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oliver-ruben/README.md

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Oliver Ruben

Senior VoIP Engineer | WebRTC and Streaming Technology Specialist

About

I am a telecommunications engineer with over a decade of deep, hands-on experience in VoIP, SIP signaling, and real-time media streaming. My background includes designing, building, and maintaining reliable, high-performance communication systems for service providers and enterprise clients.

My primary technical focus is on SIP server engineering, large-scale WebRTC deployments, media proxying, and building resilient, secure, and scalable platforms for real-time voice and video services. I am passionate about open source, protocol analysis, and robust system architecture.

Areas of expertise:

  • SIP server development and operations (Freeswitch, Kamailio, OpenSIPS, Asterisk)
  • WebRTC signaling and media path, including TURN/STUN, ICE, RTP/SRTP
  • Large-scale system design: high-availability, failover, load balancing, clustering
  • Security best practices for VoIP, including encryption, anti-fraud, and abuse mitigation
  • Monitoring and troubleshooting (Homer SIP Capture, Prometheus, Grafana)
  • Automation and deployment (Ansible, Docker, Kubernetes)
  • Scripting and backend integrations (Python, Bash, Node.js, C/C++)

Core Technologies

  • SIP Servers: Freeswitch, Kamailio, OpenSIPS, Asterisk, RTPengine
  • WebRTC Stack: TURN/STUN, ICE, SFU/MCU, SDP, NAT traversal
  • Protocols: SIP, RTP, SRTP, ZRTP, SIPREC, WebSocket, HTTP/2
  • DevOps: Docker, Kubernetes, Ansible, Linux server administration
  • Monitoring: Prometheus, Grafana, Homer, VoIPmonitor

Selected Work & Open Source Projects

  • Advanced SIP routing and dynamic least cost routing modules for Kamailio and OpenSIPS
  • WebRTC gateways and media bridges for secure connectivity between browsers and SIP endpoints
  • Automation pipelines for telecom infrastructure deployment (CI/CD, Docker Compose, Kubernetes)
  • Real-time analytics dashboards for VoIP and streaming traffic
  • Custom integrations for fraud detection and automated abuse response

For sample code and contributions, please see the pinned repositories below.

GitHub Analytics

GitHub Stats GitHub Streak

Contact

For professional inquiries, collaboration, or consultation regarding VoIP, SIP, WebRTC, or streaming solutions, please use one of the methods below:

Email [email protected]
LinkedIn oliver-ruben-99651a33b
WhatsApp +1 706 664 5169
Telegram @dsky2342

If your project, team, or organization is working in real-time communications and next-generation telephony, I welcome your message.

Popular repositories Loading

  1. oliver-ruben oliver-ruben Public

    1

  2. SIP.js SIP.js Public

    Forked from onsip/SIP.js

    A simple, intuitive, and powerful JavaScript signaling library

    TypeScript 1

  3. JsSIP JsSIP Public

    Forked from versatica/JsSIP

    JsSIP, the JavaScript SIP library

    JavaScript 1

  4. freeswitch freeswitch Public

    Forked from signalwire/freeswitch

    FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From…

    C 1

  5. kamailio kamailio Public

    Forked from kamailio/kamailio

    Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -

    C 1

  6. opensips opensips Public

    Forked from OpenSIPS/opensips

    OpenSIPS is a GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP serve…

    C 1