Abstract that optimize the media stream based


 The objective of the
coursework is to implement the Voice over Internet Protocol (VoIP) call using SIP Server (Trueconf) and analysis the result
based on the quality of service, security and using of different codec (G.711
and G.722) of VoIP call over wired (LAN) and wireless (Wi-Fi) network. Voice over Internet Protocol (VoIP) is a methodology and group of
technologies for the delivery of voice
communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. VoIP allow
to make calls free or very low cost. Voice over Internet Protocol telephony
provides many features and services which are expensive using traditional
telephony networks.  They are terms like
Internet telephony, broadband telephony and broadband phone service refers to
the devices for communication services over public internet rather than the
public switched telephone network (PSTN). VoIP having many
advantages over public switched telephone network (PSTN). Therefore, the report
is about a brief introduction on the basic principles and technologies of SIP
based VoIP networks, their advantages and disadvantages, and analysis of the
wired/wireless VoIP calls by using TrueConf SIP Server.


Principle of Operation of VoIP Systems

Voice over Internet Protocol (VoIP)
is a method for the delivery of voice
communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.  The basic principle of operation of VoIP
telephone calls are very similar to traditional telephony. The steps involved
are signalling,
channel setup, digitization of the analog voice signals, and encoding. Instead of being transmitted over a circuit-switched
network, the digital information is packetized,
and transmitted as IP packets over a packet-switched network. The media streams are transported using special
media delivery protocols that encode audio and video with audio
codecs, and video codecs (The term
codec is the short form for “encoder/decoder”).
The media streams are encoded at the transmitter end, sent over the internet
and decoded at the receiver end. The codecs must match at each end of the
communication link. Various codecs exist that optimize the media stream based
on application requirements and network bandwidth; some implementations rely
on narrowband and compressed
speech, while others support high-fidelity stereo codecs. Some popular codecs include µ-law and a-law versions of G.711, G.722, an open source voice codec known as iLBC,
and many others (“Voice over IP”, 4th December 2017, para. 1 &



in Implementation of VoIP Systems

“Voice over Internet Protocol (VoIP) has become a popular
alternative to traditional public-switched telephone network (PSTN) networks
that provides advantages of low cost and flexible advanced digital features.
The flexibility of the VoIP system and the convergence of voice and data
networks brings with it additional security risks” (Butcher, Li & Guo,
2007). Therefore, it is necessary to analyse the challenges involved in
implementing the VoIP systems. Some of the major drawbacks of VoIP systems are
as follows:

a)    Quality
of Service (QoS) –

Quality of service (QoS) is fundamental to a VoIP network’s
operation. A VoIP application is much more sensitive to delays than its
traditional data counterparts. Tools such as encryption and firewall protection
can help secure the network, but they also introduce significant delay. Jitter,
referred to nonuniform delays is another QoS issue, it can cause packets to
arrive and processed out of sequence. The Real-Time Transport Protocol (RTP),
which is used to transport voice media, is based on UDP, so packets received
out of order can’t be reassembled at the transport level, but must be reordered
at the application level, introducing significant overhead. To control jitter,
network designers can use buffers and implement QoS-supporting network elements
(especially routers) that let VoIP packets “play through” when larger data
packets are scheduled ahead of them. Another issue that must be dealt with is
the packet loss. However, VoIP packets are small, containing a payload of only
10–50 bytes, or approximately 12.5–62.5 ms, with most implementations at the shorter
end of the range. The loss of such a minuscule amount of speech is
indiscernible, or at least unworthy of complaint, by a human VoIP user (Walsh
& Kuhn, 2005).

call capacity –

over Internet Protocol (VoIP) bandwidth consumption over a network is one
of    the most important factors to
consider when deploying a VoIP infrastructure. Failure to account for VoIP
bandwidth requirements will severely limit the reliability of the VoIP system
and place a huge burden on the network infrastructure and VoIP Server in
question (Soroyewun & Obiniyi, 2015).
A link which has high bandwidth may be able to carry large volumes of data
packets, but the network links with low bandwidth can cause packets loss and
other QoS problems. Hence, proper bandwidth reservation and allocation is
essential to VoIP quality. Sharing data and voice on the same wires, which is
one of the great attractions of VoIP, is also a potential challenge for

of CODEC –

In VoIP systems, codecs serve as an
important function – they enable the conversion of analog voice signals into
digital packets that can be sent over the Internet. VoIP codecs generally
accomplish encoding/decoding and compression/decompression. The codec’s job is
to convert the human voice into digitized packets that can be routed over the
internet just like any other form of media. These voice packets can be flagged
with a higher priority through QoS settings to enable them to pass through
firewalls or another border equipment quicker. This is often necessary to
ensure that audio does not get dropped during calls. The main difference
between the various types of VoIP codecs used by providers comes from the way
they compress audio. Unfortunately, there is no perfect codec that is
acceptable for all enterprises under all circumstances. Some high-quality
codecs require licensing fees whereas others are open source, some offer better
quality while others use less bandwidth. All these aspects will affect the final
choice of codec selection by a provider, client or organization.


Eavesdropping is the major security threat in VoIP systems
especially when transmitting the data packets in wireless communication. Through eavesdropping, a
third party can obtain names, password and phone numbers, allowing them to gain
control over voicemail, calling plan, call forwarding and billing information.
This subsequently leads to service theft. VoIP systems consists of softphones
and software that are vulnerable to viruses and malware attacks. Another
security threat is call tampering, for example, the attacker can simply spoil
the quality of the call by injecting noise packets in the communication stream.


TrueConf SIP Server

Trueconf is a developer of software and
hardware solutions for video conferencing over the internet and in corporate
networks. Trueconf provides unique products like video conferencing server.
Trueconf server provides cloud service for trueconf online and trueconf mobile,
client applications for mobiles devices running on android and iOS. It helps to
organize people and video conferences in anywhere like in the office, at home
are in public places.

Trueconf software architecture is based
on Scalable Video Coding (SVC)



Setup and Configuration of TrueConf SIP Server

TrueConf Server is shipped as a software
installation package that contains the server side and client applications for
popular platforms. After the installation package is downloaded, it is launched
to begin the installation. TrueConf Web Manager port is determined during the
server installation. Since we were installing the server behind the firewall,
to complete the registration TCP port 4310 was used to access from inside to
internet and port 80 was used between server and client applications. To
complete the registration, the free registration key was obtained. After the server had been successfully registered, on
top of the activation window of web-configurator a special sign appeared on the
server status stating that the server is running and registered. Once the
server has been configured, the TrueConf client software is installed on the
client machine. Then internal and external addresses must be specified. Internal addresses and ports are used for clients to
contact this server. By default, the server uses all IP addresses of machine on
default TrueConf Server port 4307. When default settings are on, current
connections are displayed in this column. External
addresses are the ports and IP addresses or DNS names, which help
client applications to connect to the server (“TrueConf Server Administrator
Guide”, n.d.). TrueConf Server has built-in gateway for SIP protocols


Practical Implementation of
VoIP Calls

TrueConf Server application has a dedicated softphone for making video calls. The
video call between two end-users can be established by just entering the user
ID or the IP address of the other user in the address bar as shown in figure 1.
And figure 2 shows the TrueConf softphone with video call in progress.



The five main functions of the TrueConf
SIP server in implementation of the VoIP calls are first the TrueConf SIP
server determines the user location and the type of end system that will be
used by the session. Second function of the SIP server is to determine the user
availability. End users can tell the system that they are available to talk or
that they are busy and do not wish to be disturbed. The third SIP function is
the user capabilities function. This is important because different devices
have different capabilities. For example, computer can do more things than a
phone, therefore, the user capabilities function allows the SIP to determine
the media being used and of the parameters being associated with it. The fourth
SIP function is the session setup. This function is responsible for connecting
the call. It establishes session parameters for both the caller and the
recipient of the call. The fifth and final function of the TrueConf SIP server
is the session management. It allows the users to end a call, transfer a call
to someone else, add multiple users to make a conference call or make
modifications to the session parameters (“An introduction to SIP and SIP functions”, n.d.).


Results and Drawbacks

The VoIP video calls using TrueConf SIP
server were successfully made through wired connection as well as through the
wireless connection via Wi-Fi. The sample packets were captured and analysed
with the observer. Problems with VoIP audio quality are always due to network
delay, jitter and packet loss. Observer helps in tracking network factors that
affect quality and reports call quality scores, which measure the overall VoIP
network health. Figure 3 shows the traffic summary of the call made through
wired connection using codec G.711.


4 and figure 5 below, shows the statistics of the overall call quality of the
call made through wired connection using codec G.711. The total number of VoIP
packets were 32,955. And the sample packets
show how signalling, call set up/take down, data transfer occur during a
typical session, duration of the call and the average jitter rate.






Figure 6 shows the traffic summary of
the call made through wireless connection using codec G.711.



Figure 7 and figure 8 below, shows the
statistics of the overall call quality of the call made through wired
connection using codec G.711. The
total number of VoIP packets were 34,345.



VoIP enjoys a
distinct advantage and supremacy of budget scalability in comparison with the
currently operational alternative telephony systems. This allows lines to be
shared with other users and services thus helping to lower the overall costs
over the circuit switched networks. However, being predominantly a network
based on software – it is exposed to the possibility of being attacked or
harmed by the progressively rising threat of cyber-attacks from crackers in
terms of malware such as viruses and worms. The VoIP traffic, must pass across
several different types of networks — often heterogeneous in nature.
Degradation of Quality of Service (QoS) is thus experienced while the traffic
traverses across such assorted networks. The QoS parameter of VoIP traffic
varies, and can be quantified by a range of divergent metrics, such as the:
jitter, end-to-end delay and Mean Opinion Score (MOS). The Mean Opinion Score
(MOS) has been used to subjectively measure the voice quality in a telephone
network (Miraz,
Molvi, Ganie, Ali & Hussein, 2017).
For the calls made over Wi-Fi networks, as
time passes, with the increased level of VoIP traffic due to the higher number
of calls generated, the MOS decreases. Whereas, there are better MOS scores for
the calls made over wired networks.



Voice over
Internet Protocol is the digitized voice traffic intrinsically transmitted
using a data network to make telephone calls. This differs from using a
traditional analogue circuit switched public network, as now the data has been
split into packets. These packets can take any route to reach the destination.
Packetized data travel through a virtual circuit which differs from a circuit
switched network in that the circuit does not need to be reserved for the
entire duration of the call between the sender and the receiver with packet
switching. The advantages offered include the multiple routing of the VoIP traffic
ensuring a cheaper and often free of cost flow of traffic between the different
intra-packet network components such as the routers and switches. On the other
hand, there are major drawbacks as well with regard to the quality of service
and security, especially for the calls over Wi-Fi networks. The other issue to
be dealt with is the choice of codec. In this project, two types of codecs were
used – G.711 and G.722.  Codec G.711 offers narrow-band voice with low processing
requirements, and with typical bit-rates of 32-64 kbps. G.711
is currently used in a wide domain of applications,
it performs best in
local networks, therefore best suited for wired networks. Codec G.722 offers wideband voice with low DSP processing
requirement. It
provides improved speech quality due to a wider speech bandwidth of
50–7000 Hz compared to narrowband speech coders like G.711. However, it requires a
bit-rate of 48-64 kbps per channel. This codec is best suited for calls over
wireless networks.

In this coursework, we implemented the VoIP calls
using TrueConf SIP server. The video calls were established both on wired and
wireless based networks. The overview of the principle of operation of VoIP,
advantages and major drawbacks were presented in this report. Next the overview
of TrueConf SIP server, its setup and configuration process, and its function
was presented. Then the analysis of the packets captured during laboratory
investigations were presented to show how signalling, call setup/take down and
data transfer occur during a typical session. Finally,
problems of making secure video calls at
acceptable QoS levels over current Wi-Fi networks based on the lab results
using different codecs (G.711 and G.722) were discussed.





Voice over IP. (4th
December 2017). In Wikipedia.
Retrieved December 19, 2017, from https://en.wikipedia.org/wiki/Voice_over_IP

Butcher, D., Li, X., & Guo, J. (2007). Security challenge
and defense in VoIP infrastructures. IEEE Transactions on Systems, Man,
and Cybernetics, Part C (Applications and Reviews), 37(6),

Walsh, T. J., & Kuhn, D. R. (2005). Challenges in
securing voice over IP. IEEE Security & Privacy, 3(3),

Soroyewun, M. B., & Obiniyi, A. A (2015). Empirical Study
of Achievable Bandwidth Capacity of VoIP Infrastructure in an Intranet with
Open Source Tools.

TrueConf. (5th
October 2017). In Wikipedia.
Retrieved December 19, 2017, from https://en.wikipedia.org/wiki/TrueConf

TrueConf Server
Administrator Guide. (n.d.). Retrieved December 19, 2017, from https://trueconf.com/support/online-help/server-help.html

An introduction to SIP
and SIP functions. (n.d.). Retrieved January 4, 2018, from http://searchunifiedcommunications.techtarget.com/tip/An-introduction-to-SIP-part-1

Miraz, M. H., Molvi, S. A., Ganie, M. A., Ali, M., &
Hussein, A. H. (2017). Simulation and Analysis of Quality of Service (QoS)
Parameters of Voice over IP (VoIP) Traffic through Heterogeneous
Networks. arXiv preprint arXiv:1708.01572.

Mohammed, M. H., & Abdullah, W. A. N. PERFORMANCE