PGM Reliable Transport Protocol Specification

Reliable multicast is a technology used in a number of applications such as multi-receiver file transfer, streaming media, multimedia conferencing, collaborative processing etc. Pragmatic general multicast (PGM) is a reliable multicast transport protocol that runs over a best effort datagram service. Through this paper we propose an efficient loss recovery scheme for PGM. In our proposal a PGM router that receives a negative acknowledgement tries for local recovery prior to global recovery. The scheme works well when the sender/DLR is located further away from the PGM router than its receivers and the probability that a local receiver caches the packet is high. If there is a hit, the router retrieves the packet from its neighborhood, this results in a decrease in recovery latency, increase in throughput, reduction in power consumption also it alleviates the load at the sender/DLR.

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.

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.

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.

Multicast communication protocols are not immune from failures as a result of packets being dropped due to a broken link or time out processes. Therefore, it is essential to understand how these failures can affect the overall performance of multicast protocols over the Internet. This paper compares the fault-tolerance effect of two reliable multicast protocols: pragmatic general multicast (PGM) and elastic reliable multicast (ERM) in a situation where a multicast-aware node fails and the sub-nodes will have to request a repaired packet. A simulation model is developed in such a way that faults are randomly created on nodes and link for a specified period of time and the fault-tolerance effect on the two multicast protocols is analyzed. The model developed for this paper repeats the simulation for different network size, the results obtained show that the ERM protocol is better than the PGM as the size of the network increases. This finding is key while considering the improvement (or upgrade) of existing multicast protocols. The result is also significant at the early stage of designing new multicast protocols as it provides useful information in allocating scarce resource that can be appropriated to improve other infrastructure in the network.

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.

A large number of network services use IP and reliable transport protocols. For applications with constant pressure of data, loss is handled satisfactorily, even if the application is latency sensitive. For applications with data streams consisting of intermittently sent small packets, however, users experience extreme latencies more frequently. Due to the fact that such thin-stream applications are commonly interactive and time-dependent, increased delay may severely reduce the experienced quality of the service. When TCP is used for thin-stream applications, events of highly increased latency are common, caused by the way retransmissions are handled. Other transport protocols that are deployed in the Internet, like SCTP, model their congestion control and reliability on TCP, as do many frameworks that provide reliability on top of unreliable transport. We have tested several application-and transport layer solutions, and based on our findings, we propose sender-side enhancements that reduce the application-layer latency in a manner that is compatible with unmodified receivers running commodity opeating systems like Linux, *BSD, Mac OSX and Windows. We have implemented the mechanisms as modifications to the Linux kernel, both for TCP and SCTP. The mechanisms are dynamically triggered so that they are only active when the kernel identifies the stream as thin. To evaluate the performance of our modifications, we have conducted a wide range of experiments using replayed thin-stream traces captured from real applications as well as artificially generated thin-stream data patterns. From the experiments, effects on latency, redundancy and fairness were evaluated. The analysis of the performed experiments shows great improvements in latency for thin streams when applying the modifications. Surveys where users evaluate their experience of several applications' quality using the modified transport mechanisms confirmed the improvements seen in the statistical analysis. The positive effects of our modifications were shown to be possible without notable effects on fairness for competing streams. We therefore conclude that it is advisable to handle thin streams separately, using our modifications, when transmitting over reliable protocols to reduce retransmission latency.

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.

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.

In this paper we propose a transport connection model managed by reliable transport protocol with forward error correction (FEC) in the modes of selective and group reject in the form of a discrete-time Markov chain. It takes into account the influence on the transport connection throughput by protocol parameters (window size and acknowledge time-out duration), error level in the communication channels of the data transmission path, round-trip delay, and forward error correction process parameters. We assessed the benefits of the transport protocol with FEC over the classic transport protocol in the multidimensional feature space of influencing variable values.

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.

SRDTP: A reliable datagram transport protocol for distributed computing

Distributed real-time systems call for expressive modeling languages to capture and express their functional and nonfunctional requirements at early stages of the development process. The UML profile for Schedulability, Performance and Time (UML/SPT) is an object-oriented real-time modeling language. UML/SPT has been designed using the built-in extension mechanisms of UML, which makes it flexible and customizable. In this paper, we propose an extension for UML/SPT to capture multicast communications. We define a metamodel that encapsulates the main concepts involved in multicast communications, we show its relationship to UML/SPT domain model, and we introduce new stereotypes corresponding to these concepts. We illustrate the extension with the modeling of the Reliable Multicast Transport Protocol (RMTP2). Finally, we compare our approach to extend UML/SPT for multicast communications with an extension for MSC having the same purpose.