The amount of packets on-the-fly in a wireless ad-hoc network increases when the size of congestion window and the number of hop count increase. It is possible that packets may have to retransmit because large amount of on-the-fly packets may increase the media contention. Besides, packet delay can grow rapidly when the wireless network becomes congested or the channel interference remains unresolved. This thesis presents a cross-layer TCP congestion window control mechanism for multihop ad-hoc networks to dynamically adjust the size of congestion window according to the MAC-layer contention statistics measured at each hop along the routing path. With the proposed scheme, the congestion window of each traffic flow can be dynamically set to an appropriate size to reduce packet delay and increase flow throughput. For the purpose of evaluation, we perform simulations on NS-2. The simulation results have shown the advantage of our proposed scheme over the two previous works, NewReno and CWL(Congestion Window Limit), especially when the ad-hoc network is loaded with background traffic.

Abstract The throughput of a TCP connection is decided by the size of the congestion window. And cwnd increases when an acknowledgement arrives. It leads to that TCP has a bias against connections with long round-trip-time. For enhancing the fairness of TCP, we proposed a new scheme FTCP (Fair TCP). Unlike TCP, in FTCP congestion avoidance state, it compares its RTT with the standard RTT to adjust the increase amount of cwnd when an ACK arrives TCP sender. Therefore FTCP can keep the throughput increase rate of connections with different RTTs be the same. When FTCP enters timeout state, it sets appropriate slow start threshold by calculating the difference value of cwnd / 2 and the cwnd while standard connection achieves ssthresh. So that FTCP can eliminate the difference of throughput between connections with different RTT while leaving the slow start state. FTCP significantly improves the unfair bandwidth distribution between connections with different RTT. TCP connections over wireless links perform badly because of the unnecessary congestion control, inefficiency to burst packet loss, and long delay to slow down the cwnd recovery time. In proposed schemes, Snoop takes BS as a pivot point to cache the unacknowledged TCP packets. When errors occur in wireless link, Snoop retransmits the packets locally from BS instead of retransmitting these packets from sender. And Snoop shields off the duplicate ACKs caused by wireless errors to avoid sender triggering unnecessary congestion control. But Snoop adopts same retransmission style as TCP. It only retransmits one packet per continuous duplicate ACKs. Snoop recovers error packets more quickly and tolerates higher BER than TCP. But Snoop doesn¡¦t really solve the degraded performance problem of multiple errors of TCP. When the channel is the in a very bad quality, Snoop still performs badly. We proposed a new scheme, Csnoop (continuous snoop), extended from Snoop. When bursty errors happen in the wireless links, Csnoop retransmits one lost packets from the BS in first RTT and counts the number of ACKs that arrives BS to calculate the number of lost packets. And Csnoop retransmits these lost packets continuously. When local timeout happens, Csnoop infers that all packets were dropped and retransmits all packets cached in the buffer. Simulations show that Csnoop achieves better throughput compared to Snoop and TCP, especially for bad quality wireless links. Furthermore, Csnoop needs less buffer size to cache the unacknowledged packets at the base station than Snoop.

IETF proposes the Differentiated Service(DiffServ) architecture for next-generation QoS networks. The main features of a DiffServ network are high scalability and compatibility. However, lacking of supporting QoS for individual flows becomes a problem. To remedy this problem, we propose an effective TCP transmission (ETCP) scheme based on the characteristics of TCP. The goal of this thesis is to make TCP transmission more effective. Because of the network complexity, we need to modify both the Ingress and Egress nodes. At Ingress node, we measure TCP congestion window(cwnd) and Round Trip Time(RTT), while at Egress node we monitor the throughput and embed a feedback ratio into TCP header of the acknowledge packets. When the throughput is below or over the target throughput, we adjust DSCP dynamically. The adjustment depends on both measured TCP congestion window and RTT. Each DSCP value is mapped with a delay time through Delayed-FIFO. Changing the delay time in Ingress node, we are able to control TCP transmission associated with RTT more effectively. For the purpose of the demonstration, we implement our scheme on Linux platform. We discuss the performance difference between the theoretical and practical results through experiments. Besides, from the implementation, we have shown that our proposed architecture is workable and compatible with today¡¦s network environment.

Most process of lease is done manually in the past. Since the networks were used extensively and systems of Point of Sales risen quickly, the efficiency of administration of payment flow had been obviously promoted. What are mostly encrypted are information of payment flow, but what we encrypt is the information of leased products this time. We apply encryption technique to the lease system, and access information via RS-232 devices, such as SmartCard Reader and EM Reader. After processing of system and encryption, the system will transfer the processed data to the server. In this paper, we take Kaohsiung C-Bike Lease System for case study, our discussion will focus on network packet, data encryption, and real-time monitoring. We will also introduce critical features and mechanisms of the lease system and address some issues we encountered. This system encrypts data via Advanced Encryption Standard, sends out data packets via TCP/IP, and monitors leased bikes via databases and framework of multi-thread.

<p>Denna studie undersöker om, och isåfall hur, nätverksfördröjningar uppstår då små SOAP-meddelanden skickas över TCP/IP. För att utföra detta skapades ett litet nätverk och i det genomfördes mätning av svarstider för två olika SOAP-implementationer, Apache SOAP och Microsoft SOAP Toolkit. Resultaten från mätningarna analyserades och visade att det uppstår nätverksfördröjningar då små SOAP-meddelanden skickas över TCP/IP. Både Apache SOAP och Microsoft SOAP Toolkit uppvisade olika strategier i nätverkskommunikationen vilket påverkade svarstider. För Apache SOAP uppstod fördröjningar vid varje SOAP-anrop, oberoende av storleken på meddelandet som returnerades, medan Microsoft SOAP Toolkit undvek nätverksfördröjningar då meddelandet som returnerades var tillräckligt stort.</p>

<p>Detta examensarbete behandlar problemet med att testa inbyggda system i kontorsmiljö. För att göra detta och därigenom kunna göra anrop på det inbyggda systemets API, måste detta anrop skickas som ett seriellt datapaket över en seriell kommunikationslänk som TCP/IP.</p><p>Detta möjliggjordes genom att först upprätta en kommunikationslänk med protokollet TCP/IP, där användningen av POSIX-sockets tillämpades. För att packa ner och packa upp funktionsanropen till seriell data implementerades ett protokoll som följdes när detta utfördes. Hantering av data i samband med överföring över TCP/IP sköttes av ett protokoll vid namn BGSFP, ett protokoll som bygger på det tidigare protokollet TSFP.</p>

<p>TCP (Transmission Control Protocol) is a transport protocol designed for the wired Internet. In wireless networks packet losses occur more frequently due to the unreliability of the physical link. The main problem is that TCP treats all losses as congestion, which leads to a lower throughput. </p><p>Ad hoc networks are multihop wireless networks of mobile nodes, where each node can allow other packets to pass through it. Topology changes often occur and may lead to packet losses and delays, which TCP misinterprets as congestion. We want to modify TCP to recognize the differences between link failure and congestion to improve the capacity. </p><p>In our model we have built a connection in an ad hoc network where packet losses and partitions can be made. Simulation experiments show that we didn't get the problems we expected. This can be explained by low delays and because we buffered the packets during link failure. </p><p>A simple modification of TCP was made and simulated, and showed that an improvement of performance is possible. More research should be done to make a modification of TCP that would further affect the throughput.</p>

<p>Distribution of information across IP based networks is today part of our everyday life. IP is the backbone of the Internet and most office networks. We use IP to access web pages, listen to radio, and to create computation clusters. All these examples use bandwidth, and bandwidth is a limited resource.</p><p>Many applications distribute the same information to multiple receivers, but in many cases the same information is sent to a single receiver at a time, thus multiple copies of the same information is sent, thus consuming bandwidth.</p><p>What if the information could be broadcasted to all the clients at the same time, similar to a television broadcast. TCP/IP provides some means to do that. For example UDP supports broadcasting; the problem faced when using UDP is that it’s not reliable. There is no guarantee that the information actually reaches the clients.</p><p>This Bachelor thesis in Computer Science aims to investigate the problems and solutions of how to achieve reliable distribution of fixed size data sets using a non reliable multicast communication channel, like UDP, in a LAN environment.</p><p>The thesis defines a protocol (Almost Reliable Multicast Protocol – Arm-P) that provides maximum scalability for delivery of versioned data sets that are designed to work in a LAN-environment. A proof-of-concept application is implemented for testing purposes.</p>

<p>In this thesis we investigate, implement and evaluate a solution for introducing the TransmissionControl Protocol (TCP) into the software of a load generator. The load generator is a simulator usedfor simulating end-user generated activities in the Universal Mobile Telecommunication System(UMTS) network. The purpose of simulating traffic on the network is in this case to verify thefunctionality and robustness of the Radio Network Controller (RNC) node within the UMTSnetwork.TCP is a protocol that provides reliable data transfer over unreliable underlying networkprotocols. It is used as the main transport protocol of the Internet, thus it is also used in the UMTSnetwork in order to provide connectivity for user equipment, such as 3G mobile phones, to servicesover the Internet. For the load generator to be able to produce more realistic traffic scenarios is itdesirable to give it the ability to use TCP.This thesis presents a solution of the problem where an open-source implementation of the TCPfunctionality was chosen, ported to the running platform of the load generator and then tested in asimulated test environment. The choice of the open source implementation of TCP was made byperforming an investigation of available options. In the investigation an open source TCP/IP stackcalled lwIP was chosen. lwIP was then ported to the running platform of the load generator bywrapping and modify the source code. The tests of the ported TCP implementation were made in asimulated test environment with focus on testing basic TCP functionality. The tests showed that theTCP implementation produced provided the basic functionality that was asked for.</p>

In this diploma thesis, the implementation of a networking protocol, which is adopted from a previous work, is refined to a state where it is ready for production use. The protocol is tailored for the specific needs in a cluster environment which uses standard ethernet as the interconnection hardware. Thereby focuses is on developing and implementing a flow control which is very lightweigt in terms of processing overhead and copes the challenges of such a tightly coupled network gracefully. Second, the Log(G)P models for parallel computation are examined and it is shown that they do not give a usable definition of the overhead or latency parameters for protocols whose overhead is O(message size)–instead, the P-LogP model is used to characterize the protocol and for comparisons to the TCP/IP protocol suite. Last, it is demonstrated that the protocol gives usable results with real-world scientific applications.