si² uses Network Address Translator (NAT) technology. It translates IP address of the originator so that remote computers would think that the incoming request was originated by si² rather than the original host computer that resides on the LAN. When the requested data is sent back to the originator, si² will change back it IP address to its original state. This will eliminate the need for having to lease IP addresses from ISP, which is very expensive. Furthermore, the use of NAT also enable the workstations to use applications that do not support proxy servers.
As for the VPN function, si² uses the CIPE technology. CIPE encapsulates encrypted IP datagrams in UDP datagrams and sends them via the normal UDP mechanism. This is different from standard IPIP encapsulation. UDP was chosen because this way many different endpoints can easily be distinguished by port numbers; because an IP protocol number would warrant a formal registration; and because handling of UDP datagrams is easier than using a separate IP protocol number, especially in firewalled setups. Specifically, UDP can be handled by user-level applications such as a SOCKS5 relayer. A CIPE link always connects exactly two endpoints. In many ways, the link works like a PPP dial-up link. At present, each link has its own secret 128-bit key which has to be known by both ends (and nobody else). This link key (called static key in the protocol description) is used to negotiate a frequently changed dynamic key, which encrypts the actual data. It is also possible to negotiate the keys via a public key mechanism, similar to the SSH package. This removes the need for shared secret keys.
For the PABX function, si² also has that function built-in. It provides all of the features you would expect from a PABX and more. si² does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using comparatively inexpensive hardware. si² provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway). Check the Features section for a more complete list. si² needs no additional hardware for Voice over IP. For interconnection with digital and analog telephony equipment, si² supports a number of hardware devices such as single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks.