The network and the wireless communication technology are all pervasive and growing. VoIP is very popular in recent years. Every IP phone requires a public IP address, but there are no enough public IP addresses. At present to solve this problem is use a NAT (Network Address Translation). NAT is a technology used for broadband connections which allows multiple devices to share one connection for accessing the Internet. The disadvantage is that Peer to Peer VoIP applications do not work behind NAT without settings. In this thesis, I introduce how to realize the VoIP by SIP operations which enables two SIP clients to communicate each other. Later, I explain five kinds of NAT and the SIP traversal over NAT, such as STUN, ICE, RTP Relay Server, UPnP and TURN. We found RTP Relay Server is the most easy and acceptable. The major disadvantage of using a RTP Relay Server is that the media will have to travel via a third party, the relay server, on the Internet. The quality of the phone call may be affected by the relay server. Therefore, I rewrite RTP Relay Server application in python to let it be able to use SNMP to measure whether the network flow is over the threshold. When the loading on the server is over than it will bear, the server will reject the connection. Then, this connection will be dispatched to another relay server to maintain a good communication quality.

In this research, we present a investigative study on how timing-based traffic analysis attacks can be used for recovery of the speech from a Voice Over Internet Protocol (VOIP) conversation by taking advantage of the reduction or suppression of the generation of traffic whenever the sender detects a voice inactivity period. We use the simple Bayesian classifier and the complex HMM (Hidden Markov Models) classier to evaluate the performance of our attack. Then we describe the usage of acoustic features in our attack to improve the performance. We conclude by presenting a number of problems that need in-depth study in order to be effective in carrying out silence detection based attacks on VOIP systems.

As Voice-over-IP (VoIP) technology developed, VoIP services alternate the traditional PSTN gradually with their advantage of low rates. Instead of using the public switched telephone network, VoIP services exchange voice information over Internet. As the result, VoIP services have to suffer from the weaknesses of the IP network infrastructure and VoIP devices are easier to be attacked than traditional phones. In our research, we analyze authentication mechanisms of Session Initiation Protocol, and address the weakness of current authentication mechanisms and the security threats to SIP. We use limited resources to implement an authentication mechanism in our embedded SIP user agent. The results of the Invite/Bye attack experiments confirm that our authentication mechanism in Direct Call and Proxy Call can both avoid malicious Invite/Bye attack.

Because of Skype, Voice over IP(VoIP) becomes much more hot and popular. It has always been considered to be a killer application of new generation of the Internet. With its distinct economic advantage and good voice quality, it has the tendency to replace the traditional Public Service Telephone Network(PSTN). Moreover, Session Initiation Protocol(SIP) is proposed by Next Generation Network(NGN) to be the first choice of voice and multimedia network control protocol. Client-server is the key architecture of SIP. Although this architecture is simple and easy to maintain, and even it has faster response time than P2P, the server may cause problems while the client is increasing. Moreover, it is possible that the large load may cause the service stop anytime. In this paper, we discuss how to design and implement an embedded VoIP user agent based on SIP standard. It can improve efficiency of SIP signaling delivery through the user characteristic of making calls with limited resources and reduce the number of services that servers to provide while calling. We also design a graphical user interface. The interface lets users feel friendly and make the method of the system more efficient.

VOIP has become very popular since Skype software was released in 2003. Due to the convenience and lower cost brought by unique P2P technology, Skype enters people¡¦s lives quickly. Skype changes the way people communicate and the way people think about VOIP. In Taiwan, PChome Online Inc. launched Skype business by Co- Branding in 2004. People tend to accept Skype very easily. As individual users became more and more, organizational user noticed this and started to use Skype. In addition to Skype, there are other competitors in Taiwan market like Mackay, APOL, and SaveCom. Taiwan Government opened the 070 VOIP business in 2007, many companies were eager to join this market. It¡¦s predictable that the opening of 070 VOIP will change the whole industry and affect VOIP, Telecommunication and Fixed Network companies a lot. This research chose VOIP as topic. First, this research analyzed the whole market and VOIP company¡¦s marketing strategy. Then this research used empirical analysis to study customer¡¦s satisfaction, loyalty and purchasing behavior in organizational buyer¡¦s view. The studies focusing on VOIP organizational buyer in Taiwan are rare, so this research reviewed literatures related customer¡¦s satisfaction and loyalty and design scale for VOIP organizational buyer. Finally, this research chose organizational buyers of leading service provider in this market, PChome & Skype and did a questionnaire survey. The result showed that 60% of Skype business users are located in north of Taiwan. The companies from manufacturing, wholesaling and retailing industries accounted for the mainly. 80% of Skype business users are small companies with less than 100 employees. About customer¡¦s satisfaction, the most satisfying items are ¡§software interface, convenience and channel, and good image.¡¨ The least diverse items are ¡§good image, safe online purchasing and redeeming, and convenience and channel. ¡¨ About the loyalty, each question is between ¡§slightly agree¡¨ and ¡§agree.¡¨ According to the correlation analysis, customer¡¦s satisfaction is positive correlated to customer¡¦s loyalty.

As the maturation of the VoIP technique, VoIP can not only satisfy the communicating requirement of telecommunication but also provide network multimedia services. In the VoIP technique, Session Initiation Protocol (SIP) is precisely one of main protocol that is proposed by Next Generation Network (NGN) to be the first choice of voice and multimedia network control protocol. SIP also may make the union with traditional PSTN even to substitute, and is easier to use and to operate for the PSTN user. In our research, we make a network billing system in view of the SIP environment, and with the increasing of the VoIP population, using single server is unable to afford so much loading. It is possible that the large load makes the service stop anytime. Using RADIUS (remote authentication dial-in user service), an AAA (authentication, authorization, accounting) protocol, can be used for conveying accounting information between an SIP proxy server and an accounting server. To make the VoIP service work anytime, through the features of DNS (Domain Name System) and RADIUS to achieve the service that load balancing and VoIP can be provided anytime.

In the 21st century's today, because of the computer and the telecommunication are combined with each other, VOIP (Voice over Internet Protocol) substitute for PSTN(Public Switched Telephone Network) stage by stage according to its cost and the quality of voice. VOIP digitizes analogic voice by software and hardware, and uses Packet Switching to transmit the packets in the IP network, its standard protocol are H.323, SIP, etc. H.323 is more complex and has more sub-protocol such that it is restricted by technicality. To compare with H.323, SIP is free. It can achieve the goal communication with PSTN through Gateway, and constructs a simple structure of VOIP. SIP has become the important protocol of NGN (Next Generation Networks) and 3G mutimedia systems. The structure of Client-Server of SIP is simple, but as long as the users increase, the Server may cause some trouble because of the great deal of information, and the VOIP network which depends on the Server will stop its service. For this reason, we will improve the Client-Server structure in the paper, we wish that clients could communication with the others without the service of SIP Server to reduce the load, and improve the perfomance of SIP signaling.

Over the last decade, the internet industry has rapidly grown with regard to infrastructure and bandwidth. Widespread internet networks with large bandwidth connect people-to-people, people-to-machines, and machine-to-machine. Like other multimedia services, large bandwidth enables voice services to be provided over IP networks where network connectivity is not consistent. In this context, research on service quality monitoring is necessary to satisfy customers by providing consistent service quality. The major contribution of this dissertation is the development of three novel techniques to improve or measure voice quality over IP networks. This dissertation first addresses an adaptive playout buffer scheduling algorithm that enables systems to lossen delay jitter due to the legacy of packet-switched networks. The scheduling algorithm is operated by a desired quality of service, minimizing the end-to-end delay by adjusting playout delay times. Secondly, this dissertation also explores a parameter-based nonintrusive speech quality measure to monitor the quality of VoIP. During the lifetime of sound, the network parameters are estimated and used to predict the quality of speech. As a cognitive model, a machine-learning technique is exploited to map features in the feature space into the perceived speech quality scale space. Finally, this dissertation introduces a signal-based nonintrusive speech quality measure. Features for the proposed measurement are extracted from observations of the characteristics of natural speech sounds and artificial noises. The calculated features are mapped into the perceived speech quality scale. The proposed parameter-based measure achieves a high prediction accuracy while the signal-based measure reaches to a comparable performance to the official International Telecommunication Union (ITU) standard, P.563. The contributions described in this dissertation provides smart methodologies for monitoring or enhancement of VoIP service qualities. / text

Fixed Broadband Wireless Access (BWA) for the last mile is a promising technology which can offer high speed voice, video and data service and fill the technology gap between Wireless LANs and wide area networks. This is seen as a challenging competitor to conventional wired last mile access systems like DSL and cable, even in areas where those technologies are already available. More importantly the technology can provide a cost-effective broadband access solution in rural areas beyond the reach of DSL or cable and in developing countries with little or no wired last mile infrastructure. Earlier BWA systems were based on proprietary technologies which made them costly and impossible to interoperate. The IEEE 802.16 set of standards was developed to level the playing field. An industry group the WiMAX Forum, was established to promote interoperability and compliance to this standard. This thesis gives an overview of the IEEE 802.16 WirelessMAN OFDM standard which is the basis for Fixed WiMAX. An in depth description of the medium access control (MAC) layer is provided and functionality of its components explained. We have concentrated our effort on enhancing the performance of Fixed WiMAX for VoIP services, and best effort traffic which includes e-mail, web browsing, peer-to-peer traffic etc. The MAC layer defines four native service classes for differentiated QoS levels from the onset. The unsolicited grant service (UGS) class is designed to support real-time data streams consisting of fixed-size data packets issued at periodic intervals, such as T1/E1 and Voice over IP without silence suppression, while the non-real-time polling service (nrtPS) and best effort (BE) are meant for lower priority traffic. QoS and efficiency are at opposite ends of the scale in most cases, which makes it important to identify the trade-off between these two performance measures of a system. We have analyzed the effect the packetization interval of a UGS based VoIP stream has on system performance. The UGS service class has been modified so that the optimal packetization interval for VoIP can be dynamically selected based on PHY OFDM characteristics. This involves cross layer communication between the PHY, MAC and the Application Layer and selection of packetization intervals which keep the flow within packet loss and latency bounds while increasing efficiency. A low latency retransmission scheme and a new ARQ feedback scheme for UGS have also been introduced. The goal being to guarantee QoS while increasing system efficiency. BE traffic when serviced by contention based access is variable in speed and latency, and low in efficiency. A detailed analysis of the contention based access scheme is done using Markov chains. This leads to optimization of system parameters to increase utilization and reduce overheads, while taking into account TCP as the most common transport layer protocol. nrtPS is considered as a replacement for contention based access. Several enhancements have been proposed to increase efficiency and facilitate better connection management. The effects of proposed changes are validated using analytical models in Matlab and verified using simulations. A simulation model was specifically created for IEEE 802.16 WirelessMAN OFDM in the QualNet simulation package. In essence the aim of this work was, to develop means to support a maximum number of users, with the required level of service, using the limited wireless resource.

Submitted by Fernanda Rodrigues de Lima (fernanda.rlima@ufpe.br) on 2018-10-05T21:52:12Z No. of bitstreams: 2 license_rdf: 811 bytes, checksum: e39d27027a6cc9cb039ad269a5db8e34 (MD5) DISSERTAÇÃO Anderson Wesley Alves Bezerra.pdf: 1979361 bytes, checksum: ac90c7988a2d3b56b4a1fb6c7d34698a (MD5) / Approved for entry into archive by Alice Araujo (alice.caraujo@ufpe.br) on 2018-11-14T21:05:11Z (GMT) No. of bitstreams: 2 license_rdf: 811 bytes, checksum: e39d27027a6cc9cb039ad269a5db8e34 (MD5) DISSERTAÇÃO Anderson Wesley Alves Bezerra.pdf: 1979361 bytes, checksum: ac90c7988a2d3b56b4a1fb6c7d34698a (MD5) / Made available in DSpace on 2018-11-14T21:05:11Z (GMT). No. of bitstreams: 2 license_rdf: 811 bytes, checksum: e39d27027a6cc9cb039ad269a5db8e34 (MD5) DISSERTAÇÃO Anderson Wesley Alves Bezerra.pdf: 1979361 bytes, checksum: ac90c7988a2d3b56b4a1fb6c7d34698a (MD5) Previous issue date: 2017-06-27 / Em um contexto onde os serviços de rede de dados nas organizações requerem melhor aproveitamento dos recursos disponíveis, essa pesquisa consiste em um estudo voltado a determinar a viabilidade da utilização da tecnologia VoIP em dispositivos móveis utilizando o padrão IEEE 802.11, no âmbito do IFMT - Campus Cáceres - Prof. Olegário Baldo. Este Campus possui dois links de dados (4 e 20Mbps) considerados pelo Departamento de Informática insuficientes para atender à demanda dos usuários. Além disto, a rede sem fio apresenta pontos críticos de sinal em função de obstáculos e/ou por interferência. Num primeiro momento foram realizadas enquetes entre os usuários, que permitiram verificar o grau de satisfação com os serviços providos pelo Campus e o interesse na utilização de aplicações com recurso de tecnologia VoIP. O resultado das enquetes revelou a insatisfação dos usuários com a qualidade do acesso à Internet, a necessidade de mobilidade no interior do Campus e a utilização de aplicativos VoIP em dispositivos móveis, com preferência aos aplicativos WhatsApp e Messenger. Em seguida, foram realizados experimentos com o objetivo de avaliar subjetivamente e objetivamente a qualidade das chamadas de voz destas aplicações em três cenários: comunicação local com baixa carga; handover com carga média; e handover no Campus e comunicações externas. Para estas avaliações foram utilizadas duas bandas de dados para cada comunicação, 15 e 30 kbps, que foram escolhidas a partir de testes preliminares em ambiente indoor. Os dispositivos utilizados no ambiente de teste foram configurados com parâmetros de qualidade de serviço (QoS) para o protocolo de voz e com redução da área de alcance para obter maior vazão e qualidade de sinal. As avaliações subjetivas foram realizadas com usuários do próprio Campus e alcançaram resultados no mínimo razoáveis. As avaliações objetivas foram realizadas utilizando os softwares Wireshark e um emulador Android (Nox) para captura dos pacotes e cálculo das métricas: atraso, variação de atraso, perda de pacotes e o handover, cujos resultados estiveram dentro dos limites estabelecidos na literatura. Também, através de um computador, com sistema operacional pfSense, foi possível gerenciar o controle de banda de dados e o acesso à Internet, bem como determinar o consumo da banda de dados utilizado pelas aplicações. Os resultados obtidos revelam que a viabilidade do uso de aplicações VoIP em dispositivos móveis depende da readequação da infraestrutura da rede sem fio, devendo-se atentar para o distanciamento entre os pontos de acesso, obstáculos, sobreposição de canais e potência do sinal. Apesar da insatisfação dos usuários com os serviços da Internet no Campus, de modo a satisfazer o desejo deles de utilizar aplicações VoIP móveis, recomenda-se reservar 500 kbps para atender no mínimo 35 usuários simultâneos, considerando que o ideal seria aumentar a banda do link de dados e não piorar ainda mais a qualidade percebida dos serviços atuais. / In a context where data network services in organizations require a better use of available resources, this research consists of a study aimed at determining the feasibility of using VoIP technology on mobile devices using the IEEE 802.11 standard, at IFMT - Campus Cáceres - Prof. Olegário Baldo, which has two data links (20 and 4 Mbps) considered insufficient by the IT department to meet the current number of users, it also presents critical points of signal due to obstacles and/or by interference. Initially it was carried out surveys among the users, which allowed us to verify the degree of satisfaction with the services provided by the Campus and the interest in the use of applications with VoIP technology. The result of the survey revealed the dissatisfaction of the users with the quality of Internet access, the desire for mobility within the Campus and the use of VoIP applications on mobile devices, with WhatsApp and Messenger applications as a preference. Afterwards, it was performed several experiments aiming at evaluating subjectively and objectively the quality of the voice calls of these applications in three scenarios: local communication with low load; handover with medium load and handover at the Campus and external communications. The evaluations were divided into two data bands, 15 and 30 kbps, which were chosen from preliminary indoor tests. The devices used in the test environment were set up with QoS parameters for the speech protocol and with reduction of the reach area to obtain higher throughput and signal quality. Subjective evaluations were carried out with real Campus users, which achieved at least reasonable results. The objective evaluations were carried out by using the Wireshark software and an Android (Nox) emulator for the capture of the packets and subsequent metrics computation: delay, delay variation, packet loss and handover time, which were within the limits established in the literature. Also, through a computer with pfSense operating system, it was possible to manage data band control and Internet access, as well as to determine the bandwidth consumption used by the applications. The obtained results show that the feasibility of using VoIP applications in mobile devices depends on the adjustment of the infrastructure of the wireless network, and attention must be paid to the distance between the access points, obstacles, overlapping channels and signal strength. Although the uses are dissatisfied with the Campus Internet services, in order to meet their desire in using mobile VoIP application on Campus, it should be reserved 500 kbps to meet at least 35 simultaneous users, considering that it would be ideal to increase the data link and not to aggravate the current services quality even more.