Telephony Engineer

HappyRobot is the AI-native operating system for the real economy—a system that closes the circuit between intelligence and action. By combining real-time truth, specialized AI workers, and an orchestrating intelligence, we help enterprises run complex, mission-critical operations with true autonomy.

We're looking for a Telephony Engineer based in Madrid to own and scale the real-time voice infrastructure that powers HappyRobot's AI workers across Europe. You'll be the go-to person for everything telephony on this side of the Atlantic—from SIP trunking and SBC configuration to CCaaS integrations and network troubleshooting.

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  Own and operate European SBC infrastructure (OpenSIPS/Kamailio). Configure, monitor, and troubleshoot SIP trunks for enterprise clients. Ensure high-availability voice delivery with sub-second failover.

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  Integrate AI voice agents with CCaaS/UCaaS platforms such as Genesys Cloud, Five9, NICE CXone, RingCentral, Zoom, 8x8, and others. Handle OAuth flows, SIP header parsing, queue routing, and agent-availability checks.

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  Diagnose call quality issues—latency, jitter, one-sided audio, codec negotiation failures—using Wireshark, tshark, mtr, and ffmpeg. Capture and analyze SIP call flows end-to-end.

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  Optimize call flows, IVRs, routing logic, and contact center performance across client deployments.

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  Support trunk provisioning and incident resolution with carriers and telephony vendors as needed.

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  Maintain and improve telephony infrastructure in Kubernetes. Write automation scripts and tooling in Golang, JavaScript, or Python. Contribute to CI/CD pipelines and monitoring.

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  Join technical calls with enterprise clients and their telephony partners. Translate complex SIP and networking issues into clear language for non-technical stakeholders.

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  Create and maintain documentation on infrastructure, integrations, and procedures.

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  Help evaluate and deploy next-generation real-time communication tools (LiveKit, WebRTC) as the voice stack evolves.

• Deep SIP/VoIP expertise. You can read a SIP call flow in your sleep. You understand INVITEs, REGISTERs, codecs (PCMU, PCMA, G722), SRTP, DTMF (RFC 2833), and one-sided audio debugging.

• OpenSIPS or Kamailio experience. Hands-on experience configuring, scaling, and troubleshooting SIP proxy servers is essential for this role.

• Strong networking fundamentals. TCP/UDP, DNS, TLS, NAT traversal, firewall rules, port management. Proficient with Wireshark/tshark, mtr, traceroute, and packet-level debugging.

• CCaaS/UCaaS/PBX knowledge. Hands-on experience integrating with platforms like Genesys Cloud, Five9, NICE CXone, RingCentral, 8x8, Zoom, or similar contact center solutions.

• DevOps competency. Comfortable with Kubernetes, Git, and CI/CD pipelines. You don't need to be a 10x developer, but you must be able to tread water with code and leverage AI tools effectively.

• Proficiency in at least one of: Golang, JavaScript/Node.js, or Python for scripting, automation, and tooling.

• Audio engineering basics. Understanding of audio codecs, transcoding, ffmpeg, and media handling in RTP streams.

• Experience with system integration via APIs, webhooks, and middleware platforms.

• Strong troubleshooting skills across networks, voice quality issues, and platform interoperability in enterprise environments with high availability and scalability needs.

• Excellent communication and documentation skills.

• Fluent English, written and spoken.

• Networking or SIP School certifications.

• Experience with telephony vendors and SBC platforms such as Ribbon, Oracle, or Oculeus.

• Familiarity with LiveKit or WebRTC-based real-time communication frameworks.

• Experience working with European telcos and carriers.

• Carrier-side provisioning experience (number porting, trunk setup, regulatory compliance).

• Certifications in Genesys, NICE, or equivalent telephony solutions.

• Exposure to contact center analytics and reporting tools.