Adaptive playout scheduling and loss concealment for voice communication over IP networks

The quality of service limitation of today's Internet is a major challenge for real-time voice communications. Excessive delay, packet loss, and high delay jitter all impair the communication quality. A new receiver-based playout scheduling scheme is proposed to improve the tradeoff between buffering delay and late loss for real-time voice communication over IP networks. In this scheme the network delay is estimated from past statistics and the playout time of the voice packets is adaptively adjusted. In contrast to previous work, the adjustment is not only performed between talkspurts, but also within talkspurts in a highly dynamic way. Proper reconstruction of continuous playout speech is achieved by scaling individual voice packets using a time-scale modification technique based on the Waveform Similarity Overlap-Add (WSOLA) algorithm. Results of subjective listening tests show that this operation does not impair audio quality, since the adaptation process requires infrequent scaling of the voice packets and low playout jitter is perceptually tolerable. The same time-scale modification technique is also used to conceal packet loss at very low delay, i.e., one packet time. Simulation results based on Internet measurements show that the tradeoff between buffering delay and late loss can be improved significantly. The overall audio quality is investigated based on subjective listening tests, showing typical gains of 1 on a 5-point scale of the Mean Opinion Score.     

实时语音的自适应回放算法

由于丢包和延时抖动的引入而使网络传输的实时语音质量让人难以接受，目前对丢包和延时抖动提出了很多的解决方案．但是却很少把这两者结合在一起进行研究。本文提出了一种新的自适应回放算法，通过监测接收和回放队列，结合丢包的自适应恢复技术，达到语音高质量的连续回放。实验证明，该算法能在严格的平均回放延时条件下努力减小由于超时而引起的丢包，获得较好的重建语音质量。     

Playout buffering of speech packets based on a quality maximization approach

To combat jitter problems in voice streaming over packet networks, playout buffering algorithms are used at the receiver side. Most of the proposed solutions rely on two main operations: prediction of delay statistics for future packets; setting of the end-to-end delay so as to limit or avoid packet losses. In recent years, a new approach has been presented, which is based on using a quality model to evaluate the impact of both packet loss and delay on the voice quality. Such a model is used to find the buffer setting that maximizes the expected quality. In this paper, we present a playout buffering algorithm whose main contribution is the extension of the new quality-based approach to the case of voice communications affected by bursty packet losses. This work is motivated by two main considerations: most of IP telephony applications are characterized by bursty losses instead of random ones; the human perception of the speech quality is significantly affected by the temporal correlation of losses. To this purpose, we make use of the extensions proposed in the ETSI Tiphon for the ITU-T E-Model so as to incorporate the effects of loss burstiness on the perceived quality. The resulting playout algorithm estimates the characteristics of the loss process varying the end-to-end delay, weights the loss and the delay effects on the perceived quality, and maximizes the overall quality to find the optimal setting for the playout buffer. The experimental results prove the effectiveness of the proposed technique.     

Providing MAC QoS for multimedia traffic in 802.11e based multi-hop ad hoc wireless networks

Ad hoc wireless networks with their widespread deployment, now need to support applications that generate multimedia and real-time traffic. Video, audio, real-time voice over IP, and other multimedia applications require the network to provide guarantees on the Quality of Service (QoS) of the connection. The 802.11e Medium Access Control (MAC) protocol was proposed with the aim of providing QoS support at the MAC layer. The 802.11e performs well in wireless LANs due to the presence of Access Points (APs), but in ad hoc networks, especially multi-hop ones, it is still incapable of supporting multimedia traffic.One of the most important QoS parameters for multimedia and real-time traffic is delay. Our primary goal is to reduce the end-to-end delay, thereby improving the Packet Delivery Ratio of multimedia traffic, that is, the proportion of packets that reach the destination within the deadline, in 802.11e based multi-hop ad hoc wireless networks.Our contribution is threefold: first we propose dynamic ReAllocative Priority (ReAP) scheme, wherein the priorities of packets in the MAC queues are not fixed, but keep changing dynamically. We use the laxity and the hop length information to decide the priority of the packet. ReAP improves the PDR by over 28% in comparison with 802.11e, especially under heavy loads. Second, we introduce Adaptive-TXOP (A-TXOP), where transmission opportunity (TXOP) is the time interval during which a node has the right to initiate transmissions. This scheme reduces the delay of video traffic by reducing the number of channel accesses required to transmit large video frames. It involves modifying the TXOP interval dynamically based on the packets in the queue, so that fragments of the same packet are sent in the same TXOP interval. A-TXOP is implemented over ReAP to further improve the performance of video traffic. ReAP with A-TXOP helps in reducing the delay of video traffic by over 27% and further improves the quality of video in comparison with ReAP without A-TXOP. Finally, we have TXOP-sharing, which is aimed at reducing the delay of voice traffic. It involves using the TXOP to transmit to multiple receivers, in order to utilize the TXOP interval fully. It reduces the number of contentions to the channel and thereby reduces the delay of voice traffic by over 14%. A-TXOP is implemented over ReAP to further improve the performance of voice traffic. The three schemes (ReAP, A-TXOP, and TXOP-sharing) work together to improve the performance of multimedia traffic in 802.11e based multi-hop ad hoc wireless networks.     

Optimal,quality-aware scheduling of data consumption in mobile ad hoc networks

In this paper we study the delivery of quality contextual information in mobile ad-hoc networks. We consider that information has a certain quality level that fades over time. Mobile context-aware applications receive and process disseminated information given that the corresponding quality is above the lowest level. The necessity for optimally scheduling information delivery arises from the dynamic nature of the network, e.g., probabilistic spreading, caching, deferred delivery, and mobility of nodes. We propose two policies for optimal scheduling information delivery consumption based on the Optimal Stopping Theory. The mobile nodes delay the reporting of information to mobile context-aware applications in search for better quality. The proposed policies efficiently deal with the delivery of quality information in mobile ad-hoc networks.     

率失真优化的无线多跳网络多路径选择算法

随着无线网络技术的发展,基于无线多跳网的视频通信在智能交通、灾难应急和军事指挥等多个领域得到越来越广泛的应用.但是,如何保证无线视频的传输质量,是亟待解决的一个关键问题.已有多路径视频传输研究忽略了信道变化和路径间干扰.针对该问题,提出一种基于率失真预测的多路径选择优化算法.该算法不仅分析了网络拥塞对传输质量的影响,而...     

Telephony in the year 2005

The growth of packet-based voice services is leading to integration of voice and other services over packet switched data networks. This paper explores a possible path that the telephone service industry may follow as this integration is accelerated by technological advances that improve the capabilities of packet-based services while reducing their costs.     

SRPM: Secure Routing Protocol for IEEE 802.11 Infrastructure Based Wireless Mesh Networks

Infrastructure based IEEE 802.11 wireless mesh networks (WMNs) are new paradigm of low cost broadband technology. The large scale city-wide community-based coverage and multi-hop architecture are such characteristics which are vulnerable to network layer threats, and the adversary can exploit them for large scale degradation of the broadband services. So far many secure routing protocols have been proposed for ad-hoc networks, however, due to the different nature and characteristics; they cannot perform well in a WMN environment. In this paper, we discuss the limitations and challenges as well as propose an exclusive secure routing protocol for an infrastructure based wireless mesh (SRPM) network. SRPM is robust against a variety of multi-hop threats and performs well over a range of scenarios we tested.     

Voice quality prediction models and their application in VoIP networks

The primary aim of this paper is to present new models for objective, nonintrusive, prediction of voice quality for IP networks and to illustrate their application to voice quality monitoring and playout buffer control in VoIP networks. The contributions of the paper are threefold. First, we present a new methodology for developing perceptually accurate models for nonintrusive prediction of voice quality which avoids time-consuming subjective tests. The methodology is generic and as such it has wide applicability in multimedia applications. Second, based on the new methodology, we present efficient regression models for predicting conversational voice quality nonintrusively for four modern codecs (G.729, G.723.1, AMR and iLBC). Third, we illustrate the usefulness of the models in two main applications - voice quality prediction for real Internet VoIP traces and perceived quality-driven playout buffer optimization. For voice quality prediction, the results show that the models have accuracy close to the combined ITU PESQ/E-model method using real Internet traces (correlation coefficient over 0.98). For playout buffer optimization, the proposed buffer algorithm provides an optimum voice quality when compared to five other buffer algorithms for all the traces considered.     

Design and realization of ad-hoc VoIP with embedded p-SIP server

In this paper we design and implement the pseudo session initiation protocol (p-SIP) server embedded in each mobile node to provide the ad-hoc voice over Internet protocol (VoIP) services. The implemented p-SIP server, being compatible with common VoIP user agents, integrates the standard SIP protocol with SIP presence to handle SIP signaling and discovery mechanism in the ad-hoc VoIP networks. The ad-hoc VoIP signaling and voice traffic performances are analyzed using E-model R rating value for up to six hops in the implemented test-bed. We also conduct the interference experiments to imitate the practical ad-hoc VoIP environment. The analyzed results demonstrate the realization of ad-hoc VoIP services by using p-SIP server.     

Neural and fuzzy computation techniques for playout delay adaptation in VoIP networks

Playout delay adaptation algorithms are often used in real time voice communication over packet-switched networks to counteract the effects of network jitter at the receiver. Whilst the conventional algorithms developed for silence-suppressed speech transmission focused on preserving the relative temporal structure of speech frames/packets within a talkspurt (intertalkspurt adaptation), more recently developed algorithms strive to achieve better quality by allowing for playout delay adaptation within a talkspurt (intratalkspurt adaptation). The adaptation algorithms, both intertalkspurt and intratalkspurt based, rely on short term estimations of the characteristics of network delay that would be experienced by up-coming voice packets. The use of novel neural networks and fuzzy systems as estimators of network delay characteristics are presented in this paper. Their performance is analyzed in comparison with a number of traditional techniques for both inter and intratalkspurt adaptation paradigms. The design of a novel fuzzy trend analyzer system (FTAS) for network delay trend analysis and its usage in intratalkspurt playout delay adaptation are presented in greater detail. The performance of the proposed mechanism is analyzed based on measured Internet delays.     

Location awareness in ad hoc wireless mobile networks

Networks composed of dynamically repositioning mobile hosts require location awareness to provide new geographic services and to maximize routing efficiency and quality of service. Because wireless networks can operate in a 3D physical environment, exploiting mobile hosts' location information is both natural and inevitable. Emerging geographic services based on mobile ad-hoc networks (manets) must confront several challenges, including how to increase positioning accuracy and how to establish a connection from location information to the vast body of Web data, as in a tour-guide system for example     

Wireless MediaNets: application-driven next-generation wireless IP networks

Thanks to the explosive creation of multimedia contents, the pervasive adoption of multimedia coding standards and the ubiquitous access of multimedia services, multimedia networking is everywhere in our daily lives. Unfortunately, the existing best effort IP network infrastructure, originally designed with little real-time QoS requirement, has started to suffer from performance degradation on emerging multimedia networking applications. This inadequacy problem is further deepened by the prevalence of last/first-mile wireless networking, such as Wi-Fi, mobile WiMAX, and many wireless sensors and ad-hoc networks. This can be evidenced by more and more fragmentation of application-driven IP-based networks, such as for power grid distribution, networked security surveillance, intelligent transportation communication, and many other sensor networks. To overcome the QoS challenges, the next generation wireless IP networks have to be architected in a top-down manner, i.e., application-driven layered protocol design. More specifically, based on the application media data, compression schemes are applied, the subsequent Network, MAC- and PHY-layered protocols need to be accordingly or jointly enhanced to reach the optimal performance. This is the fundamental concept behind the design of Wireless MediaNets. In this survey paper, I will address the QoS challenges specifically encountered in video over heterogeneous wireless broadband networks and address several application-driven Wireless MediaNet solutions based on effective cross-layer integration of APP and MAC/PHY layers. More specifically, the congestion control for achieving airtime fairness of video streaming to maximize the link adaptation performance of Wi-Fi, the minimum latency event-driven data exchange for distributed wireless ad-hoc sensor networks, and the opportunistic multicast of scalable video live streaming over mobile WiMAX.     

无线传感器网络跨层自适应唤醒算法研究

时钟同步与能耗有效性是无线传感器网络多跳传输关键问题.提出一种自适应唤醒算法,在MAC层和网络层进行跨层优化,节点按需自适应唤醒,实现多跳网络的协同传输,能够有效降低系统能耗.该算法不需要节点间的周期性同步,减少了频繁的包交换带来的数据冲突与能量浪费,从而提高了网络可靠性和能耗效率,提高了基于IEEE802.15.4标准的多跳网络的节能效率.为了验证算法有效性,工作分别在NS-2仿真环境和实际应用场景下进行了仿真与测试验证,结果表明在传输可靠性和节能效率上均由较大提高.跨层自适应唤醒算法可进一步推广到大规模异构自组织网络中.     

P2P IP Telephony over wireless ad-hoc networks

This paper presents a new strategy to form P2P IP Telephony overlay for wireless ad-hoc networks. In the proposed strategy a structured P2P system is considered where some nodes, called super-nodes, with higher capacity form the overlay and provide registry and call routing services. As selection and admission of new super-nodes in wireless ad-hoc networks is more challenging than backbone networks, we define the strategies to select and admit new super-nodes into the overlay. On one hand, scarce resources and fluctuating link quality demand additional criteria than just node computing resources for super-nodes selection. On the other hand, the indiscriminate increase in super-node number can raise the call session setup delay and degrade the quality. This is due to the relaying of packets across multiple wireless links. In this paper, we first define the criteria to select super-nodes and then the major part of the paper is dedicated to defining the required strategies to admit new super-nodes. Our admission strategies add new super-nodes to the system whenever they are required. Since the strategy does not simply admit all eligible super-node candidates, this ensures control over the number of super-nodes and keeps the session setup delay within to the required service level threshold. We define a queuing network to model our system and evaluate the efficacy of our admission strategies with intensive simulations. Furthermore, we have implemented a P2P IP Telephony system that operates on wireless ad-hoc networks and validated the performance of our admission strategies on this real platform.     

WSN数据收集中移动sink的路径规划和蔟头节点选取问题的综合研究

针对较大规模的无线传感器网络通过多跳传输进行数据收集而引起的能量空洞问题,本文提出了一种基于移动sink的簇头节点数据收集算法（MSRDG）,该算法基于图论原理,在满足时延性的条件下,综合考虑了普通节点到簇头节点路由和移动sink遍历路经选取的问题,构建了一条通过的簇头节点尽可能多的移动轨迹。通过NS-2仿真软件对算法的性能进行评估,结果显示出该算法能减少数据的多跳传输,降低无线传感器网络节点的能量消耗,延长网络寿命。     

Real-time video streaming over multipath in multi-hop wireless networks

Given the limited wireless link throughput, high loss rate, and varying end-to-end delay, supporting video applications in multi-hop wireless networks becomes a challenging task. Path diversity exploits multiple routes for each session simultaneously, which achieves higher aggregated bandwidth and potentially decreases delay and packet loss. Unfortunately, for TCP-based video streaming, naive load splitting often results in inaccurate estimation of round trip time (RTT) and packet reordering. As a result, it can suffer from significant instability or even throughput reduction, which is also validated by our analysis and simulation in multi-hop wireless networks. To make real-time TCP-based streaming viable over multi-hop wireless networks, we propose a novel cross-layer design with a smart traffic split scheme, namely, multiple path retransmission (MPR). MPR differentiates the original data packets and the retransmitted packets and works with a novel QoS-aware multi-path routing protocol, QAOMDV, to distribute them separately. MPR does not suffer from the RTT underestimation and extra packet reordering, which ensures stable throughput improvement over single-path routing. Through extensive simulations, we further demonstrate that, as compared with state-of-the-art multi-path protocols, our MPR with QAOMDV noticeably enhances the TCP streaming throughput and reduces bandwidth fluctuation, with no obvious impact to fairness.     

Secure and robust multipath routings for advanced metering infrastructure

A Smart Grid is the modernization of the electricity grid using communication technology with the prime goals of reducing energy consumption as well as cost increasing reliability and creating new services for all participants. It comprises key components such as the Advanced Metering Infrastructure (AMI), which includes Neighborhood area network (NAN). When multi-hopping is considered in wireless communication, especially in WiFi and ZigBee, the range of the communication can be extended to communicate with the gateway collector in AMI network. Wireless mesh AMI network may have smart meters, a NAN gateway, and fixed as well as mobile repeaters. Though many techniques have been developed to secure on-demand routing protocols in wireless multi-hop ad-hoc networks, these protocols have shortcomings. In this paper, we propose two robust and secure multipath routing protocols for wireless mesh AMI networks. We have analyzed their robustness to various attacks. The simulation results show that the proposed protocols are better than existing secure routing protocols.     

A link availability-based QoS-aware routing protocol for mobile ad hoc sensor networks

With more and more wireless devices being mobile, there is a constant challenge to provide reliable and high quality communication services among these devices. In this paper, we propose a link availability-based QoS-aware (LABQ) routing protocol for mobile ad hoc networks based on mobility prediction and link quality measurement, in addition to energy consumption estimate. The goal is to provide highly reliable and better communication links with energy-efficiency. The proposed routing algorithm has been verified by NS-2 simulations. The results have shown that LABQ outperforms existing algorithms by significantly reducing link breakages and thereby reducing the overheads in reconnection and retransmission. It also reduces the average end-to-end delay for data transfer and enhances the lifetime of nodes by making energy-efficient routing decisions.     

Optimal delay estimation using complete bipartite graph method for VoIP over WLAN environments

Voice over wireless LAN (WLAN) is widely applied in modern Internet networks. In real-time wireless environments, the primary cause of an impaired speech quality is transmission delays. In wireless communications, the parameters employed for quality of service (QoS) estimations include the delay, the throughput, the blocking rate and the link cost. These parameters are significantly dependent on the type of topology pattern adopted. This study focuses primarily on the delay guarantee and optimal delay aspects of wireless communications. A tradeoff exists between the link number and the delay guarantee. A greater number of links is beneficial in reducing transmission delays, but leads to an increased cost. Conversely, a lower number of links reduces the system cost, but increases the delay. This study employs graph theory to determine the node topology pattern which minimizes the delay for voice over IP (VoIP) transmission in WLAN environments. The topology is optimized both for a variable link delay and a constant link delay. The optimization method presented in this study can be widely applied to the link topology design for multiparty conferencing in modern wireless networks.