Resource-efficient delivery of on-demand streaming data using UEP codes

We propose and analyze a new multicast scheme for delivering on-demand streaming data using unequal protection codes. The scheme allows an end user to join only one multicast channel for a data stream at any time to play out the requested data stream from its beginning after a fixed initial playout delay. The scheme tolerates packet loss during transmission, and thus, significantly reduces the cost of implementing a reliable multicast network layer to ensure delivery of all packets. Meanwhile, resource usage of the scheme, including server computing bandwidth, network bandwidth, and client's buffer space, is determined only by the original data stream length and the initial playout delay, but is independent of either the number or the arrival pattern of individual end-user requests. Thus, the scheme is totally scalable with the number of end users, fully utilizing the data delivery efficiency of a multicast network. The scheme also uses resources efficiently, e.g., with an initial playout delays of 30 s and 60 s, multicasting a 2 h video using this scheme needs only about 5.5 and 4.8 times, respectively, the server computing bandwidth and network bandwidth of those for a single unicast delivery of the same original data stream.     

突发大时延下的IP语音缓冲算法

当网络中存在突发大时延时,就会出现极大丢包率或极大端到端时延的情况,从而难以获得好的语音质量。对于这个问题,提出针对突发大时延存在下的自适应语音缓冲算法。算法通过控制语音包在语音缓冲队列中的位置来控制语音包的播放时间,从而可以尽量减小语音裂缝(Gap)的出现。算法将突发大时延存在下的最大丢包率可以扩大到20％,而一般的预测算法只能容忍5-10％的最大丢包率。通过基于听觉模型的客观音质评价(PESQ)仿真计算,以及实际应用表明本文的算法对有突发大时延存在的网络中的语音通信质量有一定的改善作用。     

Efficient De-Jitter Control for Voice Applications over Wireless Ad Hoc Networks

Voice over IP applications require a playout buffer at the receiver side to smooth network delay variations. Unfortunately, existing algorithms for dynamic playout adjustment designed for wireline networks do not operate correctly in wireless ad hoc networks. These algorithms estimate the end-to-end delay on the set of previous received audio packets. Indeed, such a delay estimation based on past history is not appropriate due to mobility which leads to random changes of the network topology. In this paper, we highlight this delay estimation problem. We show that route request AODV control messages provide more accurate delay estimation. Then, we propose a new algorithm for playout delay adjustment based on these control messages. The performance evaluation is performed by simulation using ns-2. We show that this algorithm outperforms existing playout delay adjustment algorithms. Performance criteria are loss late percentage (reliability criterion), averaged playout delay (interactivity criterion) and playout delay variation (stability criterion).     

A perceptual quality model intended for adaptive VoIP applications

Quality models predict the perceptual quality of services as they calculate subjective ratings from measured parameters. In this article, we present a new quality model that evaluates Voice over IP (VoIP) telephone calls. In addition to packet loss rate, coding mode and delay, it takes into account the impairments due to changes in the transmission configuration (e.g. switching the coding mode or re‐scheduling the playout time). Moreover, this model can be used at run time to control the transmission of such calls. It is also computationally efficient and open source. To demonstrate the potential of our model, we apply it to select the ideal coding and packet rate in bandwidth‐limited environments. Furthermore, we decide, based on model predictions, whether to delay the playout of speech frames after delay spikes. Delay spikes often occur after congestion and cause packets to arrive too late. We show a considerable improvement in perceptual speech quality if our model is applied to control VoIP transmissions. Copyright © 2005 John Wiley & Sons, Ltd.     

Error recovery for interactive video transmission over the Internet

Real-time interactive video transmission in the current Internet has mediocre quality because of high packet loss rates. Loss of packets in a video frame manifests itself not only in the reduced quality of that frame but also in the propagation of that distortion to successive frames. This error propagation problem is inherent in any motion compensation-based video codec. In this paper, we present a new error recovery scheme, called recovery from error spread using continuous updates (RESCU), that effectively alleviates error propagation in the transmission of interactive video. The main benefit of the RESCU scheme is that it allows more time for transport-level recovery such as retransmission and forward error correction to succeed while effectively masking out delays in recovering lost packets without introducing any playout delays, thus making it suitable for interactive video communication. Through simulation and real Internet experiments, we study the effectiveness and limitations of our proposed techniques and compare their performance to that of existing video error recovery techniques including H.263+ (NEWPRED). The study indicates that RESCU is effective in alleviating the error spread problem and can sustain much better video quality with less bit overhead than existing video error recovery techniques under various network environments     

Packet scheduling with playout adaptation for scalable video delivery over wireless networks

In this paper, we propose a playout deadline-aware packet scheduling for scalable video delivery over wireless networks. We develop a novel playout adaptation algorithm to reduce playback interruptions by jointly considering the active playout buffer status and adaptive playout rate. We also propose a packet priority analysis method based on the layer information of Scalable Video Coding (SVC). Based on the priority of the video packet and the adaptive playout-deadline, an optimal packet scheduling algorithm is proposed. Packets are selected for transmission to minimize the quality degradation caused as well as to reduce the playout latency. We also adopt a benchmark for the packet priority analysis by calculating the distortion impact of each packet with the consideration of the packet dependency in SVC. When compared with the state-of-the-art algorithms as well as the benchmark, our proposed scheduling algorithm shows a good trade-off between the video quality and the playout latency.     

Impact of playout buffer dynamics on the QoE of wireless adaptive HTTP progressive video

The quality of experience (QoE) of video streaming is degraded by playback interruptions, which can be mitigated by the playout buffers of end users. To analyze the impact of playout buffer dynamics on the QoE of wireless adaptive hypertext transfer protocol (HTTP) progressive video, we model the playout buffer as a G/D/1 queue with an arbitrary packet arrival rate and deterministic service time. Because all video packets within a block must be available in the playout buffer before that block is decoded, playback interruption can occur even when the playout buffer is non-empty. We analyze the queue length evolution of the playout buffer using diffusion approximation. Closed-form expressions for user-perceived video quality are derived in terms of the buffering delay, playback duration, and interruption probability for an infinite buffer size, the packet loss probability and re-buffering probability for a finite buffer size. Simulation results verify our theoretical analysis and reveal that the impact of playout buffer dynamics on QoE is content dependent, which can contribute to the design of QoE-driven wireless adaptive HTTP progressive video management.     

改善Internet视频流QoS的途径和方法

在当前Internet的尽力而为的服务模式下，网络拥塞和分组丢失不可避免，视频流必须使用有效的拥塞控制和差错控制来改善性能。本文分析了：Internet视频流QoS影响因素，提出了两种QoS解决方案：基于终端和基于网络。本文着重讨论了基于终端的QoS解决方案，在目前Internet的环境下，基于终端的QoS解决方案更具可行性。     

Improving voice over IP subjective call quality

Adaptive playout algorithms rely on estimates of network delays to calculate playout times of voice packets. Typically, network estimators are either able to react quickly to delay variations or to ignore transient noise conditions, but cannot do both. In our solution, the weighting factor that controls the estimation process is dynamically adjusted according to the observed delay variations. This results in higher quality estimates of network delays. Experimental results show that our algorithm can achieve higher subjective call quality than the basic adaptive algorithm, as measured by the ITU-T E-Model methodology.     

Calculation of packet jitter for non-poisson traffic

The packet delay variation, commonly called delay jitter, is an important quality of service parameter in IP networks especially for real-time applications. In this paper, we propose the exact and approximate models to compute the jitter for some non-Poisson FCFS queues with a single flow that are important for recent IP network. We show that the approximate models are sufficiently accurate for design purposes. We also show that these models can be computed sufficiently fast to be usable within some iterative procedure, e.g., for dimensioning a playback buffer or for flow assignment in a network.     

A New Buffer Algorithm for Speech Quality Improvement in VoIP Systems

Jitter buffer plays an important role in Voice over IP (VoIP) applications because it provides a key mechanism for achieving good speech quality to meet technical and commercial requirements. The main objective of this paper is to propose a new, simple-to-use jitter buffer algorithm as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance, in terms of enhanced user-perceived speech quality and reduced end-to-end delay. Supported by signal processing features, the new algorithm, the so-called Play Late Algorithm, alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. The results show that the new algorithm achieves the best performance under different network conditions when compared to conventional static and adaptive jitter buffer algorithms. The results reported here are based on live tests and emulated network conditions on real mobile phone prototypes. The mobile phone prototypes use AMR codec and support full IP/UDP/RTP stack with IPSec function in some of the tests. The method for perceived speech quality measurement is based on the ITU-T standard for speech quality evaluation (PESQ).

Assessing the quality of voice communications over Internet backbones

As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to meet the quality standards achieved in the public switched telephone network. Our objective in this paper is to assess to what extent today's Internet meets this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks and uses subjective voice quality measures capturing the various impairments incurred. First, we compile the results of various studies into a single model for assessing the voice-over-IP (VoIP) quality. Then, we identify different types of typical Internet paths and study their VoIP performance. For each type of path, we identify those characteristics that affect the VoIP perceived quality. Such characteristics include the network loss and the delay variability that should be appropriately handled by the playout scheduling at the receiver. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of Internet backbone paths lead to poor performance.     

A novel adaptive traffic prediction AQM algorithm

In the Internet, network congestion is becoming an intractable problem. Congestion results in longer delay, drastic jitter and excessive packet losses. As a result, quality of service (QoS) of networks deteriorates, and then the quality of experience (QoE) perceived by end users will not be satisfied. As a powerful supplement of transport layer (i.e. TCP) congestion control, active queue management (AQM) compensates the deficiency of TCP in congestion control. In this paper, a novel adaptive traffic prediction AQM (ATPAQM) algorithm is proposed. ATPAQM operates in two granularities. In coarse granularity, on one hand, it adopts an improved Kalman filtering model to predict traffic; on the other hand, it calculates average packet loss ratio (PLR) every prediction interval. In fine granularity, upon receiving a packet, it regulates packet dropping probability according to the calculated average PLR. Simulation results show that ATPAQM algorithm outperforms other algorithms in queue stability, packet loss ratio and link utilization.     

The integrated services in the Internet: state of the art

This paper is about the evolution of the Internet from a simple data network into a true multiservice network that can support the emerging multimedia applications and their protocols with appropriate performance and costs. The real-time delivery and specific bandwidth requirements of these multimedia applications have created a need for an integrated services Internet in which traditional best effort datagram delivery can coexist with additional enhanced quality of service delivery classes. The integrated services Internet will be able to commit to meet bandwidth, packet loss, and delay specifications for individual data flows by using the resource reservation protocol together with appropriate packet forward scheduling policies     

A novel congestion control protocol with AQM support for IP-based networks

Multimedia services (Real-time and Non real-time) have different demands, including the need for high bandwidth and low delay, jitter and loss. TCP is a dominant protocol on the Internet. In order to have the best performance in TCP, the congestion window size must be set according to some parameters, since the TCP source is not aware of the window size. TCP emphasizes more on reliability than timeliness, so TCP is not suitable for real-time traffic. In this paper an active Queue management support TCP (QTCP) model is presented. Source rate is regulated based on the feedback which is received from intermediate routers. Furthermore, in order to satisfy the requirements of multimedia applications, a new Optimization Based active Queue management (OBQ) mechanism has been developed. OBQ calculates packet loss probabilities based on the queue length, packets priority and delay in routers and the results are sent to source, which can then regulate its sending rate. Simulation results indicate that the QTCP reduces packet loss and buffer size in intermediate nodes, improves network throughput and reduces delay.     

Quality of service management for IPTV services support in VANETs: a performance evaluation study

In order to deliver a qualitative Internet Protocol Television (IPTV) service over vehicular ad hoc networks (VANETs), a quality of service (QoS) mechanism is needed to manage the allocate of network resources to the diverse IPTV application traffic demands. Unlike other mobile network, VANETs have certain unique characteristic that presents several difficulties in providing an effective QoS. Similarly, IPTV requires a constant stream for QoS which at the moment is quite difficult due to the inherent VANET characteristics. To provide an effective QoS that will meet the IPTV application service demands, VANETs, must satisfy the compelling real-time traffic streaming QoS requirement (i.e., minimum bandwidth allocation, packet loss and jitter). In this report, we evaluate via simulation the feasibility of deploying quality IPTV services over VANETs, by characterizing the association between the IPTV streaming quality determining factors (i.e., throughput, delay, loss, jitter) and the IPTV quality degradation, with respect to node density and node velocity. Furthermore, we used an objective QoS metric (Media-Delivery-Index) to identify, locate and address the loss or out-of-order packet. We outline how, using these information’s can support in shaping network parameters to optimize service flows. The implementation assures a priority for handling IPTV traffic, such that maximise the usage of VANETs resources, and opens the possibility that loss and delay can be minimised to a degree that could guarantee quality IPTV service delivery among vehicle in a vehicular network system.

     

Uncoordinated real-time video transmission in wireless multicode CDMA systems: An SMPT-based approach

We consider the real-time transmission of encoded video from distributed, uncoordinated wireless terminals to a central base station in a multicode CDMA system. Our approach is to employ the recently proposed simultaneous MAC packet transmission (SMPT) approach at the data link layer (in conjunction with UDP at the transport layer). We consider the real-time transmission of both video encoded in an open loop (i.e., without rate control) and video encoded in a closed loop (i.e., with rate control). We conduct extensive simulations and study quantitatively the trade-off between video quality, transmission delay (and jitter), and number of supported video streams (capacity). We find that the simple-to-deploy SMPT approach achieves significantly higher video quality and smaller delays than the conventional sequential transmission approach, while ensuring high capacity. In typical scenarios, with SMPT the probability of in-time video frame delivery is more than twice as large as with sequential transmission (for given delay bounds). Our results provide guidelines for the design and dimensioning of cellular wireless systems as well as ad hoc wireless systems.     

Improving the R-score of an adaptive VoIP codec in IEEE 802.16 networks

Worldwide Interoperability for Microwave Access (WiMAX) technology, which is based on the IEEE 802.16 standard, supports different quality of service (QoS) for different services. WiMAX is expected to support QoS in real-time applications such as Voice over Internet Protocol (VoIP). When network congestion occurs, the VoIP bit rate needs to be adjusted to achieve the best speech quality. In this study, we propose a new scheme called Adaptive VoIP Level Coding (AVLC). This scheme takes into consideration network conditions (packet delay and packet loss) and a connection’s modulation scheme. The amount of data that can be transmitted increases with the speed of the modulation scheme. When network congestion occurs, AVLC scheme prioritizes reducing the bit rate of a connection that has a slower modulation scheme to mitigate congestion. Depending on network conditions, such as modulation scheme, packet delay, packet loss, and residual time slot, we use the G.722.2 codec to adjust each connection’s bit rate. Simulations are conducted to test the performance (network delay, packet loss, number of modulation symbols, and R-score) of the proposed scheme. The simulation results indicate that speech quality is improved by the use of AVLC.     

Stochastic performance analysis of non-feedforward networks

Many Internet applications are both delay and loss sensitive, and need network performance guarantees that include bandwidth, delay/delay jitter, and packet loss rate. It is very important to quantify and exploit the capabilities of guaranteed service provisioning of communication networks. In this paper, we study the queueing behaviors of non-feedforward networks (a non-feedforward network is a network in which at least one set of acyclic traffic routes forms a cycle; a feedforward network is a network in which any set of acyclic traffic routes does not form a cycle) with FIFO scheduling discipline and Regulated, Markov On-Off, and Fractional Brownian traffic sources. We develop a new methodology to analyze the probabilistic bounds on the delays experienced by traffic. By leveraging the large deviations and fixed-point techniques, we turn probability problems into deterministic optimization problems and translate a probabilistic delay bound into a fixed point of a non-linear real function. Our contribution in this paper is the derivation of a probabilistic bound on the delays experienced by traffic in non-feedforward networks, based on an assumption, i.e., the tail probability of the difference between the beginning time of a busy interval of a server and the earliest arriving time at the corresponding network ingress of the traffic arrivals that arrive at this server during this busy interval can be bounded by the maximum of the violation probabilities of the accumulative upper stream delay bound suffered by this server‘s traffic arrivals. Consequently, our new results not only consummate the theory of stochastic analysis of network performance, but also facilitate the design of protocols and algorithms for non-feedforward networks to provide performance guarantees to various applications with diverse performance requirements.