FreeSWITCH and WebRTC Engineer

Hyderabad, Telangana | 2026-04-23

Experience : 5.00 - 10.00 years

Shift Type : 6:30 PM to 3:30AM

Qualification : BTech or MCA

Location : Hyderabad, Telangana

Mode of Operation : On-site

Number of Openings : 2

Job Description

Job Overview

We are seeking a highly skilled FreeSWITCH and WebRTC Engineer with expertise in multi-tenant architectures, Least Cost Routing (LCR), API integrations with third-party applications, and SIP routing platforms such as Kamailio or OpenSIPS. The ideal candidate will be responsible for configuring, managing, and optimizing FreeSWITCH in a multi-tenant environment, ensuring cost-effective routing and integrating with external systems via APIs. Experience with Kamailio or OpenSIPS is essential for managing SIP signaling and routing, ensuring scalable and reliable VoIP communications.

Key Responsibilities

Roles & Responsibilities :

• FreeSWITCH Management:
• Install, configure, and maintain FreeSWITCH servers to manage multi-tenant VoIP systems, ensuring isolation and scalability for each tenant.
• Implement and configure Least Cost Routing (LCR) strategies to ensure cost-effective call routing based on time, destination, and other criteria.
• Work with Kamailio or OpenSIPS to handle SIP signaling, ensuring that calls are efficiently routed and handled across multiple tenants.
• Troubleshoot and resolve issues related to FreeSWITCH, SIP signaling, and call routing in a multi-tenant environment.
• WebRTC Integration:
• Implement and maintain WebRTC solutions for real-time communication (voice, video, and chat) within a multi-tenant architecture.
• Integrate WebRTC seamlessly with FreeSWITCH for high-quality, low-latency communication.
• Ensure efficient WebRTC signaling, media handling, and security, optimizing for performance in a multi-tenant setup.
• Multi-Tenant Architecture:
• Design and manage multi-tenant VoIP systems that ensure each tenant has isolated resources, such as dedicated routing rules, billing, and call logs.
• Develop admin interfaces for managing multi-tenant deployments, with functionality to provision new tenants, monitor usage, and generate reports.
• Ensure tenant-specific call routing, billing, and user management is handled securely and efficiently.
• API Integration with Third-Party Applications:
• Integrate FreeSWITCH with third-party applications through APIs to extend functionality, such as integrating with CRM systems, billing platforms, call analytics tools, and customer support solutions.
• Work with RESTful APIs, WebSockets, and other web technologies to connect FreeSWITCH with external systems, enabling real-time data exchange.
• Develop custom API endpoints to enable external systems to interact with FreeSWITCH for call management, billing, and reporting purposes.
• Troubleshoot API integrations and ensure reliable, efficient data synchronization between FreeSWITCH and third-party systems.
• SIP Routing with Kamailio/OpenSIPS:
• Configure and manage Kamailio or OpenSIPS for SIP signaling and call routing in a high-volume, multi-tenant environment.
• Develop and maintain SIP routing logic in Kamailio/OpenSIPS to support advanced call routing features such as load balancing, failover, and dynamic routing.
• Integrate Kamailio/OpenSIPS with FreeSWITCH to provide a robust SIP infrastructure for large-scale VoIP systems.
• Implement SIP authentication, NAT traversal, and security measures for efficient and secure SIP communications across tenants.
• Troubleshoot SIP issues in the Kamailio/OpenSIPS and FreeSWITCH ecosystem to ensure reliable communication across all tenants.
• Call Routing and Least Cost Routing (LCR):
• Develop and implement dynamic LCR algorithms that optimize call routing based on real-time carrier pricing, quality of service, and other factors.
• Work with carrier APIs to fetch pricing and routing information for LCR, ensuring that calls are routed efficiently and cost-effectively.
• Integrate LCR logic with Kamailio/OpenSIPS and FreeSWITCH for a seamless call-routing solution that ensures the lowest possible cost without compromising quality.
• Telecom Protocols and Networking:
• Configure SIP, RTP, and WebRTC signaling protocols to ensure smooth communication across multiple tenants.
• Address NAT traversal, STUN, TURN, and ICE issues for SIP and WebRTC traffic.
• Optimize network settings to support high availability, low-latency communication, and scalability.
• Collaboration and Support:
• Collaborate with internal teams to design, implement, and deploy new features, integrations, or optimizations for FreeSWITCH, Kamailio/OpenSIPS, and WebRTC.
• Provide support for multi-tenant systems, LCR routing, WebRTC, and API integrations.
• Document technical configurations, processes, and troubleshooting guides to support internal teams and external customers.
  
  

Required Skills And Qualifications

• Experience:
• Proven experience with FreeSWITCH in a multi-tenant environment (at least 2-3 years of hands-on experience).
• Strong experience with Kamailio or OpenSIPS, including configuring SIP routing, load balancing, failover, and security in high-volume environments.
• Expertise in Least Cost Routing (LCR) for VoIP systems, including integration with carrier APIs for real-time pricing.
• Experience integrating WebRTC with telephony platforms (FreeSWITCH) to enable browser-based real-time communication.
• Hands-on experience with API integration, especially integrating third-party applications such as CRM systems, billing platforms, and call analytics tools.
• Proficiency with Linux server environments (Ubuntu, CentOS, etc.).
• Technical Skills:
• Expertise in SIP routing and signaling with Kamailio or OpenSIPS, including advanced routing logic and real-time adjustments.
• Strong understanding of VoIP protocols (SIP, RTP) and WebRTC protocols for real-time communication.
• Experience with API development and integration, including RESTful APIs, WebSockets, and third-party service integration.
• Proficiency with scripting languages (Lua, Python, or Bash) to automate tasks and integrate systems.
• Experience with networking protocols (STUN, TURN, ICE, etc.) for handling NAT traversal and ensuring high-quality VoIP and WebRTC communication.
• Soft Skills:
• Strong troubleshooting and problem-solving abilities, especially for complex multi-tenant systems and SIP-related issues.
• Excellent communication skills for documenting processes and collaborating with cross-functional teams.
• Ability to manage multiple projects and support multiple tenants while ensuring system reliability and scalability.
  
  

Preferred Qualifications

• Experience with cloud-based telephony solutions (AWS, Azure, etc.) and large-scale deployments.
• Familiarity with call analytics platforms and integration with CRM, billing, or customer support systems.
  
  

Required Key Skills

freeswtich, webrtc, api integrations, kamailio, opensips, call analytics, crm billing

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