The Internet was originally designed for nonreal-time data services such as interactive burst or interactive bulk transfer. In these applications, there are no requirements on the maximum amount of delays that a packet may encounter during its transit to the destination. Similarly, bandwidths required by an end user are never specified. As such, the network accepts all incoming packets without using any admission control mechanism, forwards them using a simple, first-come-first-served algorithm, and delivers them on a best-effort basis. [1] Thus, issues concerning the quality of service (QoS) delivered to an end user are rather straightforward. The QoS in present-day mobile IP is also minimal because, once again, data is delivered using the best-effort scheme. With the emergence of real-time multimedia services as envisaged by third-generation (3G) wireless systems, new QoS requirements are imposed on the networks. For example, with interactive video conferencing or streaming video and audio, the network must be able to deliver these services to the destination on a timely basis. Because flow control or retransmission is not possible for these applications, the bit error rate or packet loss ratio must be kept below a certain level; otherwise, the QoS may suffer. For instance, if the bit error rate is too high, the video in an MPEG application may never synchronize at a receiver. [2]