Transmission Buffer Research Articles

An input traffic allocation strategy and an efficient transmission technique for collisions-free WDM ring MANs

This paper presents a wavelength division multiplexing multi-ring architecture with a separate control wavelength for metropolitan area networks. We study the critical problem of low fiber bandwidth utilization that many access ring protocols introduce, especially at high load conditions. In our study this problem is effectively faced by: (1) applying an efficient slotted access algorithm to avoid both data wavelengths and receiver collisions, (2) introducing a number of buffers at each node to effectively allocate the incoming traffic into them, and (3) applying an efficient algorithm for buffer selection for transmission that combines the priority criteria of packet age and receiver collisions avoidance. In this way, we obtain maximum fiber bandwidth utilization. Especially, our aim is to achieve not only significant throughput enhancement, but also essential dropping probability and total delay reduction. Also, the number of transmission buffers per node is investigated in order to manage maximum throughput improvement, while considering the financial cost. Finally, a discrete-event simulation model based on Poisson statistics is developed for the performance measures evaluation.

A High-Performance Deadlock-Free Multicast Routing Algorithm for K-Ary N-Cubes

Current multicast routing algorithms for multiprocessor systems are based on the reservation of transmission resources and buffer space for messages before transmission. This strategy causes cyclic dependencies among messages and, in turn, deadlock situations. This paper proposes and evaluates the performance of a new multicast routing algorithm, called Deadlock-Free Multicast Routing (DFMR), which removes the root cause of deadlocks. DFMR prevents deadlocks by allowing nodes to send flits as soon as internode links are available for transmission. This is obtained by allowing any possible interleaving of flits from all messages on internode links. The resulting routing algorithm eliminates the need to implement deadlock avoidance, detection, and recovery mechanisms, but needs larger output buffers. In DFMR, message flits carry additional information overhead which is used to avoid deadlock. Simulation results show that this information overhead has a negligible impact on overall performance which is considerably greater than that of previous algorithms.

Random scheduling based on transmission delay and buffer size for hop-based burst-cluster transmission in OBS networks

Hop-based burst-cluster transmission improves fairness significantly in OBS ring networks. However, in mesh networks, the overall burst loss probability increases using the method because the transmission of a burst-cluster from an ingress node is often synchronized with a burst-cluster from another ingress node. In order to resolve the synchronization issue, we propose in this article the utilization of random scheduling for hop-based burst-cluster transmission. In random scheduling, the maximum waiting time is determined in advance from the maximum acceptable delay of an IP packet and the buffer size of the ingress node. Every time a burst-cluster is generated, a scheduler derives the actual waiting time at random based on the obtained maximum waiting time. When a timer becomes the actual waiting time, the burst cluster is transmitted from the scheduler. By simulation, we evaluate the effectiveness of random scheduling for NSFNET. Numerical examples show that the random scheduling can resolve the synchronization issue and can decrease the overall burst loss probability. In addition, it is shown that random scheduling can also improve the fairness significantly.

On the Buffer Occupancy of an IEEE 802.11 Station in a Hot-Spot

In this paper we evaluate analytically the average occupancy of the transmission buffer of a 802.11 station (STA). The station belongs to a Wi-Fi Hot-Spot and exchanges data with a fixed host. The data exchange is regulated by the Transmission Control Protocol (TCP). The research interest is motivated by the fact that several papers assume that in these conditions the STA buffer is nearly empty. On the contrary, we prove that this assumption may be wrong and discuss the consequences of this fact. We test the proposed model by means of ns2 simulation, ascertain its accuracy, and highlight its limits.

Adaptive Video Streaming Using Bandwidth Estimation for 3.5G Mobile Network

Currently deployed mobile networks including High Speed Downlink Packet Access (HSDPA) offer only best-effort Quality of Service (QoS). In wireless best effort networks, the bandwidth variation is a critical problem, especially, for mobile devices with small buffers. This is because the bandwidth variation leads to packet losses caused by buffer overflow as well as picture freezing due to high transmission delay or buffer underflow. In this paper, in order to provide seamless video streaming over HSDPA, we propose an efficient real-time video streaming method that consists of the available bandwidth (AB) estimation for the HSDPA network and the transmission rate control to prevent buffer over-flows/underflows. In the proposed method, the client estimates the AB and the estimated AB is fed back to the server through real-time transport control protocol (RTCP) packets. Then, the server adaptively adjusts the transmission rate according to the estimated AB and the buffer state obtained from the RTCP feedback information. Experimental results show that the proposed method achieves seamless video streaming over the HSDPA network providing higher video quality and lower transmission delay.

Analysis of Stop-and-Wait ARQ for a wireless channel

In this paper, we study the behavior of the transmitter buffer of a system working under a Stop-and-Wait retransmission protocol. The buffer at the transmitter side is modeled as a discrete-time infinite-capacity queue. The numbers of information packets entering the buffer during consecutive slots are assumed to be independent and identically distributed random variables. The packets are sent over an unreliable channel and transmission errors occur in a correlated manner. Specifically, the probability of an erroneous transmission is modulated by a two-state Markov chain. An expression is derived for the probability generating function of the buffer content. This expression is then used to derive several queue-length characteristics and the mean packet delay. Numerical examples illustrate the strong effect of error correlation on the system performance. The obtained analytical results are also compared with appropriate simulations.

A real-time video watermarking system with buffer sharing for video-on-demand service

Real-time watermarking for streaming video (such as VOD service) requires significant amounts of computing resources. To address this issue, we present a scalable watermarking scheme integrated in a parallel MPEG-2 engine. A content-based block selection algorithm is proposed to efficiently embed the pseudo-random watermark signatures into DCT blocks. Our watermark scheme also provides a robust way to synchronize the watermarked video to the original source at detectors and is very resilient against cumulative and temporal attack. We optimize the parallel watermark engine to achieve real-time watermarking performance. We found that the system throughput could suffer significant degradation when processing high-level MPEG-2 video (such as HDTV) due to inefficient management of memory space. Therefore, we investigated an efficient buffer management scheme consisting of two methods: First we reduced the transmission buffer in slave nodes by frames sharing between frames in the Group-of-Picture (GOP) level. Then we further reduce the buffer space by a dynamic on-demand allocation on the slave side. By solving the memory-shortage bottleneck, the proposed system can support real-time watermarking for multiple high-resolution (up to 1404 × 960) video.

An integrated rate control scheme for TCP-friendly MPEG-4 video transmission

Delivering MPEG-4 real-time video over the Internet requires the use of rate control schemes in both the compression and the transport domains. This paper discusses the practical difficulties involved in integrating MPEG-4 scalable rate control (SRC) in the compression domain with TCP-friendly rate control (TFRC) in the transport domain. In addition to the requirement for rate matching between the two schemes to ensure TCP-friendliness, users demand a high-quality, smooth media output under the varying network conditions. In this paper, an integrated rate control scheme is proposed for MPEG-4 video transported by a TFRC protocol. The proposed scheme comprises a rate smoothing control mechanism to meet the rate matching with the controlled transmission buffering delay, and an adaptive rate control for the I-frame designed to reduce the frame skipping. Through the successful integration of the two schemes, the experimental results show that the proposed scheme better preserves the motion smoothness and improves the overall perceptual quality for TCP-friendly MPEG-4 video transmissions to obtain a smooth, high visual quality video experience.

An efficient cross layer scheduler for multimedia traffic in wireless local area networks with IEEE 802.11e HCCA

This paper proposes a scheduling algorithm, namely Wireless Timed Token Protocol (WTTP), for the Hybrid Coordination Function (HCF) Controlled Channel Access (HCCA) in IEEE 802.11e. WTTP provides traffic streams with a minimum reserved rate, as required by the standard, and it accounts for two types of traffic streams simultaneously, depending on the corresponding application: constant bit rate, which are served according to their rate, and variable bit rate traffic streams. The latter are guaranteed a minimum rate, but they are also allowed to exploit unused bandwidth, which preserves small access delays in case of bursty arrivals. Additionally, WTTP shares the capacity which is not reserved for QoS traffic streams transmissions among traffic flows with no specific QoS requirements. We also propose a strategy for the QoS Access Point to infer the idle/busy status of the uplink transmission buffers, based on cross layer information made available at the MAC by the application layer, which allows channel capacity to be saved. WTTP exhibits O(1) per packet computational complexity. We evaluate the performance of WTTP via simulation under different traffic conditions, and we investigate its resilience to variations of different system parameters.

Traffic Estimation and Capacity Assignment in Multimedia Distribution Networks with Guaranteed Quality of Service

This paper considers the provisioning of transmission line bandwidth on a private network with given traffic routing for the purpose of distribution of video-on-demand service with guaranteed end-to-end quality of service. We present an architecture for video-on-demand service delivery and model the assignment of bandwidth in the distribution network as a constrained, nonlinear optimization problem. To solve this optimization problem, we develop three new auction algorithm-based solution procedures. The optimization problem assumes that there is a functional relationship between the maximum acceptable end-to-end delay and the bandwidth requirement for the links in the distribution network. Absent reliable video traffic data models for MPEG-2 format, we sample a large number of DVD-recorded movies to form a basis for randomly generated aggregate traffic streams. The aggregate traffic streams are used in a simulation experiment to measure the maximum transmission buffer occupancy for each given traffic stream for different transmission rates. Based on this simulation experiment, we derive an empirical transmission line provisioning function that guarantees delivery of all video frames without frame loss within a maximum frame delay tolerance. To illustrate the effectiveness of the proposed solution procedure for the bandwidth assignment problem, we solve to near optimality 570 small problem instances under three demand structures and 100 large problem instances with uniformly distributed demand.

A New Call Admission Control Algorithm for VoIP in IEEE 802.11 Wireless

In the past few years, the IEEE 802.11 WLAN has become very popular and widely deployed for Internet access. On the other hand, voice over IP is one of the fast growing Internet applications today. Thanks to the convergence of these two trends, it is believed that VoIP over WLAN is expected to become an important Internet applications. In this context, the so called “avalanche effect” has been identified as a real problem in a WLAN network, when operating near its capacity limit, in which the admission of an additional call may result in unacceptable QoS for all the ongoing VoIP connections. Therefore, this work proposes a new call admission control mechanism, based on the access point’s transmission buffer utilization ratio, to mitigate this problem. The obtained results show that the proposed mechanism mitigates the “avalanche effect”, leading to stable service operation, acceptable QoS levels and significant capacity gains considering tested scenarios.

Queue Analysis and Multiplexing of Heavy-tailed Traffic in Wireless Packet Data Networks

Recent research based on traffic measurements shows that Internet traffic flows have a fractal nature (i.e., self-similarity property), which causes an underestimation of network engineering parameters when using the conventional Poisson model. Preliminary field measurements demonstrate that packet data traffic in wireless communications also exhibits self-similarity. In this paper, we investigate the queuing behavior of self-similar traffic flows for data applications in a packet-switching single-server wireless network. The traffic is generated by an on-off source with heavy-tailed on periods and exponentially distributed off periods. We extend previous analysis of a relation among the asymptotic distribution of loss probability, traffic specifications, and transmission rate for a wireline system to a wireless system, taking into account wireless propagation channel characteristics. We also investigate the multiplexing of heavy-tailed traffic flows with a finite buffer for the downlink transmission of a wireless network. Computer simulation results demonstrate that assumptions made in the theoretical analysis are reasonable and the derived relationships are accurate.

SR ARQ packet delay statistics on markov channels in the presence of variable arrival rate

In this letter we investigate the packet delay statistics of a fully reliable selective repeat ARQ scheme by considering a discrete time Markov channel with non-instantaneous feedback and assigned round-trip delay m. Our focus is on studying the impact of the arrival process on the delay experienced by a packet. An exact model is introduced to represent the system constituted by the transmitter buffer, the m round-trip slots, and the channel state. By means of this model, we evaluate and discuss the delay statistics and we analyze the impact of the system parameters, in particular the packet arrival rate, on the delay statistics

Resource Allocation Under Uncertainty Using the Maximum Entropy Principle

In this paper, we formulate and solve a problem of resource allocation over a given time horizon with uncertain demands and uncertain capacities of the available resources. In particular, we consider a number of data sources with uncertain bit rates, sharing a set of parallel channels with time-varying and possibly uncertain transmission capacities. We present a method for allocating the channels so as to maximize the expected system throughput. The framework encompasses quality-of-service (QoS) requirements, e.g., minimum-rate constraints, as well as priorities represented by a user-specific cost per transmitted bit. We assume only limited statistical knowledge of the source rates and channel capacities. Optimal solutions are found by using the maximum entropy principle and elementary probability theory. The suggested framework explains how to make use of multiuser diversity in various settings, a field of recently growing interest in communication theory. It admits scheduling over multiple base stations and includes transmission buffers to obtain a method for optimal resource allocation in rather general multiuser communication systems.

A Simple Packet-Transmission Scheme for Wireless Data Over Fading Channels

In this letter, we present a simplified scheduling scheme for packet transmission over a fading channel which is modeled as a finite-state block channel. We first address the optimal minimum power-transmission policy with constraints on both average delay and packet loss. The optimal policy requires a sophisticated table lookup for implementation. In order to alleviate this problem, we design a simplified transmission policy that is based on checking for three control parameters: a transmission-rate threshold; a channel-state threshold; and the transmission buffer size. Our results show that the minimum average power with the simplified scheme is very close to that achieved by the optimal policy. Further, modeling the simplified scheme via a single-server bulk-service queue, we derive approximations to the quality of service metrics, such as average power and delay, that allow for analytical insights into the policy design.

MPEG-4 Video Transmission over Wireless Networks: A Link Level Performance Study

With the scalability and flexibility of the MPEG-4 and the emergence of the broadband wireless network, wireless multimedia services are foreseen to become deployed in the near future. Transporting MPEG-4 video over the broadband wireless network is expected to be an important component of many emerging multimedia applications. One of the critical issues for multimedia applications is to ensure that the quality-of-service (QoS) requirement to be maintained at an acceptable level. This is further challenged in that such a service guarantee must be achieved under unreliable and time-varying wireless channels. In this paper we study the link level performance of MPEG-4 video transmission over the uplink of an unreliable wireless channel. We introduce the discrete time batch Markovian arrival process (DBMAP) with two types of arrivals to model the MPEG-4 video source, which takes into account the inherent nature of the adaptiveness of the video traffic. We prove that in a hidden Markov modeled (HMM) wireless channel with probabilistic transmission, the service time for an arbitrary radio link control (RLC) burst follows phase type (PH-type) distribution. We show that the link level performance of a wireless video transmission system can be modeled by a DBMAP/PH/1 priority queue, and present computation algorithm and numerical results for the queueing model. Extensive simulations are carried out on the queueing behavior of the video transmission buffer, as well as on the packet level error behavior of the video data. The results demonstrate that video quality can be substantially improved by preserving the high priority video data during the transmission.

Real-Time Traffic Scheduling and Routing in Packet-Switched Networks Using a Least-Laxity-First Strategy

In this paper, we address the problem of transmission scheduling and buffer management at a switch in a high-speed packet-switched network. Future computer networks are expected to carry bursty real-time multimedia traffic with specific time-delay requirements. We propose a technique to schedule competing packets at a switch in a packet-switched network with two objectives--(i) maximizing the number of packets that would reach their destination before their deadline has been reached (ii) minimizing the number of packets dropped in transit in the network. The method is broadly based on the popular least-laxity heuristic in real-time process scheduling and is combined with buffer management at the switch.

A collaboration model for better reliability and performance for object-based distributed applications over the Internet

The improved consecutive message transmission and adaptive buffer control (ICMTABC) collaboration model proposed in this paper is an enhancement of the existing CMTABC version. The motivation is to provide better collaboration performance for distributed applications, such as network-based computer-supported cooperative design. The enhancement is achieved through better reliability for end-to-end communication channels over the Internet. The CMT and ABC mechanisms are adaptive; the former improves message transmission and reception and the latter prevents message loss due to buffer overflow at the reception side. The anticipative capability of the ICMTABC framework comes from the convergence algorithm (CA) element, which is an internet end-to-end performance measurement (IEPM) mechanism that predicts the mean message response time of a communication channel quickly, responsively and accurately. The CA predictions provide the basis for computing the QoS factors needed by CMT and ABC for anticipative and adaptive control.

Queueing analysis of some continuous ARQ strategies with repeated transmissions

The statistical analysis of a class of continuous automatic repeat request (ARQ) strategies is performed. In the considered class, an optimal (throughput-maximising) number of copies of each data block is sent contiguously instead of one single copy. Explicit formulas for the probability generating functions of the queue length and the delay time in the transmitter buffer are obtained.

Transmitter buffer behaviour of stop-and-wait ARQ schemes with repeated transmissions

The statistical analysis of a number of previously proposed ARQ protocols of the stop-and-wait type is performed. Specifically, a method is described to find the tail distribution (i.e. the probability of exceeding a certain threshold) for the delay time of the transmitted data blocks and the queue or buffer occupancy at the transmitter side. A performance comparison of the schemes is given, based on both throughput and tail probability considerations.