Description

Key Skills: SIP, SIPREC, Node.js, VoIP Integration, FreeSWITCH, Asterisk, Kamailio, WebRTC, RTP, RESTful APIs, Wireshark, sngrep, Networking (UDP, TCP, NAT, STUN/TURN), Call Recording, Telephony Systems, SIP Trace Analysis.

Roles and Responsibilities:

  • Design and implement telephony integrations using SIP and SIPREC protocols.
  • Develop robust backend services and APIs for call control, recording, and session management.
  • Work closely with PBX systems, SIP servers, and media servers to manage SIP call flows and media capture.
  • Integrate internal applications with third-party VoIP systems to ensure seamless telephony experiences.
  • Analyze, debug, and troubleshoot SIP signaling and RTP media streams.
  • Collaborate with DevOps, QA, and Product teams to design scalable, production-ready solutions.
  • Create and maintain comprehensive technical documentation, network diagrams, and integration guides.
  • Ensure telephony systems are secure, fault-tolerant, and optimized for scalability.

Experience Requirement:

  • 4.5-8 years of experience in telephony development or VoIP systems integration.
  • Strong hands-on experience with SIP protocol including methods like INVITE, ACK, BYE, REGISTER, and REFER.
  • Proven experience in implementing SIPREC for VoIP call recording.
  • Solid background in Node.js development for backend services and integrations.
  • Worked extensively with SIP servers such as FreeSWITCH, Asterisk, Kamailio, or OpenSIPS.
  • Hands-on experience in RTP media handling and WebRTC technologies.
  • Familiarity with analyzing SIP traces using tools like Wireshark, sngrep, or similar.
  • Strong troubleshooting skills in networking including UDP, TCP, NAT traversal, STUN, and TURN.
  • Experience building RESTful APIs and integrating them with voice platforms.

Education:  Any Graduation

Key Skills
Education

Any Graduate

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