用OPNET研究无线Ad Hoc的语音传输性能

探讨了用仿真工具OPNET来研究语音分组在无线Ad Hoc网络传输特性。通过用OPNET网络仿真工具分析了三种不同场景下,语音分组在无线Ad Hoc网络中传输质量的变化,及对三个不同场景下语音分组传输质量进行了分析比较,得到一些有用的结论,如：验证了用OPNET仿真工具研究无线Ad Hoc网络语音传输特性的有效性和可行性;在无线Ad Hoc网络中,随着跳数的增加,语音传送质量会随着跳数的增加而变差等有用结论。     

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。     

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。     

QoS requirements for a voice-over-IP PSTN

Voice over IP is already widespread in enterprise private networks and is growing in public switched voice networks as manufacturers withdraw support for earlier technologies. Packet transmission of voice can introduce new impairments, including packet loss, extra sources of delay, and the use of compressed speech coding, all of which may affect voice quality delivered to the user. Factors affecting the quality of a voice telephony connection are described, concentrating on those which are changed by the move to packet transmission, including the complex area of delay. We outline subjective testing based on users’ opinions of fragments of recorded audio material or of connections realised in a laboratory, and describe the abstraction of these results into transmission planning models to assist with design of networks and their QoS mechanisms. QoS requirements are stated for a packet technology to support a PSTN and ISDN service in the UK telecommunications environment.     

Integrated voice/data multiplexers with slotted and nongated packetservices

Integrated voice/data multiplexers that provide packet services for both voice and data traffic are discussed. A slotted service is assumed, so that packet transmissions are synchronized to slot boundaries. Nongated service, in which packets are transmitted as soon as the transmission capacity becomes available, is also assumed. The performance of nongated and slotted multiplexers is obtained by analytic and simulation approaches. In particular, a PRIO (head-of-the-line priority to voice packets) and a BVFD (busy-voice, fixed-data) multiplexer are shown to be suitable for such a nongated environment     

Coded Speech in Packet-Switched Networks: Models and Experiments

This paper considers the possibility of introducing packetized voice traffic into a packet-switched network. It is well known that the network must assure voice packets sufficient delay characteristics for conversational speech, i.e., low delay between speaker and listener and low delay jitter or variance. To reach these goals, simplified protocols and priority rules for voice handling are proposed and evaluated. A model of a packet switching node structure capable of handling both data and voice is derived for both analytical and simulation approaches. The use of low bit rate voice encoders is considered. The necessity of avoiding the transmission of silent intervals is discussed in relation to the behavior of packet voice receivers. Proposed strategies are compared by means of analytical tools and simulation experiments considering the presence of voice, interactive, and batch data packets.     

Models for Analysis of Packet Voice Communications Systems

In a packet voice communication system, packets are fed to a common queue by a number of independent voice sources and are removed from this queue on a first-come-first-serve basis for transmission over a communication link of finite capacity. Each voice source alternates between active periods, during which packets are generated at regular intervals, and inactive periods, during which no packets are generated. In this paper, we discuss three models, a semi-Markov process model, a continuous-time Markov chain model, and a uniform arrival and service model, to assess the queueing behavior of such systems. Numerical results obtained from each of the three models are compared to each other, to results obtained from a discrete event simulation program, and to results obtained from anM/D/1analysis. Parameters of the model are the average duration of active and inactive periods, the packet generation rate, the communication link capacity, and the total number of voice sources. Conclusions are drawn regarding which models appear to be most appropriate in the parameter ranges investigated.     

关于VOLTE语音质量分组丢失率高小区整治的研究

阐述4G语音分组丢失对用户感知的影响,通过大数据对语音分组丢失进行四维五域分析定界,聚焦在“单通”、“断续”、“音质“三个影响用户感知的现象,最后根据VoLTE语音质量定界法,对无线侧分组丢失问题这一影响用户感知主要原因进行分析,并列出影响分组丢失最常见因素及优化解决方法。     

Implementing VoIP: a voice transmission performance progress report

Aiming to introduce voice over IP networks and services in ways that satisfy the voice quality expectations of our customers, we have been conducting laboratory studies of how VoIP transmission affects voice quality while also carefully monitoring and managing several field implementations of VoIP. This article summarizes much of what we have learned in this work, and we hope it provides a useful progress report on the industry's evolution to VoIP. We review our data on the voice quality effects of packet loss, delay, speech coders, packet loss concealment algorithms, and the compression option of suppressing transmission during silence. Because the familiar problem of echo has emerged repeatedly in the VoIP environment, we review this issue in some detail. Packet loss and delay variation measurements made on private VoIP networks are reviewed, and the data here are encouraging. We finish by making our case that the network planning tool known as the E-model is currently an inexact predictor of VoIP network performance.     

Output and delay of multi-channel slotted ALOHA systems for integrated voice and data transmission

Packet-switched technology has been developed to offer personal communication services not only for data but also for different types of user-end equipment such as phone-type audio. To satisfy the huge service demand and multi-traffic requirements with limited bandwidth, this paper proposes an efficient procedure of multi-channel slotted ALOHA for integrated voice and data transmission in wireless information networks and presents an exact analysis with which to numerically evaluate the performance of the systems. A channel reservation policy is applied, where a number of channels (called reserved channels) are used exclusively by voice packets, while the remaining channels are used by both voice and data packets, and voice packets select the reserved channels with a given probability (called selection probability). Probability distributions for the numbers of voice and data departures and for the data packet delay are derived. Numerical results compare some cases with different numbers of channels, different numbers of reserved channels and different selection probabilities to discuss what effects they may have on channel utilization, loss probability, average packet delay, coefficient of variation of data packet delay, and correlation coefficient of packet departures. This revised version was published online in June 2006 with corrections to the Cover Date.     

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.     

Performance of packet voice transmission using IEEE 802.16 protocol

The IEEE 802.16 standard defines three types of scheduling services for supporting real-time traffic, unsolicited grant service (UGS), real-time polling service (rtPS), and extended real-time polling service (ertPS). In the UGS service, the base station (BS) offers a fixed amount of bandwidth to a subscriber station (SS) periodically, and the SS does not have to make any explicit bandwidth requests. The bandwidth allocation in the rtPS service is updated periodically in the way that the BS periodically polls the SS, which makes a bandwidth request at the specified uplink time slots and receives a bandwidth grant in the following downlink subframe. In the ertPS service, the BS keeps offering the same amount of bandwidth to the SS unless explicitly requested by the SS. The SS makes a bandwidth request only if its required transmission rate changes. In this article we study the performance of voice packet transmissions and BS resource utilization using the three types of scheduling services in IEEE 802.16-based backhaul networks, where each SS forwards packets for a number of voice connections. Our results demonstrate that while the UGS service achieves the best latency performance, the rtPS service can more efficiently utilize the BS resource and flexibly trade-off between packet transmission performance and BS resource allocation efficiency; and appropriately choosing the MAC frame size is important in both the rtPS and ertPS services to reduce packet transmission delay and loss rate     

An analysis of the performance of ARQ schemes for integrated traffic satellite-switched SFH and DS CDMA networks

The delay and throughput performance of satellite-switched Slow Frequency Hopping CDMA network for simultaneous voice and data transmission is analyzed and compared to that of a DS-CDMA system. Two ARQ schemes are suggested for data while Forward Error Correction using the same encoder is used for voice packets. The queueing analysis assumes priority for voice and two models for voice traffic are used (Markovian and IPP). The probability of successful packet transmission is derived for all systems as a function of traffic load allowing us to evaluate the systems using delay, throughput, and voice packet loss as figures of merit. Numerical results show that while voice delay is minimal DS CDMA is much more effective then SFH CDMA in all cases. One interesting result is that SFH systems perform better with S/W schemes and achieve a higher maximum throughput. It is also observed that the IPP and Markovian models gave similar results.This work was supported by an NSERC CRD (Collaborative Industrial Research and Development grant,) with Spar Aerospace, Quebec, Canada     

Discrete-Time Analysis of Integrated Voice/Data Multiplexers With and Without Speech Activity Detectors

Discrete-time analysis of two schemes for multiplexing voice and data is presented. In each scheme voice and data are multiplexed using the movable boundary frame allocation scheme. In the first scheme, speech activity detectors (SAD's) are not used, and hence, the variations in the voice traffic are only due to the on/off characteristics of voice. In the second scheme, SAD's are employed so that talker silences can he utilized for transmission of additional voice and/or data. In this scheme, the multiplexer performs digital speech interpolation as well as movable boundary frame allocation. The performance measures considered are probability of loss for voice calls, probability of speech clipping, speech packet rejection ratio, and the expected data message delay. In the case of the multiplexer with SAD, a tradeoff exists between data message delay and speech interpolation advantage. Some numerical examples are presented which illustrate the performance of the two multiplexers.     

一种用于北斗大容量通信的丢包重传机制的实现

我国北斗三号卫星导航系统的建立,解决了北斗一号和北斗二号报文通信功能存在的局限性,不仅通信速率得到了极大提升,而且通信容量扩充了近10倍,可支持语音、图片等大容量信息的传输,但是由于语音等信息容量较大,在传输之前需要经过压缩编码和数据分包重组,由于北斗卫星链路容易受到大气条件、电磁干扰等因素影响,容易出现数据丢包,因此...     

车用自组网媒体访问控制机制改进

为提升车用自组网传输音频、视频的服务质量,对基于IEEE802.11p的车用无线接入技术MAC机制进行改进,提出竞争窗口自适应EDCA机制。仿真实验表明,竞争窗口自适应EDCA机制有效地降低了车用自组网中音频、视频流的传输时延、时延抖动和丢包率,保证了车用自组网传输VoIP、视频会议、音视频流媒体等多媒体业务的服务质量。     

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     

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     

Anti-packet-loss joint encoding for voice-over-IP steganography

VoIP (voice over IP) is a kind of voice communication technology based on UDP/IP protocols. Packet loss will inevitably happen when the channel environment deteriorates, which can pose challenges to the reliable transmission of VoIP steganography. A steganographic model based on joint encoding was proposed. In this model, packet erasure coding was introduced to preprocess the secret data. And the encoded data were embedded into voice packets with minimum dis-tortion using matrix embedding. Furthermore, the influences of key parameters on the performance of joint coding were studied. The selection algorithm for optimal parameters was also given. Experimental results show that the proposed joint coding can effectively improve steganographic resistance to packet loss, and decrease the number of modifications to the voice stream.