Real-time and content-aware applications and infrastructure: a holistic approach dealing with architectural, performance and security issues

Voice over Internet protocol (VoIP) has become a popular alternative to traditional public-switched telephone network (PSTN) networks that provides advantages of low cost and flexible advanced ldquodigitalrdquo features. The flexibility of the VoIP system and the convergence of voice and data networks brings with it additional security risks. These are in addition to the common security concerns faced by the underlying IP data network facilities that a VoIP system relies on. The result being that the VoIP network further complicates the security assurance mission faced by enterprises employing this technology. It is time to document various security issues that a VoIP infrastructure may face and analyze the challenges and solutions that may guide future research and development efforts. In this paper, we examine and investigate the concerns and requirements of VoIP security. After a thorough review of security issues and defense mechanisms, we focus on attacks and countermeasures unique to VoIP systems that are essential for current and future VoIP implantations. Then, we analyze two popular industry best practices for securing VoIP networks and conclude this paper with further discussion on future research directions. This paper aims to direct future research efforts and to offer helpful guidelines for practitioners.

Voice over internet protocol (VoIP) is being widely used, while the integration of voice and data provide several opportunities. Lower cost and more flexibility are the main advantages of the VoIP which derived the attention of the enterprises to it. Unfortunately in the other extreme, some security issues may come across with the extensive use of VoIP. The purpose of this review paper is to encourage the VoIP industry to consider the importance of the security matters of their application and to find the gaps by discussing the pros and cons of each application and performing the fair comparison. This article begins by discussing the motivation of security measures in VoIP applications and a list of possible threats such as denial-of-service (DoS), Call Hijacking and Interception. These security issues are the most well known attacks that can be carried out in current VoIP deployment. Next, we discuss about two VoIP applications, which are Skype and GTalk, and their security levels have been measured briefly by highlighting some of the security issues in them. These security measurements are used to perform the comparison between the studied VoIP applications. Finally some methods have been proposed to improve the security in the VoIP applications in future state.

The usage of voice over IP (VoIP) traffic in IEEE 802.11 wireless networks is expected to increase in the near future due to widely deployed 802.11 wireless networks and VoIP services on fixed lines. However, the quality of service (QoS) of VoIP traffic in wireless networks is still unsatisfactory. In this thesis, I identify several sources for the QoS problems of VoIP traffic in IEEE 802.11 wireless networks and propose solutions for these problems. The QoS problems discussed can be divided into three categories, namely, user mobility, VoIP capacity, and call admission control. User mobility causes network disruptions during handoffs. In order to reduce the handoff time between Access Points (APs), I propose a new handoff algorithm, Selective Scanning and Caching, which finds available APs by scanning a minimum number of channels and furthermore allows clients to perform handoffs without scanning, by caching AP information. I also describe a new architecture for the client and server side for seamless IP layer handoffs, which are caused when mobile clients change the subnet due to layer 2 handoffs. I also present two methods to improve VoIP capacity for 802.11 networks, Adaptive Priority Control (APC) and Dynamic Point Coordination Function (DPCF). APC is a new packet scheduling algorithm at the AP and improves the capacity by balancing the uplink and downlink delay of VoIP traffic, and DPCF uses a polling based protocol and minimizes the bandwidth wasted from unnecessary polling, using a dynamic polling list. Additionally, I estimated the capacity for VoIP traffic in IEEE 802.11 wireless networks via theoretical analysis, simulations, and experiments in a wireless test-bed and show how to avoid mistakes in the measurements and comparisons. Finally, to protect the QoS for existing VoIP calls while maximizing the channel utilization, I propose a novel admission control algorithm called QP-CAT (Queue size Prediction using Computation of Additional Transmission), which accurately predicts the impact of new voice calls by virtually transmitting virtual new VoIP traffic.

In recent years, multimedia and game applications have experienced rapid growth at an explosive rate both in quantity and complexity. Since these applications typically demand 1010 to 1011 operations to be executed per second, higher processing capability is expected. Therefore, stream processors are becoming popular because of its performance advantages in the domains of signal processing, multimedia and etc. To provide sufficient computing capability, multi-SIMD units are employed in the stream processors. Moreover, to overcome the centralized register file constraint, hierarchical register organization is proposed and widely used in stream processors. In upper level of the hierarchy, distributed register file (DRF) becomes the dominant design and there are explicit interconnections among the DRFs managed by the compiler in a VLIW manner. Moreover, in order to further exploit the nice locality characteristics in multimedia applications, the lower level is a multi-banked register file where each bank is accessed by several SIMD units through a shared data bus. We will refer to the architecture with such characteristics as MLRM- SIMD architecture. Although such a design suits for multimedia processing, low data bandwidth caused by the shared data bus between multi-level registers severely impedes the programs mapping to MLRM-SIMD architecture. If this constraint can not be resolved well, the parallelism among multi-SIMD units would also be influenced as well as the whole program performance. Therefore, one of the major challenges to optimizing techniques for MLRM-SIMD architecture is to resolve the shared bus conflicts well. However, when generating the executable codes for MLRM-SIMD architecture, the compiler must simultaneously allocate many interdependent resources: the SIMD units on which the operations take place, the register files to hold the intermediate values and the shared bus to transfer the data for SIMD units. These conditions put very high pressure on the optimizing algorithms. Although read-read reuse operands among different SIMD units can be replicated, replicating all data would increase the register pressure and data spilling and reloading would also lead to the data bus conflicts. As there are interconnections among DRFs, some optimizations can be performed to improve the low data bandwidth. If the data is not required by many different SIMD units at the same time, they can be loaded by one SIMD unit only and transferred to other SIMD units through the interconnection between registers while being required. Such a manner would bring no data bus conflict and less local register pressure because only one data copy is saved in the DRFs. It is the major motivation of our optimizing algorithm. In this paper, we present a novel data pipeline optimization through communicating read-read source operands among different SIMD units to reduce the shared data bus conflicts. We use loop level parallelism to exploit the multi- SIMD computing ability and use read-read reuse communication to reduce the data bus conflicts. In contrast, traditional pipeline algorithms usually exploit pipeline parallelism and communicate computing results of prior stages. Therefore, we refer to our algorithm as data pipeline scheduling to distinguish it from traditional pipeline algorithm. Such a policy not only reduces the bus conflict, but also releases the register pressure. Experimental results show that it improves performance by 12% over the traditional method based on replication only. Based on the experimental results, some advice on programming is proposed for MLRM-SIMD architecture. When writing programs for MLRM-SIMD architecture, it is better to maintain the original structure in the algorithms, which would be much easier for compilers to exploit the parallelism in the programs and thus generate more efficient executable codes.

The work described in this Ph.D. thesis has been carried out within the context of computer networks and real-time multimedia applications over the Internet. Specifically, we focus on conferencing, a challenging service which experienced a wide growth during the last years. We examine the standardization efforts conducted in this field by the Internet Engineering Task Force (IETF), which are mainly focused on centralized conferencing as defined by the XCON Working Group. We actively contributed to such efforts by designing and implementing the defined framework and protocols, while also writing best current practice documents which are on the path to become Request For Comments (RFCs). The main outcome of such activities has been Meetecho, a standards-compliant multimedia conferencing and collaboration platform we developed. Meetecho has been conceived at the outset to be extensible towards a distributed architecture, yet being fully compliant with the XCON specification, in order to better fulfill scalability requirements resulting from its large-scale deployment. The distributed architecture, which is the subject of the DCON (Distributed Conferencing) proposal we submitted to the IETF, is thoroughly described herein, where we also provide the reader with the results of a scalability analysis we conducted in order to assess the actual performance improvement attainable with distribution. The figures obtained have been encouraging and definitely motivated us in pushing our proposal into the Internet standardization community. Finally, a remarkable part of this dissertation is focused on diagnosing and addressing issues that might arise when deploying multimedia architectures in the actual Internet.

Recent voice over Internet protocol (VoIP) technology is attracting more and more attention and interest in the industry. VoIP applications such as IP telephony systems involve sending voice transmissions as data packets over private or public IP networks as well as reassembled and decoded on the receiving side. Security concerns are always the major barrier that prevents many businesses from employing VoIP technologies. In this paper, we discuss various VoIP security threats and possible approaches to tackle the threats in VoIP applications

In the past, the main purpose of telephony network is to transfer the voice. But Since 1990, there have been many changes about it. First of all, wireless mobile communication network has enjoyed tremendous growth at the popularity. And the purpose of communication has been changed to data communication from voice one. Furthermore, people have started to require both data communication and voice communication. Because of these issues, VoIP (Voice over IP) is getting to be a matter of concern. And According to development of network and computer devices, people want to communicate by not only voice telecommunication but also video telecommunication. However because it uses public IP network, security issues are getting to be an important thing in VoIP. Actually, there are many solutions to protect data from attackers in VoIP System. But to serve video transmission needs more special care. To display real time video streaming on telephones need more bandwidth and computing power. This is a handicap for video streaming on the mobile device. Therefore we concentrate on the solution how to protect the video data from attackers with well quality of service. This is discussed in this paper.

The rise of Web 2.0 and the current, unprecedented diffusion of mobile devices have laid new foundations for the development of PGIS (Participatory GIS). This study evaluates the possibility of exploiting FOSS (Free and Open Source Software) tools to build up a PGIS prototype providing Web publication of user field-collected data. Besides increasing public awareness and collaboration, user-generated content should also enlarge the knowledge of specific phenomena up to the local level. A prototype architecture was designed and tested in relation to a simple, planning-related case study, i.e., the report of road pavement damages. Open Data Kit suite was used to gather georeferenced multimedia data using mobile device sensors (e.g., the GPS) and to store them into a PostgreSQL database with PostGIS spatial extension. Data was then Web-published using GeoServer. Web access was finally enabled from both traditional desktop-computers and mobile plat forms through ad hoc OpenLayers and Leaflet clientside solutions. The architecture provided support for FOSS applicability within the typical PGIS-related tasks, from field survey to data storage, management and dissemination on the Internet. This bottom-up communication paradigm, which exploits real-time, freely available user contributions, can become a potentially precious tool for making decision-processes more democratic, faster and ultimately better.

Fast secured speech transmission is certainly essential especially in real-time multimedia applications. This paper proposes an effective, low complexity, Selective On-Fly Scrambling Algorithm (SOFSA) that could be implemented on Code-Excited-Linear-Prediction (CELP) parameters while processing the speech frame in order to provide fast speech security. ITU G.723.1, which is one of the most widely used speech coders in Voice over Internet Protocol (VoIP) applications, is chosen as our test CELP coder. Subjective and objective measurements are used to prove SOFSA's utility in real-time voice applications. Results show that the implementation of SOFSA on selective CELP parameters, especially on high intelligibility density parameters, provides a form of data security without compromising neither the bandwidth of the output nor the speech quality. Thereby the proposed algorithm SOFSA could avoid the use of a separate complex data encryption algorithm with higher processing delay.

Voice over Internet Protocol (VoIP) service has been a very popular and important application over the Internet. Wireless VoIP also becomes more and more popular due to its features of low cost and convenience. Recently, wireless mesh network has been considered as a good solution for VoIP services since it is easy to deploy and provides a larger area coverage. However, the security and performance are the main challenges. In this work, we focus on the security issues and present a new protocol for securing the voice traffic over the wireless mesh network, including the client authentication, intermediate mesh nodes authentication to ensure secure multi-hop communication, voice traffic confidentiality and secure optimal routing selection by using the probing packets.

Online social network is emerging as a promising alternative for users to directly access video contents. By allowing users to import videos and re-share them through the social connections, a large number of videos are available to users in the online social network. The rapid growth of the user-generated videos provides enormous potential for users to find the ones that interest them; while the convergence of online social network service and online video sharing service makes it possible to perform recommendation using social factors and content factors jointly. In this paper, we design a joint social-content recommendation framework to suggest users which videos to import or re-share in the online social network. In this framework, we first propose a user-content matrix update approach which updates and fills in cold user-video entries to provide the foundations for the recommendation. Then, based on the updated user-content matrix, we construct a joint social-content space to measure the relevance between users and videos, which can provide a high accuracy for video importing and re-sharing recommendation. We conduct experiments using real traces from Tencent Weibo and Youku to verify our algorithm and evaluate its performance. The results demonstrate the effectiveness of our approach and show that our approach can substantially improve the recommendation accuracy.

Recent success of online social networks (OSNs) motivates the study of security issues in OSNs. A fundamental but challenging security issue in OSNs is to authenticate a friend's real identity. A solution to this issue will benefit a number of OSN security protocols. Existing solutions require users securely obtain some secret information from their friends before authentication takes place, which is not always possible in OSNs. In this paper, we propose a new authenticated key exchange protocol based on the exclusive secrets shared between friends. It provides identity authentication and key exchange in a plain setting, i.e., users do not need to securely exchange or distribute any information beforehand. The protocol is designed to work with low-entropy input information, because human beings are not good at dealing with a large amount of information. Another advantage of our protocol is its tolerance of input errors considering human error is always a possibility. We prove the security of the protocol in the universal composability (UC) framework and demonstrate its efficiency.

Over the last few years, many multimedia conferencing and Voice over Internet Protocol (VoIP) applications have been developed due to the use of signaling protocols in providing video, audio and text chatting services between at least two participants. This paper compares between two widely common signaling protocols: InterAsterisk eXchange Protocol (IAX) and the extension of the eXtensible Messaging and Presence Protocol (Jingle) in terms of delay time during call setup, call teardown, and media sessions.

The purpose of this study is to identify the key security issues within the context of software based VoIP (Voice over Internet Protocol) which are engaged in the process of data exchange on the converged networks. There is no one single solution for VoIP security; it is all depended on the type of customers, existing IT environment and carriers. In order to narrow down the focus we will concentrate on the SMEs (Small and Medium size Enterprises). This paper focuses on the key similarities and differences between softphone and hardphone, analyses VoIP security protocol and proposed a model for softphone security is presented.

Voice over Internet Protocol (VoIP) service has been a very popular and important application over the Internet. Recently, wireless mesh networks (WMNs) has been considered as a good solution for VoIP services due to low cost and convenience, easy deployment and larger area coverage. However, many relevant security issues have been raised for integrating IP Telephony into the existent applications and system infrastructure. In this paper, we propose a trust degree-based dynamic model for securing the VoIP over WMNs, in which each node maintains trust degree for other nodes and applies the trust degree to evaluate their trustworthiness. We also validate the performance of our model with simulation results