The Extensive Growth Of Networks

Published Date: 02 Nov 2017

Chapter 1

Introduction

Internet has become an integral part of our everyday life with its global evolution and the extensive growth of networks. The human speech is carried in analog wave signal [1]. To make voice telephone calls, analog circuits were used. Voice over IP technology is a prospective replacement and a development of the traditional telephony systems over the Public Switched Telephone Network (PSTN). Internet Telephony is the transmission of voice signals in digital form.

In the early 1960’s, the world's first packet switching network invented was the ARPANET [2]. Later it was integrated into the TCP/IP network protocol used on the Internet today. In the 1990s, with the Internet and Web boom, there has been an increasing demand for transmission of digital information through IP-based packets by telephone companies. In recent years, Voice over IP has developed greatly and more people are using it every day with applications such as Skype [3].

Traditional telephones use circuit networks and they can only communicate when a dedicated channel has been established between them [4]. Resources are reserved for the whole duration of the communication session. Therefore there are wastages of the network resources during silent period. Whereas Voice over IP uses packet switching and the packets have many possible paths to take. Transmitting voice calls in this manner is very cost effective since the available bandwidth is used very efficiently.

However many studies made on Voice over IP show that there are problems which occur. V.Hardman et al., 1995[5] made a study on ‘Reliable Audio for use over the Internet’ and have pointed out the problem of packet loss and the degradation of the speech intelligibility due to that. Shveni Mehta, 2005[6] also found that along with packet loss, excessive delay and high delay jitter also have an impact on the quality of communication. Due to these vulnerabilities, many researches have been conducted on these issues. The studies made by C.Perkins, O.Hodson and V.Hardman, 1998[7], Thomas J.Kostas et al., 1998 [8] and Ajay Kumar, 2006 [9] also show the adverse effect of packet loss on the speech quality.

To operate the network efficiently, there are many organizations which are working on the Quality of Service (QoS) which is very important. But the current IP networks offer only best-effort service, that is the network provides no guarantee that the data has been delivered and they are designed to support non-real time applications like FTP, Telnet and email. TCP (Transmission Control Protocols) offers reliable delivery of data by using the Synchronous (SYN) and Acknowledge (ACK) control flags which are additional overhead attached to the data and lead to excessive delay in transmission. But such delay cannot be tolerated in real-time services like video conferencing.

1.1 Motivation

Voice over IP, also known as IP Telephony, is a technology that transmits real-time voice signals using the Internet Protocol (IP) over a private data network or the public Internet. VoIP brings together different services of an organization into a single communication network.

The basic processes involved are:

Conversion of analog signals into digital form

Breaking of the digital signal into packets (compression/translation)

Transmission of the separate packets over an IP network

Reassembling of the packets at the receiving end.

Due to the problems faced by Voice over IP, users are reluctant to use the technology. Users must have at least the same quality of the transmitted speech signal acquired by the PSTN network so that IP network is adopted widespread. The QoS level of VoIP applications depends on many factors such as jitter, delay and packet loss which greatly affect the voice quality [10]. We are confronted with an important tradeoff of designing every parameter to an optimum value to meet the quality objectives while making efficient use of the network resources.

1.2 Problem statement

VoIP applications require real-time packet delivery and the quality of service is not satisfactory for the current internet system. Real-time applications have different features compared to text-based non-real time applications.

An important characteristic of most real-time applications is that the data do not need to be completely identical as the original one. Some errors and a level of loss can be afforded without affecting the performance considerably. However as the packet loss rate increases, the speech signal gets degraded. Reliability is an important issue to be considered. Reliability is the reliable delivery of packets to its destination. When packets are transmitted, they are sometimes delayed and are considered lost if they do

not reach their destination within a certain time. Therefore it is important to understand the loss behaviour and the delays.

The purpose of this project is to design loss concealment techniques which can improve the quality of the speech. Receiver-based and Sender-based techniques are used and the methods are compared to know which one gives better performance.

1.3 Outline of project

In this project, the insertion-based techniques, silence substitution and packet repetition and two FEC schemes (Reed-Solomon code and Convolutional code) are evaluated for replacement of a packet loss.

The outline of the report is as follows:

Chapter Two gives an outline of some of the studies performed on Voice over IP. An overview of the different packet recovery technique which is divided into receiver-based and sender-based and the forward error correction techniques are given. It explains some theoretical framework of digital voice communication, the Gilbert loss model and Erasure codes.

Chapter three explains the methods used to carry out the project and flowcharts are used for the explanation.

Chapter four contains all the simulation results from the implementations and an analysis of the results.

Chapter five gives an overall conclusion of the results obtained and the further works that can be done.

[1] You Don't Know Jack About VoIP, THE COMMUNICATIONS THEY ARE A-CHANGIN'.

PHIL SHERBURNE AND CARY FITZGERALD, CISCO, September 1, 2004 at:

http://queue.acm.org/detail.cfm?id=1028895

[2] A Brief History of VoIP, Document One, Joe Hallock – The Past, November 26, 2004

Evolution and Trends in Digital Media Technologies – COM 538 Masters of Communication in Digital Media University of Washington at:

http://www.joehallock.com/edu/pdfs/Hallock_J_VoIP_Past.pdf

[3] JISC Voice over IP: what it is, why people want it, and where it is going, Jane Dudman, Contributing author: Gaynor Backhouse, September 2006 at:

http://www.ciscovb.com/up/uploads/files/domain-357d053b63.pdf

[4] QuickStudy: Packet-Switched vs. Circuit-Switched Networks, By Lee Copeland, March 20, 2000 12:00 PM at: http://www.computerworld.com/s/article/41904/Packet_Switched_vs._Circuit_Switched_Networks

[5] Vicky Hardman, Martina Angela Sasse, Mark Handley, Anna Watson,, ‘’Reliable audio for use over the Internet,’’ Proc.INET ’95, 1995.

[6] Comparative Study of Techniques to minimize packet loss in VoIP, Shveni P Mehta, 21st Computer Science Seminar SB3-T2-1, 2005 at:

http://www.ewp.rpi.edu/hartford/~rhb/cs_seminar_2005/SessionB3/mehta.pdf

[7] C. Perkins, O. Hodson, and V. Hardman, "A survey of packet loss recovery techniques for streaming audio," IEEE Network, 1998, pp. 40-48.

[8] Real-Time Voice Over Packet-Switched Networks

Thomas J.Kostas, Michael S.Borella, Ikhlaq Sidhu, Guido M.Schuster, Jacek Grabiec and Jerry Mahler 3COM, IEEE Network, January/February 1998

[9] AN OVERVIEW OF VOICE OVER INTERNET PROTOCOL (VOIP)

Ajay Kumar*

Graduate student, M.S. in Computer Science Program, Rivier College

RIVIER COLLEGE ONLINE ACADEMIC JOURNAL, VOLUME 2, NUMBER 1, SPRING 2006

[10] An Analytic and Experimental Study on the Impact of Jitter Playout Buffer on the

E-model in VoIP Quality Measurement

Olusegun Obafemi, Tibor Gyires ,Yongning Tang

ICN 2011 : The Tenth International Conference on Networks