Thesis (B.A.)--Haverford College, Dept. of History, 2008. / Includes bibliographical references.

At beginning of this paper, it will introduce the SIP protocol major used on the VoIP service at present time. Introduce the development and the application of SIP. And then explain the operation of the SIP protocol by an example. Explain how a SIP client establish the communication with another one by the SIP proxy. After understanding the operation of the SIP, explain why the SIP client behind NAT will meet the fail and introduce the solutions that have developed to conquer the fail. Briefly describe each of these solutions and compare all of them. Next is to describe how the solution of statteful SIP proxy used to avoid the fail of NAT changes the behavior between the SIP clients from end-to-end to master-to-slave. The change of behavior between the SIP clients may affect the quality of the communication in some conditions. To improve this, this paper brings out an idea and implements it. Finally, test the quality of the communication, catch the packet carrying the SIP messages and packet of RTP, and watch the content of SIP messages and the routing of the RTP packet.

With the bandwidth growing and voice coding technology enhancing, VoIP(Voice over IP)becomes a very popular application in recent years. Although VoIP has good voice quality and huge price advantage, it can't replace PSTN(Public Switched Telephone Network) completely in the next few years. We therefore hope to be able to solve the connectivity issue between PSTN and VoIP. In this thesis, I design and implement a VoIP Gateway using Infineon EASY 5120 platform which has a powerful CPU with network processor, ADM5120, and has a DSP (Digital Signal Processor), Vinetic-2CPE PEB3332, to handle digital and analog voice signals. The operation method of the gateway is that PSTN analog signals are passed to Vinetic-2CPE from FXO (Foreign eXchange Office) port, and they are transferred into digital signals to be packetized to transit into Internet to another VoIP Client. My VoIP Gateway can solve the interconnection issues between PSTN and VoIP. Furthermore, in order to solve NAT traversal problem of SIP protocol, we propose and implement an integration approach which uses STUN (Simple Traversal of UDP through NAT) to cooperate with RTP Relay server to traverse various types of NAT, especial symmetric NAT. My VoIP Gateway not only can traverse any type of NAT but also can perform self-configuration according to STUN test report in various network environments.

With the development and improvement of network bandwidth, Voice over Internet Protocol (VoIP) technology begins to bloom. VoIP gradually replaces voice over telecom networks because of its low cost. For instance, VoIP technology divides voice data into different packets and transmits over Internet Protocol, so that it does not require setting up a particular path as the telecom networks does. Although there is much VoIP software available in the market, most of them need pre-installation before using. Our web-based VoIP technology allows users to take advantage of the service with either home computers or public computers. Users only need to log on to our website, then our system will provide users the client program via Java Applet with automatic installation. Users only need to close the browser to terminate the service; the program data will not be left on the computer. In order to provide users to use our service in anytime and anywhere, we need to recognize Network Address Translation (NAT) as one of our biggest barriers. Including Session Initiation Protocol (SIP), instant message and voice data packet could be blocked by NAT. So, we propose a complete solution of NAT traversal. Our system, SIP Communicator, provides a communication platform to users by supporting instant message, VoIP and session transferring. Users can log on to our system via any computers that are JRE supported to communicate with other clients.

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.

Nathaniel Ndazana Nakasa [1937 - 1965] was a South African journalist who reported for llanga Lase [Natal] in 1956 and 1957, for Drum magazine from 1958 to 1964 and wrote a column for the Saturday edition of the Rand Daily Mail in I9B4. He also founded the literary journal The Classic in 1963. This essay is the first extended treatment of Nakasa's writing, and views his journalism as part of his own 'autobiography1. As such, his writing reflects his crisis of identity, which resulted from his endeavour to sustain his vision of a broad South African humanism in the face of the apartheid policies of the Government in the late 1950s and early 1960s. Nakasa's death by suicide in New York in 1965 signalled the tragic end of his search for equality and justice. Nakasa had been labelled 'the black face behind the white mask' and is criticised, particularly by adherents of Black Consciousness, for his evident faith in the tenets of liberalism. This essay attempts to locate Nakasa in the context of opposition by those of humanist inclinations to apartheid in the fifties and sixties and to view sympathetically his commitment to justice and compassion : values which remain relevant and valid in our search for a better society in South Africa. The investigation proceeds by an analysis of his journalism as both the record of the times and, more subjectively, the projection and expression of his own crisis-ridden personality. An introduction is followed by two sections on his writing, the first dealing with his articles on Drum, the second with his sketches on the Rand Daily Mail. A brief conclusion argues for the continuing interest of Nakasa's writing. / Thesis (M.A.)-University of Natal, Durban, 1990.

Práce se zabývá problematikou firewallů a jejich vývojem. Rozdělena je na 4 hlavní části. V první (teoretické) části jsou rozpracovány některé důležité koncepty síťového provozu ? především pak ty s dopadem na zajištění bezpečného přístupu k síti. Rovněž jsou roztříděny firewally a zmíněny konkrétní dopady (přínosy) jejich nasazení. Na konci kapitoly si ještě rozebereme Standardy, které v této oblasti existují (především pak TCSEC a CC). Další část je věnována samotnému vývoji fw a především pak současným trendům posledních 5 let. Rozebrány jsou rovněž kromě jejich dopadů na bezpečnost i externality, které s nimi byly spojeny (změna chování uživatelů, rychlost, propustnost, atp.). Konec kapitoly obsahuje ještě výhledy předpokládaného vývoje v této oblasti. V praktické části se podíváme na některé nejdůležitější způsoby ochrany. V části terminace otestujeme všechny publikované metody terminace a možnosti ochrany. Nakonec v části odchozí komunikace provedeme test všech dostupných metod průniku ven ze sítě ? vyzkoušíme tedy všechny publikované techniky, jak dostat data nepozorovaně ven z počítače a zjistíme, jaké jsou možnosti obrany u konkrétní techniky. V poslední části se podíváme na některá důležitá či zajímavá praktická řešení z oblasti bezpečnosti sítí. Především pak na zabezpečení sítě Vysoké Školy Ekonomické v Praze, Checkpoint Firewall-1, a nakonec jeho nasazení na VŠE.

This thesis project concerns NAT traversal techniques and their application to P2P networking with regard to MediaSense platform. Since MediaSense open source platform, developed by Mid Sweden University, utilizes the benefits of P2P networking, it also suffers from the drawbacks provided by NAT. The issue of NAT traversal is not trivial due to the fact that the behavior of NAT devices is not standardized and vendors are free to provide their own implementations. The common knowledge is, that at least four main types of NATs exist, differing in the filtering and mapping algorithms employed. NAT traversal techniques vary accordingly. No single technique can handle all the cases. Most of the techniques can handle up to three main types of NAT. The last type is usually used in large corporate networks and is called the Symmetric NAT. The most viable, and basically the only available technique for its traversal, is data relaying. This thesis builds a NAT traversal module for the MediaSense platform. The main purpose of this module is to provide seamless NAT traversal capabilities to the platform. The module does this in several steps: UPnP enabled device discovery, NAT type determination and data relaying via the proxy. Firstly the module attempts to discover the presence of a UPnP enabled Internet Gateway Device on the network. If such a device is present on the network, a port mapping can be created, making the node located behind NAT accessible from the public Internet. If a UPnP enabled device was not found, the module will try to determine the type of NAT used. Based on the type of NAT used, the module can transit to either the proxy mode or request assistance of the STUN server to keep the created mapping alive. The resulting chapters provide the reader with the output produced by each step, conclusions the author has made while working on this project and some general ideas on future work within the subject.

Video streaming over the Internet can be very difficult under the traditional client-server model. Peer-to-peer (P2P) systems, in which each participating peer contributes its upload bandwidth to other peers while it downloads data, have been successful in file-sharing applications, and they appear to be promising in delivering video contents, too. However, the existence of network address translation (NAT) is always considered as a challenge to peer-to-peer systems. NAT has been a practical solution to the Internet Protocol version 4 (IPv4) address exhaustion problem, as it reduces the usage of IP addresses by allowing multiple private hosts to share a single public IP address, but NAT can degrade the performance of a peer-to-peer system as it limits the direction of connectivity. Measurement studies show that a considerable fraction of peer-to-peer video streaming system users are behind NAT devices, and that their uplink bandwidth is not well utilized, but the literature lacks a quantitative analysis of the impact of NAT on the performance of P2P video streaming systems. In this thesis, an extensible analytical model is built to capture the performance for P2P live streaming systems with a certain percentage of users behind NAT and cannot be reached by NAT traversal techniques, the correctness of which is verified by software simulation. A simple mechanism is proposed in this thesis, which is able to effectively improve the system performance and fairness by counteracting the negative impact of NAT, and it can also be used to reduce the usage of server bandwidth. / Graduate

Thesis (M.A.)--Columbia University. / Bibliography: p. 111-125.