A new ATM adaptation layer for TCP/IP over wireless ATM networks

This paper describes the design and performance of a new ATM adaptation layer protocol (AAL‐T) for improving TCP performance over wireless ATM networks. The wireless links are characterized by higher error rates and burstier error patterns in comparison with the fiber links for which ATM was introduced in the beginning. Since the low performance of TCP over wireless ATM networks is mainly due to the fact that TCP always responds to all packet losses by congestion control, the key idea in the design is to push the error control portion of TCP to the AAL layer so that TCP is only responsible for congestion control. The AAL‐T is based on a novel and reliable ARQ mechanism to support quality‐critical TCP traffic over wireless ATM networks. The proposed AAL protocol has been validated using the OPNET tool with the simulated wireless ATM network. The simulation results show that the AAL‐T provides higher throughput for TCP over wireless ATM networks compared to the existing approach of TCP with AAL 5. This revised version was published online in July 2006 with corrections to the Cover Date.     

Resource allocation for broadband networks

The major benefit of a broadband integrated ATM (asynchronous transfer mode) network is flexible and efficient allocation of communications bandwidth for communications services. However, methods are needed for evaluating congestion for integrated traffic. The author suggests evaluating congestion at different levels, namely the packet level, the burst level, and the call level. Congestion is measured by the probabilities of packet blocking, burst blocking, and call blocking. He outlines the methodologies for comparing these blocking probabilities. The author uses the congestion measures for a multilayer bandwidth-allocation algorithm, emulating some function of virtual circuit setup, fast circuit switching, and fast packet switching at these levels. The analysis also sheds insight on traffic engineering issues such as appropriate link load, traffic integration, trunk group and switch sizing, and bandwidth reservation criteria for two bursty services     

Promoting the use of end-to-end congestion control in the Internet

This paper considers the potentially negative impacts of an increasing deployment of non-congestion-controlled best-effort traffic on the Internet. These negative impacts range from extreme unfairness against competing TCP traffic to the potential for congestion collapse. To promote the inclusion of end-to-end congestion control in the design of future protocols using best-effort traffic, we argue that router mechanisms are needed to identify and restrict the bandwidth of selected high-bandwidth best-effort flows in times of congestion. The paper discusses several general approaches for identifying those flows suitable for bandwidth regulation. These approaches are to identify a high-bandwidth flow in times of congestion as unresponsive, “not TCP-friendly”, or simply using disproportionate bandwidth. A flow that is not “TCP-friendly” is one whose long-term arrival rate exceeds that of any conformant TCP in the same circumstances. An unresponsive flow is one failing to reduce its offered load at a router in response to an increased packet drop rate, and a disproportionate-bandwidth flow is one that uses considerably more bandwidth than other flows in a time of congestion     

Flow control and bandwidth management in next generation Internets

The Internet has traditionally relied on end-to-end congestion control performed at the transport layer, where sources reduce their offered traffic only after congestion sets in. By then, network resources have been wasted and effective throughput compromised. In next generation Internets, the delay-bandwidth product is large and bandwidth is a precious resourse. Hence, in this paper we present a link layer flow control for Internet backbones over ATM using the ABR service and flow control. We describe a backpressure mechanism that reduces packet losses and promotes effective utilization of allocated and unused resources along backbone links. We then show how to perform dynamic renegotiation of allocated backbone resources based on our flow control approach and on class based queueing. This revised version was published online in August 2006 with corrections to the Cover Date.     

Estimation of the cell loss ratio in ATM networks with a fuzzysystem and application to measurement-based call admission control

An important parameter in asynchronous transfer model (ATM)-based network design and management is the cell loss ratio (CLR) in ATM multiplexers. It is a key parameter to many vital functions in the network such as call admission control (CAC), bandwidth allocation, etc. However, the CLR depends usually on many unknown and unpredictable traffic parameters such as input traffic correlations. In this paper, we propose a simple and robust fuzzy-based algorithm to predict the CLR in large-sized systems based on both a small amount of information from small-sized systems, and the asymptotic behavior for very large systems. Unlike the model-based approaches, our approximation avoids the problem of assuming any traffic parameters or arrival process. This algorithm is used with real-time traffic measurement to propose an effective measurement-based call admission control framework for ATM networks     

A novel adaptive traffic prediction AQM algorithm

In the Internet, network congestion is becoming an intractable problem. Congestion results in longer delay, drastic jitter and excessive packet losses. As a result, quality of service (QoS) of networks deteriorates, and then the quality of experience (QoE) perceived by end users will not be satisfied. As a powerful supplement of transport layer (i.e. TCP) congestion control, active queue management (AQM) compensates the deficiency of TCP in congestion control. In this paper, a novel adaptive traffic prediction AQM (ATPAQM) algorithm is proposed. ATPAQM operates in two granularities. In coarse granularity, on one hand, it adopts an improved Kalman filtering model to predict traffic; on the other hand, it calculates average packet loss ratio (PLR) every prediction interval. In fine granularity, upon receiving a packet, it regulates packet dropping probability according to the calculated average PLR. Simulation results show that ATPAQM algorithm outperforms other algorithms in queue stability, packet loss ratio and link utilization.     

面向ABR业务的公平排队策略

在基于ＡＴＭ－ＴＣＰ技术实现ＡＢＲ中,一个信元的丢失会导致该信元所属分组破坏,致命该信元所属分组的网络中传输都变得无效,重传机制来确保传输正确性,分组多次重传浪费大量带宽,并进一步加剧网络拥塞状况。ＡＴＭ－ＴＣＰ只保证每个信元对网络访问的公平性,而忽略了ＴＣ究组的公平性。本文提出了一种与分组长度无关的公平排队策略,提出以抛弃信元的代价为权值,来决定要抛弃的信元。它不仅能保证分组的公平性,而且能使网     

基于星上处理的卫星ATM中ABR 业务拥塞控制算法研究

卫星ATM是近年来通信领域的研究热点,拥塞控制是其中很关键的一个问题.本文主要研究基于星上处理的卫星ATM网的ABR流量控制,它通过对流量的长时预测来达到控制目的.由于卫星信道的大延时,无法实现精确的长时预测,因而将预测结果直接用于ABR的拥塞控制效果不甚理想.将长时预测同动态利用因子控制的方法结合之后,在链路效率、信元丢失率、不同连接数目和暂态响应等方面都取得很好的结果.     

An analysis of statistical multiplexing in an ATM transport network

The traffic characteristics of an asynchronous transfer mode (ATM) network are analyzed by theoretical methods. A new method is proposed to express the burstiness of the cell arrival process. Both statistical multiplexing and statistical bandwidth allocation are quantitatively evaluated. When packetized video traffic and voice traffic are multiplexed, the number of multiplexable sources strongly depends on the peak bit rate of the multiplexed video sources, and statistical bandwidth allocation is ineffective without control. On the other hand, lowering the peak bit rate of video traffic effectively improves bandwidth utilization     

Enhancing satellite broadcasting services using multiresolutionmodulations

In this paper we propose to take advantage of the energy link margin that can exist of satellite connections to enrich the DVB-S services with Web-like interactive services. The exploitation of such a margin is obtained by using multiresolution modulation techniques. The system architecture analysed is asymmetrical, composed of a satellite forward link and a narrowband terrestrial reverse link. ATM is adopted to support different QoS for different types of information delivered. The satellite propagation delay and the traffic and congestion control of ATM suggest to modify the slow start and the congestion avoidance of the TCP. Our approach is based on the combination of a fixed window flow control at the transport layer with the ATM traffic and congestion control. Our analysis shows that the system performance is satisfactory if some bounds of the TCP buffer size are respected and the spacing of the resource management cells is within a given range of values     

TCP-LP: low-priority service via end-point congestion control

Service prioritization among different traffic classes is an important goal for the Internet. Conventional approaches to solving this problem consider the existing best-effort class as the low-priority class, and attempt to develop mechanisms that provide "better-than-best-effort" service. In this paper, we explore the opposite approach, and devise a new distributed algorithm to realize a low-priority service (as compared to the existing best effort) from the network endpoints. To this end, we develop TCP Low Priority (TCP-LP), a distributed algorithm whose goal is to utilize only the excess network bandwidth as compared to the "fair share" of bandwidth as targeted by TCP. The key mechanisms unique to TCP-LP congestion control are the use of one-way packet delays for early congestion indications and a TCP-transparent congestion avoidance policy. The results of our simulation and Internet experiments show that: 1) TCP-LP is largely non-intrusive to TCP traffic; 2) both single and aggregate TCP-LP flows are able to successfully utilize excess network bandwidth; moreover, multiple TCP-LP flows share excess bandwidth fairly; 3) substantial amounts of excess bandwidth are available to the low-priority class, even in the presence of "greedy" TCP flows; 4) the response times of web connections in the best-effort class decrease by up to 90% when long-lived bulk data transfers use TCP-LP rather than TCP; 5) despite their low-priority nature, TCP-LP flows are able to utilize significant amounts of available bandwidth in a wide-area network environment.     

Request rate control for QA services in DQDB/SMDS–ATM gateways

By the advancement of B-ISDN, ATM has been accepted as the future technology to integrate a variety of services as it is a fine comprise between circuit switching technology and packet switching technology. A cost-effective and simple way to realize the global network is to implement a backbone such as DQDB to interconnect the existing LANs and MANs to ATM. The major advantages of DQDB are the complete compatibility of different layers, the capability of handling real-time and non-real-time traffic and little degradation in delay performance with increase in network span and density. However, a major issue of concern in the design of a DQDB/SMDS–ATM gateway is the transparent transfer of non-real-time traffic, as ATM is basically connection-oriented in nature. In the absence of flow control schemes in high-speed networking, some sort of congestion control mechanism becomes an absolute necessity to reduce the cell loss ratio to the minimum and thus wastage in the network resources. In this paper we discuss the transfer of connectionless traffic in a DQDB–ATM interconnection. One of the two congestion control mechanisms for DQDB networks that we have proposed is discussed in detail. In the request rate control mechanism the bandwidth utilization in DQDB is restricted by controlling the request rate from the gateway. DQDB networks interconnected by gateways are modelled and simulated. The request rate congestion control mechanism is evaluated. Simplicity in implementation and effectiveness in performance are the highlights. © 1998 John Wiley & Sons, Ltd.     

Dynamics of TCP traffic over ATM networks

Investigates the performance of transport control protocol (TCP) connections over ATM networks without ATM-level congestion control and compares it to the performance of TCP over packet-based networks. For simulations of congested networks, the effective throughput of TCP over ATM can be quite low when cells are dropped at the congested ATM switch. The low throughput is due to wasted bandwidth as the congested link transmits cells from “corrupted” packets, i.e., packets in which at least one cell is dropped by the switch. The authors investigate two packet-discard strategies that alleviate the effects of fragmentation. Partial packet discard, in which remaining cells are discarded after one cell has been dropped from a packet, somewhat improves throughput. They introduce early packet discard, a strategy in which the switch drops whole packets prior to buffer overflow. This mechanism prevents fragmentation and restores throughput to maximal levels     

GSM phase 2+ general packet radio service GPRS: Architecture, protocols, and air interface

Selective packet dropping policies have been used to reduce congestion and transmission of traffic that would inevitably be retransmitted. For data applications using best-effort services, packet dropping policies (PDPs) are congestion management mechanisms implemented at each intermediate node that decide, reactively or proactively, to drop packets to reduce congestion and free up precious buffer space. While the primary goal of PDPs is to avoid or combat congestion, the individual PDP designs can significantly affect application throughput, network utilization, performance fairness, and synchronization problems with multiple Transmission Control Protocol (TCP) connections. Scalability and simplicity are also important design issues. This article surveys the most important selective packet dropping policies that have been designed for best-effort traffic in ATM and IP networks, providing a comprehensive comparison between the different mechanisms.     

TCP-Jersey for wireless IP communications

Improving the performance of the transmission control protocol (TCP) in wireless Internet protocol (IP) communications has been an active research area. The performance degradation of TCP in wireless and wired-wireless hybrid networks is mainly due to its lack of the ability to differentiate the packet losses caused by network congestions from the losses caused by wireless link errors. In this paper, we propose a new TCP scheme, called TCP-Jersey, which is capable of distinguishing the wireless packet losses from the congestion packet losses, and reacting accordingly. TCP-Jersey consists of two key components, the available bandwidth estimation (ABE) algorithm and the congestion warning (CW) router configuration. ABE is a TCP sender side addition that continuously estimates the bandwidth available to the connection and guides the sender to adjust its transmission rate when the network becomes congested. CW is a configuration of network routers such that routers alert end stations by marking all packets when there is a sign of an incipient congestion. The marking of packets by the CW configured routers helps the sender of the TCP connection to effectively differentiate packet losses caused by network congestion from those caused by wireless link errors. This paper describes the design of TCP-Jersey, and presents results from experiments using the NS-2 network simulator. Results from simulations show that in a congestion free network with 1% of random wireless packet loss rate, TCP-Jersey achieves 17% and 85% improvements in goodput over TCP-Westwood and TCP-Reno, respectively; in a congested network where TCP flow competes with VoIP flows, with 1% of random wireless packet loss rate, TCP-Jersey achieves 9% and 76% improvements in goodput over TCP-Westwood and TCP-Reno, respectively. Our experiments of multiple TCP flows show that TCP-Jersey maintains the fair and friendly behavior with respect to other TCP flows.     

Call admission control in an ATM network using upper bound of cellloss probability

Call admission control in ATM networks without monitoring network load is proposed. Traffic parameters specified by users are employed to obtain the upper bound of cell loss probability, thereby determining call admission. A cell loss probability standard is guaranteed to be satisfied under this control without assumptions of a cell arrival process. Implementation of this control to quickly evaluate cell loss probability after acceptance of a new call is discussed. The result is also applicable to ATM network dimensioning based on traffic parameters     

Active intra-ONU scheduling with proportional guaranteed bandwidth in long-reach EPONs

The unused slot remainder (USR) problem in Ethernet Passive Optical Networks and long-reach Passive Optical Networks (LR-PONs) results in both a lower bandwidth utilization and a greater packet delay. In a previous study by the current group, an Active intra-ONU Scheduling with predictive queue report scheme was proposed for resolving the USR problem by predicting the granted bandwidth in advance based on the arrival traffic estimates of the optical network units (ONUs). However, it was found that the higher bandwidth prediction error in the proposed scheme prevents the network performance from being improved. Accordingly, the present study proposes a non-predictive-based ONU scheduling method designated as Active intra-ONU Scheduling with proportional guaranteed bandwidth (ASPGB) to improve the performance of LR-PONs. In the proposed method, the maximum guaranteed bandwidth of each ONU is adapted dynamically in accordance with the ratio of the ONU traffic load to the overall system load. Importantly, the proposed dynamic bandwidth allocation approach reduces the dependence of the network performance on the granted bandwidth prediction since the maximum guaranteed bandwidth determined by the Optical Line Terminal more closely approaches the actual bandwidth demand of each ONU. To solve the idle time problem arising in the event of an excess bandwidth reallocation, ASPGB is integrated with an improved early allocation (IEA) algorithm (a kind of Just-In-Time scheduling). The simulation results show that the IEA-ASPGB scheme outperforms previously published methods in terms of bandwidth utilization and average packet delay under both balanced and unbalanced traffic load conditions.