a method for implementing, simulating and analyzing a voice over

A Method for Implementing, Simulating and

Analyzing a Voice over Internet Protocol Network

Bianca Enache

Communication Department

“Politehnica” University of Timisoara

Timisoara, Romania

[email protected]

Irina Giea

Global Service Delivery Department

Alcatel-Lucent

Timisoara, Romania

[email protected]

Abstract—In the telecommunication domain, the simulation

methodology is very important for the researchers because there

are many simulation software. This paper presents a method of

implementing a Voice over IP network in two steps. The first step

is creating a real network, then analyze the call sessions. The

second step is simulating the network and analyzing the traffic.

Keywords—VoIP network, network simulator, call sessions,

traffic flow.

I. Introduction

More and more people need to communicate. Through this direction were implemented many technologies. The Voice over IP (VoIP) is one of them. It allows the user to make/receive a call from/to its telephone or personal computer (PC) using the IP cloud. VoIP comes as an answer to the call users who want to benefit of the same security, speed and quality of service as the network users already have [8-10].

Specific protocols were designed for this new type of network, like Session Initiation Protocol (SIP). Implementing, simulating and analyzing a VoIP network can be done in different ways and using various tools and simulators. In he next chapters, we will present one of this methods[1].

II. tools and simulators

To develop our VoIP network, we used the following tools:

X-LITE, PBX Manager, Asterisk, Hammer Call Analyzer,

GNS3 and Wireshark [6-7].

For the first step, the tools (X-LITE, PBX Manager,

Asterisk and Hammer Call Analyzer) are described below.

X-LITE is a software that enables a PC to receive or

execute telephone calls, calls that can originate from another

PC or from an IP telephone. The type of calls are audio, video

or both (conferences). You can also send messages towards

different devices that supports this service. In order to be able

to use this software, the PC has to accomplish several

minimum performance conditions:

• Processor: Intel Core Duo (or its equivalent), Video

Card with support for DirectX 9.0c

• Memory: 2 GB RAM

• Space on Hard Disk: 50 MB

• Operating system: Microsoft XP Service Pack2, Vista

(32 or 64 bits), 7 or MAC OS 10.5

• Connection: connection to a network (broadband,

LAN, wireless), permanent connection to Internet

• Sound adaptor: Full-duplex, 16 bit or using a USB

Headset

PBX Manager is a software that depends on Asterisk. It

is a graphical management interface which allows the

configuration and management of a PC in order for it to act

like a conventional PBX (Private Branch Exchange). This

software allows the construction of a VOIP PBX with the

integration of capabilities and characteristics which can't be

found in the PBX conventional systems.

Asterisk is an open source program that transforms a PC

into a communication server. It is used with PBX systems

based on IP, VOIP gateways, conference servers and others.

The most important files are extensions.conf and sip.conf.

Extensions.conf defines the PBX calling plan for each user. In

the sip.conf file we can configure everything that is related

with the SIP protocol, like the creation of new users or the

definition of the SIP suppliers.

Hammer Call Analyzer can discover, isolate and fix

signaling or transmission problems. With its help you can

visualize what protocols were used, you can correlate call

sessions over multiple protocols or domains and you can also

visualize and analyze call session flows.

The simulation of the network, for the second step, was

possible using the GNS3 and Wireshark.

GNS3 is a graphical network simulator which allows the

simulation of complex networks. In order for it to function, it

is dependent on three other programs that must run

simultaneous: Dynamips (the core for GNS3 that emulates

IOS CISCO images), Dynagen (text-based software that is

necessary to Dynamips ) and Qemu (open source emulator and

virtualization tool).

Wireshark is the most common tool for analyzing the

network protocols, fixing the network problems, developing

software products and communication protocols. Wireshark

captures the traffic of packets in real time but the analyze can

be made offline. This paper is supported by the Human Resources Development Programme POSDRU/159/1.5/S/137516 financed by the European Social Fund and by the Romanian Government.

6TH INTERNATIONAL CONFERENCE ON MODERN POWER SYSTEMS MPS2015, 18-21 MAY 2015, CLUJ-NAPOCA, ROMANIA

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III. CREATING A REAL VOIP

NETWORK

In order to create the network, we connected three PCs with

a router [2-3]. Each PC is behaving like a telephone, meaning

that it can receive or make calls. connecting the PC as a

telephone was possible using three software: X-LITE, PBX

Manager and Asterisk .

For the identification of each PC, we attached an ID

number that can be used as a telephone number also. One PC

will act as a telephone and also as a PBX station. Because the

network is connected to the Internet, each PC will have the IP

address of the PBX station. This must not be confused with its

own IP address [Fig.1].

The PC that runs as both a telephone and a station is PC3,

with the calling number 3002. Each PC has the IP address in

the range of 93.115.167.x.

In order to study this network from the point of view of the

protocols involved, we captured the packets using Hammer

Call Analyzer software. In the case of our network, only one PC has this program

installed (PC1), which is why the sessions that will be studied, will be initiated only by PC1.

Fig.1 The configuration of the real VOIP network

For the capturing of the packets we chose three situations. In the first case, PC1 calls PC2 but after a few seconds PC2 is put on hold for 10 sec. during which PC1 calls PC3, a session that also lasts for 10sec. PC1 ends the session with PC3 because it's not answering. PC1 returns to the session created with PC2 and also closes shortly (Fig.2).

Fig.2 Case 1 - capture window

There are four windows in the capture program Hammer

Call Analyzer as you can see above (from left to right, up-

down): list of captured packets, call flow with the list of calls

and statistics, a detailed list of the protocols used in the

sessions, protocol list in hexadecimal format.

In the second case, PC1 calls PC3. The caller is announced

that PC3 is already in a conversation and receives two choices:

it can leave a message which PC3 will receive after it closes

the call or it can terminate the session.

We can visualize all the steps of a call session in the call-

flow window. This window presents each command the SIP

protocols requires in order to establish a correct call session.

In the figure below are shown several statistics for the packets

(Ethernet, IP, UDP-User Datagram Protocol) and other

information (Fig.3).

Fig.3 Protocol statistics

In the last case, PC1 calls PC2. PC2 answers and talks for

10sec. PC2 then puts on hold PC1 and calls PC3. PC3 answers

and talks for 10sec. before closes. PC2 returns to the call with

PC1 and closes after a short time. Because the calls go through

the PBX station PC3, the IP addresses for the called PCs will

not be showed (Fig.4).

Fig.4 Case 3 - call flow


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IV. SIMULATING THE VOIP

NETWORK

For the simulation we used a router from the 1700 platform

and three PCs. For the router to be accepted by the program

IOS images had to be loaded as well as configuring each PC to

accept connections towards the router. GNS3 has five

windows were you can see the devices available to create a

network, the network implementation area, a console that can

be used as a telnet, a summary of the network topology and a

window with the packet captures that are running (Fig.5).

Fig. 5 The network topology

Both router and the PCs have special menus where you can

modify certain information.

Once the network is created and works, you can proceed to

analyze it with Wireshark. To start the capture of packets, you

need to right-click on any connection and select "Start

Wireshark". All of the captures are for the traffic between the

router and PC2 (Fig.6).

Fig.6 Packet capture between router and PC2

The most revealing actions that can be taken to analyze a

network are the ones from the menus: capture, analyze

statistics and telephony (when using the virtual models for IP

telephones).

The capture of packets depends on different options. You

can choose the interface you are using for the capture, you can

select the filter meaning that only the packets that are passing

through the filer will be captured (Fig. 7).

Fig.7 Capture menu

In the Analyze menu you can select and visualize the

filters, you can see the TCP (Transmission Control Protocol),

UDP or SSL (Secure Socket Layer) flows and you can

validate protocols.

One of the most important menus is the Statistics menu.

The first option you can select presents a summary of the

capture that is running. There you can find information about

the topology's name, the encapsulation that is used or the size

limit of the packets. Also, you can see when the first and the

last packet arrived and the duration between these packets.

The summary also presents information about the traffic

(Fig.8).

Fig.8 Statistics menu - Flow graph

To see the protocol hierarchy used by this network, the

conversations that were captured, the input-output graphic of

the data and the traffic analysis, we used the interface between

the router and PC2 (Fig. 9).


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Fig.9 Protocol hierarchy statistics

The conversations between these devices are of two types:

Ethernet and UDP [4-5]. Both present the number of sessions

captured and the number of packets, their size and the

direction from which they were captured.

Fig. 10 Ethernet sessions

The difference is that the UDP session also presents the

port number of the device involved and the IP addresses of the

devices (Fig. 11).

Fig. 11 UDP sessions

The window opened to visualize the input-output graphic

of the traffic has options for choosing the filter. You can also

modify the axes of the graphic. The axe X is the time axe and

has a range between a few milliseconds and a few minutes.

The Y axe can linear, logarithmic or automatic. You can also

select on X axe the packets by their hour, minute or second

when their appeared (Fig. 12).

Fig. 12 Input-Output traffic graphic

The traffic analyze depends on the selected options from

the Statistics menu-flow graph (Fig. 8). Once they are selected

the window from figure Fig.13 will appear. There we can

observe each message that was received by the two devices.

For this analysis we chose the TCP flows with the standard

addresses for the source and destination. Each message

received is detailed in the right side of the window (Fig.13).

Fig. 13 Traffic analysis

V. CONCLUSIONS

Implementing the network using this method reflects many advantages. The tools and simulators are both used by the researchers and in the engineering field. The analyzers work in real-time but the troubleshooting can be done in offline mode. The capture of packets offers an accuracy of milliseconds for each packet. Using this network model, the same principle can be applied to various network sizes from a company’s intranet to a city’s Metropolitan Access Network (MAN).

Acknowledgment

This paper is supported by the Human Resources Development Programme POSDRU/159/1.5/S/137516 financed by the European Social Fund and by the Romanian Government.

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