基于自适应变速率编码的VoIP网关的研究与应用

针对当前VoIP语音质量易受网络带宽影响的问题，提出了一种基于自适应变速率编码的VoIP网关。通过实时监测RTP语音分组的丢包率，并分析服务质量，自适应地选择最适合当前网络状况的编码速率，以提供一种语音质量和网络状况的最佳组合，从而降低语音分组的丢包率、有效地保障语音服务质量。结合开源项目Asterisk，实现了基于自适应变速率编码的VoIP网关。     

VoIP应用的原理与实现

VoIP技术是利用IP网络来传递语音信号的一种技术,近年来得到了广泛的应用和发展.分析VoIP技术的原理和应用模式,并基于Java语言给出了PC-PC模式下VoIP应用程序的设计与实现.     

一种基于SoPC技术的iLBC语音编解码器设计实现

基于VoIP技术的语音通信发展迅速,单芯片VoIP处理器的设计方法成为当前的研究热点.iLBC作为专为窄带通信而设计的VoIP语音编解码器,可以在丢包率和延迟较高的网络环境中保持良好的语音通话质量,具有广泛的应用前景.传统的基于DSP处理器实现方法具有芯片面积大、功耗高等缺点,难以满足VoIP系统集成度高、低功耗和易于升级等需求.本文提出了一种基于SoPC技术的iLBC语音编解码器实现方案,并对自相关计算算法进行了并行计算硬件IP核设计,提高了系统的集成度、计算性能和可扩展性.理论分析和实验结果表明并行自相关计算结构有效减少了访存次数,可以获得接近30的加速比.     

VoIP中为提高语音质量所采用的关键技术

VoIP(Voice over IP)即IP电话,是将话音编码、压缩转换成数据包,在IP网络中进行传输的技术。为应对传统电话公司的竞争,IP电话的语音质量成为决定其未来命运的关键因素。该文首先介绍了几个界定QoS的参数和目前评价IP电话业务语音质量的三种模型：MOS模型、PSQM模型、E模型,然后重点介绍了在终端和网络上提高VoIP语音质量所采用的一些关键技术,应用于终端的技术中比较重要的是语音的编码与压缩、差错控制等,而应用于网络的技术则是解决IP QoS的两种基本模型：综合业务模型和区分业务模型。     

IEEE 802.16系统中自适应的功率节省策略

针对IEEE 802.16系统中基于自适应多速率(AMR)语音编码器的IP语音(VoIP)业务,本文提出了一个自适应的功率节省策略。该策略周期性检测双向会话的语音帧信息,以此来判断上下行业务是否均进入语音静默期,然后自适应地调整功率节省模式参数。从能量节省、丢包率、系统信令开销方面分析了所提策略的性能,并且做了仿真实验。从理论分析和仿真结果可以看出,新策略在保证一定丢包率的基础上,可以比传统策略减少13.4%以上的能量损耗。     

支持QoS的SIP代理服务器方案的设计与实现

从如何提高基于SIP协议的VoIP系统的语音质量出发，分析了该系统与MPLS技术结合处的难点，并提出了一种支持QoS的SIP代理服务器模型的解决方案。该方案的设计结构清晰、可扩展性强，解决了SIP代理服务器在IP网络中没有会话服务质量控制能力的问题，是一种好的融合VoIP及MPLS两大技术的应用模型。该方案目前已运用于基于嵌入式Linux平台的VoIP系统中，完全能够满足高质量语音的需求。     

基于网络性能的VoIP语音质量评价模型

在VoIP应用中,为了实现服务质量的监测和路径切换,通常需要测量路径的网络性能,并将网络性能映射到语音质量评价.本文提出一种基于网络性能的VoIP语音质量评价模型,该模型在E-Model的基础上进行了改进,只考虑网络性能的动态变化对语音质量的影响.新的模型考虑更少的影响因素,比E-Model更容易计算,因此更适用于VoIP系统的语音质量评价.通过实验比较了新的模型和简单的网络参数评价模型,结果显示该模型具有更好的语音质量描述能力.     

从VoIP到IP语音通信

IP语音通信使用VoIP(IP语音通话)技术,一度被看作是VoIP的同义词,但其应用领域将远远超越后者。因为它可以在IP网络上实现可靠的话音和数据协同,如果说VoIP发起了一场成本节约革命,IP语音通信则是切实提供丰富通信功能的解决方案。     

VoIP中丢包隐藏技术研究

由于在“尽力型通信”中不可避免的传输错误(如丢包和时延),VoIP的语音质量会潜在地降低。在许多端对端VoIP系统中,语音的服务质量(QoS)很大部分地取决于丢包率和接收端的丢包隐藏算法(PLC)。文中论述了丢包的原因,对当前普遍采用的几种丢包隐藏技术进行了初步分析并进行了比较。     

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。     

QoS-aware path switching for VoIP traffic using SCTP

Voice over Internet protocol (VoIP) has been a prevalent multimedia service nowadays. It allows us to transmit voice data over IP networks. However, quality of service (QoS) is a major challenge to VoIP services. It must provide similar quality to traditional public switched telephone network or cellular phone services. Therefore, QoS related protocols have become important for real-time applications. Multi-protocol label switch (MPLS) is one of the important techniques to improve the network performance from QoS point of view. It employs label swapping to speed up packet forwarding. However, when a large number of users utilize VoIP services, the network congestion issue still exists. It causes delay, jitter and packet loss that affect VoIP QoS. In this paper, we propose a QoS-aware path switching strategy by using stream control transmission protocol (SCTP) in MPLS network to improve the VoIP traffic. This was done by employing SCTP selective acknowledgment mechanism to report the transmission parameters of primary path and to determine the criteria to switch to backup path. Simulation results show significant improvement in VoIP QoS.     

Design and implementation of QoS-provisioning system for voice over IP

In this paper, we address issues in implementing voice over IP (VoIP) services in packet switching networks. VoIP has been identified as a critical real-time application in the network QoS research community and has been implemented in commercial products. To provide competent quality of service for VoIP systems comparable to traditional PSTN systems, a call admission control (CAC) mechanism has to be introduced to prevent packet loss and over-queuing. Several well-designed CAC mechanisms, such as the site-utilization-based CAC-and the link-utilization-based CAC mechanisms have been in place. However, the existing commercial VoIP systems have not been able to adequately apply and support these CAC mechanisms and, hence, have been unable to provide QoS guarantees to voice over IP networks. We have designed and implemented a QoS-provisioning system that can be seamlessly integrated with the existing VoIP systems to overcome their weakness in offering QoS guarantees. A practical implementation of our QoS-provisioning system has been realized.     

基于改进的SOM网络模型的VoIP QoS应用研究

VoIP的服务质量（QoS,Quality of Service）评估可以采用一系列可度量的参数来描述：业务可用性、吞吐量、延迟、抖动、分组丢失率等。现有的感知语音质量评价（PESQ）很难对不同环境下的网络结构进行实时和恰当的语音等级质量分类。为了能够综合考虑几种QoS相关因素,在给出改进的自组织映射神经网络模型（ESOMNN）的基础上,利用ESOM能够对高维输入数据有效分类的特点,提出了将端到端延迟、丢包率、抖动、语音编码以及测试系统标识作为ESOMNN的输入数据,在对采样数据进行训练后可自动完成语音质量评价和映射,并能根据得到的实时变量有效地评价包含多种相关因素的QoS级别。     

无线Mesh网络中VOIP的QoS性能分析

针对VoIP(Voice over IP)业务在无线Mesh网上进行传输时存在服务质量(QoS)需求难以保证、带宽利用率低的问题,介绍了VoIP的QoS影响因素,分析了端到端时延、时延抖动和丢包率等几个重要参数,并对VoIP在无线Mesh网中的传输性能进行了论述。提出了基于无线Mesh网络的QoS保证机制,可以为端到端的数据传输公平的分配带宽,并能在保证QoS下实现大规模的实时任务的多跳转发。仿真试验表明能有效降低端到端时延,有着更好的QoS性能。     

QoS management in heterogeneous home networks

An OSGi (Open Services Gateway Initiative) home gateway system manages the integration of heterogeneous home networks protocols and devices to develop ubiquitous applications. Wired and wireless heterogeneous home networks have different QoS concerns. For instance, jitter and latency are important concerns in web phones, while packet loss ratio is important in on-line video. This study adopts UPnP QoS specification version 1.0 to design an adaptive QoS management mechanism based on the RMD (Resource Management in DiffServ) architecture. This study monitors real-time network traffic, and adaptively controls the bandwidth, to satisfy the minimum but most important quality for each application in home network congestion. Simulation results indicate that the average jitter, latency and packet loss are reduced by 0.1391 ms, 0.0066 s, and 5.43%, respectively. The packet loss ratio is reduced by 4.53%, and the throughput is increased by 1.2% in high definition video stream; the packet loss ratio is reduced by 1.89% for standard definition video stream, and in VoIP (Voice over IP) the jitter and latency are reduced to 0.0407 ms and 0.0209 s, respectively.     

Analysis of Skype VoIP traffic in UMTS: End-to-end QoS and QoE measurements

In the future Internet, multi-network services will follow a new paradigm in which the intelligence of the network control is gradually moved to the edge of the network. This impacts both the objective Quality of Service (QoS) of the end-to-end connection as well as the subjective Quality of Experience (QoE) as perceived by the end user. Skype already offers such a multi-network Voice-over-IP (VoIP) telephony service today. Due to its ease of use and a high sound quality, it becomes increasingly popular in the wired Internet.UMTS operators promise to offer large data rates which should suffice to support VoIP calls in a mobile environment. However, the success of those applications strongly depends on the corresponding QoE. In this work, we analyze the theoretically achievable as well as the actually achieved quality of IP-based voice calls using Skype. This is done performing measurements in both a real UMTS network and a testbed environment. The latter is used to emulate rate control mechanisms and changing system conditions of UMTS networks. The results show in how far Skype over UMTS is able to keep pace with existing mobile telephony systems and how it reacts to different network characteristics. The investigated performance measures comprise the QoE in terms of the MOS value and the QoS in terms of network-based factors like throughput, packet interarrival times, or packet loss.     

A multi-path mechanism for reliable VoIP transmission over wireless networks

With the advances in audio encoding standards and wireless access networks, voice over IP (VoIP) is becoming quite popular. However, real-time voice data over lossy networks (such as WLAN and UMTS), still posses several challenging problems because of the adverse effects caused by complex network dynamics. One approach to provide QoS for VoIP applications over the wireless networks is to use multiple paths to deliver VoIP data destined for a particular receiver. This paper introduced cmpSCTP, a transport layer solution for concurrent multi-path transfer that modifies the standard stream control transmission protocol (SCTP). The cmpSCTP aims at exploiting SCTP’s multi-homing capability by selecting several best paths among multiple available network interfaces to improve data transfer rate to the same multi-homed device. Through the use of path monitoring and packet allotment techniques, cmpSCTP tries to transmit an amount of packets corresponding to the path’s ability. At the same time, cmpSCTP updates the transmission strategy based on the real-time information of all of paths. Using cmpSCTP’s flexible path management capability, we may switch the flow between multiple paths automatically to realize seamless path handover. The theoretical analysis evaluated the model of cmpSCTP and formulated optimal traffic fragmentation of VoIP data. Extensive simulations under different scenarios using OPNET verified that cmpSCTP can effectively enhance VoIP transmission efficiency and highlighted the superiority of cmpSCTP against the other SCTP’s extension implementations under performance indexes such as throughput, handover latency, packet delay, and packet loss.     

Evaluating perceived voice quality on packet networks using different random neural network architectures

Voice over Internet Protocol (VoIP) is one of the fastest growing technologies in the world. In VoIP speech signals are transmitted over the same network used for data communications. The internet is not a robust network and is subjected to delay, jitter, and packet loss. It is very important to measure and monitor the quality of service (QoS) the users experience in VoIP networks; this is not an easy task and usually requires subjective tests. In this paper we have analyzed three non-intrusive models to measure and monitor voice quality using Random Neural Networks (RNN). A RNN is an open queuing network with positive and negative signals. We have assessed the voice quality based on various parameters i.e. delay, jitter, packet loss, and codec. In our approach we have used the Mean Opinion Score (MOS) calculated using a Perceptual Evaluation of Speech Quality (PESQ) algorithm to generate data for training the RNN model. We have studied two feed-forward models and a recurrent architecture. We have found that the simple feed-forward architecture has produced the most accurate results compared to the other two architectures.     

An Overlay Architecture for High-Quality VoIP Streams

The cost savings and novel features associated with voice over IP (VoIP) are driving its adoption by service providers. Unfortunately, the Internet's best effort service model provides no quality of service guarantees. Because low latency and jitter are the key requirements for supporting high-quality interactive conversations, VoIP applications use UDP to transfer data, thereby subjecting themselves to quality degradations caused by packet loss and network failures. In this paper, we describe an architecture to improve the performance of such VoIP applications. Two protocols are used for localized packet loss recovery and rapid rerouting in the event of network failures. The protocols are deployed on the nodes of an application-level overlay network and require no changes to the underlying infrastructure. Experimental results indicate that the architecture and protocols can be combined to yield voice quality on par with the public switched telephone network