According to Wu, system performance and capacity degrade due to the co-channel interference between microcells (i. e. hot-spot) and macrocells in a hierarchical structure if the same channel is used. In order to avoid co-channel interference, different channels are assigned for macrocells and microcells. That means multi channels exist in the hierarchical structure. However, hard handoff is necessary for a mobile station (MS) across the boundary of microcell and macrocell because both of them use different channels. Based on a hot-spot overlaying environment, this paper proposes multi-channel access schemes to avoid serious interference between hot-spot and macrocell and thus the capacities are increased. Besides, the same channel can be used in macrocell and hot-spot simultaneously. It makes soft handoff workable and improves the system performance.

The potential network architecture of the emerging carrier class VoIP (Voice over IP) technology for NGN (Next Generation Networks) adopts distributed control architecture to take full advantage of scalability, reliability, flexibility, and interoperability. However, the design of distributed control architecture in the carrier class VoIP network is the state-of-the-art in decentralization and distribution of control. Different configurations of system elements, control scheme of inter system elements communications, signaling protocol, functional partitioning, and scheduling of jobs in call control processing may affect the system performance and QoS (Quality of Service) of MGC (Media Gateway Controller) in carrier class VoIP network. Hence, the modeling of distributed control architecture and its performance analysis are essential issues whenever optimum control architecture has to be determined to meet design requirements. Based on these reasons, this paper proposes several potential network architectures and focuses on the performance study of distributed control architecture in carrier class VoIP network. The SIGTRAN-based distributed control architecture model and the MGCP/MEGACO-based distributed control architecture model are presented. Then, we analyze the SIGTRAN-based distributed control architecture model between MGC and SG (Signaling Gateway) using WRR (Weighted Round Robin) and WF2Q (Worst-case Fair Weighted Fair Queueing) scheduling algorithms respectively. And, we analyze the MGCP/MEGACO-based distributed control architecture model between MGC and MG (Media Gateway) using M/G/1 gating service queueing model. Consequently, the results of performance analysis can be used to evaluate whether the performance of distributed control architecture model can meet the requirement of planning and design for carrier class VoIP network deployment.

The CDMA system can provide more capacity than other systems and the hierarchical layer of cells is required for system design to provide a balance between maximizing the number of users perfrequency band and minimizing the network control associated with handoff. However, the co-channel interference from microcell to macrocell and vice versa in such a two-tier structure is different from that in a homogeneous structure. In order to avoid the serious interference, different RF channels should be used in microcell and macrocell in hierarchical structure. The efficient usage of multi-channels for macrocell and microcell is of primary concern herein. In this study, we investigate the channel segregation in a two-tier cellular system. Moreover, we intentionally arrange the procedures for the MS in macrocell and microcell to choose the channel. The macrocell's (resp., microcell's ) channels to be accessed are sorted into three priority groups. In order to justify the merits of the proposed channel segregation method, we define the following three performance measures including capacity gain, response to the variation of traffic loading and system stability. Under the condition of steady-state traffic load, capacity gain is 10% on the average. If the traffic load vary, the system can respond quickly and retrieve the borrowed channels with 2tp time interval as long as appropriate system parameters are chosen.

Due to the Cell Delay Variation (CDV) at User Network Interface (UNI), it is very hard for an ATM network to perform Usage Parameter Control (UPC), which is an important job for congestion control. Based on the Generic Cell Rate Algorithm (GCRA), ATM Forum has proposed a procedure to perform the UPC. However, the severe problem is that a user has to specify the CDV Tolerance at the UNI by itself. Such a nearly unreachable constraint makes the GCRA unsuitable for UPC. In this paper, we point out that the CDV comprises two parts in which the customer and a network provider should be responsible. Thus, we propose a concept of Innocent Public Network and an Agent Protocol to realize the principle and facilitate UPC. In addition, a shaper is suggested for the customer to employ so as to prevent its performance degradation. In the proposed system, the network is no longer suffered from CDV at the UNI and the UPC can be easily preformed.

In ATM networks, two main functions for achieving traffic control and congestion control are Call Admission Control (CAC) and Usage Parameter Control (UPC). Among various UPC schemes, Leaky Bucket is a popular one. In this paper, we study the characteristics of the system in which every traffic source is regulated by an enhanced leaky-bucket before entering the multiplexer at the edge node of the ATM network. In addition to the factor of mean cell rate, peak cell rate is also taken into consideration. Based on the criteria of average waiting time at the multiplexer, we derive the performance bounds expressed as the functions of the LB parameters and numbers of connections.

Leaky Bucket based traffic parameters are widely used for traffic declaration and enforcing in an ATM network. In this paper, we investigate the characteristics of the system that every traffic source is policed by a dual leaky bucket before entering the network. In addition to mean cell rate, peak cell rate of traffic is also taken into consideration. We find the worst output pattern from the dual leaky bucket and derive the performance bound of maximum cell loss ratio encountered in the multiplexer. It is obtained as every source transmits cells according to the criteria for extreme synchronous transmission in a coincident token-generating condition.

The next generation of wireless networks must provide sufficient resources to support a broader range of services beyond the traditional voice-only services provided in current wireless systems. For this, packet scheduling is the most critical function involved in the provision of individual bandwidth, and delay guarantee to meet the required qualities of service (QoS) for the switched sessions. In this paper we introduce the concept of system capacity in the multi-code CDMA system and evaluate the system performance constrained by the required SNRth and other QoS factors. The CAC algorithm is proposed to administer the requests. Moreover, the enforcer, scheduler and shaper are proposed to maintain the required QoS. The system performance including blocking probability, mean delay and bandwidth utilization in the BS egress are evaluated and compared under different traffic loadings.

This paper presents the performance modeling application of SIP-T (Session Initiation Protocol for Telephones) signaling system based on two-class priority queueing process in carrier class VoIP (Voice over IP) network. The SIP-T signaling system defined in IETF (Internet Engineering Task Force) is a mechanism that uses SIP (Session Initiation Protocol) to facilitate the interconnection of existing PSTN (Public Switched Telephone Network) with carrier class VoIP network. One of the greatest challenges in the migration from PSTN toward NGN (Next Generation Networks) is to build a carrier class VoIP network that preserves the ubiquity, quality, and reliability of PSTN services while allowing the greatest flexibility for use of new VoIP technology. Based on IETF, the SIP-T signaling system not only promises scalability, flexibility, and interoperability with PSTN but also provides call control function of MGC (Media Gateway Controller) to set up, tear down, and manage VoIP calls in carrier class VoIP network. This paper presents the two class priority queueing model, performance analysis, and simulation of SIP-T signaling system in carrier class VoIP network focused on toll by-pass or tandem by-pass of PSTN. In this paper, we analyze the average queueing length, the mean of queueing delay, and the variance of queueing delay of SIP-T signaling system that are the major performance evaluation parameters for improving QoS (Quality of Service) and system performance of MGC in carrier class VoIP network. A mathematical model of the M/G/1 queue with two-class non-preemptive priority assignment is proposed to represent SIP-T signaling system. Then, the formulae of average queueing length, queueing delay, and delay variation for the non-preemptive priority queue are expressed respectively. Several significant numerical examples of average queueing length, queueing delay, and delay variation are presented as well. Finally, the two-class priority queueing model and performance analysis of SIP-T signaling system are shown the accuracy and robustness after the comparison between theoretical estimates and simulation results.

The emerging Wireless Access in Vehicular Environment (WAVE) architecture, which aims to provide critical traffic information and Internet services, has recently been standardized in the IEEE 802.11p specification. A typical WAVE network consists of one road-side-unit (RSU) and one or more on-board-units (OBUs), wherein the RSU supports one control channel (CCH) and one or more service channels (SCH) for the OBUs to access. Generally, an OBU is equipped with a single transceiver and needs to periodically switch between the CCH and one of the SCHs in order to receive emergency messages and service information from the CCH and to deliver Internet traffic over an SCH. Synchronizing all OBUs to alternatively access the CCH and SCHs is estimated to waste as much as 50% of the channel's resources. To improve efficiency, we propose an innovative scheme, namely coordinated interleaving access (CIA) scheme, which optimizes the SCH throughput by smartly grouping the OBUs to let them access the CCH and SCHs in an interleaved and parallel manner. To further the capability of CIA scheme, an enhanced version is also proposed to handle the case where OBUs with multiple transceivers. Performance analysis and evaluation indicates that the proposed CIA scheme achieves a significant improvement in resource. Thus it can be advantageous to adapt it into the IEEE 802.11p protocol for its adoption in multi-channel wireless vehicular networks.

Emerging requirement for higher rate data services and better spectrum efficiency is the main driving force identified for the third generation mobile radio systems. Moreover, it needs the capability of providing predictable qualities of service (QoS) for different applications. To maintain different QoS requirements, mechanisms such as call admission control (CAC) and load control, etc. are needed to achieve the required services. In this paper, we propose a CAC algorithm based on channel assignment in a multi-chip rate direct-sequence CDMA (MCR-DS-CDMA) cellular system supporting multi-rate services. Five multi-MBC (mapping of information bit rates to chip rates) channel assignment schemes and corresponding channel selection rules are proposed herein. Computer simulation, where multimedia applications are considered, is used to evaluate the system performance (e.g., blocking probability and system capacity) with different channel assignment schemes. Numerical results demonstrate that scheme 5 (i.e., Minimum-influence scheme) performs better because it provides the highest system capacity and least blocking probability.

In this paper, we study the performance of quality-based voice/data CDMA system where new and handoff traffic are considered. A call request for handoff data queues up if the signal-to-interference ratio exceeds a predefined threshold while priority is given to handoff voice calls by reserving some channels exclusively for them. The transmission rate of data users may vary according to measured SIR value. Important performance measures of the system such as blocking probability and system capacity for voice or data calls for proposed schemes are presented and compared.

In this paper we propose an effective ratebased virtual clock (ERVC) scheduling algorithm which is applied to the switching nodes in the connection-oriented high-speed networks. It is based on the effective rate which has a value between the average and peak transmission rates. The algorithm is simple but overcomes the defects of original virtual-clock algorithm. Performance results demonstrate the effectiveness of the ERVC algorithm in comparison with other methods.

Wireless access in the vehicular environment (WAVE) architecture of intelligent transportation system (ITS) has been standardized in the IEEE 802.11p specification and it is going to be widely deployed in many roadway environments in order to provide prompt emergency information and internet services. A typical WAVE network consists of a number of WAVE devices, in which one is the road-side-unit (RSU) and the others are on-board-units (OBUs), and supports one control channel (CCH) and one or more service channels (SCH) for OBU access. The CCH is used to transport the emergency messages and service information of SCHs and the SCHs could be used to carry internet traffic and non-critical safety traffic of OBUs. However, the IEEE 802.11p contention-based medium access control protocol would suffer degraded transmission efficiency if the number of OBUs contending on an SCH is large. Moreover, synchronizing all WAVE devices to periodically and equally access the CCH and an SCH will waste as much as 50% of the channel resources of the SCH [1]. As a solution, we propose an efficiency-improvement scheme, namely the agent-based coordination (ABC) scheme, which improves the SCH throughput by means of electing one OBU to be the agent to schedule the other OBUs so that they obtain the access opportunities on one SCH and access the other SCH served by RSU in a contention-free manner. Based on the ABC scheme, three different scheduling and/or relaying strategies are further proposed and compared. Numerical results and simulation results confirm that the proposed ABC scheme significantly promotes the standard transmission efficiency.

In the original shared buffer memory switch, there exists inherent unfairness since multicast cells always own higher priority to be served such that unicast cells must endure longer delay time. We propose a multicasting balancing method to overcome this unfair phenomenon in which the service order for two kinds of cells is decided by a ratio of unicast queue length and multicast queue length for each output port. Performance results are provided to verify the effectiveness.

In ITU-T Recommendation I.371, the Generic Cell Rate Algorithm (GCRA) is used to define Peak Cell Rate for the ATM network. It is further applied by the ATM Forum '93 to define Sustainable Cell Rate and Burst Tolerance so as to facilitate Usage Parameter Control and Network Parameter Control. To judge the validity of a cell according to declared GCRA parameters, the enforcer must read the clock time when the cell arrives. However, the clock of the enforcer would roll over frequently and accordingly the judgment would be incorrect. On the other hand, for a shaper in a customer premise node to dispatch cells conforming to the declared GCRA parameters, the clock would also roll over and the cell would not be dispatched correctly. To overcome the problems induced by clock roll-over, based on "time difference" concept, we propose two modified GCRA's for the enforcer and shaper, respectively. According to the proposed algorithms, we design a feasible architecture for a multi-connection shaper and simplify it for an enforcer. They are proven to perform well in spite of the inherent clock roll-over characteristics. By simulation, we evaluate the delay in the shaper and the loss in the enforcer. The features of the architectures are also discussed.

In this paper we propose a congestion control method for interconnecting connectionless MANs with ATM networks which works at the gateway of DQDB. Since connectionless traffic belong to loss sensitive data, they should experience small cell loss rate. Due to the function of congestion control in the gateway, we can prevent the network from overload which not only introduces serious cell loss at remote destination gateway but also lots of undesirable retransmissions and time delay in the ATM network. It neither needs to modify the slot format of DQDB nor to increase the overhead so the implementation is simple and cost effective. Performance results are also provided to verify the effectiveness.

The FDDI-II is a high speed and flexible backbone LAN. It can divide the capacity into one packet-switched channel with flexible bandwidth and up to 16 isochronous channels which may be allocated for a variety of real-time services such as video and voice. How to allocate and maintain isochronous bandwidth is an important issue for supporting good services to users. The FDDI-II standard proposed a centralized scheme to achieve this goal. In this paper, we propose a new scheme in a distributed fashion for the management of isochronous bandwidth. Based on our scheme, the network can support various services in a more efficient way.

In ATM networks, a usage parameter control (UPC) strategy must regulate incoming traffic according to the characteristics of the sources declared at call set-up. Among various UPC schemes, Leaky Bucket is a conventionally method having been discussed extensively. This paper examines the characteristics of the multiplexer with a sufficient buffer in which cell arrivals are policed by Enhanced Leaky-Bucket (ELB) before entering the system. In addition to the factor of mean rate, peak rate and cell delay variation (CDV) are also considered for each ELB. We find out the worst output pattern from the ELB and derive the upper bound on average waiting time as a function of the ELB parameters.

The Direct-Sequence Code Division Multiple Access (DS-CDMA) is one of the most likely candidates for the 3rd generation mobile communication systems. On the other hand, a network supporting integrated data transfer and providing predictable qualities of service (QoS) to such application is an attractive issue. This paper proposes several schemes based on the majority rule concerned with resource reservation to investigate the performance of a CDMA cellular system. The admission control is based on the residue capacity and required signal-to-noise ratio (SNR) for multimedia traffic including voice, data and video. The system performance is explored with variable spreading gain and multi-code CDMA systems under the proposed schemes. Moreover, several system parameters including the handoff probability (lifetime in a cell) and the traffic load are investigated to study the impact to system performance. The results show the trade-off between the blocking probability of new calls and dropping probability of handoff calls under diversified situations. However, it is inefficient under resource reservation due to unused capacity. In this paper, we also propose a resource-borrowing scheme to improve system performance. Simulation results show that our proposed algorithm really improves the system performance in a multimedia CDMA cellular system with resource reservation.

This paper presents the performance modeling, analysis, and simulation of SIP-T (Session Initiation Protocol for Telephones) signaling system in carrier class packet telephony network for NGN (Next Generation Networks). Until recently, fone of the greatest challenges in the migration from existing PSTN (Public Switched Telephone Network) toward NGN is to build a carrier class packet telephony network that preserves the ubiquity, quality, and reliability of PSTN services while allowing the greatest flexibility for use of new packet telephony technology. The SIP-T signaling system defined in IETF (Internet Engineering Task Force) draft is a mechanism that uses SIP (Session Initiation Protocol) to facilitate the interconnection of PSTN with carrier class packet telephony network. Based on IETF, the SIP-T signaling system not only promises scalability, flexibility, and interoperability with PSTN but also provides call control function of MGC (Media Gateway Controller) to set up, tear down, and manage VoIP (Voice over IP) calls in carrier class packet telephony network. In this paper, we derive the buffer size, the mean of queueing delay, and the variance of queueing delay of SIP-T signaling system that are the major performance evaluation parameters for improving QoS (Quality of Service) and system performance of MGC in carrier class packet telephony network focused on toll by-pass or tandem by-pass of PSTN. First, we assume a mathematical model of the M/G/1 queue with non-preemptive priority assignment to represent SIP-T signaling system. Second, we derive the formulas of buffer size, queueing delay, and delay variation for the non-preemptive priority queue by queueing theory respectively. Besides, some numerical examples of buffer size, queueing delay, and delay variation are presented as well. Finally, the theoretical estimates are shown to be in excellent consistence with simulation results.