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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.
Anforderungen
- 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
Aufgaben
- 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
Berufserfahrung
- 5 Jahre
Ausbildung
- Abgeschlossene BerufsausbildungODER
- Bachelor-AbschlussODER
- Master-Abschluss
Sprachen
- Englisch – verhandlungssicher
Tools & Technologien
- FreeSWITCH
- Kamailio
- SIP protocol
- RTP streaming
- Homer SIP capture
- NAT traversal
- STUN
- TURN
- WebRTC
- ICE frameworks
- Linux
- Redis
- PostgreSQL
- NoSQL
- Bash
- Python
- Golang
Noch nicht perfekt?
- ParloaVollzeitmit HomeofficeSeniorBerlin, München
- Innovaphone
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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.
Anforderungen
- 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
Aufgaben
- 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
Berufserfahrung
- 5 Jahre
Ausbildung
- Abgeschlossene BerufsausbildungODER
- Bachelor-AbschlussODER
- Master-Abschluss
Sprachen
- Englisch – verhandlungssicher
Tools & Technologien
- FreeSWITCH
- Kamailio
- SIP protocol
- RTP streaming
- Homer SIP capture
- NAT traversal
- STUN
- TURN
- WebRTC
- ICE frameworks
- Linux
- Redis
- PostgreSQL
- NoSQL
- Bash
- Python
- Golang
Über das Unternehmen
Synthflow AI
Branche
IT
Beschreibung
The company is a no-code platform for deploying voice AI agents that automate phone calls across contact center operations and BPO at scale.
Noch nicht perfekt?
- Parloa
Forward Deployed Engineer, VoIP(m/w/x)
Vollzeitmit HomeofficeSeniorBerlin, München - Innovaphone
Systems Engineer Innovaphone(m/w/x)
Vollzeitmit HomeofficeBerufserfahrenBerlin, Bremen, Cottbus, Düsseldorf, Greven, Westerstede - Nebius
Senior Site Reliability Engineer(m/w/x)
Vollzeitmit HomeofficeSeniorBerlin - Synthflow AI
Senior Python Software Engineer(m/w/x)
Vollzeitmit HomeofficeSeniorBerlin - emnify
Senior Backend Engineer(m/w/x)
Vollzeitmit HomeofficeSeniorBerlin, Würzburg