Call admission control of multiservice hierarchical wireless networks supporting soft handoff is studied, where users with different average call holding times are assigned to different layers, i.e., microcells in the lower layer are used to carry users with shorter call holding time, whereas macrocells in the upper layer are for users with longer call holding time. Further, to give handoff calls priority over new calls, handoff queues are provided for handoff calls that can not obtain the required channel immediately. According to whether handoff queues are provided in microcells and/or macrocells, four different call admission control schemes are proposed and studied. We derive the mathematical model of the considered system with multi- dimensional Markov process, and find the steady state probability distribution iteratively and thus the performance measures of interest: new call blocking probability, forced termination probability, and mean waiting time in handoff queue. Analytical results show that providing handoff queues in both microcells and macrocells can achieve the best blocking performance at the expense of mean waiting time in handoff queue.

Single cell reuse of the same frequency, which is possible in DS-CDMA cellular systems, yields the option of site diversity to increase link capacity. In this letter, a generalized case of site diversity transmission is considered where multiple base stations (BS's) are involved in weighted transmissions with constant total transmit power to a target mobile station (MS). A general equation of conditional bit error rate (BER) is derived based on the model of weighted transmissions combined with antenna diversity reception and rake combining. It turns out theoretically that the optimum set of weights to maximize forward link capacity makes site selection diversity transmission (SSDT) the best performer. This theoretical analysis is confirmed by performance evaluation based on the Monte-Carlo simulation.

Recent advancement in personal communication services (PCS) provides wireless multimedia services for users on the move. For establishing the convenient access environment, the evolution of IP-based integrated scheme becomes very urgent for public wireless access systems. In this paper, we clarify the well-known issues of integrating the general packet radio service (GPRS) network and IEEE 802.11 wireless local area network (WLAN). These issues include the inconsistence in transmission rate and coverage area, the difficulty in seamless handoff, the complexities of mobile IP and address translations, and so on. Based on classified cases, we propose the circular probe strategy (CPS) to solve the IP translations and measure the precise handoff latency of a mobile node roaming from one network to another. This latency information helps a mobile node and its home agent (HA) to decide the appropriate handoff timing, to maximize the data transmission rate, and to perform seamless handoff in the heterogeneous system. We also proposed the smart access point (AP) mechanism to reduce the overhead of CPS and to provide precise handoff latency information for mobile nodes (MNs) to select the most appropriate AP for association. Experiment results demonstrate that the proposed smart handoff scheme performs better than traditional algorithms in terms of handoff time in the integrated network.

Since the interference is quite related to the performance of CDMA cellular systems, it is necessary to estimate Other-Cell-Interference Factor (OCIF). Here, starting from OCIF calculation for an aeronautical communication system, we investigate the forward link performance of data packet transmission in which the capacity, throughput and delay of the system are measured. To the numerical results, one can see that the performance of the aeronautical communication system is worse than that for terrestrial cellular systems and also depends logarithmically on both the cell radius and height.

In order to exploit the benefits of soft handoff, finding good values of handoff control parameters is important. In cellular system specifications such as IS-95C and WCDMA, handoff decision algorithm includes signal strength averaging and drop timer in addition to hysteresis. This paper analyzes the effects of signal strength averaging and drop timer and their performance tradeoffs. Because averaging and drop timer are both based on time delay, one may expect that they have similar impact on soft handoff performance. The results show that the effects of averaging and drop timer are rather similar and closely connected in terms of reducing the signaling overhead. However, they have different impacts on resource usage and diversity gain of the system.

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.

Proposed CCRSVP (CandidateCasting RSVP) algorithm is a new fast handoff method for IEEE 802.11 Wireless LAN (WLAN) environments. It shows good performance in the handoff latency and the bandwidth efficiency aspect and guarantees QoS because it uses an advanced multicasting method and RSVP. CCRSVP uses L2 information (BSSID) of WLAN and starts reserving resources and multicasting packets before L2 handoff completes. Therefore, the proposed algorithm can reduce L3 handoff latency more than other methods. To show performance of CCRSVP algorithm, we calculate handoff latency and packet loss ratio of each algorithm. Also we model handoff process which uses RSVP mechanism to confirm resource efficiency. Proposed handoff model uses parameters which can distinguish each handoff algorithm. We introduce Markov chain which can analyze handoff model and analysis method which uses iteration method. In this article, the results show that the proposed algorithm shows superior bandwidth efficiency than existing L3 handoff algorithms using RSVP. To analyze bandwidth efficiency of each algorithm, we compare the blocking probability which occurs in case of absence of resource, resource usage which shows reservation quantity, the average number of ongoing session which really uses resource reserved and resource utilization. We can confirm that CCRSVP algorithm has better performance than other algorithms at each comparative item.

In this paper, we present a group communication protocol that achieves total ordering message delivery for mobile computing systems with multiple overlapping groups. Our mechanism is an efficient adaptation of the propagation-tree technique to the mobile computing environments. It takes advantages of the capability of stationary mobile support stations to overcome the deficiencies associated with mobile devices. We construct the propagation tree based on the stationary stations, rather than the mobile hosts. As a result, mobile hosts are relieved of the excessive load of forwarding messages and communications on wireless channels are confined to transmitting messages to destination processes. This is important considering that the bandwidth of the wireless channels is limited. Moreover, the proposed protocol employs a mechanism to synchronize transmissions within a wireless cell. This serves to avoid redundant transmissions of a message in a wireless network in an attempt to achieve better utilization of the network bandwidth. Our mechanism relies on a handoff operation to deal with mobility of mobile devices. The handoff procedure ensures a smooth integration of a mobile host into a new cell, while preserving reliability of communication and the total ordering property of message delivery.

This paper proposes an analytical model to demonstrate the benefit of data service in wireless networks using dynamic multi-channel scheme with channel de-allocation. The performance of a system providing buffers to voice calls to reduce the raised voice blocking probability caused by data contention is investigated. The effect of the cell dwell time and overlap area with adjacent cells on system performance are studied. All free channels are allocated to data users dynamically. For those data users using more than one channel, channels would be de-allocated for new requests, voice or data. Buffers are provided for voice calls to reduce the voice blocking probability caused by data packets contention. Handoff calls are given priority to be queued in the front of the buffer instead of providing guard channels to reduce their dropping probability. Meanwhile, the reneging time for new calls and the handoff dwell time for handoff calls are considered in our analysis to obtain an appropriate amount of buffer to voice. To compensate the blocking probability in data, guard channels are provided for data traffic. Numerical results show that the dynamic multi-channel scheme with possible de-allocation, compared with the single channel scheme, can enhance data traffic performance significantly in terms of the mean transmission time and blocking probability. A system providing an appropriate amount of buffer to voice traffic and giving priority to queued handoff calls can indeed reduce new call blocking probability and handoff call dropping probability. In addition, the proposed scheme can reduce the incomplete transmission probability of data packets.

This paper presents a network architecture with a dual interface IP handoff technique that allows smooth node mobility without using any intermediate proxy. The proposed architecture is suitable for low bit-rate time sensitive real time applications, where payload tends to be short and packet header overhead is particularly significant. Connections are established as per permanent addresses of the nodes but are carried on by the IP layer according to the temporary addresses by address translation within the end hosts. The mapping information is maintained by database servers, which can be placed in the Internet in a distributed manner. We describe the architecture and show its mobile capabilities by prototype implementation and performance evaluation. Furthermore a dual-interface handoff suitable to the proposed architecture is also introduced. Preliminary results show that the proposed architecture has significantly low overheads. It is compatible with the existing infrastructure and works fine in both IPv4 and IPv6 environments. Analysis also shows that with dual-interface handoff it is possible to achieve seamless handoff without any packet loss by exploiting overlapping coverage area and speed of the mobile node. Handoff latency is reduced significantly as compare to MIPv6. We believe that with more powerful network interface card drivers our concept of dual interface handoff can be realized.

In order to benefit from the advantages of soft handoff (SHO), it is important that the SHO parameters (the SHO thresholds; T_ADD and T_DROP are well assigned. T_ADD is the threshold used for triggering a pilot with high strength to be added to the Active Set (AS) list. The AS means the pilots associated with the forward traffic channels assigned to mobile station. In contrast, T_DROP is the threshold used for triggering a pilot with low strength to be dropped from the AS list. This paper analyzes the effects of varying SHO thresholds in a cellular code division multiple access (CDMA) system on the blocking probability based on traffic load and geometrical distances in hexagonal layout of base stations (BSs). In addition, the previously proposed traffic load equation is applied to the proposed SHO model for balancing the numbers of new and handoff calls on the forward link capacity in case of uniform traffic load. The results show that the blocking probability is more sensitive to T_DROP than to T_ADD variations.

The handoff in Mobile IP networks causes packet sequence disruption during a packet forwarding procedure and may result in performance degradation in higher layer protocols. We investigate the impact of handoff in the Mobile IPv6 networks, where an optimized routing with the smooth handoff is adopted. The impact on the packet sequence is measured by an 'unstable time period (UTP)' and a 'silence time period (STP).' The UTP explains the time duration of out-of-sequence packets while the STP reflects the blackout duration of a mobile node after the initiation of handoff procedure. With the analysis on the UTP and STP, the total transient time period (denoted as handoff time period or HTP) after the handoff initiation can be estimated. In our previous work, focusing on the UTP, the packet flow sequence under the smooth handoff is analyzed for the Mobile IPv4 networks. The proposed queuing-based analysis is extended in this work for the Mobile IPv6 networks. That is, several modifications are made to conform to Mobile IPv6 and at the same time the queuing analysis itself is improved to better model the handoff procedure. The numerical results show that the queuing delay for the handoff packets (affected by background traffic) and the involved link (or route) capacities affect the estimated UTP, STP, and HTP. In addition, two schemes such as priority queuing and buffered packet forwarding are introduced to reduce the transient period and the improvements are analyzed for comparison.

Various network resources, including wireless access services and multimedia appliances (device) are expected to be available in ubiquitous computing environments. Since resource availability can change when a user migrates from one place to another, functions to monitor the availability of resources in use and, if necessary, switch from obsolete resources to new ones are necessary for continuous service provision. This paper proposes adaptive terminal middleware called AMID that performs policy-based dynamic resource selection and host-based session management to ease network administrative tasks, and hide session failures and resource changes from applications and a user. AMID supports two kinds of mobility; session maintenance on vertical handoff and device handoff (service mobility). By AMID, a mobile host keeps entire handoff control and session state to eliminate the need for network-layer or intermediate-node mobility support, and mitigate responsibility of devices for session management. AMID realizes a Reliable Virtual Socket (RVS), on top of real sockets, which employs a seamless session handoff mechanism for resource changes, and a reliable session resume mechanism against unplanned disconnection of a wireless link. It achieves seamless session handoff through a proactive soft handoff method; to conceal setup and signaling latency, it initiates setup procedures with neighbor resources in advance of actual handoff and utilizes multiple wireless interfaces and devices redundantly. We implemented AMID and a follow-me audio application on top of it to evaluate the performance. Redirection of audio streams from built-in speakers to external ones, and handoff between 802.11b and Cellular are autonomously performed when a user migrates in the house. We confirmed that AMID achieved reliable session maintenance against wireless link failure, concealed latency of handoff management, and prevented packet loss during handoff.

The QoS(Quality of Service) guarantee mechanism is one of critical issues in the wireless network. Real-time applications like VoIP(Voice over IP) in All-IP networks need smooth handoffs in order to minimize or eliminate datagram loss as a Mobile Host(MH) transitions between network links. In this paper, we design a new DB(Dynamic Buffering) mechanism for IPv6 by which an MH can request that the router on its current subnet buffers packets on its behalf while the MH completes registration procedures with the router of a new subnet. Performance results show that our proposed buffering scheme with a dynamic buffer space allocation is quite appropriate for mobile Internet, or the All-IP environment in terms of the datagram loss rate and average waiting time.

A Virtual Cellular Network (VCN) is a wireless cellular network wherein a single Mobile Station (MS) can communicate simultaneously with more than one Base Station (BS). In this paper, we analyze handoff for two kinds of VCN: a 'Distributed Resource-control VCN' (DR-VCN) and a 'Centralized Resource-control VCN' (CR-VCN). A VCN can take advantage of the fact that the same data is received by multiple base stations. The DR-VCN is a system in which every BS controls its own channels, while the CR-VCN is a system wherein a central station controls all system channels. Results from analysis and simulation show that both the new call drop rate and handoff refusal rate of the CR-VCN are much lower than those of the DR-VCN.

This letter addresses the QoS issue of a RSVP flow during handoff events. For the QoS guarantee of real-time applications with RSVP reservation in All-IP wireless networks, mechanisms are required to minimize the resource reservation path changes and the packet loss resulting from handoff events. If the new RSVP reservation is made along the path via a new base station (BS) in advance for soft handoff, on-going RSVP flow session can be kept with the guaranteed QoS. Therefore, we propose a seamless switching scheme of RSVP branch path for soft handoff. We also show that this scheme could provide the QoS guarantee by adaptively adjusting the pilot signal threshold values for soft handoff.

In this paper, we propose two channel allocation schemes for supporting voice and multimedia traffic in hierarchical cellular systems. They are guaranteed to satisfy the required quality of service for multimedia traffic in accordance with their characteristics such as a mobile velocity for voice calls and a delay tolerance for multimedia calls. In the first, only slow-speed voice calls are allowed to overflow from macrocell to microcell and only adaptive multimedia calls can overflow from microcell to macrocell after reducing their bandwidth to the minimum channel bandwidth. In the second, in addition to the first scheme, non-adaptive multimedia calls can occupy the required channel bandwidth through reducing the channel bandwidth of adaptive multimedia calls. The proposed schemes are analyzed using the 2-dimensional Markov model. Through computer simulations, it is shown that the proposed schemes yield a significant improvement in terms of the forced termination probability of handoff calls. In particular, the second decreases the blocking probability of new calls as well as the forced termination probability of handoff calls.

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.

Traditional soft-handoff algorithms are based on the static threshold handoff algorithm recommended in the IS-95 standard. They are characterized by two parameters, an add threshold Tadd and a drop threshold Tdrop. These two parameters are assumed to have some constant values irrespective of the received signal strength of the pilot in the active set. To improve the performance of the soft-handoff, a dynamic threshold concept was proposed in previous work, where Tadd and Tdrop are dynamically determined according to the received signal strength of the pilot channel in the active set. In this study, previous work of the dynamic threshold algorithm is extended by including additional handoff criteria based on absolute signal strength and/or drop timer. Some functional forms with two new parameters, called boundary thresholds, and slope constant are proposed to dynamically determine Tadd and Tdrop. The dynamic threshold algorithms are compared with static ones in four different cases. Computer simulations show that the dynamic threshold algorithms outperform the static algorithms. We can see that the performance improvements differ from case to case. For example, when a pure dynamic threshold algorithm is compared with a pure static one (case 1), the decrease in the number of active set is about 13.7%. When the absolute threshold and the drop timer are also included in the handoff decision criteria (case 4), however, the decrease is only about 6.2%.

In this paper, we propose a new channel allocation scheme, which is designed to efficiently carry out handoffs in wireless networks. Our scheme is based on conventional handoff methods, which include the channel reservation, carrying and sub-rating methods to assign channels for handoffs. First, we reserve a number of channels only for handoffs. Second, if there is no available channel in the next cell during handoff, the mobile station is allowed to carry a movable channel into the next cell. Finally, in order to avoid co-channel interference due to channel mobility, we adopt the sub-rating method for its carried channel. We evaluated our scheme using both Markov analysis and computer simulation. The analytical and simulation results indicate that our scheme may offer better performance than conventional handoff schemes in terms of handoff blocking probability and channel utilization.