This paper investigates the performance of an integrated voice and data transmission protocol that can be used in mobile communication networks, e.g. mobile cellular and LEOGEO satellite networks. Voice and data are concentrated at place such as base station in a cellular network. Time sensitive voices are supported by the concentrator in the manner of TDMA. Loss sensitive data are collected, stored, and transmitted using idle TDMA slots. Data users transmit data to the concentration point using ALOHA protocol. Characterization of data arrivals to the concentrator is done by the method of moment matching. The emphasis of this study is on the data performance in terms of packet loss rate, average buffer occupancy, and mean packet waiting time. It is demonstrated through numerical examples that a buffer of reasonable size is good enough to offer satisfactory performance. The analysis is also validated by computer simulations.

Currently, the issues in Quality of Service, fairness and pricing strategies should have expedited the emergence of service differentiation in wireless access networks. In this paper, we propose a novel scheduling algorithm, called the Wireless Differentiated Fair Queueing (WDFQ) algorithm, to accommodate such need by providing delay/jitter controls, and fair residual bandwidth sharing for real-time and non-real-time traffic streams simultaneously. We show that the WDFQ scheme can achieve excellent performance, including timely delivery of real-time traffic, virtually error-free transmission of non-real-time traffic, and fair usage of channel bandwidth among remote stations. In addition, the location-dependent channel error property, as appeared in most wireless networks, are considered in the model and the temporary short error bursts are compensated by credits of bandwidth. The simulation results suggest that the length of retransmission period should be adapted to the error length to achieve good performance and maintain low implementation complexity.

The performance of a GMPLS switching architecture with or without the flush capability is studied. For this switching architecture, we propose a queueing model that includes the control plane, the switch buffer mechanism, and the flush mechanism. The flush capability is included to reduce the out-of-sequence problem due to dynamic path changes. The behavior of aggregated streams, the label-setup and release policies, and the mechanisms for efficient resource allocation are all covered. With the proposed model, one can select appropriate parameters for the label-setup policy and the label-release policy to match the traffic load and network environment. Key performance metrics, such as the label-setup rate, the switching ratio, the bandwidth utilization of the label switched path, the average delay, and the average packet loss rate, can all be evaluated via this mathematical model. Numerical results and simulations are used to verify the accuracy of the proposed queueing model. Furthermore, we adopt trace simulation to certify the applicability the proposed model. The trade-off among these performance metrics can be observed as well.

This study presents a novel window-based token protocol which supports two priority levels of ATM-like bursty traffic. The proposed protocol ensures that packets arriving during a given window period are always transmitted prior to those arriving during subsequent window periods. During which periods, high priority (delay-sensitive) packets are served by a nonpreemptive priority service discipline. Under this window-based token protocol, the delay bounds for packets of both priority levels are derived when the traffic streams satisfy the traffic descriptors. Hence, a call admission control policy based on delay bounds can be employed to guarantee QoS parameters such as the packet delay and packet loss-free requirement for each traffic stream. The protocol proposed herein can achieve tighter delay bounds than the IEEE 802. 5 token protocol and the timed token protocol used in FDDI, particularly for high priority bursty traffic, thereby making it highly attractive for supporting not only real-time multimedia VBR or CBR services such as in ATM networks but also ATM encapsulation such as in Cells In Frames (CIF) protocol.

The proposed GCRA (Generic Cell Rate Algorithm) traffic shaper consists of a regulator and a scheduler. It can shape multiple incoming VBR (Variable Bit Rate) cell streams simultaneously to be strictly conforming according to the GCRA algorithm when the cells depart for the ATM output link. The impact of cell emission conflicts is considered and resolved by using an EDD (Earliest-Due-Date) scheduler and a feedback signal from the scheduler to the regulator. The call admission control condition and the cell delay bound are derived. Simulation results demonstrate that the output cell streams of the proposed GCRA traffic shaper do not contain any non-conforming cells and the scheduler queue size is significantly reduced. Meanwhile, the delay performance is almost not affected by the use of the feedback mechanism.

In the current ATM AAL5 implementation, even a single cell loss event can lead to the corruption of one whole packet. Hence, it has been observed that the throughput of upper layer protocol may easily collapse on a congested ATM network. In this paper, we propose a buffer management method called Age Priority Packet Discarding (APPD) scheme to be used along with two other schemes: the Early Packet Discarding (EPD) and the Partial Packet Discarding (PPD) schemes. After describing the operations and the pseudo code of the proposed APPD scheme and how it operates with the EPD/PPD schemes, the packet level QoS of APPD and its extended versions are derived analytically under homogeneous ON-OFF source model. Numerical results obtained via analytical approach suggest that the proposed APPD scheme can more effectively and fairly reduce packet loss probability than other schemes.

Power consumption due to transmissions in base stations (BSs) has been a major contributor to communication-related CO2 emissions. A power optimization model is developed in this study with respect to radio resource allocation and activation in a multiple Component Carrier (CC) environment. We formulate and solve the power-minimization problem of the BS transceivers for multiple-CC networks with carrier aggregation, while maintaining the overall system and respective users' utilities above minimum levels. The optimized power consumption based on this model can be viewed as a lower bound of that of other algorithms employed in practice. A suboptimal scheme with low computation complexity is proposed. Numerical results show that the power consumption of our scheme is much better than that of the conventional one in which all CCs are always active, if both schemes maintain the same required utilities.

Although the performance degradation for TCP/IP over plain ATM during congestion can be reduced if switch buffer management techniques such as Early Packet Discarding (EPD) and Partial Packet Discarding (PPD) schemes are employed. However, we show via simulation that fairness among connections remains a problem. For example, the fairness among packets of different length is a well known unsolved issue for EPD. To improve fairness of TCP and UDP over an Internet backbone, we propose a new technique called the Age Priority Packet Discarding (APPD) scheme to be used along with EPD and PPD. We employ two simulation scenarios to examine the performance of APPD; the MPEG-I video over UDP protocol and the FTP over TCP protocol. The simulation shows that with APPD combined with EPD and PPD, fairness can be well maintained against different packet length, order of connections, as well as different propagation delays. In addition, packet loss probability can be reduced with APPD in all scenarios, and the improvement is especially significant for video over UDP protocol. Finally we discuss the hardware implementation technique of the APPD scheme.

In this paper we propose an MPLS switch model with a GPS-based scheduler to provide guaranteed QoS in the Internet. Important applications of this model include IP VPN (MPLS-VPN) services and MPLS networks. The considered scenarios include cases with sufficient labels or with limited number of labels under competing traffic streams. Due to tractability issues in direct analysis, we use a worst case equivalent system model to obtain the theoretical upper bounds of delays and backlogs of the original system model. Simulation results are used to verify the analytical results. We can obtain the worst case delay bounds and the maximum backlogs. Furthermore, the trade-off among the number of labels, the worst delay bounds, and the maximum backlogs can be observed. With our model, one can determine the number of required labels according to the allowable delay and the available buffer size.

A closed-loop queueing model of flow-based label switches, supporting label reservation protocols of different label-setup and release policies, is presented. This model can emulate the behavior of TCP under the label switch when the maximum window size has been achieved and the packet loss rate is negligible. The label-setup policy is that the IP controller does not start to set up a label until the accumulated packets of the same flow in the switch buffer have exceeded a triggering threshold. Meanwhile, the reserved bandwidth is released when the flow is detected idle and the label-release timer has expired. This policy can achieve higher channel utilization with minimal label processing overhead in spite of suffering from certain delay penalty. To avoid unnecessary TCP timeout or large packet delay under such policy, we also introduce a label-setup timer. Norton's theorem is applied to obtain approximate solutions of this queueing model. Although the analytical method is an approximate one, the simulation results show that the accuracy is high and this model can clearly illustrate how the label-setup and the lable-release timer affect the system performance. Besides, one can observe the trade-off between the throughput and the channel utilization.

In this paper we propose a close-loop queueing model of MPLS switch under different label-setup and release policies, supporting both traffic-driven and topology-driven connection setup procedures. This model can emulate the behavior of TCP under the MPLS switch when the maximum window size is sustained and the packet loss rate is negligible. From the proposed flow-based MPLS switch model, one can clearly observe the competition of multiple IP flow for limited number of labels, and how the label-setup policy and the label-release policy affect the system performance. We find that Norton's theorem can be applied to solve this sophisticated queueing model. Therefore, with very limited computational complexity with respect to the number of IP flows or labels, the proposed mathematical model and the approximation of label competition can be used to obtain the desired performance metrics, such as the throughput, the label-setup rate, and the channel utilization. Finally, the trade-off among performance metrics can be observed as well.

This work addresses the problem of a fast packet retransmission scheme intended for transporting delay-sensitive flows in a Cooperative Diversity (CD) environment. This cooperative fast retransmission scheme exploits the advantages of the CD environment and hybrid Automatic-Repeat-reQuest (ARQ), while allowing retransmission just one time via a cooperating user (i.e., partner) or via both the sender and the partner simultaneously. Complementary link packets are used for the retransmission whose policy can be adjusted on the basis of the qualities of channels among the sender, the partner and the receiver, as well as the application layer protocol data unit size, using the application throughput as the objective. For this scheme, we first derive the application throughput formulas which are then verified via simulations. Next, the CD-based optimized fast retransmission scheme is shown able to achieve better effective throughput (goodput) than other CD-based or non-CD-based ARQ schemes in various Nakagami-m slow-fading environments. As a result, the proposed scheme should be an excellent fast retransmission mechanism for real-time multimedia transport in many CD environments.

We proposed an improved Hierarchical Packet Fair Queueing (H-PFQ) mechanism, using ACK Spacing, for efficient bandwidth management of TCP traffic over Internet. According to the pre-determined bandwidth sharing and the class hierarchy of all TCP sessions, we design an algorithm to calculate the required time intervals between consecutive ACK packets of each TCP session to avoid packet drops due to buffer overflow. We demonstrated via computer simulations that the proposed improvement techniques may result in much better performance than merely original H-PFQ mechanism used in the forward direction in the sense that not only effective throughput of the bottleneck link is improved but also the fairness among TCP sessions can be maintained.

For direct sequence spread spectrum systems, the performance of PN sequence acquisition can be significantly affected if data modulation is present. However, the data modulation often exists during the reacquisition of a PCS radio channel. This study proposes and analyzes two shemes which are designed to improve acquisition process for PN sequence under data modulation. Both designs are based upon a PN acquisition receiver with parallel I-Q matched filters. The first scheme employs a serial search strategy with verification mode. The second scheme, which is still based upon the same parallel acquisition receiver, employs the parallel search strategy. We show that the second scheme is capable of providing faster acquisition under data modulation than the first serial search scheme using the same number of I-Q matched filter. We believe it should become a very good alternative for the acquisition of data modulated PN sequences in personal communications.