The network and the wireless communication technology are all pervasive and growing. VoIP is very popular in recent years. Every IP phone requires a public IP address, but there are no enough public IP addresses. At present to solve this problem is use a NAT (Network Address Translation). NAT is a technology used for broadband connections which allows multiple devices to share one connection for accessing the Internet. The disadvantage is that Peer to Peer VoIP applications do not work behind NAT without settings. In this thesis, I introduce how to realize the VoIP by SIP operations which enables two SIP clients to communicate each other. Later, I explain five kinds of NAT and the SIP traversal over NAT, such as STUN, ICE, RTP Relay Server, UPnP and TURN. We found RTP Relay Server is the most easy and acceptable. The major disadvantage of using a RTP Relay Server is that the media will have to travel via a third party, the relay server, on the Internet. The quality of the phone call may be affected by the relay server. Therefore, I rewrite RTP Relay Server application in python to let it be able to use SNMP to measure whether the network flow is over the threshold. When the loading on the server is over than it will bear, the server will reject the connection. Then, this connection will be dispatched to another relay server to maintain a good communication quality.