This paper presents Random Early Detection (RED) gate-ways for congestion avoidance in packet-switched networks. The gateway detects incipient congestion by com-puting the average queue size. The gateway could notify connections of congestion either by dropping packets ar-riving at the gateway or by setting a bit in packet headers. When the average queue size exceeds a preset threshold,the gateway drops or marks each arriving packet with a certain probability, where the exact probability is a func-tion of the average queue size. RED gateways keep the average queue size low while allowing occasional bursts of packets in the queue. During congestion, the probability that the gateway notifies a particular connection to reduce its window is roughly proportional to that connection's share of the bandwidth throughthe gateway. RED gateways are designed to accompany a transport-layer congestion control protocol such as TCP.The RED gateway has no bias against bursty traffic and avoids the global synchronization of many connectionsdecreasing their window at the same time. Simulations of a TCP/IP network are used to illustrate the performance of RED gateways.

We discuss findings from a large-scale study of Internet packet dynamics conducted by tracing 20 000 TCP bulk transfers between 35 Internet sites. Because we traced each 100-kbyte transfer at both the sender and the receiver, the measurements allow us to distinguish between the end-toend behaviors due to the different directions of the Internet paths, which often exhibit asymmetries. We: 1) characterize the prevalence of unusual network events such as out-of-order delivery and packet replication; 2) discuss a robust receiver-based algorithm for estimating “bottleneck bandwidth ” that addresses deficiencies discovered in techniques based on “packet pair;” 3) investigate patterns of packet loss, finding that loss events are not well modeled as independent and, furthermore, that the distribution of the duration of loss events exhibits infinite variance; and 4) analyze variations in packet transit delays as indicators of congestion periods, finding that congestion periods also span a wide range of time scales.

While today’s computer networks support only best-effort service, future packet-switching integrated-services networks will have to support real-time communication services that allow clients to transport information with performance guarantees expressed in terms of delay, delay jitter, throughput, and loss rate. An important issue in providing guaranteed performance service is the choice of the packet service discipline at switching nodes. In this paper, we survey several service disciplines that are proposed in the literature to provide per-connection end-to-end peqormance guarantees in packet-switching networks. We describe their mechanisms, their similarities and differences, and the performance guarantees they can provide. Various issues and tradeoffs in designing service disciplines for guaranteed performance service are discussed, and a general framework for studying and comparing these disciplines are presented. I.

Vegas is a new implementation of TCP that achieves between 40 and 70 % better throughput, with one-fifth to onehalf the losses, as compared to the implementation of TCP in the Reno distributionof BSD Unix. This paper motivates and describes the three key techniques employed by Vegas, and presents the results of a comprehensive experimental performance study—using both simulations and measurements on the Internet—of the Vegas and Reno implementations of TCP. 1

Vegas is an implementation of TCP that achieves between 37 and 71 % better throughput on the Internet, with one-fifth to one-half the losses, as compared to the implementation of TCP in the Reno distribution of BSD Unix. This paper motivates and describes the three key techniques employed by Vegas, and presents the results of a comprehensive experimental performance study—using both simulations and measurements on the Internet—of the Vegas and Reno implementations of TCP.

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This paper describes the INRIA Videoconferencing System (IVS), a low bandwidth tool for real-time video between workstations on the Internet using UDP datagrams and the IP multicast extension. The video coder-decoder (codec) is a software implementation of the UIT-T recommendation H.261 originally developed for the Integrated Services Digital Network (ISDN). Our focus in this paper is on adapting this codec for the Internet environment. We propose a packetization scheme, an error control scheme and an output rate control scheme that adapts the image coding process based on network conditions. This work shows that it is possible to maintain videoconferences with reasonable quality across packet-switched networks without requiring special support from the network such as resource reservation or admission control.

Existing approaches for providing guaranteed services require routers to manage per ow states and perform per ow operations [9, 21]. Such a stateful network architecture is less scalable and robust than stateless network architectures like the original IP and the recently proposed Di serv [3]. However, services provided with current stateless solutions, Di serv included, have lower exibility, utilization, and/or assurance level as compared to the services that can be provided with per ow mechanisms. In this paper, we propose techniques that do not require per ow management (either control or data planes) at core routers, but can implement guaranteed services with levels of exibility, utilization, and assurance similar to those that can be provided with per ow mechanisms. In this way we can simultaneously achieve high quality of service, high scalability and robustness. The key technique we use is called Dynamic Packet State (DPS), which provides a lightweight and robust mechanism for routers to coordinate actions and implement distributed algorithms. We present an implementation of the proposed algorithms that has minimum incompatibility with IPv4.

The available bandwidth (avail-bw) in a network path is of major importance in congestion control, streaming applications, QoS 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 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 non-intrusive, 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 some paths that cross USA and Europe. 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 roughly measure the avail-bw in a path, but TCP saturates the path, and increases significantly the path delays and jitter.