Description

You will design and implement telephony integrations using SIP and SIPREC protocols, managing call flows and media capture.

This role is on-site.

Responsibilities

• Design and implement backend services and APIs for call control, recording, and session management.
• Manage SIP call flows and media capture using PBX systems, SIP servers, and media servers.
• Integrate internal applications with third-party VoIP systems and analyze SIP signaling and RTP media streams.
• Debug and troubleshoot telephony issues using tools like Wireshark or sngrep.
• Create technical documentation, network diagrams, and integration guides.

Required Skills

• 4.5-8 years of experience in telephony development or VoIP systems integration.
• Deep knowledge of SIP protocol methods including INVITE, ACK, BYE, REGISTER, and REFER.
• Proven experience implementing SIPREC for VoIP call recording.
• Strong Node.js development skills for backend services and integrations.
• Hands-on experience with FreeSWITCH, Asterisk, Kamailio, or OpenSIPS.
• Experience with RTP media handling and WebRTC technologies.
• Strong networking knowledge covering UDP, TCP, NAT traversal, STUN, and TURN.
• Experience building RESTful APIs for voice platform integration.

Preferred Skills

• Experience designing scalable, fault-tolerant, and secure telephony systems.

Key Skills

Education

Any Graduate