Integration of Voice and Data in the Wideband Packet Satellite Network

The Wideband (packet satellite) network is an experimental 3 Mbit/s communications system developed under sponsorship of the Defense Advanced Research Projects Agency and the Defense Communications Agency. This system is being used to evaluate the use of packet transmission for efficient voice communication, voice conferencing, and integration of voice and data over a satellite channel. Each station in the Wideband network consists of an earth terminal (dedicated 5 m antenna plus associated IF/RF equipment), a burst-modem and codec unit, and a station controller. Station controllers provide interfaces to host computers (including packet speech sources) and manage the allocation of the satellite channel on a TDMA demand-assigned basis. TDMA demand-assignment is implemented using a reservation-based packet-oriented protocol capableof handling traffic at multiple priority levels. The channel protocol provides a reservation-per-message mode of service (datagrams) to support transmission from bursty traffic sources and a reservation-per-call mode of service (streams) to support traffic with more regular arrival statisticS (e.g., vioce). A distributed scheduler running in every station controller eliminates the need for a central control station and minimizes network transit delay for datagram transmission as well as stream creation, modification, and deletion. In this paper we describe the protocols and mechanisms upon which the Wideband packet satellite network is based.     

The evolution of packet switching

Over the past decade data communications has been revolutionized by a radically new technology called packet switching. In 1968 virtually all interactive data communication networks were circuit switched, the same as the telephone network. Circuit switching networks preallocate transmission bandwidth for an entire call or session. However, since interactive data traffic occurs in short bursts 90 percent or more of this bandwidth is wasted. Thus, as digital electronics became inexpensive enough, it became dramatically more cost-effective to completely redesign communications networks, introducing the concept of packet switching where the transmission bandwidth is dynamically allocated, permitting many users to share the same transmission line previously required for one user. Packet switching has been so successful, not only in improving the economics of data communications but in enhancing reliability and functional flexibility as well, that in 1978 virtually all new data networks being built throughout the world are based on packet switching. An open question at this time is how long will it take for voice communications to be revolutionized as well by packet switching technology. In order to better understand both the past and future evolution of this fast moving technology, this paper examines in detail the history and trends of packet switching.     

High-Speed Packet Switching Systems for Multimedia Communications

High-speed packet switching (HPS) systems can Provide flexible, economical, high-quaiity services for integrated voice, video, and data communications. To realize such HPS systems, methods have been developed to bring about high-speed protocol processing as well as a system architecture for facilitating high-throughput switching. Adopting the parallel processing algorithm into protocol processing allows us to achieve high-speed packet protocol processing of about 100 times faster than conventional processing. Furthermore, a fully distributed system architecture in addition to hierarchical interconnection networks can achieve high-capacity packet switching systems. The proposed HPS system is thus capable of accommodating lines of up to 10-50 Mbits/s, of providing high-throughput switching capability of 1 000 000 packets/s, and of having an average delay of less than 2 ms. Furthermore, an evaluation of network delay performances of video conferencing and voice communications indicate that HPS systems are quite suitable for handling such multimedia communications.     

Admission policies for integrated voice and data traffic in CDMApacket radio networks

The authors derive optimal admission policies for integrated voice and data traffic in packet radio networks employing code division multiple access (CDMA) with direct-sequence spread spectrum (DS/SS) signaling. The network performance is measured in terms of the average blocking probability of voice calls and the average delay and packet loss probability of data messages. The admission scheme determines the number of newly arrived voice users that are accepted in the network so that the long-term blocking probability of voice calls is minimized. In addition, new data arrivals are rejected if the mean delay or the packet loss probability of data exceeds a desirable prespecified level. A semi-Markov decision process (SMDP) is used to model the system operation. Then, a value iteration algorithm is used to derive the optimal admission control. Two models for the other-user interference of the CDMA system are considered: one based on thresholds and another based on the graceful degradation of the CDMA system performance, and their performance is compared. These admission policies find application in emerging commercial CDMA packet radio networks including cellular networks, personal communication networks, and networks of LEO satellites for global communications     

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     

A reservation random-access protocol for voice/data integratedspread-spectrum multiple-access systems

Most code-division multiple-access (CDMA) systems described in the literature provide only one single service (voice or data) and employ the strategy of “one-code-for-one-terminal” for code-assignment. This assignment, though simple, fails to efficiently exploit the limited code resource encountered in practical situations. We present a new protocol called reservation-code multiple-access (RCMA), which allows all terminals to share a group of spreading codes on a contention basis and facilitates introducing voice/data integrated services into spread-spectrum systems. The RCMA protocol can be applied to short-range radio networks, and microcell mobile communications, and can be easily extended to wide area networks if the code-reuse technique is employed. In RCMA, a voice terminal can reserve a spreading code to transmit a multipacket talkspurt while a data terminal has to contend for a code for each packet transmission. The voice terminal will drop a long delayed packet while the data terminal just keeps it in the buffer. Therefore, two performance measures used to assess the proposed protocol are the voice packet dropping probability and the data packet average delay. Theoretical performance is derived by means of equilibrium point analysis (EPA) and is examined by extensive computer simulation     

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.     

IP communications in air traffic control (ATC)

Although established some 60 years ago voice communications is still the sole means to separate and guide aircrafts. Voice communication systems are therefore amongst the most critical installations in air traffic control (ATC). Frequentis was the first to introduce PCM (pulse code modulated) based equipment for ATC worldwide and is on the leading edge of steering into the world of packetized voice communications. VoIP is the emerging voice communication technology which has already proven to provide satisfactory service for commercial applications. However, in ATC a number of requirements exceeding commercial applications have to be met. The voice communication system for ATC integrates radio communication as a prime service. Delays generated by the system therefore directly affect the performance on the radio channel and need to be extremly low. Both the radio control (push-to-talk, PTT) and the voice content need to be processed and delivered in a timely manner. SIP (session initiated protocol) based signalling represents a promising approach to tackle the delay problem. In addition, voice communication systems are required to provide high availability figures. VoIP based systems strongly rely on the communications infrastructure as they are distributed by nature. Resilient packet ring structures allow for these high availablity figures for the communications infrastructure.     

Analysis of Group 4 Facsimile Throughput

The CCITT (Consultive Committee for International Telephone and Telegraph) is developing recommendations for a new generation of facsimile equipment which is designed as Group 4. This class of equipment will transmit an ISO A4-sized page overcommunications networks having error control. Most commercial packet switched and circuit data networks have been designed primarily for the communications of short bursty messages (typically 1000-2000 bits/message between computers and data terminals. The length of a G4 message is forecast to be very long-typically 500 000 bits. There is serious concern that data networks may not handle facsimile traffic very efficiently. This paper projects the near term characteristics of data networks and Group 4 facsimile systems, and estimates the efficiency with which Group 4 messages will be transmitted over three types of data networkspacket switching (PSDN), circuit switched (CSDN), and the public switched telephone network (PSTN). Throughput has been measured by the amount of time required to send a single facsimile page of 500 000 coded bits. In all cases the overhead, as a percentage of the basic facsimile transmission time, is in the range of 50-60 percent. For each network there is a different factor is the halts are forced is transmission as a result of the network window. For the CSDN, the process if converting from voice to data mode is the dominating factor. For the PSTN, packet retransmission due to transmission errors is the dominating factor. The paper includes the assumptions and some of the analytical details of the throughput analysis. Conclusions are drawn regarding the relative transmission efficints through the three types of networks.     

MSR（ITU-T X.87/Y.1324）体系构架与系统特性

针对光城域网对综合业务的电信级需求，以城域网三网业务融合传送为主要领域，以基于弹性分组环RPR实现多业务传送为主要手段，研究了多业务传送与交换的简化方法，介绍了自主国际标准ITU-T X．87／Y．1324的协议要点、组网拓扑结构、传输架构、系统节点的组成和MSR通用帧格式。通过与RPR的比较，得出如下结论：MSR解决了语音、数据和视频等多业务(支路)分别在RPR各节点上，下的传送、支路组播、支路保护和性能监测问题。MSR成本较低，而且在同一网络平台上提供语音、数据和视频等支路业务需要这样的功能，它是目前基于分组的城域网多业务传送和出租的有效方法。     

Highly Efficient Voice—Data Integration over Medium and High Capacity Wireless TDMA Channels

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 and data traffic over two wireless channels, one of medium capacity (referring mostly to outdoor microcellular environments) and one of high capacity (referring to an indoor microcellular environment). Data message arrivals are assumed to occur according to a Poisson process and to vary in length according to a geometric distribution. We evaluate the voice packet dropping probability and access delay, as well as the data packet access and data message transmission delays for various voice and data load conditions. By combining two novel ideas of ours with two useful ideas which have been proposed in other MAC schemes, we are able to remarkably improve the efficiency of a previously proposed MAC scheme [5], and obtain very high voice sources multiplexing results along with most satisfactory voice and data performance and quality of service (QoS) requirements servicing. Our two novel ideas are the sharing of certain request slots among voice and data terminals with priority given to voice, and the use of a fully dynamic low-voice-load mechanism.     

Economic analysis of integrated voice and data networks: A case study

This paper describes the results of a study to evaluate alternative switching strategies for future integrated voice and data networks. Three fundamental problems are addressed: 1) the economics of integrating voice and data applicatiom in a common communications system; 2) the comparison of alternative switching technologies for integrated voice and data networks; 3) the cost-effectiveness of alternative voice digitization rates and strategies. Three broad switching technologies and variations thereof are investigated and compared. The switching technologies are: circuit switching, packet switching, and hybrid (circuit-packet) switching. Each switching technology can accommodate either voice or data applications separately or combined voice and data requirements in an integrated fashion. Results of studies regarding communications systems option are provided and the sensitivity of the results are tested with respect to traffic variations, cost trends of switching and transmission, and network performance variables. The significant variables which affect the results are identified and quantified. The intent of this study is to identify and quantify network technologies which demonstrate long-term low operating costs, This is a necessary effort to provide the basis for determining the most cost-effective evolutionary path for future communication systems. It is recognized that transition problems and associated costs may be other important factors determining the ultimate evolutionary path. However, determining these costs was not an objective of this study. Neverthless, this study provides a framework and a target technology for detailed evolution, planning, and cost analysis.     

OpenFlow网络测量分析系统的设计实现

OpenFlow网络目前缺少支持定量测量分析各种创新应用或机制的有效手段。以升级OpenFlow网络设备为具有本地日志功能的OpenFlow测量实体为基础,设计了一种基于集中式服务器控制测量实体进行分布式测量的机制,制定了其间的通信规程OpenFlow测量控制协议(OMCP),同时基于正则表达式、散列技术和可扩展的统计函数库等方式设计了一种分析测量日志的功能。原型系统的实验表明,OpenTrace服务器能够灵活部署和控制分布式测量任务,OpenTrace系统不仅能够定量地重现数据平面的数据流传输过程而且能够重现控制平面的控制事件交互过程,从而可为量化分析OpenFlow网络应用和新型机制提供广泛的性能数据。     

Integrated voice and data communications overview

The author reviews architectures and traffic characteristics for voice and data communications, and addresses important issues in integrated voice and data communications. He discusses some possible methods of integrating voice and data and presents an example of implementing integrated voice and data communications. He considers ISDN (integrated services digital network) as a vehicle for supporting integrated services including voice and data as well as image and video. Although the concern is mainly with switched networks, some of the concepts discussed are applicable to both switched and special-services networks     

Performance analysis of channel de-allocation schemes for dynamic resource allocation in GSM/GPRS networks

Channel de-allocation for GSM voice call (DASV) has been considered for dynamic resource allocation in GSM/GPRS networks. Two new de-allocation schemes are proposed: de-allocation for GPRS packet (DASP) and de-allocation for both GSM voice call and GPRS packet (DASVP). An analytic model with general GPRS data channel requirement is derived to evaluate the performance of the schemes in terms of GSM voice call incompletion probability, GPRS packet dropping probability, average GPRS packet transmission time and channel utilisation.     

Packetized voice/data integrated transmission on a token passingring local area network

The performance of a token-passing ring network with packetized voice/data mixed traffic is investigated through extensive simulations. Both data and voice users are modeled in the simulations. Data users produce bursty traffic. Voice traffic is modeled as having alternating talkspurts and silences, with generation of voice packets at a constant rate during talkspurts and no packet generation during silence periods. Token passing ring local area networks are shown to effectively handle both voice and data traffic. The effects of system parameters (e.g. voice packet length, talkspurt/silence lengths, data traffic intensity, and limited exhaustive service disciplines) on network performance are discussed     

Universal Digital Portable Communications: A System Perspective

In our highly mobile society, the provision of voice and data communications to a person away from his/her wireline telephone has become a major communications frontier. The early penetration of this frontier has been based on very limited portable communications approaches, e.g., cordless telephones, mobile radio telephones, and radio paging. Each of these approaches only partially satisfies portable communications needs. This paper describes an approach to providing universal digital portable communications integrated into telephone networks. A system configuration employing time-division multiple-access radio link architecture and frequency reuse is described. Issues affecting radio link transmission rates and radio system coverage are discussed. Characteristics and parameters of a possible system to supplement the wire (or fiber) loop are indicated.     

Probability distributions of interdeparture time and response timein multipacket CSMA/CD systems

An exact analysis is presented of an asynchronous (unslotted) multipacket CSMA/CD-DFT (carrier-sense multiple-access/collision detection with delayed first transmission) model and used to derive (1) the Laplace-Stieltjes transform of the probability distribution function (PDF) of the packet interdeparture time, (2) the LST of the PDF of the message interdeparture time, (3) the moment generating function (MGF) of the message response time, and (4) the MGF of the message response time conditioned on the number of packets necessary for transmission of a message. Higher moments as well as averages of these performance measures are important for performance evaluation of interconnected networks or integrated networks of voice and data. As the performance model allows each message to consist of a geometrically distributed number of constant-length packets, the result is also useful in examining the performance of error-prone CSMA/CD systems as well as the impact of random message sizes on the system performance. Numerical examples include comparisons between the DFT model and the IFT (immediate-first-transmission) model     

Expressnet: A High-Performance Integrated-Services Local Area Network

Expressnet is a local area communication network comprising an inbound channel and an outbound channel to which the stations are connected. Stations transmit on the outbound channel and receive on the inbound channel. The inbound channel is connected to the outbound channel so that all signals transmitted on the outbound channel are duplicated on the inbound channel, thus achieving broadcast communication among the stations. In order to transmit on the bus, the stations utilize a distributed access protocol which achieves a conflict-free round-robin scheduling. This protocol is more efficient than existing round-robin Schemes as the time required to switch control from one active user to the next in a round is minimized (on the order of a carrier detection time), and is independent of the end-to-end network propagation delay. This improvement is particularly significant when the channel data rate is so high, or the end-to-end propagation delay is so large, Or the packet size is so small as to render the end-to-end propagation delay a significant fraction of, or larger than, the transmission time of a packet. Moreover, some features of Expressnet make it particularly suitable for voice applications. In view of integrating voice and data, a simple access protocol is described which meets the bandwidth requirement and maximum packet delay constraint for voice communication at all times, while guaranteeing a minimum bandwidth requirement for data traffic. Finally, it is noted that the voice/data access protocol constitutes a highly adaptive allocation scheme of channel bandwidth, which allows data users to recover the bandwidth unused by the voice application. It can be easily extended to accommodate any number of applications, each with its specific requirements.     

The DARPA internet protocol suite

THE MILITARY requirement for computer communications between heterogeneous computers on heterogeneous networks has driven the development of a standard suite of protocols to permit such communications to take place in a robust and flexible manner. These protocols support an architecture consisting of multiple packet switched networks interconnected by gateways. The DARPA experimental internet system consists of satellite, terrestrial, radio, and local networks, all interconnected through a system of gateways and a set of common protocols.