QoE prediction model for mobile video telephony

Interactive online video applications, such as video telephony, are known for their vulnerability to network condition. With the increasing usage of hand-held wireless mobile devices, which are capable of capturing and processing good quality videos, combined with the flexibility in an end-user movements have added new challenging factors for application providers and network operators. These factors affect the perceived video quality of mobile video telephony applications, unlike conventional video telephony over desktop computers. We investigate this impact on video quality of mobile video telephony in varying network conditions and end-users movement scenarios. Based on 312 live traces, we quantitatively derive the correlation between the perceived video quality and the network Quality of Service (QoS) and user mobility. With the results, we develop a Quality of Experience (QoE) prediction model for mobile video telephony using Support Vector Regression techniques. The prediction models display ≈ 0.8 pearson correlation with experimental data. Our methodology and findings can be used to guide the video telephony application providers and network operators to work towards satisfying end-user experience.     

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.     

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QoS量化用户体验质量的评价模型

针对网络环境中业务供应商提供的业务不能很好的满足用户需求的问题,引入用户体验质量（QoE）并结合服务质量（QoS）参数,通过仿真网络得到相应数据,使用matlab工具分别建立用户体验质量与比特率、用户体验质量与丢包率的评价模型.运用统计分析方法对评价模型进行数据分析,并与史蒂文斯幂定律、韦伯-费希纳定律等模型进行对比验证.结果表明,构建的评价模型能更精确的体现用户体验质量与服务质量的关系,为供应商提供了重要的参考依据,并指明了服务方向.     

Developing Skype-Based Reference Services

ABSTRACT

Internet voice and video calling have demonstrated a dramatic rise in quality in the past several years and have quickly become communication technologies with a disruptive impact on society. Librarians, who regularly adopt such technologies in outreach and public services, should evaluate Web-calling programs as potential sources of innovative reference programming. This article describes the development of Skype video and Internet telephony pilot reference services at the Ohio University Libraries and considers the utility of Skype and VoIP in a library setting.     

Multipoint-to-point communications for SHE surveillance with QoS and QoE management

The bandwidth-hunger applications of SHE (Smart Home Environment) can take advantage of the multipoint-to-point (MPP) connections to aggregate more bandwidth to gain user-perceived Quality of Experience (QoE) and network Quality of Service (QoS). The receiver-centric transport-layer R²CP (Radial Reception Control Protocol) was proposed to resolve the incapability of the MPP communication in conventional TCP and UDP. However, R²CP has no consideration to discriminate the importance in a packet payload which is critical to QoE and brings an issue for critical data packets that may be dropped in great risk of network congestion. In this paper, we thus present P-R²CP (Prioritized R²CP) to effectively decrease the loss ratio of critical data packets in MPP video streaming while the network is congested. P-R²CP is a cross-layer protocol that considers both the transport-layer issues and the media content’s properties in application-layer. Then, an example on MPP-UVS (MPP ubiquitous video surveillance) is presented as UVS is now a very important Internet application that requires QoS/QoE management to protect lives and assets especially in SHE. Our experiments are conducted on different kinds of surveillance videos over MPP links with different bandwidth and packet loss inserted. The experimental results demonstrate that, as the loss of critical packets is decreased by an order and much less critical data packets are dropped, P-R²CP can highly guard not only QoS but also QoE of SHE surveillance video streaming.     

Packet loss rate mapped to the quality of experience

In order to study the influence of packet loss on the users’ quality of experience QoE and establish the Mapping model of the two when the video transmit in the network, building a NS2?+?MyEvalvid simulation platform, by the method of modifying QoS parameters to simulate different degrees of packet loss, focus on the influence of packet loss on QoE and establish the mapping model between them. Experimental results show that, packet loss has a significant influence on Quality of experience. Packet loss rate and the Quality of experience presents a nonlinear relationship, and use Matlab to establish the mapping model, this model’s accuracy is high, easy to operate, can real-time detect packet loss has influence on the user’s quality of experience (QoE).     

异构网络中贪婪资源分配策略的研究

针对当前异构网络中多种业务并存的现状,提出了一种可应用于由移动微波存取全球互通和通用移动通信系统组成的异构网络中考虑切换业务优先的贪婪资源分配策略,并通过网络中的呼损率验证了该策略的可行性;接着,在NS2环境下将该策略与公平资源分配策略在资源利用率、分组丢失率方面进行了比较.仿真结果显示,新策略可以提供良好的服务质量,前者的资源利用率和分组丢失率比后者分别改善了13％和14％左右.     

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.     

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.     

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。     

VoIP over DVB-RCS satellite systems: Trial results and the impact of adaptive speech coding using cross-layer design

VoIP is one of the most popular applications on the Internet. The DVB-S2 (digital video broadcasting-satellite version 2) and DVB-RCS (digital video broadcasting-return channel satellite) standards provide the potential for reliable and efficient voice and data transfer over satellite networks. This is important, as satellites can play a role in broadband service provision by addressing the cases of limited terrestrial access such as in rural or underserved areas. The first part of this paper presents the results of our VoIP trials with different commercial DVB-S/RCS satellite offers and popular internet telephony and videoconferencing applications (Skype, MSN, etc.). These results reveal that packet delay and jitter are strongly affected by the satellite network component as well as the type of speech codec used. Accordingly, research presented in the second part of this paper is focused on dynamic speech coding rate control adapted to the conditions of the underlying network, in which the satellite domain presents the most challenging portion of the end-to-end path. For this purpose, a novel cross-layer mechanism is proposed to facilitate and increase the accuracy of the speech coding rate adaptation mechanism. Cross-layer design is a relatively new idea aiming to exploit information exchange among layers of the protocol stack. Our simulation analyses show that the proposed cross-layer mechanism can help us in optimally adjusting the speech coding rate to maintain the user-perceived quality in terms of MOS (mean opinion score) in the face of time-varying available satellite channel capacity.     

服务功能链中基于机器学习的QoE评估与预测

在软件定义网络与网络功能虚拟化协同的网络架构下,只考虑单个服务质量（QoS）指标的服务功能链部署无法满足用户的多业务体验需求。提出一种基于机器学习的服务功能链部署模型。基于层次分析法构造MPNQ2算法以建立QoS与体验质量（QoE）的映射关系,得出影响QoE的网络参数并评估其影响权重。在此基础上,利用具备较强综合学习和泛化能力的随机森林模型对服务功能链的QoE进行预测。实验结果表明,与梯度提升决策树、线性判别分析等机器学习模型相比,随机森林模型为预测QoE的最佳模型,同时在影响QoE的网络参数中,丢包率对服务功能链的部署影响最大。     

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.     

QoE assessment model for multimedia streaming services using QoS parameters

This paper proposes a video QoE (Quality of Experience) assessment model which can assess video quality of experience with only QoS (Quality of Service) parameters and their relative importance at network layer. Since network or service providers can forecast whether to provide multimedia services above a certain level of service quality using the proposed model, they can offer and maintain optimum network environment for multimedia service such as IPTV. Through an experiment of video quality comparison we show that our QoS/QoE correlation model is closely related with video quality degradation patterns to network environmental change.     

MMS业务的QoE定义与测量

用户体验质量已成为移动通信中的一个热门话题,为了有效地测量多媒体短信业务（MMS）的用户体验质量,分析了QoE的定义以及它与传统QoS的区别,介绍了QoE的KQI和KPI之间的相互关系并建立起QoE的映射模型;然后,结合MMS业务的实现流程以及所涉及的网元设备详细定义了MMS业务中的KQIs与KPIs,最后,针对MMS业务提出了一种基于统计样本的QoE测量方案,通过测量明确KPIs与QoE之间的关系以及影响用户体验质量的因素,这对于提高用户的体验质量有重要意义。     

Admission control by implicit signaling in support of voice over IP over ADSL

Voice over DSL (VoDSL) is a technology that enables the transport of data and multiple voice calls over a single copper-pair. Voice over ATM (VoATM) and Voice over IP (VoIP) are the two main alternatives for carrying voice over DSL. ATM is currently the preferred technology, since it offers the advantage of ATM’s built-in Quality of Service (QoS) mechanisms. IP QoS mechanisms have been maturing only in recent years. However, if VoIP can achieve comparable performance to that of VoATM in the access networks, it would facilitate end-to-end IP telephony and could result in major cost savings. In this paper, we propose a VoIP-based VoDSL architecture that provides QoS guarantees comparable to those offered by ATM in the DSL access network. Our QoS architecture supports Premium and Regular service categories for voice traffic and the Best-Effort service category for data traffic. The Weighted Fair Queuing algorithm is used to schedule voice and data packets for transmission over the bottleneck link. Fragmentation of large data packets reduces the waiting time for voice packets in the link. We also propose a new admission control mechanism called Admission Control by Implicit Signaling. This mechanism takes advantage of application layer signaling by mapping it to the IP header. We evaluate the performance of our QoS architecture by means of a simulation study. Our results show that our VoIP architecture can provide QoS comparable to that provided by the VoATM architecture.     

OntoSPIT: SPIT management through ontologies

VoIP enables new ways for communication. At the same time, it provides new means, in terms of transmitting bulk unsolicited messages and calls, namely SPam over Internet telephony (SPIT). In this paper, we propose a conceptual model, based on an underlying ontology, which describes the SPIT domain. The ontology provides capabilities, such as modeling the SPIT phenomenon in a SIP-based VoIP environment, a common understanding of SPIT domain, as well as reusable SPIT-related knowledge interoperability, aggregation and reasoning. We demonstrate that the proposed ontology, combined with a set of SPIT identification criteria, as its underlying axioms and rules, could enhance the correlation and management of SPIT incidents. It could also support SPIT detection, thus facilitating the better protection of VoIP environments in a holistic, cooperative, and effective way.     

Evaluating perceived voice quality on packet networks using different random neural network architectures

Voice over Internet Protocol (VoIP) is one of the fastest growing technologies in the world. In VoIP speech signals are transmitted over the same network used for data communications. The internet is not a robust network and is subjected to delay, jitter, and packet loss. It is very important to measure and monitor the quality of service (QoS) the users experience in VoIP networks; this is not an easy task and usually requires subjective tests. In this paper we have analyzed three non-intrusive models to measure and monitor voice quality using Random Neural Networks (RNN). A RNN is an open queuing network with positive and negative signals. We have assessed the voice quality based on various parameters i.e. delay, jitter, packet loss, and codec. In our approach we have used the Mean Opinion Score (MOS) calculated using a Perceptual Evaluation of Speech Quality (PESQ) algorithm to generate data for training the RNN model. We have studied two feed-forward models and a recurrent architecture. We have found that the simple feed-forward architecture has produced the most accurate results compared to the other two architectures.     

一种视频质量监测系统的设计与实现

网络服务提供商希望能从用户的角度了解目前网络所提供的服务质量,而用户也希望获得定量的指标来评价当前网络服务质量。为此,以视频质量监测为研究对象,提出一种面向用户体验质量的网络监测系统。通过实验分析了网络传输过程中QoS参数对视频QoE的影响;提出一种将视频流转化为测试序列的视频丢包测量方法,该方法能低入侵、准确测量视频传输过程中的丢包情况;基于以上的研究成果,通过对MIB库的扩展和对MIB库轮询机制的研究,构建了面向QoE的视频服务监测系统,该监测系统结构简单、可行性强,实验表明可实时对网络中的视频服务质量进行监测。