Parloa
Forward Deployed Engineer, VoIP(m/w/x)
Full-timeWith HomeofficeSenior
Berlin, München
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SYSynthflow AI
Lead VOIP Engineer (FreeSWITCH & Kamailio)(m/w/x)
Berlin
Full-timeWith Home OfficeSenior
Nejo AI Summary
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
Find the original job posting in its most current version here. Nejo automatically captured this job from the website of Synthflow AI and processed the information on Nejo with the help of AI for you. Despite careful analysis, some information may be incomplete or inaccurate. Please always verify all details in the original posting! Content and copyrights of the original posting belong to the advertising company.
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SYSynthflow AI
Lead VOIP Engineer (FreeSWITCH & Kamailio)(m/w/x)
Berlin
Full-timeWith Home OfficeSenior
Nejo AI Summary
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
Find the original job posting in its most current version here. Nejo automatically captured this job from the website of Synthflow AI and processed the information on Nejo with the help of AI for you. Despite careful analysis, some information may be incomplete or inaccurate. Please always verify all details in the original posting! Content and copyrights of the original posting belong to the advertising company.
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.
Not a perfect match?
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