Buffer Requirements Research Articles

Improve TCP performance with a novel adaptive queue management scheme in IEEE 802.17 RPR optical networks

Resilient Packet Ring (RPR), a novel IEEE 802-series MAC scheme over optical network for Metropontan Area Networks (MAN), provides flexible data efficient delivery and optical transmission reliability. However, RPR fairness scheme causes TCP performance degradation. In this paper, an analytical traffic model is developed to study the impact of congestion control of TCP under RPR AM fairness scheme. Basically, two possible ways may be able to improve this impact, to equip a buffer to accommodate the burst or to employ a robust active queue management (AQM). However, buffer requirement in RPR networks may be different for various network scenarios and AQM scheme should be better for RPR networks. Furthermore, our study shows that TCP works not well if queue management parameters are fixed at a value. Hence, a novel adaptive queue management scheme with RPR feedback information is proposed. Available bandwidth for each ingress-aggregated flow should be passed to its upstream nodes in RPR MAC scheme. Accordingly, queue management design parameters could be adaptively tuned to achieve better queueing behavior. Through detailed simulations in various scenarios, our proposed schemes are able to improve TCP performance obviously.

The role of preservation solution on acid–base regulation during machine perfusion of kidneys

To meet the current clinical organ demand, efficient preservation methods and solutions are needed to increase the number of viable kidneys for transplantation. In the present study, the influence of perfusion solution buffering strength on renal pH dynamics and regulation mechanisms during kidney ex vivo preservation was determined. Porcine kidneys were hypothermically machine perfused for 72 h with either Unisol-UHK or Belzer-Machine Perfusion solution, Belzer-MP solution. Renal perfusate samples were periodically collected and biochemically analyzed. The UHK solution, a Hepes-based solution (35 mM), provided a more efficient control of renal pH that, in turn, resulted in minor changes in the perfusate pH relative to baseline, in response to tissue CO2 and HCO3- production. In the perfusate of Belzer-MP kidney group a wider range of pH values were recorded and a pronounced pH reduction was seen in response to significant rises in pCO2 and HCO3- concentrations. The Belzer-MP solution, containing phosphate (25 mM) as its main buffer, and only 10 mM Hepes, had a greater buffering requirement to attenuate larger pH changes.

Low-pass filtering of rate-distortion functions for quality smoothing in real-time video communication

In variable-bit-rate video coding, the video is preprocessed to collect sequence-level statistics, which are used for global bit allocation in the actual encoding stage to obtain a smoothed video presentation quality. However, in real-time video recording and network streaming, this type of two-pass encoding scheme is not allowed because the access to future frames and global statistics is not available. To address this issue, we introduce the concept of low-pass filtering of rate-distortion functions and develop a smoothed rate control (SRC) framework for real-time video recording and streaming. Theoretically, we prove that, using a geometric averaging filter, the SRC algorithm is able to maintain a smoothed video presentation quality while achieving the target bit rate automatically. We also analyze the buffer requirement of the SRC algorithm in real-time video streaming, and propose a scheme to seamlessly integrate robust buffer control into the SRC framework. The proposed SRC algorithm has very low computational complexity and implementation cost. Our extensive experimental results demonstrate that the SRC algorithm significantly reduces the picture quality variation in the encoded video clips.

Energy-Efficient TCP Operation in Wireless Sensor Networks

Many applications of wireless sensor networks require connectivity to external networks to let monitoring and controlling entities communicate with the sensors. By using the TCP/IP protocols inside the sensor network, external connectivity can be achieved anywhere in the sensor network. In such IP-based sensor networks, TCP can be used for remote management and reprogramming of sensor nodes. However, the high bit error rates in multi-hop sensor networks lead to energy-inefficiencies that reduce the lifetime of the sensor network. This paper introduces and compares two approaches to support energy-efficient operation of TCP in sensor networks: Distributed TCP Caching (DTC) and TCP Support for Sensor networks (TSS). Both concepts allow intermediate sensor nodes to cache TCP segments and to perform local retransmissions in case of errors. This allows reducing the total number of packet transmissions in the sensor network when transferring data to or from a sensor node. DTC caches and immediately forwards TCP data segments, whereas TSS does not forward a cached segment until it knows that the previous segment has been successfully received by the next hop node. We show by simulation that both approaches significantly reduce the number of TCP segment and acknowledgement transmissions. Their performance differs slightly depending on the error rate. Both approaches have also slightly different needs in buffer requirements and TCP options to be supported.

Minimizing the Buffer Size in Fault-Tolerant Video Servers for VBR Streams

To guarantee the high reliability of video services, video servers usually adopt parity-encoding techniques in which data blocks and their associated parity blocks form a parity group. For real-time video service, all the blocks in a parity group are prefetched in order to cope with a possible disk failure, thereby incurring a buffering overhead. In this paper, we propose a new scheme called Round-level Parity Grouping (RPG) to reduce the buffer overhead while restoring VBR video streams in the presence of a faulty disk. RPG allows variable parity group sizes so that the exact amount of data is retrieved during each round. Based on RPG, we have developed a storage allocation algorithm for effective buffer management. Experimental results show that our proposed scheme reduces the buffer requirement by 20% to 25%.

Adaptive cycle management in soft real-time disk retrieval

The objective of this study is to determine the right cycle management policy to service periodic soft real-time disk retrieval. Cycle-based disk scheduling provides an effective way of exploiting the disk bandwidth and meeting the soft real-time requirements of individual I/O requests. It is widely used in real-time retrieval of multimedia data blocks. Interestingly, the issue of cycle management with respect to dynamically changing workloads has not been receiving proper attention despite its significant engineering implications on the system behavior. When cycle length remains constant regardless of varying I/O workload intensity, it may cause under-utilization of disk bandwidth capacity or unnecessarily long service startup latency. In this work, we present a novel cycle management policy which dynamically adapts to the varying workload. We develop pre-buffering policy which makes the adaptive cycle management policy robust against starvation. The proposed approach elaborately determines the cycle length and the respective buffer size for pre- buffering. Performance study reveals a number of valuable observations. Adaptive cycle length management with incremental pre-buffering exhibits superior performance to the other cycle management policies in startup latency, jitter and buffer requirement. It is found that servicing low playback rate contents such as video contents for 3G cellular network requires rather different treatment in disk subsystem capacity planning and call admission criteria because relatively significant fraction of I/O latency is taken up by plain disk overhead.

Channel folding - an algorithm to improve efficiency of multicast video-on-demand systems

Recently a number of researchers have proposed and investigated new video-on-demand architectures that make use of network multicast to achieve vastly improved efficiency. Techniques such as batching, patching, periodic broadcasting, chaining, and piggybacking have been explored both in isolation and in combinations. This study investigates a new tool in the arsenal-channel folding, where aggressive client-side caching is used to merge clients from one multicast channel into the other. In particular, this channel folding algorithm is applied to a previously proposed unified video-on-demand (UVoD)) architecture to demonstrate and to quantify its impact on the performance and the tradeoff in a multicast video distribution architecture. This paper presents this channel folding algorithm in the context of UVoD) and derives a performance model to obtain the system latency, the near-optimal channel partition policy, and the client buffer requirement. Numerical results show that channel folding can double the capacity of UVoD) with a remarkably small overhead in the client buffer requirement.

Bandwidth allocation for wireless multimedia systems with most regular sequences

In integrated wireless multimedia service, isochronous traffic of different connections can be scheduled by using a most regular binary sequence (MRBS). Such a sequence schedules traffic in an evenly spaced manner to achieve any arbitrary rate asymptotically and while avoiding excessive delay or buffering requirement. Flexible slot assignment that can match requests exactly improves bandwidth efficiency in multirate operations. The most regular binary sequence provides a distributed solution for multiaccess control that is based on limited information exchange. As a generalization, the concept of a most regular code sequence (MRCS) is proposed to support variable rate transmission in wideband code division multiple access (CDMA) systems and to provide spreading factor (SF) optimization. This scheme improves channel utilization efficiency in supporting traffics of various classes and hence results in an overall capacity gain.

An Efficient Scheduling Discipline for Packet Switching Networks Using Earliest Deadline First Round Robin

This paper addresses a frame-oriented scheduling discipline, EDF-RR (earliest deadline first round robin), for OQ (output-queued) switch architecture and data traffic consisting of fixed-length cells. Bandwidth reservation for an active session is performed by holding a number of cell slots for the session in a repeatedly-transferred frame. Each cell that is going to be transferred in the frame is assigned a virtual release time and a virtual deadline according to the bandwidth reservation scheme. The transmitting order of the cells in the frame is determined by non-preemptive non-idling EDF algorithm so that cells of a backlogged session in the frame are distributed as uniformly as possible. Through the analysis applying real-time scheduling theory and network calculus as well as network simulation, EDF-RR takes the advantage of O(1) computational complexity, and possesses tight delay bounds and lenient buffer requirements. The proposed scheduling discipline is appropriate for distributed real-time systems as we show that sessions can be configured based on message traffic models and deadline requirements. Also, a modified version of EDF-RR, called EDF-DRR, can be applied as traffic regulator when jitter requirements exist among active sessions.

Accumulation-based congestion control

This paper generalizes the TCP Vegas congestion avoidance mechanism and uses accumulation , buffered packets of a flow inside network routers, as a congestion measure based on which a family of congestion control schemes can be derived. We call this model Accumulation-based Congestion Control (ACC), which fits into the nonlinear optimization framework proposed by Kelly. The ACC model serves as a reference for packet-switching network implementations. We show that TCP Vegas is one possible scheme under this model. It is well known that Vegas suffers from round trip propagation delay estimation error and reverse path congestion. We therefore design a new Monaco scheme that solves these problems by employing an out-of-band, receiver-based accumulation estimator, with the support of two FIFO priority queues from the (congested) routers. Comparisons between these two schemes demonstrate that Monaco does not suffer from the problems mentioned above and achieves better performance than Vegas. We use ns-2 simulations and Linux implementation experiments to show that the static and dynamic performance of Monaco matches the theoretic results. One key issue regarding the ACC model in general, i.e., the scalability of bottleneck buffer requirement, and a solution using a virtual queueing algorithm are discussed and evaluated.

Memory efficient progressive rate-distortion algorithm for JPEG 2000

A novel rate-distortion optimization algorithm for JPEG 2000 is proposed. In JPEG 2000 standard, the encoder generates a number of independent compressed bit streams. But it is the rate-control algorithm that generates truncation points for these bit streams in an optimal way in order to minimize the distortion according to a target bit rate. In this letter, we propose a computationally efficient rate-distortion optimization algorithm that generates the truncation points of the compressed bit streams during the encoding procedure to avoid unnecessary coding of the possible discarded bit streams for the given bit rate. This algorithm meets memory buffer requirement for the compressed bit streams quite strictly according to a given bit rate. Moreover, before the encoding process even starts, a required memory buffer size can be estimated. This algorithm can also help avoid unnecessary encoding for some parts of an image. Hence, it is memory efficient and supports progressive encoding also. In low bit rate application the proposed algorithm can greatly reduce the encoder complexity and buffer requirement.

Memory efficient progressive rate-distortion algorithm for JPEG 2000

A novel rate-distortion optimization algorithm for JPEG 2000 is proposed. In JPEG 2000 standard, the encoder generates a number of independent compressed bit streams. But it is the rate-control algorithm that generates truncation points for these bit streams in an optimal way in order to minimize the distortion according to a target bit rate. In this letter, we propose a computationally efficient rate-distortion optimization algorithm that generates the truncation points of the compressed bit streams during the encoding procedure to avoid unnecessary coding of the possible discarded bit streams for the given bit rate. This algorithm meets memory buffer requirement for the compressed bit streams quite strictly according to a given bit rate. Moreover, before the encoding process even starts, a required memory buffer size can be estimated. This algorithm can also help avoid unnecessary encoding for some parts of an image. Hence, it is memory efficient and supports progressive encoding also. In low bit rate application the proposed algorithm can greatly reduce the encoder complexity and buffer requirement.

Performance Analysis of Downward Handoff Latency in a WLAN/GPRS Interworking System

This study presents the performance analysis of a new tight coupling based WLAN/GPRS interworking architecture. The effects of network traffic on downward handoff latency are investigated. The results indicate that increasing WLAN traffic increases the overall downward handoff latency more than increasing GPRS traffic. On the other hand, increasing GPRS traffic results in higher packet buffering requirements at the Serving GPRS Support Node (SGSN).

Efficient techniques for improved QoS performance in WDM optical burst switched networks

In this paper, we address the problems of burst scheduling, network fairness, and service differentiation in wavelength division multiplexing (WDM) optical burst switched (OBS) networks. We propose two approaches— time-slotting and burst fragmentation—to provide efficient solutions for the above problems. We also propose a path selection algorithm to distribute label switched paths (LSP) based on load balancing. Time slotting refers to the quantization of time into slots of fixed sizes. Time slotting aids fast implementation of scheduling algorithms. Further, the slotting is local to the nodes and they need not be synchronized across the network. In burst fragmentation, a burst is fragmented instead of being dropped if it cannot be accommodated on any wavelength as a whole. These fragments are then sent on different wavelengths on the same path. Only in the event, that a fragment cannot be scheduled on any wavelength of the link, is the burst dropped. We develop a new scheduling algorithm based on time-slotting and fragmentation. This algorithm has several attractive features such as fast and simple implementation and improved burst dropping performance. Using the hop-based shortest path algorithm to route a burst without accounting for the link load results in poor burst dropping performance. This is because, some links could be heavily loaded forming a bottleneck for network performance. Load balancing approach aims at reducing the bottleneck of data traffic in the network. We develop a path selection algorithm based on load balancing with the objectives of improving network performance in terms of burst dropping probability and improve fairness among bursts traversing paths of different lengths. We then develop an offset-based algorithm to provide inter-class service differentiation and intra-class fairness for the bursts belonging to classes with different levels of priority. An attractive feature of this algorithm is that it works with a constraint of maximum permissible initial offset time in order to reduce the burst transfer delay and also to reduce the buffer requirements at the ingress edge routers. It also ensures that shorter-hop bursts of a low-priority class does not perform better than the longer-hop bursts belonging to a high-priority class. Extensive simulation results have been used to demonstrate the effectiveness of the proposed approaches and algorithms for various identical and non-identical traffic demand scenarios.

Inventory Requirements Planning at Lucent Technologies

The inventory requirements planning (IRP) system is a decision-support tool that Lucent has used to determine its buffering requirements for parts in a time-phased provisioning system. It measured the deviation of actual supply and demand from their planned values and, using the historical profile of these deviations, determined the safety-stock levels Lucent needed to meet the desired service levels. In doing so, it quantified the relationships among the inventory drivers: supply intervals, demand and supply variability, desired service levels, and part proliferation. It used these drivers to identify and drive initiatives to improve processes to reduce uncertainty in the provisioning process. Several Lucent locations have used the IRP system to realign inventories and to improve service in various manufacturing and distribution facilities. The IRP system's value has been recognized through several awards. It also contributed to Lucent's receiving the Malcolm Baldrige Award in 1992 (as a part of AT&T Corporation) and subsequently the INFORMS Prize in 1998.

A hybrid scheme for multicast authentication over lossy networks

For multicast communication, authentication is a challenging problem, since it requires that a large number of recipients must verify the data originator. Many of multicast applications are running over IP networks, in which several packet losses could occur. Therefore, multicast authentication protocols must resist packet loss. Other requirements of multicast authentication protocols are: to perform authentication in real-time and to have low communication and computation overheads. In the present paper, a hybrid scheme for authenticating real-time data applications, in which low delay at the sender is acceptable, is proposed. In order to provide authentication, the proposed scheme uses both public key signature and hash functions. It is based on the idea of dividing the stream into blocks of m packets. Then a chain of hashes is used to link each packet to the one preceding it. In order to resist packet loss, the hash of each packet is appended to another place in the stream. Finally, the first packet is signed. The proposed scheme resists packet loss and is joinable at any point. The proposed scheme is compared to other multicast authentication protocols. The comparison shows that the proposed scheme has the following advantages: first, it has low computation and communication overheads. Second, it has reasonable buffer requirements. Third, the proposed scheme has a low delay at the sender side and no delay at the receiver side, assuming no loss occurs. Finally, its latency equals to zero, assuming no loss occurs.

Evaluation and improvement of response time bounds for real-time applications under non-pre-emptive Fixed Priority Scheduling

Real-time applications must be guaranteed with a bounded response time. The two main approaches for computing the upper bounds are worst-case schedulability analysis and Network Calculus, both of which are based on the analysis of a deterministic majoring trajectory. The first approach is issued from the results of Liu and Layland and gives what is called the worst-case response time for a given set of periodic tasks scheduled with fixed priority. The second approach, proposed by Cruz, gives an upper bound on delay for a set of (σ, ρ)-bounded message flows. Both approaches could be used to evaluate the end-to-end delay bound in the industrial switched Ethernet (the target application) in which the main traffic is periodic with or without jitters. However, the use of either the worst-case trajectory or (σ, ρ) trajectory produces overestimated delay bounds. Therefore, to minimize this overestimation, the paper proposes a comparative study of the delay bounds evaluated by both approaches for periodic (with or without jitters) arrival processes under Fixed Priority Scheduling. For this purpose, a relationship is given between jitter and the maximum burst size for an optimal transposition from the classical task model to the (σ, ρ)-constrained model. The paper also proposes a hybrid method to reduce the upper bound given Network Calculus for a multihop network. Numerical studies show the advantage of the method for reducing the estimation of the buffering requirement in each network element.

Assessing and improving TCP rate shaping over edge gateways

Computers installed with commercial/open-source software have been widely employed as organizational edge gateways to provide policy-based network management. Such gateways include firewalls for access control, and bandwidth managers for managing the narrow Internet access links. When managing the TCP traffic, pass-through TCP flows can introduce large buffer requirements, large latency, frequent buffer overflows, and unfairness among flows competing for the same queue. So, how to allocate the bandwidth for a TCP flow without the above drawbacks becomes an important issue. This study assesses and improves TCP rate shaping algorithms to solve the above problems through self-developed implementations in Linux, testbed emulations, live Internet measurements, computer simulations, modeling, and analysis. The widely deployed TCP rate control (TCR) approach is found to be more vulnerable to Internet packet losses and less compatible to some TCP sending operating systems. The proposed PostACK approach can preserve TCR's advantages while avoiding TCR's drawbacks. PostACK emulates per-flow queuing, but relocates the queuing of data to the queuing of ACKs in the reverse direction, hence minimizing the buffer requirement up to 96 percent. PostACK also has 10 percent goodput improvement against TCR under lossy WAN environments. A further scalable design of PostACK can scale up to 750Mbps while seamlessly cooperating with the link-sharing architecture. Experimental results can be reproduced through our open sources: 1) tcp-masq: a modified Linux kernel, 2) wan-emu: a testbed for conducting switched LAN-to-WAN or WAN-to-LAN experiments with RTT/loss/jitter emulations.

Advanced mirrored-pyramid broadcasting for video-on-demand

The client's buffer requirement, the server network bandwidth, and the access latency are three key parameters for the design of video-on-demand (VoD) system. Many schemes have been proposed to provide VoD service, one of which is the mirrored-pyramid broadcasting (MPB) scheme. However, it is not optimal with regard to the client's buffer requirement, the bandwidth utilization, and the access latency. After discussing the drawbacks of the MPB scheme, we propose an advanced mirrored-pyramid broadcasting (AMPB) scheme, which has a novel channel design and a novel broadcast schedule. Experimental results show that AMPB saves up to 75% of the client's buffer and 50% of the server network bandwidth. It also achieves a better access latency.

On-Board Satellite “Split TCP” Proxy

Several satellite systems currently in operation or under development claim to support broadband Internet applications. In these scenarios, transmission control protocol (TCP) plays a critical role. Unfortunately, when used with satellite links, TCP suffers from a number of well-known performance problems, especially for higher data rates and high altitude satellites with longer delays. In response to these difficulties, the satellite and Internet research communities have developed a large gamut of solutions ranging from architectural modifications to changes in the TCP protocol. Among these, one approach requiring minimal modifications involves splitting the TCP connection in two or more segments with one segment connecting terrestrial nodes across the satellite network. In this paper, we consider an evolution of this idea: placing a TCP proxy on board the satellite that further subdivides the end-to-end connection into separate TCP connections between ground and space. We focus upon the efficient use of buffer resources on board the satellite, while at the same time enhancing TCP performance. We evaluate two TCP protocol versions, TCP NewReno and TCP Westwood. We consider various geosynchronous earth orbit satellite scenarios, with and without the split proxy, and with different channel error conditions (random errors, shadowing, etc.). Using simulation, we show that an on-board proxy provides a number of distinct advantages and can enhance throughput up to threefold for both TCP New Reno and TCP Westwood, in some scenarios, with relatively modest on-board buffering requirements. The main contributions of this paper are: the on-board split proxy concept, the buffer management strategy, the use of a realistic "urban shadowing" model in the evaluation, and the extensive comparison of the recently announced TCP Westwood with the traditional TCP New Reno.