19 research outputs found
Media gateway utilizando um GPU
Mestrado em Engenharia de Computadores e Telemátic
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Survey of unified approaches to integrated-service networks
The increasing demand for communication services, coupled with recent technological advances in communication media and switching techniques, has resulted in a proliferation of new and expanded services. Currently, networks are needed which can transmit voice, data, and video services in an application-independent fashion. Unified approaches employ a single switching technique across the entire network bandwidth, thus, allowing services to be switched in an application-independent manner. This paper presents a taxonomy of integrated-service networks including a look at N-ISDN, while focusing on unified approaches to integrated-service networks.The two most promising unified approaches are burst and fast packet switching. Burst switching is a circuit switching-based approach which allocates channel bandwidth to a connection only during the transmission of "bursts" of information. Fast packet switching is a packet switching-based approach which can be characterized by very high transmission rates on network links and simple, hardwired protocols which match the rapid channel speed of the network. Both approaches are being proposed as possible implementations for integrated-service networks. We survey these two approaches, and also examine the key performance issues found in fast packet switching. We then present the results of a simulation study of a fast packet switching network
Time and frequency domain algorithms for speech coding
The promise of digital hardware economies (due to recent advances in
VLSI technology), has focussed much attention on more complex and sophisticated
speech coding algorithms which offer improved quality at relatively
low bit rates.
This thesis describes the results (obtained from computer simulations)
of research into various efficient (time and frequency domain) speech
encoders operating at a transmission bit rate of 16 Kbps.
In the time domain, Adaptive Differential Pulse Code Modulation (ADPCM)
systems employing both forward and backward adaptive prediction were
examined. A number of algorithms were proposed and evaluated, including
several variants of the Stochastic Approximation Predictor (SAP). A
Backward Block Adaptive (BBA) predictor was also developed and found to
outperform the conventional stochastic methods, even though its complexity
in terms of signal processing requirements is lower. A simplified
Adaptive Predictive Coder (APC) employing a single tap pitch predictor
considered next provided a slight improvement in performance over ADPCM,
but with rather greater complexity.
The ultimate test of any speech coding system is the perceptual performance
of the received speech. Recent research has indicated that this
may be enhanced by suitable control of the noise spectrum according to
the theory of auditory masking. Various noise shaping ADPCM
configurations were examined, and it was demonstrated that a proposed
pre-/post-filtering arrangement which exploits advantageously the
predictor-quantizer interaction, leads to the best subjective
performance in both forward and backward prediction systems.
Adaptive quantization is instrumental to the performance of ADPCM systems.
Both the forward adaptive quantizer (AQF) and the backward oneword
memory adaptation (AQJ) were examined. In addition, a novel method
of decreasing quantization noise in ADPCM-AQJ coders, which involves the
application of correction to the decoded speech samples, provided
reduced output noise across the spectrum, with considerable high frequency
noise suppression.
More powerful (and inevitably more complex) frequency domain speech
coders such as the Adaptive Transform Coder (ATC) and the Sub-band Coder
(SBC) offer good quality speech at 16 Kbps. To reduce complexity and
coding delay, whilst retaining the advantage of sub-band coding, a novel
transform based split-band coder (TSBC) was developed and found to compare
closely in performance with the SBC.
To prevent the heavy side information requirement associated with a
large number of bands in split-band coding schemes from impairing coding
accuracy, without forgoing the efficiency provided by adaptive bit
allocation, a method employing AQJs to code the sub-band signals together
with vector quantization of the bit allocation patterns was also
proposed.
Finally, 'pipeline' methods of bit allocation and step size estimation
(using the Fast Fourier Transform (FFT) on the input signal) were examined.
Such methods, although less accurate, are nevertheless useful in
limiting coding delay associated with SRC schemes employing Quadrature
Mirror Filters (QMF)
A-Interface Over Internet Protocol For User-Plane Connection Optimization In GSM/EDGE Radio Access Network
This thesis will cover a detailed study about the main motivations and benefits from using IP as a transport protocol for specifically A-interface in GERAN for Circuit Switched User-Plane (CS-UP) connection, in addition to the required protocols.
The main study in this document will be around Real Time Protocol (RTP), Real Time Control Protocol (RTCP) negotiation for RTP packets multiplexing, for both cases, with and without RTP header compression. The focus will be about the communication between the Base Station Controller (BSC) and the Media GateWay (MGW), the bandwidth gain in accordance to the multiplexing delay for processing and buffering, the voice Quality of Service (QoS) and some other parameters
Future benefits and applications of intelligent on-board processing to VSAT services
The trends and roles of VSAT services in the year 2010 time frame are examined based on an overall network and service model for that period. An estimate of the VSAT traffic is then made and the service and general network requirements are identified. In order to accommodate these traffic needs, four satellite VSAT architectures based on the use of fixed or scanning multibeam antennas in conjunction with IF switching or onboard regeneration and baseband processing are suggested. The performance of each of these architectures is assessed and the key enabling technologies are identified
Simulation and performance of a statistical multiplexer in an ATM network
This report examines some of the issues arising m the implementation of statistical multiplexing in a broadband Integrated digital services network (B-ISDN) by analysis and simulation The BISDN concept is introduced and described. A review o f the current areas of research is given along with some of the important issues as they relate to telephone traffic. The report then
focuses on the problem o f multiplexing voice traffic.
A typical voice source is analysed and the traffic characteristics which result are described. The
concept of statistical multiplexing is mtroduced. A review of the current literature studies relating to the problems of analysing multiplexed sources is given, with particular reference to the concept of cell level and burst level queues being separate and disparate components requiring different analytical approaches. Several models are mtroduced including the 3-state model not previously described in the literature.
The queue behaviour resulting from a large number of superposed lmes is analysed as a simplified Markov process and the results are used to argue that it is not feasible to provide buffers for nodes which multiplex a large number of low intensity sources. The problem of scaling small models up to realistic situations is discussed.
An approach to simulating the problem is described along with algorithms for implementing the basic elements. A senes of results derived from the described simulation are presented and analysed. The report concludes that statistical multiplexing is feasible, but with certain limits as to the type of traffic which can be supported
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Speech coding
Speech is the predominant means of communication between human beings and since the invention of the telephone by Alexander Graham Bell in 1876, speech services have remained to be the core service in almost all telecommunication systems. Original analog methods of telephony had the disadvantage of speech signal getting corrupted by noise, cross-talk and distortion Long haul transmissions which use repeaters to compensate for the loss in signal strength on transmission links also increase the associated noise and distortion. On the other hand digital transmission is relatively immune to noise, cross-talk and distortion primarily because of the capability to faithfully regenerate digital signal at each repeater purely based on a binary decision. Hence end-to-end performance of the digital link essentially becomes independent of the length and operating frequency bands of the link Hence from a transmission point of view digital transmission has been the preferred approach due to its higher immunity to noise. The need to carry digital speech became extremely important from a service provision point of view as well. Modem requirements have introduced the need for robust, flexible and secure services that can carry a multitude of signal types (such as voice, data and video) without a fundamental change in infrastructure. Such a requirement could not have been easily met without the advent of digital transmission systems, thereby requiring speech to be coded digitally. The term Speech Coding is often referred to techniques that represent or code speech signals either directly as a waveform or as a set of parameters by analyzing the speech signal. In either case, the codes are transmitted to the distant end where speech is reconstructed or synthesized using the received set of codes. A more generic term that is applicable to these techniques that is often interchangeably used with speech coding is the term voice coding. This term is more generic in the sense that the coding techniques are equally applicable to any voice signal whether or not it carries any intelligible information, as the term speech implies. Other terms that are commonly used are speech compression and voice compression since the fundamental idea behind speech coding is to reduce (compress) the transmission rate (or equivalently the bandwidth) And/or reduce storage requirements In this document the terms speech and voice shall be used interchangeably
Quality of Service Controlled Multimedia Transport Protocol
PhDThis research looks at the design of an open transport protocol that supports a range of
services including multimedia over low data-rate networks. Low data-rate multimedia
applications require a system that provides quality of service (QoS) assurance and flexibility.
One promising field is the area of content-based coding. Content-based systems use an array
of protocols to select the optimum set of coding algorithms. A content-based transport
protocol integrates a content-based application to a transmission network.
General transport protocols form a bottleneck in low data-rate multimedia
communicationbsy limiting throughpuot r by not maintainingt iming requirementsT. his work
presents an original model of a transport protocol that eliminates the bottleneck by
introducing a flexible yet efficient algorithm that uses an open approach to flexibility and
holistic architectureto promoteQ oS.T he flexibility andt ransparenccyo mesi n the form of a
fixed syntaxt hat providesa seto f transportp rotocols emanticsT. he mediaQ oSi s maintained
by defining a generic descriptor. Overall, the structure of the protocol is based on a single
adaptablea lgorithm that supportsa pplication independencen, etwork independencea nd
quality of service.
The transportp rotocol was evaluatedth rougha set of assessmentos:f f-line; off-line
for a specific application; and on-line for a specific application. Application contexts used
MPEG-4 test material where the on-line assessmenuts eda modified MPEG-4 pl; yer. The
performanceo f the QoSc ontrolledt ransportp rotocoli s often bettert hano thers chemews hen
appropriateQ oS controlledm anagemenatl gorithmsa re selectedT. his is shownf irst for an
off-line assessmenwt here the performancei s compared between the QoS controlled
multiplexer,a n emulatedM PEG-4F lexMux multiplexers chemea, ndt he targetr equirements.
The performanceis also shownt o be better in a real environmentw hen the QoS controlled
multiplexeri s comparedw ith the real MPEG-4F lexMux scheme