Responsibilities
- Design, deploy, and maintain FreeSWITCH and Kamailio-based VoIP solutions.
- Configure and troubleshoot SIP protocols, RTP streams, and media gateways.
- Monitor and optimize high-load production FreeSWITCH and VOIP systems.
- Implement and manage SIP monitoring tools like Homer or other real-time debugging solutions.
- Develop and fine-tune call routing, failover mechanisms, and load balancing strategies.
- Ensure security, scalability, and redundancy in the VoIP infrastructure.
- Troubleshoot and resolve VoIP-related issues in production environments.
- Collaborate with development, DevOps, and support teams to improve system performance.
- Work with SIP Trunks, SBCs (Session Border Controllers), and Media Servers.
- Implement WebRTC-based voice and video solutions.
- Document system configurations, troubleshooting guides, and best practices.
Requirements
- 5+ years of experience in VOIP engineering, particularly with FreeSWITCH and Kamailio.
- 3 years of experience with k8s on production systems is a must have.
- Strong expertise in SIP protocol, RTP streaming, and VoIP call flows.
- Experience working in high-load production FreeSWITCH environments.
- Hands-on experience with Homer SIP capture or similar SIP monitoring tools.
- Deep understanding of NAT traversal, STUN/TURN, WebRTC, and ICE frameworks.
- Strong knowledge of load balancing, failover strategies, and traffic optimization.
- Proficiency in Linux administration, networking, and troubleshooting.
- Experience with database integration (Redis, PostgreSQL, or NoSQL solutions).
- Ability to write Bash/Python (or Golang) scripts for automation and monitoring.
- Strong problem-solving skills and ability to work in highly available and scalable environments.
Nice to Have
- Familiarity with WebRTC, Asterisk, OpenSIPS, or RTP Proxy is a plus.
