The Internet has made tremendous progress since its inception. The kingpin has been the transmission control protocol (TCP), which supports a large fraction of communication. With the Internet's wide-spread access, users now have increased expectations. The demands have evolved to an extent which TCP was never designed to support. Since network stacks do not provide the necessary functionality for modern applications, developers are forced to implement them over and over again --- as part of the application or supporting libraries. Consequently, application developers not only bear the burden of developing application features but are also responsible for building networking libraries to support sophisticated scenarios. This leads to considerable duplication of effort. The challenge for TCP in supporting modern use cases is mostly due to limiting assumptions, simplistic communication abstractions, and (once expedient) implementation shortcuts. To further add to the complexity, the limited TCP options space is insufficient to support extensibility and thus, contemporary communication patterns. Some argue that radical changes are required to extend the networks functionality; some researchers believe that a clean slate approach is the only path forward. Others suggest that evolution of the network stack is necessary to ensure wider adoption --- by avoiding a flag day. In either case, we see that the proposed solutions have not been adopted by the community at large. This is perhaps because the cost of transition from the incumbent to the new technology outweighs the value offered. In some cases, the limited scope of the proposed solutions limit their value. In other cases, the lack of backward compatibility or significant porting effort precludes incremental adoption altogether. In this dissertation, we focus on the development of a communication model that explicitly acknowledges the context of the conversation and describes (much of) modern communications. We highlight how the communication stack should be able to discover, interact with and use available resources to compose richer communication constructs. The model is able to do so by using session, flow and endpoint abstractions to describe communications between two or more endpoints. These abstractions provide means to the application developers for setting up and manipulating constructs, while the ability to recognize change in the operating context and reconfigure the constructs allows applications to adapt to the changing requirements. The model considers two or more participants to be involved in the conversation and thus enables most modern communication patterns, which is in contrast with the well-established two-participant model. Our contributions also include an implementation of a framework that realizes such communication methods and enables future innovation. We substantiate our claims by demonstrating case studies where we use the proposed abstractions to highlight the gains. We also show how the proposed model may be implemented in a backwards compatible manner, such that it does not break legacy applications, network stacks, or middleboxes in the network infrastructure. We also present use cases to substantiate our claims about backwards compatibility. This establishes that incremental evolution is possible. We highlight the benefits of context awareness in setting up complex communication constructs by presenting use cases and their evaluation. Finally, we show how the communication model may open the door for new and richer communication patterns. / PHD

High-performance networking (HPN) is of significance today in order to enable next-generation applications using wired and wireless networks. Some of the examples of HPN include low-latency industrial sensing, monitoring and automation using Wireless Sensor Networks (WSNs). HPN however requires protocol optimization at many layers of the open system interface (OSI) network model in order to meet the stringent performance constraints of the given applications. Furthermore, these protocols need to be impervious to denial of service (DoS) and distributed DoS (DDoS) attacks. Some of the key performance aspects of HPN are low point-to-point and end-to-end latency, high reliability of transmitted frames and performance predictability under various network load situations. This work focuses on two discrete issues in designing protocols for HPN applications. The first research issue looks at the Medium Access Control (MAC) layer of the OSI network model for designing of MAC protocols that provide low-latency and high reliability for point-to-point communication under a WSN. Existing standards in this area are governed by IEEE 802.15.4 specification which defines protocols for MAC and PHY layers for short-range, low bit-rate, and low-cost wireless networks. However, the IEEE 802.15.4 specification is inefficient in terms of latency and reliability performance and, as a result, is unable to meet the stringent operational requirements as defined by counterpart wired sensor networks. Work presented under current research issue describes new MAC protocols that are able to show low-latency transmission performance under strict timing constants for power limited WSNs. This enhancement of the MAC protocols is named extended GTS (XGTS) contained under extended CFP (ECFP) and is published under the IEEE's 802.15.4e standard. The second research issue focuses on the application layer of the OSI network model to design protocols that enhance the robustness of the text based protocols to various traffic inputs. The purpose of this is to increase the reliability of the given text based application layer protocol under a varied load. Session Initiation Protocol (SIP) is used as a case study and the work aims to build algorithms that ensure that SIP can continue to function under specific traffic conditions, which would otherwise deem the protocol useless due to DoS and DDoS attacks. Proposed algorithms investigate techniques that enhance the robustness of the SIP against parsing attacks without performing a deep parse of the protocol data unit (PDU). The desired effect of this is to reduce the time spent in parsing the SIP messages at a SIP router and as a result increase the number of SIP messages processed per unit time at a SIP router.

In response to the development of a next-generation networking (NGN) generalized multi-protocol label switching (GMPLS) technology is required for automatically switched optical network (ASON). Reconfigurable optical add-drop multiplexer (ROADM) is an indispensable device for the ASON, and the dense wavelength division multiplexed (DWDM) signals can be transmitted through the network under the management of the network administrator to configure dynamic customer needs and the desired quality of service (QoS). The ROADM can also increase the efficiency of utilizing the existing capacity of the optical fiber lines and can reduce or waive to set up additional optical fiber lines. This thesis studies a novel ROADM based on the arrayed waveguide grating (AWG) and the fiber Bragg grating (FBG) to overcome that the current ROADM cannot process that the input signal channels is greater than the wavelengths channels of AWG. Many types of ROADMs have been proposed and realized through different optical devices. Among these, hybrid optical circulator and FBG based ROADM is more attractive because of its low crosstalk and polarization insensitivity. However, it still suffers from many component counts and high insertion losses due to the use of many circulators and a multiplexer-demultiplexer pair. In this master thesis, we focus on demonstrating a novel ROADM and evaluating its crosstalk performance. It is found that the proposed ROADM has the advantage on extending the number of wavelength signal to make the system configurable and flexible.

Internet Protocol multimedia subsystem (IMS) is an architecture for services such as voice over Internet Protocol (VoIP) in IP based communication systems. IMS is standardized by the 3GPP standardization forum, and was first released in 2002. Since then IMS has not had the wide adoption by operators as first anticipated. As 3G already supported voice and video, the operators could not justify the expense of IMS. The current emergence of the fourth generation mobile communication system named Long Term Evolution (LTE) has, however, increased the need for knowledge of IMS and of creating services for it. LTE networks are IP only networks that provide low latency. In order to use LTE for making phone calls, VoIP technologies are needed. IMS is the architecture intended to be used for Voice over LTE (VoLTE). The need for tools for education within IMS was seen in 2006 by Enea Experts in Linköping, Sweden. The author of this thesis designed an IMS for educational purposes, but the project was never fully completed. This thesis will reexamine the design decisions previously made by the author. The requirements stated by the customer remain: that an IMS with basic signaling and logging should be easy to install, maintain, and evolve at a low cost. A literature study of IMS and VoLTE is presented to contribute with knowledge in these areas. The previous design and implementation made by the author is presented and analyzed. The third-party software that the previous implementation was based on is reexamined. Existing open source components are analyzed in order to identify how they can be used to solve the problem and to identify what remains to be developed in order to fulfill the requirements. New design suggestions, presented in today´s context, are proposed and verified using analytical reasoning and experiments. The outcome of the final work is new verified design decisions for the customer to use when implementing a new IMS for educational purposes. The thesis should also provide useful insights which instructors and students can use to teach and learn more about IMS. / Internet Protocol multimedia subsystem (IMS) är en arkitektur för tjänster, som IP-telefoni (Voice over Internet Protocol, VoIP), i IP baserade kommunikationssystem. IMS standardi¬seras av standardiseringsforumet 3GPP och första utgåvan släpptes år 2002. IMS fick dock inte det breda genomslag bland operatörer som förväntats. Eftersom 3G redan hade stöd för tal och video kunde operatörerna inte se skäl till ytterligare utgifter för IMS. Den fjärde generationens mobila kommunikationssystem, Long Term Evolution (LTE) är helt IP-baserat och ger lägre fördröjningar i nätet. För att kunna ringa telefonsamtal via LTE krävs VoIP-teknik. IMS är en arkitektur avsedd för att användas för Voice over LTE (VoLTE). Den nuvarande utvecklingen av LTE har därför ökat behovet av kunskap om IMS och av utveckling av IMS-tjänster. Enea Experts i Linköping insåg behovet av verktyg för utbildning inom IMS år 2006. Författaren av det här examensarbetet designade därför ett IMS för utbildningssyfte. Projektet slutfördes dock aldrig. Syftet med examensarbetet är att ompröva de tidigare designbesluten. Kundens krav kvarstår: att ett IMS med grundläggande signalering och loggning bör vara enkelt att installera, enkelt att underhålla och möjligt att utveckla till en låg kostnad. Arbetet innehåller en litteraturstudie av IMS och VoLTE för att ge en inblick i dessa områden. Den tidigare designen och implementationen presenteras och analyseras. Tredjeparts mjukvara, som den tidigare implementationen baserades på, omprövas. Befintliga programvaror med öppen källkod analyseras i syfte att kartlägga hur de kan användas för att lösa uppgiften, samt att identifiera vad som återstår att utveckla för att uppfylla kraven. Nya beslut kring design presenteras och besluten verifieras med experiment och analytiskt resonemang. Resultatet av detta examensarbete innefattar nya verifierade beslut kring design som kunden kan använda vid utveckling av ett nytt IMS för utbildningssyfte. Arbetet erbjuder också värdefulla insikter som instruktörer och elever kan använda för att undervisa samt för att lära sig mer om IMS.