IP 电话网关中接口带宽的计算与分析

分析语音数据在IP电话网关中的处理过程，计算出RTP／UDP／IP协议的封装开锁，建立了在有静音压缩条件下的网关中多路编码器输出比特率的数学表达式，并利用该表达式计算了网关中各处理器之间在各种处理话路下的最小带宽，同时分析了接口带宽利用率与网关的处理密度N的关系。     

基于DSP的回波抵消器的软件实现

随着通讯、数字信号处理和大规模集成电路技术的飞速发展,人们对语音通信质量的要求越来越高。回波抵消技术能有效的解决长距离电话网络、IP电话、免提电话和视频会议等通信系统中的回波问题,很好地改善了语音通信质量,具有广阔的市场前景。主要研究用于声音通讯系统中的声回波抵消技术。针对具体应用以及DSP芯片的特点,结合BEPNLMS算法,给出了一种基于TM S320VC 5402 DSP实现回波抵消的软件设计方案。     

IP电话语音质量的测试

IP电话具有很大的发展潜力，但存在语音质量的问题。尤其是延时、丢包、抖动等广域网环境因素是导致音质下降的主要原因。文章采用VxD机制，开发了IP网络模拟器，模拟网络环境因素对IP电话设备进行语言质量测试。     

提高IP电话音质的相关技术及其参数的计算

ＩＰ电话作为一种新型的电信业务,具有很大的发展潜力,但也存在一个致命弱点,即语音质量问题,本文在分析影响ＩＰ电话音质因素的基础上,提出了如何通过改进ＩＰ电话网关结构来改善ＩＰ电话音质的有关技术,如：缓冲技术等。并借鉴多媒体同步与表现的有关理论,建立了ＩＰ电话网关中缓冲区设置数目和预取语音包数目的计算模型,文章最后给出了一个实例,验证该计算模型的正确性。     

Vo IP 的语音质量分析与控制

分析了VoIP语音质量的影响因素，通过E模型定量地描述了语音质量与端到端延迟和丢包率的关系。为了控制VoIP的语音质量，计算出VolP系统在各种情况下的语音质量极限，提出一种自适应编码和分组封装的控制策略。将该方法应用于自行开发的IP电话网关，实际测试证明能在很大程度上提高VoIP的语音质量。     

LTE系统中适应链路层机制的TCP头压缩算法

LTE系统采用全IP承载,具有比以往无线系统更高数据速率和容量.为了能够有效利用无线带宽资源,LTE系统中采用了ROHC(robust header compression)信头压缩协议对无线分组进行压缩.针对ROHC协议算法复杂度过高实现困难以及对高速数据业务的适应性较差等特点,提出一种适应LTE链路层机制的TCP头压缩算法.仿真结果表明,相比ROHC协议,本文提出的算法具有更高的头部压缩效率,进一步提高无线带宽资源的利用效率,同时,也能更好的适应LTE系统的高带宽环境,具有较低的算法复杂度和良好的稳健性和适用性.     

Adaptive explicit congestion control based on bandwidth estimation for high bandwidth-delay product networks

Existing congestion control protocols for high bandwidth-delay product (HBDP) networks either suffer from efficiency degradation caused by insufficient feedback of the network state, or are difficult to deploy in real networks because they require modification in IP header to provide enough feedback information. Based on VCP, this paper proposes the VCP-BE protocol that can provide more accurate network state information by estimating available bandwidth for sender with an adaptive bandwidth estimator. The estimator can dynamically adjust its parameters (e.g., observing interval) to achieve different accuracy and responsiveness to adapt to the network state indicated by two ECN bits, without no more modification in IP header than VCP. Furthermore, when the network load is high, VCP-BE calculates the ratio of current throughput to the available bandwidth to judge if network is close to the overload state. With these fine-grained network state estimation, VCP-BE can adjust the congestion window more precisely than VCP, thus greatly improves its convergence speed of achieving high bandwidth utilization and fairness. Simulation results show that VCP-BE also outperforms MLCP, which uses seven bits for explicit feedback while VCP-BE only uses two ECN bits.     

Low-complexity feature-mapped speech bandwidth extension

Today's telecommunications systems use a limited audio signal bandwidth. A typical bandwidth is 0.3-3.4 kHz, but recently it has been suggested that mobile phone networks will facilitate an audio signal bandwidth of 50 Hz-7 kHz. This is suggested since an increased bandwidth will increase the sound quality of the speech signals. Since only few telephones initially will have this facility, a method extending the conventional narrow frequency-band speech signal into a wide-band speech signal utilizing the receiving telephone only is suggested. This will give the impression of a wide-band speech signal. The proposed speech bandwidth extension method is based on models of speech acoustics and fundamentals of human hearing. The extension maps each speech feature separately. Care has been taken to deal with implementation aspects, such as noisy speech signals, speech signal delays, computational complexity, and processing memory usage.     

多解空间N元非合作性流量切割调度算法研究

给出了包交换网络基于多约束限制的多级并行调度和流量均衡体系模型，根据单节点路由调度任务的非合作性在极小碎片线性映射的假定条件下推导出了路由空间切割调度算法和量化公式，在节点上实现了IP包头重组的并行化计算。实验结果表明，通过量增大、利用率提升，网络在整个路由空间里有良好的时间响应特性。     

Admission control by implicit signaling in support of voice over IP over ADSL

Voice over DSL (VoDSL) is a technology that enables the transport of data and multiple voice calls over a single copper-pair. Voice over ATM (VoATM) and Voice over IP (VoIP) are the two main alternatives for carrying voice over DSL. ATM is currently the preferred technology, since it offers the advantage of ATM’s built-in Quality of Service (QoS) mechanisms. IP QoS mechanisms have been maturing only in recent years. However, if VoIP can achieve comparable performance to that of VoATM in the access networks, it would facilitate end-to-end IP telephony and could result in major cost savings. In this paper, we propose a VoIP-based VoDSL architecture that provides QoS guarantees comparable to those offered by ATM in the DSL access network. Our QoS architecture supports Premium and Regular service categories for voice traffic and the Best-Effort service category for data traffic. The Weighted Fair Queuing algorithm is used to schedule voice and data packets for transmission over the bottleneck link. Fragmentation of large data packets reduces the waiting time for voice packets in the link. We also propose a new admission control mechanism called Admission Control by Implicit Signaling. This mechanism takes advantage of application layer signaling by mapping it to the IP header. We evaluate the performance of our QoS architecture by means of a simulation study. Our results show that our VoIP architecture can provide QoS comparable to that provided by the VoATM architecture.     

基于FPGA实现的10Gbps高速转发引擎设计分析

高速特发引擎的设计是T比特路由器设计中的关键和难点,本文围绕传输带宽需求、查表时间需求和包头处理时间需求以及器件水平等方面对基于FPGA实现的10Gbps高速转发引擎进行了详细的分析,讨论了高速转发引擎各功能模块的设计可行性,并给出了一种可行的实现方案。     

Header Detection to Improve Multimedia Quality Over Wireless Networks

Wireless multimedia studies have revealed that forward error correction (FEC) on corrupted packets yields better bandwidth utilization and lower delay than retransmissions. To facilitate FEC-based recovery, corrupted packets should not be dropped so that maximum number of packets is relayed to a wireless receiver's FEC decoder. Previous studies proposed to mitigate wireless packet drops by a partial checksum that ignored payload errors. Such schemes require modifications to both transmitters and receivers, and incur packet-losses due to header errors. In this paper, we introduce a receiver-based scheme which uses the history of active multimedia sessions to detect transmitted values of corrupted packet headers, thereby improving wireless multimedia throughput. Header detection is posed as the decision-theoretic problem of multihypothesis detection of known parameters in noise. Performance of the proposed scheme is evaluated using trace-driven video simulations on an 802.11b local area network. We show that header detection with application layer FEC provides significant throughput and video quality improvements over the conventional UDP/IP/802.11 protocol stack     

IP电话系统中的消回声处理

分析了IP电话系统中回声的特点,并针对其回声具有路径延尺、延尺抖动大等特点,提出了一种基于延尺分析的自适应回声消除器设计的新思想和相关算法。介绍了自适应回声消除器的基本结构及各个功能模块,例如：变阶自适应滤波器,语音状态检测器,延尺分析器和NLMS算法控制器。本文还详细论述了以TMS320C6201DSP为硬件平台实现能运用在IP电话网关中的自适应回声消除器及其在IP电话系统中的连接关系。在文章的最后,列举了自适应回声消除器的性能测试结果,并分析了自适应回声消除器的优点,证明了自适应回声消除器能很好地解决IP电话系统中的回声消除问题。     

新型宽带语音编解码器AMR-WB的研究

该文介绍了一种应用于第三代移动通讯系统的编解码器,同时也是第一个可同时用于无线和有线应用的编解码器,该编解码器的语音带宽拓展为50Hz到7000Hz,编码后语音的自然度很高,用在3G移动通讯系统的多媒体服务、宽带包交换网络、音频和视频会议等等。由于AMR-WB为一个全新的宽带编解码器,其标准在2001年3月刚刚通过,国外对其的研究也属于起步阶段,还没有真正进入实用阶段。尤其在国内,至今尚未见到相关的研究。故对其进行全面的分析和深入的研究是必要的,会对今后的研究打下良好的基础。     

基于SIP协议的IP电话技术

现在的IP电话系统大部分还是基于H.323协议族,而H.323沿袭了传统电信网的运行和管理模式,本身较复杂,在应用于IP电话时存在着不可避免的缺陷。会议初始化协议(SIP)是由IETF制订的用于在两个或多个用户之间建立、修改和终止多媒体会话的信令协议。文中主要介绍了基于SIP的IP电话系统协议栈结构、SIP协议的基本内容以及其在IP电话中的具体应用。最后在与H.323协议进行了比较之后,提出了一些现存的问题并且做出了相应的前景展望。     

IP电话终端系统的研究与设计

通过对IP电话终端系统功能的分析,并对作者所开发的IP电话终端系统Extraphone进行剖析,阐述了IP电话的关键技术和解决方法。提出静音抑制处理技术、话音编码方式和抖动缓冲处理综合考虑的保证话音质量的传送算法模型。     

一种核心无状态保存的自适应成比例公平带宽分配机制

提出了一种核心无状态的自适应比例公平带宽分配机制CSPAFA（core stateless proporitonal adaptive fair allocation)，在边界路由器完成基于每个流的状态处理，将所有流分成标记流和非标记流两种业务类型，采用DPS（dynamic packet state)技术将有关信息编码进IP分组头，在核心将输出链路带宽分成两部分，核心根据当前的网络负荷对标记流按服务规格成比例的分配输出链路带宽，对未标记流公平分配带宽，并且能自适应地调整两类业务的带宽共享比例，最后，给出了在NS网络仿真环境下的仿真实验结果。     

无线IP网络中一种针对实时流的报头压缩算法

无线IP网中,为了能够有效利用带宽资源,使提供实时业务具有经济上的可行性和现实性,必须采用报头压缩技术减小协议报头带来的额外开销.然而,现有的报头压缩方案在设计时没有考虑无线信道状态.为了能够更好地适应无线链路特性,提出并分析了一种无线IP网络中针对实时流的基于信道状态的健壮报头压缩算法.该算法通过对无线信道状态的精确估计来调节报头压缩器中W-LSB编码的可变滑动窗口的大小,在压缩率和抗差错健壮性之间能够实现好的平衡,并适应特性经常变化的无线链路.最后,通过仿真结果证明该算法在无线链路上的有效性.     

IP电话网关中处理器平均利用率的计算

提出了一种集成式IP电话网关的体系结构，分析了IP电话网关的数据处理过程，并采用Petri网和多机系统理论，建立了集成式IP电话网关的随机Petri网模型，推导出处理器的平均利用率的解析式，分析了处理器利用率与系统总线的带宽，存储器的结构、类型等参数的关系。     

网守与许可控制和计费

该文从网守的角度,针对许可控制和计费管理实现中所涉及的三个主要问题:协议的使用、网守系统结构和消息流程,进行了认真的分析,提出了解决方案,论证了方案的可行性和先进性,并在实际产品中通过测试。