Deployment of VoIP communications in B&A spy agency: Design and implementation

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International Journal of Computer Networks and Communications Security

VOL. 3, NO. 11, NOVEMBER 2015, 424–431

Available online at: www.ijcncs.org

E-ISSN 2308-9830 (Online) / ISSN 2410-0595 (Print)

Deployment of VoIP Communications in B&A Spy Agency:

Design and Implementation

Bilal Ahmad¹, Syed Asfandyar Gilani², Dr. Razi Iqbal³, Hasnain Sherazi⁴, Mr. Vahab Iranmanesh⁵

1, 2, 3, 4, 5

Department of Computer Science and IT, Lahore Leads University, Lahore, Pakistan.

E-mail: ¹[email protected], ²[email protected]om

ABSTRACT

VoIP, Voice over Internet Protocol is one of the emerging and bleeding edge technologies that have in most

cases sorted out the budget problems for the transmission and reception of voice communications. Indeed it

is to use the Data Network infrastructure (Packet Switching Network) for the transmission and reception of

voice calls. Though, people do prefer to use circuit switching network as it gives a dedicated path between

end users that results in a crystal clear voice quality. Also the Digital communications “Modulation for the

Transmitter end” and “Demodulation for the Receiving end” is involved (Digital Signal Processing to

provide phenomenal voice quality) but no one can regret a fact that is; occasionally it becomes too

expensive for the organization to provide dedicated paths to all employees. A promising solution is yes of

course VoIP but the deployment of this technology introduces issues like Voice Quality, Jitter/ Delay,

Quality of Service, Denial of Service etc. which is a challenge. In order to deploy VoIP communications in

an existing Network it is very crucial and important to do proper planning and to analyze the network

critically and justify whether it the Network is ready to cope with the VoIP services, as it can cause serious

effects to the Network in case it is over burden. After doing that to assess and then evaluate by testing the

network. This research is a case study based on B&A, A spy agency that has faced problems in the future

regarding to call management and expansion. This research will cover the design and implementation of

VoIP simulated in OPNET.

Keywords: VoIP, DoS, VoIP Communications, Qos, B&A Spy Agency.

Table 1: Users and the Applications Used

AN OVERVIEW OF B&A SPY AGENCY

Building

Total no. of

Applications

As proposed before that B&A is a spy agency. To

No.

floors

provide not just the voice services but with

reliability and security are the two main key aspects

of the medium enterprise organization. Reliability

means that whenever a user dials a number it don’t

wait for ages also a satisfaction from the employees

are required that they are getting clear voice quality

without any problems like listening back to their

own voices or delay etc. As B&A is a spy agency

so the share of information should be secure.

http, email, database,

printing

Same as above

including the CAD

applications

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B. Ahmad et. al / International Journal of Computer Networks and Communications Security, 3 (11), November 2015

As discussed before that this arbitrary

organization was on the way to expansion and it has

been expanded now. Presently there are a total

number of seven buildings. A total number of users

have increased to 1200 from 600 (Almost doubled).

The table below is to highlight what is included

now;



Total no of users in each building Less than or

equal to 50



Building number 1 to 4 is physically located

close to each other



Building 5 and 6 is located close to each other

Building 07 is on a separate location apart from

the above building number 1 to 6

Fig. 2. Core switch room at Building 07



Server room and core switch room are in

building 07 along with a dedicated server for

software applications

All the buildings to have their respective access

and distributed switches on their respective

floor 1.



The used topology is star

Fig. 3. Server room at Building 07

The floors are connected to access switch by

10 Base T link. Access switches are connected

to Distributed switches by 100Base T and then

to distributed switches by 1000 base X. Servers

are connected by 1000 Base X link

The users are almost doubled now causing

difficulty in managing the PBX (A Local-

Exchange). Though the infrastructure did provide

the network users of that time with a separate

cabling connection for the voice communications,

but now it is a very difficult and hectic thing to re

infrastructure [1].

As the employees, most of the time

communicates each other by doing emailing so, an

approximation to answer those questions are;



Total number of simultaneously calls during

the peak hours = (30 to 50)

Duration of each call during the peak hours =

(3 to 5 minutes)

Total number of users with dedicated numbers

= 750

Total number of simultaneously calls during

off peak hours = (30-50)

Duration of each call during off peak hours =

(15-20) minutes

Requirement to call outside the organization

(internationally) including peak and off peak

hours= (10-20)

Fig. 1. Building 1 to Building 04 on the same geographic

location



Duration of each call outside the organization

(internationally) including peak and off peak

hours = (30-45) minutes

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B. Ahmad et. al / International Journal of Computer Networks and Communications Security, 3 (11), November 2015

the question whether or not OPNET will give

me the desired results or not?



No of users allowed to call internationally =

200

5. What is the required Quality of Service

threshold? Provision of an assurance to meet it.

Is Quality of service provision is sufficient to

provide reliability and security or something

else has to be done? Identification of security

risks and the provision of an effective solution

to avoid the security risks and failure.

It was reported before that with the PBX it was

difficult to maintain the infrastructure as the

employees were facing a high waiting time during

the peak hours and they were facing the blocking

probability which was very high. The organization

was very concerned as the optimized bill was out of

the budget.

6. How can be the modification of the Network

done? What are the devices that the

organization needs to consider for example a

soft phone installed in the Computer with the

connected handset or has to be a telephonic

handset?

THE VOICE OVER IP

COMMUNICATIONS: AN

INTRODUCTION:

Voice over IP, VoIP is defined as to use the

Internet Protocol to transmit and receive voice

communications [8]. However, the main purpose of

the internet protocol is to transmit and receive data

over different applications like email, http, ftp,

www, database, printing and many more. In order

to implement VoIP communications on IP based

network.

WHAT IS SECURITY AND

RELIABILITY FOR B&A SPY AGENCY?

It is very critical to define a threshold and explain

what actually reliability and security means for the

organization. It is quite obvious that people do trust

public switched telephonic network because of the

dedicated switched network.

VoIP, Voice over Internet Protocol is one of the

most promising and bleeding edge technologies in

the field of modern Telecommunications and

Information and Technology industry [1]. No one

can regret that the deployment of VoIP reduces the

cost as compared to the old fashion circuit switch

telephony but the challenges are considerably

significant and it also introduces too many risks. As

compared to the trendy circuit switching network

the packet switching, VoIP does not show

immunity towards various security threats also

there are various challenges about the Voice

Quality, Security, Packet Loss, Jitter and so many.

The positive aspects of the deployment of VoIP

are however; it is less expensive, flexible, scalable,

reconfigurable, manageable etc. There are also so

many question marks when deploying the VoIP

Communications in an existing network.

At the time of designing the network

infrastructure the Engineers thought that circuit

switching telephony is the best solution for this

small enterprise organization and they gave

dedicated paths to all users. As far as security for

the same organization is concerned it is to provide a

guaranteed telephony service between two end

users without any problems technically speaking

delay, jitter, noise, echo etc. Also the waiting time

to establish the connection should be as little as it

can. In reality the users demands as soon as he/ she

dials up a number the connection gets established

without any waiting time.

Apart from the security perspective; the Network

Engineer perhaps didn’t realize at that time that the

same enterprise will be demanding an expansion

and then how they will manage and provide more

connections. Of course there is just a single PBX

for call routing and management and if every user

wants to make a call how many dedicated paths

they will provide.

In order to deploy VoIP communications the

following questions have to be considered;

1. What is the meaning of Security and Reliability

for B&A Spy Agency?

2. How can the network be assessed or what are

the assessment tools and the parameters that

need to be measure or calculated?

A calculation was made for the previous

telephonic network;

Availability of the medium = Failure Time /

Total Time

Unavailability = 1- Availability

3. How the signalling has to be done? What is the

VoIP architecture and is the Bandwidth

sufficient to support Voice calls?

4. How to simulate and test the network and why?

Which tool to use in this case OPNET but then

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B. Ahmad et. al / International Journal of Computer Networks and Communications Security, 3 (11), November 2015

After assuming the calculations the availability

calculated was 55% and 45% unavailability during

the busy hours which was a bit concerned.

The reason why the users were facing the system

unavailability was only the delay in the process as

there was not much link capacity before for the

PSTN. It was also causing delays as the dedicated

circuitry had to be generated for the calling end

parties.

3.3 Confidentiality and Integrity

Eaves dropping, one of the examples when a fake

user wants to listen. In order to stop or prevent

proper encryption can be used to make the data

preventable from being listened.

THE NETWORK ASSESSMENT TOOLS

As the voice communications has to be deployed

As now the organization has been expanded and

the geographical map does not allow to re

infrastructure the cabling so it is a huge problem

that has to be tackled. The prominent solution is yes

the Ethernet packed based network can be used for

voice communications. Deploying VoIP seems to

be a good choice as the network provider has

assured that the link and carrier will provide its best

service but it is again opening different obstacles as

it has to be made clear that how many errors the

system will introduce, if adding soft-phones what

will be the cost and error rate, what will be issues

originated by compatibility, is the network capable

of handling calls, how will it affect the band width

and system capacity.

“Security” for B&A means that the data cannot

be intercepted and securely communicated. The

engineers at that time planned as because of circuit

switching there will be dedicated path and the

connection will be physical so there will be no

security threats and the users will be fully satisfied.

Switching to VoIP communication there can be

three different types of security issues;

on a network that was originally designed for the

data communications and service like http, email,

www. , data base and file printing so it is very

critical and necessary to assess network whether it

will or it will not support the VoIP deployment.

As far as Data Network is concerned the

parameters are most likely bit error rate, round trip

delay, CPU utilization time, Servers down time and

load parameters, packet loss and delay etc. but for

the VoIP communication the parameters are of

course that the end users are able to listen and

speak without any disturbance produced by factors

like noise, jitter, delay and packet loss.

Wallingford (2005) discussed that the most

important parameters to assess the network are [7];

Bandwidth

Packet Loss

Delay/ Jitter

Network Utilization

In order to calculate the above factors which are

not the only once there are different tools that can

be able to monitor and give results. In this very case

the assessment tools that can be utilized are

“Observer” and “Wireshark”. Yes some online

bandwidth calculation tools can also be availed if

required. In order to simulate and test OPNET will

be used.

3.1 Denial of Service

A sort of attack that keeps the user away to use

the services. As the VoIP communications have to

be deployed on a packet based network so if the

scenario like this occurs will be

great

According to Wireshark; “It is a free and open

disappointment. One can consider many examples

source packet analysing tool. Before, in 2006 it was

when different sites based on internet get attacked. known as ethereal. The plus point is it is free and

An example of Denial of service can be when a user

is using the MAC address of the router and sending

de authenticated messages to the network. In this

scenario all the users will be unavailable to

authenticate with the network. Probably it may

the network can be assessed and monitored”.

According to Observer “It is a complete solution

not only for monitoring the network problems but

also the provision of back time analyses is assured.

Other prominent features are reporting, monitoring

occur in the wireless networks but as far as B&A for the routes, alarms, application tools, trouble

spy agency is concerned none of the out siding

users can use the network or login to it.

shooting and many more”.

SIMULATION OF THE NETWORK

USING OPNET

3.2 Service Stealing

This relates to if someone is making use of the

services that he/ she is not paying for. As far as

B&A Spy agency is concerned all employees have

to follow the strict rules and regulations.

In order to simulate the network availability and

reliability different statistics are simulated by using

OPNET to come with a concrete solution. The

reason why to use OPNET is because of the

software packed with features. Voice over IP

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B. Ahmad et. al / International Journal of Computer Networks and Communications Security, 3 (11), November 2015

communications can be deployed. Because the

Ethernet Packet based network has to be used for

the voice over IP communications so it is of great

interest that for both the data and voice the

calculations has to be made. What if the voice is

deployed on a data network and it is not providing

the best service for data. Different statistics can be

obtained to discuss the factors that cause severe

problems in a network. One of the statistics is given

below; rest of them will be discussed later in the

report.

Fig. 5. Performance Analysis for the G. 711 PCM quality

by Hens and caballero (2008)[4]

Another challenge is that as compared to the old

fashion circuit switched telephone network the

delay and latency is significantly high and there are

a number of reasons including no dedicated circuit

and increase number of equipment and communica-

tion.

Fig. 4. Simulation result for the Email

Application Traffic sent and receive

Figure above shows the simulation result for the

email traffic. After simulating the Network with the

VoIP deployment the simulation justified that as

much traffic was send was received with no delay

or error. That reflects that the network is capable of

handling traffic both data/ voice efficiently. What

the simulation justifies is that the network is

capable to handle load and there is not much link

failure issues and downtime report.

It is also important to analyze the voice results

because as a user we only can say about the voice

quality, availability etc. but has no command what

actually is running behind in the background as it is

hidden from the user.

Fig. 6. Different equipment that causes delay in VoIP by

Hens and Caballero (2008) [4]

Different types of delays explained by the same

authors in the same literature explained above are

Coding/ Decoding Delay, Packet Assembly Delay,

Serialization Delay, Queuing Delay, De-jittering

Delay,

As the codec used for this deployment is G.711,

it is already explained that it generates 8 bits after

125 microseconds. As compared to other coding

schemes the PACKETIZATION Delay for G.711 is

less and the only reason is the effective and

efficient compression takes place is less time.

After simulating the VoIP deployment in OPNET

the following graph was obtained;

DELAY

If defined for the data communication, delay

referred to the time for a successful communication

that take place between the transmitter and the

receiver. It is also referred to latency. If defined for

the voice communication the concept remains the

same the only different is that the time has been

meant for the voice communication between the

transmitter and receiver. Hens and caballero (2008)

explained that because of the one way end to end

delay increases in the voice communication/

telephonic conversation routing becomes more and

more complex [4]. Hence it is very difficult to

tackle with this characteristic.

Fig. 7. Delay for the overall network simulation graph

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B. Ahmad et. al / International Journal of Computer Networks and Communications Security, 3 (11), November 2015

Ethernet delay after the deployment shows that

MOS is an abbreviation to Mean Opinion Score.

the total delay calculated is less than 0.45ms that is To calculate MOS, Hersent, Petit and Gurle (2005)

considerably acceptable as compare to 150ms explained that for the telephone coders that do

which is too much. The most important statistic to operate between the frequency ranges of 4kbp to

obtain here is MOS.

32kbps, the most commonly used methodology is

ACR, Absolute category rating [5]. In order to

achieve MOS value this test is being done. It is also

explained in the same literature that it is not the

only way to assess the quality of voice. Other

methods are DCR, Degradation category rating and

Comparison category rating, CCR. Wallingford

(2005) explained that it is not entirely necessary to

determine the MOS value but it is a positive aspect

to consider if the number of users is satisfactory

[7].

The reason why MOS calculation is used in this

project is because the project is simulated in

OPNET and it can produce MOS graphs. In order

to justify the speech quality is according to

standards it is a good way to compare and justify.

The authors in the same literature also explained

that this calculation is based on the call quality

between two effective call parties. The rating is

made on the basis of call qualities. Though the

value is absolute but there has to be some sort of

reference nodes available in order to justify the

MOS value/ score. To do that, some sort of radio

references are inserted. Also Raake (2006)

explained and defined MOS, Mean Opinion Score.

It is indeed a standard specified by ITU-T.

Fig. 8. Jitter value simulation graph for VoIP

communications

After simulating the desired network evaluation

for the VoIP configurations, the most promising

result obtained was of course jitter. It seems that the

jitter is signficantly low in the beginning and it

almost reaches to a threshold of 0 which is ideal. Of

course it is not true as for the voice

communications using the network it is true that

one has to face jitter values but might be because of

the provision of relative nodes for the backbone and

efficient transmission media the results are very

satisfactory. Also real time traffic will definitely

introuduce jitter but the positive thing is that the

network is capable to introduce a less jitter call

quality.

MOS

One of the most serious issues in any data

network is that it uses UDP, User Data Gram

Protocol which is a connection less oriented

protocol. Though Voice over IP, VoIP

communications uses RTP, Real time transmission

protocol but it is connection less. It assures its best

effort but there is no guarantee as compared to

TCP, Transmission control protocol that provides

reliability and security in terms of effectively

handling the data. In case if the data loss occurs that

will have an adverse effect on the voice

communications. If an email is considered it would

not matter if it reaches with a little delay still

understandable but in case of voice if a packet is

loss means distorted voice quality.

Fig. 9. MOS, as a function of BR, Bit rate and the coder

used, by Hersent, Petit and Gurle (2005) [5]

Table 2: ITU Coders MOS recommended values [2]

In order to measure data loss it has to be

calculated when two parties are busy in voice

communication. For that MOS score is calculated.

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B. Ahmad et. al / International Journal of Computer Networks and Communications Security, 3 (11), November 2015

Another challenge is the number of users is not

finite. Sometimes the network will be congested but

sometimes a very few users will be accessing the

network. Again a challenge!

There can be different impairments present in the

network. Generalizing a network it comprises of:

Fig. 10. MOS value for the VoIP deployment [2]

Figure shows that the MOS value after simulation

is approximately 3.7 which are satisfied according

to the literature.

HOW MORE SERVICES CAN BE

PROVIDED TO THE NETWORK: A

CONCLUSION?

Fig. 11. Layered approach for the Network

The subscribers do use different types of devices

so the problems that can be caused from the

subscriber side using the equipment can be;

Though discussed earlier about the QoS, Quality

of service provision but the question here is

whether assurance to provide quality of service will

guarantee the best voice communication experience

with reliability and security? The answer is might

be yes but in most cases no. we have to think

beyond. On a packet based network the main

targets are;

1. An introduction to the acoustic echo because of

the improper isolation of the headphones and

speakers could destruct the voice.

2. Different end users do use different equipment.

Might be one is using the VoIP phone set or

software to access a subscriber who is

connected to the PSTN so different audio

levels and different CODECS selection for two

different users on a same call.

3. As the coaxial cable is being used so to convert

two to four wires might be a mismatch in

impedance, that will result in an echo termed as

hybrid.

1. Monitoring the Network (Identification)

2. Controlling (Precautionary measurements)

3. Management (Trouble shooting and

allocation of resources)

By doing so one will be able to minimize the

packet loss and delay but it cannot be reached to

zero. So still quality of service I must say that do

not provide a complete solution. Something has to

be done beyond the expectations. As far as Ethernet

is concerned most of the time the dominant quality

of service used is class of service represented by

CoS. The argument to justify is whether prioritizing

is sufficient. Durkin and James (2003) explained

that although MPLS is a prominent solution as it

assigns a virtual label of 32bit long header and the

promising features are segmentation, prioritization

and expedition of the traffic yet again so many

other measurements have to be done like [3];

Different techniques can be used like voice signal

processing where different CODECS can be tuned

according to the requirements [7][8]. Also there are

some echo cancellation devices and adoptive noise

filters that can stop the echo problems. One can

also use matched filters and calculation on SNR

and BER as different modulation schemes bring

their own BER. Overall it seems a promising

solution to provide B&A Spy Agency with the

VoIP technology.

The far most challenge in packet based network

is to minimize the packet loss but in order to utilize

the bandwidth effectively and efficiently one has to

use the selection of appropriate codec and in this

case it is G.711. Now in order to get higher value of

MOS one has to use coding that introduce low bit

error rate but again one has to tolerate the packet

loss in order to utilize the bandwidth.

REFERENCES

[1] Allan,

(2007) PBX Systems for IP

Migrating enterprise

Telephony:

communications. McGraw Hill Telecom

professional.

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B. Ahmad et. al / International Journal of Computer Networks and Communications Security, 3 (11), November 2015

[2] Davidson, J., Peters, J., Bhatia, M,. Kalidindi,

S,. Mukherjee, S. (2006) VoIP Fundamentals

(2nd edition). Cisco Press.

[3] Durkin, J. F. (2003) Voice enabling the data

network: H.323, MGCP, SIP, QoS, SLA’s and

Security. Cisco Press, USA.

[4] Hens, F J., caballero, M (2008) Triple Play

Building the Converged Network for IP, VoIP

and IPTV. John Wiley and Sons, West Sussex,

UK.

[5] Hersent, O, Petit J-P., Gurle D. (2005) Beyond

VoIP

Protocols

Understanding

Voice

Technology and Networking Techniques for IP

telephony. Wiley and Sons, West Sussex, UK.

[6] Raake A (2006) Speech Quality of VoIP

Assessment and Prediction. John Wiley and

Sons, West Sussex, UK.

[7] Wallingford, T (2005) Switching to VoIP. O

Reilly media, USA.

[8] Wittenberg (2009) (3rd Ed) Understanding

VoIP Technologies. NY, USA.

[9] Bates, J., Gallon, C., Bocci, M., Walker, S.,

Taylor,

T (2006) Converged Multimedia

Networks. John Wiley and Sons, UK.

[10]Nortel (2008) VoIP Technologies:

comprehensive Guide to Voice over Internet

Protocol (VoIP). Nortel Press, USA.

[11]Ransome. J. F., Rittinghouse, R. (2005) VoIP

Security. Oxford, UK.

[12]Verma, P K., Wang, L (2001) Voice over IP

Networks: Quality of service, Pricing and

Security. Berlin, Deidelberg.

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