This thesis discusses wireless network standards 802.11a/b/g/n. First part explains basic principles of networks and media access. Next the standard IEEE 802.11, general QoS parameters and their application in wireless networks, according to standard 802.11e are analyzed. Second part of the thesis verifies the acquired knowledge in simulating program Opnet - effects of the delay, jitter and packet loss on VoIP call are tested. In the last part of the thesis a network for video streaming has been designed. The video was streamed in different qualities and the influence of other network traffic (with and without the support of QoS) on the video streaming was tested.

Das Ziel dieser Arbeit ist es, ein Handover-Verfahren für ein Siemens Handset zu entwickeln. Die Entwicklungsumgebung beruht auf den Wireless-LAN Standards 802.11 der IEEE (Institute of Electrical and Electronics Engineers). Dabei liegen die Schwerpunkte auf den Standardisierungen 802.11f und 802.11i, wobei sich eine neue Arbeitsgruppe (IEEE 802.11r) direkt mit dem Thema "Handover" beschäftigen wird. Das Handset soll selbständig die Verwaltung und Einleitung des Handovers übernehmen und lediglich insofern vom Access Point unterstützt werden, dass dieser als Informationssammler dient und somit Entscheidungshilfen geben kann.

Eine kurze Einführung zu "Voice over IP" (dem Telefonieren über Datennetze). Es wird ein Überblick über technische Anforderungen und Lösungen geben. Behandelte Gebiete sind Audio-Codecs, das Transportprotokoll RTP sowie die Signalisierungsdienste SIP und H.323.

Over the course of last decade, the demand for VoIP (Voice over Internet Protocol) applications has increased significantly among enterprises and individuals due to its low cost. This increasing demand resulted in a significant increase in users who require reliable VoIP communication systems. QoS (Quality of Service) is a major issue in VoIP implementation and is a method to impel the development of real-time multimedia services like VoIP and videoconferencing. However, there are certain challenges in achieving QoS for VoIP application, which need special attentions; like latency and packet loss. The VoIP servers which are functioning on single core software/hardware model have high latency and packet loss issues due to their limited processing bandwidth. A multicore software/hardware model is the solution to cope up with the increasing demands of VoIP and yet an active research area in telecommunication. Using a multicore software/hardware model for VoIP has several challenges, one of the challenges is to design and implement QoS Benchmarking module for VoIP client and server on multicore. In this thesis the focus is on latency and packet loss of SIP messages on OpenSIPS server. This is done by performing stress testing for QoS benchmarking, where delay and call drop rate is calculated for SIP (Session Initiation Protocol) signaling messages on parallel VoIP client server model. The model is built in C for multicore and is used as a simulation tool. SIP is widely deployed protocol for call establishment, maintenance and termination in VoIP.

In contemporary IT industry, Multi-tasking solutions are highly regarded as optimal solutions, because hardware is equipped with multi-core CPUs.  With Multi-Core technology, CPUs run with lower frequencies while giving same or better performance as a whole system of processing. This thesis work takes advantage of multi-threading architecture in order to run different tasks under different cores such as SIP signaling and messaging to establish one or more SIP calls, capture voice, medical data, and packetize them to be streamed over internet to other SIP agents. VoIP is designed to stream voice over IP. There is inter-protocol communication and cooperation such as between the SIP, SDP, RTP, and RTCP protocols in order to establish a SIP connection and- afterwards- stream media over the internet. We use the Microsoft COM technology in order to better the C++ component design. It allows us to design and develop code once and run it anywhere on different platforms. Using VC++ helps us reduce software design time and development time. Moreover, we follow software design standards setup by software engineers’ society. VoIP technology uses protocols such as the SIP signaling protocol to locate the user agents that communicate with each other. Pjsip is a library that allows developers to extend their design with SIP capability. We use the PJSIP library in order to sign up our own developed VoIP module to a SIP server over the Internet and locate other user agents. We implement and use the already-designed iRTP protocol instead of the RTP to stream media over the Internet. Thus, we can improve RTP packet delays and improve Quality of Service (QoS). Since medical data is critical and must not be lost, the iRTP guarantees no loss of medical data. If we want to stream voice only, we would not need iRTP, because RTP is a good protocol for voice applications. Due to the increasing Internet traffic, we need to use a reliable protocol that can detect packet loss of medical data. iRTP resolves the issue and leverages QoS. This thesis work focuses on streaming medical data and medical voice-calls using VoIP, even over small bandwidths and in high traffic periods. The main contribution of this thesis is in the parallel design of iRTP and the implementation of this very design in order to be used with Multi-Core technology. We do so via multi-threading technology to speed up the streaming of medical data and medical voice-calls. According to our tests, measurements, and result analyses, the parallel design of iRTP and the multithreaded implementation on VC++ leverage performance to a level where the average decrease in delay is 71.1% when using iRTP for audio and medical data instead of the nowadays applied case of using an RTP stream for audio and multiple TCPs streams for medical data .

This master thesis presents, examines, designs, implements, and evaluates with respect to energy efficiency a secure and robust VoIP system. This system utilizes a Session Initiation Protocol (SIP) infrastructure assisted by a cloud service, specifically focusing on small to medium sized enterprises (SME) and homes. This research focuses on using inexpensive, flexible, commodity embedded hardware (specifically a Linksys WRT54GL wireless router for the local site with a customized operating system, specifically DD-WRT). The idea is to reduce the local site's power consumption to very low levels by examining which functions can be done in a cloud service rather than at the local site. The thesis presents the design of a low-power IP telephony system for the local site and the cloud site. A number of different usage scenarios and desirable features are described. The methodology for conducting a set of experiments is defined to perform stress-testing and to evaluate the low- power IP telephony system's design. The experiments concern the overall power consumption of the local site under various configurations, the VPN link's call capacity, the QoS metrics for the VoIP calls, the session request delay (SRD) and the registration request delay (RRD). The results from these experiments show that there is a potential for significant power savings when using the proposed design for an IP telephony system. / Detta examensarbete presenterar, undersöker, utformar, implementerar, och försöker att utvärdera ett säkert och robust VoIP-system med energieffektivitet. Detta system använder en Session Initiation Protocol (SIP)-infrastruktur med hjälp av en molntjänst med särskild inriktning på, små, och medelstora företag (SME) och hemmanvändare. Denna forskning fokuserar att använda en prisvärt, billig, flexibel, med program inbyggd hårdvara (speciellt en Linksys WRT54GL trådlös router för den lokala platsen med ett anpassat operativsystem DD-WRT). Tanken är att minska energiförbrukningen på, den lokala platsen till mycket låga nivåer genom att undersöka vilka funktioner, som kan köras på, ett molntjnst snarare än på, den lokala platsen. Avhandlingen presenterar utformningen av ett IP-telefonisystem på, den lokala platsen med ett lågt strömbehov och på, molntjänsten. Ett antal olika användningsförhållanden och önskvärda egenskaper är beskrivna. Metodiken för att genomföra en rad experiment definieras för att utföra stresstester och för att utvärdera designen av IP-telefonisystem med ett lågt effektbehov. I försöken experimenteras den totala energiförbrukningen av den lokala platsen under olika konfigurationer, VPN-länkens samtalskapacitet, QoS-mätning för VoIP-samtal, Session Request Delay (SRD) och Registration Request Delay (RRD). Resultaten från dessa experiment visar att det finns en potential för betydande energibesparing när du använder den föreslagna designen för en IP-telefoni system.

Using a Key escrow agent in conjunction with Voice over IP (VoIP) communication ensures that law enforcements agencies (LEAs) can retrieve the session key used to encrypt data between two users in a VoIP session. However, the use of a single escrow agent has some drawbacks. A fraudulent request by an evil employee from the LEA can lead to improper disclosure of a session key. After the escrow agent reveals the key this evil person could fabricate data according to his/her needs and encrypt it again (using the correct session key). In this situation the persons involved in the communication session can be accused of crimes that he or she or they never committed. The problems with a single escrow agent becomes even more critical as a failure of the escrow agent can delay or even make it impossible to reveal the session key, thus the escrow agent might not be able to comply with a lawful court order or comply with their escrow agreement in the case of data being released according to this agreement (for example for disaster recovery). This thesis project focused on improving the accessibility and reliability of escrow agents, while providing good security. One such method is based on dividing the session key into M chunks and escrowing the chunks with M escrow agents. Using threshold cryptography the key can be regenerated by gathering any N-out-of-M chunks. The value of M and N may differ according to the role of the user. For a highly sophisticated session, the user might define a higher value for M and N for improved, availability, reliability, and security. For a less confidential or less important session (call), the value of M and N might be smaller. The thesis examines the increased availability and increased reliability made possible by using multiple escrow agents. / Med en nyckel förvaringsinstitut som tillsammans med Röst över IP (VoIP) kommunikation säkerställer att brottsbekämpande organ (LEAs) kan hämta sessionsnyckeln används för att kryptera data mellan två användare i en VoIP-session. Däremot har användningen av ett enda förvaringsinstitut visa nackdelar. En bedräglig begäran av en ond arbetstagare från LEA kan leda till otillbörligt röjande av en sessionsnyckel. Efter förvaringsinstitut avslöjar nyckeln detta onda person kunde fabricera uppgifter i enlighet med hans eller hennes behov och kryptera den igen (med rätt sessionsnyckel). I denna situation personer som deltar i kommunikationssession kan anklagas för brott som han eller hon eller de aldrig begått. Problemen med en enda förvaringsinstitut som blir ännu mer kritisk som ett misslyckande av förvaringsinstitut kan försena eller till och med gör det omöjligt att avslöja sessionsnyckeln, vilket förvaringsinstituten kanske inte kan följa en laglig domstolsbeslut eller uppfyller sina depositionsavtalets när det gäller data frisläppas i enlighet med detta avtal (till exempel för katastrofer). Detta examensarbete fokuserar på att förbättra tillgängligheten och tillförlitligheten i spärrade medel, samtidigt som god säkerhet. En sådan metod bygger på att dela upp sessionsnyckeln till M bitar och escrowing i bitar med M förvaringsinstituten. Använda tröskel kryptografi nyckeln kan genereras genom att samla alla N-out-of-M bitar. Värdet på M och N kan variera beroende på användarens roll. För en mycket sofistikerad session kan användaren definiera ett högre värde för M och N för förbättrad tillgänglighet, tillförlitlighet och säkerhet. För en mindre konfidentiell eller mindre viktigt session (telefonsamtal), kan värdet på M och N vara mindre. I avhandlingen analyseras den ökade tillgänglighet och ökad tillförlitlighet möjligt genom att använda flera spärrade medel.

Modern technologies are growing and Voice over Internet Protocol (VoIP) technology is able to function in heterogeneous networks. VoIP gained wide popularity because it offers cheap calling rates compared to traditional telephone system and the number of VoIP subscribers has increased significantly in recent years. End users need reliable and acceptable call quality in real time communication with best Quality of Service (QoS). Server complexity is increasing to handle all client requests simultaneously and needs huge processing power. VoIP Servers will increase processing power but the engineering tradeoff needs to be considered e.g. increasing hardware will increase hardware complexity, energy consumption, network management, space requirement and overall system complexity. Modern System-on-Chip (SoC) uses multiple core technology to resolve the complexity of hardware computation. With enterprises needing to reduce overall costs while simultaneously improving call setup time, the amalgamation of VoIP with SoC can play a major role in the business market. The proposed VoIP Server model with multiple processing capabilities embedded in it is tailored for multicore hardware to achieve the required result. The model uses SystemC-2.2.0 and TLM-2.0 as a platform and consists of three main modules. TLM is built on top of SystemC in an overlay architectural fashion. SystemC provides a bridge between software and hardware co-design and increases HW & SW productivity, driven by fast concurrent programming in real time. The proposed multicore VoIP Server model implements a round robin algorithm to distribute transactions between cores and clients via Load Balancer. Primary focus of the multicore model is the processing of call setup time delays on a VoIP Server. Experiments were performed using OpenSIP Server to measure Session Initiation Protocol (SIP) messages and call setup time processing delays. Simulations were performed at the KTH Ferlin system and based on the theoretical measurements from the OpenSIP Server experiments. Results of the proposed multicore VoIP Server model shows improvement in the processing of call setup time delays.

歷史上出現許多既有領導企業被新興企業取代的例子，其背後的原因一直引起高度興趣。Christensen (1997)提出的破壞性科技 (disruptive technology)觀念為此現象提供一個解釋的方向。自破壞性創新的提出以來，已有眾多學者投入相關研究探討，但至今尚無一個廣泛被學界所接受且可適用於不同產業的模型，因此仍迫切需要更多的研究投入和實務驗證。評估破壞性創新的困難，就如預測未來般充滿著不確定性。對既有企業而言，藉由評估破壞性創新，可以對即將面對的衝擊早一步採取因應措施，使競爭優勢得以維持；對新興企業而言，評估破壞性創新即是找出成長的利基，並可依此進入主流市場甚至擊倒既有領導企業。 有鑑於評估破壞性創新的重要性和困難度，本研究提出評估成功的破壞性創新時應考量的關鍵構面，包含創新提供的性能對應於主流市場的需求、低價格或創造新的價值、市場擴散速度、以及社會環境。本研究以VoIP產業為破壞性創新個案來驗證各項構面。研究結果顯示PSTN的通話品質和功能已超過既有需求，讓VoIP有進入低階市場的機會。同時現階段VoIP的通話品質已可滿足大眾主流市場的需求，並提供較便宜的解決方案。尤其對既有網路人口而言，VoIP不僅是便宜且簡易的解決方案，更提供了多項應用服務的整合。VoIP破壞性創新正在市場上快速擴散，尤其在美國與歐洲等地區。透過評估構面的提出，使評估成功的破壞性創新能以更系統化的方式進行。評估構面的提出可為未來完整模型的建立提供一個發展方向，同時提供一個分析創新潛力的方法，產業界可以依據分析後的結果擬定相關策略。

Approved for public release, distribution is unlimited / Recent advances in information technology communications have brought about increases in bandwidth and processing speeds to encourage the growth of Internet Protocol Telephony (IPT), a method of transmitting voice conversations over data networks. Many organizations are replacing portions of their traditional phone systems to gain the benefits of cost savings and enhanced feature sets through the use of IPT. The Coast Guard has an interest in exploiting this technology, and has taken its first steps by implementing IPT at Headquarters Support Command in Washington D.C. This thesis investigates the successful implementation practices and security policies of commercial, educational, and government organizations in order to create recommendations for IPT security policies and implementation practices relevant to the Coast Guard. It includes the discussion of the public switched telephone network, an overview of IPT, IPT security issues, the safeguards available to counter security threats, the tradeoffs (e.g., voice quality, cost) required to mitigate security risks, and current IPT security policy and implementation guidance. It is supported by the study and analysis of the IPT system at Coast Guard Headquarters. The Coast Guard gains an understanding of the advantages, limitations, and security issues that it will face as it considers further implementation of IPT. / Lieutenant, United States Coast Guard