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C. Weinstein

Bio: C. Weinstein is an academic researcher from Massachusetts Institute of Technology. The author has contributed to research in topics: Packet switching & Fast packet switching. The author has an hindex of 1, co-authored 1 publications receiving 155 citations.

Papers
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Journal ArticleDOI
TL;DR: Large-scale packet speech multiplexing experiments could not be carried out on ARPANET or SATNET where the network link capacities severely restrict the number of speech users that can be accommodated, but experiments are currently being carried out using a wide-band satellite-based packet system designed to accommodate a sufficient number of simultaneous users to support realistic experiments in efficient statisticalmultiplexing.
Abstract: The integration of digital voice with data in a common packet-switched network system offers a number of potential benefits, including reduced systems cost through sharing of switching and transmission resources, flexible internetworking among systems utilizing different transmission media, and enhanced services for users requiring access to both voice and data communications. Issues which it has been necessary to address in order to realize these benefits include reconstitution of speech from packets arriving at nonuniform intervals, maximization of packet speech multiplexing efficiency, and determination of the implementation requirements for terminals and switching in a large-scale packet voice/data system. A series of packet speech systems experiments to address these issues has been conducted under the sponsorship of the Defense Advanced Research Projects Agency (DARPA). In the initial experiments on the ARPANET, the basic feasibility of speech communication on a store-and-forward packet network was demonstrated. Techniques were developed for reconstitution of speech from packets, and protocols were developed for call setup and for speech transport. Later speech experiments utilizing the Atlantic packet satellite network (SATNET) led to the development of techniques for efficient voice conferencing in a broadcast environment, and for internetting speech between a store-and-forward net (ARPANEI) and a broadcast net (SATNET). Large-scale packet speech multiplexing experiments could not be carried out on ARPANET or SATNET where the network link capacities severely restrict the number of speech users that can be accommodated. However, experiments are currently being carried out using a wide-band satellite-based packet system designed to accommodate a sufficient number of simultaneous users to support realistic experiments in efficient statistical multiplexing. Key developments to date associated with the wide-band experiments have been 1) techniques for internetting via voice/data gateways from a variety of local access networks (packet cable, packet radio, and circuit-switched) to a long-haul broadcast satellite network and 2) compact implementations of packet voice terminals with full protocol and voice capabilities. Basic concepts and issues associated with packet speech systems are described. Requirements and techniques for speech processing, voice protocols, packetization and reconstitution, conferencing, and multiplexing are discussed in the context of a generic packet speech system configuration. Specific experimental configurations and key packet speech results on the ARPANET, SATNET, and wide-band system are reviewed.

155 citations


Cited by
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Patent
12 Jan 2010
TL;DR: In this paper, a packet-based, hierarchical communication system, arranged in a spanning tree configuration, is described in which wired and wireless communication networks exhibiting substantially different characteristics are employed in an overall scheme to link portable or mobile computing devices.
Abstract: A packet-based, hierarchical communication system, arranged in a spanning tree configuration, is described in which wired and wireless communication networks exhibiting substantially different characteristics are employed in an overall scheme to link portable or mobile computing devices. The network accommodates real time voice transmission both through dedicated, scheduled bandwidth and through a packet-based routing within the confines and constraints of a data network. Conversion and call processing circuitry is also disclosed which enables access devices and personal computers to adapt voice information between analog voice stream and digital voice packet formats as proves necessary. Routing pathways include wireless spanning tree networks, wide area networks, telephone switching networks, internet, etc., in a manner virtually transparent to the user. A voice session and associate call setup simulates that of conventional telephone switching network, providing well-understood functionality common to any mobile, remote or stationary terminal, phone, computer, etc.

1,080 citations

Journal ArticleDOI
Scott Shenker1
TL;DR: This work addresses some of the fundamental architectural design issues facing the future Internet, including whether the Internet should adopt a new service model, how this service model should be invoked, and whether this service models should include admission control.
Abstract: The Internet has been a startling and dramatic success. Originally designed to link together a small group of researchers, the Internet is now used by many millions of people. However, multimedia applications, with their novel traffic characteristics and service requirements, pose an interesting challenge to the technical foundations of the Internet. We address some of the fundamental architectural design issues facing the future Internet. In particular, we discuss whether the Internet should adopt a new service model, how this service model should be invoked, and whether this service model should include admission control. These architectural issues are discussed in a nonrigorous manner, through the use of a utility function formulation and some simple models. While we do advocate some design choices over others, the main purpose here is to provide a framework for discussing the various architectural alternatives. >

1,072 citations

Patent
22 Nov 1999
TL;DR: In this article, the authors describe a system and method for communicating voice and data over a packet-switched network that is adapted to coexist and communicate with a legacy PSTN.
Abstract: The present invention describes a system and method for communicating voice and data over a packet-switched network that is adapted to coexist and communicate with a legacy PSTN. The system permits packet switching of voice calls and data calls through a data network from and to any of a LEC, a customer facility or a direct IP connection on the data network. The system includes soft switch sites, gateway sites, a data network, a provisioning component, a network event component and a network management component. The system interfaces with customer facilities (e.g., a PBX), carrier facilities (e.g., a LEC) and legacy signaling networks (e.g., SS7) to handle calls between any combination of on-network and off-network callers. The soft switch sites provide the core call processing for the voice network architecture. The soft switch sites manage the gateway sites in a preferred embodiment, using a protocol such as the Internet Protocol Device Control (IPDC) protocol to request the set-up and tear-down of calls. The gateway sites originate and terminate calls between calling parties and called parties through the data network. The gateway sites include network access devices to provide access to network resources. The data network connects one or more of the soft switch sites to one or more of the gateway sites. The provisioning and network event component collects call events recorded at the soft switch sites. The network management component includes a network operations center (NOC) for centralized network management.

1,024 citations

Proceedings ArticleDOI
01 Oct 1992
TL;DR: This paper considers the support of real-time applications in an Integrated Services Packet Network (ISPN), and proposes an ISPN architecture that supports two distinct kinds of real time service: guaranteed service, which involves pre-computed worst-case delay bounds, and predicted service which uses the measure performance of the network in computing delay bounds.
Abstract: This paper considers the support of real-time applications in an Integrated Services Packet Network (ISPN). We first review the characteristics of real-time applications. We observe that, contrary to the popular view that real-time applications necessarily require a fixed delay bound, some real-time applications are more flexible and can adapt to current network conditions. We then propose an ISPN architecture that supports two distinct kinds of real-time service: guaranteed service, which is the traditional form of real-time service discussed in most of the literature and involves pre-computed worst-case delay bounds, and predicted service which uses the measure performance of the network in computing delay bounds. We then propose a packet scheduling mechanism that can support both of these real-time services as well as accommodate datagram traffic. We also discuss two other aspects of an overall ISPN architecture: the service interface and the admission control criteria.

919 citations

Journal ArticleDOI
TL;DR: Simulation work is reported indicating that packet reservation multiple access (PRMA) allows a variety of information sources to share the same wireless access channel and achieves a promising combination of voice quality and bandwidth efficiency.
Abstract: Simulation work is reported indicating that packet reservation multiple access (PRMA) allows a variety of information sources to share the same wireless access channel. Some of the sources, such as speech terminals, are classified as periodic and others, such as signaling, are classified as random. Packets from all sources contend for access to channel time slots. When a periodic information terminal succeeds in gaining access, it reserves subsequent time slots for uncontested transmission. Both computer simulations and a listening test reveal that PRMA achieves a promising combination of voice quality and bandwidth efficiency. >

890 citations