A Heuristic Buffer Management and Retransmission Control Scheme for Tree‐Based Reliable Multicast

We propose a heuristic buffer management scheme that uses both positive and negative acknowledgments to provide scalability and reliability. Under our scheme, most receiver nodes only send negative acknowledgments to their repair nodes to request packet retransmissions while some representative nodes also send positive acknowledgments to indicate which packets can be discarded from the repair node's buffer. Our scheme provides scalability because it significantly reduces the number of feedbacks sent by the receiver nodes. In addition, it provides fast recovery of transmission errors since the packets requested from the receiver nodes are almost always available in their buffers. Our scheme also reduces the number of additional retransmissions from the original sender node or upstream repair nodes. These features satisfy the original goal of tree‐based protocols since most packet retransmissions are performed within a local group.     

Reliable Multicast in Multi-Access Wireless LANs

Multicast is an efficient paradigm for transmitting data from a sender to a group of receivers. In this paper, we focus on multicast in single channel multi-access wireless local area networks (LANs) comprising several small cells. In such a system, a receiver cannot correctly receive a packet if two or more packets are sent to it at the same time, because the packets collide. Therefore, one has to ensure that only one node sends at a time. We look at two important issues. First, we consider the problem of the sender acquiring the multi-access channel for multicast transmission. Second, for reliable multicast in each cell of the wireless LAN, we examine ARQ-based approaches. The second issue is important because the wireless link error rates can be very high.We present a new approach to overcome the problem of feedback collision in single channel multi-access wireless LANs, both for the purpose of acquiring the channel and for reliability. Our approach involves the election of one of the multicast group members (receivers) as a leader or representative for the purpose of sending feedback to the sender. For reliable multicast, on erroneous reception of a packet, the leader does not send an acknowledgment, prompting a retransmission. On erroneous reception of the packet at receivers other than the leader, our protocol allows negative acknowledgments from these receivers to collide with the acknowledgment from the leader, thus destroying the acknowledgment and prompting the sender to retransmit the packet.Using analytical models, we demonstrate that the leader-based protocol exhibits higher throughput in comparison to two other protocols which use traditional delayed feedback-based probabilistic methods. Last, we present a simple scheme for leader election.     

A new TCP for persistent packet reordering

Most standard implementations of TCP perform poorly when packets are reordered. In this paper, we propose a new version of TCP that maintains high throughput when reordering occurs and yet, when packet reordering does not occur, is friendly to other versions of TCP. The proposed TCP variant, or TCP-PR, does not rely on duplicate acknowledgments to detect a packet loss. Instead, timers are maintained to keep track of how long ago a packet was transmitted. In case the corresponding acknowledgment has not yet arrived and the elapsed time since the packet was sent is larger than a given threshold, the packet is assumed lost. Because TCP-PR does not rely on duplicate acknowledgments, packet reordering (including out-or-order acknowledgments) has no effect on TCP-PR's performance. Through extensive simulations, we show that TCP-PR performs consistently better than existing mechanisms that try to make TCP more robust to packet reordering. In the case that packets are not reordered, we verify that TCP-PR maintains the same throughput as typical implementations of TCP (specifically, TCP-SACK) and shares network resources fairly. Furthermore, TCP-PR only requires changes to the TCP sender side making it easier to deploy.     

Scalable reliable multicast using multiple multicast channels

We examine an approach for providing reliable, scalable multicast communication, involving the use of multiple multicast channels for reducing receiver processing costs and reducing network bandwidth consumption in a multicast session. In this approach a single multicast channel is used for the original transmission of packets. Retransmissions of packets are done on separate multicast channels, which receivers dynamically join and leave. We first show that protocols using an infinite number of multicast channels incur much less processing overhead at the receivers compared to protocols that use only a single multicast channel. This is due to the fact that receivers do not receive retransmissions of packets they have already received correctly. Next, we derive the number of unwanted redundant packets at a receiver due to using only a finite number of multicast channels, for a specific negative acknowledgment (NAK)-based protocol. We then explore the minimum number of multicast channels required to keep the cost of processing unwanted packets to a sufficiently low value. For an application consisting of a single sender transmitting reliably to many receivers we find that only a small number of multicast channels are required for a wide range of system parameters. In the case of an application where all participants simultaneously act as both senders and receivers a moderate number of multicast channels is needed. Finally, we present two mechanisms for implementing multiple multicast channels, one using multiple IP multicast groups and the other using additional router support for selective packet forwarding. We discuss the impact of both mechanisms on performance in terms of end-host and network resources     

AIRMAIL: A link-layer protocol for wireless networks

This paper describes the design and performance of a link-layer protocol for indoor and outdoor wireless networks. The protocol is asymmetric to reduce the processing load at the mobile, reliability is established by a combination of automatic repeat request and forward error correction, and link-layer packets are transferred appropriately during handoffs. The protocol is namedAIRMAIL (AsymmetrIc Reliable Mobile Access In Link-layer). The asymmetry is needed in the design because the mobile terminals have limited power and smaller processing capability than the base stations. The key ideas in the asymmetric protocol design consist of placing bulk of the intelligence in the base station as opposed to placing it symmetrically, in requiring the mobile terminal to combine several acknowledgments into a single acknowledgment to conserve power, and in designing the base stations to send periodic status messages, while making the acknowledgment from the mobile terminal eventdriven. The asymmetry in the protocol design results in a one-third reduction of compiled code. The forward error correction technique incorporates three levels of channel coding which interact adaptively. The motivation for using a combination of forward error correction and link-layer retransmissions is to obtain better performance in terms of end-to-end throughput and latency by correcting errors in an unreliable wireless channel in addition to end-to-end correction rather than by correcting errors only by end-to-end retransmissions. The coding overhead is changed adaptively so that bandwidth expansion due to forward error correction is minimized. Integrity of the link during handoffs (in the face of mobility) is handled by window management and state transfer. The protocol has been implemented. Experimental performance results based on the implementation are presented.     

On the throughput of multicasting with incremental forward error correction

In this paper, we consider a multicasting model that uses incremental forward error correction (FEC). In this model, there is one sender and r/sup n/ receivers. The sender uses an ideal (n,n(1-p),np) FEC code to code a group of n(1-p) data packets with additional np redundant packets so that any set of n(1-p) packets received by a receiver can be used to recover the original n(1-p) data packets. Packets to the receivers are lost independently with probability q. For this model, we prove several strong laws of large numbers for the asymptotic throughput as n /spl rarr/ /spl infin/. The asymptotic throughput is characterized by the unique solution of an equation in terms of p, q, and r. These strong laws not only provide theoretical justification for several important observations made in the literature, but also provide insights that might have impact on future design of multicasting protocols.     

Improving the optimum generalised stop-and-wait ARQ scheme

Instead of waiting for the acknowledgments from all the copies of a single data block sent, as in the optimum generalised stop-and-wait ARQ scheme, the transmitter in the proposed scheme starts sending an optimum number of copies of the next block in the queue, soon after receiving the positive acknowledgment from the receiver, thereby further improving the throughput efficiency.     

A comparison of sender-initiated and receiver-initiated reliablemulticast protocols

Sender-initiated reliable multicast protocols based on the use of positive acknowledgments (ACKs) can suffer performance degradation as the number of receivers increases. This degradation is due to the fact that the sender must bear much of the complexity associated with reliable data transfer (e.g., maintaining state information and timers for each of the receivers and responding to receivers' ACKs). A potential solution to this problem is to shift the burden of providing reliable data transfer to the receivers-thus resulting in receiver-initiated multicast error control protocols based on the use of negative acknowledgments (NAKs). We determine the maximum throughputs for generic sender-initiated and receiver-initiated protocols for two classes of applications: (1) one-many applications where one participant sends data to a set of receivers and (2) many-many applications where all participants simultaneously send and receive data to/from each other. We show that a receiver-initiated error control protocol which requires receivers to transmit NAKs point-to-point to the sender provides higher throughput than a sender-initiated counterpart for both classes of applications. We further demonstrate that, in the case of a one many application, replacing point-to-point transfer of NAKs with multicasting of NAKs coupled with a random backoff procedure provides a substantial additional increase in the throughput of a receiver-initiated error control protocol over a sender-initiated protocol. We also find, however, that such a modification leads to a throughput degradation in the case of many-many applications     

基于机会式网络编码的低时延广播传输算法

为了提高无线网络中数据包广播传输的效率,本文提出了一种基于机会式网络编码的广播传输算法.该算法在发送端按一定顺序选择不同终端的丢包,并采用异或运算编码重传包,在终端采用从重传包中解码数据包的方法恢复丢包.该算法优先恢复时间重要性较高的丢包,并使多个终端同时从单个重传包恢复其丢包,因此有效地提高了广播传输效率并降低了传输...     

An Adaptive Delayed Acknowledgment Strategy to Improve TCP Performance in Multi-hop Wireless Networks

In multi-hop wireless networks, transmission control protocol (TCP) suffers from performance deterioration due to poor wireless channel characteristics. Earlier studies have shown that the small TCP acknowledgments consume as much wireless resources as the long TCP data packets. Moreover, generating an acknowledgment (ACK) for each incoming data packet reduces the performance of TCP. The main factor affecting TCP performance in multi-hop wireless networks is the contention and collision between ACK and data packets that share the same path. Thus, lowering the number of ACKs using the delayed acknowledgment option defined in IETF RFC 1122 will improve TCP performance. However, large cumulative ACKs will induce packet loss due to retransmission time-out at the sender side of TCP. Motivated by this understanding, we propose a new TCP receiver with an adaptive delayed ACK strategy to improve TCP performance in multi-hop wireless networks. Extensive simulations have been done to prove and evaluate our strategy over different topologies. The simulation results demonstrate that our strategy can improve TCP performance significantly.     

Adaptive Coding and Modulation for Multicast Transmission in Packet Radio Networks

A low-complexity protocol is described and evaluated for adaptation of the modulation and coding for multicast transmission in half-duplex packet radio networks. The adaptive multicast transmission protocol is designed to compensate for changes in propagation conditions that occur from packet to packet during a session with one sender and multiple receivers. The protocol relies on simple receiver statistics to obtain the control information for adapting the modulation and coding, and it also provides scheduling to avoid collisions among acknowledgments from the receivers. The throughput provided by the protocol is compared with performance results for hypothetical ideal adaptive multicast transmission protocols that are given perfect channel state information. We illustrate the importance of adaptive modulation and channel coding in systems that employ fountain coding for packet erasure correction.     

A Broadcast Protocol for File Transfers to Multiple Sites

A retransmission Protocol for a broadcast connection (point-to-multipoint) is proposed and its performance characteristics are considered. The protocol is designed for transfers of large files Over a satellite channel that is time-shared to carry both the data from the broadcasting transmitter and the set of acknowledgments from the multiple receiver sites. A mathematical model of the transmission system that includes separate error processes for uplink and downlink errors on data transmissions, and similar processes for errors on the acknowledgment frames as well, is used to analyze the performance of the scheme. Exact analytical expressions for the relative throughput of the channel are obtained for two special cases: 1) the uplink is error-free; and 2) the acknowledgments are error-free. For the general case, upper and lower bounds are derived and are shown to be virtually indistinguishable for many practical sets of system parameters. The results demonstrate that the broadcasting of large files to multiple receivers can be done both efficiently and reliably.     

Performance Analysis of TCP with Delayed Acknowledgments in Multi-hop Ad-hoc Networks

The acknowledgment strategy has great potential to increase TCP throughput when it runs over 802.11 MAC protocol. In particular, TCP acknowledgments (ACK) carry out an extensive number of medium accesses as they compete in the same route as data packets for media. In this paper, we first propose a dynamic TCP-MAC interaction strategy which tries to reduce the number of induced ACKs by monitoring the channel condition. To this end, the total collision probability collected along the path from sender to receiver in MAC layer has been used to properly set the number of delayed ACKs (DA) in TCP. Based on the estimated collision probability, TCP sender dynamically adjusts itself to the channel condition by delaying less ACKs in high traffic conditions and more ACKs in low traffic conditions. The simulation results show a throughput improvement up to 15% over the existing method called Dynamic Adaptive Acknowledgment (TCP-DAA) and much more over the regular TCP in different scenarios dealing with a dynamic loss rate. In addition, we show that our proposed strategy does not always benefit from a fixed delay policy along with a fixed congestion window size. In fact, the optimal number of delayed ACKs is based on the path length of a TCP connection and a large delay window may solely improve TCP throughput in short ranges with less number of flows. However, in a longer path congestion window limit provides more throughput gain.     

一种新的基于虚拟队列的无线多播网络编码调度策略

网络编码由于其传输效率高的特性,近年来在无线多播网络中得到广泛的应用。针对无线多播网络中丢包自动重传效率低的问题,该文提出一种新的基于虚拟队列中数据包到达时间的编码调度策略(CSAT)。在CSAT策略中,为了提高编码效率,采用虚拟队列来存放初始以及未被所有接收者接收到的数据包。考虑到队列的稳定性,CSAT策略按照一定的比率从主次队列选择发送;在次队列发送数据包时,结合了编码和非编码两种方式,根据数据包到达队列的先后,选取能够使较多数据包参与编码的方式发送。仿真结果表明,该文所提的CSAT编码调度策略在有效提高了数据包传输效率的同时,提高了网络的吞吐量并降低了平均等待时延。     

Formal specification and verification of safety and performance ofTCP selective acknowledgment

We present a formal specification of the selective acknowledgment (SACK) mechanism that is being proposed as a new standard option for TCP. The formal specification allows one to reason about the SACK protocol; thus, we are able to formally prove that the SACK mechanism does not violate the safety properties (reliable, at most once, and in order message delivery) of the acknowledgment (ACK) mechanism that is currently used with TCP. The new mechanism is being proposed to improve the performance of TCP when multiple packets are lost from one window of data. The proposed mechanism for implementing the SACK option for TCP is sufficiently complicated that it is not obvious that it is indeed safe, so we think it is important to formally verify its safety properties. In addition to safety, we are also able to show that SACK can improve the time it takes for the sender to recover from multiple packet losses. With the additional information available at a SACK sender, the round-trip time that a cumulative ACK sender waits before retransmitting each subsequent packet lost after the very first loss can be saved. We also show that SACK can improve performance even with window sizes as small as four packets and in situations where acknowledgment packets are lost     

Layered multicast congestion control by particle swarm optimization in heterogeneous environment

To enhance multicast throughput in heterogeneous environment,a new layered multicast congestion control scheme is proposed.With the goal of maximizing global satisfaction of the whole group,allocating sending rate in each layer is formulated to an optimization problem.Since the problem is noncovexity,the sender uses particle swarm optimization to search a set of optimal layers rates.The new scheme also eliminates ’lowest-first’ phenomenon by proposing a feedbacks suppression algorithm named equal-probability sampling (EPS).Upon EPS all the receivers send feedbacks at equal probability without bias.Simulation results prove that the new scheme can enhance global satisfaction and multicast throughput efficiently,compared with the traditional layered multicast congestion control scheme based on representatives.     

Resequencing delay and buffer occupancy in selective repeat ARQwith multiple receivers

The authors consider a selective repeat (SR) automatic-repeat-request (ARQ) protocol with one source and multiple receivers. Each receiver acknowledges all packets and handles its resequencing buffer based only on the packets that it receives error-free. An analysis of the resequencing delay and buffer occupancy at a receiver is presented. The authors construct a model that makes it possible to derive steady-state results, taking into consideration such system parameters as number of receivers, propagation delay, packet error probabilities, and acknowledgments. The authors focus on two measures of occupancy. The first measure corresponds to the number of packets waiting to be resequenced whereas the second measure includes, in addition, the buffer space reserved for packets that cause resequencing delays. The main results are the distribution of the resequencing delay and the distribution of the number of packets occupying the receiver's buffer     

Performance analysis of CSMA with priority acknowledgments(CSMA/PA) on noisy data networks with finite user population

An analytical procedure is developed to calculate delay and throughput performance of CSMA/PA (carrier sense multiple access with priority acknowledgments). Time-axis slotting and Markov chain analysis are used to determine the channel backlog per cycle. A cycle includes information packet access and transmission time followed by acknowledgment transmission or time-out. Throughput and delay performance are then calculated in terms of the number of network terminals, the network packet error probability, the packet generation rate, and the mean information packet retransmission delay. Fixed-length information and acknowledgment packets are assumed. Calculated results show excellent agreement with measured delay and throughput performance for two distinct five-node intrabuilding power line networks operating at data rates from 1.2 to 9.6 kb/s     

Network‐coding‐based cache policy for loss recovery enhancement in reliable multicast

In local loss recovery schemes, a small number of recovery nodes distributed along the transmission paths save incoming packets temporarily in accordance with a specified cache policy and retransmit these packets if they subsequently receive a request message from a downstream receiver. To reduce the recovery latency, the cache policy should ensure that the recovery nodes are always able to satisfy the retransmission requests of the downstream receivers. However, owing to the limited cache size of the recovery nodes and the behavior of the cache policy, this cannot always be achieved, and thus some of the packets must be retransmitted by the sender. Accordingly, this paper develops a new network‐coding‐based cache policy, designated as network‐coding‐based FIFO (NCFIFO), which extends the caching time of the packets at the recovery nodes without dropping any of the incoming packets. As a result, the lost packets can be always recovered from the nearest recovery nodes and the recovery latency is significantly reduced. The loss recovery performance of the NCFIFO cache policy is compared with that of existing cache policies by performing a series of simulation experiments using both a uniform error model and a burst error model. The simulation results show that the NCFIFO cache policy not only achieves a better recovery performance than existing cache policies, but also provides a more effective solution for managing a small amount of cache size in environments characterized by a high packet arrival rate. Copyright © 2011 John Wiley & Sons, Ltd.     

Measurement and Classification of Out-of-Sequence Packets in a Tier-1 IP Backbone

We present a classification methodology and a measurement study for out-of-sequence packets in TCP connections going over the Sprint IP backbone. Out-of-sequence packets can result from many events including loss, looping, reordering, or duplication in the network. It is important to quantify and understand the causes of such out-of-sequence packets since it is an indicator of the performance of a TCP connection, and the quality of its end-end path. Our study is based on passively observed packets from a point inside a large backbone network-as opposed to actively sending and measuring end-end probe traffic at the sender or receiver. A new methodology is thus required to infer the causes of a connection's out-of-sequence packets using only measurements taken in the "middle" of the connection's end-end path. We describe techniques that classify observed out-of-sequence behavior based only on the previously and subsequently-observed packets within a connection and knowledge of how TCP behaves. We analyze numerous several-hour packet-level traces from a set of OC-12 and OC-48 links for tens of millions connections generated in nearly 7600 unique ASes. We show that using our techniques, it is possible to classify almost all out-of-sequence packets in our traces and that we can quantify the uncertainty in our classification. Our measurements show a relatively consistent rate of out-of-sequence packets of approximately 4%. We observe that a majority of out-of-sequence packets are retransmissions, with a smaller percentage resulting from in-network reordering