HAMS: Scalable peer-to-peer multimedia streaming model in heterogeneous networks

In a peer-to-peer (P2P) overlay network, a large number and various types of peer processes are interconnected in networks and are cooperating by using multimedia contents like movies and music. Here, multimedia contents are in nature distributed to peers in various ways like downloading and caching to the peers. Multimedia streaming is a key technology to realize multimedia applications in networks. In multimedia streaming applications, multimedia contents are required to be reliable and continuously delivered to processes in a real-time manner. Some contents peer may not send packets of a content at a required rate due to limited computation resource and a communication channel may not support enough Quality of Service (QoS) due to congestions and faults. Thus, P2P overlay networks are in nature heterogeneous. In this paper, we newly discuss a heterogeneous asynchronous multi-source streaming (HAMS) model where multiple contents peers transmit packets of a multimedia content to a requesting leaf peer to increase the throughput, reliability, and scalability in P2P overlay networks. Here, some pair of channels between contents and leaf peers may support different QoS. Peers may be faulty and some pair of contents peers may have different transmission rates. Finally, we show the HAMS model can support higher throughput and shorter transmission time than the other models in the evaluation.     

Neural network prediction model for a real-time data transmission

Both the real-time transmission and the amount of valid transmitted data are important factors in real-time multimedia transmission through the Internet. They are mainly affected by the channel bandwidth, delay time, and packet loss. In this paper, we propose a predictive rate control system for data transmission, which is designed to improve the number of valid transmitted packets for the transmission of real-time multimedia over the Internet. The one-step-ahead round-trip delay time and packet loss are predicted using a prediction algorithm and then these predicted values are used to determine the transmission rate. A real-time multimedia transmission system was implemented using a TCP-friendly algorithm, in order to obtain the measurement data needed for the proposed system. Neural network modeling was performed using the collected data, which consisted of the round-trip time (RTT) delay and packet loss rate (PLR). In addition, the performance of the neural network prediction model was verified through a validation process. The transmission rate was determined from the values of RTT delay and PLR, and a data transmission test for an actual system was performed using this transmission rate. The experiment results show that the algorithm proposed in this study increases the number of valid packets compared with the TCP-friendly algorithm.     

DSMAC:一种适于无线多媒体传感器网络的信道接入协议

多媒体传感器网络作为一种多媒体信息获取和处理方式,已在军事、民用及商业领域中显示出广阔的应用前景.信道接入协议能否高效地使用无线信道是保证无线多媒体传感器网络通信的最关键的因素之一.分析支持多媒体业务传输的无线传感器网络信道接入协议的要求,提出适于多媒体传感器网络提供区分服务的信道接入协议--DSMAC(different service medium access control),对实时业务与非实时业务实现了区分服务,在信道接入帧内的随机竞争期实现突发业务及时接入,支持突发多媒体业务实时传输,并提出了多信道簇间传输方式,避免了隐终端冲突.最后,对协议的服务区分、实时性、吞吐量以及能量有效性等性能进行了仿真实验,验证了其优良性能.     

Cross-Layer Packetization and Retransmission Strategies for Delay-Sensitive Wireless Multimedia Transmission

Existing wireless networks provide dynamically varying resources with only limited support for the quality of service required by the bandwidth-intense, loss-tolerant and delay-sensitive multimedia applications. This variability of resources does not significantly impact delay insensitive data transmission (e.g., file transfers), but has considerable consequences for multimedia applications. Recently, the research focus has been to adapt existing algorithms and protocols at the lower layers of the protocol stack to better support multimedia transmission applications and conversely, to modify application layer solutions to cope with the varying wireless networks resources. In this paper, we show that significant improvements in wireless multimedia performance can be obtained by deploying a joint application-layer adaptive packetization and prioritized scheduling and MAC-layer retransmission strategy. We deploy a state-of-the-art wavelet coder for the compression of the video data that enables on-the-fly adaptation to changing channel conditions and inherent prioritization of the video bitstream. We pose the cross-layer problem as a distortion minimization given delay constraints and derive analytical solutions by modifying existing joint source-channel coding theory aimed at fulfilling rate, rather than delay, constraints. We also propose real-time algorithms that explicitly consider the available information about previously transmitted packets. The obtained results show significant improvements in terms of video quality as opposed to ad-hoc optimizations currently deployed, while the complexity associated with performing this optimization in real time, i.e., at transmission time, is limited     

一种报文大小可变的TFRC改进算法

基于Internet的实时多媒体数据传输是一种报文发送速率固定,报文大小变化的应用。该文分析了这类应用对TFRC的影响,通过对TFRC协议的扩展,提出了一种支持报文大小可变应用的改进TFRC拥塞控制算法。这种算法在接收方采用了对报文数量进行加权的方法来计算丢失事件率以支持报文大小变化的应用。同时在网络仿真器ns2中实现了这种改进算法。仿真实验表明：这种改进算法能够支持报文大小变化,报文发送速率固定的应用,并且具有TCP友好性,与TCP相比具有较平缓的流量抖动。     

A protocol for deadlock-free dynamic reconfiguration in high-speedlocal area networks

High-speed local area networks (LANs) consist of a set of switches interconnected by point-to-point links, and hosts linked to those switches through a network interface card. High-speed LANs may change their topology due to switches being turned on/off, hot expansion, link remapping, and component failures. In these cases, a distributed reconfiguration protocol analyzes the topology, computes the new routing tables, and downloads them to the corresponding switches. Unfortunately, in most cases, user traffic is stopped during the reconfiguration process to avoid deadlock. These strategies are called static reconfiguration techniques. Although network reconfigurations are not frequent, static reconfiguration such as this may take hundreds of milliseconds to execute, thus degrading system availability significantly. Several distributed real-time applications have strict communication requirements; Distributed multimedia applications have similar, although less strict, quality of service (QoS) requirements. Both stopping packet transmission and discarding packets due to the reconfiguration process prevent the system from satisfying the above requirements. Therefore, in order to support hard real-time and distributed multimedia applications over a high-speed LAN, we need to avoid stopping user traffic and discarding packets when the topology changes. In this paper, we propose a new deadlock-free distributed reconfiguration protocol that is able to asynchronously update routing tables without stopping user traffic. This protocol is valid for any topology, including regular as well as irregular topologies. It is also valid for packet switching as well as for cut-through switching techniques and does not rely on the existence of virtual channels to work. Simulation results show that the behavior of our protocol is significantly better than for other protocols based on stopping user traffic     

Impact of out-of-sequence processing on the performance of data transmission

Application Level Framing (ALF) was proposed by Clark and Tennenhouse as an important design principle for developing high performance applications. ALF relies in part on the ability of applications and protocols to process packets independently one from the other. Thus, performance gains one might expect from the use of ALF are clearly related to performance gains one might expect from applications that can handle and process packets received out-of-sequence, as compared to application that require in-sequence delivery (FTP, TELNET, etc.). In this paper, we examine how the ability to process out-of-sequence packets impacts the efficiency of data transmission. We consider both the impact of application parameters such as the time to process a packet by the application, as well as network parameters such as network transmission delay, network loss rate and flow and congestion control characteristics. The performance measure of interest are total latency, buffer requirements, and jitter. We show, using experimental and simulation results, that out-of-sequence processing is beneficial only for very limited ranges of transmission delays and application processing time. We discuss the impact of this on the architecture of communication systems dedicated to distributed multimedia applications.     

PLUS: a probe-loss utilization streaming mechanism for distributed multimedia presentation systems

We present a new flow and congestion control scheme, PLUS (Probe-Loss Utilization Streaming protocol), for distributed multimedia presentation systems. This scheme utilizes probing of the network situation and an effective adjustment mechanism to data loss to support multimedia presentations. The proposed scheme is also designed to scale with increasing number of PLUS-based streaming traffic and to live in harmony with TCP-based traffic. The novelty of the PLUS protocol is that it utilizes the knowledge of its future bottleneck bandwidth in probing the current network situation. This can be achieved by a priori knowledge of the multimedia data before a presentation is requested by a client. Compression schemes like MPEG introduce dependencies on media units. I frames are needed to successfully decode P and B frames, and P frames are needed to decode B frames. A loss of an I or P frame automatically eliminates dependent media units. Our probing scheme increases the successful transmission of critical I and P packets without the overhead of error-correction-schemes. Probing is done using B-frame packets. The advantage is that we use data packets as probe packets. With the PLUS protocol we address the need to avoid congestion rather than react to it. Experiments demonstrate the effectiveness of the approach in utilizing network resources and decreasing loss ratios.     

园区网中QoS解决方案的研究

随着网络技术的发展,网络用户的增多以及各种大型实时的多媒体应用在园区网上实施,园区网在大学、公司和医院等企事业单位的生产和发展过程中发挥着越来越重要的作用,同时网络应用对网络服务的QoS提出更高的要求。本文介绍在园区网实施QoS方案的主要策略;分析DiffServ模型,DiffServ通过对园区网中的通信数据包进行分类和策略控制来达到调节网络资源的目的;探讨区分服务模型的实现,从而实现园区网的QoS保证。     

基于Spines覆盖网络的流媒体并行传输研究

以Spines覆盖网络通用平台为基础,对流媒体数据多流并行传输的相关问题进行了研究。简要介绍了Spines网络,根据流媒体实时并行传输的要求,对Spines网络中的相关协议进行了改进,并引入了具有QoS保证的覆盖网络多路路由策略、DHT技术及LEACH协议。     

Providing MAC QoS for multimedia traffic in 802.11e based multi-hop ad hoc wireless networks

Ad hoc wireless networks with their widespread deployment, now need to support applications that generate multimedia and real-time traffic. Video, audio, real-time voice over IP, and other multimedia applications require the network to provide guarantees on the Quality of Service (QoS) of the connection. The 802.11e Medium Access Control (MAC) protocol was proposed with the aim of providing QoS support at the MAC layer. The 802.11e performs well in wireless LANs due to the presence of Access Points (APs), but in ad hoc networks, especially multi-hop ones, it is still incapable of supporting multimedia traffic.One of the most important QoS parameters for multimedia and real-time traffic is delay. Our primary goal is to reduce the end-to-end delay, thereby improving the Packet Delivery Ratio of multimedia traffic, that is, the proportion of packets that reach the destination within the deadline, in 802.11e based multi-hop ad hoc wireless networks.Our contribution is threefold: first we propose dynamic ReAllocative Priority (ReAP) scheme, wherein the priorities of packets in the MAC queues are not fixed, but keep changing dynamically. We use the laxity and the hop length information to decide the priority of the packet. ReAP improves the PDR by over 28% in comparison with 802.11e, especially under heavy loads. Second, we introduce Adaptive-TXOP (A-TXOP), where transmission opportunity (TXOP) is the time interval during which a node has the right to initiate transmissions. This scheme reduces the delay of video traffic by reducing the number of channel accesses required to transmit large video frames. It involves modifying the TXOP interval dynamically based on the packets in the queue, so that fragments of the same packet are sent in the same TXOP interval. A-TXOP is implemented over ReAP to further improve the performance of video traffic. ReAP with A-TXOP helps in reducing the delay of video traffic by over 27% and further improves the quality of video in comparison with ReAP without A-TXOP. Finally, we have TXOP-sharing, which is aimed at reducing the delay of voice traffic. It involves using the TXOP to transmit to multiple receivers, in order to utilize the TXOP interval fully. It reduces the number of contentions to the channel and thereby reduces the delay of voice traffic by over 14%. A-TXOP is implemented over ReAP to further improve the performance of voice traffic. The three schemes (ReAP, A-TXOP, and TXOP-sharing) work together to improve the performance of multimedia traffic in 802.11e based multi-hop ad hoc wireless networks.     

基于AFDX自适应优先调度算法的实时性分析

通过分析AFDX终端系统关键技术,在AFDX网络虚拟链路的调度策略中提出采用自适应优先调度算法;并分析数据包在终端系统中的处理过程,进行网络演算建模,进而确定出AFDX网络针对不同类型数据的传输速率和服务时延.结果证明该算法较好地满足了航空电子网络数据传输实时性需求,并为AFDX智能网络管理模型中的参数模型和调度策略提...     

无线传感器网络自适应实时路由协议

针对无线传感器网络中节点能源和带宽受限等问题,提出一种基于位置的自适应实时路由协议。该协议自动调整数据包在不同时刻的实际传输速率,选择较匹配的传输路径,满足了不同应用情况下不同层次的网络实时性需求,提高了网络节点的能量有效性。OMNET++软件平台上的仿真实验结果证明,该协议提高了网络的实时性、数据包有效到达率,并延长了网络的寿命。     

A new MAC protocol design based on TDMA/FDD for QoS support in WATM networks

Several wireless networking solutions have been developed to provide different types of services for various end user applications. Currently, wireless networking infrastructures are not suitable for multimedia applications each requiring a different QoS support with various traffic parameters. Due to the success of ATM technology in the wired network, WATM concept and related researches are of importance in the information technology area. Main objective of WATM, which promises seamless transmission of different traffics such as voice, data and video over wireless medium, is to implement high bit rate and QoS guaranteed data transfer, already well achieved by ATM technology over wired medium. To support QoS guaranteed data transfer over error-prone and low bandwidth wireless medium, an effective MAC protocol must be designed and utilized. In this paper, a new TDMA/FDD based MAC protocol, maintaining QoS requirements of real-time wireless multimedia applications, is proposed. The main contribution of this study is the new guarantee-based scheduling algorithm used in the proposed MAC to support required level of QoS guarantee for all multimedia traffic types in wireless medium. Computer modeling and simulation of the new approach providing CBR, VBR, ABR and UBR ATM services are realized using OPNET Modeler. Simulation results are also presented together with comparisons those of a WATM counterpart which uses PRMA/DA MAC protocol.     

WSN中SPIN路由协议的改进

无线传感器网络在实时应用中存在节点能量有限、数据传播延时大等问题。为此,提出一种改进的SPIN路由协议。通过比较最小跳数的数目控制数据的传播方向,选择一条到达Sink节点实时性能最优的路径。仿真结果显示,改进协议可以减少传输过程中数据包的数量,降低网络能耗,延长网络生命周期。     

An intelligent scheduling scheme for real-time traffic management using Cooperative Game Theory and AHP-TOPSIS methods for next generation telecommunication networks

The emerging and exponential growth of telecommunication networks have developed a variety of smart and powerful devices to handle a wide range of multimedia applications such as Voice over IP (VoIP), video streaming, etc. 3GPP introduced Long Term Evolution (LTE) in release 8 and LTE-Advanced (A) in release 10 to support multimedia traffic as these technologies offers high data rate, high bandwidth, and low latency. It also created new challenges to handle resource allocation and power optimization of User Equipment (UE). The paper explores radio resource allocation and power consumption problem of UE in LTE environment. An intelligent scheduling scheme developed is based on Cooperative Game Theory (CGT) method and AHP-TOPSIS method. It distributes resources in a fair way among a number of applications and UE are prioritized based on certain criteria like delay, throughput history, UE buffer space and channel conditions and preferences. In LTE, Discontinuous Reception (DRX) has been adopted to conserve the battery life of UE. DRX periodically switches off the radio interfaces to conserve the battery life but it may breach Quality of Service (QoS). Therefore, the DRX parameters need to be further optimized to satisfy QoS and minimize power consumption of UE. DRX parameters are dynamically adjusted on the basis of current load and channel condition of the network. Power saving operations are numerically analyzed. Simulation results show that the expert and an intelligent system can distribute resources in a fair way among UE, improves the battery consumption of UE up to 85% and packets transmission delay by 10% as compared to existing scheme for real-time applications.     

一种多媒体传感器网络容错分簇算法

在无线多媒体传感器网络分簇算法设计中,针对如何满足QoS需求并尽可能提高能量效率问题,提出了一种容错分簇算法。根据节点的剩余能量和质心选举簇头,采用容错机制和能量有效策略组织成簇,并动态调整数据包在簇头间的传输速率。仿真实验结果表明该分簇算法满足多媒体数据传输的可靠性和实时性需求,能有效延长网络的生命周期。     

MPEG流封装成RTP协议包的具体实现

MPEG流是由多个层次构成手复杂的媒体流，实时传输协议（RTP）是针对多媒体数据进行实时传输的协议。MPEG流依据RTP以流媒体的形式进行实时传输是目前MPEG视音频传输的一个有效的方法。本文给出了将MPEG流封装成RTP协议包的具体实现。     

One-time packet key exchange scheme for secure real-time multimedia applications

To secure multimedia communications, existing encryption techniques usually encrypt the whole data stream using the same session key during a session. The use of the session key confronts with tradeoff problem between session key creation latency and security for the real-time multimedia stream. The main feature of our proposed scheme is to selectively encrypt RTP packets using different one-time packet keys in the same session for real-time multimedia applications. The packet key, which has already been used, will never be reused throughout the same session. The use of the one-time packet key enables to improve security strength of real-time multimedia. To solve the issue of the real-time packet key exchanges related to the timely use of the one-time packet keys, this paper suggests the one-time packet key exchange method that does not need to occur on a packet-by-packet basis.     

支持实时多媒体传输的应用层组播系统

由于IP组播并未取得预期的成功，研究人员又提出了由终端主机来代替路由器实现群组通信功能。针对实时多媒体传输的特点以及目前单组播网络混合存在的现状，该文提出了一利，新的支持实时多媒体传输的应用层组播系统。在系统中，多个网关形成一层覆盖网，由网关完成数据的复制、分发以及组的成员管理，从而在应用层实现了群组通信的功能。