VoIP Developer (FreeSwitch - Asterisk)

Job Purpose

We are looking for an experienced VoIP Developer with 4+ years of hands-on experience in FreeSWITCH and VoIP technologies. The candidate will be responsible for developing, configuring, integrating, troubleshooting, and maintaining high-performance telephony and communication solutions.

Key Responsibilities

FreeSWITCH Development & Administration

• Develop, configure, and maintain FreeSWITCH-based VoIP solutions.
• Configure SIP profiles, gateways, extensions, dial plans, IVR, call routing, and call flows.
• Develop and customize FreeSWITCH XML dial plans, ESL, APIs, and modules.
• Integrate FreeSWITCH with dialers, CRM, contact centre, and third-party applications.
• Monitor and optimize FreeSWITCH performance and production environments.

SIP & VoIP Integration

• Configure and troubleshoot SIP trunks, SIP gateways, SIP signalling, and RTP media.
• Integrate FreeSWITCH with SBC, Asterisk, FusionPBX, VICIdial, CUCM, EPBX, and third-party PBX systems.
• Handle SIP authentication, codec configuration, NAT traversal, and interoperability issues.
• Work with SIP, RTP, SRTP, WebRTC, TCP/IP, and UDP.

Troubleshooting & Production Support

• Troubleshoot call failures, one-way/no audio, dropped calls, registration issues, and routing problems.
• Diagnose latency, jitter, packet loss, codec mismatch, SIP signalling, and RTP issues.
• Analyze FreeSWITCH logs and SIP traces for root-cause identification.
• Perform RCA and implement preventive solutions for recurring incidents.
• Coordinate with telecom carriers, network teams, and third-party vendors.

Qualification & Experience

• Bachelor's degree in Computer Science, IT, Electronics, Telecommunications, or equivalent.
• 4+ years of professional experience in VoIP/Telephony development.
• Strong hands-on FreeSWITCH experience is mandatory.
• Strong understanding of SIP, RTP, VoIP protocols, call flows, and telephony architecture.
• Experience with Asterisk, FusionPBX, VICIdial, SBC, or CUCM will be an added advantage.

Required Technical Skills

• FreeSWITCH – Mandatory
• SIP / SIP Trunking / SIP Profiles
• RTP / SRTP / WebRTC
• FreeSWITCH XML Dial Plan
• ESL (Event Socket Library) & FreeSWITCH APIs
• IVR & Call Flow Development
• SIP Gateway & PBX Integration
• SBC Configuration & Integration
• Linux / Linux CLI
• TCP/IP / UDP / NAT
• Codec Configuration & Troubleshooting
• SIP Trace & VoIP Debugging
• MySQL / PostgreSQL
• Python / PHP / Node.js or similar programming knowledge

Key Competencies

• Strong analytical and troubleshooting skills.
• Ability to design and implement complex VoIP call flows.
• Strong understanding of telephony and networking concepts.
• Experience handling production-critical VoIP environments.
• Good documentation and RCA skills.
• Ability to work independently and coordinate with internal teams, carriers, and vendors.