How to use STUN function in VigorPhone 350 When we use an IP phone or other SIP device to dial VoIP phone call via NAT, a common problem that we might face is “communication”. Refer to the following example, IP phone connects to Internet via NAT and register an SIP account with the number of 111 to SIP server. Another IP phone connects to Internet directly and registers an SIP account with the number of 222. When both users (111 and 222) communicate, only 111 can hear the sound from 222 and 222 cannot hear any voice from 111 due to the problem of NAT. This is because 222 asks 111 to send the voice to LAN IP and port number of 222, but not the true IP and port number of NAT server. Therefore, no voice can be transmitted to 222. To solve the problem of one-way call, using the function of STUN server (Simple Traversal of User Datagram Protocol through NATs ) in IP phone to pass through NAT will be the best choice. After enabling the STUN function, IP phone will send a message to the STUN server before dialing out. The STUN server will notify current IP address and port number used by NAT device to the IP phone according to the message sent by IP phone. Next, IP phone will reply correct IP address and port number to the remote client based on the answer of STUN server. Thus, the problem of one-way call can be avoided. Here, we use VigorPhone350 as an example to explain how to configure STUN server. Open General SIP Settings >> Other Settings. Click On for STUN and type the IP address and port number (usually 3748) for STUN server. 1 Below shows STUN servers free of charge that you can find in Internet. z z z stun.ekiga.net stun.ideasip.com stun01.sipphone.com Note: STUN function is suitable for most of NAT except symmetrical NAT (bi-directional NAT). 2