During devastating natural disasters, numerous people want to make calls to check on their families and friends in the stricken areas, but many call attempts on mobile cellular systems are blocked due to limited radio frequency resources. To reduce call blocking and enable as many people as possible to access mobile cellular systems, placing a limit on the holding time for each call has been studied [1],[2]. However, during a catastrophe, emergency calls, e.g., calls to fire, ambulance, or police services are also highly likely to increase and it is important that the holding time for these calls is not limited. A method of limiting call holding time to make provision for emergency calls while considering the needs of ordinary callers is proposed. In this method, called the HTL-E method, all calls are classified as emergency calls or other according to the numbers that are dialed or the terminal numbers that are given in advance to the particular terminals making emergency calls, and only the holding time of other calls is limited. The performance characteristics of the HTL-E method were evaluated using computer simulations. The results showed that it reduced the rates of blocking and forced call termination at handover considerably, without reducing the holding time for emergency calls. The blocking rate was almost equal for emergency and other calls. In addition, the HTL-E method handles fluctuations in the demand for emergency calls flexibly. A simple method of estimating the holding-time limit for other calls, which reduces the blocking rate for emergency and other calls to the normal rate for periods of increased call demand is also presented. The calculated results produced by this method agreed well with the simulation results.

It has been a very important issue to evaluate the performance of transmission control protocol (TCP), and the importance is still growing up because TCP will be deployed more widely in future wireless as well as wireline networks. It is also the reason why there have been a lot of efforts to analyze TCP performance more accurately. Most of these works are focusing on overall TCP end-to-end throughput that is defined as the number of bytes transmitted for a given time period. Even though each TCP's fast recovery strategy should be considered in computation of the exact time period, it has not been considered sufficiently in the existing models. That is, for more detailed performance analysis of a TCP implementation, the fast recovery latency during which lost packets are retransmitted should be considered with its relevant strategy. In this paper, we extend the existing models in order to capture TCP's loss recovery behaviors in detail. On the basis of the model, the loss recovery latency of three TCP implementations can be derived with considering the number of retransmitted packets. In particular, the proposed model differentiates the loss recovery performance of TCP using selective acknowledgement (SACK) option from TCP NewReno. We also verify that the proposed model reflects the precise latency of each TCP's loss recovery by simulations.

Congestion control in the Internet consists of two main components: the TCP Additive-Increase Multiplicative-Decrease (AIMD) mechanism on sending windows implemented by end-users, and the Active Queue Management (AQM) scheme implemented in the routers which improves the effectiveness of congestion control. TCP connection is regarded as a feedback control system. Comparably, AQM is classified as a flow controller. There are several kinds of time delays in the network, such as propagation delay, queuing delay in the buffer of the router, etc. The time delays cause degradation of performance and instability of the network. A Smith Predictor is commonly used in feedback control of plants with significant time delays to implement effective compensation. In this paper, a Smith Predictor-based PI-controller for AQM (SPPA) is proposed, which uses a TCP reference model and an average Round-Trip Time (RTT) to reduce unfavorable effects of time delays in TCP networks. The drop probability is calculated by a Proportional-Integral (PI) controller based on the prediction error. When a mismatch exists in between the actual model of the TCP process and the reference model employed by the SPPA, we demonstrate conditions under which the network is stable. The performance, robustness and effectiveness of the proposed SPPA are all evaluated using simulations. The performance of the SPPA is compared with some typical AQMs, such as the Adaptive RED, the PI-controller, and the Proportional-Differential (PD) controller.

In this letter, we propose a protocol sensitive random early detection algorithm for active queue management to improve fairness between TCP and UDP flows and to reduce delay time with small overheads. The algorithm classifies the packets into responsive and unresponsive flows, and applies the RED algorithm individually to each classified group. Using ns-2 simulations, we showed the effectiveness of the proposed PSRED algorithm compared with several well-known AQM schemes, such as RED and RED-PD algorithms.

Adaptive Virtual Queue (AVQ) introduces a novel implementation algorithm for Active Queue Management (AQM). The stability criterion for AVQ was deduced in literature [1], but it lacks practicability due to the difficulty of solving the transcendental equation. In this letter, the AVQ stability is further investigated based on the characteristic roots of delay-differential equation. Another stability criterion explicitly associated with parameters of network configuration is deduced and the upper bound of delay time for stable AVQ algorithm is determined. Finally, the conclusion is validated through simulation experiments.

TCP performance in the IEEE 802.11-based multihop ad hoc networks is extremely poor, because the congestion control mechanism of TCP cannot effectively deal with the problem of packet drops caused by mobility and shared channel contention among wireless nodes. In this paper, we present a cross-layer method, which adaptively adjusts the TCP maximum window size according to the number of RTS (Request To Send) retry counts of the MAC layer at the TCP sender, to control the number of TCP packets in the network and thus decrease the channel contention. Our simulation results show that this method can remarkably improve TCP throughput and its stability.

In multicast congestion control, the receiver of the worst congestion level is selected as the representative and transmission rate of the sender is adjusted to TCP throughput of the representative. This approach has high scalability and TCP friendliness. However, when this approach is applied in wireless communications, wireless-caused packet loss will cause to frequent change of the representative. This is because degradation of wireless channel quality causes bursty packet loss at a corresponding receiver. Fading, one of main reasons of wireless channel degradation, is expected to be recovered after short time period, which leads to frequent change of the representative. This frequent change of representative makes the sender adjust its transmission rate to the tentative worst receiver, which brings severe performance degradation to wireless multicast. We call this technical problem, the wireless-caused representative selection fluctuation problem. Wireless-caused representative selection fluctuation problem is one of scalability problems in the wireless multicast, because this problem remarkably happens for large scale multicast. We propose two possible solutions for this problem, an end-to-end approach and a network support approach. Performance evaluation in various situation show that an end-to-end approach is sensitive for its inferring error but a network support approach shows good performance improvement.

This paper presents an adaptive control design for the ABR traffic congestion control in ATM networks. Firstly, we consider a control-based mathematical model to the ABR traffic congestion control problem. Then the feedback pole placement control design is applied to the ATM ABR traffic congestion control problem for the case of known delays. Finally, by using the online plant parameter estimation algorithm and modifying the controller parameters adaptively in real time, a method to treat the case of unknown time-varying delays is proposed. Several design modifications are introduced to solve practical control issues such as bounded command rate constraint, output buffer saturation and bounded values to the plant parameter estimation algorithm. Simulations are implemented to verify the proposed control design. It is shown that while considering these practical control issues, the control method satisfies the requirements of fairness to users, network efficiency, unknown time-varying delays, queue length control and good convergence performance at an acceptable computation effort.

Active Queue Management (AQM) can maintain smaller queuing delay and higher throughput by purposefully dropping packets at the intermediate nodes. Most of the existing AQM schemes follow the probability dropping mechanism originated from Random Early Detection (RED). In this paper, we develop a novel packet dropping mechanism for AQM through designing a two-category classifier based on the Fisher Linear Discriminate approach. The simulation results show that the new scheme outperforms other well-known AQM schemes, such as RED, AdaptiveRED, AVQ, PI, REM etc., in the integrated performance. Additionally, our mechanism is simple since it requires few CPU cycles, which makes it suitable for the high-speed routers.

Available Bit Rate (ABR) flow control is an effective measure in ATM network congestion control. In large scale and high-speed network, the simplicity of algorithm is crucial to optimize the switch performance. Although the binary flow control is very simple, the queue length and allowed cell rate (ACR) controlled by the standard EFCI algorithm oscillate with great amplitude, which has negative impact on the performance, so its applicability was doubted, and then the explicit rate feedback mechanism was introduced and explored. In this study, the model of binary flow control is built based on the fluid flow theory, and its correctness is validated by simulation experiments. The linear model describing the source end system how to regulate the cell rate is obtained through local linearization method. Then, we evaluate and analyze the standard EFCI algorithm using the describing function approach, which is well-developed in nonlinear control theory. The conclusion is that queue and ACR oscillations are caused by the inappropriate nonlinear control rule originated from intuition, but not intrinsic attribute of the binary flow control mechanism. The simulation experiments validate our analysis and conclusion. Finally, the new scheme about parameter settings is put forward to remedy the weakness existed in the standard EFCI switches without any change on the hardware architecture. The numerical results demonstrate that the new scheme is effective and fruitful.

Current IP network has become the dominant paradigm for all networking environments. The significant cause of packet losses in such heterogenous networks is no longer limited to network congestion. Traditional TCP interprets every packet loss as caused by congestion which may be not the case in the current Internet. Misinterpretation of wireless random loss as an indication of network congestion results in TCP slowing down its sending rate unnecessarily. In this paper, we propose a new variant of TCP Vegas named RedVegas. By using the innate nature of Vegas and congestion indications marked by routers, RedVegas may detect random packet losses precisely. Through the packet loss differentiation, RedVegas reacts appropriately to the losses, and therefore the throughput of connection over heterogeneous networks can be significantly improved.

As the Internet proliferates, there has been a growing interest in supporting multiparty collaborative applications. It has led to the emergence of many-to-ma ny reliable multicast. Congestion control is a key task in reliable multicast along with error control. However, existing tree-based congestion control schemes such as TRAMCC and MTCP are designed for one-to-many reliable multicast and have some drawbacks when they are used for many-to-many reliable multicast. We propose an efficient congestion control mechanism, MTRMCC, for tree-based many-to-many reliable multicast protocols. The proposed scheme is based on the congestion windowing mechanism and a rate controller is used in addition. The feedback for error recovery is exploited for congestion control as well to minimize the overhead at the receivers. The ACK timer and the NACK timers are set dynamically reflecting the network traffic changes. The rate regulation algorithm in the proposed scheme is designed to help the flows sharing the same link to achieve the fair share quickly. The performance of the proposed scheme is evaluated using network simulator ns-2. The simulation results show that the proposed scheme outperforms TRAMCC in terms of intra-session fairness and supports responsiveness, TCP-friendliness, and scalability.

Using the Proportional Integral Derivative (PID) principle of classical feedback control theory, this paper develops two general congestion control algorithms for routers implementing Active Queue Management (AQM) while supporting TCP/IP traffic flows. The general designs of non-interacting (N-PID) and interacting (I-PID) congestion control algorithms are tailored for practical network scenarios using the Ziegler-Nichols guidelines for tuning such controllers. Discrete event simulations using ns are performed to evaluate the performance of an existing F-PID and new N-PID and I-PID algorithms. The performance of N-PID and I-PID is compared mutually as well as with the F-PID algorithm. It reveals that N-PID and I-PID have higher speed of response but lower stability margins than F-PID. In general the accurate following of the target queue size by the PID principle congestion control algorithms, while providing high link utilization, low loss rate and low queuing delays, is also demonstrated.

In this paper, we propose a scalable Extended Differentiated-Services (EDS) architecture to guarantee edge-to-edge explicit rate allocation. In presence of flows with explicit rate allocation, to share bandwidth fairly, a new fairness definition is proposed. Based on EDS and the proposed fairness definition, a scalable fair Edge-to-Edge Congestion Control Algorithm with Explicit Rate Allocation (ECC-ERA) is presented to solve the bandwidth assurance problem facing Differentiated Service architecture, where EDS uses congestion control packets to carry the flow-related states and congestion control information. By designing efficiency control and fairness control separately, the ECC-ERA can achieve good scalability to link capacity, round-trip time and number of flows. It will be shown that EDS plus ECC-ERA outperforms the general Diff-Serv bandwidth guarantee approaches. The main advantages of EDS+ECC-ERA are as follows: (1) it not only can guarantee explicit rate allocation, but also can guarantee near-zero packet loss in core routers, high utilization, lower and smoother queueing delay, better fairness and better protection from unresponsive traffic. (2) Neither resource pre-reservation nor sophisticated scheduling mechanisms are required. The simple FIFO at core routers is enough. (3) EDS plus EC-ERA is very efficient and can be used as end-to-end QoS building block.

To accommodate best-effort multimedia Internet protocol (IP) connections in mobile environments, we introduced new criteria for TCP-friendliness and developed a control algorithm for the transient response and stability in the packet transmission rate. We improved the maximum throughput and QoS guaranteed congestion control algorithm (MAQS) by using these two solutions, and solved the following problems that Reno and conventional congestion control algorithms have: (1) network congestion cannot be avoided when the round-trip time (RTT) is short and the holding time is long, (2) the packet transmission rate of a long-RTT connection is small when it is multiplexed with short-RTT connections, (3) the packet transmission rate cannot be adjusted quickly when the channel capacity changes according to hand-off.

Reliable multicast is an interesting application of distributing data to lots of clients at the same time. In heterogeneous environment, it is necessary to adjust the transmitting rate corresponding to the bandwidth of receivers. Placed at a network bottleneck point, an active server can buffer the multicast packets and control the transmitting rate to the downstream multicast receivers independently so as to absorb bandwidth differences. If wireless and wired receivers coexist, the best position for the active server is at the edge of the wired and wireless links because the bandwidth of wireless receivers are lower than that of wired receivers. However, it is not enough that an active server only controls the transmitting rate in such environment because wireless receivers tend to lose packet by the wireless transmission error. This paper proposes a scheme in which the active server independently controls a reliable multicast scheme that is robust against packet loss due to wireless transmission error. Simulation results show that rate-based reliable multicast congestion control is more appropriate than window-based control for wireless links. We also show that FEC applied only to the wireless link improves the throughput of wireless multicast receivers. Finally, we show that combining rate-based reliable multicast congestion control scheme with FEC only for the wireless link makes reliable multicast more practical and friendly with TCP even if packets are lost due to transmission errors.

In this letter, we evaluate the performance of several adaptive and non-adaptive AQM schemes for congestion control in a dynamic network environment with variable bandwidth links. The AQM schemes examined are RED, BLUE, Adaptive RED, REM, AVQ and PI controller. We compare their queueing performance and show that none of them can derive stable queue length and low packet drop rate simultaneously in networks where both input traffic and available output bandwidth are time varying. Adaptive and efficient algorithms should be designed and applied in order to improve the adaptiveness and robustness of congestion control in dynamic networks such as Internet.

We proposed a new buffer management scheme for GFR services through FIFO queuing discipline. Proposed scheme can provide minimum bandwidth guarantee for GFR VCs as well as improve the fairness among the competing GFR VCs on a single FIFO queue. From simulation result, we demonstrate the proposed scheme fulfills the requirements of GFR services as well as improves the TCP throughput

In this paper, we propose a nonlinear control model to characterize the AQM algorithm-GREEN. Based on this model, we analyze its performance and prove that there exists a stable oscillation when in equilibrium. Furthermore, we also investigate the effects of the factors such as bandwidth, round trip time, and load level on the amplitude and frequency of the oscillation. Theoretical analysis and simulation results indicate that GREEN algorithm is insensitive to the network conditions when the link rate and the round trip time are relatively small and becomes more sensitive to the change of network conditions when the bandwidth delay product is relatively high. For GREEN the adaptability to a wide range of network conditions is based on the compromising of the efficiency.

Although TCP is widely used in the Internet, new specifications are still proposed and implemented. In the circumstance above, it is highly possible that some errors are detected on the communication between new and old implementations. Several test tools were developed so far. However, they do not have enough functions to allow test operators to modify test sequences suitable for their test purposes. We have developed a TCP tester which generates test sequence using test scenario. The tester performs exceptional TCP protocol behavior only when the condition specified in the test scenario is satisfied. Otherwise, it performs normal TCP behavior. The tester is implemented by modifying TCP module of NetBSD with SACK code developed by Pittsburgh Supercomputing Center. We have also evaluated implementations of congestion control and SACK algorithms using the tester.