Submitted by Fabio Sobreira Campos da Costa (fabio.sobreira@ufpe.br) on 2017-03-02T13:44:12Z No. of bitstreams: 2 license_rdf: 1232 bytes, checksum: 66e71c371cc565284e70f40736c94386 (MD5) Dissert_Alexandre_Santos_CIN.pdf: 2195136 bytes, checksum: bc5b11111393e727688fe5580cf6f0b4 (MD5) / Made available in DSpace on 2017-03-02T13:44:12Z (GMT). No. of bitstreams: 2 license_rdf: 1232 bytes, checksum: 66e71c371cc565284e70f40736c94386 (MD5) Dissert_Alexandre_Santos_CIN.pdf: 2195136 bytes, checksum: bc5b11111393e727688fe5580cf6f0b4 (MD5) Previous issue date: 2016-07-28 / Com o crescimento da adoção de comunicação de voz em redes de dados, tecnologia conhecida como Voice Over Internet Protocol (VoIP) , tornaram-se evidentes, tanto vantagens obtidas, quanto problemáticas decorrentes do contraste entre a natureza das redes de dados e as necessidades do tráfego VoIP. A internet não possui, por padrão, formas inteligentes de proteção e priorização do tráfego, levando à utilização de tecnologias externas para suprir estas necessidades. Todavia, a utilização destas tecnologias tem tornado-se cada vez mais complexa em virtude de um fenômeno conhecido como ossificação das redes IP, conforme Mckeown et al. (2008). Em busca de solucionar os problemas enfrentados pelas redes atuais e prover maior flexibilidade e capacidades inovadores, surge a Software Defined Network (SDN), tecnologia que visa trazer uma série de vantagens para as redes de dados atuais através de sua arquitetura. Neste trabalho é proposta a VoIPath, uma solução baseada em SDN que implementa uma técnica de seleção de caminhos com base em atributos específicos, relevantes para o tráfego VoIP. Para processar estes atributos e identificar os melhores caminhos para o tráfego VoIP é empregado um algoritmo criado com programação linear, onde um volume de tráfego VoIP é dividido entre diferentes caminhos, com base em opções de execução da solução. Para controlar a propagação do tráfego é empregada uma tecnologia baseada em SDN a Virtual Tenant Networks (VTN), através desta o tráfego é conduzido por uma rede lógica, chamada de tenant. O mapeamento destas redes lógicas na rede SDN física garante a passagem do tráfego pelos caminhos selecionados. Por fim é realizada uma série de experimentos para demonstrar a efetividade da solução, bem como melhorias na qualidade do áudio, calculadas com base em um modelo de avaliação de qualidade de tráfego VoIP. / With increasing adoption of voice communications through data networks, technology known as VoIP, both advantages and issues have become apparent. The second group caused by the contrast between the nature of data networks and the needs of VoIP traffic. By default, data networks do not have smart structures to protect and prioritize traffic. In order to achieve these characteristics external technologies are deployed, but the utilization of such strategy has become complex due to a phenomenon described as network ossification (MCKEOWN et al., 2008). In order to provide solutions to the problems of the actual data networks architecture and provide greater flexibility, comes in scene the SDN technology. The SDN architecture unleashes a new horizon of possibilities to the networks. This work proposes the VoIPath solution, which consists of a implementation of a SDN based path selection technique, oriented by specific relevant attributes to the VoIP traffic. A algorithm created with linear programming is employed to compute these attributes and identify the best paths to the VoIP traffic, which is divided through the network, in according to execution parameters. In order to control the traffic propagation is employed the VTN technology, which consists on an SDN based solution. Through VTN technology the VoIP traffic is forwarded through an logical network called tenant. The mapping of the tenant over the physical network ensures that the selected paths are employed. To ensure the solution’s effectiveness a series of experiments are performed and the impact of its’s execution on the audio is calculated with the usage of an VoIP quality measurement evaluation model.

When using Voice over IP (VoIP) in Wireless Mesh Networks the overhead induced by the IEEE 802.11 PHY and MAC layer accounts for more than 80% of the channel utilization time, while the actual payload only uses 20% of the time. As a consequence, the Voice over IP capacity is very low. To increase the channel utilization efficiency and the capacity several IP packets can be aggregated in one large packet and transmitted at once. This paper presents a new hop-by-hop IP packet aggregation scheme for Wireless Mesh Networks. The size of the aggregation packets is a very important performance factor. Too small packets yield poor aggregation efficiency; too large packets are likely to get dropped when the channel quality is poor. Two novel distributed protocols for calculation of the optimum respectively maximum packet size are described. The first protocol assesses network load by counting the arrival rate of routing protocol probe messages and constantly measuring the signal-to-noise ratio of the channel. Thereby the optimum packet size of the current channel condition can be calculated. The second protocol, which is a simplified version of the first one, measures the signal-to-noise ratio and calculates the maximum packet size. The latter method is implemented in the ns-2 network simulator. Performance measurements with no aggregation, a fixed maximum packet size and an adaptive maximum packet size are conducted in two different topologies. Simulation results show that packet aggregation can more than double the number of supported VoIP calls in a Wireless Mesh Network. Adaptively determining the maximum packet size is especially useful when the nodes have different distances or the channel quality is very poor. In that case, adaptive aggregation supports twice as many VoIP calls as fixed maximum packet size aggregation.

Telephony plays a fundamental role in our society. It enables remote parties to interact and express themselves over great distances. The telephone as a means of communicating has become part of every day life. Organisations and industry are now looking at Voice over IP (VoIP) technologies. They want to take advantage of new and previously unavailable voice services. Various interested parties are seeking to leverage the emerging VoIP technology for more flexible and efficient communication between staff, clients and partners. <o>VoIP is a recent innovation enabled by Next Generation Network (NGN). It provides and enables means of communication over a digital network, specifically the Internet. VoIP is gaining wide spread adoption and will ultimately replace traditional telephony. The result of this trend is a ubiquitous, global and digital communication infrastructure. VoIP, however, still faces many challenges. It is not yet as reliable and dependable as the current Public Switched Telephone Network (PSTN). The employed communication protocols are immature with many security flaws and weaknesses. Session Initiation Protocol (SIP), a popular VoIP protocol does not sufficiently protect a users privacy. A user’s information is neither encrypted nor secured when calling a remote party. There is a lack of control over the information included in the SIP messages. Our specific concern is that private and sensitive information is exchanged over the public internet. This dissertation concerns itself with the communication path chosen by SIP when establishing a session with a remote party. In SIP, VoIP calls are established over unknown and untrusted intermediaries to reach the desired party. We analyse the SIP headers to determine the information leakage at each chosen intermediary. Our concerns for possible breach of privacy when using SIP were confirmed by the findings. A user’s privacy can be compromised through the extraction of explicit private details reflected in SIP headers. It is further possible to profile the user and determine communication habits from implicit time, location and device information. Our research proposes enhancements to SIP. Each intermediary must digitally sign over the SIP headers ensuring the communication path was not be altered. These signatures are added sequentially creating a chain of certified intermediaries. Our enhancements to SIP do not seek to encrypt the headers, but to use these intermediary signatures to reduce the risk of information leakage. We created a model of our proposed enhancements for attaching signatures at each intermediary. The model also provides a means of identifying unknown or malicious intermediaries prior to establishing a SIP session. Finally, the model was specified in Z notation. The Z specification language was well suited to accurately and precisely represent our model. This formal notation was adopted to specify the types, states and model behaviour. The specification was validated using the Z type-checker ZTC. Copyright / Dissertation (MSc)--University of Pretoria, 2010. / Computer Science / unrestricted

This master thesis deals with the design and implementation of an application for analyzing Voice over IP quality using NetFlow. In the beginning, there is summarized basic information about VoIP technology and NetFlow - its principles, the most used protocols, factors that have influence on call quality and call quality rating methods. Later there is presented proposal of application and then described its implementation. The created application was tested on samples, which simulate calls in network with delays and packet-loss. Within testing was made the comparison with commercial application and the results are discussed.

Workshop Mensch-Computer-Vernetzung Stand und Perspektiven des Einsatzes einer "Voice over IP"-Lösung für die Telefonie am URZ der TU Chemnitz auf Grundlage der VOCAL-Server-Suite

Voice over the Internet Protocol (VoIP) is one of the latest and most successful Internet services. It takes advantage of Wireless Local Area Networks (WLANs) and broadband connections to provide high quality and low cost telephony over the Internet or an intranet. This project exploits features of VoIP to create a communication scenario where various conversations can be held at the same time, and each of these conversations can be located at a virtual location in space. The report includes theoretical analysis of psychoacoustic parameters and their experimental implementation together with the design of a spatial audio module for the Session Initiation Protocol (SIP) User Agent “minisip”. Besides the 3D sound environment this project introduces multitasking as an integrative feature for “minisip”, gathering various sound inputs connected by a SIP session to the “minisip” interface, and combining them altogether into a unique output. This later feature is achieved with the use of resampling as a core technology. The effects of traffic increment to and from the user due to the support of multiple streams are also introduced. / Röst över Internet Protocol (VoIP) är en av de senaste och mest framgångsrika Internettjänsterna. Det utnyttjar Trådlösa Nätverk och bredband för att erbjuda högkvalitativ och billig telefonering över Internet eller ett Intranät. Det här projektet använder sig av VoIP för att skapa ett kommunikationsscenario där flera olika konversationer kan hållas samtidigt och där varje konversation kan placeras på en virtuell plats i rymden. Rapporten innehåller en teoretisk analys av psykoakustiska parametrar och deras experimentella genomförande tillsammans med design av en 3D ljud modul för Session Initiation Protocol (SIP) User Agent ”minisip”. Förutom ljudmiljön i 3D introducerar projektet multitasking som en integrerbar del av ”minisip”. Alla tänkbara ljudkällor baserade på SIP förbindelser samlas med ”minisip” interfacet och kombineras till en enda utsignal. Detta uppnås med hjälp av resampling som kärnteknologi. Effekterna av att mer trafik når användaren på grund av stödet av multiple källor introduceras också.

The very nature of mobile phones makes them ideal vehicles to study both individuals and organizations: people habitually carry a mobile phone with them and use it as a medium for much of their communication. The information available from today's phones includes the user's location, people nearby, and communication (call and SMS logs), as well as application usage and phone status (idle, charging, and so on). The main goal of this project is to combine some of the new technologies of voice over IP (VoIP) with context awareness services for mobile users and create a demonstrator for a typical routine of a student in Kista. We used context awareness together with the SIP Express Router to make a system more intelligent for the user. In this thesis the definition of CPL scripts and how they could exploit context information to provide SIP service that would be useful to a student were examined. A simple test was conducted to measure the overhead of using context awareness by the SIP proxy when processing CPL scripts. / Mobila telefoner gör dem ideala medel för att studera både individer och organisationar: personer bär ofta en mobil telefon med dem och använder den som ett medel för mycket av deras kommunikation. Informationen som är tillgänglig från dagens telefoner inkluderar användares läge, personer som är närliggande och kommunikation, såväl som applikationanvändning och telefon status. Målet av detta projekt är att kombinera som några av de nya teknologierna av röst över IP (VoIP) med kontextuppmärksamma servar för mobila användare och skapar en demonstrant för en typisk rutin av en studerande i Kista. Vi använde kontextuppmärksamma med SIP Express Router för att göra ett system mer intelligent för användare. I detta examensarbetet undersöker vi CPL skrifter och hur de skulle kunna exploatera kontext information för att ge den SIP tjänsten som är användbar till en studerande. Ett enkelt test förades för att mäta det över huvudet av att använda kontextuppmärksamma av den SIP proxyen när det arbetar med CPL skrifter.

With its enormous potential Voice over Internet Protocol is one of the latest buzzwords in information technology. Despite the numerous advantages of Voice over IP, it is a major technical challenge to achieve a similar call quality as experienced in the ordinary Public Switched Telephone Network. This thesis introduces standardized Internet protocols for Voice over IP, such as Session Initiation Protocol (SIP), Real-time Transport Protocol (RTP), in its background chapter. In order to provide better Quality of Service (QoS) Voice over IP applications should support a feedback mechanism, such as the Real-time Control Protocol (RTCP), and use accurate timing information, provided by the Network Time Protocol (NTP). Additionally this thesis considers synchronization issues in calls with two and more peers. After a rather academic overview of Voice over IP, the open source real-time application “minisip”, a SIP user agent, and its operation and structure for handling audio streams will be introduced. Minisip was extended by an implementation of NTP and RTCP to provide a test platform for this thesis. A clear conclusion is that the addition of global time helps facilitate synchronization of multiple streams from clients located any where in the network and in addition the ability to make one-way delay measurements helps SIP user agents to provide better quality audio to their users. / Röst över IP, eller Internettelefoni baserad på “Voice over Internet Protocol” (VoIP), har med sin stora potential blivit ett av de senaste modeorden inom informationsteknologin. Vid sedan av ett antal fördelar med VoIP så innebär det en stor teknisk utmaning att uppnå en likadan samtalskvalitet som i det vanliga, fasta, telenätet. I den här uppsatsen beskrivs hur tjänstevalitet för VoIP kan förbättras genom att noggrant tidssynkronisera de (två eller flera) klienter som deltar i ett telefonsamtal. För detta krävs dels en återkopplingsmekanism, såsom “Real-time Control Protocol” (RTCP), samt en gemensam tidsuppfattning i de inblandade klienterna, vilket kan uppnås med hjälp av “Network Time Protocol” (NTP). Dessa protokoll, liksom de övriga Internet-standarder som VoIP baseras på (såsom “Session Initiation Protocol” (SIP) och “Real-time Transport Protocol” (RTP), beskrivs inledningsvis i uppsatsen. För studien har en SIP-klient baserad på öppen källkod använts (“Minisip”), och utökats med NTP och RCTP funktionalitet för att testa den föreslagna förbättringen av VoIP. En tydlig slutsats är att kännedom om en “global tid” möjliggör synkronisering av multipla ljudströmmar från klienter som befinner sig på olika nätverk. Möjligheten att mäta paketfördröjningen (envägs) bidrar också till en förbättrad ljudkvalitet.

The focus of this thesis is on transferring data between mobile devices. It analyzes the resource consumption of using the Wireless LAN interface in some situations, and proposes and evaluates some new applications involving the transfer of voice and data files. Measurements show the high consumption of battery power due to the operation of a wireless network interface, even if it is not “actively” used to transfer data, while the memory consumption of the running applications is very limited. This thesis includes also an application to transfer files between two or more Pocket PCs: it is proposed as an addition to the VoIP application Minisip. Finally I have developed an application in order to explore the possibility to make a voice call by transferring real-time encoded text using UDP and/or RTP streams. This could be used together with speech-to-text and text-to-speech conversion applications at the end points to allow a “voice conversation” even on wireless links with very limited capacity, whereas a standard VoIP conversation would not be affordable. / Fokus i detta examensarbete ligger på överföring av data mellan mobila enheter. Det analyserar resursutnyttjandet vid användande av Wireless LAN-anslutning i några situationer, samt brukar analysen för att föreslå och utvärdera ett antal nya applikationer för överföring av röstsamtal och filer. Mätningar visar den höga konsumtion av batterikraft som anslutning till ett trådlöst nätverk gränsnitt, även då det inte används "aktivt" för dataöverföring, medan de applikationer som använts här behöver bara mycket begränsad minneskapacitet. I detta examensarbete ingår även ett program för överföring av filer mellan två eller flera mobila apparater, föreslaget att ingå i Minisip, en VoIP applikation. Dessutom föreslås och utvärderas en applikation som undersöker möjligheterna att vid röstsamtal överföra realtidskodad text via RTP och/eller UDP stream. Använd tillsammans med en röst-till-text- och text-till-röstomvandling i ändpunkterna möjliggör denna genomförande av röstsamtal även då mycket begränsade trådlösa anslutningar nyttjas. / <p>Exchange student from Pisa.</p>

Voice over IP (VoIP) can be defined as the ability to make phone calls and to send faxes (i.e., to do everything we can do today with the Public Switched Telephone Network, PSTN) over IP−based data networks with a suitable quality of service and potentially a superior cost/benefit ratio. There is a desire to provide (VoIP) with the suitable security without effecting the performance of this technology. This becomes even more important when VoIP utilizes wireless technologies as the data networks (such as Wireless Local Area Networks, WLAN), given the bandwidth and other constraints of wireless environments, and the data processing costs of the security mechanisms. As for many other (secure) applications, we should consider the security in Mobile VoIP as a chain, where every link, from the secure establishment to the secure termination of a call, must be secure in order to maintain the security of the entire process. This document presents a solution to these issues, providing a secure model for Mobile VoIP that minimizes the processing costs and the bandwidth consumption. This is mainly achieved by making use of high− throughput, low packet expansion security protocols (such as the Secure Real−Time Protocol, SRTP); and high−speed encryption algorithms (such as the Advanced Encryption Standard, AES). In the thesis I describe in detail the problem and its alternative solutions. I also describe in detail the selected solution and the protocols and mechanisms this solution utilizes, such as the Transport Layer Security (TLS) for securing the Session Initiation Protocol (SIP), the Real−Time Protocol (RTP) profile Secure Real−Time Protocol (SRTP) for securing the media data transport , and the Multimedia Internet KEYing (MIKEY) as the key−management protocol. Moreover, an implementation of SRTP, called MINIsrtp, is also provided. The oral presentation will provide an overview of these topics, with an in depth examination of those parts which were the most significant or unexpectedly difficult. Regarding my implementation, evaluation, and testing of the model, this project in mainly focused on the security for the media stream (SRTP). However, thorough theoretical work has also been performed and will be presented, which includes other aspects, such as the establishment and termination of the call (using SIP) and the key−management protocol (MIKEY). / Voice over IP (VoIP) kan defineras som förmågan att göra ett telefonsamtal och att skicka fax (eller att göraallting som man idag kan göra över det publika telefonnätet) över ett IP−baserat nätverk med en passande kvalitet och till lägre kostnad, alternativt större nytta. VoIP måste tillhandahållas med nödvändiga säkerhetstjänster utan att teknikens prestanta påverkas. Detta blir allt viktigare när VoIP används över trådlösa länktekniker (såsom trådlösa lokala nätverk, WLAN), givet dessa länkars begränsade bandbredd och den bearbetningkraft som krävs för att exekvera säkerhetsmekanismerna. Vi måste tänka på VoIPs säkerhet likt en kedja där inte någon länk, från säker uppkoppling till säker nedkoppling, får fallera för att erhålla en säker process. I detta dokument presenteras en lösning på detta problem och innefattar en säker modell för Mobile VoIP som minimerar bearbetningskostnaderna och bandbreddsutnyttjandet. Detta erhålls huvudsakligen genom utnyttjande av säkerhetsprotokoll med hög genomströmning och låg paketexpansion, såsom "Secure Real− time Protocol" (SRTP), och av krypteringsprotokoll med hög hastighet, såsom "Advanced Encryption Standard" (AES). I detta dokument beskriver jag problemet och dess alternativa lösningar. Jag beskriver också den valda lösningen och dess protokoll och mekanismer mer detaljerat, till exempel "Transport Layer Security" (TLS) för att säkra "Session Initiation Protocol" (SIP), SRTP för att skydda transporten av data och "Multimedia Internet KEYing" (MIKEY) för nyckelhantering. En implementation av SRTP, kallad MINIsrtp, finns också beskriven. Beträffande praktiskt arbete och tester av lösningsmodellen har detta projekt fokuserats på skyddandet av datatransporten (SRTP), dess implementation och prestanda. Emellertid har en grundlig teoretisk undersökning genomförts, vilken innefattar andra aspekter såsom telefonsamtalets uppkoppling och nedkoppling (med hjälp av SIP) och valet av passande nyckelhanteringsprotokoll (MIKEY) för att stödja SRTP.