From the Publisher:
The most up-to-date introduction to the field of computer networking, this book's top-down approach starts at the application layer and works down the protocol stack. It also uses the Internet as the main example of networks. This all creates a book relevant to those interested in networking today. By starting at the application-layer and working down the protocol stack, this book provides a relevant introduction of important concepts. Based on the rationale that once a reader understands the applications of networks they can understand the network services needed to support these applications, this book takes a "top-down" approach that exposes readers first to a concrete application and then draws into some of the deeper issues surrounding networking. This book focuses on the Internet as opposed to addressing it as one of many computer network technologies, further motivating the study of the material. This book is designed for programmers who need to learn the fundamentals of computer networking. It also has extensive material making it of great interest to networking professionals.

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Computer Networking: A Top-Down Approach Featuring the Internet

Reliable communication protocols require that all the intended recipients of a message receive the message intact. Automatic Repeat reQuest (ARQ) techniques are used in unicast protocols, but they do not scale well to multicast protocols with large groups of receivers, since segment losses tend to become uncorrelated thus greatly reducing the effectiveness of retransmissions. In such cases, Forward Error Correction (FEC) techniques can be used, consisting in the transmission of redundant packets (based on error correcting codes) to allow the receivers to recover from independent packet losses.Despite the widespread use of error correcting codes in many fields of information processing, and a general consensus on the usefulness of FEC techniques within some of the Internet protocols, very few actual implementations exist of the latter. This probably derives from the different types of applications, and from concerns related to the complexity of implementing such codes in software. To fill this gap, in this paper we provide a very basic description of erasure codes, describe an implementation of a simple but very flexible erasure code to be used in network protocols, and discuss its performance and possible applications. Our code is based on Vandermonde matrices computed over GF(pr), can be implemented very efficiently on common microprocessors, and is suited to a number of different applications, which are briefly discussed in the paper. An implementation of the erasure code shown in this paper is available from the author, and is able to encode/decode data at speeds up to several MB/s running on a Pentium 133.

/pdf/effective-erasure-codes-for-reliable-computer-communication-4sfugth909.pdf

Effective erasure codes for reliable computer communication protocols

We use the measured round trip delays of small UDP probe packets sent at regular time intervals to analyze the end-to-end packet delay and loss behavior in the Internet. By varying the interval between probe packets, it is possible to study the structure of the Internet load over different time scales. In this paper, the time scales of interest range from a few milliseconds to a few minutes. Our observations agree with results obtained by others using simulation and experimental approaches. For example, our estimates of Internet workload are consistent with the hypothesis of a mix of bulk traffic with larger packet size, and interactive traffic with smaller packet size. We observe compression (or clustering) of the probe packets, rapid fluctuations of queueing delays over small intervals, etc. Our results also show interesting and less expected behavior. For example, we find that the losses of probe packets are essentially random unless the probe traffic uses a large fraction of the available bandwidth. We discuss the implications of these results on the design of control mechanisms for the Internet.

/pdf/end-to-end-packet-delay-and-loss-behavior-in-the-internet-mgzpeb25au.pdf

End-to-end packet delay and loss behavior in the internet

We use the measured round trip delays of small UDP probe packets sent at regular time intervals to characterize the end-to-end packet delay and loss behavior in the Internet. By varying the interval between probe packets, it is possible to study the structure of the Internet load over different time scales. In this paper, the time scales of interest range from a few milliseconds to a few minutes. Our observations agree with results obtained by others using simulation and experimental approaches. For example, our estimates of Internet workload are consistent with the hypothesis of a mix of bulk traffic with larger packet size, and interactive traffic with smaller packet size. The interarrival time distribution for Internet packets is consistent with an exponential distribution. We also observe a phenomenon of compression (or clustering) of the probe packets similar to the acknowledgement compression phenomenon recently observed in TCP. Our results also show interesting and less expected behavior. For example, we find that the losses of probe packets are essentially random when the probe traffic uses a small fraction of the available bandwidth.

Characterizing End-to-End Packet Delay and Loss in the Internet

The Internet provides a single class best effort service. From an application's point of view, this service amounts in practice to providing channels with time-varying characteristics such as delay and loss distributions. One way to support real time applications such as interactive audio given this service is to use control mechanisms that adapt the audio coding and decoding processes based on the characteristics of the channels, the goal being to maximize the quality of the audio delivered to the destinations. In this paper, we describe and analyze a set of such control mechanisms. They include a jitter control mechanism and a combined error and rate control mechanism. These mechanisms have been implemented and evaluated over the Internet and the MBone. Experiments indicate that they make it possible to establish and maintain reasonable quality audioconferences even across fairly congested connections.

Control mechanisms for packet audio in the Internet

A technique for fiber-optic networks based on forward-error correction (FEC) that allows the destination to reconstruct missing data packets by using redundant parity packets that the source adds to each block of data packets is presented. Methods for generating several types of parity packets are described, along with decoding techniques and their implementations. Algorithms are presented for packet interleaving and selective rejection of packets from node buffers, both of which disperse missing packets among many blocks, thereby reducing the required coding complexity. Performance evaluation, by both analytic and simulation models, shows that this technique can result in a reduction of up to three orders of magnitude in the packet loss rate. >

N. Shacham

Papers

Packet recovery in high-speed networks using coding and buffer management