Setup your SIP SBC within minutes         

The Mizu SIP SBC can be used to control SIP signaling and media streams. You can install it on the edge of your VoIP network to enforce security and to perform various tasks such as validation of SIP sessions and NAT handling.
It can be used also as a Gateway or as a B2B SIP proxy. The SIP SBC will also solve any incompatibility issues between devices.
The SIP SBC is implemented in native C++ and it can be installed on Windows OS as an NT service, offering sub millisecond packet routing delays. 
Have a look at the SBC documentation for the details (Online | PDF | WinHelp). More general details about SBC's can be found here

It was never been easier to setup a SIP SBC.

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SIP SBC Features

  • GUI: comfortable graphical user interface for configuration, management and monitoring (real-time, CDR's, statistics)
  • NAT: the SBC has built-in NAT traversal, support capable to gracefully handle all types of NAT's, firewalls and routers
  • Security: SIPS, DoS and DDoS attack protection with intelligent firewall, lawful intercept, topology-hiding, access control, call limits, rate limiter, stream and session protection, blacklist, fraud call detection and various other thresholds
  • Transparency: no modifications will be required on your SIP server or for your SIP clients. The SBC can act as a transparent SIP proxy forwarding all authorization requests to your server (you will have to manage the trunks/extensions/routing/dial-plans on your server exactly as you did it before)
  • Internetworking: compatible with all SIP devices, will resolve any incompatibility issues between devices
  • Protocol conversion: SIP, WebRTC (optional), UDP/TCP/TLS, RTP/SRTP/DTLS
  • SIP trunks: manage inbound/outbound trunks allowing lower cost SIP trunking
  • Routing: intelligent routing of calls can be done by various rules such as priority, quality and load-balancing
  • Failover: it can detect failed servers and reroute the calls to others or temporarily disable failed servers 
  • RTP relay: bypass, route or off-route the media with various enhancements such as NAT handling, DTMF, QoS and stream control. It can be set to automatic (will route the media only if necessary) or enforced by manual settings
  • Transcoding: transcode between various codec's such as Opus, Speex, G.729, G.723, G.711 (PCMU/PCMA), GSM, iLBC (this is done automatically by default only when necessary and can be overridden manually)
  • Encryption: encrypt/decrypt between clear SIP/RTP and SIPS-TLS/SRTP (and also DTLS for WebRTC calls)
  • Recording: voice call recording, chat recording, detailed logs and CDR's
  • Optional modules: billing, user management, registrar, class 5 features, WebRTC-SIP, push notifications, tunneling, high-speed load balancer



  • Basic (20 simultaneous calls, 50 users): Free Download
  • Standard (200 simultaneous calls, 2 000 users): $2900 Buy Now
  • Advanced (2 000 simultaneous calls, 100 000 users): $5900 Buy Now
  • Gold (unlimited calls and users): Contact Us

We can also accept payment by card or wire-transfer.