The topic of the diploma thesis is " Application of simulation model of access network". The aim of the thesis was to create a part of the access network in which were tested simulations for TCP and UDP traffic. The theoretical part of the thesis deals with general information about the GNS3 program and the possibilities which this program allows. The next part of the thesis describes the Mikrotik RouterOS program, which was used in the topology. The next section describes the types of routing in IP networks, as well as the types of packet delivery. The practical part deals with the creation of test topologies and verification of the performance of the simulation tool. Furthermore, the practical part deals with the programming of a network generator for TCP and UDP traffic in the network. The generator was created by using a Bash script and a combination of the dd and nc commands. At the end, the thesis describes the results of simulations in terms of transmission speed, network throughput, packet size distribution and in the case of TCP also bidirectional delay – RTT.

This Diploma thesis describes the procedure of setting the parameters of communication blocks and functions in PLC and PC, so that it is possible to remotely control part of the production process. TCP / IP protocol is used for communication between devices. Part of the production technology is represented by the created 3D model, which serves for three-dimensional interpretation of the simulation of the mathematical model of this part of the process. Part of this work is the design of this model and description of its dynamics.

This diploma thesis describes a possibility of application protocol implementation in OPNET Modeler simulation environment. It presumes that this application protocol is going to use TCP protocol for their communication on transport layer. The first part of thesis is focused on a description of TCP. It is a connection oriented, reliable and confirmed protocol which maintains sequence of transmitted data. This data is after receiving positively confirmed. In the second chapter are described the main functions of OPNET Modeler simulation environment. OPNET is hierarchical divided into four editors. Each editor has a specific function by creating a network and setting his behavior. There is also focused on the two lowest layers of OPNET Modeler and their components in detail, which are participating at usage TCP on transport layer to communication. Implemented applications communicate by sockets, which are created and destroyed on request. Communication is controlled by manager process, whose function is maintenance particular connections and redirect dataflow into relevant process. This manager put in action as well a process, which simulates a single TCP. In the practical part I made two applications: one is a client type and a second one is a server type. Both applications are using TCP on transport layer. The establishment of connection initiates a client, who is sending a request to the server for a data. Then server sends back data in desired quantity. After sending the entire data, client terminates the connection. The result of simulation is statistics, where we pictured the size of the transferred data, a number of transferred packets and other parameters typical for TCP.

The Master´s Thesis describes laboratory exercises for Advanced communication technology subject. Exercises are created in Opnet IT Guru aplication. Each exercise has detailed instructions according to which can be realized whole exercise. Each exercise also contains individual task, where are checked knowledge gained from studies of example exercise. Exercises are based on issues of routing in autonomous systems, protocol BGP, protocols TCP and UDP and issues of QoS in IP networks.

In order to easily send data between two end elements without congestion, methods that suitably control flow of date and evaluate possible overload state are necessary. One such method is to control the data flow directly on the transport layer. This layer offers a range of mechanisms dedicated to deal with this issue. The aim of this paper is divided into three parts. The first part describes the integration of transport layer TCP/IP model, and the ability to process TCP data stream. The second part describes methods to manage congestion, their integration by usage environment. It mainly focuses on methods of TCP Reno and TCP Vegas. Their simulation and analysis on transmission the data stream stream. The third part deals with the analysis in detail of TCP Vegas. Analyzes possible parameters for alpha a beta within the TCP Vegas, and a combination of TCP Vegas and TCP Reno.

Buffer sizing is an important network configuration parameter that impacts the quality of data traffic. Falling memory cost and the fallacy that ‘more is better’ lead to over provisioning network devices with large buffers. Over-buffering or the so called ‘bufferbloat’ phenomenon creates excessive end-to-end delay in today’s networks. On the other hand, under-buffering results in frequent packet loss and subsequent under-utilization of network resources. The buffer sizing problem has been studied extensively for wired networks. However, there is little work addressing the unique challenges of wireless environment. In this dissertation, we discuss buffer sizing challenges in wireless networks, classify the state-of-the-art solutions, and propose two novel buffer sizing schemes. The first scheme targets buffer sizing in wireless multi-hop networks where the radio spectral resource is shared among a set of con- tending nodes. Hence, it sizes the buffer collectively and distributes it over a set of interfering devices. The second buffer sizing scheme is designed to cope up with recent Wi-Fi enhancements. It adapts the buffer size based on measured link characteristics and network load. Also, it enforces limits on the buffer size to maximize frame aggregation benefits. Both mechanisms are evaluated using simulation as well as testbed implementation over half-duplex and full-duplex wireless networks. Experimental evaluation shows that our proposal reduces latency by an order of magnitude.

The Transmission Control Protocol (TCP), used in the Internet, was not developed for a mobile, wireless environment. One reason why TCP might encounter problems in such an environment is rapid changes of the link characteristics. These rapid changes can occur due to handover between two subnetworks (macro handover), e.g., when a mobile node switches between different access networks. A possible and realistic handover scenario could be when a mobile node is roaming between a high bandwidth local area network (LAN) with limited coverage and a low bandwidth radio link with wide area coverage. The goals of this thesis were to set up a realistic environment for measurements of the handover performance of TCP, and to observe TCP behavior when the link characteristics suddenly change. Further objectives were to analyze the results and propose solutions for improving the performance. The mobility management in the measurement setup is handled by Mobile IP. Handovers are performed between a wireless LAN (WaveLAN) and a PPP link over a GSM circuit switched data connection. Our investigation shows that several spurious TCP timeouts occur after handover from the fast link to the slow link, triggering unnecessary retransmissions and hence resulting in TCP performance degradation. To avoid unnecessary retransmissions, we suggest a resetting of the retransmission timeout value (RTO) at the moment of handover. In the case of handover from the low bandwidth link (PPP) to the high bandwidth link (WaveLAN), our measurements show that queued packets in the link layer buffer continue to flow over the PPP link even after the handover. The high bandwidth available after the switch is thus poorly utilized before the buffer of the low bandwidth link has been emptied. The IP sending process should delay putting packets in the queue of a slow link, thus avoiding large link layer queues and enabling utilization of the high bandwidth link faster. This could be achieved by flow control between the IP layer and the link layer. After the packets have started flowing over the WaveLAN, the RTO value is unnecessarily high, which could result in extensive delays in the case of packet losses. To alleviate the problems we recommend resetting the RTO value or modifying the algorithm for calculating the RTO value to faster adapt to sudden and significant decrease of the round-trip time (RTT) in the case of handover. In both handover scenarios mentioned above we have found that a small window size is favorable to mitigate the negative effects due to the rapid changes of the link characteristics. The use of Active Queue Management to avoid large window sizes would be an interesting approach for future investigations. Another interesting approach could be to have flow control between the IP layer and the link layer to avoid a large link layer queue when the handover from [one link to another occurs.]

With Ericsson’s goal to have optimal user experience at 5G’s 2020 release, it is very important to optimize transport protocols and techniques to manage the increasing amount of data traffic. Additionally, it will be important to manage handovers between very high speed 5G networks and older networks. Today most of the traffic is video on demand and the amount of this kind of traffic is expected to increase. Moreover, the current amount of data traffic will increase by an order of magnitude over the next few years. This thesis focuses on radio access networks and the difficulties they face in delivering high speed data traffic. This thesis analyzes one of the most used TCP protocols, CUBIC, as well as a new transport protocol developed by Aspera, called the Fast and Secure Protocol. Aspera’s FASP is a new transport protocol that promises full link utilization. FASP is built upon UDP and uses advanced round trip time measurements and queuing delay to detect the available bandwidth between two communicating hosts. This thesis project also provides methods to realize experiments to assess the limitations of transport protocols. These experiments are conducted in an environment that resembles the upcoming 5G radio access network. Results have shown that both delay and packet loss affect TCP more than we expected and that high packet loss is devastating. In contrast, Aspera’s FASP is very resistant to both delay and packet loss. These results and analysis provide a foundation upon which others can build. / Med Ericssons mål att ha optimal användarupplevelse vid släppet av 5G år 2020 är det oerhört viktigt att optimera transportprotokoll och tekniker för att hantera den ökande mängden datatrafik. En annan viktig aspekt kan vara att hantera överlämningar mellan 5G nätverk och äldre radionätverk. Idag är den största trafiken streamad video och prognoser visar att den sortens trafik bara kommer att öka. Prognoserna visar också att all trafik kommer att öka mångfaldigt de närmaste åren. Denna thesis kommer att fokusera på svårigheterna just inom radionätverk. Denna thesis kommer att analysera ett av vårt mest använda transportprotokoll CUBIC TCP, den kommer också att analysera ett helt nytt transportprotokoll utvecklat av Aspera, Fast and Secure Protocol. Aspera lovar fullt utnyttjande av den mellanliggande länken. FASP är byggt ovanpå UDP och använder avancerade tur- och returtidsmätningar för att använda all outnyttjad bandbredd. Denna thesis visar även hur man kan göra experiment för att hitta begränsningar i transportprotokoll. Alla dessa experiment kommer utförs i en miljö som efterliknar det nya 5G-nätverket. Resultatet visar att både förlora paket tillsammans med en hög fördröjning påverkar mycket mer än väntat och att frekvent förlora paket är förödande för TCP. Asperas FASP är i motsats mycket tålig mot både paketförlust och hög fördröjning. Detta resultat och denna analys lägger en grund var andra kan arbeta vidare.

New generations of wireless cellular networks, including 3G and 4G technologies, are envisaged to support more mobile users and a variety of wireless multimedia services. With an increasing demand for wireless multimedia services, the performance of TCP becomes a bottleneck as it cannot differentiate between the losses due to the nature of air as a medium and high data load on the network that leads to congestion. This misinterpretation by TCP leads to a reduction in the congestion window size thereby resulting in reduced throughput of the system. To overcome this scenario Radio Link Protocols are used at a lower layer which hides from TCP the channel related losses and effectively increases the throughput. This thesis proposes enhancements to the radio link protocol that works underneath TCP by identifying decisive frames and categorizing them as {\em crucial} and {\em non-crucial}. The fact that initial frames from the same upper layer segment can afford a few trials of retransmissions and the later frames cannot, motivates this work. The frames are treated differentially with respect to FEC coding and ARQ schemes. Specific cases of FEC and ARQ strategies are then considered and it is shown qualitatively as how the differential treatment of frames can improve the performance of the RLP and in effect that of TCP over wireless networks.

In this thesis, a new model for the slow start phase based on the discrete evolutions of congestion window is developed, and we integrate this part into the improved TCP steady state model for a better prediction performance. Combining these short and steady state models, we propose an extensive stochastic model which can accurately predict the throughput and latency of the TCP connections as functions of loss rate, round-trip time (RTT), and file size. We validate our results through simulation experiments. The results show that our model?s predictions match the simulation results better than the Padhye and Cardwell's stochastic models, about 75% improvement in the accuracy of performance predictions for the steady state and 20% improvement for the short-lived TCP flows.