Voice Over IP (VoIP) telephony is becoming widespread, and is often integrated into computer networks. Because of his, it is likely that malicious software will threaten VoIP systems the same way traditional computer systems have been attacked by viruses, worms, and other automated agents. While most users have become familiar with email spam and viruses in email attachments, spam and malicious traffic over telephony currently is a relatively unknown threat. VoIP networks are a challenge to secure against such malware as much of the network intelligence is focused on the edge devices and access environment. A novel security architecture is being developed which improves the security of a large VoIP network with many inexperienced users, such as non-IT office workers or telecommunication service customers. The new architecture establishes interaction between the VoIP backend and the end users, thus providing information about ongoing and unknown attacks to all users. An evaluation of the effectiveness and performance of different implementations of this architecture is done using virtual machines and network simulation software to emulate vulnerable clients and servers through providing apparent attack vectors.

隨著網路技術的發展，目前網路電話(VoIP)已有逐漸取代傳統電話的趨勢。尤其能夠允許多人同時在線上進行會談是其最大的優勢之一。但在多人參與網路會談時，因為聲音在空間中傳遞或反射等因素，使得由喇叭發出的聲音再次被麥克風收回，造成回音的產生。會談中只要有一位使用者的裝置發生回音時，回音訊號就會在與會者之間擴散，使得所有使用者均會受到影響，進而嚴重影響網路通話的進行。此狀況在參與會談人數越多時，發生機率越高，且對通話品質影響越嚴重。 傳統電話在一對一通話時，通常使用遠端回音消除機制(Near End Echo Canceller)，由接收端在接收聲音後先暫存在記憶體中再播放，再將麥克風擷取的聲音與事先暫存的訊號反向後混合，以抵銷回音。網路會談的環境下，由於沒有標準的聽筒設備，使得回音發生的時間難以預估。且多人參與的網路會談中，由於收聽者所聽到的聲音可能混合多個使用者說話的聲音與回音，使得回音訊號難以偵測。另外，由於網路傳輸的特性，回音訊號到達的時間與順序都難以預估，這使得回音消除機制在多人網路回談中經常失效。 本研究提出藉由語音動態偵測(Voice Activity Detection-VAD)的方式分辨回音訊號，藉由本研究所提出的語音能量VAD判定機制，能夠有效區別正常語音與回音的差異，即可有效的消除回音，同時發揮靜音抑制（Slience Suppression）的效果，阻擋不含語音內容的封包，降低網路頻寬耗用。本研究以自行開發的VoIP軟體進行實地測試實驗，實驗中顯示，我們的方法能消除85%以上的回音。 / With the prosperous development of Internet technology, traditional phone service is being replaced gradually byVoice-over-IP (VoIP) technology. One of the critical problem that is yet to be improved is the echo problem. Due to the difference in working environment, conventional echo cancellation technology may not work well on VoIP system. The echo problem is becoming more critical as the number of participants in a talk session increases. As long as one user fails to depress echos, every other participant in the conference will be infected. The more participant, the higher probability of echo infection. We propose an energy based Voice Activity Detection (VAD) mechnism that effectively differentiate echo from speech signal. Our VAD algrouthm records a user’s speech volume, and based on this information to determine whether the frame is echo or not. By applying this mechnism to network conference, we can filter out echo frames and suppress slience at same time to save bandwidth consumption. We experimented on a self-developed VoIP software platform, the experiment result shows that our method can eliminate more than 85% of the echo.

Approved for public release, distribution is unlimited / Converging voice and data networks has the potential to save money and is the main reason Voice over Internet Protocol (VoIP) is quickly becoming mainstream in corporate America. The potential VoIP offers to more efficiently utilize the limited connectivity available to ships at sea makes it an attractive option for the Navy. This thesis investigates the usefulness of VoIP for the communications needs of a unit level ship. This investigation begins with a review of what VoIP is and then examines the ship to shore connectivity for a typical unit level ship. An OMNeT++ model was developed and used to examine the issues that affect implementing VoIP over this type of link and the results are presented. / Lieutenant Commander, United States Navy

En este trabajo se propone y desarrolla una topología en k-hipercubos que resuelve los principales inconvenientes asociados a la topología en hipercubo convencional. Los resultados obtenidos son muy prometedores, con aplicaciones tanto en el campo de la voz sobre IP, como en muchos otros campos que precisen de un intercambio de información muchos a muchos. Sobre la topología propuesta se define el protocolo Darkcube, que es una propuesta de protocolo totalmente distribuido basado en el concepto de darknet, posibilitando la realización de conversaciones muchos a muchos incluyendo audio, vídeo, texto y datos de geoposicionamiento, entre otros. También se propone un método de codificación de coordenadas de geoposicionamiento que resulta especialmente eficiente en el aprovechamiento del ancho de banda sobrante en las comunicaciones muchos a muchos que proporciona Darkcube. Durante el desarrollo de este trabajo, se ha implementado el simulador DarkcubeEmu; herramienta que posibilita la obtención de resultados relevantes en términos de la calidad de la comunicación. Finalmente, utilizando como base el protocolo Darkcube, se propone un protocolo de seguridad que traslada un esquema de infraestructura de clave pública a un protocolo totalmente distribuido, como es Darkcube; garantizando, de esta forma, la confidencialidad en las comunicaciones y la legitimidad de la identidad asociada a cada uno de sus miembros.

<p>Mobile telephony technology like GSM made portable telephony a possibility. The arising and development of the Internet made a revolutionary change to communication and interchange of information. Bluetooth wireless technology revolutionizes personal connectivity by providing freedom from wired connections. Combining these technologies together brings the concept of Mobile@Home of Ericsson. </p><p>Mobile@Home is a fixed-mobile convergence concept using the fixed network to carry present and future mobile services (e.g. voice, video, mail and Internet access) all the way to the home or office. By combining the high bandwidth of the fixed access network with the wireless technology of Bluetooth, Mobile@Home makes it possible to deliver high bandwidth to the mobile phone. Mobile@Home requires a Bluetooth enabled mobile phone and a Bluetooth enabled HBS (Home Base Station), placed at the home or office. By means of fast IP access (ADSL, cable modem etc.) the HBS connects into the standard mobile core network through a HBSC (Home Base Station Controller). </p><p>The purpose of this thesis is the generation of simulated traffic between the HBS and HBSC and to analyze its behavior. This primary involves generation of signaling through an internal protocol, provided by Ericsson, for management and call control, and generation of GSM EFR (Enhanced Full Rate) voice streams over the RTP (Real Time Protocol) protocol. The simulation will consist of both the HBS and MS (Mobile Station). A set of HBS: s with attached MS will call one another through the HBSC. In this assignment only the GSM signaling will be considered because of time and scope limitations. The goal is to validate the RTP traffic generated towards the HBSC. Parameters like packet loss, packet delay and erroneous packets will be analyzed.</p>

<p>This thesis is an experimental study to evaluate the performance of different routing protocols in commonly deployed scenarios. This study mainly focuses on how much time each protocol consumes while recovering from a link-loss. It provides a guide line for the best routing solutions for ISPs, individual organizations or other types of providers which are engaged in providing reliable real-time communications to their subscribers. Such communications may include vehicle trafficking data, online TV programs (IPTV), voice over IP telephony (VoIP), weather forecasts, tracking systems and many other services which totally depend upon the reliability of real-time data streams, where any major loss in received data may bring significant negative results in the integrity of the entire application.</p><p>This work experimentally observes and tracks the loss of UDP packets when changes in the network topology occur. In order to make this observation in real network topologies, a custom-designed software tool has been developed. The tool is capable of delivering enough resources to a tester in evaluating the performance of routing protocols. All the test results derived from the software tool are statistically evaluated and on the basis of the outcome a better proposition can be provided to network administrators which face inconsistent topological issues.</p>

<p>Statistical distributions are used to model behaviour of real VoIP traffic. We investigate call holding and inter-arrival times as well as speech patterns. The consequences of using an inappropriate model for network dimensioning are briefly discussed. Visual examination is used to compare well known distributions with empirical data. Our results support the general opinion that the Exponential distribution is not appropriate for modelling call holding time. We find that the distribution of talkspurt periods is well modelled by the Lognormal distribution and the silence periods by the generalized Pareto distribution. It is also observed that the call inter-arrival times tend to follow a heavy tailed distribution.</p>

This thesis is an experimental study to evaluate the performance of different routing protocols in commonly deployed scenarios. This study mainly focuses on how much time each protocol consumes while recovering from a link-loss. It provides a guide line for the best routing solutions for ISPs, individual organizations or other types of providers which are engaged in providing reliable real-time communications to their subscribers. Such communications may include vehicle trafficking data, online TV programs (IPTV), voice over IP telephony (VoIP), weather forecasts, tracking systems and many other services which totally depend upon the reliability of real-time data streams, where any major loss in received data may bring significant negative results in the integrity of the entire application. This work experimentally observes and tracks the loss of UDP packets when changes in the network topology occur. In order to make this observation in real network topologies, a custom-designed software tool has been developed. The tool is capable of delivering enough resources to a tester in evaluating the performance of routing protocols. All the test results derived from the software tool are statistically evaluated and on the basis of the outcome a better proposition can be provided to network administrators which face inconsistent topological issues.

Due to their massive user base and request load, large-scale Internet applications have mainly focused on goals such as performance and scalability. As a result, many of these applications rely on weaker but more efficient and simpler authentication mechanisms. However, as recent incidents have demonstrated, powerful adversaries are exploiting the weaknesses in such mechanisms. While more robust authentication mechanisms exist, most of them fail to address the scale and security needs of these large-scale systems. In this dissertation we demonstrate that by taking into account the specific requirements and threat model of large-scale Internet applications, we can design authentication protocols for such applications that are not only more robust but also have low impact on performance, scalability and existing infrastructure. In particular, we show that there is no inherent conflict between stronger authentication and other system goals. For this purpose, we have designed, implemented and experimentally evaluated three robust authentication protocols: Proxychain, for SIP-based VoIP authentication; One-Time Cookies (OTC), for Web session authentication; and Direct Validation of SSL/TLS Certificates (DVCert), for server-side SSL/TLS authentication. These protocols not only offer better security guarantees, but they also have low performance overheads and do not require additional infrastructure. In so doing, we provide robust and practical authentication mechanisms that can improve the overall security of large-scale VoIP and Web applications.

pyAXL ist eine Programmierschnittstelle (API) zur Steuerung des "Cisco Call Manager", eine enterprise-VoIP-Verwaltung. Der Vortrag zeigt, wie pyAXL aufgebaut ist. An ein paar Beispielen wird die Verwendung von pyAXL demonstriert.