Analysis of packet loss in a continuous-time finite-buffer queue with multimedia traffic streams

The main peculiarity of multimedia services is the presence of intermedia relationships, that is, time relationships between the various media making up the multimedia stream. These, in fact, affect the characteristics of the traffic offered to the network and therefore they have to be taken into account when network resource allocation and management strategies are investigated. To achieve this aim we propose, in this study, to model a multimedia source as the superposition of heterogeneous correlated ON–OFF processes, each of which models a monomedia source in a continuous-time environment. The performance of a finite-size queue driven by a number of heterogeneous multimedia sources modelled according to the proposed paradigm is studied. Finally, assuming a master/slave relationship between the monomedia streams composing the multimedia stream, a case study is presented in order to show how intermedia relationships and their effects on buffer performance can be described through the auto- and crosscorrelation functions of the compound processes and the autocorrelation function of the multimedia source as a whole. ©1997 by John Wiley & Sons, Ltd.     

Robust rate control for integrated services packet networks

Research on congestion-control algorithms has traditionally focused more on performance than on robustness of the closed-loop system to changes in network conditions. As the performance of the control loop is strictly connected with the quality of service, these systems are natural candidates to be approached by the optimal control theory. Unfortunately, this approach may fail in the presence of transmission delay variations, which are unavoidable in telecommunication systems. In this paper, we first show the fragility of optimal controllers and demonstrate their instability when the control delay is not known exactly. Then we propose a robust control algorithm based on a classical proportional integral derivative scheme which does not suffer from this fragility phenomenon. Its stability versus the control delay variations, as well as versus sources that transmit less than their computed share, is studied with Nyquist analysis. The control algorithm is implemented within a simulator in the framework of the asynchronous transfer mode (ATM) ABR transfer capability. The final part of the paper shows some selected results assessing the performance of the control algorithm in a realistic network environment. ABR was chosen as an example, but the control studied here can be applied in any data network to obtain a robust and reliable congestion-control scheme.     

基于实测流量的网际分组丢失率分析

分组丢失率是评价网络性能的一个重要指标。提出了一种利用完整TCP流计算实测数据中分组丢失率的方法,并基于实测数据对CERNET江苏省网与外部不同网络分区的分组丢失率进行分析。算法给出了基于实测数据的分组丢失率包括首次分组丢失率pfl、整体分组丢失率psl和重传率prp 3个测度。统计分析表明,在绝大多数情况下,pfl比psl小,而prp比pfl和psl明显要高,验证了分组丢失对网络性能的影响以及重传率明显比分组丢失率高。     

Blocking probability analysis for packet circuit emulation services

Standardization of pseudo-wire and circuit emulation services on IP packet networks is underway. The ability to quickly find circuits in networks supporting circuit emulation is directly computed using blocking probability analysis. When all of the sequenced flows have the same maximum packet size, i.e., they are homogeneous, blocking probabilities are computed using Erlang's B formula. This paper describes analysis tools for the heterogeneous case and introduces the concept of steady-state fragmentation blocking probability, which controls the size of state transition diagrams used in the analysis. Simulation results using these tools are presented.     

Fair channel-adaptive rate scheduling in wireless networks with multirate multimedia services

In this paper, we study the fundamental problem of allocating with efficiency and fairness the available resources in a code-division multiple-access wireless system that supports multirate multimedia services. Our proposed approach adopts the use of dynamically assigned data rates that match the channel capacity in order to improve the system throughput and overcome the problems associated with the location-dependent and time-dependent errors and channel conditions, the variable system capacity and the transmission power limitation. We introduce and describe two new algorithms, namely the channel adaptive rate scheduling (CARS) and fair channel adaptive rate scheduling (FCARS) algorithms. CARS improves the system throughput by adjusting the transmission rates according to the varying channel conditions and performs an iterative procedure to determine the power index that a user can accept by its current channel condition and transmission power. Based on the assignment of CARS, FCARS achieves the objective of fairness by further compensating the lagging users, while still maintaining all the constraints imposed by the system. The performance evaluation process confirms that our approach achieves, simultaneously, the design objectives of both high throughput and fairness and demonstrate the corresponding improvements.     

一种新的VANET网络链路丢包率估计算法

在车载自组织网络（VANET）协议体系中,网络链路层丢包率的估计和预测十分重要,因为它不仅决定了信息传输的效果,并且其大小将直接影响对上层服务协议的执行。因此,提出了一种基于后验期望估计（PEE）算法来预测和估算链路丢包率。根据对实验数据结果的分析,使用很少的探测包就可以快速准确地对链路丢包率进行估算,而且PEE算法优于最大似然估计（MLE）算法和期望传输次数（EXT）算法。     

Queuing theory based analysis for packet jitter of mixed services

Exploring theory and methods to analyze the impacts of non real-time services on jitter performance of real-time services is a quite challenging but meaningful job. This article puts forward a general theoretical method to calculate packet jitter in network node bearing mixed services. Based on queuing theory and Markov theory, the network node is modeled as a double-queue single-server and limited-cache queuing system with thresholds. Two-state Markov-modulated Bernoulli process （MMBP-2） and interrupted Bernoulli process （IBP） are used to model the arrival processes of real-time services and small data services respectively in a packet switched network. In order to depict the interaction between real-time services and small data services, a four-dimensional discrete-time Markov chain is implemented to describe the transition of the system states. By solving the system model, expressions for packet jitter are obtained. The results given by the model are then compared with the simulation results obtained by network simulator, version 3 （NS-3）.     

Performance analysis of wavelet packet multirate multicarrier multicode CDMA for wireless multimedia communications

Different kinds of Multirate (MR) communication systems, such as multicode (MCD) scheme and variable spreading length (VSL) schemes, have been considered for accommodating information sources with different data rates in Multicarrier code-division multiple access (MC-CDMA). In this paper, we propose the use of MCD scheme for MR services in MC/MCD-CDMA system that employs wavelet packets (WPs) as subcarriers. The bit error rate (BER) performance for the system was investigated by means of analytical methods and numerical results in a slow fading frequency selective Nakagami channel. The performance analysis includes the effects of diversity techniques, channel intensity profile, diversity order and fading parameter. Also, the effects of different service rates and number of users in each service rate were investigated. The performance of the system was compared to that of MC/MCD-CDMA based on sinusoidal carrier. Results reveal that BER performance is proportional to the service rate and our proposed system outperform the other system.     

Performance analysis of cellular mobile communication networks supporting multimedia services

This paper illustrates the development of an analytical model for a communication network providing integrated services to a population of mobile users, and presents performance results to both validate the analytical approach, and assess the quality of the services offered to the end users. The analytical model is based on continuoustime multidimensional birth–death processes, and is focused on just one of the cells in the network. The cellular system is assumed to provide three classes of service: the basic voice service, a data service with bit rate higher than the voice service, and a multimedia service with one voice and one data component. In order to improve the overall network performance, some channels can be reserved to handovers, and multimedia calls that cannot complete a handover are decoupled, by transferring to the target cell only the voice component and suspending the data connection until a sufficient number of channels become free. Numerical results demonstrate the accuracy of the approximate model, as well as the effectiveness of the newly proposed multimedia call decoupling approach.     

Cross-Layer optimal connection admission control for variable bit rate multimedia traffic in packet wireless CDMA networks

Next generation wireless code division multiple access (CDMA) networks are required to support packet multimedia traffic. This paper addresses the connection admission control problem for multiservice packet traffic modeled as Markov modulated Poisson process (MMPP) with the quality of service (QoS) requirements on both physical layer signal-to-interference ratio (SIR) and network layer blocking probability. Optimal linear-programming-based algorithms are presented that take into account of SIR outage probability constraints. By exploiting the MMPP traffic models and introducing a small SIR outage probability, the proposed algorithms can dramatically improve the network utilization. In addition, we propose two reduced complexity algorithms that require less computation and can have satisfactory approximation to the optimal solutions. Numerical examples illustrating the performance of the proposed schemes are presented.     

减少调度自动化设备通讯串口丢包率的措施

丢包率是指在使用系统检测过程中对数据进行访问时,数据包因为各种物理原因或者软件原因导致的信道丢失现象。在自动化通讯设备出现串口丢包的情况下,会导致设备通信不通畅,信号遗失等不良后果。本文从软硬件方面对通讯串口的丢包问题进行分析,并提出防范措施。     

QoS guarantees for multimedia services on a TDMA-based satellitenetwork

This article describes a proposed network architecture for both the ground and space segments of a satellite network supporting multimedia services with quality of service (QoS) guarantees. The space segment proposal is geared toward a TDMA-based satellite network comprising many nongeostationary satellites, while the ground segment proposal does not put any restriction an the satellite network. The ground segment proposal is based on the new ATM adaptation layer 2 (AAL2) which we consider to have a major role to play in offering an efficient way to provide multimedia services over ATM networks. Indeed, it allows easy encapsulation of the complete set of media component sessions which forms a multimedia transaction into a single ATM virtual channel connection. An additional advantage of AAL2 is that it can run over any arbitrary slotted access system independent of the ATM format (i.e., without transporting the ATM cell headers). This is important for efficient interfacing of the ground and space segments. The space segment proposal aims at providing a robust, easily managed and controlled architecture to guarantee QoS by using a combination of connectionless and connection-oriented techniques. This is achieved by: defining a fixed logical network architecture (i.e., independent of the motion of the satellites); limiting the functionality of the switches in the sky; and using adequate addressing and routing procedures     

Performance models of statistical multiplexing in packet videocommunications

Models and results are presented that assess the performance of statistical multiplexing of independent video sources. Presented results indicate that the probability of buffering (or delaying) video data beyond an acceptable limit drops dramatically as the number of multiplexed sources increases beyond one. This demonstrates that statistical or asynchronous time-division multiplexing (TDM) can efficiently absorb temporal variations of the bit rate of individual sources without the significant variations in reception quality exhibited by multimode videocoders for synchronous TDM or circuit-switched transmission. Two source models are presented. The first model is an autoregressive continuous-state, discrete-time Markov process, which was used to generate source data in simulation experiments. The second model is a discrete-state, continuous-time Markov process that was used in deriving a fluid-flow queuing analysis. The presented study shows that both models generated consistent numerical results in terms of queuing performance     

基于网络丢包率的混合FSO/RF系统切换技术研究

利用C++ Builder编程软件提供的相关网络组件,提出了基于网络丢包率的混合FSO/RF系统切换判决机制,在网络中实现了对链路的监控功能,并采用无线路由器、光纤收发器等器件搭建了混合FSO/RF测试系统,对基于网络丢包率的混合系统切换机制进行了测试,结果表明采用网络切换判决方案,能够自动选择链路通信,保证数据不间断...     

Demonstration of photonic fast packet switching at 700 Mbit/s data rate

Selfrouting of optical data through a photonic packet switch, with user data at 700 Mbit/s, is demonstrated. The switch is transparent to the bandwidth of the optical data thus allowing essentially unlimited payload bit rate.<>     

Efficient radio resource management for wireless multimedia communications: a multidimensional QoS-based packet scheduler

Future wireless multimedia systems will support a variety of services with diverse range of capabilities and bit rates. For these systems, it is highly desired for real-time conversational and non-real-time services to efficiently share the available channels and bandwidth in an optimized way. The partitioned resource shaping with either fixed or a slow changing dynamic, proposed for conventional packet scheduling techniques, proves difficult and inefficient under fast-changing dynamics of radio channel and traffic. By taking into account almost all the aspects (dimensions) of quality-of-service (QoS) provisioning, the proposed unified fast dynamic multidimensional QoS-based packet scheduler (MQPS) in this paper elegantly and efficiently encapsulates features of many possible packet scheduling strategies. MQPS applies an optimization and tuning mechanism to packet scheduling weights to adopt the most appropriate packet scheduling and channel assignment strategy in response to the varying traffic and radio channel conditions. As an example, the technique is applied to a high-speed downlink packet access (HSDPA) system. It is shown that MQPS provides significantly better performance than existing techniques by satisfying all the requirements of a successful QoS provisioning to maximum possible level simultaneously.     

Towards a new generation of transport services adapted to multimedia applications

A partial order and partial reliable connection (POC) is an end-to-end transport connection authorized to deliver objects in an order that can differ from the transmitted one. The service provided by such a connection is also authorized to lose some objects in a controlled manner. The POC approach establishes a conceptual link between connectionless best-effort (CL) and connection-oriented reliable (co) protocols. POC is motivated by heterogeneous CL networks such as Internet which are plagued by unordered delivery of packets and losses, which in turn tax the performances of current protocols. Moreover, it has been shown, that out of order delivery is able to economize the use of network resources such as memory and bandwidth, and reduces end-to-end transit delay. To take advantage of POC’S benefits, applications must be able to relax some transport constraints. In this paper a temporal extension of POC, called TPOC (for temporal POC), is introduced. TPOC offers a conceptual framework, which allows the Qos of distributed multimedia applications to be taken into account. The architecture for offering a TPOC transport service is introduced and evaluated for the transport of MPEG video streams. We demonstrate that POC connections fill not only the conceptual gap between CL protocols (such as UDP) and CO protocols (such as TCP) but also provides demonstrable performances improvements for the transport of multimedia streams such as MPEG video.     

On the effects of the packet size distribution on the packet loss process

Real-time multimedia applications have to use forward error correction (FEC) anderror concealment techniques to cope with losses in today’s best-effort Internet. The efficiency of these solutions is known however to depend on the correlation between losses in the media stream. In this paper we investigate how the packet size distribution affects the packet loss process, that is, the distribution of the number of lost packets in a block, the related FEC performance and the average loss run length. We present mathematical models for the loss process of the MMPP+M/D/1/K and the MMPP+M/M/1/K queues; we validate the models via simulations, and compare the results to simulation results with an MPEG-4 coded video trace. We conclude that the deterministic packet size distribution (PSD) not only results in lower stationary loss probability than the exponential one, but also gives a less correlated loss process, both at a particular average link load and at a particular stationary loss probability as seen by the media stream.Our results show that for applications that can only measure the packet loss probability, the effects of the PSD on FEC performance are higher in access networks, where a single multimedia stream might affect the multiplexing behavior. Our results show that the effects of the PSD on FEC performance are higher in access networks, where a single multimedia stream might affect the multiplexing behavior and thus can improve the queuing performance by decreasing the variance of its PSD.     

Dimensioning the packet loss burstiness over wireless channels: a novel metric,its analysis and application

The packet loss burstiness over wireless channels is commonly acknowledged as a key impacting factor on the performance of networking protocols. An accurate evaluation of the packet loss burstiness, which reveals the characteristics and performance of the wireless channels, is crucial to the design of wireless systems and the quality‐of‐service provisioning to end users. In this paper, a simple yet accurate analytical framework is developed to dimension the packet loss burstiness over generic wireless channels. In specific, we first propose a novel and effective metric to characterize the packet loss burstiness, which is shown to be more compact, effective, and accurate than the metrics proposed in existing literature for the same purpose. With this metric, we then develop an analytical framework and derive the closed‐form solutions of the packet loss performance, including the packet loss rate and the loss‐burst/loss‐gap length distributions. Lastly, as an example to show how the derived results can be applied to the design of wireless systems, we apply the analytical results to devise an adaptive packetization scheme. The proposed packetization scheme adaptively adjusts the packet length of transmissions based on the prediction of the packet loss rate and loss‐burst/loss‐gap lengths of the wireless channel. Via extensive simulations, we show that with the proposed packetization scheme, the channel throughput can be enhanced by more than 10% than the traditional scheme.Copyright © 2012 John Wiley & Sons, Ltd.     

Personal multimedia services over a common home and access networking environment

This article presents a new end-to-end architecture model that will enable the deployment of a plethora of different multimedia services from diverse suppliers competitively coexisting over a common access and home networking environment. The model is focused on personal multimedia communication services and terminals. The proposed model is a segmentation of the end-to-end multimedia chain into several business segments. The interfaces among segments and functional entities inside each segment are identified and defined in the work. In order to study the impact level of the results achieved, a comparison with the approximation to NGN provided by current standardization bodies in the field (3GPP and TISPAN) is also included in the article. This comparison highlights the advantages of using this model as a solution to offer PMC services.