VOIP Developer

Alchemilla Marketing Private Limited; India; Remote; INR 1,283,000 - 1,674,000; Full time

Job summary

Alchemilla Marketing Private Limited in India is seeking a skilled VOIP Developer to design, implement, and maintain SIP-based systems, WebRTC applications, and real-time communication services across our platforms.

You will work on Asterisk, FreeSWITCH, LiveKit, and related technologies, develop backend services in Go or Node.js, and ensure robust, scalable communication infrastructure in a remote-first, collaborative engineering culture.

Qualifications

• Experience with VOIP technologies such as SIP, RTP, WebRTC, Asterisk, FreeSWITCH or LiveKit.

Responsibilities

• Design, develop, and maintain VOIP solutions using Asterisk, FreeSWITCH, LiveKit, WebRTC, and related technologies.
• Build and integrate SIP-based systems, call routing logic, IVR flows, conferencing, recording, and real-time media services.
• Develop backend services and APIs using Go, Node.js, or similar technologies.
• Troubleshoot SIP signaling, RTP/media issues, NAT traversal, registrations, call quality, and gateway integrations.
• Work with Linux/Unix-based systems to deploy, monitor, and maintain communication services.
• Collaborate with product, infrastructure, and support teams to deliver reliable voice and video communication features.
• Document system architecture, call flows, deployment steps, and operational procedures.

Skills

VOIP expertise

SIP/RTP

WebRTC

Go

Node.js

Linux

Tools

Asterisk

FreeSWITCH

LiveKit

Job description

At Alchemilla, we are looking for a skilled and motivated VOIP Developer to help us build and maintain modern communication platforms. In this role, you will work on SIP systems, PBX integrations, WebRTC applications, and real-time communication infrastructure.

Responsibilities:

• Design, develop, and maintain VOIP solutions using Asterisk, FreeSWITCH, LiveKit, WebRTC, and related technologies.
• Build and integrate SIP-based systems, call routing logic, IVR flows, conferencing, recording, and real-time media services.
• Develop backend services and APIs using Go, Node.js, or similar technologies.
• Troubleshoot SIP signaling, RTP/media issues, NAT traversal, registrations, call quality, and gateway integrations.
• Work with Linux/Unix-based systems to deploy, monitor, and maintain communication services.
• Collaborate with product, infrastructure, and support teams to deliver reliable voice and video communication features.
• Document system architecture, call flows, deployment steps, and operational procedures.

Requirements:

• Strong experience with VOIP technologies such as SIP, RTP, WebRTC, Asterisk, FreeSWITCH, or LiveKit.
• Experience with Go, Node.js, or similar backend development technologies.
• Good understanding of networking concepts, NAT, firewalls, TLS, DNS, and TCP/UDP protocols.
• Familiarity with Linux or Unix server environments.
• Ability to debug SIP traces, logs, media paths, and production communication issues.
• Experience with databases, REST APIs, queues, or event-driven systems is a plus.
• Knowledge of cloud deployments, containers, and monitoring tools is preferred.
• Strong problem-solving skills and ability to work independently in a remote environment.

What We Offer:

• Competitive salary of ₹1,15,000 per month.
• Remote-first work environment.
• Opportunity to work on real-world communication infrastructure and modern VOIP platforms.
• A collaborative engineering culture with room for technical ownership.
• Exposure to SIP, WebRTC, PBX systems, and scalable backend services.

Empowering your business with cutting-edge technology solutions.