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.

Land mobile radio (LMR) systems are used for communication by public safety and other government and commercial organizations. LMR systems offer mission-critical or even life-critical service in the day-to-day activities of such organizations. Traditionally, a variety of different LMR systems have been deployed by different organizations, leading to a lack of radio interoperability. A voice application that connects LMR systems via a packet-switched network is called an LMR Voice over IP (LMRVoIP) system and is a potential solution to the interoperability problem. LMRVoIP systems are time critical, i.e., are delay and jitter sensitive. Transmission of LMRVoIP traffic in a congested packet-switched network with no quality of service (QoS) or priority mechanisms in place could lead to high delays and extreme variations in delay, i.e., high jitter, thus resulting in poor application performance. LMRVoIP systems may also have performance issues with the use of virtual private networks (VPNs). To the best of our knowledge, there has been no prior thorough investigation of the performance of an LMRVoIP system with different queuing schemes for QoS and with the use of VPN. In this thesis, we investigate the performance of an LMRVoIP system with different queuing schemes and with the use of VPN. An experimental test bed was created to evaluate four QoS queuing schemes: first-in first-out queuing (FIFO), priority queuing (PQ), weighted fair queuing (WFQ), and class-based weighted fair queuing (CBWFQ). Quantitative results were obtained for voice application throughput, delay, jitter, and signaling overhead. Results show that, compared to a baseline case with no background traffic, LMRVoIP traffic suffers when carried over links with heavy contention from other traffic sources when FIFO queuing is used. There is significant packet loss for voice and control traffic and jitter increases. FIFO queuing provides no QoS and, therefore, should not be used for critical applications where the network may be congested. The situation can be greatly improved by using one of the other queuing schemes, PQ, WFQ, or CBWFQ, which perform almost equally well with one voice flow. Although PQ has the best overall performance, it tends to starve the background traffic. CBWFQ was found to have some performance benefits over WFQ in most cases and, thus, is a good candidate for deployment. The LMRVoIP application was also tested using a VPN, which led to a modest increase in latency and bandwidth utilization, but was found to perform well. / Master of Science

Studien behandlar och beskriver dels hur tolkar upplever att taltolka hörande personer som använder teckenspråk, dels hur hörande personer som använder teckenspråk upplever att bli taltolkade. Två fokusgrupper fick berätta om sina erfarenheter av denna företeelse; teckenspråkstolkar i en grupp och hörande teckenspråkiga i en annan grupp. Genom kvalitativ innehållsanalys identifierades fem teman med de viktigaste aspekterna i frågan. Dessa teman var: 1. Hörandes val av språk i tolkade sammanhang 2. Strategier för att hantera att bli taltolkad 3. Tolkens språkliga utmaningar 4. Ett gott samarbete är avgörande 5. CEFR – självbedömning är svårt. Genom dessa teman framkom att taltolkning av hörande är ett komplext fenomen som kräver särskild hantering. Ett väl utarbetat samarbete är en stor fördel och underlättar tolkens arbete. Det krävs också god kännedom om anledningen till att hörande ibland väljer att teckna i tolkade sammanhang. Det gör att tolken kan slappna av och koncentrera sig på tolkuppdraget. God insikt hos hörande gällande sin teckenspråkskompetens är också eftersträvansvärt. Vanan av att bli taltolkad underlättar situationen eftersom hörande då utarbetar strategier för att tåla att parallellt med den egna produktionen höra taltolkningen. / This study examines interpreters’ experiences of interpreting hearing signers into spoken language and describes the phenomenon of being the hearing person who signs in these interpreted contexts. Two focus groups were given the chance to share their stories about the situation - one group of sign language interpreters and one group of hearing signers. Through content analysis five themes were identified concerning the most important aspects. 1. Hearing persons’ choice of language in an interpreted situation 2. Strategies to manage being interpreted into spoken language 3. The interpreters’ linguistic challenges 4. Good cooperation is vital 5. CEFR- self-evaluation is hard. These themes revealed that giving voice to hearing signers is a complex phenomenon requiring special management. Well developed cooperation is an advantage and facilitates the interpreters work. Good knowledge about the reasons why hearing signers choose to sign makes it easier for the interpreters to relax and concentrate on the task. Hearing people need to have good understanding of their language proficiency as well as being accustomed to and develop strategies for managing to hear the interpreters voice over.

Background. SIP and RTP are two protocols that are widely used, and they play an important role in VoIP services. VoIP is an integral part of many communication services, e.g., Microsoft Teams, Skype, Discord, and communications over cellular networks (VoLTE and VoWiFi). Since these technologies are so widely used, a high level of security is paramount. Objectives. The aim of this study is threefold: (1) To investigate if it is possible to create a penetration testing methodology for SIP and RTP, where the target group is penetration testers with no previous knowledge of these protocols. (2) To identify previously discovered vulnerabilities and attacks. (3) Due to the lack of domain experts, a methodology of this kind will hopefully help penetration testers without prior knowledge, easing them into a new work area. Further, the aim is to increase awareness of potential vulnerabilities in such systems. Methods. Through a literature review, threat modeling, and exploratory penetration testing on three different testbeds, several vulnerabilities and attacks were identified and validated. From the results, a methodology was compiled. For evaluation purposes, it was evaluated by a third party, who tested it on a testbed and gave feedback. Results. The results from our research show that SIP and RTP are susceptible to a wide array of different attacks even to this day. From our literature study, it was determined that most of these attacks have been known for a long time. Using exploratory penetration testing, we managed to verify most of these attacks on three different systems. Additionally, we discovered a few novel attacks that we did not find in previous research. Conclusions. Our literature study suggests that SIP and RTP based systems are relatively susceptible to multiple attacks. Something we also validated during the exploratory testing phase. We successfully executed multiple existing attacks and some new attacks on three different testbeds. The methodology received mostly positive feedback. The results show that many of the participants appreciated the simplicity and concrete model of the methodology. Due to the low number of participants in the evaluation, an improvement to the study and results would be to increase the population and also have multiple novice penetration testers test several different systems. An increase in the number of testbeds would also further support the results and help generalize the methodology. / Bakgrund. SIP och RTP  är två protokoll som är vitt använda och spelar en väldigt viktig roll i VoIP-tjänster. VoIP utgör en viktig del i många kommunikationstjänster, t.ex. Microsoft Teams, Skype och Discord, men även i kommunikation över mobilnätet (VoLTE och VoWiFi). Eftersom dessa teknologier används i så stor utsträckning, är säkerhet av största vikt. Syfte. Syftet med denna studie är trefaldig: (1) Undersöka om det är möjligt att utforma en penetration testningsmetod för SIP och RTP, för en målgrupp av penetrationstestare utan förkunskaper kring dessa protokoll. (2) Att identifiera sårbarheter och attacker från tidigare studier. (3) På grund av brist på kompentens inom området penetrationstestning och telekommunikation kan en sådan här metod förhoppningsvis hjälpa till att introducera penetrationstestare utan tidigare erfarenhet till det här specifika området. Ytterligare är också målet att att öka medvetheten när det kommer till sårbarheter i sådana system. Metod. Genom en literaturstudie, hotmodellering och utforskande penetrationstestning på tre olika testmiljöer har ett flertal sårbarheter och attacker identifieras och utförts. Från resultatet utformades en metod för penetrationstesning, som sedan evaluerades genom att en tredje part testade metoden och gav återkoppling som rör metodens format och struktur. Resultat. Resultaten från vår studie visar att SIP och RTP är sårbara för en rad olika attacker än idag. Resultaten från vår litteraturstudie visar att många av dessa attacker har varit kända under en lång tid. Vi lyckades verifiera de flesta av dessa attacker genom utforskande penetationstestning på tre olika system. Dessutom lyckades vi identifiera ett antal nya attacker som inte tidigare nämnts i forskning inom området. Slutsatser. Resultaten från vår litteraturstudie visar att system som använder sig av SIP och RTP är relativt sårbara för en mängd olika attacker. Detta bekräftades i den utforskande testningen, där ett flertal kända samt nya attacker utfördes framgångsrikt. Den interna evalueringen i studien visar på att metoden kan appliceras framgångsrikt på ett flertal olika system, med begränsningen att endast tre system testats. Resultaten från den externa evalueringen, där penetrationstestare blev tillfrågade att utvärdera och testa metoden visar att de hade en relativt positiv inställning till metoden. För att ytterligare underbygga detta påstående krävs en större population, både för testningen och utvärderingen. Det krävs också att en större mängd testmiljöer används för att kunna generalisera metoden.

Thesis (MPhil)--Stellenbosch University, 2012. / ENGLISH ABSTRACT: This study investigates the translation strategy oortekening (Afrikaans for “resigning” or “redrawing”) as it is found in the bilingual context of Breytenbach’s volume of poems Oorblyfsel/ Voice Over (2009). The term oortekening (retracing), coined by Odendaal (2011), refers to corresponding parts of the Afrikaans and English versions in the volume which differs semantically but which are alike phonologically and orthographically. The argument is made that, in the context of the bilingual layout of the volume, the translation strategy oortekening invites bilingual readers to take part in an interpretative game as they read the two versions side-by-side. It is further argued that currently there are not adequate theories in translation studies to describe and understand this interpretative play. This study therefore uses theories from outside translation studies to describe the translation phenomenon in this volume and in this way provide a theoretical framework in which the unusual nature of the translation of the volume can be understood. Chapter one, an introduction, is followed by Chapter two which gives an overview of the main approaches in translation theory, describing how the problem statement would be described and understood by each. The linguistic approaches which are covered includes structural linguistics, text and sociolinguistics and functionalism. The literary approaches include descriptive translation studies and more ideologically orientated approaches. Enquiry is made into which aspects of each approach are useful to explain the volume and which are not. Chapter three applies three concepts from outside translation studies to oortekening in the context of the bilingual layout of the volume: Pratt’s contact zone (1987, 1991, 1992, 1996), Bhabha’s in-between (1993) and Lecercle’s remainder (1990). These concepts are used to analyse examples of oortekening from the volume. In the last chapter the findings of the study are discussed. The relationship between the two languages in the volume is compared to the historical and current relationship between Afrikaans and English. The chapter ends with a discussion of the variety of possible further studies based on this volume and this study. / AFRIKAANSE OPSOMMING: Hierdie studie ondersoek die vertaalstrategie oortekening soos wat dit in die tweetalige konteks van Breytenbach se bundel Oorblyfsel/ Voice Over (2009) voorkom. Die term oortekening is aan Pieter Odendaal (2011) ontleen en verwys na ooreenkomstige dele van die Afrikaanse en Engelse weergawes in die bundel wat semanties verskil maar fonologies en ortografies ooreenkom. Daar word geargumenteer dat die vertaalstrategie oortekening in die konteks van die tweetalige uitleg van die bundel tweetalige lesers nooi om aan ’n interpretatiewe spel deel te neem soos wat hulle die twee weergawes saam lees. Daar word verder geargumenteer dat daar tans in die vertaalkunde nie gepaste teorieë is om die interpretatiewe spel te beskryf en te begryp nie. Die studie gebruik dus teorieë van buite die vertaalkunde om die vertaalverskynsel in hierdie bundel te beskryf en om so ’n teoretiese raamwerk voor te stel waarbinne die ongewone aard van die bundel se vertaalaspek begryp kan word. Die inleidende hoofstuk word gevolg deur hoofstuk twee wat ’n oorsig gee van die vernaamste vertaalteoretiese benaderings en hoe die probleemstelling volgens elk van die benaderings beskryf en verstaan word: die linguistiese benaderings, vanaf die strukturele linguistiek, teks- en sosiolinguistiek na die funksionalisme; en die literêre benaderings, wat deskriptiewe vertaalstudies en meer ideologies gefundeerde benaderings insluit. Daar word ondersoek ingestel na watter aspekte van elk van die benaderings nuttig is om die bundel te beskryf en watter tekortkominge elke benadering bied. Hoofstuk drie maak drie konsepte van buite die vertaalkunde van toepassing op oortekening in die konteks van die tweetalige uitleg van die bundel: Pratt (1987, 1991, 1992, 1996) se contact zone as die kontaksone, Bhabha (1993) se in-between as die (in)tussen(in) en Lecercle (1990) se remainder as die oorblyfsel. Hierdie konsepte word gebruik om voorbeelde van oortekening uit die bundel te analiseer. In die laaste hoofstuk word die bevindinge van die studie bespreek. Die verhouding tussen die twee tale in die bundel word vergelyk met die historiese en huidige verhouding tussen Afrikaans en Engels. Daar word laastens gewys op ’n verskeidenheid verdere navorsingsmoontlikhede wat hierdie bundel en hierdie studie bied.

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.

This study represents an investigation into the effect of multiple production techniques on the recall and retention of information of university students. The purpose of the study was to determine how the ‘Voice of God’ narration affects recall and retention of information compared to an on-camera interview. In documentary video, conveying information is one of the key goals of the director. In most cases, the conveyance of information in itself is insufficient. The director seeks the emotional participation of the audience so that they can become aware of a specific issue. In this mediated environment, the producer’s attempts at reaching the audience are interwoven with the producer’s capability to facilitate the audience’s recall of information. By using the ‘Voice of God’ narration, a director can enhance the narrative and make the information more understandable. Theory suggests that the use of multiple production techniques can either have a negative or positive impact on the processing of information. This statement is based on different experiments that were done by researchers on how various production techniques affect the information processing of an individual. The theoretical basis of the study is rooted in the metatheory, cybernetics. Within cybernetics, the narrative theory explains the structure of the story and how it is conveyed to an audience. Voice-over in documentary video is situated in the narrative theory because the structure of the information the voice-over gives to an audience is of utmost importance. From the theoretical basis, this study uses Lang’s (2000) limited capacity model of mediated message processing to investigate the effect of narration (voice-over) as embedded in documentary video. Specifically, it addresses the mediator’s (in documentary video, the producer’s) goal of maximum information recall by the receiver of the message. The application of Lang’s model is outlined in an empirical design that explores recall of message content and the retention of information in two ways; the recall and retention of information presented through an on-camera interview and the recall and retention of information presented by a narrator whilst images are shown that do not include an image of the narrator him/herself (Voice of God narration). Two experiments were designed for the purpose of this study in which 37 students from the North-West University’s Potchefstroom campus participated. The students were divided into two groups; group one watched the video where the information is presented by an on-camera-interview, and group two watched the video where the information is presented by a ‘Voice of God’ narrator. Two questionnaires were given to the groups at two different times. The results obtained suggest that there is no significant difference in the production techniques and the recall and retention of information. Based on the results, certain recommendations are made for future research, which include modifying the message and research design. / MA (Communication Studies), North-West University, Potchefstroom Campus, 2014

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

O projeto VoIPFix surgiu da necessidade de uma ferramenta que complementasse as demais existentes no ramo de análise de redes de computadores para telefonia IP. Ele foi construído para ser uma ferramenta de gerenciamento eficiente e exclusiva para VoIP, com funcionalidades necessárias para dar suporte ao profissional de rede de computadores e telefonia IP a observar e diagnosticar problemas de VoIP. / The VoIPFix project arose from the need for a tool to complement similar tools in the analysis of computer networks for IP telephony. It was built to be an efficient and unique management tool for VoIP, with advanced features required to support the computer network and IP telephony professionals to observe and diagnose problems related to VoIP.