Heaven’s Light is Our Guide

Department of Electronics & Telecommunication Engineering

Rajshahi University of Engineering & Technology

Laboratory Report
on
ETE 4216: Sessional Based on Telecommunication Engineering

Submitted by:
Student Name
Roll No. 1704XX

Submitted to:
Md. Tarek Hassan
Lecturer
Dept. of ETE, RUET

Month, Year
Contents

List of Figures ii

List of Tables iii

Experiment 1: Design and Simulation of an Inter–Campus Telecommunication

Network Using Riverbed Modeler Software. 1

Experiment 2: MOS Value Analysis of a Voice Over IP Networks (VoIP) Using

Riverbed Modeler Software. 1

Appendices 6

Experiment 3: Appendix 7

i
List of Figures

Experiment 1: 1
1.1 A Basic Telecommunication Networks. 2
1.2 10BaseT LAN based Telecommunication Network. 2
1.3 100BaseT LAN based Telecommunication Network. 3
1.4 1000BaseT LAN based Telecommunication Network. 3
1.5 Email Profile Results in 10BaseT LAN. 4
1.6 Voice Profile Results in 10BaseT LAN. 4
1.7 Video Profile Results in 10BaseT LAN. 5
1.8 Email Profile Results in 100BaseT LAN. 5
1.9 Voice Profile Results in 100BaseT LAN. 6
1.10 Vide0 Profile Results in 100BaseT LAN. 6
1.11 Email Profile Results in 1000BaseT LAN. 7
1.12 Voice Profile Results in 1000BaseT LAN. 7
1.13 Vide0 Profile Results in 1000BaseT LAN. 8
1.14 Email Traffic Results in LANs. 9
1.15 Voice Traffic Results in LANs. 9
1.16 Video Traffic Results in LANs. 10

2.1 Whole VoIP Network. 2

2.2 Voice Traffic Results when packet discard ratio is (a)0.5 percent; (b)one percent;
(c)Five percent. 3
2.3 Voice MOS value, end to end delay, traffic sent and traffic received profile when
packet discard ratio is (a)0.5 percent; (b)one percent; (c)Five percent. 4

ii
List of Tables

Experiment 1: 1

2.1 Mean Opinion Score (MOS) 3

iii
 Heaven’s Light is Our Guide

Rajshahi University of Engineering & Technology

Department of Electronics & Telecommunication Engineering

ETE 4216: Sessional Based on Telecommunication Engineering

Experiment 1
Design and Simulation of an Inter–Campus Telecommunication Network
Using Riverbed Modeler Software

Submitted by: Submitted to:

Ruksana Akter Md. Tarek Hassan
Roll: 1704002 Lecturer
Session: 2017-18 Dept. of ETE, RUET

Date of Experiment:14/03/2023
Date of Submission: 2/05/2023

Report (Teacher’s Section) Viva

□ Excellent □ Excellent
□ Very Good □ Very Good
□ Good ——————————– □ Good
□ Moderate Signature □ Moderate
□ Poor □ Poor
Experiment 1
Design and Simulation of an Inter–Campus Telecommunication Network
Using Riverbed Modeler Software.

Objectives

The main objectives of this experiment are:

• To design an inter-campus telecommunication network
• To understand the basic operation of a riverbed modeler while designing a scenario
• To observe the results from traffic for Email, Voice, and Video.
• To analyze the traffic for using different LANs e.g., 10BaseT, 100BaseT, and 1000BaseT

1.1 Theory

A broadcast communications arrange may be a gather of hubs interconnected by broadcast

communications joins that are utilized to trade messages between the hubs. The joins may
utilize a assortment of innovations based on the strategies of circuit exchanging, message ex-
changing, or parcel exchanging, to pass messages and signals. Numerous hubs may participate
to pass the message from an starting hub to the goal hub, through numerous arrange jumps.
For this directing work, each hub within the arrange is relegated a arrange address for distin-
guishing proof and finding it on the organize. The collection of addresses within the organize
is called the address space of the arrange.
Cases of broadcast communications systems incorporate computer systems, the Web, the
open exchanged phone arrange (PSTN), the worldwide Message organize, the aeronautical
ACARS network,[1] and the remote radio systems of cell phone media transmission suppliers.
Transmission of Data in Telecommunication Networks: Telecommunication networks for
transmitting data formerly used numerous different protocols. A basic telecommunication net-
work is represented in fig 1.1:

1
 Figure 1.1: A Basic Telecommunication Networks.

1.2 Required Apparatus/Softwares

1. A Highly Configured PC

2. Riverbed Modeler.

1.3 Block Diagram/Network Scenarios

In this experiment, the whole process can be explored through a block diagram/network sce-
nario in parts shown in Fig. 1.2

Figure 1.2: 10BaseT LAN based Telecommunication Network.

2
 Figure 1.3: 100BaseT LAN based Telecommunication Network.

Figure 1.4: 1000BaseT LAN based Telecommunication Network.

1.4 Result Analysis & Discussion

1.4.1 Results for 10BaseT LAN Connection:

In this section email, voice and video profile results are represented. In figure 1.5 trafice sent
and trafic received in packets per second is mentioned. 10BASE-T is an Ethernet standard for
local area networks. 10BASE-T supports 10 megabits per second (Mbps) transmission speed
over twisted-pair cabling with a maximum length of 100 meters (m).

3
Figure 1.5: Email Profile Results in 10BaseT LAN.

Figure 1.6: Voice Profile Results in 10BaseT LAN.

4
 Figure 1.7: Video Profile Results in 10BaseT LAN.

1.4.2 Results for 100BaseT LAN Connection:

Figure 1.8: Email Profile Results in 100BaseT LAN.

In this section email, voice and video profile results are represented. In figure 1.5 trafice
sent and trafic received in packets per second is mentioned for 100baseT LAN. 100baseT means
it supports 100Mbps data transmission. For voice profile jitter(sec), traffic received(packets/sec)

5
Figure 1.9: Voice Profile Results in 100BaseT LAN.

Figure 1.10: Vide0 Profile Results in 100BaseT LAN.

6
and traffic sent(packets/sec) are shown. And also shown the traffic received and sent for video
profile.

1.4.3 Results for 1000BaseT LAN Connection:

Figure 1.11: Email Profile Results in 1000BaseT LAN.

Figure 1.12: Voice Profile Results in 1000BaseT LAN.

7
 Figure 1.13: Vide0 Profile Results in 1000BaseT LAN.

In this section email, voice and video profile results are represented. In figure 1.5 trafice
sent and trafic received in packets per second is mentioned for 1000baseT LAN. 100baseT
means it supports 1000Mbps data transmission. For voice profile jitter(sec), traffic received(packets/sec)
and traffic sent(packets/sec) are shown. And also shown the traffic received and sent for video
profile.

1.4.4 Results for Comparing 10BaseT, 100BaseT, and 1000BaseT LAN Connection:

Fig. 1.14 shows that there is some differences in traffic sent and received among 10BaseT,
100BaseT 1000BaseT LAN connections, but fluctuation was observed in received traffic. In
some portions, 1000BaseT is performing well.The 100BaseT is performing at best comparing
with 10baseT and 1000baseT.

8
 Figure 1.14: Email Traffic Results in LANs.

Figure 1.15: Voice Traffic Results in LANs.

Fig. 1.15 shows that there is a significant difference in traffic sent and received and Jitter
among 1000BaseT, 100BaseT 1000BaseT LAN connections for voice traffic profile, but fluc-

9
tuation was observed in received traffic and traffic sent. For jitter there is no fluctuation was
observed. Here for voice traffic sent 1000baseT performed better. Like this in some cases
100baseT performs better, 10baseT also perfoms good for some cases.

Figure 1.16: Video Traffic Results in LANs.

Fig. 1.16 shows results of traffic sent and received among 1000BaseT, 100BaseT 1000BaseT
LAN connections video profiles. With video communication, traffic is more difficult to define.
It may consists of a still picture or image or slow scan motion pictures. Conference televi-
sion also has a number of variants. Here for video conferencing traffic received 10baseT LAN
perfomrs better.

1.5 Conclusion

In this experiment, we performed the simulation for analyzing an inter-campus network for
email, voice and video traffic at a time using three different LANs. These LANs are 10baseT,
100baseT and 1000baseT LAN. And also the video, voice and emal profiles for these LANs
were compared.

10
 Heaven’s Light is Our Guide

Rajshahi University of Engineering & Technology

Department of Electronics & Telecommunication Engineering

ETE 4116: Sessional Based on Telecommunication Engineering.

Experiment 2
MOS Value Analysis of a Voice Over IP Networks (VoIP) Using Riverbed
Modeler Software.

Submitted by: Submitted to:

Ruksana Akter Md. Tarek Hassan
Roll: 1704002 Lecturer
Session: 2017-18 Dept. of ETE, RUET

Date of Experiment : 02/05/2023

Date of Submission : 23/05/2023

Report (Teacher’s Section) Viva

□ Excellent □ Excellent
□ Very Good □ Very Good
□ Good ——————————– □ Good
□ Moderate Signature □ Moderate
□ Poor □ Poor
Experiment 2
MOS Value Analysis of a Voice Over IP Networks (VoIP) Using Riverbed
Modeler Software.
2.1 Objectives

The main objectives of this experiment are

• To implement a long-distance VoIP network

• To observe MOS value of VoIP services.

2.2 Theory

The Mean Opinion Score, the nifty little metric that can help us to measure and improve the
quality of our VoIP calls. VoIP Quality is exceedingly dependent on organize execution, which
suggests that numerous arrange issues like parcel misfortune, idleness, and jitter can cause tall
levels of VoIP corruption. To dodge humiliating choppy voice calls, or lagginess amid your
following client assembly, we’re running you through how to degree VoIP Quality with MOS
Score (Cruel Conclusion Score). VoIP Quality alludes to the by and large level of sound and
visual execution experienced amid a Voice over Web Convention (VoIP) call. This incorporates
variables such as call clarity, flag quality, delay, resound, and other sound and visual twists that
can affect the client involvement. VoIP Quality can be affected by a assortment of components,
counting the quality of the web association, the sort of gadget utilized for the call, and the or-
ganize framework supporting the call. Measuring and checking VoIP Quality is imperative to
guarantee that clients are able to communicate successfully and productively, without encoun-
tering baffling sound or visual interferences.
The Mean Opinion Score (MOS) is a subjective metric that is obtained by having a group
of people rate the quality of audio samples. MOS scores are widely used in the telecommu-
nications industry to assess the quality of VoIP calls. We’ll expand on MOS Score in the
next section. The Mean Opinion Score, usually referred to as MOS Score, has been the most
commonly-used metric to measure the overall voice call quality for decades. Standardized by
the International Telecommunications Union (ITU-T), MOS Score refers to:

1
 A numerical measure of the human-judged overall quality of voice and video sessions.

2.3 Required Apparatus

1. Laptop or Desktop PC.

2. Riverbed Modeler.

3. Power Cables.

2.4 VoIP Network

Figure 2.1: Whole VoIP Network.

Voice over Internet Protocol (VoIP), is a technology that allows us to make voice calls
using a broadband Internet connection instead of a regular (or analog) phone line. Some VoIP
services may only allow us to call other people using the same service, but others may allow
us to call anyone who has a telephone number - including local, long distance, mobile, and
international numbers. Also, while some VoIP services only work over our computer or a
special VoIP phone, other services allow us to use a traditional phone connected to a VoIP
adapter.

2
2.5 Data Table

Table 2.1: Mean Opinion Score (MOS)

MOS Quality Impairment

5 Excellent Imperceptible
4 Good Perceptible but not annoying
3 Fair Perceptible but not annoying
2 Poor Annoying
1 Bad Very Annoying

2.6 Result and discussion

2.6.1 Result Analysis

(a) Voice Traffic Results when packet dis- (b) Voice Traffic Results when packet dis-
card ratio is 0.5 percent card ratio is one percent

(c) Voice Traffic Results when packet dis-

card ratio is five percent

Figure 2.2: Voice Traffic Results when packet discard ratio is (a)0.5 percent; (b)one percent;
(c)Five percent.

Voice quality on calls is highly subjective, there exist various ways to asses the score; a
common and a more practical option being algorithms that are used to predict MOS scores.Figure
2.2 represents the voice MOS, voice packet end to end delay(sec), voice traffic received(bytes
per sec) and voice traffic sent(bytes per sec) for packet discard ratios of 0.5,1 and 5 percents
respecivetly. A packet discard happens when a received packet has a transmission or format

3
 error, or when the receiving device doesn’t have enough storage room for it. While some
discarding is inevitable, excessive discards can point to several problems including: A network
device is misconfigured.

2.6.2 Comparison

(a) Voice MOS profile for packet discard (b) Voice packet end to end delay profile
ratios of 0.5, 1 and 5 for packet discard ratios of 0.5, 1 and 5

(c) Voice traffic sent profile for packet (d) Voice traffic received profile for
discard ratios of 0.5, 1 and 5 packet discard ratios of 0.5, 1 and 5

Figure 2.3: Voice MOS value, end to end delay, traffic sent and traffic received profile when
packet discard ratio is (a)0.5 percent; (b)one percent; (c)Five percent.

MoS (Mean Opinion Score) is a subjective measure of the quality of a voice call that is
rated by users on a scale of 1 to 5. The MoS score reflects the perceived quality of the call,
based on various factors such as signal strength, voice clarity, delay, and distortion. If the IP
cloud discards more packets, it can lead to a decrease in signal strength, voice clarity, and an
increase in delay and distortion. These factors can negatively impact the user’s perception of
call quality, resulting in a lower MoS score. In simple terms, if more packets are discarded
by the IP cloud, the call quality will deteriorate, and the MoS score will decrease accordingly.
Figure 2.3 represents the comparison between voice mos values, voice packet end to end delay,
voice traffic sent and voice traffic received for different packet discard ratios. As we know that

4
if the IP cloud discards more packets, it can lead to a decrease in signal strength, voice clarity,
and an increase in delay and distortion so for 0.5 voice profilr shows good performance.

2.7 Conlcusion

There are several ways to improve the MoS (Mean Opinion Score) value for VoIP (Voice over
Internet Protocol) calls as follows: QoS (Quality of Service): Implement QoS to prioritize
voice traffic over data traffic and ensure that voice packets are given priority over other types
of data packets. Bandwidth: Ensure that the available bandwidth is sufficient for the number
of concurrent VoIP calls. Typically, at least 100 kbps per call is required for high-quality
VoIP.Jitter buffering: Implement jitter buffering to reduce the impact of delay and packet loss
on call quality.Codec selection: Choose a codec that provides good quality at low bit rates. The
G.729 codec is a good option for VoIP as it provides good quality at low bit rates etc.

5
Appendix

6
Experiment 3
Appendix