An efficient modeling and dimensioning approach for integrated streaming and elastic traffic

Recent studies show that both data traffic and real-time traffic grow very fast in wired and wireless networks. To provide better performance guarantee, these applications need efficient network modeling and planning. In this paper, the problem where the total bandwidth of a link is shared by streaming traffic (real time traffic such as voice or video etc.) and elastic traffic (such as data) is studied. Integrating streaming traffic and elastic traffic presents a unique dimensioning problem. This paper considers dimensioning a link to satisfy both quality of service (QoS) requirements for streaming traffic, such as loss probability, and elastic traffic, such as mean waiting (delay) time. The Erlang loss model is applied to streaming traffic and a bursty traffic model is applied to the elastic traffic. Efficient dimensioning algorithms based on classical Markovian models and time-scale decomposition are then proposed. Numerical results show that the proposed methods have good accuracy.     

Network dimensioning and performance of multiservice, multirateloss networks with dynamic routing

We address a multiservice, multirate loss network environment with dynamic routing. In this setting, we consider multiple traffic load periods (multihour) during the day, and by observing network dynamics, we present a network dimensioning model that consists of two steps: a bandwidth estimation step, followed by a multicommodity flow model for multiple services and traffic loads. For network operations, we discuss a probabilistic admission control policy and three multiservice routing schemes. We have used a ten-node network with multiple asymmetric traffic data sets (partially extracted from an actual network) for our study. It was found that the capacity obtained using the analytic network dimensioning model provides a good estimate of network capacity required for meeting the grade-of-service goal for each service type in each traffic load period; this observation is based on a simulated network environment that uses the proposed admission control and the dynamic routing schemes. Our observation suggests that it may not be not necessary for the dimensioning model to explicitly incorporate an admission control policy, but admission control is needed for network operation to provide desirable grade-of-service     

Dimensioning of the LTE access network

This paper proposes efficient analytical models to dimension the necessary transport bandwidths for the Long Term Evolution (LTE) access network satisfying the QoS targets required by different services. In this paper, we consider two major traffic types: elastic traffic and real time traffic. For each type of traffic, individual dimensioning models are proposed for both the S1 interface and the X2 interface. For elastic traffic the dimensioning models are based on the Processor Sharing models; while for real time traffic the dimensioning models are based on the fundamental queuing models. For validating these analytical dimensioning models, a developed LTE system simulation model is used. Extensive simulations are performed for various traffic and network scenarios. The analytical results derived from the proposed dimensioning models are compared with the simulation results. The presented results demonstrate that the proposed analytical models can appropriately estimate the required performances for different service classes and priorities. Hence they are suitable to be used for dimensioning of the LTE access network with different traffic and network conditions.     

Network architecture and traffic transport for integrated wireless communications over enterprise networks

A novel network architecture based on the IEEE 802.6 metropolitan area networks (MAN) is proposed to integrate the wireless and wired segments of a regional enterprise network (REN) within a city. This architecture functions like a distributed switch for all types of services, reducing traffic congestion by sharing the high capacity link dynamically and facilitating signaling, mobility management, call processing and network management through its distributed functions, transport facilities and broadcasting capability. It also serves as a peripheral gathering network of REN traffic for transport over a wide area ATM/BISDN, enabling integration of an enterprise's regional networks into a global EN. Two major wireless applications, i.e., wireless PABX (WPABX) and wireless LAN (WLAN) are discussed to illustrate the advantages of this MAN‐based architecture. Although a REN is likely to support a wide range of different services, voice and data will continue to be the predominant traffic generated by WPABXs and WLANs, respectively, and are also representative of isochronous and asynchronous multimedia traffic carried by future wireless networks. We compare the traffic capacity of several voice transport alternatives under integrated (voice/data) network traffic with various data traffic loads, and study voice and data integration under three different integration schemes by simulations. Results indicate that the MAN‐based architecture is most effective employing queue arbitrated (QA) access for asynchronous traffic, pre‐arbitrated access for constant bit‐rate isochronous traffic, and the new reservation arbitrated (RA) access for variable bit‐rate isochronous traffic, under a scheme that permits full sharing between QA and RA traffic. This revised version was published online in July 2006 with corrections to the Cover Date.     

Approach for voice quality and throughput estimation in wireless convergent networks

A common communications convergence scenario which is being adopted in personal communications relates to the combination of wireless and cellular networks by the use of multimode terminals. Since most of the wireless networks were initially dimensioned only for data communications, this paper shows how voice over wireless LAN dimensioning could be addressed under the optimal network throughput and the perspective of voice quality, using a simple approach. The maximum number of simultaneous users resulting from throughput is limited by the collisions taking place in the shared medium with the statistical contention protocol. The voice quality is conditioned by the delay and the packet loss in the contention protocol. Both approaches are analyzed within the scope of the voice codecs commonly used in voice over wireless LANs, to conclude that voice dimensioning based on network throughput and voice quality show complementary results. Additionally the use of low rate codecs in voice over wireless LANs is advantageous for the network performance point of view but may produce poor voice quality results. Mid range codecs like G729 could represent a trade-off for quality throughput. For these reasons, voice quality and wireless network throughput have to be taken into account in the network admission control, design and deployment to ensure a satisfactory user experience. The impact of handoff interval of wireless convergent networks on the conversation quality need also be assessed for a proper network design.     

Policing of realistic VBR video traffic in an ATM network

ATM has been accepted by CCITT as the transport mechanism for the future BISDN and will also be widely used in future customer premises networks. Networks based on the ATM principle are expected to provide a very flexible communications infrastructure allowing customers to make effective use of a wide variety of offered services. To provide this flexibility with an acceptable quality of service while operating the network in an economic way, elaborate traffic management functions will be necessary to control the traffic flows within the network. This paper will study one of these functions—the so-called ‘usage parameter control’ or ‘policing’ function—in some detail to illustrate some of the problems that arise and point out possible solutions. The mechanisms chosen to implement the policing function will be the ‘leaky bucket’ mechanism, the ‘jumping window’ mechanism and the ‘moving window’ mechanism. The input streams used to assess the mechanisms represent different types of video communication—videophone, video conference and entertainment video—coded according to different variable bit-rate (VBR) algorithms. In contrast to most of the previous studies, where artificial, statistical traffic sources have been used, the sources used in this paper are directly based on measured ‘real-life’ video data. This ensures that all the statistical properties of the actual traffic stream are preserved and allows identification of the different factors that influence the dimensioning and the performance of the policing mechanism. The results of this study show that the uncertainty about the key parameters at call set-up and the considerable impact of single scenes make the proper dimensioning of policing mechanisms difficult. Furthermore, it seems not to be practical to use the long term mean bit-rate as the key traffic control parameter for these sources. Results indicating that the long-term cell loss ratio is not a sufficient measure for the quality of service are also presented. A comparison of the mechanisms shows that from a performance perspective, the ‘leaky bucket’ mechanism is superior to the two window mechanisms. This work is relevant to evolving standards for both BISDN traffic management and variable bit-rate video coding.     

QoS‐based radio network dimensioning for LTE networks with heavy real‐time traffic

Applications of video streaming and real‐time gaming, which generate large amounts of real‐time traffic in the network, are expected to gain considerable popularity in Long Term Evolution networks. Maintaining the QoS such as packet delay, packet loss ratio, median, and cell border throughput requirements in networks dominated by real time traffic, is critical. The existing dimensioning methodology does not consider QoS parameters of real‐time traffic in network dimensioning. Moreover, exhaustive and time‐consuming simulations are normally required to evaluate the performance and QoS of real‐time services. To overcome this problem, we propose an improved radio network dimensioning framework that considers the QoS of real‐time traffic in network dimensioning. In this framework, an analytical model is proposed to evaluate the capacity and performance of real‐time traffic dominant Long Term Evolution networks. The proposed framework provides a fast and accurate means of finding the trade‐off between system load, packet delay, packet loss ratio, required median, and cell border throughput. It also provides network operators with an analytical means for obtaining the minimum number of sites required by jointly considering coverage, capacity and QoS requirements. The accuracy of the proposed model is validated through simulations. Copyright © 2012 John Wiley & Sons, Ltd.     

Analysis of Bandwidth Gain Over Various Timer$_$CU of AAL2 for Voice Traffic Multiplexing

An asynchronous transfer mode adaptation layer type 2 (AAL2) transmission scheme commonly is used to deliver the voice and tha data traffic between Node-B and the radio network controller on the universal mobile terrestrial device network. To predict the AAL2 multiplexing performance, we analyzed the bandwidth gain and the cell-packing density using discrete Markov chain model for the voice service and validated these results with simulations. We also performed a detailed simulation for the voice and the data services in a concentrator. Based on the analysis, we proposed an engineering guideline for selecting the optimal Timer$_$CU in a Node-B. We found that there is no major benefit in using the AAL2 multiplexing in a concentrator. The benefit of the AAL2 multiplexing in$ I_ ub$for the data service was much less than that for the voice service. They also depended heavily on the traffic load.     

Optimized dimensioning of resilient optical grids with respect to grade of services

An optical grid network geographically integrates distributed computing/information resources with high speed communications. Network dimensioning, maximization of services, and job scheduling are some of today key arising issues in optical grids. Since the last decade, many projects have been conducted in order to provide computational and information facilities in the academic as well as in the business communities. In this paper, we study the network dimensioning and the maximization of IT services in optical grids. We propose a scalable optimization model for maximizing IT services under link transport capacities. We assume the use of the anycast routing principle to identify the server nodes for executing the jobs, and a shared path protection mechanism in order to offer protection against single link/node failures. We also investigate different calculation methods of the link transport capacities in order to maximize the grade of services, while taking into account the bandwidth requirements. Computational results are presented on different traffic distributions. They show that the proposed link dimensioning can save more than 35 % bandwidth in optical grid networks, in comparison with the classical link dimensioning strategies. We also investigate the different protection schemes against single link failures, single node failures, single node and server node failures, and compare their bandwidth requirements, as well as their impact on the grade of services (GoS). Results show that there is no significant increase of the bandwidth requirements and no meaningful impact on the GoS when moving from a single link protection scheme to a single node (including server nodes) protection scheme.     

Capacity of a wireless LAN with voice and data services

The capacity of a carrier sense multiple-access wireless LAN with voice and data services using the TCP/IP protocol is analyzed to obtain a lower bound for the capacity of the wireless networks with voice and data services. The voice traffic is given a higher priority over the data traffic to accommodate the delay requirement for an acceptable quality of service. This is implemented by assigning the TCP protocol for data and the UDP protocol for voice. The relationship between the data throughput and the number of the voice users supported in this environment is analyzed by using a nonpreemptive queuing model. While the analysis in this paper can be applied to any voice encoding system, the improved multiband excitation voice encoding technique is adopted to provide a low transmission rate with an acceptable quality     

Hierarchical Ethernet Transport Network Architecture for backhaul cellular networks

Cellular traffic has been going through major changes in recent years. With the introduction of broadband services in 3G/4G and the continuously increasing provided data rates in high-speed packet access and Long Term Evolution (LTE), a broad range of cellular applications has emerged, changing the characteristics of cellular traffic. The traditional circuit-switched voice traffic has been taken over by packet-oriented data traffic. This shift in traffic has driven operators to prefer packet-oriented network technologies over circuit-switch technology when implementing their cellular networks. The current offered technologies, including PBB-Traffic Engineering and MPLS-TP, lack important functionality required for LTE such as automation (simple management), traffic engineering, protection, Quality of Service, and scalability. We propose a scalable Hierarchical Ethernet Transport Network Architecture (HETNA), a layer 2 transport technology that addresses these issues and brings a viable solution for cellular networks. The suggested architecture can handle streaming, real-time, multicasting, and other applications. Both connection-oriented transport services and connectionless-oriented services are supported. HETNA was simulated and prototyped, showing significant improvements over regular Ethernet in terms of buffers and control messages that enable this network to function.     

Analysis and engineering of a voice/data packet multiplexer

This paper investigates performance and engineering issues concerning a multiplexer scheme that has been implemented in AT&T's Integrated Access Terminal (IAT) to transport packetized voice and data traffic on shared facilities. The multiplexer serves voice and data traffic according to a dynamic bandwidth allocation scheme in order to simultaneously meet their performance requirements. A bit-dropping procedure is employed for voice packets to provide a graceful degradation of voice quality under overload conditions. An analytical model is developed for the multiplexer service scheme that estimates performance parameters given the voice and data offered loads. The model is used to demonstrate the capacity advantages of dynamic bandwidth allocation, and to generate load-service curves that illustrate the tradeoffs of carrying different combinations of voice and data traffic on the multiplexer. Sensitivity of voice and data performance to the multiplexer time-slice parameters is also investigated. The model is readily embedded in a design approach that determines the bandwidth required to carry the voice and data traffic demands while satisfying all desired performance objectives     

无线网络中的QoS和策略管理测试(续一)

上一期中,我们论述了策略管理和QoS在无线网络中的作用,并介绍了3GPP所定义的实现方式。在本期中,我们将继续讨论QoS和策略管理的测试方法,并介绍IXIA在一个北美运营商实验室里实现VoLTE测试的成功案例。6无线网络的服务质量验证全球运营商正花费数十亿美元购买设备和频谱来升级他们的移动宽带网络。通过网络升级,运营商旨在     

VoIP reliability: a service provider's perspective

Voice over IP services offer important revenue-generating opportunities, as well as many technical challenges in providing high-quality services. Users have come to expect highly available telecommunications services with high-quality voice. Service providers need reliable high-performance networks to meet user expectations, and must be able to guarantee performance and reliability to their customers. In converged voice and data networks, the network infrastructure must deliver very high quality and availability for some customer needs, while also providing low-cost high-capacity bandwidth for other needs. The use of quality of service mechanisms to provide prioritization for various traffic types is a key element needed for voice and data network convergence. However, it is not sufficient if the underlying networks are unreliable. The focus of this article is to address the reliability aspects of VoIP services, including the underlying IP networks.     

Interference and SIR in integrated voice/data wireless DS-CDMAnetworks-a simulation study

The objective of this study is to characterize the multiple-access interference in a DS-CDMA integrated voice/data wireless network. The system model is designed to accommodate bursty packet-based data services in addition to stream-based services on a shared spectrum basis. A common packet data channel (CPDC) is employed to transport bursty data services, Consequently, this study incorporates features associated with packet-based services, such as higher transmitter power, short inter-arrival times, and short service time durations, as well as features associated with stream-based services, in order to provide a complete interference characterization. The results of a detailed and precise simulation study are presented in which the influence of traffic burstiness, voice activity, spatial distribution of mobiles, and transmitter powers on the fluctuation of the interference and signal-to-interference ratio (SIR) is assessed. The results obtained also quantify the impact of interference caused by CPDC services on stream-based services. The interference and SIR statistics are evaluated in terms of cumulative distribution functions. Since a DS-CDMA network is interference limited, studies of this type are essential for system design, capacity evaluation, and bandwidth management     

针对密集城区场景的VoLTE语音性能研究

在LTE网络大规模部署的现阶段,4G用户规模和业务量持续攀高,4G手机渗透率迅速提升,而语音业务作为运营商重要的收入来源,其重要性不言而喻.针对LTE的语音终极实现方案VoLTE(Voice over LTE),研究了实现VoLTE的关键技术,分析了影响语音质量的多种因素,提出了一种基于E-model的VoLTE语音性能评估方法,在LTE网络率先部署的密集城区场景中,基于不同网络配置和并发用户数,对VoLTE业务性能进行仿真,并采用E模型进行语音质量分析,最终实现对VoLTE的语音性能评估.     

Dimensioning IP access links carrying data traffic

This paper proposes simple dimensioning rules for high speedip access links carrying data traffic. Assuming a finite source population and fair bandwidth sharing among user flows, we derive formulas relating capacity, demand and performance. These formulas allow link dimensioning for a target quality of service expressed in terms of useful per-flow throughput. They constitute a data traffic model equivalent of the Engset model for telephone access networks. Performance is shown to be largely independent of detailed traffic characteristics such as the statistical distributions of flow size and think time. Simple approximations are derived for two distinct performance regimes corresponding to transparency and saturation, respectively. Extensions to the basic model account for a heterogeneous user demand, unfair bandwidth sharing or different access rate classes. In any case, the key parameter for dimensioning is the offered traffic, defined as the average data rate a user would generate in the absence of congestion.     

Buffer and bandwidth allocation for TCP data traffic: experimental results

This paper discusses the dimensioning of buffers and the bandwidth allocation for data traffic in the ATM network. Data traffic is notoriously complex and bursty, making such dimensioning a difficult task. However, the COMBINE project, when dimensioning their InterWorking Units (IWUs), adopted a Poissonian packet arrival model, based upon the argument that burstiness at timescales higher than that of a packet arrival are a problem to be tackled by flow control at higher layers. This paper presents experimental results from the COMBINE testbed that show that this hypothesis was justified and that good TCP goodput was obtained based upon this dimensioning approach, due to TCP's ability to adapt to network congestion. However, it is also shown that it was the TCP algorithm that was ultimately responsible for controlling the packet loss ratio in the network and not the bandwidth allocation or buffer size. The results highlight the importance of taking into account the mutual influence between the ATM layer and the transport layer congestion control algorithms.     

Packet CDMA versus dynamic TDMA for multiple access in anintegrated voice/data PCN

A comparative evaluation of dynamic time-division multiple access (TDMA) and spread-spectrum packet code-division multiple access (CDMA) approaches to multiple access in an integrated voice/data personal communications network (PCN) environment are presented. After briefly outlining a cellular packet-switching architecture for voice/data PCN systems, dynamic TDMA and packet CDMA protocols appropriate for such traffic scenarios are described. Simulation-based network models which have been developed for performance evaluation of these competing access techniques are then outlined. These models are exercised with example integrated voice/data traffic models to obtain comparative system performance measures such as channel utilization, voice blocking probability, and data delay. Operating points based on typical performance constraints such as voice blocking probability 0.01 (for TDMA), voice packet loss rate 10^-3 (for CDMA), and data delay 250 ms are obtained, and results are presented     

Application of cost models over traffic dimensioning with QoS restrictions

Network operators and Internet service providers are offering “Triple Play” products integrating services with different quality of service (QoS) requirements. The provision of QoS guarantees implies the revision of current dimensioning methods and consequences for costing and pricing. This paper proposes a cost model which considers QoS parameters, based on the Total Element Long Run Incremental Cost (TELRIC) model, calculating the cost of a network element and distributing it over the different services whose traffic uses it, taking into account the QoS requirements of each service. For this purpose, three traffic engineering methods are analyzed: complete traffic aggregation by “Over-engineering,” complete traffic segregation by separated virtual tunnels, and partial traffic aggregation by priority queuing. As an example, the cost model is applied to the connection in a Next Generation Network aggregation network for estimating the influence of QoS and traffic engineering on the cost estimation under the TELRIC model.