IP telephony protocols support (SIP, H323)
UserGate v.5 supports the SIP and H323 protocols, which allow using the UserGate Server as a VoIP gateway for software IP phones, as well as for conventional IP phones.

The SIP proxy option with connection condition control (state full proxy) is added to version 5. A SIP proxy can be enabled in UserGate in “Services – Proxy Settings” and it always works in transparent mode listening to ports 5060 TCP and 5060 UDP.

Details (the connection status (registration, call, waiting, etc.), user name (or number), call duration and the number of bytes sent and received) are available through the Administration Console “Sessions” section.The same information is stored in the UserGate statistics database.

For SIP proxy usage please specify the UserGate Server IP as the default gateway in the TCP/IP options section of the user workstation, as well as DNS server address.

The setup of the client side is illustrated by the example of SJPhone software phone and the Sipnet provider. Run SJPhone, right-click on its icon in the system tray, choose the item “Options” from the context menu and create a new profile. Enter the profile name (Fig. 9), for example “sipnet.ru”, as a profile type and specify “Calls through SIP Proxy”.

Figure 9. SJPhone profile creation.

You should define your VoIP provider proxy server address in the same “Profile Options” dialog box. When closing the dialog, please enter your username and password for authorization on your VoIP provider server.

Figure 10. SJPhone profile settings.

Built-in H323 protocol support enables you to use UserGate Server as a Gatekeeper. In the H323 settings you should specify the interface where all the client requests are listened, the port number, as well as the H323 gateway address and port. For authorization on UserGate Gatekeeper, the user should specify its UserGate user name, password (or workstation IP address) and phone number. These points should be same as those in the UserGate user settings for authorization.