Sobre

I’m looking for an experienced VoIP engineer to configure an SBC (Session Border Controller) to handle SIP call routing for my system. What I need: Configuration of OpenSIPS or Kamailio as an SBC Dynamic caller ID rewriting based on rules Routing outbound calls to my SIP trunk (outbound route) NAT handling and topology hiding SIP header manipulation as needed Basic security hardening (rate-limit, anti-fraud, user auth) Support for high CPS routing (discador / dialer scenario) Optional: Integration with rtpengine (if needed for RTP relay) Environment: VPS: 16 vCPU / 16 GB RAM SIP trunk already available Discador cloud system sending calls to the SBC Deliverables: Fully working SBC with tested call flow Routing logic documented Instructions to manage numbers, caller IDs and routing rules Optional: simple web panel or flat config files for editing rules Ideal freelancer: Strong experience with OpenSIPS / Kamailio Knowledge of SBCs, SIP signaling, NAT, RTP, SIP trunks Experience with high-volume call routing Able to deliver clean, maintainable configuration If you have experience with similar setups, please send examples of previous projects. Looking forward to working with you!