP Developer to design, build, and maintain enterprise-grade telecom and unified communications solutions.

You will work with product, QA, DevOps, and support to deliver scalable telecom applications for global customers. The role requires hands-on expertise with FreeSWITCH, Asterisk, OpenSIPS, Kamailio, FusionPBX, SIP/RTP, and Linux; you will implement SIP-based routing, IVR, conferencing, and integrations with CRM and REST APIs while

Qualifications

• Bachelor's degree in Computer Engineering, Information Technology, or a related field.
• 5+ years of experience in VoIP or Telecom Software Development.
• Strong programming skills in C, C++, Python, PHP, or Java.
• Hands-on experience with Linux environments and networking concepts.
• Good understanding of REST APIs, SQL databases, and system integration.

Responsibilities

• Design, develop, and maintain VoIP and telecom applications.
• Configure, customize, and optimize FreeSWITCH, Asterisk, OpenSIPS, Kamailio, and FusionPBX.
• Develop SIP-based call routing, IVR, conferencing, recording, and telephony solutions.
• Integrate VoIP platforms with CRM systems, REST APIs, and third-party applications.
• Troubleshoot SIP, RTP, codec, NAT, and signalling issues.
• Optimize system performance, scalability, and high availability.
• Perform debugging, testing, and code reviews.
• Collaborate with cross-functional teams to deliver new features and product enhancements.
• Prepare technical documentation and follow development best practices.

Skills

FreeSWITCH

OpenSIPS

Asterisk

Kamailio

FusionPBX

C

C++

Python

PHP

Java

Linux

REST APIs

SIP

RTP

WebRTC

Education

Bachelor's degree in Computer Engineering, Information Technology, or a related field

Tools

Docker

Kubernetes

Job description

About the Role

Elision Technologies Pvt. Ltd. is looking for an experienced VoIP Developer to develop, customize, and maintain enterprise-grade telecom and unified communication solutions. The ideal candidate should have strong expertise in FreeSWITCH, OpenSIPS, Asterisk, Kamailio, and FusionPBX, along with a solid understanding of SIP-based communication systems and VoIP technologies.

You will work closely with Product, QA, DevOps, and Support teams to build scalable, secure, and high-performance telecom applications for global customers.

Key Responsibilities

• Design, develop, and maintain VoIP and telecom applications.
• Configure, customize, and optimize FreeSWITCH, Asterisk, OpenSIPS, Kamailio, and FusionPBX.
• Develop SIP-based call routing, IVR, conferencing, recording, and telephony solutions.
• Integrate VoIP platforms with CRM systems, REST APIs, and third-party applications.
• Troubleshoot SIP, RTP, codec, NAT, and signalling issues.
• Optimize system performance, scalability, and high availability.
• Perform debugging, testing, and code reviews.
• Collaborate with cross-functional teams to deliver new features and product enhancements.
• Prepare technical documentation and follow development best practices.

Required Qualifications

• Bachelor's degree in Computer Engineering, Information Technology, or a related field.
• 5+ years of experience in VoIP or Telecom Software Development.
• Strong programming skills in C, C++, Python, PHP, or Java.
• Hands-on experience with Linux environments and networking concepts.
• Good understanding of REST APIs, SQL databases, and system integration.

Required Technical Skills

• FreeSWITCH
• OpenSIPS
• Asterisk
• SIP & RTP Protocols
• VoIP & IP Telephony
• WebRTC
• REST APIs
• TCP/IP & Networking

Preferred Skills

• Experience with Contact Center Solutions, Predictive Dialers, and Auto Dialers.
• Knowledge of Cloud Telephony, Unified Communications (UC), and IP PBX.
• Experience with Docker, Kubernetes, or cloud platforms.
• Familiarity with AI-powered Voice Bots and speech technologies is an added advantage.

Key Skills

• FreeSWITCH
• OpenSIPS
• Asterisk

Why Join Elision Technologies?

• Work on innovative VoIP, Contact Center, and AI-powered communication products.
• Opportunity to build scalable telecom solutions for global enterprises.
• Continuous learning and career growth opportunities.
• Competitive salary and performance-based incentives.

Required Skills

WebRTC IP & RTP Protocols OPENSIP Freeswitch Kamailio FusionPBX REST APIs & Linux VoIP & Asterisk