Showing papers on "Throughput published in 1997"

A comparison of mechanisms for improving TCP performance over wireless links

Abstract: Reliable transport protocols such as TCP are tuned to perform well in traditional networks where packet losses occur mostly because of congestion. However, networks with wireless and other lossy links also suffer from significant losses due to bit errors and handoffs. TCP responds to all losses by invoking congestion control and avoidance algorithms, resulting in degraded end-to end performance in wireless and lossy systems. We compare several schemes designed to improve the performance of TCP in such networks. We classify these schemes into three broad categories: end-to-end protocols, where loss recovery is performed by the sender; link-layer protocols that provide local reliability; and split-connection protocols that break the end-to-end connection into two parts at the base station. We present the results of several experiments performed in both LAN and WAN environments, using throughput and goodput as the metrics for comparison. Our results show that a reliable link-layer protocol that is TCP-aware provides very good performance. Furthermore, it is possible to achieve good performance without splitting the end-to-end connection at the base station. We also demonstrate that selective acknowledgments and explicit loss notifications result in significant performance improvements.

The performance of TCP/IP for networks with high bandwidth-delay products and random loss

Abstract: This paper examines the performance of TCP/IP, the Internet data transport protocol, over wide-area networks (WANs) in which data traffic could coexist with real-time traffic such as voice and video. Specifically, we attempt to develop a basic understanding, using analysis and simulation, of the properties of TCP/IP in a regime where: (1) the bandwidth-delay product of the network is high compared to the buffering in the network and (2) packets may incur random loss (e.g., due to transient congestion caused by fluctuations in real-time traffic, or wireless links in the path of the connection). The following key results are obtained. First, random loss leads to significant throughput deterioration when the product of the loss probability and the square of the bandwidth-delay product is larger than one. Second, for multiple connections sharing a bottleneck link, TCP is grossly unfair toward connections with higher round-trip delays. This means that a simple first in first out (FIFO) queueing discipline might not suffice for data traffic in WANs. Finally, while the Reno version of TCP produces less bursty traffic than the original Tahoe version, it is less robust than the latter when successive losses are closely spaced. We conclude by indicating modifications that may be required both at the transport and network layers to provide good end-to-end performance over high-speed WANs.

Associativity-Based Routing for Ad Hoc Mobile Networks

Abstract: This paper presents a new, simple and bandwidth-efficient distributed routing protocol to support mobile computing in a conference size ad-hoc mobile network environment. Unlike the conventional approaches such as link-state and distance-vector distributed routing algorithms, our protocol does not attempt to consistently maintain routing information in every node. In an ad-hoc mobile network where mobile hosts (MHs) are acting as routers and where routes are made inconsistent by MHs‘ movement, we employ an associativity-based routing scheme where a route is selected based on nodes having associativity states that imply periods of stability. In this manner, the routes selected are likely to be long-lived and hence there is no need to restart frequently, resulting in higher attainable throughput. Route requests are broadcast on a per need basis. The association property also allows the integration of ad-hoc routing into a BS-oriented Wireless LAN (WLAN) environment, providing the fault tolerance in times of base stations (BSs) failures. To discover shorter routes and to shorten the route recovery time when the association property is violated, the localised-query and quick-abort mechanisms are respectively incorporated into the protocol. To further increase cell capacity and lower transmission power requirements, a dynamic cell size adjustment scheme is introduced. The protocol is free from loops, deadlock and packet duplicates and has scalable memory requirements. Simulation results obtained reveal that shorter and better routes can be discovered during route re-constructions.

M-TCP: TCP for mobile cellular networks

Abstract: Transport connections set up over wireless links are frequently plagued by problems such as - high bit error rate (BER), frequent disconnections of the mobile user, and low wireless bandwidth that may change dynamically. In this paper, we study the effects of frequent disconnections and low variable bandwidth on TCP throughput and propose a protocol that addresses this problem. We discuss the implementation (in NetBSD) of our protocol called M-TCP and compare its performance against other mobile TCP implementations. We show that M-TCP has two significant advantages over other solutions: (1) it maintains end-to-end TCP semantics and, (2) it delivers excellent performance for environments where the mobile encounters periods of disconnection.

Solutions to hidden terminal problems in wireless networks

Abstract: The floor acquisition multiple access (FAMA) discipline is analyzed in networks with hidden terminals. According to FAMA, control of the channel (the floor) is assigned to at most one station in the network at any given time, and this station is guaranteed to be able to transmit one or more data packets to different destinations with no collisions. The FAMA protocols described consist of non-persistent carrier or packet sensing, plus a collision-avoidance dialogue between a source and the intended receiver of a packet. Sufficient conditions under which these protocols provide correct floor acquisition are presented and verified for networks with hidden terminals; it is shown that FAMA protocols must use carrier sensing to support correct floor acquisition. The throughput of FAMA protocols is analyzed for single-channel networks with hidden terminals; it is shown that carrier-sensing FAMA protocols perform better than ALOHA and CSMA protocols in the presence of hidden terminals.

Performance modeling of asynchronous data transfer methods of IEEE 802.11 MAC protocol

Abstract: To satisfy the needs of wireless data networking, study group 802.11 was formed under IEEE project 802 to recommend an international standard for Wireless Local Area Networks (WLANs). A key part of standard are the Medium Access Control (MAC) protocol needed to support asynchronous and time bounded delivery of data frames. It has been proposed that unslotted Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) be the basis for the IEEE 802.11 WLAN MAC protocols. We conduct performance evaluation of the asynchronous data transfer protocols that are a part of the proposed IEEE 802.11 standard taking into account the decentralized nature of communication between stations, the possibility of “capture”, and presence of “hidden” stations. We compute system throughput and evaluate fairness properties of the proposed MAC protocols. Further, the impact of spatial characteristics on the performance of the system and that observed by individual stations is determined. A comprehensive comparison of the access methods provided by the 802.11 MAC protocol is done and observations are made as to when each should be employed. Extensive numerical and simulation results are presented to help understand the issues involved.

A comparison of sender-initiated and receiver-initiated reliable multicast protocols

Abstract: Sender-initiated reliable multicast protocols based on the use of positive acknowledgments (ACKs) can suffer performance degradation as the number of receivers increases. This degradation is due to the fact that the sender must bear much of the complexity associated with reliable data transfer (e.g., maintaining state information and timers for each of the receivers and responding to receivers' ACKs). A potential solution to this problem is to shift the burden of providing reliable data transfer to the receivers-thus resulting in receiver-initiated multicast error control protocols based on the use of negative acknowledgments (NAKs). We determine the maximum throughputs for generic sender-initiated and receiver-initiated protocols for two classes of applications: (1) one-many applications where one participant sends data to a set of receivers and (2) many-many applications where all participants simultaneously send and receive data to/from each other. We show that a receiver-initiated error control protocol which requires receivers to transmit NAKs point-to-point to the sender provides higher throughput than a sender-initiated counterpart for both classes of applications. We further demonstrate that, in the case of a one many application, replacing point-to-point transfer of NAKs with multicasting of NAKs coupled with a random backoff procedure provides a substantial additional increase in the throughput of a receiver-initiated error control protocol over a sender-initiated protocol. We also find, however, that such a modification leads to a throughput degradation in the case of many-many applications.

ARQ error control for fading mobile radio channels

Abstract: In this paper, we study the correlation properties of the fading mobile radio channel Based on these studies, we model the channel as a one-step Markov process whose transition probabilities are a function of the channel characteristics Then we present the throughput performance of the Go-Back-N and selective-repeat automatic repeat request (ARQ) protocols with timer control, using the Markov model for both forward and feedback channels This approximation is found to be very good, as confirmed by simulation results

Implementation and performance evaluation of Indirect TCP

Abstract: With the advent of small portable computers and the technological advances in wireless communications, mobile wireless computing is likely to become very popular in the near future. Wireless links are slower and less reliable compared to wired links and are prone to loss of signal due to noise and fading. Furthermore, host mobility can give rise to periods of disconnection from the fixed network. The use of existing network protocols, which were developed mainly for the high bandwidth and faster wired links, with mobile computers thus gives rise to unique performance problems arising from host mobility and due to the characteristics of wireless medium. Indirect protocols can isolate mobility and wireless related problems using mobility support routers (MSRs) as intermediaries, which also provide backward compatibility with fixed network protocols. We present the implementation and performance evaluation of I-TCP, which is an indirect transport layer protocol for mobile wireless environments. Throughput comparison with regular (BSD) TCP shows that I-TCP performs significantly better in a wide range of conditions related to wireless losses and host mobility. We also describe the implementation and performance of I-TCP handoffs.

Effect of traffic self-similarity on network performance

Abstract: Recent measurements of network traffic have shown that self- similarity is an ubiquitous phenomenon present in both local area and wide area traffic traces. In previous work, we have shown a simple, robust application layer causal mechanism of traffic self-similarity, namely, the transfer of files in a network system where the file size distributions are heavy- tailed. In this paper, we study the effect of scale- invariant burstiness on network performance when the functionality of the transport layer and the interaction of traffic sources sharing bounded network resources is incorporated. First, we show that transport layer mechanisms are important factors in translating the application layer causality into link traffic self-similarity. Network performance as captured by throughput, packet loss rate, and packet retransmission rate degrades gradually with increased heavy-tailedness while queueing delay, response time, and fairness deteriorate more drastically. The degree to which heavy-tailedness affects self-similarity is determined by how well congestion control is able to shape a source traffic into an on-average constant output stream while conserving information. Second, we show that increasing network resources such as link bandwidth and buffer capacity results in a superlinear improvement in performance. When large file transfers occur with nonnegligible probability, the incremental improvement in throughput achieved for large buffer sizes is accompanied by long queueing delays vis-a- vis the case when the file size distribution is not heavy- tailed. Buffer utilization continues to remain at a high level implying that further improvement in throughput is only achieved at the expense of a disproportionate increase in queueing delay. A similar trade-off relationship exists between queueing delay and packet loss rate, the curvature of the performance curve being highly sensitive to the degree of self-similarity. Third, we investigate the effect of congestion control on network performance when subject to highly self-similar traffic conditions. We implement an open-loop congestion control using unreliable transport on top of UDP where the data stream is throttled at the source to achieve a fixed arrival rate. Decreasing the arrival rate results in a decline in packet loss rate whereas link utilization increases. In the context of reliable communication, we compare the performance of three versions of TCP--Reno, Tahoe, and Vegas--and we find that sophistication of control leads to improved performance that is preserved even under highly self-similar traffic conditions. The performance gain from Tahoe to Reno is relatively minor while the performance jump from TCP Reno to Vegas is more pronounced consistent with quantitative results reported elsewhere.© (1997) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Fair scheduling in wireless packet networks

Abstract: Fair scheduling of delay and rate-sensitive packet flows over a wireless channel is not addressed effectively by most contemporary wireline fair scheduling algorithms because of two unique characteristics of wireless media: (a) bursty channel errors, and (b) location-dependent channel capacity and errors. Besides, in packet cellular networks, the base station typically performs the task of packet scheduling for both downlink and uplink flows in a cell; however a base station has only a limited knowledge of the arrival processes of uplink flows.In this paper, we propose a new model for wireless fair scheduling based on an adaptation of fluid fair queueing to handle location-dependent error bursts. We describe an ideal wireless fair scheduling algorithm which provides a packetized implementation of the fluid model while assuming full knowledge of the current channel conditions. For this algorithm, we derive the worst-case throughput and delay bounds. Finally, we describe a practical wireless scheduling algorithm which approximates the ideal algorithm. Through simulations, we show that the algorithm achieves the desirable properties identified in the wireless fluid fair queueing model.

Error control and energy consumption in communications for nomadic computing

Abstract: We consider the problem of communications over a wireless channel in support of data transmissions from the perspective of small portable devices that must rely on limited battery energy. We model the channel outages as statistically correlated errors. Classic ARQ strategies are found to lead to a considerable waste of energy, due to the large number of transmissions. The use of finite energy sources in the face of dependent channel errors leads to new protocol design criteria. As an example, a simple probing scheme, which slows down the transmission rate when the channel is impaired, is show? to be more energy efficient, with a slight loss in throughput. A modified scheme that yields slightly better performance but requires some additional complexity is also studied. Some references on the modeling of battery cells are discussed to highlight the fact that battery charge capacity is strongly influenced by the available "relaxation time" between current pulses. A formal approach that can track complex models for power sources, including dynamic charge recovery, is also developed.

TCP extensions for space communications

Abstract: The space communication environment and mobile and wireless communication environments show many similarities when observed from the perspective of a transport protocol. Both types of environments exhibit loss caused by data corruption and link outage, in addition to congestion-related loss. The constraints imposed by the two environments are also similar—power, weight, and physical volume of equipment are scarce resources. Finally, it is not uncommon for communication channel data rates to be severely limited and highly asymmetric. We are working on solutions to these types of problems for space communication environments, and we believe that these solutions may be applicable to the mobile and wireless community. As part of our work, we have defined and implemented the Space Communications Protocol Standards-Transport Protocol (SCPS-TP), a set of extensions to TCP that address the problems that we have identified. The results of our performance tests, both in the laboratory and on actual satellites, indicate that the SCPS-TP extensions yield significant improvements in throughput over unmodified TCP on error-prone links. Additionally, the SCPS modifications significantly improve performance over links with highly asymmetric data rates.

Scheduling and performance limits of networks with constantly changing topology

Abstract: A communication network with tine-varying topology is considered. The network consists of M receivers and N transmitters that, in principle, may access every receiver. An underlying network state process with Markovian statistics is considered that reflects the physical characteristics of the network affecting the link service capacity. The transmissions are scheduled dynamically, based on information about the link capacities and the backlog in the network. The region of achievable throughputs is characterized. A transmission scheduling policy is proposed that utilizes current topology state information and achieves all throughput vectors achievable by any anticipative policy. The changing topology model applies to networks of low-Earth orbit (LEO) satellites, meteor-burst communication networks, and networks with mobile users.

Window-based error recovery and flow control with a slow acknowledgement channel: a study of TCP/IP performance

Abstract: With the envisaged growth in Internet access services over networks with asymmetric links such as asymmetric digital subscriber line (ADSL) and hybrid fiber coax (HFC), it becomes crucial to evaluate the performance of window-based protocols over systems in which the reverse link is considerably slower than the forward link. Even if the actual bandwidth asymmetry is moderate, high effective asymmetries can result because of bidirectional traffic. Our objective is to determine, whether TCP/IP performs reasonably in a setting in which the reverse link is the primary bottleneck. Our main results are as follows. (1) For both the prevalent Tahoe version with Fast Retransmit and the Reno version of TCP we determine the throughput as a function of buffering, round-trip times and normalized asymmetry (taken to be the ratio of the transmission time of ACKs in the reverse path to that of data packets in the forward path). We identify three modes of operation which are dependent on the forward buffer sizes and the normalized asymmetry. (2) Asymmetry increases the TCP's already high sensitivity to random packet losses that might be caused by transient bursts in real-time traffic. Specifically, random loss leads to significant throughput deterioration when the product of the loss probability, the asymmetry and the square of the bandwidth delay product is large. (3) Congestion in the reverse path adds considerably to the TCP's unfairness when multiple connections share the reverse link. Link bandwidth sharing is unfair even for connections with identical round-trip times and hence use of per connection buffer allocation on the reverse path appears essential.

On the effect of traffic self-similarity on network performance

Abstract: Recent measurements of network traffic have shown that self-similarity is an ubiquitous phenomenon present in both local area and wide area traffic traces. In previous work, we have shown a simple, robust application layer causal mechanism of traffic self-similarity, namely, the transfer of files in a network system where the file size distributions are heavy-tailed. In this paper, we study the effect of scale-invariant burstiness on network performance when the functionality of the transport layer and the interaction of traffic sources sharing bounded network resources is incorporated. First, we show that transport layer mechanisms are important factors in translating the application layer causality into link traffic self-similarity. Network performance as captured by throughput, packet loss rate, and packet retransmission rate degrades gradually with increased heavy-tailedness while queueing delay, response time, and fairness deteriorate more drastically. The degree to which heavy-tailedness affects self-similarity is determined by how well congestion control is able to shape a source traffic into an on-average constant output stream while conserving information. Second, we show that increasing network resources such as link bandwidth and buffer capacity results in a superlinear improvement in performance. When large file transfers occur with nonnegligible probability, the incremental improvement in throughput achieved for large buffer sizes is accompanied by long queueing delays vis-a-vis the case when the file size distribution is not heavy-tailed. Buffer utilization continues to remain at a high level implying that further improvement in throughput is only achieved at the expense of a disproportionate increase in queueing delay. A similar trade-off relationship exists between queueing delay and packet loss rate, the curvature of the performance curve being highly sensitive to the degree of self-similarity. Third, we investigate the effect of congestion control on network performance when subject to highly self-similar traffic conditions. We implement an open-loop congestion control using unreliable transport on top of UDP where the data stream is throttled at the source to achieve a fixed arrival rate. Decreasing the arrival rate results in a decline in packet loss rate whereas link utilization increases. In the context of reliable communication, we compare the performance of three versions of TCP-Reno, Tahoe, and Vegas-and we find that sophistication of control leads to improved performance that is preserved even under highly self-similar traffic conditions. The performance gain from Tahoe to Reno is relatively minor while the performance jump from TCP Reno to Vegas is more pronounced consistent with quantitative results reported elsewhere.

Using channel state dependent packet scheduling to improve TCP throughput over wireless LANs

Abstract: In recent years, a variety of mobile computers equipped with wireless communication devices have become popular. These computers use applications and protocols, originally developed for wired desktop hosts, to communicate over wireless channels. Unlike wired networks, packets transmitted on wireless channels are often subject to burst errors which cause back to back packet losses. In this paper we study the effect of burst packet errors and error recovery mechanisms employed in wireless MAC protocols on the performance of transport protocols such as TCP. Most wireless LAN link layer protocols recover from packet losses by retransmitting lost segments. When the wireless channel is in a burst error state, most retransmission attempts fail, thereby causing poor utilization of the wireless channel. Furthermore, in the event of multiple sessions sharing a wireless link, FIFO packet scheduling can cause the HOL blocking effect, resulting in unfair sharing of the bandwidth. This observation leads to a new class of packet dispatching methods which explicitly take wireless channel characteristics into consideration in making packet dispatching decisions. We compare a variety of channel state dependent packet (CSDP) scheduling methods with a view towards enhancing the performance of transport layer sessions. Our results indicate that by employing a CSDP scheduler at the wireless LAN device driver level, significant improvement in channel utilization can be achieved in typical wireless LAN configurations.

Leap forward virtual clock: a new fair queuing scheme with guaranteed delays and throughput fairness

Abstract: We describe an ejjicient fair queuing scheme, Leap Forward Virtual Clock, that provides end-to-end delay bounds simdar to WFQ, along with throughput fairness. Our scheme can be implemented with a worstca.se time O (log log N) per packet (inclusive of sorting costs), which improves upon all previously known schemes that guarantee delay and throughput fairness similar to WFQ. Interestingly, both the classical virtual clock and the Self- Clocked Fair Queuing schemes can be thought of as special cases of our scheme, by setting the leap forward parameter appropriately.

Guaranteed quality-of-service wireless access to ATM networks

Abstract: We study the problem of wireless access to asynchronous transfer modes (ATMs). We consider three classes of ATM sources: constant bit rate (CBR), variable bit rate (VBR), and available bit rate (ABR). We propose a polling scheme with nonpreemptive priority. Under such a scheme, we derive sufficient conditions such that all the CBR sources satisfy their jitter constraints and all the VBR sources satisfy their delay constraints. The remaining bandwidth is used by the ABR sources, for which we adapt a random access scheme proposed by Chen and Lee (1994). For this random access scheme, we derive the throughput-offer load characteristic, and thus the capacity. Based on this, we propose adaptive random access schemes that track the offer load to its optimal value. Our simulations show that our adaptive schemes maintain a high throughput with respect to the whole range of system load.

Investigation of the IEEE 802.11 medium access control (MAC) sublayer functions

Abstract: Analysis of the draft IEEE 802.11 wireless local area network (WLAN) standard is needed to characterize the expected performance of the standard's ad hoc and infrastructure networks. The performance of the medium access control (MAC) sublayer, which consists of distributed coordination function (DCF) and point coordination function (PCF), is determined by simulating asynchronous data traffic in a 1 Mbps ad hoc network, and asynchronous data and packetized voice traffic in a 1 Mbps infrastructure network. The simulation models incorporate the effect of burst errors, packet size, RTS threshold and fragmentation threshold on network throughput and delay. The results show that the IEEE 802.11 WLAN can achieve a reasonably high efficiency when the medium is almost error-free, but may degrade appreciably under harsh fading. The results also show that time-sensitive traffic such as packet voice can be supported together with other intensive traffic such as packet data. However, an echo canceller is required for packet voice systems.

Method and apparatus for measurement of peak throughput in packetized data networks

Abstract: A system for measuring peak throughput in packetized data networks includes a remote monitoring probe (12) and console (16). The probe (12) is connected to the network to monitor network activity, while the console is in communication with the probe via the network or other communications medium. The probe maintains a plurality of counters associated with different ranges of percentage of utilization of network bandwidth. For each sampling interval, the probe measures the network and individual network circuit bandwidth utilization and increments the appropriate counters associated with the percentage ranges encompassing the measured bandwidth utilizations. The console polls the probe for the percentage counter data to display the network bandwidth utilization in the form of a bar graph and pie chart. The network bandwidth may then be adjusted based on the displayed data.

Low-loss TCP/IP header compression for wireless networks

Abstract: Wireless is becoming a popular way to connect mobile computers to the Internet and other networks. The bandwidth of wireless links will probably always be limited due to properties of the physical medium and regulatory limits on the use of frequencies for radio communication. Therefore, it is necessary for network protocols to utilize the available bandwidth efficiently. Headers of IP packets are growing and the bandwidth required for transmitting headers is increasing. With the coming of IPv6 the address size increases from 4 to 16 bytes and the basic IP header increases from 20 to 40 bytes. Moreover, most mobility schemes tunnel packets addressed to mobile hosts by adding an extra IP header or extra routing information, typically increasing the size of TCP/IPv4 headers to 60 bytes and TCP/IPv6 headers to 100 bytes. In this paper, we provide new header compression schemes for UDP/IP and TCP/IP protocols. We show how to reduce the size of UDP/IP headers by an order of magnitude, down to four to five bytes. Our method works over simplex links, lossy links, multi-access links, and supports multicast communication. We also show how to generalize the most commonly used method for header compression for TCP/IPv4, developed by Jacobson, to IPv6 and multiple IP headers. The resulting scheme unfortunately reduces TCP throughput over lossy links due to unfavorable interaction with TCP's congestion control mechanisms. However, by adding two simple mechanisms the potential gain from header compression can be realized over lossy wireless networks as well as point-to-point modem links.

System for placing packets on network for transmission from sending endnode to receiving endnode at times which are determined by window size and metering interval

Abstract: A method for controlling communications between endnodes in a packet-switched computer network includes dynamic window sizing and dynamic packet metering. Dynamic window sizing regularly probes the network to determine whether higher throughput is available at an increased window size, and avoids unnecessary decreases in window size after a packet is dropped or all available data has been transmitted. Dynamic packet metering regularly adjusts the rate at which packets are transmitted in response to changes in the measured propagation rate of packets through the network. To avoid unnecessary ack packets, acks are bundled together and piggybacked on returning data packets when appropriate. The invention provides control even if the address of an endnode changes. The invention also supports multiplexing several logical connections over a single transport session and combining data from several connections in a single packet, as well as construction of packets in a network-layer-independent format.

Data processor having controlled scalable input data source and method thereof

Abstract: A multimedia communication system processes and multiplexes different types of data, including data from an adaptive data rate data source and a nonadaptive data rate data source, to substantially increase data throughput over a communication channel. The system includes: a first data source, responsive to a control signal, including a video image processor constructed to capture images and to present the images as a first type of data at a rate determined as a function of the control signal; at least one additional data source generating at least one additional data signal; a data signal processor that determines an available bandwidth factor for the communication channel, generates the control signal in response to this factor, collects the first type of data at a rate that varies in response to the available channel bandwidth of the modem, and collects the at least one additional type of data at at least one established rate. A communication channel modem transmits the data to a receiving terminal as the data is presented by the data signal processor.

System and method for the efficient control of a radio communications network

Abstract: A technique for optimizing throughput on a communications channel shared by multiple users. A communications channel that must be shared by a large number of devices has the potential of being very inefficient because of collisions or overlapping of transmissions by the various devices. The system combines a carrier sense, multiple access (CSMA) mode with a time division multiple access (TDMA) mode to achieve a channel utilization greater than 90 percent. The remote units send a poll request to a base station using the CSMA mode and receive a poll signal from the base station with a poll sequence. The remote units send their data in their assigned time slot. The remote units do not have to all be in radio contact with each other to maintain synchronization. Each remote unit selects the base station that it wishes to communicate with based on signal strength of various base stations. The remote units may switch from one base station to another by addressing the selected base station and using the selected base station's synchronization data pattern in radio transmissions from the remote unit. The synchronization data pattern may be different for each base station or may be identical for groups of base stations to provide broader regional control of the communications network. The base station will only communicate with remote units using the synchronization code for that base station. The system also recovers data from a more powerful signal that collides with a weaker signal by examining the received data for the synchronization code from the more powerful signal.

Dynamic global packet routing in wireless networks

Abstract: We consider schemes for reuse-efficient packet access in wireless data networks. We show that computing the maximum ergodic packet arrival rate is NP-hard. We give an upper bound on the maximum ergodic throughput in terms of the eigenvalues of matrices related to the path-gain matrix. We present simple, practical heuristic algorithms which exhibit good throughput and packet delay and report on results of preliminary simulations. More sophisticated algorithms that yield optimal throughput are also presented. A recent result of McKeown, Anantharam and Walrand (1996) on scheduling of input-queued switches is obtained as a by-product.

Bandwidth reservation for multiple file transfer in a high speed communication network

Abstract: A method and device for transmission of data units over a communication network, particularly over a network where data units are transmitted with constant bit rate Bandwidth reservation is optimized in the communication network for more than one file transfer within a greater time period between two dedicated end systems, running automatically without needing user interaction The bandwidth is controlled by characterizing a given transmission path with an optimal bandwidth learned from characterization results Based on observations of a data path during transmission phase, for instance by a packet trace, the bandwidth used and level of utilization are determined and an optimal bandwidth is stored in memory and used for a next file transfer to the same end system The advantage of the optimization procedure is high throughput, ie, short delay, by keeping a high bandwidth utilization with regard to low overallocation The user does not have to estimate a vague value for the bandwidth reservation Additionally, the system is adaptive so that a new optimal bandwidth occurs after a reconfiguration

Leap forward virtual clock: a new fair queuing scheme with guaranteed delays and throughput fairness

Abstract: We describe an efficient fair queuing scheme, leap forward virtual clock, that provides end-to-end delay bounds similar to weighted fair queuing (WFQ), along with throughput fairness. Our scheme can be implemented with a worst-case time O(loglogN) per packet (inclusive of sorting costs), which improves upon all previously known schemes that guarantee delay and throughput fairness similar to WFQ. Interestingly, both the classical virtual clock and the self-clocked fair queuing schemes can be thought of as special cases of our scheme, by setting the leap forward parameter appropriately.

Remote communications server system

Abstract: A remote communications server system enables multiple remote users simultaneous access to a network. The system connects a plurality of on-line sessions across multiple communication lines to the network. A unique combination of multiple buses, multiple processors, and a segmented design creates a distributed processing communications system having high throughput without the stability problems associated with gigabit bus speeds. Furthermore, the system supports a mixture of communication links and allows the substitution of one service type for another without affecting the remaining communication links. The system is scalable in that segments can be added as needed and the number of lines handled by a segment can be increased.

UBR+: improving performance of TCP over ATM-UBR service

Abstract: ATM-UBR service responds to congestion by dropping cells when switch buffers become full. TCP connections running over UBR experience low throughput and high unfairness. For 100% TCP throughput, each switch needs buffers equal to the sum of the window sizes of all the TCP connections. Intelligent drop policies can improve the performance of TCP over UBR with limited buffers. The UBR+ service proposes enhancements to UBR for intelligent drop. The early packet discard scheme improves throughput but does not attempt to improve fairness. The selective packet drop scheme based on per-connection buffer occupancy improves fairness. The fair buffer allocation scheme further improves both throughput and fairness.