Very high speed and high capacity packet switching for broadbandISDN

The authors discuss and propose a very-high-speed and high-capacity packet-switching (HPS) architecture for a future broadband ISDN (integrated-services digital network). The HPS network accommodates various communication services, such as voice, high-speed data, high-speed still picture, and video services. The proposed architecture has three significant principles: a high-speed oriented simple network protocol, separation of signaling and network control from data transfer, and hardware switching. These principles provide fast- and high-throughput transmission for data packets and reliable transmission and processing for call-control packets. The HPS protocol structure is addressed, which provides high flexibility for various communications services as well as high-speed capability. A 3-Gb/s capacity and building-block-structured packet-switching system architecture, using bus- and loop-type switch fabric, is also presented     

Paris: An approach to integrated high-speed private networks

This paper describes a design of a high-speed packet switching system for integrated voice, video and data communications. The system makes use of a simplified network architecture in order to achieve the low packet delay and high nodal throughput necessary for the transport of voice and video. A prototype of this system has been implemented and is now being tested under a variety of packet traffic loads. We have demonstrated that this system provides a cost-effective solution for private integrated networks.     

Integrated Services Packet Network Using Bus Matrix Switch

For several years, Fujitsu has been researching and developing high-speed packet switching networks, the result being an integrated multimedia information networks architecture. This architecture has already been applied to LAN systems and can now be applied to wide area corporate networks thanks to the new developments described in this paper. The most important technology-the bus matrix switch-provides a quantum leap in processing capacity up to several Gbit/s (few millions of packets every second), while keeping a very low switching delay. The performance evaluation based on our prototype models is also considered in detail. For example, a voice delay of less than 20 ms is obtained for five-hop communication. The new technologies will greatly improve for computer networks and will also enable basic and broadband ISDN.     

Congestion control for multimedia services

The problem of congestion control in high-speed networks for multimedia traffic, such as voice and video, is considered. It is shown that the performance requirements of high-speed networks involve delay, delay-jitter, and packet loss. A framing congestion control strategy based on a packet admission policy at the edges of the network and on a service discipline called stop-and-go queuing at the switching nodes is described. This strategy provides bounded end-to-end delay and a small and controllable delay-jitter. The strategy is applicable to packet switching networks in general, including fixed cell length asynchronous transfer mode (ATM), as well as networks with variable-size packets     

Packet CDMA versus dynamic TDMA for multiple access in anintegrated voice/data PCN

A comparative evaluation of dynamic time-division multiple access (TDMA) and spread-spectrum packet code-division multiple access (CDMA) approaches to multiple access in an integrated voice/data personal communications network (PCN) environment are presented. After briefly outlining a cellular packet-switching architecture for voice/data PCN systems, dynamic TDMA and packet CDMA protocols appropriate for such traffic scenarios are described. Simulation-based network models which have been developed for performance evaluation of these competing access techniques are then outlined. These models are exercised with example integrated voice/data traffic models to obtain comparative system performance measures such as channel utilization, voice blocking probability, and data delay. Operating points based on typical performance constraints such as voice blocking probability 0.01 (for TDMA), voice packet loss rate 10^-3 (for CDMA), and data delay 250 ms are obtained, and results are presented     

Output and delay process analysis for slotted CDMA wirelesscommunication networks with integrated voice/data transmission

This paper presents the output and delay process analysis of integrated voice/data slotted code division multiple access (CDMA) network systems with random access protocol for packet radio communications. The system model consists of a finite number of users, and each user can be a source of both voice traffic and data traffic. The allocation of codes to voice calls is given priority over that to data packets, while an admission control, which restricts the maximum number of codes available to voice sources, is considered for voice traffic so as not to monopolize the resource. Such codes allocated exclusively to voice calls are called voice codes. In addition, the system monitoring can distinguish between silent and talkspurt periods of voice sources, so that users with data packets can use the voice codes for transmission if the voice sources are silent. A discrete-time Markov process is used to model the system operation, and an exact analysis is presented to derive the moment generating functions of the probability distributions for packet departures of both voice and data traffic and for the data packet delay. For some cases with different numbers of voice codes, numerical results display the correlation coefficient of the voice and data packet departures and the coefficient of variation of the data packet delay as well as average performance measures, such as the throughput, the average delay of data packets, and the average blocking probability of voice calls     

Software Protocol for Handling ISDN in the Packet-Switched Network

The Integrated Services Digital Network (ISDN) provides basic architecture for existing, as well as future residential plus business communications. ISDN overlayed with CCS#7 of a digital PSTN (Public Switched Telephone Network) can be the ultimate, ubiquitous network for circuit switch (voice, data), packet switch (voice, data), and private line (voice, data) applications. Assuming that the present ISDN has to interwork in the present physically separate overlayed networks (voice and data), significant problems are expected to emerge for designing hardware and linking softwares for handling packet traffic. In this paper, the software-related problems, when ISDN packet distribution nodes have to handle an ISDN interface, will be outlined with an ISDN software protocol solution. An approximation of the delay involved in the telephone switching system which is part of ISDN processing as well as the delay for the interface gateways, the HOST computer nodes, and the LAN and WAN computer nodes will be identified and formulated to reflect the total performance measure defined. Major emphasis is given to flow and congestion control performance measures in the ISDN Gateways, which are analyzed and simulated with the assistance of the basic delay table transfer software model developed for the IMPS and gateways in the ARPANET, MILNET, and MINET. The performance evaluation of this basic ISDN interfacing software, which only involved one ISDN level, i.e., the HOST or gateway and its related subnetworks, is simulated on sections of these networks to illustrate its congestion control effectiveness. There are six mathematical software techniques to account for end-to-end delay, which form the basis for the solution to these ISDN software-hardware problems in the Interface Gateways connecting the electronic switch to the computer network components.     

Design of a broadcast packet switching network

An overview is given of a system designed to handle a heterogeneous and dynamically changing mix of applications. It is based on fiber-optic transmission systems and high-performance packet switching and can handle applications ranging from low-speed data to voice to full-rate video. A novel feature is a flexible multipoint connection capability suitable for broadcast and conferencing applications. The architecture of a switching systems that can be used to support this network is described     

On Multiple Traffic Type Integration over Wireless TDMA Channels with Adjustable Request Bandwidth

A new medium access control (MAC) protocol for mobile wireless communications is presented and investigated. We explore, via an extensive simulation study, the performance of the protocol when integrating voice, video and data packet traffic over a wireless channel of high capacity (referring to an indoor microcellular environment). Depending on the number of video users admitted into the system, our protocol varies: a) the request bandwidth dedicated to resolving the voice users contention, and b) the probability with which the base station grants information slots to voice users, in order to preserve full priority for video traffic. We evaluate the maximum voice capacity and mean access delay, as well as the aggregate channel throughput, for various voice and video load conditions, and the maximum voice capacity, aggregate channel throughput and average data message delays, for various video, voice and data load conditions. As proven by the comparison with a recently introduced efficient MAC scheme (DPRMA), when integrating voice and video traffic our scheme obtains higher voice capacity and aggregate channel throughput. When integrating all three traffic types, our scheme achieves high aggregate channel throughput in all cases of traffic load.     

Integrated video and voice transmission using packet reservation multiple access

This paper proposes a new protocol for the integration of voice and video transmission over the packet reservation multiple access (PRMA) system that is a modification of reservation‐ALOHA protocol. We focus on low bit‐rate video applications like video conferencing and visual telephony for wireless communications. The ITU–T H.263 standard provides a solution to the need for low bit‐rate video compression under 64 kbytes/s. The proposed protocol assumes that each voice terminal follows a traffic pattern of talk spurts and silent gaps with fixed permission probability (p=0.3), and each video terminal has the higher permission probability (p=1) to access the available slot based on ITU–T H.263 standard. Again, we present a ‘pseudo‐reservation’ scheme to release slots reserved by video terminals according to the contents of each video transmission buffer, and the active voice terminals can temporarily access the additional slots to improve the performance without sacrificing the video capacity of the system. The packet dropping probability of the active voice terminals and bandwidth utilization of the system are superior to the original PRMA, as indicated in simulation results. Copyright © 2001 John Wiley & Sons, Ltd.     

Seamless SIP-based mobility for multimedia applications

Application-level protocol abstraction is required to support seamless mobility in next-generation heterogeneous wireless networks. Session initiation protocol (SIP) provides the required abstraction for mobility support for multimedia applications in such networks. However, the handoff procedure with SIP suffers from undesirable delay and hence packet loss in some cases, which is detrimental to applications like voice over IP (VoIP) or streaming video that demand stringent quality of service (QoS) requirements. In this article we present a SIP-based architecture that supports soft handoff for IP-centric wireless networks. Soft handoff ensures that there is no packet loss and that the end-to-end delay jitter is kept under control.     

Switching for IP-based multimedia satellite communications

This paper discusses the structure and performance of an Internet protocol (IP)-based satellite communications system to provide multimedia services. Uplink scheduling and switching to support IP differentiated services (DiffServ) traffic in a multibeam environment are addressed. End-to-end performance of a multibeam satellite communications system using an on-board switch is evaluated using simulation. Aggregate real-time and non-real-time traffic using different DiffServ classes is considered and the effects of their burstiness and long-range dependent behavior on the queueing performance are examined. Multiple-frequency time-division multiple-access is used on the uplink in conjunction with a dynamic capacity allocation scheme. Higher priority is given to voice and video real-time traffic to avoid delay variation. On-board downlink queue for non-real-time traffic is provided to achieve high statistical multiplexing gain.     

A communication protocol for mobile satellite systems affected byrain attenuation

A protocol that supports real-time data rate selection and change during rain events is presented. The protocol is developed with emphasis on being efficient yet robust to the primary channel impairment in such mobile satellite systems. The system architecture is briefly presented and the analytical framework from which the protocol originates is pointed out. Link, connection, and packet types are introduced, and the protocol procedures and design rationale are discussed. The detailed presentation focuses on link setup with the appropriate data rate(s) and the real-time switching of data rates during a voice conversation to either preserve the link or enhance its quality during rain attenuation events     

Synchronous Composite Packet Switching-A Switching Architecture for Broadband ISDN

A new switching architecture for broadband ISDN, "Synchronous Composite Packet Switching (SCPS)," is proposed and evaluated. It efficiently integrates circuit and packet switching functions on a single switching system and accommodates very high speed-up to several tens of Mbit/s-communication services, such as very high speed bursts of data, still picture, and motion video, as well as 64 kbit/s or less voice and data services. The SCPS system comprises plural switch modules and plural Very high speed synchronous loops. In the SCPS system, messages on plural circuit switched channels are assembled into quasi-packets, called "composite packets," and switched synchronously between switch modules, maintaining complete time transparency and short absolute delay time. A system parameter design to obtain high system efficiency and appropriate system modularity is explained, and an example for a very large capacity transit switch of 4 Gbit/s throughput is presented. System implementation problems to realize the SCPS principle, such as efficient implementation of the composite packet assembling and loop transmission functions, are investigated and an experimental system constructed for circuit switching part is presented. The most remarkable characteristic of the SCPS is that it efficiently integrates64 times nkbit/s circuit switching with packet switching. Moreover, the SCPS system retains compatibility with existing networks and the possibiliy of evolution toward a future broadband ISDN. On the basis of the above investigations and experimental system construction, the authors conclude that the SCPS is one of the most practical switching architectures for the coming broadband ISDN era.     

A distributed media access protocol for packet radio networks and performance analysis. Part 1: Network capacity

A distributed time-slot assignment protocol is developed for a mobile multi-hop broadcast packet radio network, using time division multiple access channel access and virtual circuit switching. The protocol eliminates the single point failure mode of centralized network management and the delays of centralized processing. It is applicable to the user-to-user communications functions of such systems as the U. S. Army's enhanced position location and reporting system (EPLRS). The important functions of the distributed protocol, including time-slot assignment, virtual circuit set-up, and network synthesis, are identified, and implementing algorithms are presented and verified. The performance analysis of the protocol is divided into two parts. In this paper, Part 1 of the performance analysis, the capacity of a network using this protocol is studied and a tool is developed to design the network capacity by trading off among the network area, the transmission range, and the number of packet radio units. Since these results are not in closed form, numerical results provide insight into these parameters. In Part 2 the network set-up time and network data rate are analysed and a hierarchical architecture for the distributed protocol is proposed and analysed.     

Elastic basket switching

This paper proposes a new integrated switching system, ‘elastic basket switching’, for broadband and multimedia communications, including voice and high-speed data. In elastic basket switching (EBS), it is possible flexibly and efficiently to handle multimedia information by adaptively assigning communication resources according to communication requests and bandwidth of switched information. For continuous information, such as voice, EBS functions just as a circuit switching system, and for burst data it achieves high-efficiency bandwidth usage equivalent to a packet switching system by demand-assign type time-slot assignment. The detail of EBS and its application to a departmental system-orientated PBX are described. The traffic handling capability and details of the hardware structure are presented. The experimental system, including use of LSIs in the main parts of EBS is also described.     

Architecture, Design, and Development of the System 75 Office Communications System

AT&T-IS System 75 is a new customer-premises digital communications system. It is designed to complement the larger AT&TIS System 85 in the intermediate size range of 20-400 stations and makes use of the same terminals and adjuncts. System 75 utilizes a distributed switching and control architecture and employs VLSI for a compact cost-effective realization. Integrated voice and data communications are supported with full voice conferencing capabilities and high-speed data switching. A process and message-based software architecture supports a large number of features and services, including a flexible user-oriented maintenance and administration capability, accessible from a local or remote system management terminal. This paper describes the system, the switch, common control, software architecture, system management capabilities, and product testing methods.     

On the design of asynchronous optical packet switch architectures with shared delay lines and converters

Optical packet switching (OPS) is a promising technology to enable next-generation high-speed IP networks. A major issue in OPS is packet contention that occurs when two or more packets attempt to access the same output fiber. In such a case, packets may be dropped, leading to degraded overall switching performance. Several contention resolution techniques have been investigated in the literature including the use of fiber delay lines (FDLs), wavelength converters (WCs), and deflection routing. These solution typically induce extra complexity to the switch design. Accordingly, a key design objective for OPS is to reduce packet loss without increasing switching complexity and delay. In this paper, we investigate the performance of contention resolution in asynchronous OPS architectures with shared FDLs and WCs in terms of packet loss and average switching delay. In particular, an enhanced FDL-based and a novel Hybrid architecture with shared FLDs and WCs are proposed, and their packet scheduling algorithms are presented and evaluated. Extensive simulation studies show that the performance of proposed FDL-based architecture outperforms typical OPS architectures reported in the literature. In addition, it shown that, for the same packet loss ratio, the proposed hybrid architecture can achieve up to 30% reduction in the total number of ports and around 80% reduction in the overall length of fiber as compared to the FDL-based architectures.     

Design of transmission and multiplexing systems for broadbandpacket networks

Dynamic time-division multiplexing (DTDM) is a flexible network transport technique capable of handling both continuous and bursty traffic effectively. By using three different multiplexing architectures in the network, DTDM permits graceful evolution of the existing circuit switching network into a flexible broadband packet communications network supporting integrated voice, data, and video traffic. The first multiplexing stage uses a packet assembler to multiplex different broadband services into a common DTDM-format serial bit stream. The second multiplexing stage uses a statistical packet multiplexer to concentrate network traffic for more efficient use of transmission facilities. The third multiplexing stage uses a synchronous time-division multiplexer for high-speed point-to-point transparent transmission. The multiplexer uses a simple tributary synchronization scheme based on positive and negative block justification, which combines the concept of controlled-slip and bit-stuffing techniques while maintaining information integrity. A generic CMOS LSI chip has been designed for use in the three-stage multiplexing system     

3G wireless and cdma2000 1/spl times/ evolution in Korea

The cdma2000/spl reg/ I/spl times/EV-DV system is designed to meet the ever-increasing demand for high-speed packet data transmission while providing the same level of revenue generated by conventional voice communications on existing cdma2000 1/spl times/ systems. It supports concurrent voice and high-speed data on a single cdma 1.25 MHz carrier, and offers improved flexibility for operators to manage data and voice services cost efficiently. However, the Korean market has proved that providing higher bandwidth to the user does not guarantee the success of mobile data service. Users tend not to use expensive mobile wireless data services such as video streaming, video on demand, and MP3 music download service that they feel do not provide enough justification for their costs. The problem LG Telecom faces now as a mobile operator is not only to evolve technologies that enable various services, but also to find a way to provide attractive services at reasonable prices. This article describes experiences in the wireless data market in Korea as well as the market needs and driving forces for cdma2000 1/spl times/EV-DV developments.