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.     

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。     

一种VoIP流量识别的新方法*

目前传统的基于端口的流量识别方法不适合VoIP流量识别,基于载荷分析的流量识别方法应用于实时监测存在开销过大的问题,基于统计模型的流量识别方法也不适合实时监测。为此,针对VoIP的传输特性,提出并实现了一种新的VoIP流量识别方法。实验表明,只根据传输层信息,该方法的误报率在10%左右。     

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.     

Accurate resource estimation for homogeneous VoIP aggregated traffic

Modern VoIP codecs like G.729, G.723.1 or AMR can generate traffic during voice inactivity periods for Comfort Noise Generation (CNG). This feature alters the classical on–off pattern typically used to model the traffic generated by codecs with a Silence Suppression scheme. Therefore, the traffic generated due to CNG leads to severe inaccuracies in the dimensioning analysis done through traditional models based on multiplexing on–off sources like MMPP or fluid model.This paper addresses the VoIP dimensioning issue. First, we extend the traditional MMPP and fluid analytical models to include those traffic sources which perform the CNG feature. Second, we propose a simple but efficient algorithm which can be applied in dimensioning or admission control to find out the bandwidth reservation required to guarantee delay and loss in a packet-switch multiplexer node for VoIP traffic. Results are validated by simulations and VoIP traces and demonstrate a significant improvement in accuracy with respect to current on–off-based approaches.     

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.     

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.     

VoIP development in China

Several factors have spurred the explosive growth of VoIP phone use in China, including customer incentives such as improved voice quality and lower cost per call, and provider incentives such as higher profits and upgrade paths to next-generation technologies. The voice over Internet protocol, also called IP telephony, offers a new type of service that uses the Internet protocol, intranets, and extranets to deliver voice information. In contrast to traditional telephone services, which operate through a circuit-switched network, VoIP uses a packet-switched network. This distinction results in differences in implementation, quality of service (QoS), and operating costs. Since the service was introduced to the public in China in April 1999, VoIP toll telephone traffic has increased with astonishing speed. By the end of 2002, VoIP toll telephone traffic had surpassed traditional toll telephone traffic in China in both domestic longdistance and international call areas, including phone calls to and from Hong Kong, Macao, and Taiwan.     

基于MPLS的IP语音网络恢复性能仿真分析*

提出了一种基于MPLS 的IP语音网络恢复性能评估方案,该方案利用OPNET Modeler工具构建一个基于MPLS的IP语音网络仿真平台,分析与评估基于MPLS的两种故障恢复技术。仿真结果表明,采用基于MPLS OAM技术的快速重路由算法可以有效地缩短流量重路由时间与改进IP语音网络服务质量,验证了基于MPLS OAM技术的FRR算法在VoIP网络系统中实现QoS性能网络优化的可行性。     

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。     

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.     

基于流统计特征的VoIP媒体网关检测方法*

针对已有VoIP检测方法对协议解析依赖性较大,且检测范围不尽全面的问题,提出基于流统计特征的VoIP媒体网关检测方法.该方法通过对RTP流平均包到达间隔作同流计算与同源多流统计,得出对应网络实体的平均包到达间隔概率分布.根据同一IP地址所对应的平均包到达间隔分布的概率峰值判定媒体网关,从而大幅提高VoIP业务检测效率.     

在WLAN实现VoIP的新兴多路访问协议

VoIP是一种在延时、抖动和丢包率上对WLAN系统要求很严格的实时应用。对于CSMA/CA而言,如果语音和数据传输同时存在的话,想要语音质量满足通话要求,就要在CSMA/CA上增加优先级控制。文章提出了一种新兴的多路访问控制协议,这个协议使基于自治分布式控制的无线局域网能够满足VoIP的要求。     

基于SIP的VoIP流量识别方法研究*

分析了已有流量识别方面的主要方法,针对基于SIP的VoIP流量,提出一种结合协议特征和协议流程分析的综合流量识别方法,并基于Libpcap库实现了对应的识别工具.     

VoIP业务入侵检测模型

提出一种SIP信令协议的入侵检测方法,用以加强VoIP业务环境的安全。重点对基于SIP的VoIP业务环境的安全威胁和业务流量分析,利用数据挖掘算法和改进的贝叶斯算法构建针对SIP下的入侵检测模型。实验结果表明,该方法可以对VoIP业务环境下的网络攻击进行有效检测。     

一种基于C4.5决策树的VoIP流量识别方法

针对互联网中常见的VoIP应用类型,重点从分类特征的选取出发,提出了一种基于C4.5决策树算法的VoIP流量识别方法,并分别利用模拟数据和真实网络数据对其进行了测试。结果表明,该方法识别准确率高,对含有不同协议种类VoIP的网络数据具有较强的适应能力。     

改进的WiMAX上行链路调度机制

基于IEEE 802.16e协议的实时业务调度机制,以及对VoIP业务特性的分析,提出了一种改进的上行链路调度机制,该调度机制自适应地调整数据带宽分配周期。从理论上分析了所提出的调度机制和已有调度机制的VoIP用户容量。系统仿真结果表明,在保证系统容量的前提下,所提出的调度机制可以有效地降低用户的上行时延,从而提高了服务质量。     

QoS-aware path switching for VoIP traffic using SCTP

Voice over Internet protocol (VoIP) has been a prevalent multimedia service nowadays. It allows us to transmit voice data over IP networks. However, quality of service (QoS) is a major challenge to VoIP services. It must provide similar quality to traditional public switched telephone network or cellular phone services. Therefore, QoS related protocols have become important for real-time applications. Multi-protocol label switch (MPLS) is one of the important techniques to improve the network performance from QoS point of view. It employs label swapping to speed up packet forwarding. However, when a large number of users utilize VoIP services, the network congestion issue still exists. It causes delay, jitter and packet loss that affect VoIP QoS. In this paper, we propose a QoS-aware path switching strategy by using stream control transmission protocol (SCTP) in MPLS network to improve the VoIP traffic. This was done by employing SCTP selective acknowledgment mechanism to report the transmission parameters of primary path and to determine the criteria to switch to backup path. Simulation results show significant improvement in VoIP QoS.     

VoWLAN中QoS延迟性能分析与改进

VoWLAN（VoIP over WLAN）技术是一种利用无线局域网（WLAN）来传输VoIP帧的技术。VoWLAN中的语音业务是对延时较为敏感的实时业务，因此对其作延迟分析尤为重要。重点介绍了VoWLAN系统中AP节点处延迟产生过程，并采用一种新的具有优先级的M/G/1排队分析模型对AP节点处的延迟进行详细分析，该模型比以往所采用的单一的基于泊松过程的分析模型更精确；探讨了减少延迟的方法，提出新的点协调功能（PCF）方式下站点轮询算法，该算法可以更好的支持VoWLAN系统的VoIP等实时业务；最后用OPNET仿真工具进行了验证。     

Energy-aware QoS uplink scheduling for UGS traffic flows in WiMAX networks

This paper presents a new energy-aware QoS scheduling and call admission control algorithm for WiMAX IEEE 802.16e wireless access standard. The scheduling algorithm works at MAC layer and is designed towards minimizing power consumption at mobile stations supporting multiple Unsolicited Grant Service (UGS) connections, while meeting the QoS requirements of the connections. The algorithm uses a novel idea to fill an active OFDM frame as much as possible in order to increase the number of OFDM frames in sleep mode at mobile station. The algorithm also considers the dynamic nature of connection joining and termination. We used Voice-over-IP (VoIP) traffic connection models for UGS traffic flows to simulate and validate algorithm. The simulation model was tested using VoIP codec types with different rates and QoS requirements. Simulation results show that a power saving in the range of 50–75% can be easily achieved at the mobile station under low-to-moderate traffic intensities.