What are suspicious VoIP delays?

Voice over IP (VoIP) is unquestionably the most popular real-time service in IP networks today. Recent studies have shown that it is also a suitable carrier for information hiding. Hidden communication may pose security concerns as it can lead to confidential information leakage. In VoIP, RTP (Real-time Transport Protocol) in particular, which provides the means for the successful transport of voice packets through IP networks, is suitable for steganographic purposes. It is characterised by a high packet rate compared to other protocols used in IP telephony, resulting in a potentially high steganographic bandwidth. The modification of an RTP packet stream provides many opportunities for hidden communication as the packets may be delayed, reordered or intentionally lost. In this paper, to enable the detection of steganographic exchanges in VoIP, we examined real RTP traffic traces to answer the questions, what do the “normal” delays in RTP packet streams look like? and, is it possible to detect the use of known RTP steganographic methods based on this knowledge?     

Online FPS games: effect of router buffer and multiplexing techniques on subjective quality estimators

First Person Shooters are a genre of online games in which users demand a high interactivity, because the actions and the movements are very fast. They usually generate high rates of small packets which have to be delivered to the server within a deadline. When the traffic of a number of players shares the same link, these flows can be aggregated in order to save bandwidth. Certain multiplexing techniques are able to merge a number of packets, in a similar way to voice trunking, creating a bundle which is transmitted using a tunnel. In addition, the headers of the original packets can be compressed by means of standard algorithms. The characteristics of the buffers of the routers which deliver these bundled packets may have a strong influence on the network impairments (mainly delay, jitter and packet loss) which determine the quality of the game. A subjective quality estimator has been used in order to study the mutual influence of the buffer and multiplexing techniques. Taking into account that there exist buffers which size is measured in terms of bytes, and others measured in packets, both kinds of buffers have been tested, using different sizes. Traces from real game parties have been merged in order to obtain the traffic of 20 simultaneous players sharing the same Internet access. The delay and jitter produced by the buffer of the access router have been obtained using simulations. In general, the quality is expected to be reduced as the background traffic grows, but the results show an anomalous region in which the quality rises with the background traffic amount. Small buffers present better subjective quality results than bigger ones. When the total traffic amount gets above the available bandwidth, the buffers measured in bytes add to the packets a fixed delay, which grows with buffer size. They present a jitter peak when the offered traffic is roughly the link capacity. On the other hand, buffers which size is measured in packets add a smaller delay, but they increase packet loss for gaming traffic. The obtained results illustrate the need of knowing the characteristics of the buffer in order to make the correct decision about traffic multiplexing. As a conclusion, it would be interesting for game developers to identify the behaviour of the router buffer so as to adapt the traffic to it.     

一种提高TCP与UDP数据流公平性的拥塞控制机制

在UDP业务流逐步占据大部分网络带宽的情况下,如何在网络层对UDP包进行拥塞控制显得尤为重要。在网络拥塞时,由于UDP包本身缺乏反馈机制,将会产生严重的丢包或者UDP抢占TCP带宽的现象。文中提出了基于网络层的发送端缓冲队列管理的拥塞控制机制,可以均衡TCP和UDP的带宽使用,同时通过对路由器ICMP网络拥塞报文的处理,建立了高效的流量调节策略,并进行了网络仿真实验。实验结果表明,基于该机制的拥塞控制可以有效改善网络不同业务流带宽使用的不公平性。     

A comprehensive survey on scheduler for VoIP over WLAN

Voice over Internet Protocol (VoIP) technology has observed rapid growth in the world of telecommunications. VoIP offers high-rate voice services at low cost with good flexibility, typically in a Wireless Local Area Network (WLAN). In a voice conversation, each client works either as a sender or a receiver depending on the direction of traffic flow over the network. A VoIP technologically requires high throughput, less packet loss and a high fairness index over the network. The packets of VoIP streaming may experience drops because of competition among the different kinds of traffic flow over the network. A VoIP application is also sensitive to delays and requires voice packets to arrive on time from the sender to the receiver without any delay over a WLAN. To date, scheduling of VoIP traffic is still an unresolved problem. The objectives of this survey paper are to discuss fundamental principles of VoIP-related schedulers and identify current scheduler issues. This survey paper also identifies the importance of the scheduling techniques over WLANs. Related research work for real-time applications specifically for VoIP will also be highlighted.     

多维IP分类算法分析研究

路由器中将分组分类成"流"的过程称为分组分类,属于同一流中的所有分组遵循相同的预定规则且路由器对其进行相似的处理。非 "尽力而为"的服务需要对分组进行分类，例如：防火墙，QoS，区分服务等。该文描述3种不同的分类方法并比较分析各种分类（Packet Classification）算法的查找时间复杂度、存储开销。     

基于语音质量预测的VoIP自适应抖动缓冲算法

抖动缓冲是解决VoIP系统延时抖动问题的有效方法。为实现抖动缓冲的动态调整,获得更好的VoIP通话质量,提出了一种基于语音质量预测的自适应抖动缓冲算法。算法采用Pareto分布为延时建模,通过E-Model方法预测突发丢包模式下的瞬时语音质量,以最大化语音质量为目标,自适应选择出最优的抖动缓冲区大小。实验仿真结果表明,所提算法明显优于已有算法,能够有效提高VoIP系统的语音质量。     

Comparison of load-balancing approaches for multipath connectivity

During past years, the so-called resource pooling principle in data networks has been studied more carefully. For example, the recent work on routing on the Internet over multiple paths and Multipath TCP both seek to make the best possible use of multiple connecting paths between two end points. In deployments where multiple users could share multiple paths, one of the very first questions that comes to mind is, should we schedule packets from the users on a per-flow or per-packet basis? In this paper we study networking scenarios in which several networks are connected to each other via multiple paths. We seek to understand how a multi-homed router should schedule packets and packet flows out towards other networks. Our primary interests are to study path utilization and analyze the bandwidth fairness of various approaches using different traffic loads.     

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。     

确保服务中一种基于动态阈值的数据包标记算法

确保转发过程中带宽享用的公平性问题一直是区分服务网络研究的热点，影响这种公平性的因素包括回路响应时间RTT、数据包大小、目标速率及聚流中包含的单流数量等．确保服务的实现依赖于在边界路由器执行的数据包标记策略和在核心路由器执行的队列管理策略，基于动态阈值的数据包标记策略DTBM的目的就是处理异质的TCP流之间带宽享用的公平性问题。DTBM通过测量局部吞吐量来调整标记算法中的阈值，以改变不同颜色的标记概率从而达到公平带宽享用的目的。DTBM的主要优点在于其实现简单、对参数不是很敏感并且对端结点主机是“透明”的，仿真实验表明，和其他几种标记算法相比，DTBM能有效地消除上述因素的影响，具有更好的公平性。     

无线Mesh网络中VOIP的QoS性能分析

针对VoIP(Voice over IP)业务在无线Mesh网上进行传输时存在服务质量(QoS)需求难以保证、带宽利用率低的问题,介绍了VoIP的QoS影响因素,分析了端到端时延、时延抖动和丢包率等几个重要参数,并对VoIP在无线Mesh网中的传输性能进行了论述。提出了基于无线Mesh网络的QoS保证机制,可以为端到端的数据传输公平的分配带宽,并能在保证QoS下实现大规模的实时任务的多跳转发。仿真试验表明能有效降低端到端时延,有着更好的QoS性能。     

LTE系统中提高TCP性能的资源调度算法研究

LTE可以提供真正无处不在基于IP的移动宽带业务,但随着承载网的IP化,网络拥塞、丢包、抖动、延时等质量问题将影响到LTE业务层的QoS质量.作为LTE无线资源管理的核心,研究并设计一个良好的资源调度算法是提高数据业务的性能和终端用户的体验是一个亟待解决的重要任务.本文通过借鉴LTE对VoIP数据分组的半持续调度算法的思想,提出了一种LTE的无线资源调度的改进方案.方案将TCP确认包映射到具有更高优先级的空闲逻辑信道,从而降低了ACK包在无线信道中发生丢弃和拥塞的概率,避免了TCP拥塞控制机制的频繁开启.仿真结果表明,本文提出的TCP确认包映射转换方案在RTT时延、吞吐量等方面均有一定的提升,具有一定的稳定性和性能优势.     

Enhanced bulk scheduling for supporting delay sensitive streaming applications

Providing end-to-end delay guarantees for delay sensitive applications is an important packet scheduling issue with routers. In this paper, to support end-to-end delay requirements, we propose a novel network scheduling scheme, called the bulk scheduling scheme (BSS), which is built on top of existing schedulers of intermediate nodes without modifying transmission protocols on either the sender or receiver sides. By inserting special control packets, which called TED (Traffic Specification with End-to-end Deadline) packets, into packet flows at the ingress router periodically, the BSS schedulers of the intermediate nodes can dynamically allocate the necessary bandwidth to each flow to enforce the end-to-end delay, according to the information in the TED packets. The introduction of TED packets incurs less overhead than the per-packet marking approaches. Three flow bandwidth estimation methods are presented, and their performance properties are analyzed. BSS also provides a dropping policy for discarding late packets and a feedback mechanism for discovering and resolving bottlenecks. The simulation results show that BSS performs efficiently as expected.     

QoS management in heterogeneous home networks

An OSGi (Open Services Gateway Initiative) home gateway system manages the integration of heterogeneous home networks protocols and devices to develop ubiquitous applications. Wired and wireless heterogeneous home networks have different QoS concerns. For instance, jitter and latency are important concerns in web phones, while packet loss ratio is important in on-line video. This study adopts UPnP QoS specification version 1.0 to design an adaptive QoS management mechanism based on the RMD (Resource Management in DiffServ) architecture. This study monitors real-time network traffic, and adaptively controls the bandwidth, to satisfy the minimum but most important quality for each application in home network congestion. Simulation results indicate that the average jitter, latency and packet loss are reduced by 0.1391 ms, 0.0066 s, and 5.43%, respectively. The packet loss ratio is reduced by 4.53%, and the throughput is increased by 1.2% in high definition video stream; the packet loss ratio is reduced by 1.89% for standard definition video stream, and in VoIP (Voice over IP) the jitter and latency are reduced to 0.0407 ms and 0.0209 s, respectively.     

Short-circuiting the congestion signaling path for AQM algorithms using reverse flow matching

Recently, we introduced a new congestion signaling method called ACK spoofing, which offers significant benefits over existing methods, such as packet dropping and Explicit Congestion Notification (ECN). Since ACK spoofing requires the router to create a ‘short circuit’ signaling path, by matching marked data packets in a congested buffer with ACK packets belonging to the same flow that are traveling in the opposite direction, the focus of this paper is evaluating the feasibility of reverse flow matching. First, we study the behavior of individual flows from real bi-directional Internet traces to show that ACK spoofing has the potential to significantly reduce the signaling latency for Internet core routers. We then show that reverse flow matching can be implemented at reasonable cost, using essentially the same hardware as the packet filtering logic commonly employed in Layer 2 transparent bridges. Finally, we show that this architecture can be scaled to accommodate worst-case traffic patterns on multi-gigabit links that would render ordinary route caching algorithms completely ineffective.     

基于Speex语音引擎的VoIP系统设计与实现*

论述了一套基于Speex语音引擎和RTP的VoIP系统设计和开发,介绍了该系统服务器端和客户机端的软件实现。该系统具有点对点通信、算法延时小、丢包补偿和延时补偿性能好等特点,并具有多方通话功能。性能对比实验表明,该系统的通话质量优于几套流行的开源VoIP软件,能满足实际应用的要求。     

基于E-Model的语音帧分组传输性能研究

voIP的语音帧分组大小是实时语音传输的关键参数。为提高网络效率和最大话路数,采用EModel的方法分析了RTP包中语音帧个数、语音长度、丢包概率和抖动缓冲区大小对语音质量的影响,给出了不同带宽时的最佳传输分组大小。仿真结果表明,在保证最基本的话音质量情况下,为不同链路确定合适的分组语音帧数能有效提高链路的最大话路数。     

EVoIP: Energy efficient voice over IP privacy

Due to the convergence of telecommunication technologies and pervasive computing, voice is increasingly being transmitted over IP networks, in what is commonly known as Voice over IP (VoIP). Despite many advantages offered by this technology, VoIP applications inherit many challenging characteristics from the underlying IP network related to quality of service and security concerns. Traditional ways to secure data over IP networks have negative effects on real-time applications and on power consumption, which is scarce in power-constrained handheld devices. In this work, a new codec-independent Energy Efficient Voice over IP Privacy (E²VoIP²) algorithm is devised to limit the overhead of the encryption process, without compromising the end-to-end confidentiality of the conversation. The design takes advantage of VoIP stream characteristics to encrypt selected packets using a secure algorithm, while relaxing the encryption procedure in-between these packets. We evaluated experimentally the difficulty of conducting known plaintext attacks on VoIP by demonstrating that a sound recorded simultaneously by different sources results in apparently random encoded files. Regarding E²VoIP², experimental and simulation results show a substantial improvement in terms of the number of CPU cycles which results in a reduction of latency and a reduction in consumed power with respect to that of the SRTP. In addition, the proposed method is flexible in terms of the balance between security and power consumption.     

HMM delay prediction technique for VoIP

This paper proposes a new algorithm for predicting audio packet playout delay for voice conferencing applications that use silence suppression. The proposed algorithm uses a hidden Markov model (HMM) to predict the playout delay. Several existing algorithms are reviewed to show that the HMM technique is based on a combination of various desirable features of other algorithms. Voice over Internet protocol (VoIP) applications produce packets at a deterministic rate but various queuing delays are added to the packets by the network causing packet interarrival jitter. Playout delay prediction techniques schedule audio packets for playout and attempt to make a reasonable compromise between the number of lost packets, the one-way delay and the delay variation since these criteria cannot be optimized simultaneously. In particular, this paper will show that the proposed HMM technique makes a good compromise between the mean end-to-end delay, end-to-end delay standard deviation and average packet loss rate.     

Adaptive load balancing algorithm for multiple homing mobile nodes

In places where mobile users can access multiple wireless networks simultaneously, a multipath scheduling algorithm can benefit the performance of wireless networks and improve the experience of mobile users. However, existing literature shows that it may not be the case, especially for TCP flows. According to early investigations, there are mainly two reasons that result in bad performance of TCP flows in wireless networks. One is the occurrence of out-of-order packets due to different delays in multiple paths. The other is the packet loss which is resulted from the limited bandwidth of wireless networks. To better exploit multipath scheduling for TCP flows, this paper presents a new scheduling algorithm named Adaptive Load Balancing Algorithm (ALBAM) to split traffic across multiple wireless links within the ISP infrastructure. Targeting at solving the two adverse impacts on TCP flows, ALBAM develops two techniques. Firstly, ALBAM takes advantage of the bursty nature of TCP flows and performs scheduling at the flowlet granularity where the packet interval is large enough to compensate for the different path delays. Secondly, ALBAM develops a Packet Number Estimation Algorithm (PNEA) to predict the buffer usage in each path. With PNEA, ALBAM can prevent buffer overflow and schedule the TCP flow to a less congested path before it suffers packet loss. Simulations show that ALBAM can provide better performance to TCP connections than its other counterparts.     

快速数据包分流算法研究

基于“流”的数据包分类算法已经在第四层交换等领域中得到了应用,该类算法的特点是流表的容量大,流表的更新速度较快.“快速的数据包分流算法”采用了散列算法的基本思想,并引入了流的局部性原理来加速散列查找的过程,用软件对该算法进行了仿真测试,并在最后从时间复杂度和空间复杂度两个方面对其进行了性能分析.实验结果表明,该算法具有良好的时间复杂度和空间复杂度,可以实现快速的分流.