A throughput analysis of reliable multicast transport protocols

Tree-based reliable multicast protocols are well-known for being scalable for one-to-many sessions, but their behaviour in terms of throughput and bandwidth requirements has not been identifed for many-to-many sessions. In this paper, existing tree-based reliable multicast protocols are categorised into four classes according to their feedback/retransmission mechanisms. Analysing and comparing them using a new spatial multicast loss model that considers the correlation of packet loss events can yield realistic analysis results. The analysis reveals that for many-to-many sessions, tree-based protocols with unicast NACK, unicast retransmission, and periodic polling are most scalable with respect to throughput and network bandwidth for practical session sizes. The result is contrary to the one-to-many case and it is shown that the reason is that reducing the sender's processing time is a key factor for scaling one-to-many reliable multicast, whereas reducing the receiver's processing time is far more important factor for scalability of many-to-many reliable multicasts.

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.

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.

Even though tree-based reliable multicast protocols are known to be most scalable for one-to-many sessions, there is still an open question whether these protocols are also scalable for many-to-many sessions. In this paper, we analyze and compare two promising multicast protocols - the receiver-initiated protocol with NACK suppression and the tree-based protocol - using a new spatial loss model. The proposed model considers the correlation of packet loss events for more realistic analysis unlike the previous work. The analysis results show that the tree-based protocol achieves much higher throughput than the receiver-initiated protocol for a many-to-many session as the number of participants in the session becomes larger.

SRDTP: A reliable datagram transport protocol for distributed computing

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.

This paper contains a detailed description of the congestion control algorithm of TRAM, a tree-based reliable multicast protocol. This algorithm takes advantage of regular acknowledgements from the receivers that propagate back to the sender via the repair tree. This scalable feedback mechanism is used to collect receiver credits. Complementing the windowing mechanism, packet transmission is smoothed by using a data rate commensurate with the window size. Additional details, such as how to prune slow receivers, and how to implement the rate scheduler on non-real-time systems are also discussed. The performance of the congestion control algorithm is then evaluated in extended LANs, and wide area networks. The fairness of bandwidth-sharing with other (TCP) traffic is also evaluated.

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

This paper presents the design of a reliable multicast transport protocol. The aim of the protocol is to provide a service equivalent to a sequence of reliable sequential unicasts between a client and a number of servers, whilst using the broadcast nature of some networks to reduce both the number of packets transmitted and the overall time needed to collect replies.

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.

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.

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.

We explore the performance of reliable data communication in mobile computing environments. Motion across wireless cell boundaries causes increased delays and packet losses while the network learns how to route data to a host's new location. Reliable transport protocols like TCP interpret these delays and losses as signs of network congestion. They consequently throttle their transmissions, further degrading performance. We quantify this degradation through measurements of protocol behavior in a wireless networking testbed. We show how current TCP implementations introduce unacceptably long pauses in communication during cellular handoffs (800 ms and longer), and propose an end-to-end fast retransmission scheme that can reduce these pauses to levels more suitable for human interaction (200 ms). Our work makes clear the need for reliable transport protocols to differentiate between motion-related and congestion-related packet losses and suggests how to adapt these protocols to perform better in mobile computing environments.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Reliable end-to-end delivery service is one of the most important issues for wireless sensor networks in large-scale deployments. In this paper, a reliable data transport protocol, called the Data Forwarding Protocol (DFP), is proposed to improve the end-to-end delivery rate with minimum transport overhead for recovering from data loss in large-scale wireless sensor environments consisting of low speed mobile sensor nodes. The key idea behind this protocol is the establishment of multi-split connection on an end-to-end route, through the Agent Host (AH), which plays the role of a virtual source or a sink node. In addition, DFP applies the local error control and the local flow control mechanisms to multi-split connections, according to network state. Extensive simulations are carried out via ns-2 simulator. The simulation results demonstrate that DFP not only provide up to 30% more reliable data delivery, but also reduces the number of retransmission generated by data loss, compared with the TCP-like end-to-end approach.

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.