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Study of Next Generation Networks (NGN) Using simulation modeling

Report submitted in partial fulfillment of the requirement for the degree of

B.Sc In

Electrical and Electronic Engineering

Supervisor: Dr. Iman Abuel Maaly By

Rifga Mokhier Altaher

Department of Electrical and Electronic Engineering University of Khartoum

June 2007

Dedication

I dedicate my project

With my all love

Endless thanks

Best wishes to

My parents for the great trust they have

put upon me.

My sisters and brother, who supported

and encouraged me.

My partner, Enas, for doing it with me

hand by hand, step by step

My supervisor, Dr. Iman Abuel Maaly who

guided me the way to do such a project

i

Acknowledgement

I would like to thank my supervisor,

Doctor Iman Abuel Maaly for her patience,

understanding and support. I would also

like to thank the engineers at Canar Telecom

Company and everyone who have

encouraged me to finally finish my thesis.

Finally, I’m deeply grateful to all the staff

of electrical and electronic engineering

department.

ii

TABLE OF CONTENS Dedication………………………………………………………………..i

Acknowledgement…………………………………………………….…ii

Table of contents………………………………………………………...iii

List of figures……………………………………………………………vi

List of tables…………………………………………………………….vii

Abstract………………………………………………………………….viii

Chapter 1

1. Introduction…………………………………………………………….1

1.1 introduction ………………………………………………………...2

1.2 Definition…………………………………………………………...2

1.3 problem statement…………………………………………………..4

1.4 Objectives…………………………………………………………..4

1.5 Methodology………………………………………………………..4

1.6 Thesis outline……………………………………………………....5

Chapter 2

2. NGN basic concepts…………………………………………………...6

2.1 Introduction………………………………………………………...7

2.2 Migration to NGN………………………………………………….7

2.2.1 Wait & See…………………………………………………….7

2.2.2 Internet offload………………………………………………...7

2.2.3 Voice over broadband………………………………………….8

2.2.4 Replace transit switches………………………………………..8

2.3 The Softswitch……………………………………………………...8

2.3.1 Definition……………………………………………………....8

2.3.2 The Softswitch concept………………………………………...9

2.4 NGN characteristics……………………………………………….10

2.5 conclusion…………………………………………………………11

Chapter 3

3. NGN architecture……………………………………………………...12

3.1 NGN layers………………………………………………………...13

3.2 Edge Access Layer………………………………………………...13

3.2.1 Integrated Access Device …………………………………….14

3.2.2 Media Gateways (MGs)………………………………………14

iii

3.2.2.1 Media Gateways Characteristics……….……………………14

3.2.2.2 Types of Media Gateways…………..………..……………..15

1. Access Media Gateway (AMG)…………………………...15

2. Trunk Media Gateway (TMG)……………………………..15

3. Signaling Media Gateway (SG)…………………………....15

4. Universal media gateway (UMG)………………………….16

3.3 Core Switching Layer……………………………………………....16

3.4 Network Control Layer…………………………………………......17

3.5 Service Management Layer……………………………………........18

3.5.1 Integrated Operation Support System (IOSS)……………...…..18

3.5.2 Policy server………………………………………………...….18

3.5.3 Application Server……………………………………………...18

3.5.4 Location Server…………………………………………………19

3.5.5 Media Resource Server…………………………………………19

3.5.6 Service Control Point…………………………………………...19

3.6 Conclusion…………………………………………………………..20

Chapter 4

4. VoIP Implementation in NGN………………………………………….21

4.1 Introduction……………………………………………………...…..22

4.2 Problem definition and decomposition………………………………22

4.2.1 The Scenarios…………………………………………………….22

4.2.1.1 PC to PC phase………………………………………….…..23

4.2.1.2 PC to PHONE phase………………………………………...23

4.2.1.3 PHONE to PHONE phase…………………………………..24

4.2.2 Signaling Architecture…………………………………………...24

4.3 Session Initiation protocol (SIP)……………………………………..25

4.3.1 Definition………………………………………………………..25

4.3.2 Comparison between SIP & H.323……………………………...25

4.3.3 SIP architecture………………………………………………….26

4.3.3.1 SIP Entities………………………………………………….26

4.3.3.2 SIP messages…………………………………………….…..27

4.3.4 Session Description Protocol (SDP)………………………….…..31

4.3.5 SIP Addressing ……………………………………………….…..31

4.3.6 Inter-working between SIP network and NGN …………………..32

4.4 Real Time Protocol (RTP)………………………………………….….33

iv

Chapter 5

5. Switching Design& Evaluation…………………………………………..34

5.1 Introduction……………………………………………………...…...35

5.2 Java Interface to SIP …………………………………………...…….35

5.3 Application architecture………………………………………...…….35

5.3.1 User Agent …………………………………………………...….35

5.3.1.1 User Agent fundamentals……………………………………35

5.3.1.2 User Agent demonstration……………………………...…...37

5.3.2 Proxy and Registrar server…………………………………...…..38

5.3.2.1 Proxy capabilities…………………………………………....38

5.3.2.2 How to start Proxy? ………………………………………....39

5.3.2.3 System Administrator………………………………………..39

5.3.3 SIP/PSTN Gateway ………………………………………………40

5.3.3.1 Preface ……………………………………………………..…40

5.3.3.2 Basic structure of PSTN Gateway…………………………....40

5.3.3.3 PSTN connection part to the Gateway …………………….....40

5.3.3.4 How PSTN Gateway works? ………………………………...41

5.3.3.5 Detailed explanation of Gateway functionality……………….42

5.4 Testing and Analysis results…………………………………………....43

5.4.1 Testing example……………………………………………………43

5.4.2 Evaluation of results ………………………………………………44

Chapter 6

6. Conclusion and Comments…………………………..……………………46

6.1 Conclusion……………………………………………………………..47

6.2 Recommendations……………………………………………………...48

References……………………………………………………………………49

Appendix ……………………………………………………………………..50

v

List of Figures 1.1 Evolution from PSTN to NGN…………………………………………3

2.1 Softswitch……………………………………………………………....9

2.2 Next Generation Network in a variety of network deployments………11

3.1 NGN Layer Architecture………………………………………………13

3.2 Edge Access layer devices…………………………………………….13

3.3 Network Control layer…………………………………………………17

3.4 Separation of call control from bearer…………………………………17

3.5 Service Management Layer……………………………………………18

3.6 NGN architecture………………………………………………………20

4.1 Decomposition of the problem………………………………………...23

4.2 SIP Entities…………………………………………………………….27

4.3 SIP message syntax ……………………………………………………28

4.4 Setting up a SIP session………………………………………………..31

4.5 Interworking between SIP network & NGN…………………………..32

5.1 User Agent architecture………………………………………………..36

5.2 The stack properties User Interface window…………………………..37

5.3 Authentication of a User Agent ……………………………………….37

5.4 User Agent calling another User Agent………………………………..38

5.5 Proxy server configuration…………………………………………….39

5.6 Gateway architecture…………………………………………………..41

5.7 PSTN Gateway………………………………………………………...42

vi

List of Tables 4.1 Comparison between SIP & H.323 …………….………………..26

4.2 SIP response codes……………………………………………….30

vii

viii

Abstract

This thesis aims to study the Next Generation Network (NGN), which refer to

the idea of one network that cannot only cost effectively, deliver and manage all the

voice, video, and data communications option available today, but one that can also

adopt and grow to handle any new communication options that will inevitably

evolve. Moreover, the main goal of this thesis is to examine the Next Generation

Network’s capability in providing a reliable voice communication service while

keeping the cost at minimum.

Next Generation Network (NGN) uses the IP network as its core network, i.e.

voice is transferred over the IP network (VoIP). Thus, the VoIP implementation in

NGN is included in this thesis. This implementation is done in stages. Firstly the

desired scenario must be specified either PC to PC or PC to PHONE or PHONE to

PHONE which is our concern. Then, an appropriate signaling protocol is chosen for

setting up the IP telephony session between the scenario’s parties, the Session

Initiation Protocol (SIP) is our preferred option over the NGN protocols. After the

signaling takes place, the parties are now ready to transfer media using the Real Time

Protocol (RTP). Hence, in order to achieve the thesis’s goal, the Session Initiation

Protocol (SIP) has been simulated. SIP entities and the software application

components used for simulation purposes are built and interconnected using Java

based software packages.

The results obtained from this simulation of SIP VoIP application which

works on platforms that support NGN protocol functionalities did not deliver perfect

level of quality standards, but it had offered voice with acceptable limits in addition

to the unlimited bandwidth due to the packet-based IP network . These results

would definitely serve both service provider and customer side's benefit.

CHAPTER 1

INTRODUCTION

Chapter 1- Introduction _________________________________________________________________________________________________________________________________________

1.1 Introduction In recent years, the telecom network has been developing rapidly, and the

integrated communication ability has been enhanced greatly. However, the network

faces more and more pressure from the gradual integration of telephone network,

computer network, and cable television network. Because the network load is

increasing and service demands become diversified, the carriers have to provide

more new services to attract users, which are hardly provided by the PSTN (Public

Switched Telephony Network) or PLMN (Public Land Mobile Network).

Meanwhile, the rapid-developing data networks are taking over some services from

the PSTN and PLMN and play an important role in bearing voice service. Whereas,

the Voice over IP (VoIP) based on the H.323 protocol can only meet the basic

requirement for packet voice service, and cannot provide abundant service functions.

[1]

1.2 Definition In the conditions mentioned above, the Next Generation Network (NGN),

which is based on the softswitch technology, comes into being.

NGN is a milestone in the telecom field. It indicates the arrival of the new

generation telecom network. In terms of the development, NGN is a step from the

traditional circuit switched PSTN to the packet-based IP network. NGN bears all

the services of the PSTN, offloads large amount of data transmission to the IP

network to reduce the load over PSTN, and supports new services and enhances

traditional services by taking full advantage of IP technology. Figure (1.1) shows

the evolution from PSTN to NGN. [1]

___________________________________________________________________________________________________________________________________

Next Generation Network Final Year Project - 2007 2


Tandem /toll exchange

LE

Packet core network

Trunk gateway

LE

Soft switch Call control

___________________________________________________________________________________________________________________________________

Next Generation Network

ININ NMSNMS App ServerApp Server Policy ServicePolicy Service

ISUP SG

PSTNswitchswitchswitch

STPSTPTMG

PLMN

WMG

SIPPhone

SIPPhone

H.323PhoneH.323Phone

PCPhone

PCPhone

IADIADAMGAMG

Packet core network

Packet core network

Soft switchSoft switchSoft switchSoft switch

Figure (1.1) evolution from PSTN to NGN.

The Next Generation Network (NGN) seamlessly blends the Public Switched

Telephony Network (PSTN) and the Public Switched Data Network (PDSN),

creating a single multi-service network. Rather than large, centralized, proprietary

switch infrastructure, this next generation architecture pushes central office (CO)

functionality to the edge of the network. The result is a distributed network

infrastructure that leverages new, open technologies to reduce the cost of market

entry dramatically, increase flexibility, and accommodate both circuit switched voice

and packet switched data. [1]

The NGN has the following features:

• Openness: NGN can be divided into several functional modules according to

different networks interworked and different functions provided. These

modules can not only be developed independently, but also act as a whole.

Meanwhile, such openness also enables the carriers to choose the best

products in the market as per their own requirements, without worrying about

the inter-working among different devices.

Final Year Project - 2007 3


• High efficiency: since NGN can separate service from call control, it provides

good conditions for the real independence of service from the network and

effective minimization of the development period for a new service.

Accompanying with the inter-working of multiple networks, many new

services are emerging.

• Multimedia: real-time transmission of voice, video and other streams is

another outstanding advantage of NGN.

• Low cost: compared with current PSTN, the adoption of relative cheaper

networks such as IP network as the transmission bearer in NGN greatly

reduces the communication cost. This advantage is more obvious in toll calls

and international calls. [1]

1.3 Problem Statement Transmission of voice through IP network using simple in-hand equipment is

our considered problem definition. This embraces a research on several stages for the

suitable scenarios to express and maintain the desired results, the means of

implementation, in addition to the achievement and evaluation of results.

1.4 Objectives The main objective of this thesis is to demonstrate the robust features of

providing voice services over NGN architecture. Our study would use simple tools in

order to prove NGN's capabilities via the current available individual VoIP

Protocols. Another objective is to highlight the impact of the convergence between

circuit-switched networks and IP networks by means of modeling the gateway

adaptation layer, using the familiar existing devices discussed in brief, shortly in this

chapter, and in more details later in the thesis.

1.5 Methodology

In order to approach the desired aim of this project, Java based software

packages implemented by NIST were used to build and interconnect the software

application components used for simulation purposes. The Session Initiation Protocol

(SIP), which was our option for signaling between the application elements, also

called as SIP entities was also implemented using the same technique. This included ___________________________________________________________________________________________________________________________________



the use of the JAIN (Java API's for Integrated Networks)-SIP to interface the SIP

protocol stack, implemented by NIST Corporation, to JAVA programming language.

The implementation of logical interfaces to the PSTN gateway used (typically a

voice modem) was accomplished through JTAPI phone dialer application (Java

Telephony application programmable interface) attached to our SIP environment in

such a way that replicates the real facility optimized by the NGN. Also it would

contain a simple demonstration example of inter-working between networks

elements of diverse nature. This would bring to mind the amount of impact the

integration process.

1.6 Thesis outline

This target is covered by a six chapter discussion gradually advancing stage

to stage in order to help understand all the related issues to the main concepts of the

thesis. These chapters followed the following structure: Chapter 1 is an introduction

to the main idea and the reasons that stood behind the choice of this topic in certain.

Also a highlight to the included simulation is found here. Chapter 2 and chapter 3

would introduce the backbone topic (the NGN) discussed and explained in more

details and in depth analysis. In Chapter 4 the simulation parameters are defined,

initialized and the pre-testing stage is established to obtain desired results. Chapter 5

contains the testing stage of the designed application and holds an evaluation to the

obtained results, this involves a comparison between the expected pre-testing results

and the obtained one, in order to analyze and discuss them. Chapter 6 represents a

typical conclusion to the work accomplished and our recommendations for future

work points.

___________________________________________________________________________________________________________________________________


CHAPTER 2

Next Generation Network basic concepts

Chapter 2- Next Generation Networks basic concepts _________________________________________________________________________________________________________________________________________

2.1 Introduction The NGN is mainly a concept that refers to packet-based networks enabling

convergence between voice and data on one hand, and between fixed and mobile

communications on the other hand. These networks provide multi-purpose services

through various access technologies. IP as the most wide-spread standard is regarded

to be a federative element in the new architecture. The NGN concept involves

decoupling of services and networks allowing them to be offered separately and to

evolve independently. The focus is on soft switch technology and packet voice and

the roles they play in the migration toward fully converged local networking. [1][2]

2.2 Migration to NGN Many options are available to service providers to pave their way to NGN and it

is very important to know how to evaluate them to determine whether it offers a

feasible NGN migration strategy. We review here the migration options, knowing that

each operator will find his way to the target NGN architecture by combining one or

several of the following strategies:

2.2.1 Wait & see The easiest path for an operator seems to continue with the existing technology,

waiting for NGN products to mature, keep investing in class 4 and 5 legacy switches.

This wait & see approach is reliable and well-understood but does not position the

service provider for a converged network strategy and keeps the service intelligence

in the legacy network, i.e. it does not represent a future-proof investment.[2]

2.2.2 Internet offload This strategy only consists in freeing resources for voice traffic in the switches.

Offload switches provide cost saving through more efficient use of legacy switches.

But this solution does not change fundamentally the PSTN architecture, as voice is

still carried over TDM, Separately from data backbone. [2]

___________________________________________________________________________________________________________________________________



2.2.3 Voice over broadband This approach provides packet voice from the access network, by carrying voice and

data on the same connection going out from the customer premises. [2]

2.2.4 Replace transit switches Traditional local-exchange switches are all based on circuit switching

techniques. Within the switch fabric, voice traffic is represented as 64kbps streams.

At the input and output ports of the switch, the 64kbps streams are time division

multiplexed into higher-speed digital facilities. The intelligence of the switch that

performs call routing and feature processing is integrated tightly with the circuit-

switching fabric. The economic advantages of packet voice are driving both the

access and core voice networks away from circuit switching and towards packet

switching. As packet voice becomes widely adopted for both access and core

networking, the traditional local-exchange switch represents an island of circuit

switching that connects these two packet voice networks. The packet-to-circuit

conversion that must be carried out at both input and output of the local-exchange

switch, however this introduces undesirable additional cost and transmission delays

into the voice path . If a local-exchange switching solution was available that was

capable of delivering local voice services and custom calling features directly over a

packet-switching infrastructure, then unnecessary packet-to-circuit conversions could

be avoided. This has the dual effects of reducing cost and improving quality, and it

moves the voice network a major step closer to the ultimate goal—homogeneous end-

to-end packet voice. [1]

2.3 The Soft switch 2.3.1 Definition Soft switch, shown in figure (2.1), is the generic name for a new approach

to telephony switching that has the potential to address all the shortcomings of

traditional local-exchange switches identified above. This section explains the basic

concept and describes the functional components of local-exchange soft-switching.

[1]

___________________________________________________________________________________________________________________________________



Figure (2.1) Soft Switch

2.3.2 The Soft switch Concept By far the most complex part of a local-exchange switch is the software that

controls call processing. This software has to make call-routing decisions and

implement the call processing logic for hundreds of custom calling features. Today’s

local-exchange switches run this software on proprietary processors that are integrated

tightly with the physical circuit-switching hardware itself. The inability of existing

local-exchange switches to deal directly with packet voice traffic, however, is a major

barrier to packet voice migration. In the future, delivery of local telephony will come

over a purely packet-based infrastructure. But for years to come, the migration path to

end-to-end packet voice will require working with a hybrid network handling both

packet and circuit voice. One possible solution to this is to create a hybrid device that

can switch voice in both packet and circuit formats, with all the necessary call

processing software integrated into this switch. While this approach may help address

the question of migration, it does not necessarily lower the cost of local-exchange

switching or improve the prospect for differentiated local voice services. The telecom

industry appears to have reached broad consensus that the best answer lies in

separating the call processing function from the physical switching function and

connecting the two via a standard protocol. In softswitch terminology, the physical

switching function is performed by a media gateway (MG), while the call processing

logic resides in a media gateway controller (MGC).

There are a number of reasons why this separation of functionality is believed to be

the best approach:

• It opens the way for smaller and more agile players who specialize in call

processing software and in packet-switching hardware respectively to make an ___________________________________________________________________________________________________________________________________



impact in an industry that has been dominated by large, vertically integrated

vendors.

• It enables a common software solution for call processing to be applied in a

number of different kinds of networks, including combinations of circuit-

based networks and packet voice networks using multiple different packet

voice formats and physical transports.

• It allows standardized commodity computing platforms, operating systems,

and development environments to be leveraged, thereby bringing considerable

economies to the development, implementation, and processing aspects of

telephony software.

• It allows a centralized intelligence in a service provider’s voice network to

remotely control switching devices located in customer premises.

• Being service driven it provides all the means needed to offer new services

and customize existing ones in order to generate future revenues. [1][2]

2.4 Next Generation Network Characteristics This next-generation switching architecture represents an entirely new

approach to delivering services that is specifically designed to accomplish the

following services:

• Deliver robust switching functionality at a cost that is an order of magnitude

lower than traditional, proprietary Class-5 switches.

• Distribute switching functionality to the edge of the network.

• Protect existing investments by supporting all current analog and digital

network standards, interfaces, media, and service elements.

• Reduce the number of network elements by combining a range of telephony,

application, and service-delivery functions.

• Enable new service creation through programmability and the flexibility of

an open application programming interface (API).

• Provide a high degree of scalability, enabling network operators to expand

their subscriber base rapidly and cost-effectively.

• Promote extensibility through open architecture design and, thus, take

advantage of future technological advances.

___________________________________________________________________________________________________________________________________



• Redefine true, carrier-class design for maximum fault tolerance and zero

downtime.

• Reduce operating costs by employing advanced remote maintenance and

diagnostics capabilities.

• Increase revenues by shortening time to market, reducing upfront costs,

and providing remote management capabilities.

• NGN provides converged services based on open common services platform.

It uses a common IP core network which combines all types of access links

and user services. This is shown in figure (2.2).

Figure (2.2) Next-Generation in a variety of

Network deployments

2.5 Conclusion This chapter has described a migration path for broadband packet-voice

access: a migration moving from a transport-only solution that relies on a

conventional local-exchange switch toward a full-fledged local-exchange soft

switching and access solution that delivers packet voice dial tone.

___________________________________________________________________________________________________________________________________


CHAPTER 3

Next Generation Network Architecture

Chapter 3- Next Generation Network Architecture _________________________________________________________________________________________________________________________________________

3.1 Next Generation Network Layers NGN is divided into four layers: edge access layer, core switching layer,

network control layer and service management layer. The NGN architecture is shown

in figure (3.3). Inter-working between different layers is realized via open standard

protocols, which provides NGN with considerable advantages and flexibilities.

Edge Access

Core Switch

Network Control

Service Management

Figure (3.1) The NGN Layer Architecture

3.2 The Edge Access Layer Edge access layer is used to connect subscribers and terminals by a variety of

means, and convert the original information format to the suitable one that can be

transferred over the network. Devices at this layer, as shown in the figure below, are:

IAD AMGBroadBand

Access

WMG

PLMN/3G

SG TMG

PSTN

UMG

Packet Core Network

Edge Access

Figure (3.2) Edge Access Layer Devices

___________________________________________________________________________________________________________________________________



3.2.1 Integrated Access Device (IAD) It is a type of subscriber access device used in the NGN architecture. It

introduces data, audio, video and other services of the subscribers to the packet-based

network. IAD encapsulates the voice-band signals into IP packets with standard voice

codec and compression technologies, and sends the IP packets over the IP network to

the called MG, which will make a reverse conversion on the IP packets to recover the

original voice-band signals. [1]

3.2.2 Media Gateways (MGs) Controlled by the Soft switch, the function of the media gateway is to adapt

user data to the backbone network based on a packet switching technology (IP or

TDM e.g.). It terminates voice calls from the TDM side, compresses and pocketsize

voice data, and delivers the compressed voice packets to the packet network. On the

opposite way of a call, it receives the voice packet from the packet network,

unpacketizes and decompresses them, and delivers them to the TDM side. [1]

3.2.2.1 Media Gateway Characteristics MGs always have a master/slave relationship with the Media Gateway

Controller (MGC) that is achieved through a control protocol such as Media Gateway

Control Protocol (MGCP). MGs serve the following functions:

1- Perform of media processing functions such as:

• Media transcoding.

• Media Packetization.

• Echo cancellation.

• Jitter buffer management.

• Packet loss compensation.

2- Perform of media insertion functions such as:

• Call progress tone generation.

• DTMF generation.

• Comfort noise generation.

3- Perform of signaling and media event detection functions:

• DTMF detection.

___________________________________________________________________________________________________________________________________



• On/off-hook detection.

• Voice activity detection.

4- May have the ability to perform digit analysis based on a map downloaded

from the MGC

5- Provides a mechanism for the MGC to inspect the state and capabilities of the

endpoints in call. [1]

3.2.2.2 Types of Media Gateways MGs are divided to four different types depending on their functions:

1- Access Media Gateway

It is another example of subscriber level gateways, but with much higher

capacity than the IADs. AMGs provide narrowband and broadband service access.

The AMG provides narrowband and broadband service access. The AG transfers

subscriber line data such as voice, modem and fax across the NGN through media

stream conversion. The AMG interacts with the Soft switch device by MGCP,

accepting control from the latter, reporting subscriber line status and processing

subscriber calls. [1]

2- Trunk Media Gateway (TMG)

Here is the first network to network interfacing technique comes to scene.

TMGs are resident between the circuit switched network and the IP packet switched

network, achieving the conversion function between the public switched telephone

network (PSTN) and the IP network. The TMG interacts with the Soft switch device

through MGCP, accepting control from the latter and carrying out establishment and

disconnection of calls and other services. [1]

3- Signaling Media Gateway (SG)

Located at the interface layer of the Signaling System No. 7 and the Internet

Protocol (IP) this device realizes the signaling conversion function between the public

switched telephone network (PSTN) and the IP network. Signaling gateway function

provides a gateway for signaling between a VoIP network and the PSTN (SS7/TDM)

based. For wireless mobile networks it also provides a gateway for signaling between

IP based mobile core network and PLMN that is based on either SS7/TDM or

BICC/TDM. The primary role is to encapsulate and transport PSTN (ISUP or INAP)

or PLMN (MAP or CAP) signaling protocols over IP. [1]

___________________________________________________________________________________________________________________________________



4- Universal Media Gateway (UMG)

This device is a combined version of a TMG and an SG, but has lower signaling

conversion capabilities. It converts the media stream and signaling between different

formats. It can act as a built-in SG or AMG. It can connect a variety of devices

including PSTN exchange (PBX), access network, network access server (NAS) and

base station controller (BSC) for wireless networks applications. [1]

The Media Gateway functionality would be discussed in more depth later on in

the next sections of this chapter.

3.3 The Core Switching Layer Core switching layer adopts the packet technology, and is composed of the

devices distributed over the backbone network and the Metropolitan Area Network

(MAN), such as routers and layer-3 switches. It is used to provide subscribers with a

uniform and integrated transmission platform with high reliability, Quality of service

(QoS) assurance and a large capacity. [1]

However, as the Internet has emerged as the network of choice for providing

the converged services approved by NGN, the demands placed on the IP core

network, in terms of speed and bandwidth, have strained the resources of the existing

Internet infrastructure. This transformation of the network towards a packet- and cell-

based infrastructure has introduced uncertainty into what has traditionally been a

fairly deterministic network. In addition to the issue of resource constraints, another

challenge relates to the transport of bits and bytes over the backbone to provide

differentiated classes of service to users. The exponential growth in the number of

users and the volume of traffic adds another dimension to this problem. Class of

service (CoS) and QoS issues must be addressed to in order to support the diverse

requirements of the wide range of network users. [1]

In sum, despite some initial challenges, the Core Layer still plays an important role in

the routing, switching, and forwarding of packets through the next-generation network

in order to meet the service demands of the network users.

___________________________________________________________________________________________________________________________________



3.4 Network Control Layer Network control layer, shown in figure (3.3), is responsible for implementing

call control. Its core technology is soft switching, which is used to achieve basic real-

time call control and connection control functions. [1]

Control Layer SoftSwitch

Figure (3.3) Network Control Layer

The control layer handles the call setup and controls the media gateways. Major

components at this layer are the Soft switches. The SoftSwitch is a central device in

the Telecommunication network which connects calls from one phone line to another

entirely by means of software running on a computer system. This work was formerly

done by hardware. The SoftSwitch architecture, show in figure3.4, involves the

separation of media path (voice packets over IP e.g.) and media conversion functions

from the call control and signaling functions. It is mainly split up into two basic

components:

• Media Gateway Controller (MGC) which handles the call control

functionalities. Also known as Call Agents.

• Media Gateways which are responsible of media conversion.

SoftSwitch SoftSwitch

3G AccessAMG IAD Broadband

Access PSTN TMG SG

PLMN

IP Core Network

UMG

Call control path

Talking Path UMG

Figure (3.4) Separation of Call control from bearer

___________________________________________________________________________________________________________________________________



The advantage of the SoftSwitch is its distributed architecture. For a network

operator, it is possible to use different network components from different vendors,

such that the best in each class may be chosen in each area. For equipment vendors, it

is possible to focus efforts on one area and not to have to develop acquire expertise in

all area. [1]

3.5 The Service Management Layer The service management layer is mainly used to provide supplementary

value added services and operation support based on established calls. It comprises of

Application and feature servers, as shown in figure (3.5), like:

Service Management IOSS

PolicyServer Application

Server SCPLocationServer MRS

Figure (3.5) The Service Management Layer

3.5.1 Integrated Operation Support System (IOSS) The acronym of integrated Operation Support System, which includes two

parts: Network Management System (NMS) for managing the NGN network elements

in a centralized way, and integrated charging system. [1]

3.5.2 Policy Server This server is used to manage the policies of the subscribers, such as Access

Control List (ACL), bandwidth, traffic, and Quality of Service (QoS). [1]

3.5.3 Application Server It is responsible for generating and managing logics of various value added

services and intelligent network services, and providing innovation platform for

developing third-party services by means of open APIs. As a physically separated

component, Application Server is independent of the SoftSwitch equipment. This

___________________________________________________________________________________________________________________________________



contributes to the separation of service from call control and is beneficial to the

introduction of new services. [1]

3.5.4 Location Server This is used to dynamically manage the routes between the Softswitch

equipment in the NGN, indicate reach ability of the destinations of calls, ensure the

best efficiency of call routing table, prevent the routing table from being oversized

and impractical, and abates the complexity of routes. [1]

3.5.5 Media Resources Server It is used to enable the media processing functions in the basic and enhanced

services. The functions include service tone provision, conference service, Interactive

Voice Response (IVR), recorded announcements and advanced tone service. [1]

3.5.6 Service Control Point It is the core component of the traditional Intelligent Network (IN), and is used

to store subscriber data and service logics. According to the call events reported by

the Service switching point (SSP), SCP starts an appropriate service logic, retrieves

the service database and the subscriber database based on the started service logic,

and then sends proper call control instructions to the corresponding SSP to instruct the

SSP how to perform next, thus realizing various intelligent calls. That is the main

function of the SCP. [1]

3.6 Conclusion After this comprehensive realization of the NGN architecture, the complete

NGN’s four-layer architecture model is shown in figure (3.6).

___________________________________________________________________________________________________________________________________



IAD AMGBroadband

Access

WMG

PLMN/3G

SG TMG

PSTN

UMG

Service Management

Packet Core Network

Network Control

Core Switching

Edge Access

SoftSwitch SoftSwitch

iOSS PolicyServer

ApplicationServer

SCPLocationServer

RADIUSServer

MRS

Figure (3.6) NGN Architecture

___________________________________________________________________________________________________________________________________


CHAPTER 4

VoIP Implementation in NGN

Chapter 4- VoIP Implementation In NGN _________________________________________________________________________________________________________________________________________

4.1 Introduction

Voice over Internet Protocol (VoIP) is one benefit of the convergence

between data and telecommunications introduced by NGN platforms. Companies

today are seeing the value of transporting voice over IP networks to reduce telephone

and facsimile costs and to set the stage for advanced multimedia applications and

services such as unified messaging, in which voice, fax, and e-mail are all combined.

NGN systems were very successful in the matter of standardization and

interoperability of VoIP service. The adoption of several VoIP protocols not only

enables the NGN infrastructure to represent a reliable VoIP service provider, but also

introduces improved functionalities to the users, indeed with irresistible cost

proposals. [3]

4.2 Problem Definition and Decomposition The main problem is transferring voice throughout IP networks instead of

using circuit switched PSTNs, as these are the transport routes for media in the NGN

platform. Now to simulate the mechanisms deployed by the NGN infrastructure used

to provide VoIP service, four stages of implementation have been followed:

• Identification of desired scenarios in the simulation.

• Specification of the Signaling Protocol most suitable to accomplish the call

processes in the required scenarios successfully.

• Determination of the appropriate Media Control Technique.

• Design & Implementation.

4.2.1 The Scenarios

In order to make VoIP simple to implement, figure 4.1 decomposes the main

problem into sub problems, each of which represents a simple scenario that is an

undividable sub problem.

___________________________________________________________________________________________________________________________________



Figure (4.1) decomposition of the problem.

4.2.1.1 PC to PC phase Beginning with the pc to pc phase, which is considered the smallest unit of

all scenarios in our simple hierarchy, the main objective of this part is to define

the pure IP based signaling and call setup. The SIP (Session Initiation Protocol) as

would be later on discussed was chosen as the desired method to initiate a voice

session in an IP environment. Now in order to transmit voice between two PC's

includes the following concerns:

• Voice capture:

As a hardware device the user needs a sound card. Plug the microphone into

its appropriate place. User will use the microphone to capture the voice that he/she

wants to send to destination.

• Voice play

Any typical presentation device can be used, such as, sound players (speakers).

4.2.1.2 PC to PHONE phase The second phase's objective is to demonstrate the interoperability between

two different networks (IP-TDM) whereas a hardware adjustment is recommended in

order to secure the desired compatibility and convergence between these networks.

Now when talking about gradual migration to pure IP solutions, current TDM

structured networks must be considered in our scenarios. There are a lot of feasible

___________________________________________________________________________________________________________________________________



hardware devices from different vendors that undertake the process of converting

digital representation of voice into analog and vice versa so the choice should depend

on the project requirement. The project requirement is to provide a reliable VoIP

telephony application with acceptable quality of service but the main requirement is to

reduce the cost as much as possible. There are two recommended solutions in this

field:

1- Dialogic cards: dialogic cards provide high quality telephony applications (QoS)

simple to implement because it comes with libraries helps programmers but are

expensive.

2- Dial up modem provides a medium quality of service more complex to implement

but less cost.

The optimal solution that achieves the goal of the project is to use dialup

modem since the concern is the cost plus using a regular modem can be easily

developed to any better hardware such as dialogic cards.

4.2.1.3 PHONE to PHONE phase This is the most efficient scenario that is similar most to the actual NGN

concept of pushing access to the edge of the network and transporting media in RTP

(Real Time Protocol)/UDP/IP streams in the IP core network. This demonstration

explains why exactly are NGNs cost effective at network management techniques, in

addition to extreme utilization of the open bandwidth offered in Packet based

Networks. Analog voice is transferred from PSTN (TDM) networks at the originating

point, packetized in Media Gateways and sent through the IP core. At the destination,

after accomplishing a successful signaling session, a gateway once again reconstructs

the transformed data into the original TDM form, therefore ready for the end device.

4.2.2 The Signaling Architecture The previously mentioned scenarios need to be implemented in such a manner

that provides a kind of entity interconnection that is analogous to the one actually

applied in the NGN infrastructure. This means that each participant of a certain

scenario must be defined carefully to a centralized controller and must be totally

distinguishable from other entities. To provide Interoperability between different

entities in our scenarios, VoIP applications need a standard signaling protocol. There ___________________________________________________________________________________________________________________________________



are many available signaling protocols in the NGN environment such as H323, SIP

and MGCP. Our choice has landed on the SIP (Session Initiation Protocol) for several

reasons discussed in detail in the next section of this chapter.

4.3 Session Initiation Protocol (SIP) 4.3.1 Definition Session Initiation Protocol or SIP is the IETF standard for voice or

multimedia session establishment over the Internet. SIP is an application level

protocol used for call setup management and teardown. The SIP architecture is similar

to HTTP (client-server protocol) architecture. Its message structure was based on

SMTP (email), with the simple, text-based, extensible form that had helped to make

email so successful. It comprises requests that are sent from the SIP user client to the

SIP Server which processes the request and responds to the client. SIP makes minimal

assumptions about the underlying transport protocol and it provides reliability and

does not depend on the underlying protocol’s characteristics .The SIP protocol due to

simplicity and easier implementation was our preferred option over other NGN

protocols. [3][4]

4.3.2 Comparison between H.323 and SIP H.323 and SIP both support VoIP and multimedia communications, but SIP is a

relatively new protocol as compared to H.323 and hence, has been able to avoid all

the problems associated with H.323. Table (4.1) shows some of the comparison

features between H.323 and SIP from which it is clear that SIP protocol is more

scalable, extensible, with less complexity and easy on implementation, without need

for special parser, customization and call forking with Third-party call control.[1][4]

___________________________________________________________________________________________________________________________________



Table (4.1) Comparison between SIP & H.323

Feature

H.323

SIP

Architecture Stack Implementation Element Implementation

Complexity Complex Simple

Standards body ITU IETF

Protocol Mostly TCP Mostly UDP

Protocol

Encoding

Binary (ASN.1, Q.931) Text (HTTP-ish)

Server

processing

State-full State-less, Transaction oriented

Addressing Aliases, email SIP URLs

Call Setup

delay

V1: 6-7x RTT to V3: 1.5-2.5x

RTT

1.5x RTT

Mid-call failure Fail Live

Loop Detection V1:No, v3: Path Value Yes – “via” field, time, hops

Manageability Yes No

Call control Yes Yes

Emphasis Telephony Multimedia, multicast

Third party call In al versions No Yes

Fault tolerance v1 No,v1 No, v3 backup Yes

4.3.3 SIP Architecture SIP communication is made up of messages that are sent between the devices

using UDP, TCP, or another transport protocol. These messages are either requests or

responses and contain a set of headers, which are the parameters of the message, and

one or more message bodies, as required by the application. A single SIP request and

all its responses form a SIP transaction. Different types of transaction are used for

different protocol functions. SIP transactions can exist within or outside a SIP dialog,

and transactions are used to establish and terminate dialogs. [1] 4.3.3.1 SIP Entities The main components of a system employing SIP which are shown in figure

(4.2) are mentioned below:

___________________________________________________________________________________________________________________________________



Figure (4.2) SIP Entities

1- User Agents (UA) are endpoint devices that terminate the SIP signaling. They

can be clients (UAC) that initiate requests, servers (UAS) that respond to

requests, or more normally a combination of the two.

2- Proxies are devices in the signaling path between User Agents that route

requests on towards their destination. They may add parameters to the requests

and may reject requests, but they may not initiate requests or respond

positively to any request that they receive.

3- Registrars are specialized User Agent Servers that handle REGISTER

requests. SIP devices use REGISTER requests to dynamically register their

current location, and this enables them to be contacted when mobile.

4- Location servers maintain a database holding the location of all known User

Agents within a domain.

5- Redirect Servers are specialized User Agent Servers that respond to requests

by redirecting them to another device. [1][4]

4.3.3.2 SIP Messages

SIP uses messages for call connection and control. There are two types of SIP

messages: requests from client to server and responses (status messages) from server

to client. For all messages, the general format is:

• A start line.

• One or more header fields.

• An empty line.

___________________________________________________________________________________________________________________________________



• A message body (optional).

• Each line must end with a carriage return-line feed (CRLF).

SIP message syntax is shown in figure (4.3). Message header provides additional

information regarding the request or response. The message body normally describes

the type of session to be established, including a description of the media to be

exchanged. [1]

Figure (4.3) SIP message syntax

Request: SIP request starts with a request-Line which begins with a method token,

followed by the Request-URI and the protocol version, and ending with CRLF. The

request-line specifies the type of request being issued. SIP uses six types (methods) of

requests:

1- INVITE

The INVITE method indicates that the user or service is being invited to

participate in a session. The message body contains a description of the session to

which the callee is being invited. For two-party calls, the caller indicates the type of

___________________________________________________________________________________________________________________________________



media it is able to receive and possibly the media it is willing to send as well as their

parameters such as network destination. A success response MUST indicate in its

message body which media the callee wishes to receive and MAY indicate the media

the callee is going to send.

2- ACK

The ACK request confirms that the client has received a final response to an

INVITE request.

3- BYE

The user agent client uses BYE to indicate to the server that it wishes to

release the call. A BYE request is forwarded like an INVITE request and MAY be

issued by either caller or callee.

4- CANCEL

The CANCEL request cancels a pending request with the same Call-ID, To,

From and CSeq (sequence number only) header field values, but does not affect a

completed request. (A request is considered completed if the server has returned a

final status response).

5- OPTIONS

This method queries the capabilities of servers.

6- REGISTER

A client uses the REGISTER method to register the address listed in the To

header field with a SIP server. [1][4]

Response: The start of a SIP response is a status line. This contains a status code, which

is a three digit number indicating the outcome of the request. The status line will also

contain a reason phrase which provides a textual description of the outcome. The

status codes defined in SIP have values between 100 and 699, with the first digit of

the reason code indicating the class of response as shown in table (4.2). [1]

___________________________________________________________________________________________________________________________________



Table (4.2) SIP response codes

ExamplesDescription 100 trying.

180 ringing.

181 call is being forwarded.

Informational.1xx

200 OKSuccess.2xx

300 multiple choices.

301 moved permanently.

302 moved temporarily.

Redirection.3xx

400 bad request

401 unauthorized.

406 not acceptable.

408 request timeout.

415 unsupported media type.

Client error.4xx

502 bad gateway.

503 service unavailable.

505 version not supported.

Server error.5xx

600 busy everywhere.

603 decline.

Global failure.6xx

As an example, if client A wants to set up an IP telephony session with client B,

A sends an INVITE request to B. The INVITE message contains a payload with a

description of the session he/she wants to set up with B. When B accepts the call, his

user agent sends a message with a response code of 200. Finally, A sends an

acknowledgement to B confirming that he/she received the response from the callee.

This three-way handshake procedure is shown in figure (4.4). [5]

___________________________________________________________________________________________________________________________________



Figure (4.4) setting up a SIP session

4.3.4 Session Description Protocol (SDP)

Although a SIP message body can carry many different types of information,

the most common message body is the session information describing the media to be

exchanged between the parties. The session description includes media information

such as RTP payload type, address and ports. The format of the description will

normally be according to SDP. SIP uses SDP in an answer/offer mode. A caller sends

an invite with a SDP description that describes the set of media formats that

comprises an offer by the caller. The called party responds with a SDP description

that aligns with the offered SDP description. This exchange results in an agreement

between the two parties as to the type of the media used. The SDP text message

includes: • Session name and purpose.

• Time the session is active.

• Media comprising the session.

• Information to receive the media (address…etc). [1][5]

4.3.5 SIP addressing

SIP uses email-style addresses to identify users. SIP addresses are known as

Uniform Resource Indicators (URIs) and take the form User@ Host, where the User

can be a name or telephone number and the Host can be a domain or IP address. [1]

___________________________________________________________________________________________________________________________________



4.3.6 Inter-Working between SIP network & NGN The SIP entities, as shown in figure (4.5), are distributed in the NGN layering

scheme forming an ideal IP voice service model.

R

Figure (4.5) inter-working between SIP network and NGN.

The NGN control layer, represented in the Softswitch, uses the SIP protocol

to perform call control functionalities on an IP based platform. The most interesting

issue that both SIP & NGN share is the entire separation between the signaling and

media paths in addition to the simplicity of the text based SIP messages used for call

control. Therefore SIP is considered one of the favored options when optimizing a

VoIP service in the NGN. As shown in the diagram above a SIP application server

(Proxy, redirect & location server) is connected to the softswitch where it maintains

centralization as a call control layer element. For SIP users in the edge access layer no

need of any interfacing to the transport layer (IP core) as the SIP is already supplied

with IP intelligence in addition to accommodating several media transport protocols

over IP networks (RTP, TCP, UDP). When interoperability is required when

connecting to other network users, such as PSTN, the proxy servers are easily directed

via the Softswitch to negotiate with Media Gateways which are responsible of

successful interfacing between different integrated networks in the NGN environment.

[2] ___________________________________________________________________________________________________________________________________



4.4 Real Time Protocol (RTP) After setting up a call connection between two parties using SIP, the packetized

segments of compressed speech are carried over the IP network using the Real-Time

Transfer Protocol (RTP). RTP provides end-to-end delivery services for data with

real-time characteristics, such as interactive audio and video or simulation data, over

multicast or unicast network services. Applications typically run RTP on top of UDP

to make use of its multiplexing and checksum services; both protocols contribute parts

of the transport protocol functionality. However, RTP may be used with other suitable

underlying network or transport protocols. RTP supports data transfer to multiple

destinations using multicast distribution if provided by the underlying network. [3][5]

RTP itself doesn’t provide any mechanism to ensure timely delivery or to

provide other QoS guarantees, but relies on lower-layer services to do so. It doesn’t

guarantee delivery or prevent out-of-order delivery, nor does it assume that the

underlying network is reliable and delivers packets in sequence. The sequence

numbers included in RTP allow the receiver to reconstruct the sender’s packet

sequence. [3]

RTP consists of two closely-linked parts:

1. Real Time Transfer Protocol (RTP), to carry data that has real-time properties.

2. Real Time Control Protocol (RTCP), to monitor the quality of service (QoS) and

to convey information about the participants in an on-going session. [3]

___________________________________________________________________________________________________________________________________


CHAPTER 5

Simulation Design & Evaluation

Chapter 5- Simulation Design & Evaluation _________________________________________________________________________________________________________________________________________

5.1 Introduction This is the most interesting part in the project. It shows exactly the

accomplished work. It also explains the design and implementation of all components

and how they interact together to achieve the project’s goal. All the software packages

that have been used to construct this simulation are products of NIST. These are

illustrated below.

5.2 Java Interface to SIP (JAIN-SIP)

JAIN-SIP is the standardized Java interface to the Session Initiation Protocol for

desktop and server applications. It enables interoperability between stacks and

different applications over the SIP protocol stack. When a complete application is

constructed of a number of units, such as the one used in the design (next section),

there must be an obvious interface between the different functionalities to provide a

standard, uniform, SIP messaging scheme applied all over the application operation

time. Here the JAIN-SIP is responsible for providing methods to format SIP

messages, in addition to enabling applications to send and receive SIP messages. [6]

5.3 The Application Architecture The implemented application consists of four components:

• User agent

• Proxy server

• Registrar server

• PSTN gateway

5.3.1 The User Agent (UA) 5.3.1.1 Fundamentals Used by the end user to communicate with other user agents. This application

consists of four major classes as shown in figure 5.1.

1- Sip Communicator is the main class which centralizes the control of other classes.

2- Sip Manager:

It is responsible for managing all sip operations, with the assistance of

subprograms, operations are:

___________________________________________________________________________________________________________________________________



• Registering: When Sip Communicator starts up it sends REGISTER request to

location server or registrar server.

• Unregistering: Sends unregistered request when user exits.

• Call initiation: When user wants to call another user he/she initiates a call, Sip

Manager sends INVITE request to specific user.

• Handling incoming messages: Sip-Manager handles all messages which are

delivered by Sip-Provider whether they are requests or responses

Figure (5.1) User agent architecture

3- Media Manager

Responsible for playing, capturing and streaming voice data, with the assistance

of subprograms, includes:

• Audio transmission: Sending encoded captured audio using RTP.

• Audio reception: Receiving, decoding and playing audio streams.

4- GUI (Graphical User Interface) Manager

Responsible for displaying and updating graphical user interface, operations

include:

• Respond to user actions and initiates the corresponding events (e.g. when dial

button is pressed the GuiManger informs the SipManger to invite the callee).

• Update the interface (e.g. when SipManager reports an incoming call,

GuiManger displays an Alerting call). [6]

___________________________________________________________________________________________________________________________________



5.3.1.2 User Agent demonstration

• At start up user can configure the stack properties as shown in figure (5.2):

Figure (5.2) The Stack Properties User Interface Window

• In order to register, a user is asked to enter user name and password, this is

shown in figure (5.3). The Registrar compares user name and password with a

predefined user name and password stored in an xml file.

Figure (5.3) Authentication of a User Agent

___________________________________________________________________________________________________________________________________



• The registration status is displayed on the window after registration process is

accomplished successfully, the user agent is then capable of establishing a call

as being a registered user at the SIP proxy.

• When a user agent is to make a call, simply the login name of a registered

callee is needed, and a dial option is provided on the window to enable call

initiation. The invitation process is then passed to the proxy, which on behalf

of the call party sends an invitation to the desired called party.

• Figure (5.4) demonstrates a call request and shows the invitation process:

Figure (5.4) A user agent calling another user agent

5.3.2 Proxy and Registrar server This package contains the source code of a java based SIP proxy built on top

of the JAIN-SIP-1.1 API. The proxy also functions as a SIP registrar and a SIP

presence server. [6]

5.3.2.1 Proxy capabilities

• Registration upload:

You can specify a set of registrations that will be uploaded into the proxy at

start-up time. The file to modify is "registrations.xml" located in the "configuration/"

directory.

___________________________________________________________________________________________________________________________________



• Forking capability:

The proxy can fork the INVITE requests it receives to different location (e.g.

work and home) for a single user. [6]

5.3.2.2 How to start the proxy • Change to the proxy directory (gov/nist/sip/proxy).

• Edit the configuration file (configuration/configuration.xml).

Use the build.xml file provided in the directory to start the proxy, this configuration is

shown in figure (5.5). [6]

Figure (5.5) Proxy server configuration

5.3.2.3 System Administrator System administrator job is to add, delete and modify users' information. Users’

information includes users names, passwords and a 4 digits number to receive calls

from PSTN, these information stored in an xml file loaded by the proxy at start up.

The 4 digits number associated with each user name used at gateways as the PSTN

caller can only enter digits to refer to callee, these digits must be then mapped by the

gateway. When a gateway registers, proxy attaches each users name with the

corresponding number in the register response message as a string named Contact, so

the gateway can do the mapping. [1]

___________________________________________________________________________________________________________________________________



5.3.3 SIP/PSTN Gateway 5.3.3.1 Preface This is the part of most importance to our contribution in this project. This

gateway is a simplified simulation of the Signaling Gateways (SGs) which represent

the interface of any kind of network access in the edge layer of the NGN model to

maintain suitable interoperability with the NGN intelligence layer. The actual

Signaling gateway performs transformations of different signaling formats in the

(Time Division Multiplexing) TDM circuit switched infrastructures such as ISUP

(ISDN User Part in SS7) to an IP call control form, such as SIP, which is understood

by the Softswitch. Media gateways (MGs) cooperate together with SGs in order to

accomplish complete call initiation and progress in media flow between the integrated

networks.

Our design would not follow the same mechanisms performed by the SG, but it would

obtain very close results in the achievement of establishing a SIP call (NGN/IP

format) to a PSTN user (TDM format).

5.3.3.2 Basic Structure of PSTN gateway It is an application connects the IP network with the PSTN. The problem is

well known and can be divided it into two sub problems: 1- An application that deals with SIP user agents. This was already maintained by

designing the SIP User Agent.

2- An application that dials, receives and answers PSTN phone calls.

So, the solution is an application with two interfaces one with the network that is a

regular SIP user agent, the other with the PSTN that is a regular telephony application

which have the capability of dialing and answering regular phone calls. [1]

5.3.3.3 PSTN Connection part of the gateway As mentioned earlier in this thesis, voice modems are the hardware required

to connect to PSTN. So it is necessary to have a method that makes voice modems

dial, answer, transmit and receive audio streams. Many solutions are available, but the

chosen one which suits this application requirement is Java Telephony API (JTAPI)

that incorporates telephony functionality into java applications.

5.3.3.4 How PSTN Gateway works

___________________________________________________________________________________________________________________________________



As illustrated in figure (5.6), the gateway component is a user agent with a

JTAPI manager.

Figure (5.6) Gateway architecture

Basically the JTAPI manager is a simple JTAPI phone which has the capability to dial

and receive incoming phone calls. It exchanges events with SipCommunicator to

accomplish Gateway job. In a pc to phone scenario, the Gateway is supposed to dial

the callee number and then instruct the SipCommunicator to answer the ALERTING

call. This is achieved through a sequence of events mechanism:

1- A pc invites the Gateway.

2-Gateway takes the number to dial from a header called MyHeader in the

INVITE request (to be explained soon in details)

3- JTAPI manager dials callee number.

4- JTAPI manager fires a XHandleAnswerRequesEvent to Sip Manager:

5- Sip Manager catches the XHandleAnswerRequesEvent, then answers the

ALERTING CALL (sends OK to the INVITE).

6- In a phone to pc scenario, Gateway is supposed to detect DTMF tones which

represent the callee address then:

7- JTAPI manager fires the XUserCallInitiationEvent to SipManeger.

8- Sip Manager catches the XUserCallInitiationEvent, and then sends an INVITE

to callee (after mapping the detected digits to callee address).

9- In both scenarios when a terminal is dropped (whether the Gateway is the call

initiator or not), JTAPI manager fires XhandleHangupRequest event to

___________________________________________________________________________________________________________________________________



SipManeger. When SipManeger catches the event it ends the current call

(termination of media streams, remove call from GUI, etc…). [1][6]

The PSTN gateway configuration is shown in figure (5.7).

Figure (5.7) PSTN Gateway

5.3.3.5 Detailed explanation of Gateway functionality There are three scenarios:

• PC to PHONE scenario

Suppose a user agent wants to call a phone. Phone numbers are

categorized according to area codes e.g. a phone number in Khartoum state begins

with (01). So the user agent will call 01xxxxxxxx. Each area has a Gateway which is

responsible for routing calls to it. Initially each Gateway registers to proxy with its

area code. When the user agent sends INVITE to 01xxxxxxxx, user agent will check

the callee address (01xxxxxxxx) if its digits are all numbers it sends an INVITE

containing the first two digits in addition to the dialed number, to invite the desired

gateway. The Gateway is now ALERTING, it will extract the dialed number in the

received request and dials the number it contains through the modem using XTAPI.

After placing the call, the Gateway answers the alerting call (sends OK to caller) and

opens the media streams. Now caller (pc) and callee (phone number) are connected.

[1][6]

___________________________________________________________________________________________________________________________________



• PHONE to PC and PHONE to PHONE scenario

PHONE to PC and PHONE to PHONE are a very similar to the gateway, the

different is whether the gateway should calls a local sip user agent or forward the call

to the appropriate gateway.

The steps in both scenarios are:

Step1: PSTN user calls the gateway.

Step2: Gateway answers the call automatically and asks the caller to enter 4 digits.

Step3: Gateway maps the retrieved 4 digits using the Contact string received at

registering and comes up with the callee address.

Step4: The gateway must check whether the resulting callee name is a sip user agent

or a PSTN phone number.

Step5: If the resulted callee name is just a local sip user agent the gateway construct a

simple INVITE request, but if it turned to be a PSTN number the gateway calls the

appropriate gateway the same way as the PC to PHONE scenario.

(The gateway may ask the caller to start the number with ‘*’ if he wishes to call a

PSTN number).

Step6: The rest is like a normal PC to PC scenario. [1][6]

5.4 Testing and Analysis of Results By accomplishing the design and implementation stage of the application

components, the testing stage may be immediately executed. This would take us to the

scenarios previously suggested in the last chapter in order to carefully observe and

evaluate the obtained results after testing the performance of the application.

5.4.1 Testing Example • As an example consider a company with two branches one in Khartoum the

other in Madani. Each branch has a single telephone line. Branches are

connected with an IP network (WAN). An Employee uses his name to register

to the server and then can be called or can call other employees by dialing

their names. This is typically a Pc to Pc scenario.

• Any outside caller calling an internal employee from Khartoum or Madani

dials the company phone number (Khartoum number if calling from

___________________________________________________________________________________________________________________________________



Khartoum). If gateway phone line is not busy, it will answer and plays a

welcome message to him. It asks caller for callee number. Company policies

determine employees who can be reached from outside (from PSTN). Each

one of those employees will have a number correspondents to him because

employees are registered with there names and a PSTN caller can not specify

names using a telephone or a cell phone. When gateway answers it starts

detecting any digits pressed by the caller to determine callee using the JTAPI

DTMF tone digits detection which detects any tone digits in the phone line (*,

#, any numbers from 0 to 9).

• After detecting callee number gateway maps the digits to a name and then

send an INVITE to that user. Now the caller is connected to the callee pc

through the gateway, if BUSY, TEMPORARY UNAVAILABLE is returned,

the caller listens to busy tone and gateway hangs up to release the modem. If

callee answers an RTP stream is opened between callee and the gateway. Now

the gateway is connected to a PSTN user so gateway instead of playing the

received stream in its speakers it plays it in the modem so caller can hear

callee. And instead of playing received audio from modem in its speakers it

plays it in the microphone (sound card) so callee can receive caller voice.

• When an employee wants to call a PSTN user it dials its number preceded by

gateway number (INVITE gateway). Gateway dials the callee number and

then answered the ALERTING call. Caller is now connected to callee as in the

same previous scenario (Phone to Pc).

5.4.2 Evaluation of Results Consider this scenario to show how national calls cost between Khartoum

and Madani is reduced. An employee in Khartoum can call a PSTN user in Port Sudan

and still charged as calling locally. He/She will dial callee number preceded by

Madani gateway address. So connection from Khartoum to Madani will be through

the IP network, in Madani the gateway will dial callee phone number which will be

charged locally. Same concept can be applied using the internet which is a

conventional IP network but connects users located in large geographic areas. So a

user can call from Sudan to USA and only charged as he/she is calling locally in

USA. The last case introduced here is rather a more critical one, as VoIP protocols

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have taken various schemes and protocols all over IP environment, in addition to

facing a lot of challenges in their way to mature and standardize such as bandwidth

requirements, delays, jitter, echo, reliability and quality of service.

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CHAPTER 6

Conclusion and Comments

Chapter 6- Conclusion and comments _________________________________________________________________________________________________________________________________________

6.1 Conclusion The project achieves its final goal that is implementing and demonstrating the

Next Generation Network VoIP technology through building a reliable phone to

phone communication service. Although the quality of voice can be much better by

using a more advanced hardware, the achieved quality is satisfactory.

Moreover, one of the most important aims of this project is to reveal the

compelling feature of deploying Next Generation Network architecture. The

methodology followed was a proper choice of the scenarios desired. Also, a typical

model of VoIP service over the SIP protocol architecture has been properly

constructed and successfully verified using Java software packages and an appropriate

SIP protocol stack. Simulation of the implemented application in its different

anticipated scenarios has been executed accurately; this was done in the testing stage

where the quality of the voice service delivered was considered to be acceptable as

much as necessary. In other words, the system replied for all scenarios of the tests.

Also, the packages used are simple to set up and install and has attractive user

interfaces which are very easily familiarized to users without any previous knowledge

about the specific details of the model. It enables users to start up voice connection

from any location and at any time.

Also, we can conclude that, the traditionally PSTN is considered to be the best

provider for voice services, it presents a very high Quality of Services (QoSs) with

high costs and poor utilization of the bandwidth. In contract, the pure Voice over IP

(VoIP) networks which provide a patchy voice services by using packet voice. Their

quality of services (QoSs) is very low comparing with the PSTNs, also, they have a

very high utilization of the bandwidth with extremely low costs (almost for free). As

mentioned before, the NGN (Next Generation Network) is the architecture that

seamlessly blends the PSTN and the data network (IP network) creating a single

multi-service network. Assume that, we are going to provide voice services managed

by a voice service provider using an NGN platform. NGN with IP as its core bears all

the services offered by other networks (such as PSTNs), offloads large amount of data

transmission to the IP network to reduce the load over the network elements. NGN's

structural design allows flexible dimensioning of bandwidth, eliminating the need for

fixed size trunks groups for voice, thus making it easier to manage network structures.

The result of approval of NGN architecture is a limitless bandwidth (distance not a

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Chapter 6- Conclusion and comments _________________________________________________________________________________________________________________________________________

factor), cheap processing with everywhere simple always-on user interfaces,

guaranteed service quality and regulation in addition to reliability and security. With

NGN, not only the computer network, even the cable television network will be

converged into the IP network.

6.2 Recommendations In this report, there are some aspects that have not in any way proposed or studied

which remains to be solved. These are:

• Implementing SIP security aspects to assure security of the application

users. Another branch that is left open for any future candidates is a careful

consideration of the security aspects that could be adjusted to the

application, in order to add authentication and confidentiality attributes to

the package.

• Current proxy server has the ability of serving a limited number of

concurrent users (approximately 16). For commercial use it is better to use

a proxy with more capabilities (number of concurrent users, billing

system...etc.).

This project opens the door for new generation of applications which have

unlimited services and benefits…

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References _________________________________________________________________________________________________________________________________________

References 1. Ahmad S. Malik, Next Generation Network, ETISALAT

ACADEMY,

2. Next Generation Network, TEKOnsult (Telecom Engineering and

Consulting).

3. Akef J. Esmeirat, IP Networks, ETISALAT ACADEMY, October

2005.

4. M. Hardley, H. schulzrinne, E. Eschooler, J. Rosenberg, Session

Initiation Protocol, March 1999.

5. Fredrik Thernelius, SIP, NAT and Firewalls, may 2000.

6. www. Java .com, last visited March 2007.

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Appendix _________________________________________________________________________________________________________________________________________

Appendix

Program Code For The User Agent

import java.awt.*; import java.awt.event.*; import javax.swing.*; import javax.swing.plaf.ColorUIResource;

public class main extends JFrame { Button Start; Button Config; Button Exit; Toolkit tk = Toolkit.getDefaultToolkit(); Image image = tk.getImage("picture.jpg");

public main() { ColorUIResource textColor = new ColorUIResource(48, 63, 112);

ColorUIResource color = new ColorUIResource(242, 242, 242);

setResizable(false); setSize(400, 400); setDefaultCloseOperation(JFrame.EXIT_ON_CLOSE); setVisible(true);

mainLayout customLayout = new mainLayout();

getContentPane().setFont(new Font("Helvetica", Font.PLAIN, 18)); getContentPane().setLayout(customLayout); getContentPane().setBackground(color);

Start = new Button("Start"); Start.setBackground(color); Start.setForeground(textColor);

getContentPane().add(Start);

Config = new Button("Configration"); Config.setBackground(color); Config.setForeground(textColor); getContentPane().add(Config);

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Appendix _________________________________________________________________________________________________________________________________________

Exit = new Button("Exit"); Exit.setBackground(color); Exit.setForeground(textColor); getContentPane().add(Exit);

setSize(getPreferredSize());

addWindowListener(new WindowAdapter() { public void windowClosing(WindowEvent e) { System.exit(0); } });

//===========================================================

Start.addActionListener(new ActionListener() { public void actionPerformed(ActionEvent ee) {

hide(); dispose(); SipCommunicator sipCommunicator = new SipCommunicator(); sipCommunicator.launch();

}//end actionPerformed }//end ActionListener );//end addActionListener() Config.addActionListener(new ActionListener() { public void actionPerformed(ActionEvent ee) { PropertiesConfig window = new PropertiesConfig();

window.setTitle("Properties Configration"); window.pack(); window.show(); hide();

}//end actionPerformed }//end ActionListener );//end addActionListener()

Exit.addActionListener(new ActionListener() { public void actionPerformed(ActionEvent ee) { System.exit(0); }//end actionPerformed ___________________________________________________________________________________________________________________________________


Appendix _________________________________________________________________________________________________________________________________________

}//end ActionListener );//end addActionListener()

//===========================================================

} public void paint(Graphics g) { g.drawImage(image, 0, 10, this); }

public static void main(String args[]) {

main window = new main();

window.setTitle("main"); window.pack(); window.show(); } }

class mainLayout implements LayoutManager {

public mainLayout() { }

public void addLayoutComponent(String name, Component comp) { }

public void removeLayoutComponent(Component comp) { }

public Dimension preferredLayoutSize(Container parent) { Dimension dim = new Dimension(0, 0);

Insets insets = parent.getInsets(); dim.width = 575 + insets.left + insets.right; dim.height = 279 + insets.top + insets.bottom;

return dim; }

public Dimension minimumLayoutSize(Container parent) { ___________________________________________________________________________________________________________________________________


Appendix _________________________________________________________________________________________________________________________________________

Dimension dim = new Dimension(0, 0); return dim; }

public void layoutContainer(Container parent) { Insets insets = parent.getInsets();

Component c; c = parent.getComponent(0); if (c.isVisible()) {c.setBounds(insets.left+32,insets.top+120,168,40);} c = parent.getComponent(1); if (c.isVisible()) {c.setBounds(insets.left+368,insets.top+120,168,40);} c = parent.getComponent(2); if (c.isVisible()) {c.setBounds(insets.left+200,insets.top+120,168,40);} } }

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