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Lead VOIP Engineer (FreeSWITCH & Kamailio)(m/w/x)
Designing and optimizing high-load FreeSWITCH/Kamailio VoIP systems for voice AI agents. Deep understanding of NAT traversal, STUN/TURN, WebRTC, and ICE frameworks essential. Focus on scaling voice AI agent deployments.
Requirements
- 5+ years of experience in VOIP engineering
- Strong expertise in SIP protocol, RTP streaming, and VoIP call flows
- Experience 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)
- Familiarity with WebRTC, Asterisk, OpenSIPS, or RTP Proxy is a plus
- 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
Tasks
- 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
Work Experience
- 5 years
Education
- Vocational certificationOR
- Bachelor's degreeOR
- Master's degree
Languages
- English – Business Fluent
Tools & Technologies
- FreeSWITCH
- Kamailio
- SIP protocol
- RTP streaming
- Homer SIP capture
- NAT traversal
- STUN
- TURN
- WebRTC
- ICE frameworks
- Linux
- Redis
- PostgreSQL
- NoSQL
- Bash
- Python
- Golang
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Lead VOIP Engineer (FreeSWITCH & Kamailio)(m/w/x)
Designing and optimizing high-load FreeSWITCH/Kamailio VoIP systems for voice AI agents. Deep understanding of NAT traversal, STUN/TURN, WebRTC, and ICE frameworks essential. Focus on scaling voice AI agent deployments.
Requirements
- 5+ years of experience in VOIP engineering
- Strong expertise in SIP protocol, RTP streaming, and VoIP call flows
- Experience 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)
- Familiarity with WebRTC, Asterisk, OpenSIPS, or RTP Proxy is a plus
- 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
Tasks
- 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
Work Experience
- 5 years
Education
- Vocational certificationOR
- Bachelor's degreeOR
- Master's degree
Languages
- English – Business Fluent
Tools & Technologies
- FreeSWITCH
- Kamailio
- SIP protocol
- RTP streaming
- Homer SIP capture
- NAT traversal
- STUN
- TURN
- WebRTC
- ICE frameworks
- Linux
- Redis
- PostgreSQL
- NoSQL
- Bash
- Python
- Golang
Like this job?
BetaYour Career Agent finds similar jobs for you every day.
About the Company
Synthflow AI
Industry
IT
Description
The company is a no-code platform for deploying voice AI agents that automate phone calls across contact center operations and BPO at scale.
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