It is well-known that wide-area networks face today several performance and reliability problems. In this work, we propose to solve these problems by connecting two or more local-area networks together via a Redundant Array of Internet Links (or RAIL) and by proactively replicating each packet over these links. In that sense, RAIL is for networks what RAID (Redundant Array of Inexpensive Disks) was for disks. In this paper, we describe the RAIL approach, present our prototype (called the RAILedge), and evaluate its performance. First, we demonstrate that using multiple Internet links significantly improves the end-to-end performance in terms of network-level as well as application-level metrics for Voice-over-IP and TCP. Second, we show that a delay padding mechanism is needed to complement RAIL when there is significant delay disparity between the paths. Third, we show that two paths provide most of the benefit, if carefully managed. Finally, we discuss a RAIL-network architecture, where RAILedges make use of path redundancy, route control and application-specific mechanisms, to improve WAN performance.
Our work fits in this scope as follows. It is related to multi-homing and overlay approaches in that it tries to improve end-to-end performance by connecting edge-networks via several different ISPs and by exploiting their path diversity. We compare to related work as follows. The novel aspect we are focusing on is proactive replication of every packet over the available paths in a single RAIL. This aspect is orthogonal to the online decision of switching traffic between RAILs (i.e. sets of paths). However, in this paper we still explore how to choose and manage the physical paths that constitute a single RAIL. Similarly to @cite_0 @cite_14 , we are looking at application-level metrics, particularly for VoIP and TCP. In contrast to the media-streaming work, we transmit redundant as opposed to complementary descriptions, operating on the assumption that bandwidth is not the issue. Our delay padding algorithm resembles playout buffering @cite_28 in that it tries to smooth out the network delay jitter; however, it is implemented at an edge device instead of the end-point, and acts only as a playout-proxy without dropping packets.
@cite_0 @cite_28 @cite_14
1976099036 2033837429
End-to-end packet delay is an important performance parameter on the Internet, because it heavily affects the quality of real-time applications. Currently, however, because the packet transmission quality (e.g., transmission delay, jitter, packet loss) may vary dynamically, it is not easy to handle real-time traffic. For UDP-based real-time applications, a smoothing buffer (playout buffer) is typically used at the client to compensate for variable delays. The issue of playout control has been studied previously, and several algorithms for controlling the playout buffer have been proposed. These studies considered the network parameters (e.g., packet loss ratio and playout delay), but not the quality perceived by end users. We first clarify the relations between mean opinion score (MOS) of played audio and the network parameters (e.g., packet loss, packet transmission delay, and transmission rate). Then, utilizing the MOS function, we propose a new playout buffer algorithm that considers the user's perceived quality for real-time applications. Our simulation and implementation tests show that the algorithm can enhance the perceived quality more effectively than existing algorithms. This paper explores the feasibility of improving the performance of end-to-end data transfers between different sites through path switching. Our study is focused on both the logic that controls path switching decisions and the configurations required to achieve sufficient path diversity. Specifically, we investigate two common approaches offering path diversity - multi-homing and overlay networks - and investigate their characteristics in the context of a representative wide-area testbed. We explore the end-to-end delay and loss characteristics of different paths and find that substantial improvements can potentially be achived by path switching, especially in lowering end-to-end losses. Based on this assessment, we develop a simple path-switching mechanism capable of realizing those performance improvements. Our experimental study demonstrates that substantial performance improvements are indeed achievable using this approach. The available bandwidth (avail-bw) in a network path is of major importance in congestion control, streaming applications, quality-of-service verification, server selection, and overlay networks. We describe an end-to-end methodology, called self-loading periodic streams (SLoPS), for measuring avail-bw. The basic idea in SLoPS is that the one-way delays of a periodic packet stream show an increasing trend when the stream's rate is higher than the avail-bw. We have implemented SLoPS in a tool called pathload. The accuracy of the tool has been evaluated with both simulations and experiments over real-world Internet paths. Pathload is nonintrusive, meaning that it does not cause significant increases in the network utilization, delays, or losses. We used pathload to evaluate the variability ("dynamics") of the avail-bw in Internet paths. The avail-bw becomes significantly more variable in heavily utilized paths, as well as in paths with limited capacity (probably due to a lower degree of statistical multiplexing). We finally examine the relation between avail-bw and TCP throughput. A persistent TCP connection can be used to measure roughly the avail-bw in a path, but TCP saturates the path and increases significantly the path delays and jitter. Delay and packet loss dramatically affect the quality of voice-over-IP (VoIP) calls and depend on the playout buffer scheme implemented at the receiver. The choice of playout algorithm can't be based on statistical metrics without considering the perceived end-to-end conversational speech quality. The authors present a method for evaluating various playout algorithms that extends the E-model concept by estimating user satisfaction from time-varying transmission impairments. This article evaluates several playout algorithms and shows a correspondence between the authors' results and those obtained via statistical loss and delay metrics. Network variations in video streaming require sufficient data to be prebuffered at the client prior to playout. This receiver buffer prevents the display process from starvation in case of network congestion. Prebuffering, however, is also responsible for the major part of the delay between requesting a media stream and playing it at the receiver. We show how adaptive media playout can be employed to reduce the delay introduced by the receiver buffer while preserving the same resilience against buffer underflow as in non-adaptive media playout. For adaptive media playout we adjust the playout speed of the media packets depending on the condition of the channel and the current client buffer fullness. We employ a two-state Markov channel model to analyze the buffer underflow-delay trade-off for our adaptive playout strategy and show that for typical parameters the average end-to-end delay can be reduced by 1 to 2 seconds. Packet loss and end-to-end delay limit delay sensitive applications over the best effort packet switched networks such as the Internet. In our previous work, we have shown that substantial reduction in packet loss can be achieved by sending packets at appropriate sending rates to a receiver from multiple senders, using disjoint paths, and by protecting packets with forward error correction. In this paper, we propose a path diversity with forward error correction (PDF) system for delay sensitive applications over the Internet in which, disjoint paths from a sender to a receiver are created using a collection of relay nodes. We propose a scalable, heuristic scheme for selecting a redundant path between a sender and a receiver, and show that substantial reduction in packet loss can be achieved by dividing packets between the default path and the redundant path. NS simulations are used to verify the effectiveness of PDF system. In theory, multi homed Internet hosts, that is, hosts simultaneously connected to multiple Internet service providers (ISP) should see increased access capacity, be able to circumvent possible last-mile congestion problems, and experience improved end-to-end quality of service (QoS). In practice however, the advantages one can gain from multi homing are highly dependent on the path switching mechanism used, that is, on dynamically deciding which ISP should be used as a first-hop. This paper is a first step toward understanding the trade-off between performance improvements multi homing can help achieve and the complexity of the decisions that must be made. We measure changes in end-to-end network layer metrics (loss, latency, jitter) over the different paths available from a multi homed host to a large population of Internet hosts. Our measurements indicate that (1) in over 60 of the cases, one only needs to reevaluate the service provided by each ISP every minute to improve the performance of a specific metric, and that (2) in approximately 85 of the cases, decisions to switch from one ISP to another can be treated independently of the service metric of interest. We conclude that multi homing could in practice result in noticeable performance improvements. The quality of service limitation of today's Internet is a major challenge for real-time voice communications. Excessive delay, packet loss, and high delay jitter all impair the communication quality. A new receiver-based playout scheduling scheme is proposed to improve the tradeoff between buffering delay and late loss for real-time voice communication over IP networks. In this scheme the network delay is estimated from past statistics and the playout time of the voice packets is adaptively adjusted. In contrast to previous work, the adjustment is not only performed between talkspurts, but also within talkspurts in a highly dynamic way. Proper reconstruction of continuous playout speech is achieved by scaling individual voice packets using a time-scale modification technique based on the Waveform Similarity Overlap-Add (WSOLA) algorithm. Results of subjective listening tests show that this operation does not impair audio quality, since the adaptation process requires infrequent scaling of the voice packets and low playout jitter is perceptually tolerable. The same time-scale modification technique is also used to conceal packet loss at very low delay, i.e., one packet time. Simulation results based on Internet measurements show that the tradeoff between buffering delay and late loss can be improved significantly. The overall audio quality is investigated based on subjective listening tests, showing typical gains of 1 on a 5-point scale of the Mean Opinion Score. Multimedia communications often require intramedia synchronization for video data to prevent potential playout discontinuity resulting from network delay variation (jitter) while still achieving satisfactory playout throughput. In this paper, we propose a neural network (NN) based intravideo synchronization mechanism, called the intelligent video smoother (IVS), operating at the application layer of the receiving end system. The IVS is composed of an NN traffic predictor, an NN window determinator, and a window-based playout smoothing algorithm. The NN traffic predictor employs an on-line-trained back-propagation neural network (BPNN) to periodically predict the characteristics of traffic modeled by a generic interrupted Bernoulli process (IBP) over a future fixed time period. With the predicted traffic characteristics, the NN window determinator determines the corresponding optimal window by means of an off-line-trained BPNN in an effort to achieve a maximum of the playout quality (Q) value. The window-based playout smoothing algorithm then dynamically adopts various playout rates according to the window and the number of packets in the buffer. Finally, we show that via simulation results and live video scenes, compared to two other playout approaches, IVS achieves high-throughput and low-discontinuity playout under a mixture of IBP arrivals. Measuring real-time end-to-end network path performance metrics is important for several distributed applications such as media streaming systems (e.g., for switching to paths with higher bandwidth and lower jitter) and content distribution systems (e.g., for selecting servers with lower latency). However, it is challenging to perform such end-to-end pairwise measurements in large distributed systems while achieving high accuracy and avoid interfering with existing traffic. On the end hosts, the measurements can overload the machine by causing interference among themselves and other processes. On the network, the measurement packets from different hosts can interfere among themselves and with other flows on bottleneck links. In this paper, we propose a system to monitor end-host and network resources and adapt the number of measurements according to the observed load. Our scheme avoids interference by measuring only a small subset of network paths and reconstructing the entire network path properties from the partial, indirect measurements. Our simulation experiments and real testbed experiments on PlanetLab show that our path selection algorithm working with resource constraints does not adversely affect the accuracy of inference and our system can effectively adapt to the changing resource usage at the end hosts. Video communication over lossy packet networks such as the Internet is hampered by limited bandwidth and packet loss. This paper presents a system for providing reliable video communication over these networks, where the system is composed of two subsystems: (1) multiple state video encoder decoder and (2) a path diversity transmission system. Multiple state video coding combats the problem of error propagation at the decoder by coding the video into multiple independently decodable streams, each with its own prediction process and state. If one stream is lost the other streams can still be decoded to produce usable video, and furthermore, the correctly received streams provide bidirectional (previous and future) information that enables improved state recovery for the corrupted stream. This video coder is a form of multiple description coding (MDC), and its novelty lies in its use of information from the multiple streams to perform state recovery at the decoder. The path diversity transmission system explicitly sends different subsets of packets over different paths, as opposed to the default scenarios where the packets proceed along a single path, thereby enabling the end- to-end video application to effectively see an average path behavior. We refer to this as path diversity. Generally, seeing this average path behavior provides better performance than seeing the behavior of any individual random path. For example, the probability that all of the multiple paths are simultaneously congested is much less than the probability that a single path is congested. The resulting path diversity provides the multiple state video decoder with an appropriate virtual channel to assist in recovering from lost packets, and can also simplify system design, e.g. FEC design. We propose two architectures for achieving path diversity, and examine the effectiveness of path diversity in communicating video over a lossy packet network. In this paper, we propose a novel transport protocol that effectively utilizes available bandwidth and diversity gains provided by heterogeneous, highly lossy paths. Our Multi-Path LOss-Tolerant (MPLOT) protocol can be used to provide significant gains in the goodput of wireless mesh networks, subject to bursty, correlated losses with average loss-rates as high as 50 , and random outage events. MPLOT makes intelligent use of erasure codes to guard against packets losses, and a Hybrid-ARQ FEC scheme to reduce packet recovery latency, where the redundancy is adaptively provisioned into both proactive and reactive FECs. MPLOT uses dynamic packet mapping based on current path characteristics, and does not require packets to be delivered in sequence to ensure reliability. We present a theoretical analysis of the different design choices of MPLOT and show that MPLOT makes an optimal trade-off between goodput and delay constraints. We test MPLOT, through simulations, under a variety of test scenarios and show that it effectively exploits path diversity in addition to aggregating path bandwidths. We also show that MPLOT is fair to single-path protocols like TCP-SACK. Delivery of real time streaming applications, such as voice and video over IP, in packet switched networks is based on dividing the stream into packets and shipping each of the packets on an individual basis to the destination through the network. The basic implicit assumption on these applications is that shipping all the packets of an application is done, most of the time, over a single path along the network. In this work, we present a model in which packets of a certain session are dispersed over multiple paths, in contrast to the traditional approach. The dispersion may be performed by network nodes for various reasons such as load-balancing, or implemented as a mechanism to improve quality, as will be presented in this work. To study the effect of packet dispersion on the quality of voice over IP (VoIP) applications, we focus on the effect of the network loss on the applications, where we propose to use the Noticeable Loss Rate (NLR) as a measure (negatively) correlated with the voice quality. We analyze the NLR for various packet dispersion strategies over paths experiencing memoryless (Bernoulli) or bursty (Gilbert model) losses, and compare them to each other. Our analysis reveals that in many situations the use of packet dispersion reduces the NLR and thus improves session quality. The results suggest that the use of packet dispersion can be quite beneficial for these applications. The use of forward error correction (FEC) coding is often proposed to combat the effects of network packet losses for error-resilient video transmission on packet-switched networks. On the other hand, path diversity has recently been proposed to improve network transport for both singledescription (SD) and multiple-description (MD) coded video. In this work we model and analyze an SD coded video transmission system employing packet-level FEC in combination with path diversity. In particular, we provide a precise analytical approach to evaluating the efficacy of path diversity in reducing the burstiness of network packet-loss processes. We use this approach to quantitatively demonstrate the advantages of path diversity in improving end-to-end video transport performance using packet-level FEC. An important issue in IP-based QoS networks is the effective management of packets at the router level. Specifically, if the arriving packets cannot all be stored in a buffer, or if the packets have deadlines by which they must be delivered, the router needs to identify the packets that should be dropped. In recent work, [6] propose a model, called buffer management with bounded delay, which can be thought of as an online scheduling problem on a single machine: packets arrive at a network switch and are stored in a buffer of size B. Each packet has a positive weight and a deadline, with the weight representing the value of transmitting the packet by its deadline. At each integer time step, exactly one packet can be transmitted, and the objective is to maximize the total weight of the transmitted packets. If B = ∞, this is the online version of the scheduling problem 1| p j = 1, r j , d j |Σ w j U j . (We assume that r j and d j are integers.) As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to meet the quality standards achieved in the public switched telephone network. Our objective in this paper is to assess to what extent today's Internet meets this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks and uses subjective voice quality measures capturing the various impairments incurred. First, we compile the results of various studies into a single model for assessing the voice-over-IP (VoIP) quality. Then, we identify different types of typical Internet paths and study their VoIP performance. For each type of path, we identify those characteristics that affect the VoIP perceived quality. Such characteristics include the network loss and the delay variability that should be appropriately handled by the playout scheduling at the receiver. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of Internet backbone paths lead to poor performance. Although packet-pair probing has been used as one of the primary mechanisms to measure bottleneck capacity, cross-traffic intensity, and available bandwidth of end-to-end Internet paths, there is still no conclusive answer as to what information about the path is contained in the output packet-pair dispersions and how it is encoded. In this paper, we address this issue by deriving closed-form expression of packet-pair dispersion in the context of a single-hop path and general bursty cross-traffic arrival. Under the assumptions of cross-traffic stationarity and ASTA sampling, we examine the statistical properties of the information encoded in inter-packet spacings and derive the asymptotic average of the output packet-pair dispersions as a closed-form function of the input dispersion. We show that this result is different from what was obtained in prior work using fluid cross-traffic models and that this discrepancy has a significant impact on the accuracy of packet-pair bandwidth estimation. We study routing and scheduling in packet-switched networks. We assume an adversary that controls the injection time, source, and destination for each packet injected. A set of paths for these packets is admissible if no link in the network is overloaded. We present the first on-line routing algorithm that finds a set of admissible paths whenever this is feasible. Our algorithm calculates a path for each packet as soon as it is injected at its source using a simple shortest path computation. The length of a link reflects its current congestion. We also show how our algorithm can be implemented under today's Internet routing paradigms.When the paths are known (either given by the adversary or computed as above), our goal is to schedule the packets along the given paths so that the packets experience small end-to-end delays. The best previous delay bounds for deterministic and distributed scheduling protocols were exponential in the path length. In this article, we present the first deterministic and distributed scheduling protocol that guarantees a polynomial end-to-end delay for every packet.Finally, we discuss the effects of combining routing with scheduling. We first show that some unstable scheduling protocols remain unstable no matter how the paths are chosen. However, the freedom to choose paths can make a difference. For example, we show that a ring with parallel links is stable for all greedy scheduling protocols if paths are chosen intelligently, whereas this is not the case if the adversary specifies the paths. There has been extensive work on network architectures that support multi-path routing to improve performance in wireless mesh networks. However, previous work uses ad-hoc design principles that cannot guarantee any network-wide performance objectives such as conjointly maximizing resource utilization and improving fairness. In parallel, numerous theoretical results have addressed the issue of optimizing a combined metric of network utilization and fairness using techniques based on back-pressure scheduling, routing and flow control. However, the proposed theoretical algorithms are extremely difficult to implement in practice, especially in the presence of the 802.11 MAC and TCP. We propose Horizon, a novel system design for multi-path forwarding in wireless meshes, based on the theoretical results on back-pressure. Our design works with an unmodified TCP stack and on top of the existing 802.11 MAC. We modified the back-pressure approach to obtain a simple 802.11-compatible packet-forwarding heuristic and a novel, light-weight path estimator, while maintaining global optimality properties. We propose a delayed reordering algorithm that eliminates TCP timeouts while keeping TCP packet reordering to a minimum. We have evaluated our implementation on a 22-node testbed. We have shown that Horizon effectively utilizes available resources (disjoint paths). In contrast to previous work, our design not only avoids bottlenecks but also optimally load-balances traffic across them when needed, improving fairness among competing flows. To our knowledge, Horizon is the first practical wireless system based on back-pressure. Multipath transport protocols such as Multipath TCP can concurrently use several subflows to transmit a TCP flow over potentially different paths. Since more than one subflow is used, an efficient multipath scheduling algorithm is needed at the sender. The objective of the scheduler is to identify the subflow over which the current data packet should be sent. This paper compares the most important types of schedulers for multipath transfers. We model their performance analytically and derive key metrics, most notably the resulting end-to-end delay over heterogeneous paths. Our results show that a scheduler minimizing the packet delivery delay yields the best overall performance, but it is complex to realize. An alternative scheduler based on the sender queue size is simpler and has sufficient performance for relatively small asymmetry between the multiple paths. Our model results are confirmed by measurements with a real multipath transport protocol.