Evaluating the influence of multiplexing schemes and buffer implementation on perceived VoIP conversation quality

This work presents a study of RTP multiplexing schemes, which are compared with the normal use of RTP, in terms of experienced quality. Bandwidth saving, latency and packet loss for different options are studied, and some tests of Voice over IP (VoIP) traffic are carried out in order to compare the quality obtained using different implementations of the router buffer. Voice quality is calculated using ITU R-factor, which is a widely accepted quality estimator. The tests show the bandwidth savings of multiplexing, and also the importance of packet size for certain buffers, as latency and packet loss may be affected. The customer’s experience improvement is measured, showing that the use of multiplexing can be interesting in some scenarios, like an enterprise with different offices connected via the Internet. The system is also tested using different numbers of samples per packet, and the distribution of the flows into different tunnels is found to be an important factor in order to achieve an optimal perceived quality for each kind of buffer. Grouping all the flows into a single tunnel will not always be the best solution, as the increase of the number of flows does not improve bandwidth efficiency indefinitely. If the buffer penalizes big packets, it will be better to group the flows into a number of tunnels. The router processing capacity has to be taken into account too, as the limit of packets per second it can manage must not be exceeded. The obtained results show that multiplexing is a good way to improve customer’s experience of VoIP in scenarios where many RTP flows share the same path.     

PowerTrust: A Robust and Scalable Reputation System for Trusted Peer-to-Peer Computing

We propose a new fair scheduling technique, called OCGRR (output controlled grant-based round robin), for the support of DiffServ traffic in a core router. We define a stream to be the same-class packets from a given immediate upstream router destined to an output port of the core router. At each output port, streams may be isolated in separate buffers before being scheduled in a frame. The sequence of traffic transmission in a frame starts from higher-priority traffic and goes down to lower-priority traffic. A frame may have a number of small rounds for each class. Each stream within a class can transmit a number of packets in the frame based on its available grant, but only one packet per small round, thus reducing the intertransmission time from the same stream and achieving a smaller jitter and startup latency. The grant can be adjusted in a way to prevent the starvation of lower priority classes. We also verify and demonstrate the good performance of our scheduler by simulation and comparison with other algorithms in terms of queuing delay, jitter, and start-up latency     

OCGRR: A New Scheduling Algorithm for Differentiated Services Networks

We propose a new fair scheduling technique, called OCGRR (Output Controlled Grant-based Round Robin), for the support of DiffServ traffic in a core router. We define a stream to be the same-class packets from a given immediate upstream router destined to an output port of the core router. At each output port, streams may be isolated in separate buffers before being scheduled in a frame. The sequence of traffic transmission in a frame starts from higher-priority traffic and goes down to lower-priority traffic. A frame may have a number of small rounds for each class. Each stream within a class can transmit a number of packets in the frame based on its available grant, but only one packet per small round, thus reducing the intertransmission time from the same stream and achieving a smaller jitter and startup latency. The grant can be adjusted in a way to prevent the starvation of lower priority classes. We also verify and demonstrate the good performance of our scheduler by simulation and comparison with other algorithms in terms of queuing delay, jitter, and start-up latency.     

自相似背景流量模型下的CBR流延迟抖动分析

实时业务是网络中快速增长的业务类型,但网络中需要传送多种业务的混合流量。实时 业务的性能取决于分组延迟抖动。过大的分组延迟抖动将导致语音的中断,画面的停顿和跳跃。延 迟抖动主要是背景流量在边缘路由器的干扰引起的。以往的延迟抖动分析都是假设背景流量为泊松 过程,研究表明这种假设已经不符合当前网络流量的特性。本文将对自相似背景流量下的CBR流的 延迟抖动进行分析,给出其分布函数,并以仿真结果验证其与泊松流量对CBR流的不同影响。     

On the challenge and design of transport protocols for MMORPGs

Although MMORPGs are becoming increasingly popular as well as a highly profitable Internet business, there is still a fundamental design question: Which transport protocol should be used—TCP, UDP, or some other protocol? In this paper, we first evaluate whether TCP is suitable for MMORPGs, and then propose some novel transport strategies for this genre of games. Our analysis of a trace collected from a TCP-based MMORPG called ShenZhou Online indicates that TCP is unwieldy and inappropriate for MMORPGs. We find that the degraded network performance problems are due to the following characteristics of MMORPG traffic: 1) tiny packets, 2) a low packet rate, 3) application-limited traffic generation, and 4) bi-directional traffic. Since not all game packets require reliable transmission or in-order delivery, transmitting all packets with a strict delivery guarantee causes high delays and delay jitters. Therefore, our proposed transport strategies assign game packets with appropriate levels of transmission guarantee depending on the requirements of the packets’ contents. To compare the performance of our approach with that of existing transport protocols, we conduct network simulations with a real-life game trace from Angel’s Love. The results demonstrate that our strategies significantly reduce the end-to-end delay and delay jitter of packet delivery. Finally, we show that our strategies effectively raise satisfaction levels of the game players.     

Detecting Malicious Packet Losses

In this paper, we consider the problem of detecting whether a compromised router is maliciously manipulating its stream of packets. In particular, we are concerned with a simple yet effective attack in which a router selectively drops packets destined for some victim. Unfortunately, it is quite challenging to attribute a missing packet to a malicious action because normal network congestion can produce the same effect. Modern networks routinely drop packets when the load temporarily exceeds their buffering capacities. Previous detection protocols have tried to address this problem with a user-defined threshold: too many dropped packets imply malicious intent. However, this heuristic is fundamentally unsound; setting this threshold is, at best, an art and will certainly create unnecessary false positives or mask highly focused attacks. We have designed, developed, and implemented a compromised router detection protocol that dynamically infers, based on measured traffic rates and buffer sizes, the number of congestive packet losses that will occur. Once the ambiguity from congestion is removed, subsequent packet losses can be attributed to malicious actions. We have tested our protocol in Emulab and have studied its effectiveness in differentiating attacks from legitimate network behavior.     

Multimedia streaming gateway with jitter detection

This paper investigates a novel active buffer management scheme, "Jitter Detection" (JD) for gateway-based congestion control to stream multimedia traffics in packet-switched networks. The quality of multimedia presentation can be greatly degraded due to network delay variation or jitter when transported over a packet-switched network. Jitter degrades the timing relationship among packets in a single media stream and between packets from different media streams and, hence, creates multimedia synchronization problems. Moreover, too much jitter will also degrade the performance of the streaming buffer in the client. Packets received by the client will be rendered useless if they have accumulated enough jitter. The proposed active buffer management scheme will improve the quality of service in multimedia networking by detecting and discarding packets that accumulated enough jitter, such as to maintain a high bandwidth for packets within the multimedia stream's jitter tolerance. Simulation results have shown that the proposed scheme can effectively lower the average received packet jitter and increase the goodput of the received packets when compared to random early detection (RED) and DropTail used in gateway-based congestion control. Furthermore, simulation results have also revealed that the proposed scheme can maintain the same TCP friendliness when compared to that of RED and DropTail used for multimedia streams.     

How to reduce packet dropping in a bufferless NoC

Networks on‐chip (NoCs) interconnect the components located inside a chip. In multicore chips, NoCs have a strong impact on the overall system performance. NoC bandwidth is limited by the critical path delay. Recent works show that the critical path delay is heavily affected by switch port buffer size. Therefore, by removing buffers, switch clock frequency can be increased. Recently, a new switching technique for NoCs called Blind Packet Switching (BPS) has been proposed, which is based on removing the switch port buffers. Since buffers consume a high percentage of switch power and area, BPS not only improves performance but also reduces power and area. In BPS, as there are no buffers at the switch ports, packets cannot be stopped and stored on them. If contention arises packets are dropped and later reinjected, negatively affecting performance. In order to prevent packet dropping, some techniques based on resource replication have been proposed. In this paper, we propose some alternative and complementary techniques that do not rely on resource replication. By using them, packet dropping is highly reduced. In particular, packet dropping is completely removed for a very wide network traffic range. Moreover, network throughput is increased and packet latency is reduced. Copyright © 2010 John Wiley & Sons, Ltd.     

基于多跳间时延协作的最久分组优先算法

提出了一种基于多跳间时延协作的Crossbar调度算法。该算法以分组头中记录的时延为权重对分组进行调度，通过控制分组在各跳上的时延来达到调节端到端时延的目的。算法还使路由器避免了维护每个流的状态信息以及对单个流进行的复杂的队列管理和调度。计算机仿真表明，算法具有较高的资源利用率、较低的端到端时延抖动和较低的分组丢弃率等特点。     

A versatile model for predicting the performance of deflection-routing networks

Deflection routing can be used in networks whose stations have the same number of input and output links. Fixed-length packets arrive synchronously on the station's input links at the beginning output link that offers the shortest path to its destination. Since the number of packet buffers at each output link is finite, the simultaneous contention of two packets for the last buffer of the common output link must be resolved by “deflecting” one of the packets according to a specified criterion (e.g. at random, by destination proximity, or by packet age). Deflection routing can therefore be used with as few as one packet buffer per output link.

The potentially unbounded number of routes that a given packet can take makes analyzing the performance of such networks difficult. Using independence assumptions, we develop an efficient, high-fidelity performance model of deflection routing that allows us to estimated the mean end-to-end packet delay in a network that has any given two-connected topology, a single packet buffer at each output port, and an arbitrary traffic matrix.     

多通道并行传输中接收缓存阻塞的缓解方法

针对多通道并行传输中的接收缓存阻塞问题,分析了引起接收缓存阻塞的原因,提出一种改进的缓解接收缓存阻塞的数据包调度方法,综合考虑通道的带宽、时延和丢包率,引入通道质量的评价函数,优化多通道之间的数据包调度,选择质量最好的通道进行传输,减少由于通道特性不同造成的接收端数据包乱序;提出一种改进的数据包重传策略,基于时延和丢包率选择能使数据包最快到达接收端的通道进行重传;提出一种根据通道的带宽-延迟积估算所需接收缓存大小的方法。仿真实验表明,所提出的调度方法和重传策略能够有效地减轻接收缓存阻塞,与CMT-SCTP相比具有更优的性能,所提出的缓存大小的估算方法也能够准确估算所需接收缓存的大小。     

A sensitive network jitter measurement for covert timing channels over interactive traffic

In order to reflect the network transmission quality, some network state feedback mechanisms are provided in the network protocol. In the RTP, the jitter of the packet transmission delay is fed back through the jitter field in the RTCP packet. This feedback value is a very important reference data when the covert timing channel is established. However, the sending frequency of the RTCP packet is low and the feedback value of the RTCP packet are only the jitter value of the last RTP packet associated with this RTCP packet when it is sent. Therefore, the jitter feedback mechanism in the existing RTCP protocol has the problem of lack of feedback on the network state during the period between two RTCP data packets. As a result, the feedback value is highly susceptible to extreme values, which prevents it from providing an accurate numerical reference for establishing covert channels. Therefore, in this paper, a buffer was established between the last RTCP packet and the current RTCP packet. And we choose to set the interval is n RTP packets and record the corresponding position jitter value in the buffer. The data in the buffer is averaged, and the mean value is weighted and averaged with the jitter value of the current RTCP packet as a new jitter feedback value. The effect of the extreme value on the feedback value is reduced, thereby it contribute to the improvement of the feedback energy for the state of the network. In addition, the bit error rate generated by establishing a simple covert timing channel for data transmission under different network conditions is compared with the change of two jitter feedback values. It is verified that there is a positive correlation between the feedback value of the new feedback mode and the error rate. through the comparison It is verified that the new feedback method can provide a more accurate reference for the establishment of covert channels.

     

Per-stream loss behavior of ∑MAP/M/1/K queuing system with a random early detection mechanism

In this paper, the matrix-analytic approach is applied to explore the per-stream loss behavior of the multimedia traffic under RED scheme. We constructed a ∑MAP/M/1/K queuing model for the RED mechanism with multimedia traffic which follows a continuous-time Markovian arrival process (MAP). In addition to evaluating the long-term per-stream packet drop probabilities, we examine the bursty nature of per-stream packet drops by means of conditional statistics with respect to dropped periods and the probability that the queuing system stays in the dropped period. The dropped period corresponds to having more than a certain number of packets in router buffer; non-dropped period corresponds to the opposite. These performance measures describe the quality of service provided by the router to particular multimedia traffic streams in the presence of background multimedia traffic.     

实时语音的自适应回放算法

由于丢包和延时抖动的引入而使网络传输的实时语音质量让人难以接受，目前对丢包和延时抖动提出了很多的解决方案．但是却很少把这两者结合在一起进行研究。本文提出了一种新的自适应回放算法，通过监测接收和回放队列，结合丢包的自适应恢复技术，达到语音高质量的连续回放。实验证明，该算法能在严格的平均回放延时条件下努力减小由于超时而引起的丢包，获得较好的重建语音质量。     

Using Parallel DRAM to Scale Router Buffers

This paper addresses the design of high-performance buffers for high-end Internet routers. The buffers are typically implemented using a combination of SRAM and DRAM technologies in order to simultaneously meet the routers' high speed and capacity requirements. The major challenge in designing router buffers is to maintain multiple flow queues in the memory, unlike computer memory buffers (i.e., memory system). The major objective is to minimize the use of expensive but fast SRAM while providing acceptable delay guarantees to packets. In this paper, we first investigate hybrid SRAM/DRAM solutions proposed in the past. We show that one of the architectural limitations of these solutions is that the required SRAM size grows linearly with the number of flows in the system. This prevents the solutions from scaling to support a large number of flows. We then break down this shortcoming by proposing a parallel hybrid SRAM/DRAM (PHSD) architecture. We design a series of memory management algorithms (MMAs) for PHSD, based on tradeoffs between the complexity of the MMAs and the guarantee of in-order delivery of packets (segmentations). We perform a detailed analysis of the proposed algorithms and conduct extensive simulations to show that PHSD can significantly outperform solutions proposed in the past in terms of the SRAM requirements and packet delay.     

Subjective Quality of Speech Over Packet Networks as a Function of Packet Loss, Delay and Delay Variation

An expanding proportion of voice traffic is being carried by packet networks. Speech quality can be impaired in qualitatively new ways in packet networks when packets are lost or the spacing between them is distorted. Three parameters that characterize the performance of packet networks were examined for their relative impact on speech quality as judged by human observers: network delay or latency, packet loss, and packet delay variation or jitter. We manipulated these variables via a network emulator made available by NIST. This report summarizes five laboratory experiments that examined the variables in a variety of experimental procedures for presenting and judging speech. The experiments agreed in showing that the relative importance of the variables for affecting speech quality was, in decreasing order: packet loss, jitter, delay. The effect on speech quality of 200 ms of network delay was shown to be equivalent to the effect of one percentage point of packet loss. Many consumers also traded off some speech quality for a free, added feature, unified messaging.     

面向非全互连3D NoC的自适应单播路由算法

针对在非全互连三维片上网络（3D NoC）架构中的硅通孔（TSV）表只存储TSV地址信息,导致网络拥塞的问题,提出了记录表结构。该表不仅可以存储距离路由器最近的4个TSV地址,也可存储相应路由器输入缓存的占用和故障信息。在此基础上,又提出最短传输路径的自适应单播路由算法。首先,计算当前节点与目的节点的坐标确定数据包的传输方式;其次,检测传输路径是否故障,同时获取端口缓存占用信息;最后,确定最佳的传输端口,传输数据包到邻近路由器。两种网络规模下的实验结果表明,与Elevator-First算法相比,所提算法在平均延时和吞吐率性能指标上有明显的优势,且在网络故障率为50%时,Random和Shuffle流量模型下的丢包率分别为25.5%和29.5%。     

Port partitioning and dynamic queueing for IP forwarding

With the increase of internet protocol (IP) packets the performance of routers became an important issue in internet/working. In this paper we examine the matching algorithm in gigabit router which has input queue with virtual output queueing. Dynamic queue scheduling is also proposed to reduce the packet delay and packet loss probability. Port partitioning is employed to reduce the computational burden of the scheduler in a switch which matches the input and output ports for fast packet switching. Each port is divided into two groups such that the matching algorithm is implemented within each pair of groups in parallel. The matching is performed by exchanging the pair of groups at every time slot. Two algorithms, maximal weight matching by port partitioning (MPP) and modified maximal weight matching by port partitioning (MMPP) are presented. In dynamic queue scheduling, a popup decision rule for each delay critical packet is made to reduce both the delay of the delay critical packet and the loss probability of loss critical packet. Computational results show that MMPP has the lowest delay and requires the least buffer size. The throughput is illustrated to be linear to the packet arrival rate, which can be achieved under highly efficient matching algorithm. The dynamic queue scheduling is illustrated to be highly effective when the occupancy of the input buffer is relatively high.Scope and purposeTo cope with the increasing internet traffic, it is necessary to improve the performance of routers. To accelerate the switching from input ports to output in the router partitioning of ports and dynamic queueing are proposed. Input and output ports are partitioned into two groups A/B and a/b, respectively. The matching for the packet switching is performed between group pairs (A, a) and (B, b) in parallel at one time slot and (A, b) and (B, a) at the next time slot. Dynamic queueing is proposed at each input port to reduce the packet delay and packet loss probability by employing the popup decision rule and applying it to each delay critical packet.The partitioning of ports is illustrated to be highly effective in view of delay, required buffer size and throughput. The dynamic queueing also demonstrates good performance when the traffic volume is high.     

Investigation of the M2/G2/1/∞,N queue with restricted admission of priority customers and its application to HSDPA mobile systems

This paper investigates a queuing system for QoS optimization of multimedia traffic consisting of aggregated streams with diverse QoS requirements transmitted to a mobile terminal over a common downlink shared channel. The queuing system, proposed for buffer management of aggregated single-user traffic in the base station of High-Speed Downlink Packet Access (HSDPA), allows for optimum loss/delay/jitter performance for end-user multimedia traffic with delay-tolerant non-real-time streams and partially loss tolerant real-time streams. In the queuing system, the real-time stream has non-preemptive priority in service but the number of the packets in the system is restricted by a constant. The non-real-time stream has no service priority but is allowed unlimited access to the system. Both types of packets arrive in the stationary Poisson flow. Service times follow general distribution depending on the packet type. Stability condition for the model is derived. Queue length distribution for both types of customers is calculated at arbitrary epochs and service completion epochs. Loss probability for priority packets is computed. Waiting time distribution in terms of Laplace–Stieltjes transform is obtained for both types of packets. Mean waiting time and jitter are computed. Numerical examples presented demonstrate the effectiveness of the queuing system for QoS optimization of buffered end-user multimedia traffic with aggregated real-time and non-real-time streams.     

Adaptive management algorithms for a fixed jitter buffer

Improved jitter buffer management algorithms that synchronize the time of the arrival of voice IP packets with their generation time and take into account their order of entry into the codec are suggested. The suggested algorithms provide higher quality of voice communication (especially under strong network loading) in comparison with the known management algorithms for fixed jitter buffers.