See discussions, stats, and author profiles for this publication at: https://www.researchgate.

net/publication/363566693

Voice Over Internet Protocol (VOIP): Case Study Analysis

Technical Report · September 2022

DOI: 10.13140/RG.2.2.16127.28324

CITATIONS READS

0 1,237

1 author:

Chamoth Madushan Jayasekara

University of Plymouth
35 PUBLICATIONS 7 CITATIONS

SEE PROFILE

All content following this page was uploaded by Chamoth Madushan Jayasekara on 15 September 2022.

The user has requested enhancement of the downloaded file.

Voice Over Internet Protocol (VOIP):
Case Study Analysis

By: GPDCM Jayasekara

Case Overview

Scenario

Voice over Internet Protocol (VoIP) is simply the transport of voice traffic over the Internet or any IP-based
packet-switched network. This is in contrast with the traditional telephone network, carrying voice data over
dedicated circuit-switched transmission lines. A major challenge in implementing VoIP is to ensure
sufficient bandwidth is available and access to the available bandwidth is controlled and prioritized.
Sufficient bandwidth is required to maintain high-quality voice. On the other hand, controlling the
bandwidth limits bandwidth-hogging applications and guarantees access for delay-sensitive applications.

A major issue with IP, however, is that it does not provide any guarantee of service. Another shortfall of IP
is that packets can arrive out of order at their destination. In fact, in extreme cases some packets may be
severely delayed or may not arrive at all.

Another choice of protocol that can be used in conjunction with IP is user datagram protocol (UDP). Unlike
TCP, UDP does not guarantee in-sequence and error-free packet delivery. Yet, despite UDP’s unreliable
nature, it provides faster packet delivery compared to TCP. For voice communications packet loss of about
five percent is generally acceptable. However, voice traffic is very delay sensitive. Consequently, UDP
happens to be a proper choice for transporting voice traffic.

The purpose of this coursework is to familiarize you with building a test VoIP lab setup using Open-source
tools and experimenting how different network impairments would affect VoIP quality. The report should be
based on your lab setup activities, investigations, data analysis and research.
Question 01

Task 01:

The quality of a VoIP connection is one of the main factors to be considered in designing and deploying
VoIP based systems. The quality of VoIP is influenced by:

• VoIP protocol, for instance choice of codecs

• User devices in use
• Underlying technology, such as: IP packet, available bandwidth, latency, and delay variation.

You need to assess the quality of VoIP (primarily as it is perceived by users) under different (realistic)
circumstances, including (but not limited to): various codecs, different networking environment (high/low
latency, high/low available bandwidth, etc.) and various types of telephony equipment.

First students need to setup a VoIP lab using below components.

• Asterisk server installed in an Ubuntu server

• A VoIP client (e.g. – X-Lite) installed in a separate host

• Students can use a Windows or Ubuntu host machine and install an Ubuntu VM running in the host
machine

After setting up of lab is done, students need to test and explain how VoIP call quality is varied with

i. Codec in use

ii. Latency of the network

iii. Bandwidth of the network

iv. Jitter in the network

The answer section to task 1 should include clear evidence of lab setup configuration and testing of
scenarios given above. To measure VoIP call quality, students may use MOS scores and subjective
observations.
Task 02:

The student will also investigate whether the use of Quality-of-Service technology will further stretch the
`usability' of VoIP especially during times when resources are scarce. Following Quality of Service
management approaches can be taken into consideration in constructing the technical answer to this part.

Using the same lab setup, students need to simulate below QoS scenarios and test how they affect the final
VoIP call quality.

• DiffServ
• IntServ
• 802.1p/q VLAN

The answer to task 2 should contain detail explanation on how these three different technologies affect VoIP
system quality and students need to include all simulation results in their answer script.

Task 03:

In IP and traditional telephony, network engineers have always made a clear distinction between two
different phases of a voice call. The first phase is "call setup," and includes all the details needed to get two
telephones talking. Once the call has been setup, the phones enter a "data transfer" phase of the call using an
entirely different family of protocols to move the voice packets between the two phones. In the world of
VoIP, SIP is a call setup protocol that operates at the application layer.

Explain in detail how SIP messages call flow is happened in following given scenarios. You will use a
suitable tool such as Wireshark or tcpdump to simulate these call flows and extract relevant log captures.

1. A successful call between two SIP users within the same domain

2. A successful call between two SIP users within two domains

3. A failed call attempt between two SIP users within the same domain

The following approach is proposed for this assignment:

1. Literature study, focusing on VoIP technology, Quality of Service and perceived quality for speech.

2. Various experiments carried out to get quantitative figures on the `usability and quality' of VoIP.

3. Recommendations given on (especially) the minimal requirements for networks/links to support

VoIP.
Question 02

a) In VoIP world, usage of appropriate Voice Codecs plays a major role when it comes to determining
required resources for the system. What are the main functions of Voice Codecs in a VoIP system?

b) G.711 and G.729 are two of the most used Codecs in Voice communication. Compare these two
Codecs and clearly mention out advantages of each Codec.

c) Based on standard values and reasonable assumptions, calculate the sample size of a VoIP
communication based on G.729 Codec?

d) What are the three main types of QoS Architecture models in VoIP systems? Briefly explain how
each model addresses different QoS requirements.
Acknowledgement
First, I would like to show my deepest gratitude & respect to Mr. Duminda Nishad, for helping & guiding me
throughout this research. The completion of this research report gives me much fascination, as it would not
have been successful without the hard work, dedication & determination that has been put into this. I was able
to provide with optimum commitment to the research. Thank you, dear sir for making this a success!
Table of Contents
Case Overview................................................................................................................................. 2
Acknowledgement ............................................................................................................................ 6
1.0 TASK 01 ................................................................................................................................. 8
1.1 Setting Up VOIP Setup .............................................................................................................. 8
1.2 Applications Used ................................................................................................................... 8
1.3 Installing Ubuntu OS ................................................................................................................. 9
1.4 Installing Asterix Server ........................................................................................................... 10
1.5 Asterisk Configuration ............................................................................................................. 12
1.6 Changing NAT to Bridget Adapter (Same LAN) ......................................................................... 15
1.7 Setting Up Soft Phones ............................................................................................................ 16
1.8 Testing SIP call between 7001 Laptop & 7003 PC ...................................................................... 18
1.10 Proof of call with Wireshark Capture ....................................................................................... 20
1.11 Latency Of The Network ........................................................................................................ 23
1.8 Codec in use ........................................................................................................................... 24
G.729 CODEC .......................................................................................................................... 25
G.711 CODEC .......................................................................................................................... 26
1.12 Bandwidth Of The Network .................................................................................................... 27
1.13 Jitter In The Network ............................................................................................................. 28
TASK 02 ......................................................................................................................................... 29
DiffServ ................................................................................................................................... 29
IntServ ..................................................................................................................................... 29
802.1p/q VLAN ........................................................................................................................ 29
TASK 03 ......................................................................................................................................... 30
A successful call between two SIP users within the same domain ....................................................... 30
Proof of call with Wireshark Capture ........................................................................................... 32
A successful call between two SIP users within two domains ............................................................ 35
A failed call attempt between two SIP users within the same domain.................................................. 37
Question 02 ..................................................................................................................................... 43
a) In VoIP world, usage of appropriate Voice Codecs plays a major role when it comes to determining
required resources for the system. What are the main functions of Voice Codecs in a VoIP system? ...... 43
b) G.711 and G.729 are two of the most used Codecs in Voice communication. Compare these two
Codecs and clearly mention out advantages of each Codec. ............................................................... 44
c) Based on standard values and reasonable assumptions, calculate the sample size of a VoIP
communication based on G.729 Codec? .......................................................................................... 45
d) What are the three main types of QoS Architecture models in VoIP systems? Briefly explain how
each model addresses different QoS requirements. ........................................................................... 45
REFERENCES .............................................................................................................................. 46
About Author ................................................................................................................................... 47
END REPORT ................................................................................................................................. 48

1.0 TASK 01
Introduction to Voice Over IP

Voice over internet protocol, sometimes known as VoIP, is a technology that allows users to make free
phone calls or video calls over the internet. Voice over IP (VoIP) is a technology that allows you to transmit
your voice from an analogue transmission to a digital one. After that, you can make a direct phone call. IP
telephony, broadband, and broadband telephony are all terms that can be heard. VoIP services are subject to
fluctuations. As previously stated, VoIP frameworks give clients the option of selecting their own
equipment. Calls can now be made using a variety of devices, including traditional telephones, IP
telephones, personal computers, and cell phones, ensuring that telephone calls are no longer restricted to a
single device. As a result, VoIP has made tremendous strides forward in the communication field. Voice
over IP (VoIP) is an example (VoIP). Voice over Internet Protocol (VoIP) is a method of communicating
over the internet.

1.1 Setting Up VOIP Setup

1.2 Applications Used

These are applications we will be using that we will be using:

So as for the soft phone we will be using Micro SIP for the PC (Windows) & for the
mobile phones we will be using Line Phone.

We will be using Wireshark to capture packets between the conversation.

For the virtual machine we will be using Oracle VM Virtual Box

1.3 Installing Ubuntu OS
Ubuntu

Known as Ubuntu, it is a complete Linux operating system that is provided for free, with both community
and expert assistance. Founded on the principles outlined in the Ubuntu Manifesto, the Ubuntu community
believes that software should be available free of charge, that software tools should be usable by people
speaking their native language and despite any disabilities, and that people should have the freedom to
customize and alter their software in any way they see fit. The system is made by a company called
Canonical Ltd that is based in the United Kingdom.

Installing Ubuntu OS in Virtual Box, and the installation took several minutes to setup.
1.4 Installing Asterix Server
Asterisk Server

Digium's Asterisk is an open-source VoIP server software that can be installed on a PC and run as a server.
Telecommunications applications can be built with Asterisk as well. You have complete access to Asterisk's
source code because it is an open-source technology. Using Asterisk's capabilities and components, you can
create your own services and features. You can download and use Asterisk software for free. Asterisk, in
general, does not necessitate additional hardware. Asterisk's main selling point is its ability to transform any
PC into an IP PBX. Although Asterisk was designed to run on Linux, it is also available for use on BSD,
Windows, and Mac OS X servers.

Installing Asterix Server to Ubuntu OS by using “sudo apt-get install asterisk -y” so this will install all the
relevant requirements for the asterisk server.

Finished installing the asterisk server on ubuntu OS

After installation checking whether asterisk is professionally installed by using “sudo asterisk -r” this is to
check availability of the of the asterisk server.

Later to check the asterisk server’s configuration files we use us “ls” and there are 3 main configuration files
that we must configure for the asterisk server to setup properly and work for the VOIP setup.

Those are the 1) sip.conf 2) extension.conf 3) voicemail.conf configuration files.

1.5 Asterisk Configuration
After all the relevant packages have been professionally installed & working then we should proceed to
create the users of the VOIP system.

Then we enter “sudo gedit sip.conf” where it will prompt a text editor when, Entered So we have decided to
create 4 users (7001, 7002, 7003, 7004) with passwords equal to their usernames.

Below are the 4 assigned users [7001, 7002, 7003, 7004]

Then we enter “sudo gedit extension.conf” where it will prompt a text editor when entered, so after this is
opened, we should proceed adding the extensions to add to the process of its incoming calls.
Then we use “sudo gedit voicemails.conf” to edit the voicemails configuration of a call session.

After all the configuration files have been configured and set, we must go to “sudo asterisk -r” to make sure
the relevant changes have been applied. So, once we are inside the asterisk server command line we type in
a “reload” to save things off.

Then we can run the Asterisk server.

1.6 Changing NAT to Bridget Adapter (Same LAN)
So, as we need to get the same IP network address for the server and host machine or mobile phone to
connect as for a same LAN network, we need to change the NAT and make the connection using a Bridge
adapter so that all can be connected to the same LAN and communication will be possible.

So, as clearly stated below we can see that the IP address has been changed to 192.168.1.12 where this will
be the asterisk servers IP and all the devices in the LAN shall use this as its domain to communicate with
each other users.

The network address is 192.168.1.0 & The asterisk server IP is 192.168.1.12

1.7 Setting Up Soft Phones
In our scenario we have decided to use 3 users to SIP communicate with each other they are:

1) 7001 for our laptop running windows 11 (HP Notebook) connected to SIP Server via 192.168.1.12
For the below example we have used Micro Sip Softphone for Windows Laptop, then Online.

2) 7002 for our smartphone running android (Techno Cameo 16) connected to SIP Server via
192.168.1.12 For the below example we have used Lin Phone Softphone for Android Phone
Then Online.
 Lin Phone
Lin phone is an open-source application that allows users to make free audio/video calls as well as
text messages. Any time you have a Wi-Fi or 3G/4G internet connection, you can be reached by
anyone at any time, even if the app is closed. Due to the separation between the user interfaces and
the core engine, Lin phone can support a variety of different user interfaces on top of the same set of
functionalities, which is especially useful. Telecommunications requires two processes: media (voice
or video transmission, encoding and decoding) and signalling (routing calls, ringing, accepting a call
etc.) Lin phone combines these two aspects to manage most of these tasks. Without prior knowledge
of VoIP or telecoms, a programmer can easily integrate video calls and instant messaging into any
software.

Advantages of Lin Phone

Improved experience with instant messaging
HD video calls with a video preview are available.
Lin phone can work with most PBXs and SIP servers, but not all of them.
Graphical user interface that is simple to utilize
Communications that are secure

3) 7003 for our pc running windows 7 (MSI) connected to SIP Server via 192.168.1.12
For the below example we have used Micro Sip Softphone for Windows PC, then Online.

4) 7004 is a reserved user just in case one of the above fails.

1.8 Testing SIP call between 7001 Laptop & 7003 PC
So, we are going to initiate a SIP between 192.168.1.11 and 192.168.1.13

So, after that we initiate a call from 7001 to 7003 using the soft phone.
So now we can see the call has been successfully transferred to 7003 by 7001

Sender: Laptop [email protected] to PC [email protected]

Here we can see that both parties can hear each other, and mics and speakers are successfully emitting
sounds.
1.10 Proof of call with Wireshark Capture
So, this is the packet capture of the ongoing call

And now we must go to telephony and click on VOIP calls where it will redirect you to a page below and
now as this is a single session of duration of 24 seconds, we can see that 9 packets have been sent and the
protocol used is SIP.
And after clicking on the flow sequence, we can identify that the call sessions activity and its flow of the SIP
and, we could identify the CODEC in use which is G-711

And we could play stream so study the packet movements and its playback by properly analyzing its content
section by section in a clear graph.
Below is the filtered SIP occurred during the call.
1.11 Latency Of The Network
So, in the below diagram we can see that the latency check of network, so we pinged the PC 192.168.1.13 to
the asterisk server 192.168.1.12, first there was a 25% packet loss and then 0%

Every business needs a good phone system. So, when it comes to choosing a phone system, business is
always the most important thing to think about first.

• Cost • The use of technology to make things run more quickly • Quality

This is the packet capture in Linux Ubuntu using TCP DUMP in terminal to analyze the network
1.8 Codec in use
In the case of VoIP codecs, it converts the analog audio signal into a digital package or digital compressed
system for transmission, and then converts the compressed audio signal back to the analog audio signal. The
VoIP codec controls the frequency of calls and interruptions, as well as the location from which calls are
made over the internet.

In the asterisk server we know that it supports multiple codecs such as:

G.711 (A-law & μ-Law); G.719 (pass through); G.722; G.729a; etc.…

In Micro Sip

In Asterix

In Lin Phone

So, there are some similarities between them for example:

G.711, Speed, GSM, G-729, G-722, iLBC, Opus, SILK, PCMU, PCMA

CODEC COMPARISON G.711 vs G.729

Further explained further below

G.729 CODEC
So, here we can clearly see due to latency some packets have been dropped due to high compression. O we
can see that out of 1142 packets there is a 0.6% packet loss which is 7 packets in total. This diagram is taken
in Wireshark by accessing the RTP streams of ethernet 3

Here, we can see that the expected packets amount was 1149 but received amount is 1142 so there is a
packet loss which could degrade the original message in a call. So, the codec in use is G.729, this because of
its high compression and due to that there is a high chance of loosing packets in this form off codec
compression, whereas it is a hybrid.

G.729 is a hybrid codec with8Kbps bandwidth requirement, MOS of 4.0 & 15ms coder delay
G.711 CODEC
So, here we can clearly see there is 0 latency in the packets no packets have been dropped. Now we can see
that out of 1142 packets there is a 0.0% packet loss which is 0 packets in total. This diagram is taken in
Wireshark by accessing the RTP streams of ethernet 3

So here we can see that compression in g.711 is low that it has 0 packet loss, but for this waveform signal
this requires a high bandwidth.

G.711 is a waveform codec with MOS of 4.3 & 64 Kbps bandwidth requirement.
1.12 Bandwidth Of The Network
In network terms, it is a measure of how much information two or more biases can convey to each other
across a network connection. We consider data from point A to point B just as relevant as data from point of
significance to point B for the sake of our water force. The volume, bandwidth, and variety of the
information that is delivered have an impact on how well the media, which is analogous to a communication
network connection, performs because of the information being sent.

So here is an example of the bandwidth of the network that I am currently using G.729:

Ping is 22ms Download Speed 41Mbps | Upload Speed 2.7 Mbps Jitter 13ms

And this is a speed test on the network when a SIP call is taking place with G.711:

Ping 40ms (+18) | Download Speed 19Mbps (+20) | Upload Speed 3.6Mbps (+1) | Jitter 18ms (+5)

So, we can see there is an increase in bandwidth requirement in the 2-codec used and G.711 needs more
bandwidth while G.729 uses lesser bandwidth due to its high compression.
1.13 Jitter In The Network
Jitters are unwanted sounds or noise or unwanted bits of data in a stream of signals that could deviate if there
is a lot of jitters occurred in a call. Many VoIP phone calls have been lost because of the jittery nature of
internet communication. An organization that relies on guest calls or a good business connection may suffer
if its VoIP calls are not clear and easy to understand amongst the company's top executives. As a network
performance concern, network failure continues to result in arrests, detentions, and package loss.

Jitter is a change in latency, or the time delay between when a message is transmitted and when it is
received, as previously stated. This difference is measured in milliseconds (MS) and is defined as data
transmission interruptions. The Jitter network blockchain caused many network connections to hit small
bags of data to give each business the same IP network address at the same time.

The long-term network transformation is a unique feature of VoIP jitter. Quiescence grows as the time
allotted for a single data bag passing through it decreases.

So here in this stream analysis of a sample SIP call between 7001 and 7003 there we can see that in the RTP
stream analysis we could see the jitters as for the above example there

The max jitter is 11ms min jitter is 0.6ms mean jitter is 5.6

Jitters in G.711 was 18ms

Jitters in G.729 was 13ms

TASK 02
DiffServ
DiffServ is a network traffic classification system that divides network traffic into several classes. DiffServ
then assigns distinct policies to each of these separated classes, allowing them to provide varying degrees of
service to their respective customers. The Diffserv reduces the strain on network devices and allows for easy
scaling as the network grows.

Diffserv functions are Packet classification, Packet Marking & Traffic Conditioning while Congestion
Management Diffserv is a simple and scalable networking architecture that is used for classifying and
managing network traffic while also delivering network Quality of Service (QoS).

IntServ
In addition to these two essential industry standards, Cisco IOS Software also supports a third. There are a
variety of services, as well as Integrated Services, to choose from. Different services allow the package's
"class" label to be written directly to the package, as opposed to the IntServ type, which requires a label to
inform package drivers that special package QoS treatment is required. Different services also allow the
package's "class" label to be written directly to the package. Integrated Services provides reliable Quality of
Service systems that are capable of dynamically updating traffic, allowing diverse services to reach the
highest possible Quality of Service scalability while maintaining high levels of reliability. Even while these
techniques are rewarding, they are not self-contained.

Remote video, social media, visual effects, and a virtual world are all features that the IntServ architecture
(RFC 1633, June 1994) desperately needs. It establishes a method for providing top-to-bottom Quality of
Service applications that are rapidly required by explicitly managing network infrastructure to provide QoS
and streamline users. To stabilize and maintain QoS, it uses the "infrastructure provisioning" and
"acceptance management" systems as the main building blocks.

End-to-end routing devices are used in conjunction with the Resource Reservation Protocol (RSVP) by
Integrated Services to clearly specify the QoS of application traffic on the network. If any connection device
along the path can maintain the required bandwidth, the startup may begin to send data.

802.1p/q VLAN
802.1p is a MAC layer-based quality of service (QoS) / class of service (Cos) system (Layer 2). It is possible
to add software that supports 802.1p and identify a value indicating an important level of Ethernet interval.
For unsatisfactory communications, such as VoIP, 802.1p can help ensure a high level of communication.

Traffic is driven by boxes and other 802.1p components based on the value required. Only VLAN slots
designed to send and receive VLAN signal traffic are covered by 802.1p. This includes both the exit route
out of the firebox and the exit route into the firebox.

When the VLAN interface is configured with 802.1p, Firebox displays traffic from that interface. The
Firebox adds 802.1q tags to Layer 2 Ethernet frames and uses the Layer 3 IP header area code (PCP) 802.1p
and 802.1q tags to list the value of the IP path.
TASK 03
A successful call between two SIP users within the same domain
So, we are going to initiate a SIP between 192.168.1.11 and 192.168.1.13

So, after that we initiate a call from 7001 to 7003 using the soft phone.
So now we can see the call has been successfully transferred to 7003 by 7001

Sender: Laptop [email protected] to PC [email protected]

Here we can see that both parties can hear each other, and mics and speakers are successfully emitting
sounds.
Proof of call with Wireshark Capture
So, this is the packet capture of the ongoing call

And now we must go to telephony and click on VOIP calls where it will redirect you to a page below and
now as this is a single session of duration of 24 seconds, we can see that 9 packets have been sent and the
protocol used is SIP.
And after clicking on the flow sequence, we can identify that the call sessions activity and its flow of the SIP
and, we could identify the CODEC in use which is G-711

And we could play stream so study the packet movements and its playback by properly analyzing its content
section by section in a clear graph.
Below is the filtered SIP occurred during the call.

SIP messages call flow

1) So, the call flow happens as it has some main characteristics and messages that must be taken place
for a SIP communication to occur. So, the first one is INVITE where the sender asks a receiver to
join to a SIP session.
2) Then we have TRYING 100 where the receiver end will check whether the end-user is in position to
be in call,
3) If yes, then RINGING 180 so in this phase the VOIP will check whether the end-user has enough
resources to participate in a call
4) If yes, then it will send OK 200 saying to the sender that he is available to participate to the call.
5) The sender then gets the OK 200 signal & sends an acknowledgement saying that if you’re ok and I
am ok then let’s initiate a call.
6) Then the call will take place in an RTP session between the two end users.
7) Then when the sender or receiver decides to ruminate the call, they will send a BYE to one another.
8) Then the receiver or sender receives the BYE and sends an OK, so after this the call will be
successfully terminated on both ends.
A successful call between two SIP users within two domains
So, for this scenario we need two domains so in one domain I have installed an Asterisk server into an
Ubuntu OS and for the other domain I have installed an Asterisk server into a CENT OS.

And I settled the Linux ubuntu as well, where a SIP proxy needed to be installed on both because a proxy
server needs to be installed between 2 domains to communicate with each other.

Setting up the phone to connect with the CENT OS, where soft phone Lin phone is used.
Here we can see that the Laptop is connected to the SIP server, where it is now initiating a call with the
other domains Phone via CENT OS.

Here below we can see that the CENT OS server has successfully received the call and has successfully
forwarded it to the Lin Phone and properly answered the SIP call and now runs in RTP.

This is the packet capture in Linux Ubuntu using TCP DUMP in terminal to
analyze the network communication between the two domains.
A failed call attempt between two SIP users within the same domain
So, the plan is to initiate 2 Sip calls between users [email protected] Laptop and [email protected] PC
so in this we are going to use Codec G.711 & switch [email protected] ‘s Codec to a G.729 & by doing
this this should terminate the call successfully.

G.711 is selected in both [email protected] Laptop and [email protected] PC

Then [email protected] Laptop calls [email protected] PC

Then [email protected] PC receives the call from [email protected]

The call has been successfully initiated on both ends

Then [email protected] PC

[email protected] Laptop
So now [email protected] changes its codec from G.711 to G.729

So, as we can see the packets dropped and the call was terminated successfully, the call was a failed attempt
because the codec was changed in between the SIP.
So now in the packet capture we can filter out the SIP sessions using SIP in the filter tab.

So now we must go to the telephony and click on the VOIP calls to set this tab up and we must click and
select on the flow sequence to check the flow of the SIP.
So, now we have clearly identified the flows we can see that the call has been terminated immediately with
an error message. Where it is unidentified.

SIP messages call flow

9) So, the call flow happens as it has some main characteristics and messages that must be taken place
for a SIP communication to occur. So, the first one is INVITE where the sender asks a receiver to
join to a SIP session.
10) Then we have TRYING 100 where the receiver end will check whether the end-user is in position to
be in call,
11) If yes, then RINGING 180 so in this phase the VOIP will check whether the end-user has enough
resources to participate in a call
12) If yes, then it will send OK 200 saying to the sender that he is available to participate to the call.
13) The sender then gets the OK 200 signal & sends an acknowledgement saying that if you’re ok and I
am ok then let’s initiate a call.
14) Then the call will take place in an RTP session between the two end users.
15) Then when the sender or receiver decides to ruminate the call, they will send a BYE to one another.
16) Then the receiver or sender receives the BYE and sends an OK, so after this the call will be
successfully terminated on both ends.
Question 02
a) In VoIP world, usage of appropriate Voice Codecs plays a major role when it comes to
determining required resources for the system. What are the main functions of Voice
Codecs in a VoIP system?

When it comes to estimating the number of resources needed for a VoIP system, the use of proper Voice
Codecs is critical. Voice Codecs are used for what purposes in a VoIP system?

VoIP codec is a modern technology that specifies how Voice over Internet Protocol (VoIP) phone
conversations' audio, bandwidth, and comparison functions will be implemented. It will be put to effective
use very soon. There are complex algorithms involved in using Voice-over IP codes.

People use VoIP codecs to do this. In this case, it makes a digital package out of the analogue audio signal
and compresses it so that it can be sent over the internet quickly. In the end, when you get it back, it's still an
audio signal that has been squeezed together Voice-over IP codecs decide how often and how long calls and
conversations are interrupted when they happen over the internet. There are diverse types of codecs. If the
Voice over IP provider has a lot of databases, loyalty isn't a big deal for most telephone calls. How well the
codec works with diverse types of devices is what determines how well it will work overall.

All codecs for voice-over-internet protocol are used for the same thing. That means they get information
quickly and move it around. [1] VoIP business competitions make sure that the telephone call doesn't use
too much bandwidth and that the call is clear and sharp.

Voice over IP converts audio waves into digitized data packages that may be sent over short distances
(LAN). Quality of Service can speed up some data bundles. With this functionality, voice over IP data
packets can override less critical ones. This allows VoIP codecs to accommodate more calls. The sole
difference between Voice over IP codecs is the sound. It is required for audio transmission due to the limited
bandwidth. Files and text documents accept a lot of space, and there is no compression, therefore this is a
problem for the organization.

In VoIP, Voice Codecs do the following things:

• Encoding and decoding

• The compression and decompression of files

• The process of encrypting and decrypting things.

To get better sound quality, you should both raise the sample rate and depth. It's changed, but. This adds to
the amount of information needed to show the voice. To avoid this trade-off, the codec also adds
compression algorithms that help keep throughput high and reduce the size of the digitized volume. Even
though compression can cut down on noise a little, it can also improve the quality and volume of voice
media, which makes it more efficient.

When we use the Internet to make voice calls, like through Voice over IP or other digital networks, the voice
is stored in digital data, as well as other things, too. In the same way, the data is combined for faster delivery
and a better way to talk to each other on the phone This encoding comes from codecs (which are abbreviated
to code conversion).
b) G.711 and G.729 are two of the most used Codecs in Voice communication. Compare
these two Codecs and clearly mention out advantages of each Codec.

G.711 and G.729 are two of the most widely used Codecs in voice communication, and they are both
standardized. Compare and contrast these two codecs, highlighting the distinct advantages of each codec.
Sometimes, codecs are used as a medium to increase the capacity of a network's bandwidth operation by
compressing the data transmitted during a phone call and dismembering the data when the call is made. If we
consider the example of a 10Gb data connection, only a few calls will be able to be transmitted over this medium.
However, because Codec will be used to transfer this data, the number of calls required to transfer the connection
will be significantly increased.

This Codec is responsible for two major performances, which are U-law and A-Law, respectively. In countries
such as Japan and North America, U-law is commonly used, whereas A-law is commonly used in the remaining
countries of the world. A feature of the G. 711 codec is its ability to provide aesthetically pleasing VoIP call
quality while operating in the background and without the use of contractions. It is because of this that the call
quality is more like that of using an ISDN phone all the time. This Codec is supported by many VoIP providers.

G. 729 is a number that represents the number of people who have died in the year 729.

As of today, it is believed that G. 729 codec provides the most fashionable position of calling at a rate of at least
8Kbps (kilobits per nanosecond), which means that the only codec that is suitable for admitting additional calls
from a bandwidth is G. 711. Before implementing Codec G. 729, it is recommended that you reduce the amount
of bandwidth used to transfer call data packets. Furthermore, it necessitates additional CPU processing time
allocation to make calls. In this way, one will be able to find many VoIP phones that are capable of handling
one G 729 call while also providing sufficient processing power at a specified time. G. 729 recommends a
product that requires the stoner to purchase a license from a dispensary to use it legally.

The G729 algorithm is a globally recognized algorithm that is supported by many carriers, service providers,
and VoIP equipment manufacturers worldwide. Companies that invest in products that support this technology
can have a safer effect on their customers while also ensuring that they are not spending too much money on
transportation.
c) Based on standard values and reasonable assumptions, calculate the sample size of a
VoIP communication based on G.729 Codec?

Based on standard values and reasonable assumptions, calculate the sample size of a VoIP communication based
on G.729 Codec?

[E ]-Bytes [D ]-Bytes [C ]-Bytes [B ]-Bytes [A ]- Bytes

Second Layer Tunnel header Possible firewall Either Digitized Voice

Header header IP+UDP+RTP, The size of this
Header section depends on |
or on the codec type
cRTP and the
Header amount (msec) of
analog voice
that is digitized and
encapsulated |
in each IP packet.

A = Amount of digitized voice per packet (Bytes)

= CODEC Bandwidth (bps) x Packetization Period (in Sec) / 8 (bytes)

F = Total Frame Size (bits) = 8 x (E + D + C + B + A)
R = Packet Rate = 1/ (Packetization Period in Seconds)
Bandwidth per call (kbps) = F x R divided by 1000

If the total frame size is 66 bytes, which is equivalent to 528 bits, and the packet rate is 50 pps, multiplying
528 by 50 results in a total bandwidth of 26400 bits per second, or 26.4 Kbps

d) What are the three main types of QoS Architecture models in VoIP systems? Briefly
explain how each model addresses different QoS requirements.

Quality of service (QoS) is used to provide different routes for diverse types of traffic, to control
delays and redundancy, and to reduce the number of packages lost. When the network is over or
under congested, Quality of Service (QoS) can be used to emphasize that the transmission of
business traffic is critical.
QoS (Quality of Service) is a system architecture for controlling network resources such as
bandwidth, delay, loss, and packet loss. All Quality of Service (QoS) systems are designed to have
an impact on a minimum of one, if not all, of the features listed above.

There are three major architecture planning models in this document.

• Best Efforts on Your Part

• Integrated Services are provided.

• Services that are distinct from one another.

REFERENCES

Voice over Internet Protocol (VoIP) Based IP PBX System Design. (2016).
International Journal of Science and Research (IJSR), 5(2), pp.1380–1385.

Air call Blog. (2020). Experiencing VoIP Issues? Here’s How to Improve Call
Quality Fast. [online] Available at: https://aircall.io/blog/call-center/voip-issues/
[Accessed 14 Apr. 2021].

Call Hippo. (2019). Check Out How VoIP QoS Can Improve Your Call Quality.
[online] Available at: https://callhippo.com/telephony/quality-service-qos-
importance-voip# [Accessed 17 Jan. 2022].

Nextiva Blog. (2020). What Are VoIP Codecs & How They Affect Call Sound
Quality? [online] Available at: https://www.nextiva.com/blog/voip-codecs.html
[Accessed 15 Apr. 2021].

Goode, B. (2002). Voice over Internet protocol (VoIP). Proceedings of the

IEEE, 90(9), pp.1495–1517. end
About Author

Personal Quote
“You'll never get what you want if you don't pursue it. If you don't ask,
you'll always get a no. If you don't move forward, you remain
stationary.”

Who Is GPDCM Jayasekara?

“My name is GPDCM Jayasekara, I consider myself an optimistic & creative individual who has
passion in the field of computer networks & automation & hope to create a difference in the industry.
I am a smart and diligent collaborator who enjoys exceeding targets. I am also enthusiastic and
willing to learn new things that would challenge me both personally and academically, which would
help me be a better version of myself.

Personal Objective
To carve a niche for myself as a professional in the computer network industry with a reputed and
well-managed organization where my potential is utilized to the fullest, thereby leading to the growth
of both the organization as well as my career in the organization. Further, I’m interested in pursuing
higher studies in computer security in which I can help improve my knowledge for the betterment of
the organization and our society.
 END REPORT

View publication stats