改进IP电话传输质量

简要介绍Ｉｎｔｅｒｎｅｔ电话的发展状况，指出其发展中所要解决的问题，着重分析了网络中影响话音传输质量的因素，并采用模拟的方法对它们作出评价。最后，提出了提高话音传输质量的一种新方法和改进话音质量的建议。     

基于DSP＋ARM＋TINI平台的网络话音终端设计

为了解决分布式通信系统中话音信号点对点和点对多点传输问题,提出了以IP网络为基础的话音终端设计。从硬件及软件两个方面分别介绍了基于DSP ARM TINI平台的网络话音终端的设计,充分克服网络传输的弊端,有效保证网络传输质量,提高话音数据播放音质。     

IP电话中的差错控制技术

ＩＰ电话在传送话音分组时,为了获得较小的时间延迟,所使用的传输协议一般不保证话音分组的可靠传递。本文介绍在应用层改善话音质量方面的研究情况。首先分析了影响ＩＰ电话话音质量的主要原因,然后比较了几种差错控制技术的优缺点,最后着重研究了目前普遍被看好的话音分组差错控制技术──前向纠错（ＦＥＣ）机制的基本思想和实现方法。     

IP电话中的差错控制技术

ＩＰ电话在传送话音分组时,为了获得较小的时间延迟,所使用的传输协议一般不保证话音分组的可靠性传递。本介绍在应用层改善话音质量方面的研究情况。首先分析了影响ＩＰ电话话音质量的主要原因,然后比较了几种差错控制技术的优缺点,最后着重研究了目前普遍被看好的话音分组差错控制技术－－前向纠错（ＦＥＣ）机制的基本思想和实现方法。     

通信系统延时对话音质量的影响

论文在介绍了话音的是在数的基础上，得到了话音的时间参数与可觉察的最小响应时间增量的关系；介绍了延时对话音质量的技术评定，得到了各种话音的觉察门限，主观印象等级分和传输效率；最后介绍了移动卫星通信系统中传输延时情况。     

传输损伤对话音通信质量的影响

本文主要分析了影响话音传输性能的各种损伤因素,对损伤因素在网络中的分配和损伤因素对话音质量的综合评价提出了一些思路。     

基于以太网的数字话音传输系统设计

针对在工程应用中如何通过以太网进行话音传输提出了一种设计方案,分析了话音在网络上传输的特点,介绍了一种基于以太网的数字话音传输系统方案。系统以自带网络协议的嵌入式ARM微控制器LM3S9B96为核心平台,采用IP上传送语音（Voice over IP,VoIP）技术实现话音的以太网传输。对系统的话音实际传输效果进行了仿真测试分析,结果表明,话音清晰、失真度和时延小,整体性能满足实际话音通信的要求。     

机载移动通信卫星中继链路的高效传输方法

为了提高机载移动通信卫星中继链路的数据传输效率,给出了一种高效传输方法,先通过转换话音编码方式对语音进行压缩,再把话音数据作为有效负载封装成IP数据包,采用RTP复用和IP/UDP/RTP报头压缩等技术来提高话音数据的传输效率.最后通过仿真说明了这种高效传输方法能提升给机载移动通信的性能.     

基于AC48624+PXA270的话音编解码设计

针对PCM话音在专用网络中传输占用带宽大、传输效率低的问题,文章提出了一种基于AC48624+PXA270的话音编解码设计,可以实现对24路PCM话音的G729A编码压缩,有效降低了话音信号的传输带宽,提高了传输效率。     

VoIP话音质量分析与测试

随着VoIP发展的日趋成熟,VoIP单凭粗放低价进行市场竞争日趋艰难,更多人把关注点投向VolP服务质量.首先从VoIP的传输层面和业务层面分析影响VoIP话音质量的主要因素,然后提出相应的应对方法,最后提出测试与评估VoIP话音质量的指标体系以及测试方法.     

VoIP transmission mechanism based on TCP

As the widespread employment of firewalls on the Internet, user datagram protocol (UDP) based voice over Internet protocol (VoIP) system will be unable to transmit voice data. This paper proposed a novel method to transmit voice data based on transmission control protocol (TCP). The method adopts a disorder TCP transmission strategy, which allows discontinuous data packets in TCP queues read by application layer directly without waiting for the retransmission of lost data packets. A byte stream data boundary identification algorithm based on consistent overhead byte stuffing algorithm is designed to efficiently identify complete voice data packets from disordered TCP packets arrived so as to transmit the data to the audio processing module timely. Then, by implementing the prototype system and testing, we verified that the proposed algorithm can solve the high time delay, jitter and discontinuity problems in standard TCP protocol when transmitting voice data packets, which caused by its error control and retransmission mechanism. We proved that the method proposed in this paper is effective and practical.     

Design and performance analysis for data transmission in GSM/GPRSsystem with voice activity detection

To efficiently utilize the bandwidth of cellular mobile systems and offer service of high quality to both voice and data users, we propose a protocol to integrate packet-switched data traffic into current time-division multiple-access (TDMA)-type circuit-switched digital voice systems. We analyze the performance of the proposed system, which transmits data packets in the silent periods of a conversation with voice activity detection and adapts itself to the GSM/GPRS system, which uses the idle channels to provide data services. We show that the proposed protocol can increase the bandwidth utilization efficiency and improve the throughput/delay performance of the data transmission while minimizing the impact on the current GSM/GPRS service     

A packet reservation multiple access (PRMA)-based algorithm for multimedia wireless system

The major issue in the wireless multimedia system design is the selection of a suitable channel sharing media access control (MAC) protocol. The design challenge is to identify a wireless "multimedia capable" MAC protocol that provides a sufficient degree of transparency for many different kinds of services. This protocol should guarantee different quality of service (QoS) parameters for different types of traffic while in the same time achieving high throughput. In this paper a MAC protocol to serve different kinds of traffic, namely voice, data, and, real time variable bit rate (rt-VBR) video is proposed. The transmission time scale is divided into frames. Each frame is subdivided into N time slots. In this protocol, a fixed number of slots M out of 150 time slots are reserved at the beginning of every frame to transmit some of the video packets arriving during the frame interval. The rest of the video packets contend with the voice and data packets for the remaining time slots of this frame as in normal packet reservation multiple access (PRMA). One objective of this paper is to find the optimum value of M allowing the maximum number of voice and data users to share the RF channel with one video user. Another objective is to find the optimum permission probabilities of sending contending voice, data, and video packets allowing the maximum number of users sharing the RF channel. The dropping probability requirement for video is examined.     

A Priority Based Opportunistic Routing Mechanism for Real-Time Voice Service in Mobile Ad hoc Networks

Traditional routing protocols send traffic along pre-determined paths and have been shown ineffective in coping with unreliable and unpredictable wireless medium which is caused by the multi-path fading. The most difference between the opportunistic routing and the traditional routing mechanism is that the opportunistic routing mechanism can use several lossy broadcast links to support reliable transmission. In this paper, an opportunistic routing mechanism for real-time voice service is proposed. This mechanism is based on the dynamic source routing (DSR) protocol with some modifications, the routing messages of DSR are used to construct the forwarder list, which guides the data packets forwarding process. The forwarder nodes have priorities to restrict the number of duplicated packets. Simultaneous flows can be supported well by our mechanism. Simulations show that our mechanism can effectively decrease the data packets transmission times and the amount of the control messages and reduce the end-to-end delay for real-time voice service, the quality of service can be supported well over the unstable wireless channel.     

Medium access control protocol for the integration of voice/data on a local area network

The analysis and performance of an integrated medium access protocol based on R-VT-CSMA for packet voice and data transmission on a broadcast-bus LAN is examined. This protocol, called adaptive R-VT-CSMA uses variable length packets and the strong correlation between successive voice frames to increase the throughput.<>     

Block Reservation Time Division Multiple Access (BRTDMA) Protocol for a High Capacity Wireless Network

Next generation high capacity wireless networks need to support various types of traffic, including voice, video and data, each of which have different Quality of Service (QoS) requirements for successful transmission. This paper presents an advanced reservation packet access protocol BRTDMA (Block Reservation Time Division Multiple Access) that can accommodate voice and data traffic with equal efficiency in a wireless network. The proposed BRTDMA protocol has been designed to operate in a dynamic fashion by allocating resources according to the QoS criteria of voice and data traffic. Most of the existing reservation protocols offers reservation to voice traffic while data packets are transmitted using contention mode. In this paper we propose a block reservation technique to reserve transmission slots for data traffic for a short duration, which minimizes the speech packet loss and reduce the end-to-end delay for wireless data traffic. The optimum block reservation length for data traffic has been studied in a cellular mobile radio environment using a simulation model. Simulation results show that the BRTDMA protocol offers higher traffic capacity than standard PRMA protocol for integrated voice and data traffic and offers flexibility in accommodating multimedia traffic.     

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.     

An integrated CDMA code assignment model for voice and data traffic in wireless communications

Wireless personal communication requires a provision of integrated services of multimedia traffic, such as voice and data, over the radio link. The multiple access protocols of code-division multiple-access (CDMA) techniques have been widely investigated in the recent literature. This paper presents an innovative multiple access protocol for CDMA-based wireless communication systems by fully utilizing the characteristics of voice and data traffic. In other words, a voice terminal can reserve a spreading code to transmit packets in multiple talk spurts, while a data terminal can transmit packets by either using the unassigned codes or borrowing the codes from the voice terminals during their silent periods. We build mathematical models for voice and data subsystems, respectively. Two performance parameters, the average dropping probability for voice packets and the average transmission delay for data packets, are derived based on the equilibrium point analysis. The effects of the two performance parameters on the system performance are discussed by varying the code reservation intervals of the voice terminals.     

Power control and capacity analysis for a packetized indoormultimedia DS-CDMA network

This paper proposes a packetized indoor wireless system using direct-sequence code-division multiple-access (DS-CDMA) protocol. The indoor radio environment is characterized by slow Rayleigh fading with or without lognormal shadowing. The system supports multimedia services with various transmission rates and quality of service (QoS) requirements and allows for seamless interfacing to asynchronous transfer mode (ATM) broadband networks. All packets are transmitted with forward error correction (FEC) using convolutional code for voice packets and Bose-Chaudhuri-Hocquenghem (BCH) code for data packets with an automatic retransmission request (ARQ) protocol and for video packets without ARQ. A queueing model is used for servicing data transmission requests. A power control algorithm is proposed for the system, which combines closed-loop power control with channel estimation to give the best performance. The cell capacity of each traffic type and various multimedia traffic configurations in both single-cell and multiple-cell networks are evaluated theoretically under the assumption of perfect power control. The effect of power control imperfection on the capacity using the proposed power control algorithm is investigated by computer simulation     

QoS evaluation of sender-based loss-recovery techniques for VoIP

Voice over Internet protocol (VoIP) is a technology that transports voice data packets across packet-switched networks using the Internet protocol (IP). Losing packets in the network is inevitable, and losing voice packets degrades audio quality. There are many loss-recovery techniques that designers can use to mitigate the undesired effects of packet loss. Some of these loss-recovery techniques use sender-based procedures, and others use receiver-based procedures. We examine several well-known sender-based loss-recovery techniques and evaluate the feasibility and effectiveness of each one in real-time interactive VoIP applications. We analyze the bandwidth requirements, buffering delays, and perceptual sound qualities of these techniques. We study the effectiveness of these approaches under various packet-loss conditions, and we also compare the effectiveness of these techniques against a speech codec that has high degree of packet-loss robustness