Multichannel bandwidth allocation in a broadband packet switch

The problem of bandwidth allocation in a packet switch supporting broadband services is addressed. To reduce the performance constraints imposed by limiting a data link to a single broadband packet channel, the author introduces the concept of channel group as a set of broadband packet channels that is viewed as a single data-link connection by routing entities. He uses a two-step bandwidth allocation scheme. At connection setup time, a call is allocated to a channel group. At transmission time, specific channels of a group are optimally allocated to the packets destined to the group. Because of the statistical smoothing of the large number of sources served by a channel group, the traffic performance of the switch is improved. This scheme also allows super-rate switching, i.e., the support of services with peak bandwidth exceeding the capacity of a single packet channel. The author shows the feasibility of this scheme in a Batcher-banyan switch, by implementing in hardware the bandwidth allocation at transmission time. Performance improvements obtained by this scheme are also provided in different traffic environments     

Hopfield neural‐network‐based dynamic resource allocation scheme for non‐real‐time traffic in wireless networks

Dynamic resource allocation (DRA) plays a fundamental role in current and future wireless networks, including 3G systems. In this paper, a scheduling DRA scheme for non‐real‐time (NRT) packet services in wireless system is proposed based on the use of Hopfield neural networks (HNN). The scheme exploits the fast response time of HNN for solving NP optimization problems and has been particularized for the downlink transmission in a UMTS system, although it could be easily extended to any other radio access technology. The new DRA scheme follows a delay‐centric approach, since it maximizes the overall system resource utilization while minimizing the packet delay. Simulation results confirm that the proposed HNN‐based DRA scheme is effective in supporting different types of NRT services, while achieving efficient utilization of radio resources. Copyright © 2008 John Wiley & Sons, Ltd.     

High-throughput time group access MCR-SS-CSMA/CA for wireless ad hoc networks with layered-tree topology

Multi-hop packet transmission error due to packet collision is a serious issue for realizing large-scale wireless ad hoc networks. In order to solve the issues, a multicode-reception spread-spectrum (MCR-SS)-CSMA/CA scheme is proposed as multiple access scheme for layered-tree networks. This scheme is further improved by adding a time group access (TGA) technique to gain higher interference reduction between nodes. The proposed TGA scheme divides the layered-tree network into upper subnets group and lower subnets group, thus reducing collision probability among subnets. Network throughput and data delivery ratio for the proposed scheme are analyzed by simulations based on the IEEE 802.15.4/ZigBee specification. Simulation results show that the proposed schemes can improve packet delivery ratio and network throughput remarkably.     

Interference management using basestation coordination in broadband wireless access networks

This paper proposes a transmission‐scheduling algorithm for interference management in broadband wireless access networks. The algorithm aims to minimize the cochannel interference using basestation coordination while still maintaining the other quality of service (QoS) requirements such as packet delay, throughput and packet loss. The interference reduction is achieved by avoiding (or minimizing) concurrent transmission of potential dominant interferers. Dynamic slot allocation based on traffic information in other cells/sectors is employed. In order to implement the algorithm in a distributed manner, basestations (BSs) have to exchange traffic information. Both real‐time and non‐real‐time services are considered in this work. Results show that significant reduction in the packet error rate can be achieved without increasing the packet delay at low to medium loading values and with a higher but acceptable packet delay at high loading values. Since ARQ schemes can also be used for packet error rate reduction, we compare the performance of the proposed scheme with that of ARQ. Results indicate that although ARQ is more effective in reducing packet error rate, the proposed algorithm incurs much less packet delay particularly at medium to high loading. Copyright © 2006 John Wiley & Sons, Ltd.     

FLC-Based MBWIMA/UMTS Protocol for Multimedia Services in UTRA TDD Mode

An intelligent medium access control (MAC) protocol based on fuzzy logic control (FLC) is proposed and compared with a general packet radio system in UMTS (GPRS/UMTS), priority scheme and the movable boundary wireless integrated multiple access in UMTS (MBWIMA/UMTS) protocols. The integrated video/voice/data services of UMTS in UTRA TDD mode have different transmission properties. By fuzzy logic control, the resources of the wireless communication can be intelligent assigned for different types of mediums. The voice-video dropping probability and data packet delay are input to FLC to optimally select the maximum number of voice/video slots. Voice activity detector (VAD) and multiple access interference in single cell are also considered in the simulations.     

Channel-aware priority-based groupwise packet scheduling for non-real time data service with burst-switching DS/CDMA system

The algorithm of scheduling scheme of channel-aware priority-based groupwise transmission is investigated for non-real time data service for the uplink direct sequence code division multiple access (DS/CDMA) system using the burst-switching scheme to support the integrated voice/data service. The proposed scheme optimally determines the transmission-time groups and assigns optimal data rates to the users with packets in the transmission-time group depending on priority metric, which involves several parameters such as delay threshold, waiting time, length of packet, and state of the channel, in a way of minimizing the average transmission delay. Simulation results show that the proposed algorithm gives better performance of average transmission delay and packet loss probability than any other conventional algorithms.     

一种适用于虚拟机网络的数据包高效传输方法

为了提高数据包在云计算数据中心中基于虚拟机构成网络中的传输性能,提出了一种基于网络编码的高效数据包传输方法.基于网络编码机制,采用对传输过程中丢失数据包高效的编码组合策略,多个虚拟机终端可以在一次多播或广播传输中获取多个从交换机优先传输的数据包,因此,提出的方法可以提高基于虚拟机网络的多播及广播业务的数据包传输延迟,并提高多播及广播业务的网络吞吐量.仿真结果表明提出的方法在典型信道条件下均获得了较好的数据包传输时延及网络吞吐量性能.     

A medium access control scheme for TDD-CDMA cellular networks with two-hop relay architecture

In this paper, we propose a multihop medium access control (mMAC) scheme for time division duplexing-code division multiple access (TDD-CDMA) cellular networks with two-hop relay architecture to support packet data transmission. The proposed mMAC is based on joint CDMA/PRMA (packet reservation multiple access) protocol and it includes BCH code selection, power control and multihop relaying. Simulation results reveal that cellular networks with two-hop relay architecture with the proposed mMAC scheme can substantially provide a good performance as well as larger cell coverage as compared to conventional TDD-CDMA single-hop cellular networks.     

移动P2P流媒体中基于有效容量的邻居节点选择机制

For Peer-to-Peer (P2P) streaming services in mobile networks, the selection of appropriate neighbour peers from candidate peers with demanding data is an important approach to improve Quality-of-Service (QoS). This paper proposes a novel Effective Capacity Peer Se-lection (ECPS) scheme based on effective capacity. In the ECPS scheme, the neighbour peer selection problem was modeled using the Multiple Attribute Decision Making (MADM) theory, which considered multiple factors of candidate peers, including Signal to Interference and Noise Ratio (SINR), residen-cy time, power level, security, moving speed, and effective capacity. This model could increase the suitability of ECPS for wireless mobile environments. Then, the Technique for Order Preference by Similarity to Ideal Solu-tion (TOPSIS) was used to solve the MADM problem and identify the preferred neighbour peers. Simulation results show that the ECPS scheme can improve the network throughput, reduce packet delay by about 82%, and al-most double the packet delivery ratio of the mobile P2P streaming service.     

A preemptive polling protocol for applications in wireless LANs

This paper deals with a radio‐based communication network with a single radio channel shared by several data terminals for transmissions to a single hub station. In particular, the communication system considered is a potential provider of wireless LAN‐like services. The focus is on the capacity of combining a preemptive polling‐based multiple access scheme with a Selective Repeat ARQ technique to counteract the effect of the nonstationary transmission channel. The nonstationary transmission channel has been modeled as a two‐state Markov chain with parameters related to actual propagation conditions. Typical outdoor/indoor environments have been considered. The main idea is that of making the service interruption of the preemptive polling scheme dependent on the propagation conditions of the transmission channel by monitoring the outcome of the data packet transmission attempts. A performance comparison clearly reveals the superiority of this preemptive polling scheme with respect to the classical cyclic polling scheme. This revised version was published online in June 2006 with corrections to the Cover Date.     

Low latency handover scheme for 5G dual-connectivity scenario

A novel handover scheme was proposed for the secondary node (SN) unchanged 5G dual-connectivity scenarios.In the proposed novel scheme,the SN connection was maintained for data packet transmission during the handover,however,both the main node (MN) and the SN were completely disconnected in the legacy scheme.The transmission delay during handover was decreased greatly by the proposed scheme.Firstly,the legacy handover scheme was analyzed and its deficiency was figured out.Then,the novel mechanism’s signaling interaction was elaborated and the time sequence models for the novel scheme and the legacy scheme were further established.Finally,based on the time sequence model,the performance evaluation processes were carried out in terms of mathematical modeling and experimental simulations.The analysis results demonstrate that the proposed novel scheme reduces the single packet transmission delay,the average transmission delay and the total transmission delay,and has good performance advantages.     

无人机短包通信中基于NOMA传输的安全性能分析

针对物联网（IoT）通信的低延时需求,为了保证数据传输的灵活性,本文构建一种基于短包传输的无人机（UAV）通信网络。由于非正交多址接入（NOMA）技术能够增加可服务的地面用户数量,故将该技术应用到无人机短包通信（UAV-SPC）系统中可以解决多用户的安全传输问题。与正交多址（OMA）技术相比,NOMA可有效提高用户接入公平性和频谱利用率,因此被广泛用于下行链路的通信传输。为解决复杂的安全传输问题,首先证明在功率和译码错误率约束的条件下,分别存在最优的功率分配,数据传输包长和系统传输比特数使目标用户的平均安全吞吐量最大。在此基础上,通过本文所提算法得到安全传输问题的优化解。实验结果验证了该算法的稳定性和可行性。此外,与基准方案相比,本文所提方案可有效降低短包传输的通信时延,提高系统中目标用户的平均安全吞吐量。     

Interleaved coding and sequential transmission scheme for packet‐level forward error correction over burst loss channels

Burst packet loss is a common problem over wired and wireless networks and leads to a significant reduction in the performance of packet‐level forward error correction (FEC) schemes used to recover packet losses during transmission. Traditional FEC interleaving methods adopt the sequential coding‐interleaved transmission (SCIT) process to encode the FEC packets sequentially and reorder the packet transmission sequence. Consequently, the burst loss effect can be mitigated at the expense of an increased end‐to‐end delay. Alternatively, the reversed interleaving scheme, namely, interleaved coding‐sequential transmission (ICST), performs FEC coding in an interleaved manner and transmits the packets sequentially based on their generation order in the application. In this study, the analytical FEC model is constructed to evaluate the performance of the SCIT and ICST schemes. From the analysis results, it can be observed that the interleaving delay of ICST FEC is reduced by transmitting the source packets immediately as they arrive from the application. Accordingly, an Enhanced ICST scheme is further proposed to trade the saved interleaving time for a greater interleaving capacity, and the corresponding packet loss rate can be minimized under a given delay constraint. The simulation results show that the Enhanced ICST scheme achieves a lower packet loss rate and a higher peak signal‐to‐noise‐ratio than the traditional SCIT and ICST schemes for video streaming applications.     

Hierarchy Bayesian model based services awareness of high-speed optical access networks

As the speed of optical access networks soars with ever increasing multiple services, the service-supporting ability of optical access networks suffers greatly from the shortage of service awareness. Aiming to solve this problem, a hierarchy Bayesian model based services awareness mechanism is proposed for high-speed optical access networks. This approach builds a so-called hierarchy Bayesian model, according to the structure of typical optical access networks. Moreover, the proposed scheme is able to conduct simple services awareness operation in each optical network unit (ONU) and to perform complex services awareness from the whole view of system in optical line terminal (OLT). Simulation results show that the proposed scheme is able to achieve better quality of services (QoS), in terms of packet loss rate and time delay.     

MIMO广播信道中联合多用户速率和延迟约束调度的研究

该文给出了一种多入多出高斯广播信道中具有延迟约束的多用户无线数据包的传输方案。首先,在信道的QR分解及脏纸编码基础上,通过贪婪算法获取多用户分集与包延迟约束之间的有效结合。其次,将包到达速率、可达服务速率、用户数以及传输天线数构成一优化问题,得到最佳的用户组合及调度周期。最后,在不同用户数及发射天线数情况下对该方案进行了性能仿真,仿真结果表明:在满足包传递最小延迟等待要求的同时,得到了传输容量的最大化。     

A Study on Differentiated Service Queuing Scheme with an Overflow Buffer Allocation Within a UMTS Core Network

According to the QoS features of the four types of UMTS traffic, this study proposes a priority-based queuing scheme to support differentiated services among all UMTS traffic; it bases on packet transmission priorities of four types of UMTS traffic to handle packets forwarding in a gateway within a UMTS core network. In the proposed queuing scheme, a static dedicated logical queuing buffer allocation and a dynamic overflow buffer allocation are used to support packet enqueuing/dequeuing jobs. In this study, the ns2 network simulator is used as a simulation platform and different categories of simulation scenarios are executed. Moreover, the IETF DiffServ scheme is used as a comparison to understand the packet forwarding performance of the proposed scheme. By analyzing simulation data, two important points can be drawn. The proposed queuing scheme supports a differentiated packet forwarding behavior among four types of UMTS traffic. And, the differentiated packet forwarding behaviors with the proposed queuing scheme are similar to the packet forwarding behavior with the IETF DiffServ scheme.     

A Study on Adaptive Time Token Priority-Based Queuing Scheme

Several famous priority-based queuing schemes operated in a gateway to support differentiated services among internet traffic. Examining packet forwarding operations in these queueing schemes, they only support a priority-based service either in a packet enqueuing process or in a packet dequeuing process. If a queuing scheme can support priority-based services in both packet enqueuing/dequeuing processes; it would enhance differentiated service performance for internet traffic. This study proposes a priority-based queuing scheme with an adaptive time token allotment measure to support a differentiated packet forwarding process for different types of IP traffic both in packet enqueuing/dequeuing processes. Depending on packet sizes and packet forwarding priorities of IP traffic, the proposed queuing scheme assigns fix and adaptive time token thresholds dynamically to logical queuing buffers separately. With assigned time tokens, logical queuing buffers allow arrival IP packets to be enqueued in a differentiated way. Moreover, the proposed queuing scheme uses a transferred WRR dequeuing measure to enhance a differentiated packet forwarding process. The simulation results show that the proposed queuing scheme supports a differentiated packet forwarding process for different types of IP traffic. The differentiated packet forwarding performance supported by the proposed scheme is close to the IETF DiffServ scheme; this result shows that the proposed scheme can support differentiated packet forwarding performance for different types of IP traffic with a lower operation cost.     

Design of transmission and multiplexing systems for broadbandpacket networks

Dynamic time-division multiplexing (DTDM) is a flexible network transport technique capable of handling both continuous and bursty traffic effectively. By using three different multiplexing architectures in the network, DTDM permits graceful evolution of the existing circuit switching network into a flexible broadband packet communications network supporting integrated voice, data, and video traffic. The first multiplexing stage uses a packet assembler to multiplex different broadband services into a common DTDM-format serial bit stream. The second multiplexing stage uses a statistical packet multiplexer to concentrate network traffic for more efficient use of transmission facilities. The third multiplexing stage uses a synchronous time-division multiplexer for high-speed point-to-point transparent transmission. The multiplexer uses a simple tributary synchronization scheme based on positive and negative block justification, which combines the concept of controlled-slip and bit-stuffing techniques while maintaining information integrity. A generic CMOS LSI chip has been designed for use in the three-stage multiplexing system     

Improving the R-score of an adaptive VoIP codec in IEEE 802.16 networks

Worldwide Interoperability for Microwave Access (WiMAX) technology, which is based on the IEEE 802.16 standard, supports different quality of service (QoS) for different services. WiMAX is expected to support QoS in real-time applications such as Voice over Internet Protocol (VoIP). When network congestion occurs, the VoIP bit rate needs to be adjusted to achieve the best speech quality. In this study, we propose a new scheme called Adaptive VoIP Level Coding (AVLC). This scheme takes into consideration network conditions (packet delay and packet loss) and a connection’s modulation scheme. The amount of data that can be transmitted increases with the speed of the modulation scheme. When network congestion occurs, AVLC scheme prioritizes reducing the bit rate of a connection that has a slower modulation scheme to mitigate congestion. Depending on network conditions, such as modulation scheme, packet delay, packet loss, and residual time slot, we use the G.722.2 codec to adjust each connection’s bit rate. Simulations are conducted to test the performance (network delay, packet loss, number of modulation symbols, and R-score) of the proposed scheme. The simulation results indicate that speech quality is improved by the use of AVLC.     

自适应动态分组预约多址协议

该文提出一种基于自适应传输的动态分组预约多址协议。用来在业务量和信道都时变的环境下，灵活、高效地支持需要各种不同质量需求的无线多媒体通信。在这个方案中根据特点不同将空闲时隙分为“自由”、“长可用”或“短可用”三种，以“自由”时隙数目作为系统实时业务量的判决标准，根据系统实时业务量分别采用不同的多址接入策略，将长数据和短数据安排在语音传输的空隙传输，同时采用了自适应传输体制来进一步提高系统性能。仿真实验表明，通过合理选择系统参数，可以有效地增加系统总容量。