Connectionless data from existing network applications compose a large portion of the workload during an early ATM deployment, and are likely to make up an important portion of ATM's workload even in the long term. For providing a connectionless service over the ATM network, we compare two approaches; an indirect and a direct approaches, which are adopted by International Telecommunication Union-Telecommunication (ITU-T) as generic approaches. Our main subject of this paper is to compare network costs of two approaches by taking into account several cost factors such as transmission links, buffers, and connectionless servers in the case of the direct approach. Since the cost of the direct approach heavily depends on the configuration of a virtual connectionless overlay network, we propose a new heuristic algorithm to construct an effective connectionless overlay network topology. The proposed algorithm determines an optimal number of connectionless servers and their locations to minimize the network cost while satisfying QoS requirements such as maximum delay time and packet loss probability. Through numerical examples, we compare the indirect and direct approaches, the latter of which is constructed by means of our proposed algorithm.

This paper considers an efficient scheduling policy for Bluetooth Medium Access Control (MAC) and its parameter optimization method. The proposed algorithm improves performance as well as supports Quality of Service (QoS) simultaneously. Since Bluetooth is basically operated with a Round Robin (RR) scheduling policy, many slots may be wasted by POLL or NULL packets when there is no data waiting for transmission in the queues of the polled pair. To overcome this link wastage problem, several algorithms have been proposed. However, they have some limitations such as a heavy signaling overhead or no consideration of QoS. Therefore, we have proposed an efficient Bluetooth MAC scheduling algorithm, Differentiated K-Fairness Policy (Diff-KFP), which guarantees improved throughput and delay performance, and it can also lead to differentiated services. That is, if the parameter of the proposed algorithm is optimized, we can satisfy the QoS requirement of each master-slave pair and thereby keep communications in progress from interruption, which is a source of throughput degradation. Simulation results show that our algorithm has remarkably improved the performance and gratifies the QoS requirements of various applications.

In this paper, we propose a Distributed Request based CDMA Reservation ALOHA protocol to support multi-class services, such as voice, data, and videophone services, efficiently in multi-rate transmission cellular systems. The proposed protocol introduces a frame structure composed of an access slot and an transmission slot and an adaptive access permission probability based on the estimated number of contending users for each service, in order to control MAI by limiting the access to slots. It can provide voice service without the voice packet dropping probability through the proposed code assignment scheme, unlike other CDMA/PRMA protocols. The code reservation is allowed for voice and videophone services. The low-rate data service basically uses the remaining codes among the codes reserved for the voice service, but it can also use the codes already assigned to voice calls during the their silent periods to utilize codes efficiently. On the other hand, the high-rate data service uses the codes reserved for itself and the remaining codes among the codes reserved for the videophone service. Using the analytic method based on the Markov-chain subsystem model for each service including the handoff calls in uplink cellular systems, we show that the proposed protocol can guarantee the constant GoS for the handoff calls even with a large number of contending users through the proposed code assignment scheme and the access permission probability. Also, we show that the data services are integrated efficiently on the multi-rate transmission environment.

To prevent performance degradation of TCP due to packet losses in the smooth handoff by the route optimization extension of Mobile IP standard, a few packet buffering methods have been proposed. The packet buffering at the BS recovers the packets dropped during an inter-subnetwork handoff, by forwarding the buffered packets at the previous BS to the new BS to which the mobile host is connected after handoff. However, when the mobile host user moves to a congested BS in a new foreign subnetwork, those buffered packets are likely to be dropped at the new BS. Thus, as well as the TCP connections of the mobile host which have moved into the new BS, the already existing TCP connections of the new BS experience severe performance degradation. This effect is due to the increased congestion by the forwarded burst packets; all of the TCP connections can initiate their congestion control algorithms simultaneously, i.e., global synchronization. This paper will consider a general case where a mobile host user moves into a congested BS of a new foreign subnetwork. We analyze the influence of the packet buffering on the TCP performance in the new BS, for the Drop-Tail and Random Early Detection (RED) buffers. Simulation results show that although the RED buffer gives better handoff performance than the Drop-Tail buffer, it cannot avoid a large decrease in the TCP throughputs due to global synchronization, when a TCP connection is added at the BS by an inter-subnetwork handoff. Finally, we discuss some methods that can address the negative effect of the packet buffering method.

It is well-known that TCP often experiences severe performance degradation in mobile networks since packet losses not related to network congestion occur frequently due to host mobility. In this paper, we propose a new packet buffering method to address such a problem without the scalability problem in Mobile IP based networks. For this purpose, we first investigate the performance of TCP Tahoe without considering packet buffering through the simulation. Our simulation result shows that in most cases, the smooth handoff by the route optimization extension of Mobile IP standard cannot prevent the degradation of TCP performance due to handoffs, although it is designed to reduce the number of packets dropped during the handoff. It also shows that in utilizing the route optimization extension, the TCP performance sometimes becomes worse even than the case of the base Mobile IP unless its smooth handoff makes less than four packets be dropped during the handoff. Such results mean that at least for TCP, the smooth handoff is not useful unless the route optimization extension supports the buffering method, which makes handoffs be transparent to transport layer protocols by recovering the packets dropped during the handoff. We then investigate the effects of packet buffering on the performance of TCP. We modify the route optimization extension in order to support packet buffering at the base station, but it is a very minor change. Finally, we discuss some problems that should be addressed to recover the packets dropped during the handoff by the buffering method without giving a worse impact on the performance of TCP, and propose our solution to solve those problems.

In the wireless ATM network, the key issue is to guarantee various QoS (Quality of Service) under the conditions of the limited radio link bandwidth and error prone characteristics. In this paper, we show a combination method of the error correction schemes, which is suitable to establish multimedia wireless ATM Networks while keeping an efficient use of the limited bandwidth. We consider two levels of FEC; a bit-level and a cell-level to guarantee cell loss probabilities of real time applications. By combining two levels of FEC, various requirements on cell loss can be met. We then apply the bit-level FEC and ARQ protocol for the data communication; tolerant to the delay characteristics. Through the analytical methods, the required overheads of FECs are examined to satisfy the various QoS requirements of CBR connections. The mean delay analysis for the UBR service class is also presented. In numerical examples, we show how the combination scheme to guarantee various cell loss requirements affects the call blocking probability of the CBR service class and the delay of UBR service class.

In this paper, we investigate a call admission control (CAC) problem in a multimedia wireless ATM network that supports various multimedia applications based on micro/pico cellular architectures. Due to reduced wireless cell size (compared to conventional cellular networks), forced termination of calls in progress becomes a more serious problem in the wireless ATM network. Another problem specific to the multimedia wireless network is how to avoid an excessive delay of non-realtime applications under the presence of various realtime applications with priority over non-realtime applications. We consider two service classes; CBR for realtime applications and UBR for non-realtime applications, and then propose a new CAC scheme that addresses above two problems while minimizing a blocking probability of newly arriving calls of CBR. Through the analytical methods, we derive the blocking probabilities and forced termination probabilities of CBR calls and the average packet delay of UBR connections. We also present a method that decides the optimal CAC threshold values in our CAC scheme.