Improving latency for interactive, thin-stream applications over reliable transport

Performance and adaptability of the transport protocol are among the most critical parts of the TCP/IP protocol stack since these factors are directly related to the user's interaction and application experience. In this work we analyze specific problems of a reliable transport protocol's performance caused by rapidly changing topology in MANETs in addition to classical transport layer challenges. Then we propose the Wireless Transport Protocol (WTP) - a new lightweight, bandwidth-efficient, and reliable cross-layer transport protocol capable of adjusting its performance to suit highly dynamic environments, such as MANETs. Experimental results show that simplifying the protocol reduces its processing, while cross-layer optimization enables more precise adjustments to network and connection fluctuations, hence increasing transport protocol efficiency by up to 9% under different conditions as compared to standard TCP protocol.

This paper introduces a reliable multicast transport protocol RCCMP (reliable congestion controlled multicast protocol) that has been designed to be simple, scalable, reliable and TCP-friendly. The congestion control is a central part of the protocol, where the feedback of the worst receiver is used to control a transmission window in a TCP-like fashion. The scalability issue is addressed with an exponential timers scheme, that is also used to estimate the number of receivers involved in the communication. The protocol neither needs support from network elements nor maintains state information dependent of the number of receivers. The paper presents a number of simulation results on its performance. Besides, some design choices are evaluated.

This paper introduces a XCP framework based reliable multicast transport protocol XRMCCP (XCP framework based Reliable Multicast Congestion controlled protocol) that has been designed to be simple, scalable and reliable for high bandwidth-delay product (BDP) networks. We observe that TCP Friendly Throughput Equation (TFTE), which many current multicast transport protocols use to estimate available rates of multicast sessions, has limits of performance in high BDP networks. XRMCCP generalizes XCP framework to support one-to-many transport congestion control. The scalability issue is addressed with an exponential timers scheme that is also used to estimate the number of receivers involved in the communication. The paper presents a number of simulation results on its performance. Besides, some design choices are evaluated.KeywordsCongestion ControlRound Trip TimeCongestion WindowFairness IndexBottleneck LinkThese keywords were added by machine and not by the authors. This process is experimental and the keywords may be updated as the learning algorithm improves.

New wireless sensor network applications (e.g., military surveillance) require higher reliability than a simple best effort service could provide. Classical reliable transport protocols like Transmission Control Protocol (TCP) are not well suited for wireless sensor networks due to both the characteristics of the network nodes (low computing power, strong energy constraints) and those of the main applications running on those nodes (low data rates). Recent researches present new transport protocols for wireless sensor networks providing various type of reliability and using new mechanisms for loss detection and recovery, and congestion control. This paper presents a survey on reliable transport protocol for WSNs.

This chapter presents a distributed program that implements the reliable transport service (Chap. 20) over an LRD (loss, reorder, duplication) channel, as illustrated in Fig. 21.1. In computer networking, such a program is referred to as a transport protocol. Following this terminology, we refer to the system at each address as a “tp” system, where “tp” is short for transport protocol. We refer to the messages exchanged between tp users as “data blocks”, and reserve the term “messages” for the messages exchanged between the tp systems; the former would be sent inside the latter.

Pragmatic General Multicast (PGM) is a reliable multicast transport protocol for applications that require ordered or unordered, duplicate-free, multicast data delivery from multiple sources to multiple receivers. PGM guarantees that a receiver in the group either receives all data packets from transmissions and repairs, or is able to detect unrecoverable data packet loss. PGM is

Consistent data delivery in constrained networks such as wireless sensor networks (WSN), which has a significant role in Internet of Things (IoT), is vital due to the high probability of packet loss, not only because of the unreliable channel but also from link and buffer congestions. Thus, a reliable transport protocol should be used. Data caching at intermediate nodes is one way of improving the reliability performance of transport protocols in WSNs. However, it is not yet known up to what extent data caching alone can alleviate congestion in WSNs. This study investigates and analyses how a cache-based transport protocol, that has no congestion control mechanism and has a fixed transmission window, performs under congestion states. The transport protocol, called DTSN, is evaluated under link and buffer congestion states and the optimal performance is determined in terms of goodput, transmission cost and cache hit metrics. The results of our analysis reveal that intermediate caching was able to perform significantly during congestions in WSN. However, the amount of traffic being sent and intermediate node's cache memory should be taken into account. The optimum caching performance during link contentions is reached at higher transmission window and cache size values, while for buffer congestions, it is achieved at lower transmission window but higher cache size values, respectively. These values obtained can guarantee optimum usage of cache while ensuring congestion avoidance. In addition, the correlation that is established can serve as a basis for developing congestion window management for transport protocols in WSN that employ data caching.

The last few years have seen a great number of new network applications. Many of these applications are characterized by their multipoint features and by their needs of high bandwidth. To allow the deployment of these applications at a large scale in the Internet without running the risk of network implosion, an adapted congestion control mechanism has to be used. In this paper we propose to use the Fair Scheduler paradigm for end-to-end congestion control for the design of a congestion control algorithm to be used with the Reliable Multicast Transport Protocol RMTP. The FS paradigm supposes that all the network routers use a fair scheduler queueing mechanism. This hypotheses allowed us to have worst case bounds on the queueing delay that a block of RMTP data packets experiences between the sender and any receiver in the multicast group. Based on these bounds the sender will regulate its transmission rate in order to avoid a network congestion. Two different versions of this algorithm are presented in detail. The coexistence RMTP/TCP will be the issue of a suite of tests that intend to evaluate the performance of the new algorithm over a Fair Queueing (FQ) scheduled network with the presence of TCP flows sharing the same network resource as the RMTP flow. The same tests will be rerun over a FIFO scheduled network in order to have an idea about the advantages of using the scheduling discipline as a network support for congestion control in the Internet.

We analyze the maximum throughput that known classes of reliable multicast transport protocols can attain. A new taxonomy of reliable multicast transport protocols is introduced based on the premise that the mechanisms used to release data at the source after correct delivery should be decoupled from the mechanisms used to pace the transmission of data and to effect error recovery. Receiver-initiated protocols, which are based entirely on negative acknowledgments (NAKS) sent from the receivers to the sender, have been proposed to avoid the implosion of acknowledgements (ACKS) to the source. However, these protocols are shown to require infinite buffers in order to prevent deadlocks. Two other solutions to the ACK-implosion problem are tree-based protocols and ring-based protocols. The first organize the receivers in a tree and send ACKS along the tree; the latter send ACKS to the sender along a ring of receivers. These two classes of protocols are shown to operate correctly with finite buffers. It is shown that tree-based protocols constitute the most scalable class of all reliable multicast protocols proposed to date.

For IoT applications that require near real-time wireless connections, the traditional TCP protocol is not adept enough due to the size of its packet header, its conservative Additive Increase Multiplicative Decrease (AIMD) congestion control algorithm and its slow start strategy. This paper mainly evaluates three kinds of reliable UDP-based transport protocols namely, RUBDP, UDT, and PA-UDP, for use in IoT applications. The evaluation was done on a test bed that consists of two machines connected by wireless ad-hoc network. We consider the effect of data size by monitoring throughput of the implemented protocol. This paper also discusses the CPU usage on the sender and receiver side of each protocol. The paper findings are, PA-UDP protocol has the best throughput performance but requires high CPU utilization during the start and end of transmission at the receiver side. Meanwhile, UDT protocol is most efficient in term of CPU usage although it provides lesser throughput. RBUDP, on the other hand, performed consistently in wireless network but CPU usage is very high at the sender side. The conclusion drawn on this paper is UDT protocol is the most suitable protocol of the three for IoT applications because it has consistent performance and is CPU friendly.

This work presents new experimental results examining the loss tolerant unicast congestion control behavior of the NACK-Oriented Reliable Multicast (NORM) transport protocol documented initially as Internet Proposed Standard, Request for Comments (RFC) 5740. While RFC 5740 specifies a reliable multicast UDP-based transport protocol, NORM also has the ability to provide loss tolerant UDP unicast reliable transport that is well-suited for lossy wireless environments where protocols like TCP may not perform well. In addition, we introduce the design and use of a unicast SOCKS proxy for the NORM protocol called the NACK-Oriented Proxy (NORP) and validate the results of NORP performance against other NORM unicast experimental results. The availability of NORP will reduce the transition risk and improve the use of NORM unicast capabilities by not requiring end system application modifications. Our results examine a number of unicast congestion control variants of the NORM protocol, including an explicit congestion notification (ECN) mode, within an emulated lossy link modeled network environment. We include COTS hardware routing and radio-to-router interface (R2RI) within the test environment to examine and validate the use and performance of ECN-based packet marking and improved cross-layer flow control when available. The initial experiments demonstrate the significant advantage of using ECN-based assistance within error-prone wireless environments. We also show increased performance of an additional loss tolerant mode that does not require ECN infrastructure support. Coexistent transport fairness results show that NORM unicast modes are friendly to both inter-protocol flows (e.g., TCP flows coexisting) and to intra-protocol flows (e.g., Other NORM flows coexisting). We conclude by summarizing findings and discussing future work planned.

In wireless multimedia sensor networks (WMSNs) a sensor node may have different types of sensor which gather different kinds of data. To support quality of service (QoS) requirements for multimedia applications having a reliable and fair transport protocol is necessary. One of the main objectives of the transport layer in WMSNs is congestion control. We observe that the information provided may have different levels of importance and argue that sensor networks should be willing to spend more resources in disseminating packets carrying more important information. In WMSN event-driven applications, it is critical to report the detected events in the area, resulting in sudden bursts of traffic due to occurrence of spatially-correlated or multiple events, causing loss of data. Also, nodes have very limited power due to hardware constraints. Packet losses and retransmissions resulting from congestion, cost precious energy and shorten the lifetime of sensor nodes. Till now, in WMSNs, Congestion control techniques are based on detection of congestion and recovery, but they cannot eliminate or prevent the occurrence of congestion. Collision is a symptom of congestion in the wireless channel and can result in a time-variant channel capacity. The main intention of this protocol is to be used as a mechanism for reducing congestion in the network by free resources to set accurate rates and priority data needs. If two nodes send their packets in the shortest path to the parent node in a crowded place, a source node must prioritize the data and uses data that have lower priorities of a suitable detour nodes consisting of low or non- active consciously. Proposed algorithm is applied to the nodes near the base station (which convey more traffic) after the congestion detection mechanism detected the congestion. The results obtained from simulation test done by NS-2 simulator indicate that the proposed model is more innovative and presents better performance in compare with CCF, PCCP and DCCP protocols.

Scalability is of paramount importance in the design of reliable multicast transport protocols, and requires careful consideration of a number of problems such as feedback implosion, retransmission scoping, distributed loss recovery, and congestion control. In this article, we present a reliable multicast architecture that invokes active services at strategic locations inside the network to comprehensively address these challenges. Active services provide the ability to quickly and efficiently recover from loss at the point of loss. They also exploit the physical hierarchy for feedback aggregation and effective retransmission scoping with minimal router support. We present two protocols, one for packet loss recovery and another for congestion control, and describe an experimental testbed where these have been implemented. Analytical and experimental results are used to demonstrate that the active services architecture improves resource usage, reduces latency for loss recovery, and provides TCP-friendly congestion control.

A lot of research on mobile ad hoc networks has focused on the development of routing and transport protocols. Specifically, this research is focused on the performance of TCP over wireless mobile ad hoc networks. Reliable transport protocols such as TCP are tuned to perform well in traditional networks where packet losses occur mostly due to congestion. However, networks in wireless links perform poorly and suffer from losses due to reasons such as bit error and handoffs. TCP responds to all losses by invoking congestion control and avoidance algorithms resulting in degraded end-to-end performance in wireless systems. The paper sets out the basis for research into the design and development of a new algorithm that is integrated with Mobile IPv6 in order to be implemented in future needs, and thus improve the performance of TCP in such networks. It is proposed to use OMNET++ as the simulation tool to compare the performance of TCP using different approaches with that of the general TCP.

Existing works have approached the problem of reliable transport in ad hoc networks by proposing mechanisms to improve TCP's performance over such networks, In this paper, we show through detailed arguments and simulations that several of the design elements in TCP are fundamentally inappropriate for the unique characteristics of ad hoc networks. Given that ad hoc networks are typically stand-alone, we approach the problem of reliable transport from the perspective that it is justifiable to develop an entirely new transport protocol that is not a variant of TCP. Toward this end, we present a new reliable transport layer protocol for ad hoc networks called ATP (ad hoc transport protocol). We show through ns2-based simulations that ATP outperforms default TCP as well as TCP-ELFN and ATCP.