The recent ITU-T Recommendation P.862, known as the Perceptual
Evaluation of Speech Quality (PESQ) is an objective end-to-end
speech quality assessment method for telephone networks and speech
codecs through the measurement of received audio quality. To
ensure that certain network distortions will not affect the
estimated subjective measurement determined by PESQ, the algorithm
takes into account packet loss, short-term and long-term time
warping resulted from delay variation. However, PESQ does not work
well for time-scale audio modification or temporal clipping. We
investigated the factors that impact the perceived quality when
time-scale modification is involved. An objective measurement of
time-scale modification is proposed in this research, where the
cross-correlation values obtained from time-scale modification
synchronization are used to evaluate the quality of a time-scaled
audio sequence. This proposed objective measure has been verified
by a subjective test.
The third-generation (3G) wireless network is a convergence of
several types of telecommunication networks to support various
wireless data services. Wireless LAN also supports mobility via
mobile IP. As a result, the convergence and mobility have
potential vulnerability in security. In this paper, a
Denial-of-Service (DoS) attack which can waste wireless resource
by sending a large number of nuisance packets to the spoofed
destination address of IP packets is introduced. To effectively
prevent the attack, fast detection, reliability, and efficiency
with small overhead are suggested as requirements in a detection
system. We propose a detector using Hidden Markov Model (HMM) to
achieve these requirements and reduce the influences of the attack
as fast as possible. The generation of the HMM for the detector
are discussed and the operation of the detector are described.
Weighting factors and second order Markov models are employed to
improve the reliability of the detector. The proposed system is
compared with the existing sequential detection approach in terms
of the false alarm rate and optimum detection time interval to
evaluate the performance of the detectors. Our simulation results
using ns-2 simulator shows that the proposed HMM detector is
reliable and fast to detect the attack due to its dynamic
Traditional TCP performance degrades over lossy links, as the TCP sender assumes that packet loss is caused by congestion in the network path and thus reduces the sending rate by cutting the congestion window multiplicatively, and a mechanism to overcome this limitation is investigated in this research. Our scheme identifies the network path condition to differentiate whether congestion happens or not, and responds differently. The basic idea of separating congestion and non-congestion caused losses is to compare the estimated current available bandwidth and the average available bandwidth. To minimize the effect of temporary fluctuation of measurements, we estimate the available bandwidth with a higher weight on stable measurements and a lower weight on unstable fluctuations. In our scheme, packet loss due to congestion invokes the TCP Newreno procedure. In cases of random loss that is not related to congestion, the multiplicative decrease of the
sending rate is avoided to achieve higher throughput. In addition, each duplicate acknowledgement after a fast retransmission will increase the congestion window to fully recover its sending rate. Extensive simulation results show that our differentiation algorithm achieves high accuracy. Accordingly, the TCP connection over lossy link with the proposed scheme provides higher throughput than TCP Newreno.
An enhanced CSMA/CA (Carrier Sense Multiple Access with Collision Avoidance) protocol to be used in the Medium Access Control (MAC) layer of the IEEE 802.11 standard for wireless local area networks (wireless LANs) is proposed in this work. In wireless LANs, the CSMA/CA protocol supports asynchronous data transfer, and adopts an
acknowledgement mechanism to confirm successful transmissions and a handshaking mechanism to reduce collisions. In both cases, a binary exponential backoff mechanism is used. The enhanced protocol improves the exponential backoff scheme by dynamically adjusting the contention window (CW) around the optimal value. Moreover, an analytical model based on the Markov chain is developed to analyze the system performance in terms of throughput and delay. Numerical
results are presented to show the effect of the proposed backoff mechanism.
In traditional packet voice or the emerging 2.5G and 3G wireless data services, smooth and timely delivery of audio is an essential requirement in Quality of Service (QoS) provision. It has been shown in our previous work that, by adapting time-scale modification to audio signals, an adaptive play-out algorithm can be designed to minimize packet dropping at the receiver end. By stretching the audio frame duration up and down, the proposed algorithm could adapt quickly to accommodate fluctuating delays including delay spikes.
In this paper, we will address the packet audio QoS with emphasis on end-to-end delay, packet loss, and delay jitter. The characteristics of delay and loss will be discussed. Adaptive playback will enhance the audio quality by adapting to the transmission delay jitter and delay spike. Coupled with Forward Error Correction (FEC) schemes, the proposed delay and loss concealment algorithm achieves less overall application loss rate without sacrificing on the average end-to-end delay. The optimal solution of such algorithms will be discussed. We also investigate the stretching-ratio transition effect on perceived audio quality by measuring the objective Perceptual Evaluation of Speech Quality (PESQ) Mean Opinion Score (MOS).
Proc. SPIE. 4740, Digital Wireless Communications IV
KEYWORDS: Internet, Transmitters, Data modeling, Multiplexing, Data transmission, Telecommunications, Forward error correction, Data communications, Global system for mobile communications, General packet radio service
Throughput estimation of users in a wireless packet data communication system utilizing the General Packet radio Service (GPRS) is investigated in this work. GPRS allows dynamic allocation of the bandwidth to a mobile terminal according to its traffic demand, which results in better resource utilization and a lower communication cost. The proposed user throughput estimation scheme relies on packet data traffic modeling. Since future wireless data services are mainly in web browsing, we consider a traffic model for today's WWW traffic that consists of ON/OFF two states with the ON period containing a sequence of document transmissions. A realistic traffic model is derived based on actual traffic data measurements. The throughput and transmission delay for fixed coding schemes can be obtained accordingly. This research effort contributes to the capacity planning of GPRS for its data services.
Packet delay and loss are two essential problems to the transmission of real-time voice over the best-effort Internet. Much effort has been involved in packet-level error control and delay jitter concealment. In our previous work, a time-scale modification of audio signal with per-packet adaptive playout algorithm is proposed to minimize the packet dropping at the receiver-end due to delay jitter. This work further extends the applicability of packet-based Synchronized OverLap-and-Add (SOLA) algorithm to an integrated delay/error concealment. Considering the timing relationship with the Forward Error Correction (FEC) parity-packet for loss recovery, the proposed adaptive playout algorithm estimates and adapts to packet loss as well as delay jitter. To enhance the playout quality, per-packet time-stretching factor is bounded by the content classification module, which classifies the audio signal into different categories. We also investigate the impact of stretching-ratio transition strategy to the perceived quality. To demonstrate the proposed adaptive playout, both analysis and performance evaluation is provided by comparing it to the silenced-based adaptive playout.
For Internet audio applications, much effort has been involved in
packet-level error control and delay jitter concealment. In this
paper, a packet-based time-scale modification scheme for speech
signal is applied to provide adaptive delay concealment at the
receiver of an Internet voice session. The adaptive playout
algorithm strives to minimize receiver packet droppings for
late-arrival packets and premature packets while keeping the
end-to-end delay constrained. By stretching the voice segment
up/down and incorporating the silence interval, the proposed
algorithm could adapt quickly to accommodate fluctuating delays
including delay spikes. The evaluation verifies the performance of
the proposed adaptive playout, which improves the received speech
intelligence under a tightly bounded average playout delay.
The recently developed MPEG-4 technology enables the coding and transmission of natural and synthetic audio-visual data in the form of objects. In an effort to extend the object-based functionality of MPEG-4 to real-time Internet applications, architectural prototypes of multiplex layer and transport layer tailored for transmission of MPEG-4 data over IP are under debate among Internet Engineering Task Force (IETF), and MPEG-4 systems Ad Hoc group. In this paper, we present an architecture for interactive MPEG-4 speech/audio transmission system over the Internet. It utilities a framework of Real Time Streaming Protocol (RTSP) over Real-time Transport Protocol (RTP) to provide controlled, on-demand delivery of real time speech/audio data. Based on a client-server model, a couple of low bit-rate bit streams (real-time speech/audio, pre- encoded speech/audio) are multiplexed and transmitted via a single RTP channel to the receiver. The MPEG-4 Scene Description (SD) and Object Descriptor (OD) bit streams are securely sent through the RTSP control channel. Upon receiving, an initial MPEG-4 audio- visual scene is constructed after de-multiplexing, decoding of bit streams, and scene composition. A receiver is allowed to manipulate the initial audio-visual scene presentation locally, or interactively arrange scene changes by sending requests to the server. A server may also choose to update the client with new streams and list of contents for user selection.