Loss probabilities for messages with redundant packets feeding afinite buffer

The purpose of this paper is to obtain the distribution of the number of lost packets within a sequence of n consecutive packet arrivals into a finite buffer M/M/1 queue. We obtain explicit expressions for the multidimensional generating function of these probabilities based on a recursive scheme introduced by Cidon et al. (1993). We then analyze the loss probabilities of a whole message, and analyze the effect of adding redundant packets. We show that in both heavy traffic as well as in light traffic conditions, adding redundant packets results in decreasing the message loss probabilities     

Packet throughput in slotted ALOHA DS/SSMA radio systems withrandom signature sequences

Packet throughput figures are obtained for direct sequence spread spectrum multiple access (DS/SSMA) slotted ALOHA radio systems where all users employ random signature sequences from bit-to-bit within all transmitted packets. These calculations use an improved Gaussian approximation technique which gives accurate bit error probabilities and also incorporates the effect of bit-to-bit error dependence within each packet in the multiaccess interference environment. Numerical results are given for packets which employ varying amounts of block error control, and a comparison is made with results obtained by other methods which ignore the effects of bit-to-bit error dependence within each packet in the multiaccess interference environment. Numerical results are given for packets which employ varying amount of block error control, and a comparison is made with results obtained by other methods which ignore the effects of bit-to-bit error and/or employ less-accurate Gaussian approximations to the probability of data bit error. Maximum throughput per unit bandwidth figures are calculated which compare favorably to similar figures for narrowband signaling techniques     

Performance analysis of a random packet selection policy formulticast switching

This paper studies a random packet selection policy for multicast switching. An input packet generates a fixed number of primary copies plus a random number of secondary copies. Assuming a constant number of contending packets during a slot, the system is modeled as a discrete time birth process. A difference equation describing the dynamics of this process is derived, the solution of which gives a closed form expression for the distribution of the number of packets chosen. Then this result is extended to the steady state distribution through a Markov chain analysis. It is shown that the old packets have larger fanout than the fresh packets and the copy distribution of the mixed packets is determined. The packet and copy throughput taking into account the old packets have been obtained. We determined the mean packet delay as well as an upperbound for packet loss probabilities for finite buffer sizes. The asymptotic distribution of the number of packets is also given for large switch sizes under saturation by applying results from the renewal theory. Finally, simulations are done to determine the performance of the switch under mixed (unicast plus multicast) traffic     

Queueing analysis of delay constrained voice traffic in a packetswitching system

Unlike data traffic, the voice packet stream from a node has very high correlation between consecutive packets. In addition, in order for the speech to be properly reconstructed, a delay constraint must be satisfied. A queueing model that accurately predicts packet loss probabilities for such a system is presented. Analytical results are obtained from an embedded bivariate Markov chain and are validated by a simulation program. Based on this model, the impact of the delay constraint, talkspurt detection thresholds, and packet size on packet loss are studied. Two schemes, named `instant' and `random', for discarding late packets are considered. Simulation results show that better performance can be obtained by using the latter scheme     

Sizing a packet reassembly buffer at a host computer in an ATMnetwork

This paper develops a queueing model of a buffer that collects cells for reassembly into packets for a protocol layer above the asynchronous transfer mode (ATM) layer. Whenever the buffer fills with all packets incomplete, a packet must be sacrificed to make room for others. The queueing model estimates the equilibrium fraction of packets sacrificed under one algorithm for selecting the packet to be sacrificed. The paper also uses simulation to compare three sacrifice algorithms. The model's predicted packet loss probabilities bound from above the loss probabilities in the simulations of the different algorithms. Applications to sizing the buffer for a prescribed loss probability are given     

Multimedia Broadcast Authentication Based on Batch Signature [Advances in Mobile Multimedia]

Conventional block-based broadcast authentication protocols overlook the heterogeneity of receivers in mobile computing by letting the sender choose the block size, divide a broadcast stream into blocks, associate each block with a signature, and spread the effect of the signature across all the packets in the block through hash or coding algorithms. They suffer from some drawbacks. First, they require that the entire block with its signature be collected before authenticating every packet in the block. This authentication latency can lead to the jitter effect on real-time applications at receivers. Second, the block-based approach is vulnerable to packet loss in mobile computing in the sense that the loss of some packets makes the other packets unable to be authenticated, especially when the block signature is lost. Third, they are also vulnerable to DoS attacks caused by the injection of forged packets. In this article we propose a novel broadcast authentication protocol based on an efficient cryptographic primitive called a batch signature. Our protocol supports the verification of the authenticity of any number of packets simultaneously and avoids the shortcomings of the block-based approach.     

Buffering management schemes for optical variable length packets under limited packet sorting

This paper studies several buffering strategies for optical packet switching (OPS) under limited packet sorting. Three schemes, which are able to sort newly arrived packets based on packet’s length as well as capability of finding the minimum buffer occupancy, are analyzed and compared. Results show that all three proposed schemes could improve OPS performance considerably in terms of probability of packet loss (PPL) and probability of information loss (PIL). In addition, the simulation results show that not all the newly arrived packets need to be sorted in order to obtain minimum packet loss probability. Since the amount of packets and thus the packet processing time is significant in OPS, it is possible that not all the packets can be processed using one of the buffering strategies. An important finding of this paper is that if only 10% of the packets are sorted, the PPL is comparable to the minimum packet loss value obtained when 100% of the packets are sorted.     

Scheduling algorithms for shared fiber-delay-line optical packet Switches-part II: the three-stage clos-network case

In all-optical packet switching, packets may arrive at an optical switch in an uncoordinated fashion. To prevent packet loss in the switch, fiber delay lines (FDLs) are used as optical buffers to store optical packets. However, assigning FDLs to the arrival packets to achieve high throughput, low delay, and low loss rate is not a trivial task. In the authors' companion paper, several efficient scheduling algorithms were proposed for single-stage shared-FDL optical packet switches (OPSs). To further enhance the switch's scalability, this work was extended to a multistage case. In this paper, two scheduling algorithms are proposed: 1) sequential FDL assignment and 2) multicell FDL assignment algorithms for a three-stage optical Clos-Network switch (OCNS). The paper shows by simulation that a three-stage OCNS with these FDL assignment algorithms can achieve satisfactory performance.     

Keeping the packet sequence in optical packet-switched networks

This paper deals with optical packet switches with limited buffer capabilities, subject to asynchronous, variable-length packets and connection-oriented operation. The focus is put on buffer scheduling policies and queuing performance evaluation. In particular a combined use of the wavelength and time domain is exploited in order to obtain contention resolution algorithms that guarantee the sequence preservation of packets belonging to the same connection. Four simple algorithms for strict and loose packet sequence preservation are proposed. Their performance is studied and compared with previously proposed algorithms. Simulation results suggest that by accepting some additional processing effort it is possible to guarantee very low packet loss probabilities while avoiding the out-of-sequence delivery.     

异步光分组交换网的流量建模

研究了异步光分组交换网的流量特性，提出了网络流量的解析模型和近似模型。研究表明，在采用计时门限光分组组装算法的情况下，如输入IP流具有短程相关特性(ShortRangeDependent)，则光分组的到达间隔时间呈负指数分布，光分组的长度趋于高斯分布。     

Output and delay of multi-channel slotted ALOHA systems for integrated voice and data transmission

Packet-switched technology has been developed to offer personal communication services not only for data but also for different types of user-end equipment such as phone-type audio. To satisfy the huge service demand and multi-traffic requirements with limited bandwidth, this paper proposes an efficient procedure of multi-channel slotted ALOHA for integrated voice and data transmission in wireless information networks and presents an exact analysis with which to numerically evaluate the performance of the systems. A channel reservation policy is applied, where a number of channels (called reserved channels) are used exclusively by voice packets, while the remaining channels are used by both voice and data packets, and voice packets select the reserved channels with a given probability (called selection probability). Probability distributions for the numbers of voice and data departures and for the data packet delay are derived. Numerical results compare some cases with different numbers of channels, different numbers of reserved channels and different selection probabilities to discuss what effects they may have on channel utilization, loss probability, average packet delay, coefficient of variation of data packet delay, and correlation coefficient of packet departures. This revised version was published online in June 2006 with corrections to the Cover Date.     

On spatially disjoint lightpaths in optical networks

Within the communication networks, a delayed constrained data packet is the one that will be dropped if not being served before a certain deadline time, which causes data packet loss affecting the quality of service (QoS). In this paper, we study the blocking probability and the mean delay of such delay constrained packets in an asynchronous single-wavelength optical buffer in optical packet switching networks, where the packet arrival process follows the Poisson process and the packet-length distribution is assumed to be general. We obtain the integral equations of the modeled system and the exact expressions of blocking probabilities and the mean delays. Numerical examples are provided to validate the results with interesting observations being highlighted.     

Hierarchical constellation for multi-resolution data transmission over block fading channels

We propose a new technique for multi-resolution video/image data transmission over block fading channels. The proposed scheme uses an adaptive scheduling protocol employing a retransmission strategy in conjunction with a hierarchical signal constellation (known also as nonuniform, asymmetric, multi-resolution constellation) to give different transmission priorities to different resolution levels. Transmission priorities are given in terms of average packet loss rate as well as average throughput. Basically, according to the transmission scheduling and channel state (acknowledgment signal) of the previous transmission, it dynamically selects packets from different resolution levels to transmit for the current transmission. The bits from the selected packets are assigned to different hierarchies of a hierarchical 4/16-quadrature amplitude modulation to transmit them with different error protections. The selection of packets for transmission and the assignment of these selected packets to different hierarchies of the hierarchical constellation are referred to as the scheduling protocol in our proposed scheme. We model this protocol by a finite state first order Markov chain and obtain the packet loss rate and the packet transmission rate over Nakagami-m block fading channel in closed-form. Some selected numerical results show that the proposed scheme can control the relative packet loss rate and the packet transmission rate of different resolution levels by varying the priority parameter (or equivalently, the asymmetry) of the hierarchical constellation and the maximum number of allowed retransmissions.     

分组网络上的视频流传输面临的挑战及对策

视频已成为通信、娱乐中十分重要的媒介，视频流传输在因特网上得到了快速发展及广泛应用。分组网络所呈现出的异构性、时变性等本质属性，使得基于分组网络的视频流传输面临许多挑战。首先，当分组从一个节点传输到另一个节点时，由于其所经历的实际网络路径不同，路由器及通信信道的性能和表示这些分组连接的参数(如带宽、分组丢失概率及分组延迟)也会彼此不同。甚至会出现几个数量级大的差异。此外，上述参数还会随着时间发生变化。本从端主机的角度出发，分析在分组网络视频流传输过程中面临的挑战，并提出相应对策。     

Variable IP packets buffer management schemes for optical packet switching

Effective buffering of optical packets is essential to the efficient working of optical packet switches. In this paper three new schemes, which involve sorting and finding the least occupied buffer, are proposed. Their performance is compared with the common round-robin scheme. The results show that all these new schemes are able to enhance the optical packet switch performance significantly in terms of packet drop/loss probability. In addition, the results show that not all the newly arrived packets need to be sorted in order to obtain the minimum packet drop probability. As computation/processing time is significant in optical packet switching, partial sorting of the newly arrived packets with tolerable packet drop probability appears to be a viable proposition. Conversely, a complete sort of newly arrived packets wastes packet processing time unnecessarily while significantly increasing the packet drop probability.     

The probability of multiple correct packet receptions indirect-sequence spread-spectrum networks

We provide methods to evaluate the probabilities P(l, m-l|K), l=0, 1, ..., m and m⩽K of exactly l correct packet receptions in a group of m receivers, given that K packets are transmitted simultaneously from users employing direct-sequence spread spectrum (DS/SS) signalling schemes. This quantity is useful for the design and performance evaluation of protocols for admission control and dynamic code allocation in multiple-access spread spectrum packet radio networks intended for terrestrial or satellite applications. The evaluations are carried out for DS/SS networks employing BPSK modulation with coherent demodulation and convolutional codes with Viterbi decoding. Systems with geographically dispersed receivers and systems with colocated receivers are considered. Approximations based on the independent receiver operation assumption (IROA) and the Gaussian multivariate distribution are developed, and their accuracy is checked against the exact expressions derived for synchronous systems. The joint first error event approximation (JFEEA) is also developed for coded systems and compared to the IROA     

Managing QoS requirements for video streaming: from intra‐node to inter‐node

Streaming video over IP networks has become increasingly popular; however, compared to traditional data traffic, video streaming places different demands on quality of service (QoS) in a network, particularly in terms of delay, delay variation, and data loss. In response to the QoS demands of video applications, network techniques have been proposed to provide QoS within a network. Unfortunately, while efficient from a network perspective, most existing solutions have not provided end‐to‐end QoS that is satisfactory to users. In this paper, packet scheduling and end‐to‐end QoS distribution schemes are proposed to address this issue. The design and implementation of the two schemes are based on the active networking paradigm. In active networks, routers can perform user‐driven computation when forwarding packets, rather than just simple storing and forwarding packets, as in traditional networks. Both schemes thus take advantage of the capability of active networks enabling routers to adapt to the content of transmitted data and the QoS requirements of video users. In other words, packet scheduling at routers considers the correlation between video characteristics, available local resources and the resulting visual quality. The proposed QoS distribution scheme performs inter‐node adaptation, dynamically adjusting local loss constraints in response to network conditions in order to satisfy the end‐to‐end loss requirements. An active network‐based simulation shows that using QoS distribution and packet scheduling together increases the probability of meeting end‐to‐end QoS requirements of networked video. Copyright © 2005 John Wiley & Sons, Ltd.     

On the error probability of coded frequency-hopped spread-spectrummultiple-access systems

A simple, exact calculation is presented of the probability distribution of the number of hits in a block of n symbols in a frequency-hopped, spread-spectrum, multiple-access communication system. While the sequence of hits is not Markovian, there is an underlying Markovian structure that allows the probability distribution of the number of hits to be calculated in a recursive fashion. Knowing the probability distribution of the number of hits makes it possible to calculate the probability of error for a system employing error correcting codes for several different types of receivers, including receivers with both errors and erasures. The numerical results show that both the approximation obtained by assuming the actual sequence of hits is Markovian and the approximation obtained by assuming the hits are independent are very good. When the number of frequency slots is not too small (less than five), calculations show that assuming the independence of hits gives an error probability accurate to within 1% of the actual error probability. Assuming the hits are Markovian gives error probabilities which are accurate to within 0.001%     

Packet loss due to encryption in space data systems

This paper analyzes the probabilities of data packet loss for both an encrypted channel in self-synchronous cipher feedback mode and a nonencrypted channel, in the space data systems. Simulation results show reasonable agreement with analytical results. When channel bit error probability is 10^-5 and the total number of packets per frame is 3, the analytical model gives 0.39% packet loss while the simulation gives 0.22% packet loss due to encryption. Although the analysis is performed for the space data systems, the resulting derived equations with minor change will be useful in many packet communication applications     

Performance Evaluation of a Packetized Voice System--Simulation Study

Introduction of the packet switching technique into digitized voice communication may afford great advantages in efficient use of the channel, compared to both circuit-switched and DSI systems. Detailed characteristics, however, have not been obtained because of difficulty in the exact analysis. Hence, simalation models are developed in this paper for the packetized voice transmission system, and various characteristics such as tranmission delays and loss probability of voice packets are obtained. We further evaluate three types of voice packet reassembly strategy at the receiving terminal, and obtain the optimal packet length, which keeps both overall packet transmission delay and packet loss probabilty less than a certain permissible value. Comparison among three strategies is also stated.