The rtPS algorithm is designed to support real-time service flows, such as MPEG video or tele-conference, that generate variable size data packets periodically. In this algorithm, the BS assigns UL resources that are sufficient for unicast bandwidth request to the voice users. This is called the polling process. The duration for which the BS continues to poll an SS with rtPS connection is negotiated in the initialization process of the connection. The SSs utilize the assigned polling resources to send their bandwidth requests, reporting the exact bandwidth need for their rtPS connection. The BS in response then allocates the exact bandwidth requested to the SS for transmission of the data. Figure 1 illustrates this dynamic polling process.
Because rtPS always carry out polling process, it is able to adaptively determine suitable resource allocation from frame to frame. This adaptive request-grant process goes on until the connection is terminated. Because of the dynamic request-grant process, the algorithm has more optimum data transport efficiency than the UGS algorithm. The algorithm is able to dynamically follow all data rate of the voice codec without any resource wastage as illustrated in Figure 2 (allocated and utilized resources are equal). This is a major advantage over the UGS algorithm. The drawback of the rtPS algorithm however is that the dynamic polling process causes MAC overhead and access delay. Hence rtPS has more MAC overhead and larger access delay than the UGS.
