AAL2分组话音复接器的研究与实现

该文研究在ATM虚通路带宽利用率一定的条件下,AAL2分组话音复接器性能随ATM虚通路输出速率的增加而变化的情况。得出结论:当ATM虚通路带宽利用率一定时,ATM虚通路输出速率越高,AAL2分组话音复接器的分组丢弃概率和平均分组排队时延越小。并提出了一种AAL2分组话音复接器的实现方案。该方案可以随着ATM虚通路输出速率的增加,方便地复接多个E1话音电路上的话音数据。     

AAL2分组话音复接器缓冲器队列容量的确定方法

该文研究AAL2分组话音复接器缓冲器队列容量的确定方法。提出并从理论上证明用话音分组的最大排队时延为9ms作为确定缓冲器队列容量的标准,可很好地满足分组话音业务服务质量要求的结论,并推导出缓冲器队列容量及门限值的计算公式。仿真结果表明:按作者提出的方法确定缓冲器队列容量及门限值,可获得较低的分组丢弃概率和较小的平均分组排队时延;在满足分组话音业务服务质量要求的前提下,减少了话音分组缓冲器队列的容量,是一种很好的确定缓冲器队列容量和门限值的方法。     

无比特丢弃的AAL2分组话音系统最优分组长度的研究

该文根据分组话音业务的特点,结合分组话音业务服务质量要求,特别是分组丢弃概率和端到端分组传送时延的要求,研究AAL2分组话音系统中AAL2分组最优长度的确定方法,得出结论:对于无比特丢弃的AAL2分组话音系统,当话音采用32kb/s的编码时,AAL2分组的最优长度大约为31个字节;当话音采用16kb/s的编码时,AAL2分组的最优长度大约为27个字节。此时AAL2分组的分组头开销小,话音分组的丢弃概率和端到端分组传送时延低,所得的分组话音质量高。     

带比特丢弃的AAL2分组话音复接器业务模型及性能分析

AAL2主要用于在ATM虚通路上有效传输话音、传真和话带数据,并可利用话音压缩技术和静默检测及消除技术来提高统计复用增益.本文分析了用服务速率随系统状态改变的M/D'/1/K泊松排队系统,为在输入端进行比特丢弃的AAL2分组话音复接器建立模型的缺陷,提出了话音分组到达率随系统状态改变的M'/D/1/K泊松排队模型,并进行了性能分析.计算机仿真证实作者提出的模型是合理的.     

AAL2分组话音复接器缓冲器队列门限值的确定方法

该文研究了带比特丢弃的AAL2分组话音复接器缓冲器队列门限值的确定方法,提出用话音分组作为缓冲器队列门限值的单位,给出了确定门限值的计算公式,并对输出链路容量为384kb/s的情况进行了计算机仿真。仿真结果表明,作者提出的门限值的确定方法可获得较小的平均分组时延和较低的平均分组丢失率,计算简便,易于实现,是一种很好的确定缓冲器队列门限值的方法。     

AAL2分组话音系统信元装配时延的研究

该文研究将AAL2分组填入ATM信元载荷域时的信元装配时延。得出结论:ATM信元装配时延由话音源编码速率、分组占用时长以及接入AAL2分组话音复接器中的话音源个数确定。当话音源编码速率较低,接入AAL2分组话音复接器中的话音源数较小时,信元装配时延可能很大,需要设置定时器以限制信元装配时延,例如当话音源编码速率为8kb/s时,可令定时器的取值为3ms;当话音源编码速率为32kb/s时,若分组占用时长为5ms,一般无需使用定时器。     

AAL2在第三代移动通信系统无线接入网中的应用

ITU－T和ATM论坛推出的AAL2协议可有效承载低比特率对时延敏感的应用业务，并可通过在一条ATM连接上复接多路话音和数据分组来提高网络资源的利用率。文中研究AAL2分组话音系统中的关键技术，并将AAL2与AAL1相比较，指出AAL2在承载话音业务上的独特优势以及在实现过程中需要注意的问题，最后根据AAL2技术的特点研究其在第三代移动通信系统无线接入网中的应用方案。     

多速率话音适配研究与应用

ATM技术以定长分组进行交换和传递,是一种比较理想的承载话音业务的技术。在研究话音数据的适配方式和信元适配时延的基础上,根据AAL2适用于低速率话音和节省带宽的优点,结合工程应用和实践,选择了AAL2的话音适配方式。来自不同编解码和不同话路的话音数据经过AAL2适配,统一成以CID为标识的定长AAL2信元,在信元适配时限内填充成ATM信元,实现了多种速率编码话音包的AAL2承载和AAL2交换。     

一种新的支持多业务的CDMA网络可升降级动态带宽分配与优化策略

未来无线通信网络的主要发展方向是支持多种业务.在3GPP对UMTS的规范中,将业务按其属性对服务质量(Quality of Service,QoS)要求的不同分为4类:会话类、流媒体类、交互类和背景类,除话音业务外其余3种业务都是可变比特速率业务.对该网络用户资源分配(主要是带宽的分配)若采用传统的固定分配方法,必定陷入资源利用率低下和用户QoS得不到保障的两难境地.本文针对宽带CDMA网络,提出了一种针对无线多媒体业务的动态带宽分配与优化策略,在保证用户QoS的前提下,尽可能提高资源利用率.仿真结果表明,对比传统的网络资源管理策略,该策略大大改善了系统的性能,提高了系统资源利用率.     

适合多媒体传输的窄带群路复接器

介绍了一种窄带群路复接器的技术实现方案,重点探讨了复接器电路的模块化设计思想和适合多媒体传输并能在较低速率上传送话音和传真的电路设计。详细讨论了以下几种技术的具体实现方法:采用ATC压缩算法的话音编码速率可达8kbps;结合传真电话自动识别功能,自动恢复出传真信号,能极大地压缩传输带宽;每个数据接口均具有同步、异步传输能力;动态带宽分配技术有效地管理和使用线路带宽;使所研制的群路复接器真正在窄带上实现了数据、话音和传真的综合传输,不仅可扩大原有数字通信系统的容量,而且有利于节约传输费用和增加用户的实际经济效益。     

Voice over ATM using AAL2 and bit dropping: performance and calladmission control

Asynchronous transfer mode (ATM) adaptation layer 2 (AAL2) has been designed for efficient transport of voice, fax, and voiceband data (VBD) traffic over an ATM virtual circuit. The protocol helps achieve low latency and high bandwidth efficiency while applying suitable compression methods on voice/VBD/fax calls and silence elimination on voice calls. We analyze the performance and capacity of an ATM multiplexer based on AAL2 adaptation. We assume that embedded adaptive differential pulse code modulation (ADPCM) is used to compress voice, and silence elimination is used to achieve statistical multiplexing gain. The embedded ADPCM coding scheme allows selective dropping of less significant bits of voice during congestion in the ATM/AAL2 multiplexer. We compare the call capacities of voice multiplexers with and without bit dropping (BD). The performance models and results presented are based on fairly general assumptions and can be used for traffic engineering and call admission control in land-line or wireless ATM systems for a variety of voice/voiceband compression algorithms. A generalized algorithm for call admission control is also described     

Traffic smoothing effects of bit dropping in a packet voicemultiplexer

The performance of a packet voice multiplexer queue in which the less significant bits of voiced packets are dropped during states of congestion in the multiplexer is examined. Using the results of simulation and analytical modeling, it is illustrated that bit dropping of voice packets significantly smooth the burstiness of superposition packet voice traffic by speeding up the packet service rate during critical periods of congestion in the queue. The smoothing effect renders it possible to approximate the superposition by a Poisson process for modeling a packet voice multiplexer with bit dropping. By comparison with a simulation, an analytical model based on the Poisson assumption is shown to produce quite accurate performance predictions. The results indicate that significant capacity and performance advantages are gained in the multiplexer as a result of the bit-dropping scheme     

Analysis and engineering of a voice/data packet multiplexer

This paper investigates performance and engineering issues concerning a multiplexer scheme that has been implemented in AT&T's Integrated Access Terminal (IAT) to transport packetized voice and data traffic on shared facilities. The multiplexer serves voice and data traffic according to a dynamic bandwidth allocation scheme in order to simultaneously meet their performance requirements. A bit-dropping procedure is employed for voice packets to provide a graceful degradation of voice quality under overload conditions. An analytical model is developed for the multiplexer service scheme that estimates performance parameters given the voice and data offered loads. The model is used to demonstrate the capacity advantages of dynamic bandwidth allocation, and to generate load-service curves that illustrate the tradeoffs of carrying different combinations of voice and data traffic on the multiplexer. Sensitivity of voice and data performance to the multiplexer time-slice parameters is also investigated. The model is readily embedded in a design approach that determines the bandwidth required to carry the voice and data traffic demands while satisfying all desired performance objectives     

B—ISDN的AAL1规程及其发展

本文首先从支持面向连接的恒定比特率业务，在收侧恢复源端定量关系，结构数据转移和比特误码及信元丢失的检测与校正等功能要求方面详细介绍了基本的ＡＡＬ１规程，然后以减少时延为为目标讨论了对ＡＡＬ１规程可能的改进方案最后介绍适于支持低比特率话音的候选的ＡＡＬ规程。     

A call admission control scheme for packet data in CDMA cellular communications

In wireless cellular communication systems, call admission control (CAC) is to ensure satisfactory services for mobile users and maximize the utilization of the limited radio spectrum. In this paper, we propose a new CAC scheme for a code division multiple access (CDMA) wireless cellular network supporting heterogeneous self-similar data traffic. In addition to ensuring transmission accuracy at the bit level, the CAC scheme guarantees service requirements at both the call level and the packet level. The grade of service (GoS) at the call level and the quality of service (QoS) at the packet level are evaluated using the handoff call dropping probability and the packet transmission delay, respectively. The effective bandwidth approach for data traffic is applied to guarantee QoS requirements. Handoff probability and cell overload probability are derived via the traffic aggregation method. The two probabilities are used to determine the handoff call dropping probability, and the GoS requirement can be guaranteed on a per call basis. Numerical analysis and computer simulation results demonstrate that the proposed CAC scheme can meet both QoS and GoS requirements and achieve efficient resource utilization.