and Jae - Hyun Kim School of Electrical and Computer Engineering , Ajou University Republic of Korea VoIP

VoIP services have been considered as one of the most important services in the next generation wireless systems. VoIP service requires the same quality of service (QoS) requirement as constant bit rate services. For this reason, the IEEE 802.16 standard has defined an unsolicited grant service (UGS) to guarantee the QoS. However, the UGS is inadequate to support VoIP services with silence suppression because of the waste of radio bandwidth in the silent-periods. In the UGS, a base station (BS) periodically allocates a maximum-size radio bandwidth (grant) during the silent-periods even though a subscriber station (SS) does not have a packet to transmit in the silent-periods. To solve this problem, (Lee et al., 2005) proposed an extended real time polling service (ertPS) to support VoIP services with silence suppression. The ertPS can manage the grant-size according to the voice activity in order to save the radio bandwidth in silent-period. Unfortunately, the waste of radio bandwidth and the increase of access delay can still exist when the ertPS is applied to the system because the grant-size and grant-interval used by the ertPS cannot correspond with the packet-size and the packet-generation-interval of the VoIP services in the application layer. Recently, the IEEE 802.16’s Task Group m (TGm), which was approved by IEEE to develop an amendment to IEEE 802.16 standard in 2006, published the draft evaluation methodology document in which several kinds of VoIP speech codecs are considered such as G.711, G.723.1, G.729, enhanced variable rate codec (EVRC), and adaptive multi-rate (AMR) (Srinivasan, 2007). These VoIP speech codecs generate packets with different packet-size and packet-generation-interval as shown in Table 1. However, the IEEE 802.16 standard does not define the QoS parameter generation method, because they have focused on only medium access control (MAC) and physical (PHY) layer. For this reason, IEEE 802.16 based systems need the QoS parameter mapping algorithm to obtain the features of the VoIP services in the application layer. Hong and Kwon (Hong & Kwon, 2006) proposed the QoS parameter mapping algorithm to exploit the feature of the VoIP services in IEEE 802.16 systems which statistically measures the peak data rate of VoIP services and calculates the QoS parameters. However, the algorithm needs significant time to measure the VoIP traffic to perform the statistical analysis, and the QoS parameters cannot correspond to the features of the VoIP services when the number of samples of the VoIP traffic is not sufficient to analyze the features of the VoIP service. To overcome these problems, this chapter designs a cross-layer QoS parameter mapping scheme which exploits the information of the VoIP speech codec included in the session description protocol (SDP) to generate the QoS parameters for VoIP scheduling algorithms. Sung-Min Oh and Jae-Hyun Kim School of Electrical and Computer Engineering, Ajou University Republic of Korea VoIP Features Oriented Uplink Scheduling Scheme in Wireless Networks 10