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Sip developer - 100% Remote

Innovative IT Solutions IncUnited States🇺🇸United States

Quick overview

SeniorityMid Senior
Work modeRemote
LocationUnited States
EncryptionLoad BalancingLinux

Job description

Key Responsibilities

  • Design, develop, configure, and maintain VoIP and SIP-based communication systems using FreeSWITCH and Kamailio.

  • Architect scalable, highly available, and fault-tolerant telephony solutions for healthcare and medical applications.

  • Develop and customize FreeSWITCH dial plans, call routing logic, SIP profiles, gateways, and media handling.

  • Configure and manage Kamailio as a SIP proxy, registrar, load balancer, and routing engine.

  • Implement SIP signaling, call routing, session management, and interoperability between FreeSWITCH, Kamailio, and third-party telecom platforms.

  • Integrate telephony systems with healthcare applications, contact centers, patient engagement platforms, appointment scheduling systems, and clinical communication workflows, as applicable.

  • Troubleshoot SIP signaling, RTP media, call setup failures, audio quality issues, latency, packet loss, and call drops.

  • Implement security measures for SIP communications, including authentication, access control, encryption, and protection against unauthorized access.

  • Optimize system performance, scalability, availability, and call quality across high-volume communication environments.

  • Collaborate with software engineering, infrastructure, networking, and healthcare application teams to deliver end-to-end communication solutions.

  • Monitor production environments, analyze logs and call traces, perform root-cause analysis, and resolve complex telephony issues.

  • Support system testing, deployment, upgrades, documentation, and ongoing maintenance.

Required Skills and Qualifications

  • Strong hands-on experience in FreeSWITCH and Kamailio — mandatory.

  • Proven experience in designing and implementing SIP-based VoIP and telephony solutions.

  • Strong understanding of SIP, SDP, RTP, SRTP, and related VoIP protocols.

  • Experience with SIP routing, dial plans, gateways, call flows, and media handling.

  • Experience integrating FreeSWITCH and Kamailio with external applications, APIs, and telecom services.

  • Strong troubleshooting skills in SIP signaling, call routing, audio quality, and network connectivity.

  • Experience with Linux environments, networking fundamentals, and scripting or programming for telephony automation.

  • Experience with high availability, load balancing, scalability, and production-grade telecom systems.

  • Previous experience in the medical or healthcare domain is required.

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