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.     

A Protocol for Packet Network Intercommunication

A protocol that supports the sharing of resources that exist in different packet switching networks is presented. The protocol provides for variation in individual network packet sizes, transmission failures, sequencing, flow control, end-to-end error checking, and the creation and destruction of logical process-to-process connections. Some implementation issues are considered, and problems such as internetwork routing, accounting, and timeouts are exposed.     

A Combined Forward Error Control and Multiple Access Protocol for Wireless Voice/Data Integration

Different strategies for error control are investigated, in conjunction with a MAC-level protocol, operating in a multimedia noisy cellular environment, where a base station co-ordinates mobile users within each cell. The channel multiplexing structure is based on Time Division (TDM), and the slots in each frame are dynamically assigned to the users and their service classes. The access rights are decided by an algorithm, namely, the Independent Stations Algorithm (ISA), at the cell base station, which broadcasts this information to the users. Each single mobile station is associated a bit error probability, which changes with time in a random fashion. Bit error probabilities are estimated and the protection level of the Forward Error Correction (FEC) code is dynamically adapted to the new conditions. The performance of the scheme is analyzed by simulation in the presence of mixed voice and data traffic, and with different error handling techniques (FEC and Automatic Retransmission Request (ARQ)).     

Stability and Optimization Analysis for the PRMA Protocol for Voice and Burst Data Integration in a Personal Communication Network

Personal Communication Networks (PCNs) have become an important filed of activity and they are expected to play a fundamental role in next-generation telecommunication systems. In particular, the emerging personal communication services, such as portable computing, paging, personal e-mail, etc. have led to the need of an efficient integration of voice and data traffic in a same PCN by means of a suitable multiple-access scheme. This paper considers stability and optimization issues for the Packet Reservation Multiple Access (PRMA) protocol used for integrating voice and burst data traffic in a PCN. An important result here devised is that the optimized PRMA protocol permits to be achieved good performance even under heavy traffic load conditions.     

Packetized Service Integration Network for Dedicated Voice/Data Subscribers

This paper describes a new integrated service network using packetized voice/data technology which was developed in consideration of the sophisticated service demands for voice, data, and facsimile communication. A major problem in constructing this network is in the local distribution network, i.e., the subscriber interface unit, the local end office, and the subscriber loop which connects them. The concepts, choice of design parameters, and laboratory test results of the packetized service integration network are briefly discussed.     

分组卫星话音／数据综合的混合多址协议性能研究

本文讨论了一种用于分组话音／数据综合的混合多址协议，该协议是固定分配与随机访问的混合，并且赋予话音分组优先传输权，从而保证了无重传话音分组有较小的丢失概率。本文进行了详细的理论分析，并得出了性能比较特性，所得结果认为这是一个兼顾话音／数据综合的较好协议，且具有一定的灵活性。     

无冲突NC-IPRMA协议下的数据--话音综合

提出了一个面向微蜂窝通信的支持数据─话音综合业务的无冲突分组预约多址协议─—NC-IPRMA协议,通过M／M／N／／M．排队模型分析了NC-IPRMA协议的话音子系统性能,通过系统仿真分析了数据一话音综合性能。研究结果表明,在支持数据─话音综合业务方面,NC-IPRMA协议比PRMA和IPRMA协议更高效,这包括更高的信道利用率,更短的数据分组时延和更大的话音终端容量。由于采用排队等待方式接入时隙信道,NC-IPRMA协议彻底消除了以竞争方式接入信道的PRMA协议（及其各种改进方案）存在的重负荷下的不稳定问题。     

A Channel Access Protocol for Integrated Voice/Data Applications

A new multiaccess protocol is proposed for an integrated voice/data application. The protocol, which is a variation of virtual time CSMA (VT-CSMA), takes advantage of the periodicity of voice packets and possesses a number of important features. With this protocol, voice stations appear to have a dedicated time-division multiplexed (TDM) slot, and the delay of a voice packet is bounded by the length of a frame (defined to be the period between two consecutive voice packets from a voice station). Also, the amount of data added to the channel has little effect on the voice traffic. When silence detection is used, many more voice conversations can be supported without losing the dedicated-slot characteristic. This is in contrast to a movingboundary TDM system where the excessive bandwidth saved by silence detection can only be used for data. The protocol requires no global synchronization and is easy to implement. Simulation results are presented to evaluate its performance.     

多媒体终端中声音和数据的集成传输

本文描述了采用包复用方式在固定带宽内集成传输声音和多媒体数据的多媒体终端通信系统，系统中的声音编码采用了静默检测技术，声音编码的速率可以根据信道的拥挤程度在３２ｋｂｉｔ／ｓ和１６ｋｂｉｔ／ｓ之间动态地变化。本文提出了一种利用增减静默抽样来同步声音编解码时钟的方法，本文还提出了利用数据队列的短时平均长度来判断信道繁忙程度的算法，在多媒体数据突发性强、数据量大时，该算法比利用声音或数据队列的瞬时长度判断更为准确。     

Characterizing Superposition Arrival Processes in Packet Multiplexers for Voice and Data

This paper analyzes a model of a multiplexer for packetized voice and data. A major part of the analysis is devoted to characterizing the aggregate packet arrival process resulting from the superposition of separate voice streams. This is done via the index of dispersion for intervals (IDI), which describes the cumulative covariance among successive interarrival times. The IDI seems very promising as a measurement tool to characterize complex arrival processes. This paper also describes the delays experienced by voice and data packets in the multiplexer using relatively simple two-parameter approximations.     

数据和话音业务的融合与固定网和移动网的结合

以电话业务为主的电路交换网络正面临着以数据为主的ＩＰ网络的竞争。传统的话音业务正面对话音、数据、视频等多媒体业务需求的挑战。移动通信与Ｉｎｔｅｒｎｅｔ的高速发展为各种新的移动Ｉｎｔｅｒｎｅｔ增值业务的推出提供了新的商机。根据预测分析,到２００３～２００４年,全球移动电话用户数将和固定电话用户数接近;到２００５年,全球数据业务量将占整个通信业务量的绝大部分。为了适应市场发展和用户的需求,数据和话音业务的融合、固定网络和移动网络的结合,乃至有线通信运营者和移动通信运营者的合作与互相渗透,将成为新世纪…     

Optimal Packet Length for Packet Voice Communication

It is shown that for typical operating parameters, the optimal packet length for a single link packet voice system is of the order of 300-700 bits. This is contrary to both the optimal length of a data packet (approximately 1000 bits), and documented experimental implementations of such network architecture for voice (approximately 1000 bits).     

谈谈高速分组网简单数据链路(SDL)协议

介绍一种新的成帧协议即简单数据链路协议(SDL),它以使用长度指示符和报头CRC为基础,特别适合于IPoverSDH以及下一代宽带化、多业务的Internet网,然后具体分析了用于SDL中的新的扰码器,评估了SDL的各种性能.     

BGP-S: A Protocol for Terrestrial and Satellite Network Integration in Network Layer

To accomplish network layer integration of terrestrial and satellite IP networks, special exterior gateway protocols are needed. In this work, a new exterior gateway protocol called Border Gateway Protocol – Satellite version (BGP-S) is introduced that enables automated discovery of paths that go through the satellite network. This protocol is designed to work in only one terrestrial gateway in every Autonomous System and enables the forwarding of discovered paths in the Internet using the BGP-4 protocol. The performance of BGP-S is investigated through simulations.     

Packet Reservation Window Multiple Access for Microcellular Voice/Data Transmission

We propose a new packet reservation multiple access (PRMA) scheme for the joint transmission of voice and data traffics in a microcellular medium. The collision resolution protocol within the system is based on a modification of the window random access algorithm, which has superior properties compared to the conventional slotted Aloha. The proposed algorithm, which we call packet reservation window multiple access (PRWMA), works in distinct modes for voice and data without prioritization, and the user performs slightly different operations depending on the information type. Simulation results show that PRWMA outperforms PRMA by a significant margin in terms of voice user capacity.     

用UDP协议实现网络语音广播

本文介绍了用UDP协议实现局域网中语音通信的基本原理,基本流程和软件实现,以及在消防监控系统中的应用.     

Techniques for Packet Voice Synchronization

Packet switching has been proposed as an effective technology for integrating voice and data in a single network. An important aspect of packet-switched voice is the reconstruction of a continuous stream of speech from the set of packets that arrive at the destination terminal, each of which may encounter a different amount of buffering delay in the packet network. The magnitude of the variation in delay may range from a few milliseconds in a local area network to hundreds of milliseconds in a long-haul packet voice and data network. This paper discusses several aspects of the packet voice synchronization problem, and techniques that can be used to address it. These techniques estimate in some way the delay encountered by each packet and use the delay estimate to determine how speech is reconstructed. The delay estimates produced by these techniques can be used in managing the flow of information in the packet network to improve overall performance. Interactions of packet voice synchronization techniques with other network design issues are also discussed.     

Performance of Integrated Packet Voice/Data Token-Passing Rings

This paper presents performance results that indicate that packetized voice service can be provided on a token-passing ring without adversely affecting the performance of data traffic. This is accomplished by introducing a relatively mild priority structure: stations are limited to a single packet transmission per medium access, and voice packets are given access priority over data packets at the same station. In addition, voice traffic is allowed longer packet lengths than data traffic. Several versions of this basic scheme are considered: 1) the number of active stations is constrained so that voice packets are guaranteed access within one packetization period, 2) no guarantee on access time is provided and voice packets are discarded when the waiting time exceeds one packetization period, and 3) no guarantee on access time is provided and voice packets are buffered until they can be transmitted.     

A Medium Access Control Protocol for Voice/Data Integrated Wireless CDMA Systems

In this paper, a medium access control protocol is proposed for integrated voice and data services in wireless local networks. Uplink channels for the proposed protocol are composed of time slots with multiple spreading codes per slot based on slotted code division multiple access (CDMA) systems. The proposed protocol uses spreading code sensing and reservation schemes. This protocol gives higher access priority to delay‐sensitive voice traffic than to data traffic. The voice terminal reserves an available spreading code to transmit multiple voice packets during a talkspurt. On the other hand, the data terminal transmits a packet without making a reservation over one of the available spreading codes that are not used by voice terminals. In this protocol, voice packets do not come into collision with data packets. The numerical results show that this protocol can increase the system capacity for voice service by applying the reservation scheme. The performance for data traffic will decrease in the case of high voice traffic load because of its low access priority. But it shows that the data traffic performance can be increased in proportion to the number of spreading codes.     

有线数字电视光分组交换网络物理层协议配置

从光分组的产生、编码方案、光分组的再生、光分组同步以及光分组交换的流量分配等入手,讨论了下一代有线数字电视光分组交换网络的物理结构、物理层协议配置及光-电、电-电的多种优化设计方案.