A VoIP Traffic Identification Scheme Based on Host and Flow Behavior Analysis

With the development of network and multimedia coding techniques, more and more Voice over Internet Protocol (VoIP) applications have emerged. The traffic identification on VoIP applications becomes an important issue in network management and traffic analysis. In this paper, a new traffic identification scheme, which combines traffic flow statistic analysis with host behavior estimation, is proposed to identify the VoIP traffic at the transport layer of the Internet. The host IP addresses and the port numbers are examined as the host behavior to distinguish the VoIP traffic from traditional traffic flows. The packet size has been modeled by a function of entropy while the inter-packet time has been modeled by the self-adaptive estimation. The experiment results show that our scheme could obtain a stable performance. At the same time, the proposed scheme could maintain its validity when existing VoIP applications are updated or the new ones admitted. Both accuracy and flexibility can be improved.     

Formal analysis for robust anti-SPIT protection using model checking

Anti-SPIT policies counter the SPam over Internet Telephony (SPIT) by distinguishing bots launching unsolicited bulks of VoIP calls from human beings. We propose an Anti-SPIT Policy Management mechanism (aSPM) that detects spam calls and prevents VoIP session establishment by the Session Initiation Protocol (SIP). The SPIN model checker is used to formally model and analyze the robustness of the aSPM mechanism in execution scenarios with parallel SIP sessions. In case of a possible design flaw, the model checker provides a trace of the caught unexpected behavior (counterexample), that can be used for the revision of the mechanism’s design. Our SPIN model is parameterized, based on measurements from experiments with VoIP users. Non-determinism plays a key role in representing all possible anti-SPIT policy decisions, in terms of the SIP messages that may be exchanged. The model checking results provide evidence for the timeliness of the parallel SIP sessions, the absence of deadlocks or livelocks, and the fairness for the VoIP service users. These findings ensure robust anti-SPIT protection, meaning that the aSPM mechanism operates as expected, despite the occurrence of random SPIT calls and communication error messages. To the best of our knowledge, this is the first analysis for exhaustively searching security policy flaws, due to complex interactions between anti-SPIT measures and the SIP protocol services.     

Traffic analysis attacks on Skype VoIP calls

Skype is one of the most popular voice-over-IP (VoIP) service providers. One of the main reasons for the popularity of Skype VoIP services is its unique set of features to protect privacy of VoIP calls such as strong encryption, proprietary protocols, unknown codecs, dynamic path selection, and the constant packet rate. In this paper, we propose a class of passive traffic analysis attacks to compromise privacy of Skype VoIP calls. The proposed attacks are based on application-level features extracted from VoIP call traces. The proposed attacks are evaluated by extensive experiments over different types of networks including commercialized anonymity networks and our campus network. The experiment results show that the proposed traffic analysis attacks can greatly compromise the privacy of Skype calls. Possible countermeasure to mitigate the proposed traffic analysis attacks are analyzed in this paper.     

无线VoIP系统语音加密算法的研究

随着无线网络的飞速发展和VoIP技术的日渐成熟,无线VoIP技术应运而生。由于无线网络安全的脆弱性和VoIP系统本身的安全问题使得无线VoIP系统在安全方面存在着各种隐患。为提高无线VoIP的安全性,采用高压缩率的语音编解码技术G.729提高无线VoIP的通话质量;采用高级加密标准(AES算法)加解密采用G.729压缩过的语音信息;选用椭圆曲线密码(ECC算法)传输AES算法中用到的会话密钥;利用混沌系统可以提供可重复的随机数序列且其序列仅与系统参数和初值有关的性质确保了会话密钥的保密性。     

Context-aware adaptive IP mobility anchoring

The novel distributed mobility management trend is a promising direction to cope with the increasing mobile data traffic and flatten network architectures. Most of the novel mobility approaches distribute the mobility anchors through the access level, as opposed to the centralized mobility anchoring model. Other recent approaches argue that mobility anchors closer to the content servers may be the solution to optimize the mobility performance. However, none of the mobility anchoring models is ideal for all scenarios, since it depends on the user, the session and the network. Hence, we propose an IP mobility approach driven by the context of the user, sessions and the network, where the mobility anchors for IP address allocation and for routing/forwarding are distributed through the network nodes, while the mobility context is managed by the mobile devices. Although each session is properly anchored in the establishment phase, the routing/forwarding is adapted over time, according to the user, the session and the network context: the proposed approach is able to signal different mobility anchors to optimize the routing path to new and ongoing sessions of the user. The outcome of the evaluation shows that the proposed approach overall reduces the data cost, the data delay, the tunneled packets and the tunnel length, when compared with other anchoring models.     

VoIP网关中多链路SIP信令模块的设计与实现

文中在基于Linux操作系统的VoIP网关中，以单链路SIP信令实现为基础，提出了采用多进程架构的多链路SIP信令模块的设计实现方案。该设计继承了SIP协议固有的简单，可扩展性强等优点，而且多进程架构的设计结构清晰，能同时支持多路VoIP会话的建立与维护，解决了多用户的VoIP服务接入问题。该方案目前已用于基于Linux平台的VoIP网关的信令实现，完全满足VoIP网关的多链路处理与维护的需求。     

交通数据中的会话识别

会话识别因其能够提供对用户行为模式的深入理解而备受关注。交通数据会话是指用户为了完成某个任务而经过的交通路口序列。该文中我们采用超时和统计语言模型两种方法来进行会话识别。超时方法主要考察相邻交通路口之间的时间间隔对会话识别的影响,而统计语言模型则考虑路口序列的全局规律性。我们在交通数据集上进行了大量的实验,并通过比较分析两种方法性能上的差异得知时间因素比全局规律性在会话识别中的影响更大。     

Application-aware scheduling for VoIP in Wireless Mesh Networks

Wireless Mesh Networks (WMNs) are seen as a means to provide last mile connections in Next Generation Networks (NGNs). Because of their auto-configuration capabilities and the low deployment cost WMNs are considered to be an efficient solution for the support of multiple voice, video and data services in NGNs. This paper looks at the optimal provision of resources in WMNs for Voice over IP (VoIP) traffic, which has strict performance requirements in terms of delay, jitter and packet loss. In WMNs, because of the challenges introduced by wireless multi-hop transmissions and limited resources, providing performance quality for VoIP comparable to the voice quality in the traditional circuit-switched networks is a major challenge.This paper analyses different scheduling mechanisms for TDMA-based access control in mesh networks as specified in the IEEE 802.16-2004 WiMAX standard. The performance of the VoIP applications when different scheduling mechanisms are deployed is analysed on a variety of topologies using ns-2 simulation and mathematical analysis. The paper concludes that on-demand scheduling of VoIP traffic – typically deployed in 802.11-based WMNs – is not able to provide the required VoIP quality in realistic mesh WiMAX network scenarios and is therefore not optimal from a network operator’s point of view. Instead, it is shown, that continuous scheduling is much better suited to serve VoIP traffic. The paper then proposes a new VoIP-aware resource coordination scheme and shows, through simulation, that the new scheme is scalable and provides good quality for VoIP service in a wide range of network scenarios. The results shown in the paper prove that the new scheme is resilient to increasing hop count, increasing number of simultaneous VoIP sessions and the background traffic load in the network. Compared to other resource coordination schemes the VoIP-aware scheduler significantly increases the number of supported calls.     

Service level agreements (SLAs) parameter negotiation between heterogeneous 4G wireless network operators

We are currently witnessing a growing interest of network operators to migrate their existing 2G/3G networks to 4G technologies such as long-term evolution (LTE) to enhance the user experience and service opportunities in terms of providing multi-megabit bandwidth, more efficient use of radio networks, latency reduction, and improved mobility. Along with this, there is a strong deployment of packet data networks such as those based on IEEE 802.11 and 802.16 standards. Mobile devices are having increased capabilities to access many of these wireless networks types at the same time. Reinforcing quality of service (QoS) in 4G wireless networks will be a major challenge because of varying bit rates, channel characteristics, bandwidth allocation and global roaming support among heterogeneous wireless networks. As a mobile user moves across access networks, to the issue of mapping resource reservations between different networks to maintain QoS behavior becomes crucial. To support global roaming and interoperability across heterogeneous wireless networks, it is important for wireless network operators to negotiate service level agreement (SLA) contracts relevant to the QoS requirements. Wireless IP traffic modeling (in terms of providing assured QoS) is still immature because the majority of the existing work is merely based on the characterization of wireless IP traffic without investigating the behavior of queueing systems for such traffic. To overcome such limitations, we investigate SLA parameter negotiation among heterogeneous wireless network operators by focusing on traffic engineering and QoS together for 4G wireless networks. We present a novel mechanism that achieves service continuity through SLA parameter negotiation by using a translation matrix, which maps QoS parameters between different access networks. The SLA matrix composition is modeled analytically based on the G/M/1 queueing system. We evaluate the model using two different scheduling schemes and we derive closed form expressions for different QoS parameters for performance metrics such as packet delay and packet loss rate. We also develop a discrete event simulator and conduct a series of simulation experiments in order to understand the QoS behavior of corresponding traffic classes.     

DEVS-Based modeling of VoIP spam callers’ behavior for SPIT level calculation

At the present time the VoIP (Voice over Internet Protocol) service is generally accepted as an alternative for people seeking cheaper means to make a phone call. Users of VoIP service may fall anywhere along a spectrum between types at two extremes: one of which is an ordinary caller who doesn’t use the telephone for commercial purposes, while the other is a person who generates spam calls for commercial purposes. The focus of this paper concerns modeling of spam callers’ behavior to calculate the SPIT (Spam over Internet Telephony) level for management of the quality of service. From the perspective of a VoIP service provider’s view, spam callers are also a type of customer and sometimes they are valuable for increasing revenue. However, if a service provider does not manage spam calls, it can harm their business, because ordinary users might not receive phone calls using the phone numbers of the VoIP service. Thus, there is a trade-off between revenues and usability in managing spam calls in the VoIP service. This work presents a model of spam caller’s behavior using the DEVS (Discrete Event System Specification) formalism. The DEVS formalism is applicable as a model of behavior, by defining the state and state transition of the target of the model. In our model, we use six main parameters to define the states and state transitions. Each state is represented by a number which indicates the SPIT level of a caller. If the value is 1.0 then the caller is more similar to a spam caller. Based on the model definition, we constructed a SPIT level Calculation UI (User Interface) that is used to manage spam calls to improve VoIP service quality.     

TOP-N选择Markov预测模型

分析了访问用户和浏览器的行为,研究了现存的Markov预取模型,并分析了Markov预测模型的本质,在此基础上,提出了基于TOP N选择的Markov预测模型。该模型利用Web访问日志中请求次数大于N的URL生成TOP N,根据用户的访问会话生成Markov链。如果用户当前的访问会话与Markov链匹配,该Markov的下一URL在TOP N中,就把它取到本地缓存。实验表明,该预测模型能有效提高预测精度和命中率,在一定程度上还减少了带宽的需求。     

基于改进的SOM网络模型的VoIP QoS应用研究

VoIP的服务质量（QoS,Quality of Service）评估可以采用一系列可度量的参数来描述：业务可用性、吞吐量、延迟、抖动、分组丢失率等。现有的感知语音质量评价（PESQ）很难对不同环境下的网络结构进行实时和恰当的语音等级质量分类。为了能够综合考虑几种QoS相关因素,在给出改进的自组织映射神经网络模型（ESOMNN）的基础上,利用ESOM能够对高维输入数据有效分类的特点,提出了将端到端延迟、丢包率、抖动、语音编码以及测试系统标识作为ESOMNN的输入数据,在对采样数据进行训练后可自动完成语音质量评价和映射,并能根据得到的实时变量有效地评价包含多种相关因素的QoS级别。     

Detailed Analysis of Skype Traffic

Skype is beyond any doubt the VoIP application in the current Internet application spectrum. Its amazing success has drawn the attention of telecom operators and the research community, both interested in knowing its internal mechanisms, characterizing its traffic, understanding its users' behavior. In this paper, we investigate the characteristics of traffic streams generated by voice and video communications, and the signaling traffic generated by Skype. Our approach is twofold, as we make use of both active and passive measurement techniques to gather a deep understanding on the traffic Skype generates. From extensive testbed experiments, we devise a source model which takes into account: i) the service type, i.e., SkypeOut calls or calls between two Skype clients, ii) the selected source Codec, iii) the adopted transport layer protocol, and iv) network conditions. Leveraging on the use of an accurate Skype classification engine that we recently proposed, we study and characterize Skype traffic based on extensive passive measurements collected from our campus LAN.     

Framework based on stochastic L-Systems for modeling IP traffic with multifractal behavior

This paper presents and compares a set of traffic models, and associated parameter fitting procedures, based on so-called stochastic L-Systems, which were introduced by biologist A. Lindenmayer as a method to model plant growth. Starting from an initial symbol, an L-System generates iteratively sequences of symbols, belonging to an alphabet, through successive application of production rules. In a traffic modeling context, the symbols are interpreted as packet arrival rates or mean packet sizes, and each iteration is associated to a finest time scale of the traffic. These models are able to capture the multiscaling and multifractal behavior sometimes observed in Internet traffic. We describe and compare four traffic models, one characterizing the packet arrival process, and the other three characterizing both the packet arrival and the packet size processes. The models are tested with several measured traffic traces: the well-known pOct Bellcore, a trace of aggregate WAN traffic and two traces of specific applications (Kazaa and Operation Flashing Point). We assess the multifractality of these traces using Linear Multiscale Diagrams. The traffic models are evaluated by comparing, for the measured traffic and for traffic generated according to the inferred models, the probability mass function, the autocovariance function and the queuing behavior. Our results show that the L-System based traffic models that characterize both the packet arrival and packet size processes can achieve very good fitting performance in terms of first- and second-order statistics and queuing behavior.     

城域网出口链路VoIP通话时长的建模与预测

基于IP技术的语音分组传输（VoIP）电话业务目前被广泛部署于全国范围内的城域网。通过对某城域出口链路上实际采集的VoIP综合统计数据,针对有季节变动规律的单因素时间序列,推导建立了ARIMA（2,1,2）×（2,1,2）24季节乘积模型,并以此作为预测的基础,进行了该链路VoIP通话时长的预测。实验表明,模型的预测平均误差为8．38%。应用该模型检测将来超越阀值的可能发生时间,可以预先采取措施,保证VoIP的服务质量。     

基于语音质量预测的Speex自适应码率控制算法

为实现IP语音(VoIP)质量的动态管理与控制,提出了一种基于语音质量预测的自适应码率控制算法。通过实时预测VoIP通话的瞬时语音质量和总体语音质量,自适应地调整Speex编码参数,从而根据需要选择最佳编码速率。实验仿真结果表明,提出的算法能够有效减少网络拥塞,提高VoIP系统的语音质量。     

The changing usage of a mature campus-wide wireless network

Wireless Local Area Networks (WLANs) are now commonplace on many academic and corporate campuses. As “Wi-Fi” technology becomes ubiquitous, it is increasingly important to understand trends in the usage of these networks. This paper analyzes an extensive network trace from a mature 802.11 WLAN, including more than 550 access points and 7000 users over seventeen weeks. We employ several measurement techniques, including syslog messages, telephone records, SNMP polling and tcpdump packet captures. This is the largest WLAN study to date, and the first to look at a mature WLAN. We compare this trace to a trace taken after the network’s initial deployment two years prior.We found that the applications used on the WLAN changed dramatically, with significant increases in peer-to-peer and streaming multimedia traffic. Despite the introduction of a Voice over IP (VoIP) system that includes wireless handsets, our study indicates that VoIP has been used little on the wireless network thus far, and most VoIP calls are made on the wired network.We saw greater heterogeneity in the types of clients used, with more embedded wireless devices such as PDAs and mobile VoIP clients. We define a new metric for mobility, the “session diameter”. We use this metric to show that embedded devices have different mobility characteristics than laptops, and travel further and roam to more access points. Overall, users were surprisingly non-mobile, with half remaining close to home about 98% of the time.     

基于H.323协议的VoIP语音流量识别

通过分析H.323协议和H.323协议的会话流程,基于H.323协议的VoIP通信过程中出现的会话特征,提取出通信双方的元组信息来识别整个会话流量,设计出相应的流存储、搜索、更新方案和识别VoIP流量的算法。实验表明,该方法能够准确识别基于H.323协议的VoIP语音流量。     

Design and realization of ad-hoc VoIP with embedded p-SIP server

In this paper we design and implement the pseudo session initiation protocol (p-SIP) server embedded in each mobile node to provide the ad-hoc voice over Internet protocol (VoIP) services. The implemented p-SIP server, being compatible with common VoIP user agents, integrates the standard SIP protocol with SIP presence to handle SIP signaling and discovery mechanism in the ad-hoc VoIP networks. The ad-hoc VoIP signaling and voice traffic performances are analyzed using E-model R rating value for up to six hops in the implemented test-bed. We also conduct the interference experiments to imitate the practical ad-hoc VoIP environment. The analyzed results demonstrate the realization of ad-hoc VoIP services by using p-SIP server.     

mVoIP for P2P service based authentication system using AA authentication server

VoIP (Voice over Internet Protocol), which provides voice calls as well as additional services at cheaper prices than PSTN (Public Switched Telephone Network), is gaining ground over the latter, which had been the dominant telephone network in the past. This kind of a VoIP service is evolving into a dedicated mVoIP service for the smartphone which allows calls to be made at cheap prices using a WiFi network, as the number of smartphone users is skyrocketing as of late. While an increase in the user base is expected for mVoIP, a packet network is an open network which means anyone can easily gain access and so there can be various problems. To mitigate this, in this paper an authentication system is designed which has an AA (Attribute Authority) server added to VoIP in order to increase security and discriminate user access. In this paper a system for addressing security vulnerabilities from the increase in the use of VoIP services and providing differentiated services according to user access privileges is designed. This paper is organized as follows: Chapter 1 gives the introduction; Chapter 2 is on related research; Chapter 3 describes the proposed technique and system; Chapter 4 implements the system and analyzes its the performance; and Chapter 5 gives the conclusions.