Traditional reliable unicast protocols (e.g., TCP), known as sender-initiated schemes, do not scale well for one-to-many reliable multicast due mainly to implosion losses caused by excessive rate of feedback packets arriving from receivers. So, recent multicast protocols have been devised following the receiver- initiated approach: scalability (in terms of control traffic, protocol state and end-systems processing requirements) is achieved by making the sender independent from receivers; the sender does not know the membership of the destination group. However, this comes with a cost: the lack of knowledge about and control of receivers at the sender has negative implications with respect to throughput, network cost (bandwidth required), and degree of reliability offered to applications. This thesis follows an alternative approach: instead of adopting the receiver-initiated scheme, it greatly enhances the scalability of the sender-initiated scheme, by means of polling-based feedback and hierarchy. The resulting protocol is named PRMP: polling-based Reliable Multicast protocol. Its unique implosion avoidance mechanism polls receivers at carefully planned timing instants achieving a low and uniformly distributed rate of feedback packets. The sender retains controls of receivers: the main PRMP mechanisms are based on a one-to-many sliding window mechanism, which efficiently and elegantly extends the abstraction from reliable unicasting to reliable multicasting. The error control mechanism of PRMP incorporates the use of NACKs and selective, cumulative acknowledgment of packets; additionally, it can wait and judiciously decide between multicast and selective unicast retransmissions. The flow control mechanism prevents unnecessary losses caused by the overrunning of receivers, despite variations in round-trip times and application speeds. The scalability provided by the polling mechanism is further extended by an hierarchic organization to exploit distributed processing and local recovery: receivers are organized according to a tree-structure. However, unlike other tree-based protocols, PRMP is "fully-hierarchic": each parent node forwards data via multicast to its children, and retains/explores the control of and knowledge about its children while autonomously applying error, flow, congestion and session controls in the communication with them. Two congestion control mechanisms, one window-based and another rate-based, have been incorporated to PRMP. As shown through simulation experiments, the resulting protocol q,chieves high though put with cost- effective reliable multicasting. They also show the scalability and effectiveness of PRMP mechanisms. PRMP can achieve reliable multicast with the same kind of reliability guarantees provided by TCP but without incurring prohibitive costs in terms of network cost or recovery latency found in other protocols.

Please read the abstract in the section “front” of this document / Dissertation (MSc (Applied Science))--University of Pretoria, 2007. / Electrical, Electronic and Computer Engineering / MSc / unrestricted

In this thesis, a new transport protocol, the Concurrent Multi-Path Real-time Transmission Control Protocol (cmpRTCP) is proposed. The proposed protocol has been designed to handle real-time streams (video and audio) over IP-networks. One of the key strengths of this protocol lies in its ability to intelligently exploit the availability of multiple paths between multi-homed hosts for concurrent transmission of unicast real-time streams. This work describes the architecture and operation of cmpRTCP in detail. In addition, the limitations of currently used transport protocols in handling real-time streams are also discussed. These limitations of other protocols have played a vital role in the design process of the proposed protocol. Experiments to evaluate the performance of cmpRTCP against other protocols and the results obtained therein are also documented in this work. Results show that cmpRTCP is a best effort protocol that tries to maximize the amount of data that is successfully delivered to the destination in a timely manner under varying drop and delay conditions of the network.

<p>The emergence of real-time applications, like Voice over IP and video conferencing, in IP networks implies a challenge to the underlying infrastructure. Several real-time applications have requirements on timeliness as well as on reliability and are accompanied by signaling applications to set up, tear down and control the media sessions. Since neither of the traditional transport protocols responsible for end-to-end transfer of messages was found suitable for signaling traffic, the Stream Control Transmission Protocol (SCTP) was standardized. The focus for the protocol was initially on telephony signaling applications, but it was later widened to serve as a general purpose transport protocol. One major new feature to enhance robustness in SCTP is multihoming, which enables for more than one path within the same association.</p><p>In this thesis we evaluate some of the mechanisms affecting transmission performance in case of a switchover between paths in a multihomed SCTP session. The major part of the evaluation concerns a failure situation, where the current path is broken. In case of failure, the endpoint does not get an explicit notification, but has to react upon missing acknowledgements. The challenge is to distinguish path failure from temporary congestion to decide  when to switch to an alternate path. A too fast switchover may be spurious, which could reduce transmission performance, while a too late switchover also results in reduced transmission performance. This implies a tradeoff which involves several protocol as well as network parameters and we elaborate among these to give a coherent view of the parameters and their interaction. Further, we present a recommendation on how to tune the parameters to meet  telephony signaling requirements, still without violating fairness to other traffic.</p><p>We also consider another angle of switchover performance, the startup on the alternate path. Since the available capacity is usually unknown to the sender, the transmission on a new path is started at a low rate and then increased as acknowledgements of successful transmissions return. In case of switchover in the middle of a media session the startup phase after a switchover could cause problems to the application. In multihomed SCTP the availability of the alternate path makes it feasible for the end-host to estimate the available capacity on the alternate path prior to the switchover. Thus, it would be possible to implement a more efficient startup scheme. In this thesis we combine different switchover scenarios with relevant traffic. For these combinations, we analytically evaluate and quantify the potential performance gain from utilizing an ideal startup mechanism as compared to the traditional startup procedure.</p>

When used in a wireless mesh network, TCP has shortcomings in the areas of throughput and fairness among traffic flows. Several methods have been proposed to deal with TCP's weakness in a wireless mesh, but most have been evaluated with simulations rather than experimentally. We evaluate several major enhancements to TCP – pacing, conservative windows, and delayed ACKs – to determine whether they improve performance or fairness in a mesh network operating in the BYU Computer Science building. We also draw conclusions about the effectiveness of wireless network simulators based on the accuracy of reported simulation results.

Low power and Lossy Networks (LLNs) such as wireless sensor networks are currently used in many important applications fields such as remote environment monitoring and target tracking. This deployment has been enabled by the availability, especially in recent years, of embedded micro-controller devices that are smaller and cheaper. These devices are equipped with wireless interfaces, with which they can communicate with each other to form a network. In this thesis we focus on studying the energy consumption of reliable transport protocols over LLNs. Recently, much research has been carried out to improve the reliability and the congestion control on low power networks. Some of these works have considered TCP inappropriate for this kind of networks. Indeed, the idea of deploying TCP was rejected due to its header overhead, its end-to-end retransmission mechanism, its large rate of acknowledgment, and the impact of the lower layers fragmentation on the energy consumption. Nonetheless, the use of standard TCP/IP protocols offers the advantage of a seamless connectivity between the wireless network and the Internet. TCP allows easily the use of standard applications (HTTP, SSH) for some tasks like reprogramming of nodes or firmware updates, without the need of deploying complex proxies in border routers. In the first part of this work, we study the energy consumption of TCP and the ways that reduce its energy consumption. We study one of the proposed TCP algorithms to reduce the end-to-end retransmissions cost and we propose some improvements that allow it to reduce the energy consumption. Then, we study the compression of the TCP header over low-power and lossy networks and we consider IPv6 over Low power Wireless Personnel Area Networks (6LoWPAN) as an example. We propose a new TCP header compression algorithm that reduces the TCP header size to about six bytes. In the second part, we propose a mathematical model that allows to estimate the energy consumption of wireless nodes. Using the model, we study the tradeoff between sending long and short TCP segments and their impact on the energy consumption. Finally, we study the impact of a new fragment recovery mechanism on the energy performance of TCP.

<p>Due to the cheap and flexible framework that the underlying IP-technology of the internet provides, IP-networks are becoming popular in more and more contexts. For instance, telecommunication operators have started to replace the fixed legacy telephony networks with IP-networks. To support a smooth transition towards IP-networks, the Stream Control Transmission Protocol (SCTP) was standardized. SCTP is used to carry telephony signaling traffic, and solves a number of problems that would have followed from using the Transmission Control Protocol (TCP) in this context. However, the design of SCTP is still heavily influenced by TCP. In fact, many protocol mechansisms in SCTP are directly inherited from TCP. Unfortunately, many of these mechanisms are not adapted to the kind of traffic that SCTP is intended to transport: time critical message-based traffic, e.g. telephony signaling.In this thesis we examine, and adapt some of SCTP's mechanisms to more efficiently transport time critical message-based traffic. More specifically, we adapt SCTP's loss recovery and message bundling for timely message transfers. First, we propose and experimentally evaluate two loss recovery mechanisms: a packet-based Early Retransmit algorithm, and a modified retransmission timeout management algorithm. We show that these enhancements can reduce loss recovery times with at least 30-50%, in some scenarios. In addition, we adapt the message bundling of SCTP to better support timely message delivery. The proposed bundling algorithm can in some situations reduce the transfer time of a message with up to 70%.In addition to these proposals we have also indentified and reported mistakes in some of the most popular SCTP implementations. Furthermore, we have continously developed the network emulation software KauNet to support our experimental evaluations.</p>

As multimedia processing and networking technologies, products and services evolve, the number of users communicating, collaborating and entertaining over the IP networks is growing rapidly. With the emergence of pervasive and ubiquitous multimedia services, this proliferation creates an abundant increase in the amount of the Internet backbone traffic. This brings the problem of efficient transmission of real-time and time-sensitive media content to the fore. Effective multimedia services demand appropriate application-specific and media-aware solutions, without which the full benefits of such services will not be realized. Poor approaches often lead to system performance degradations such as unacceptable presentation quality perceived by the users, possible network collapses due to the high-bandwidth nature of the multimedia applications, and poor performance observed by other data-oriented applications due to the unresponsiveness of multimedia flows. From a networking perspective, traditional approaches consider the application data as "sacred" and do not differentiate any part of it from the rest. While this keeps the data-delivery mechanisms, namely, the transport-layer protocols, as plain as possible, it also precludes these mechanisms from interpreting the media content and tailoring their actions according to the importance of the content. Given that this naive approach cannot satisfy the specific needs of each and every one of the today's emerging applications ranging from videotelephony to video-on-demand, from distance education to telemedicine, from remote surveillance to online video gaming, the study of Multimedia Transport Protocols (MMTP) is overdue. An MMTP solution basically integrates the multimedia content information into the responsible data-delivery mechanisms along with the requirements of the invoking application and network characteristics to deliver the highest level of service quality. In other words, an MMTP solution offers a unified environment where all cooperating protocol components interact with each other and make the best use of this collaboration to fulfill their respective duties. The focus of this thesis is on the design and evaluation of a set of end-to-end and system-level MMTP solutions for scalable, reliable, and high quality multimedia services in ever-changing, complex and heterogeneous computing and communication environments.

Due to the high availability of cheap hardware, wireless multi-hop networks and in particular Wireless Mesh Networks (WMNs) are becoming popular in more and more contexts. For instance, IEEE 802.11 based WMNs have already started to be deployed as means to provide Internet access to rural areas in the developing world. To lower the cost and increase the coverage in such deployments, the wired network is extended with a wireless backbone of fixed mesh routers. With advances in technology and reduction in price comes also the possibility for more powerful wireless nodes, having multiple radios that allow transmitting on different channels in parallel. To be a successful platform for providing general Internet access, wireless multi-hop networks must provide support for common Internet applications. As most of the applications in the Internet today use the Transmission Control Protocol (TCP), TCP performance is crucial. Unfortunately, the design of TCP’s congestion control that made it successful in today’s Internet makes it perform less than optimal in wireless multi-hop networks. This is due to, among others, TCP’s inability to distinguish wireless losses from congestion losses. The current trend for operating system designers is also to focus TCP development on high-speed fixed networks, rather than on wireless multi-hop networks. To enable wireless multi hop networks as a successful platform there is therefore a need to provide good performance using TCP variants commonly deployed in the Internet. In this thesis, we develop novel proposals for the network layer in wireless multi-hop networks to support TCP traffic more efficiently. As an initial study, we experimentally evaluate different TCP variants, with and without mobile nodes, in a MANET context. Our results show that TCP Vegas, which does not provoke packet loss to determine available bandwidth, reduces the stress on the network while still providing the same or slightly increased performance, compared to TCP Newreno. We further propose and evaluate packet aggregation combined with aggregation aware multi-path forwarding to better utilize the available bandwidth. IP layer packet aggregation, where small packets are combined to larger ones before sent to the link layer, has been shown to improve the performance in wireless multi-hop networks for UDP and small packet transfers. Only few studies have been made on the impact of packet aggregation on TCP traffic, despite the fact that TCP traffic constitutes the majority of the Internet traffic. We propose a novel aggregation algorithm that is specifically addressing TCP relevant issues like packet reordering, fairness and TCP timeouts. In a typical WMN scenario, the aggregation algorithm increases TCP performance by up to 70 % and decreases round trip time (RTT) by up to 40 %. A detailed evaluation of packet aggregation in a multi radio setting has shown that a naive combination of multi path routing and packet aggregation can cause valuable aggregation opportunities to be lost. Therefore, we propose a novel combined packet aggregation and aggregation aware forwarding strategy that can reduce delay, packet loss and increase TCP performance by around 30 %.

Due to the cheap and flexible framework that the underlying IP-technology of the internet provides, IP-networks are becoming popular in more and more contexts. For instance, telecommunication operators have started to replace the fixed legacy telephony networks with IP-networks. To support a smooth transition towards IP-networks, the Stream Control Transmission Protocol (SCTP) was standardized. SCTP is used to carry telephony signaling traffic, and solves a number of problems that would have followed from using the Transmission Control Protocol (TCP) in this context. However, the design of SCTP is still heavily influenced by TCP. In fact, many protocol mechansisms in SCTP are directly inherited from TCP. Unfortunately, many of these mechanisms are not adapted to the kind of traffic that SCTP is intended to transport: time critical message-based traffic, e.g. telephony signaling.In this thesis we examine, and adapt some of SCTP's mechanisms to more efficiently transport time critical message-based traffic. More specifically, we adapt SCTP's loss recovery and message bundling for timely message transfers. First, we propose and experimentally evaluate two loss recovery mechanisms: a packet-based Early Retransmit algorithm, and a modified retransmission timeout management algorithm. We show that these enhancements can reduce loss recovery times with at least 30-50%, in some scenarios. In addition, we adapt the message bundling of SCTP to better support timely message delivery. The proposed bundling algorithm can in some situations reduce the transfer time of a message with up to 70%.In addition to these proposals we have also indentified and reported mistakes in some of the most popular SCTP implementations. Furthermore, we have continously developed the network emulation software KauNet to support our experimental evaluations.