Freeswitch Developer

Job Title:

Position Overview

We are seeking an experienced FreeSWITCH and WebRTC Engineer with expertise in building and maintaining multi-tenant VoIP systems, implementing Least Cost Routing (LCR), and integrating with SIP routing engines such as Kamailio or OpenSIPS. This is a hands-on role that requires in-depth knowledge of SIP, RTP, NAT traversal, scalable system design, and VoIP architecture. You will be responsible for WebRTC integration, API development, and creating seamless communication with third-party platforms like CRMs and billing systems.

Key Responsibilities

FreeSWITCH Management

• Deploy, configure, and maintain FreeSWITCH in a multi-tenant environment.
  
• Implement tenant isolation using profiles, dial plans, and custom routing logic.
  
• Ensure high availability, performance, and security of VoIP services.
  
• Manage call routing, media services, and codec configurations.
  

Least Cost Routing (LCR)

• Design and implement dynamic LCR based on carrier rates, call quality, and real-time data.
  
• Automate LCR updates via carrier APIs.
  
• Integrate with FreeSWITCH, Kamailio/OpenSIPS, and backend logic for cost-effective routing.
  

WebRTC Integration

• Integrate WebRTC (voice/video/chat) with FreeSWITCH and manage signaling via Kamailio/OpenSIPS or WebSocket-based systems.
  
• Ensure optimal media quality, security (SRTP/DTLS), and NAT traversal using STUN/TURN/ICE.
  
• Troubleshoot WebRTC-related issues across browsers and devices.
  

API Integration & SDK Development

• Develop and maintain RESTful APIs, WebSocket services, and custom SDKs to integrate FreeSWITCH with:
  
• CRM platforms
  
• Billing engines
  
• Analytics and reporting tools
  
• Contact center software
  
• Build custom API endpoints for external applications to manage users, call flows, billing, and more.
  

SIP Routing with Kamailio/OpenSIPS

• Configure and optimize Kamailio/OpenSIPS for SIP signaling, registration, NAT traversal, load balancing, and failover.
  
• Write and maintain SIP routing logic (dispatcher, dialog, tm modules, etc.).
  
• Integrate SIP proxy with FreeSWITCH for media handling and routing.
  

Multi-Tenant VoIP Architecture

• Design and maintain scalable multi-tenant environments.
  
• Implement tenant-level configurations for call logs, routing, access control, and monitoring.
  
• Build admin portals/tools for provisioning, analytics, and reporting per tenant.
  

Troubleshooting, Monitoring, and Optimization

• Use tools like Wireshark, sngrep, HOMER, and FreeSWITCH logs to analyze and resolve SIP/WebRTC issues.
  
• Monitor system performance, uptime, and call quality (MOS, Jitter, Packet Loss).
  
• Implement logging, alerting, and diagnostics to proactively manage service reliability.
  

Required Skills & Qualifications

Core Experience

• 2–3+ years of hands-on experience with FreeSWITCH in multi-tenant VoIP systems.
  
• Strong expertise in Kamailio or OpenSIPS for SIP proxy and routing.
  
• Proficiency in WebRTC, media relay handling, signaling, and NAT traversal.
  
• Solid experience in implementing Least Cost Routing (LCR).
  
• Expertise in Linux system administration (Ubuntu/CentOS).
  

Technical Proficiency

• Advanced knowledge of SIP, RTP, SRTP, DTLS, ICE, STUN, TURN.
  
• Familiarity with VoIP security practices (SIP authentication, firewalling, DDoS prevention).
  
• Programming/scripting experience in Bash, Lua, Python, or Perl for automation and configuration.
  
• Experience with REST APIs, WebSockets, and third-party web services.
  

Soft Skills

• Strong analytical and debugging skills across the full VoIP stack (signaling, media, API).
  
• Excellent documentation and communication skills.
  
• Ability to collaborate effectively with DevOps, Product, and Customer Support teams.
  

Preferred Qualifications

• Experience with cloud-based VoIP deployments (AWS, GCP, Azure).
  
• Knowledge of call center platforms (ACD, IVR, queue management).
  
• Familiarity with monitoring tools like Grafana, Prometheus, or Zabbix.
  
• Experience with call analytics, billing systems, and voice fraud detection.
  
• Hands-on experience with Docker, Kubernetes, or other containerization solutions.
  
• Contributions to open-source VoIP projects (FreeSWITCH, Kamailio, OpenSIPS).
  

Why Join Us?

• Work on cutting-edge VoIP and real-time communication technologies.
  
• Help build systems that power enterprise-scale multi-tenant telephony solutions.
  
• Collaborate with top talent in the RTC and open-source community.
  
• Competitive salary, flexible work arrangements, and ample career growth opportunities.