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Sip developer - 100% Remote

Innovative IT Solutions Inc
United States
11 days ago
Apply on www.dice.com
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Role details

Contract type
Contract
Employment type
Full-time (> 32 hours)
Working hours
Regular working hours
Job source
www.dice.com

Tech stack

Automatic Call Distributor Application Programming Interfaces (APIs) User Authentication VoIP Computer Programming Network Congestion Fault Tolerance Interoperability Session Initiation Protocols Linux System Administration Networking Basics Network Connections
+10 more
Performance Tuning Session Management Software Engineering System Testing Scripting Enterprise Software Applications Load Balancing System Availability Low Latency Kamailio

Job description

  • 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.

Requirements

  • 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.

Benefits & conditions

Depends on Experience

Apply for this position

This job is hosted externally. Click below to view the full posting and apply.

Apply on www.dice.com
Prepare application

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