IJSRED-V2I3P59

International Journal of Scientific Research and Engineering Development-– Volume 2 Issue 3, May 2019
Available at www.ijsred.com
ISSN : 2581-7175 ©IJSRED: All Rights are Reserved Page 501
A Survey: Issues And Challenges Of VOIP traffic
Over WiMAX
Anurag Saini1
, Avnish Kansal2
, Jasmeet Singh3
Department of Electrical & Electronics Engineering
2, 3
Department of Computer Science Engineering
Calorx Teachers' University
Ahmedabad, India
Abstract:
With the introduction of different wireless communication
technologies, the demand of users for real time services like voice
along with data called VoIP is increasing tremendously. In order
to provide high quality services, the new technologies like
WiMAX has to consider all the quality parameters attached with
a service. The paper thus discusses the WiMAX support to such
services along with all the possible features of VoIP, its quality
parameters like delay, jitter, throughput, etc. and emphasis on
what work has been done in this perspective so as to improve the
quality of the VoIP traffic by highlighting the various aspects,
methodology and limitations of previous work in this field. It has
been shown that, there still remain some issues and
considerations, which could be taken into account so as to have
high quality VoIP traffics.
Keywords— Quality of Service; VoIP issues; WiMAX
I. INTRODUCTION
According to the need of modern telecommunication
system, wireless networking has become an essential part.
The demand of high speed data transfer with high quality
is being the leading factor for the evolution of technologies
like WiMAX and WLAN and is still increasing day by day
[1]. Therefore, new ways to improve quality and speed of
connectivity are being searched for. Moving towards the
fourth generation communication networks, integrated
networks are coming into operation. WiMAX (worldwide
interoperability for microwave access) is type of
broadband wireless access which offers the additional
functionality of portability, nomadicity and mobility.
WiMAX accommodates various applications such as Web
surfing and quicker file downloads, but also enables
several multimedia applications, namely real-time audio
and video streaming, multimedia conferencing, and
interactive gaming. WiMAX connections are also being
used for voice telephony using voice over Internet Protocol
(VoIP) technology [2]. More narrowly if we see that one of
the most desired requirements of today’s smartphone users
is the need of always on connection, means the voice
telephony along with the data. For example, one is
downloading a data file and a call comes in between, the
requirement is that the file downloading should not stop by
the call interference it should go parallel with the voice and
that too with high speed. The paper studies how such
technologies are able to support such requirements, and
what issues are there behind carrying such data.
Voice over IP is expected to be a low cost
communication medium. The voice codecs are big
constraints which influence the quality of the voice in a
high data rate communication network. Hence, before real
time deployment of VoIP, it is essential to evaluate the
voice performance over the wireless networks.
Thus, in the coming sections we have given the
in-depth state of art of VoIP traffic over the WiMAX
networks. Section II discusses the WiMAX technology,
VoIP and its architecture; section III gives the details of
the issues and challenges faced by the networks to carry
VoIP traffic. The related work on different quality
parameters of VoIP has been discussed in Section IV along
with the conclusion, describing what work could be done
in this area in section V.
II. WIMAX TECHNOLOGY
WiMAX is developed by the IEEE 802 group with the aim
of providing the users a high internet speed at far range
called ‘last mile’ coverage with an ‘always on
connectivity’. It provides voice telephony, along with
faster Web surfing and quicker file downloads and also
enables several multimedia all at the same time, such as
real-time audio and video streaming, VoIP, video
conferencing, and interactive gaming. It provides services
with high quality [3]. WiMAX offers a variety of
applications each with different traffic patterns and QoS
requirements which must be fulfilled to get high user
satisfaction. The QoS is a term that refers to the “collective
effect of service,” as perceived by the user or more
narrowly it refers to meeting certain requirements such as
throughput, packet error rate, delay, and jitter associated
with a given application [4].
RESEARCH ARTICLE OPEN ACCESS

WiMAX architecture consists of broadly three subsections
namely the Access Service Network, the Connectivity
Service Network and the IP network. They are shown in
Fig.1.
Fig.1 WiMAX Architecture.
Access Service Networks: It consists of number of
transceivers called as base stations BS and the SSs. The
communication is possible through the electromagnetic
waves called radio frequencies, on which the information
is carried, the BS are used for this purpose. The subscriber
stations are the devices through which the user access the
service, which could be a mobile phone, laptop, dongles
etc.
Connectivity Service Network: It is the core network
which is the basic wireless communication providing
network or the standard network 2G, where all the calls
procedures, requests, authentication, management and
billing are done.
IP Network: It is the one which connects the WiMAX
access network with the core network through internet
protocol which further enables the users to get connected
to the communication network along with the global
internet which provides them real time and non-real time
services.
In order to rapidly converge on a worldwide standard, a
number of versions are there in the IEEE 802.16 family
described in table 1 [5].
Table 1. WiMAX Specifications
Parameters 802.16 802.16 - 2004 802.16e - 2005
Frequency
Band
10GHz –
66GHz
2GHz – 11GHz 2GHz – 11GHz
for fixed, 2GHz
– 6GHz for
mobile
applications
Application Fixed LOS Fixed NLOS Fixed and
mobile NLOS
MAC Point to Point to Point to
Architecture multipoint,
Mesh
Architecture
multipoint,
Mesh
Architecture
multipoint,
Mesh
Architecture
Transmission
Scheme
Single Carrier
Only
Single Carrier,
256 OFDM or
2048 OFDM
Single Carrier,
256 OFDM or
Scalable
OFDM with
128,512,1024
or 2048
subcarriers
Modulation QPSK, 16
QAM, 64
QAM
QPSK, 16
QAM, 64
QAM
QPSK, 16
QAM, 64 QAM
Data Rate 32Mbps-
134.4Mbps
1Mbps-
75Mbps
1Mbps-75Mbps
Multiplexing Burst
TDM/TDMA
Burst
TDM/TDMA/
OFDMA
Burst
TDM/TDMA/
OFDMA
Channel
Bandwidth
20MHz,
25MHz,
28MHz
1.75MHz,
3.5MHz,
7MHz, 14MHz,
1.25MHz,
5MHz,
10MHz,
15MHz,
8.75MHz
1.75MHz,
3.5MHz,
7MHz, 14MHz,
1.25MHz,
5MHz,
10MHz,
15MHz,
8.75MHz
Duplexing TDD and
FDD
TDD and FDD TDD and FDD
A. VoIP
VoIP has been widely accepted for its cost effectiveness.
VoIP converts the analog voice signal from a telephone or
computer into a digital packetized signal that can be
transmitted over the internet. Before transmitting the
analog voice signals they are compressed and encoded into
digital voice streams with the help of codecs. VoIP system
is divided into three indispensable components, namely
codec, packetizer and the playout buffer [6]. The general
overview of VoIP architecture is shown as in Fig. 2.
Sender
Encoder Packetizer Network
PlayoutDepacketizerDecoder

Fig.2.VoIP Architecture.
The output digital voice streams are packed into constant-
bit-rate (CBR) voice packets using packetizer. A two way
conversation is very sensitive to packet loss, delay and
jitter. Hence a playout buffer must be used at the receiver
end to smooth the speech by eliminating the delay jitter.
Quality of noise sensitive VoIP is generally measured in
terms of jitter, MOS and packet end-to-end delay [7].
Perceived voice with zero jitter, high MOS and low packet
end-to-end delay is assumed to be the best. To obtain
digital format of the analog signal process is utilized which
is called encoding and the converse is called decoding and
both are performed by voice codecs. As bandwidth is
enormous concern, compression techniques are utilized to
reduce bandwidth consumption. But problem related by
using codecs is the overhead of algorithmic delay, thus
codec is assumed to provide good quality even after
compression with minimum delay. The different types of
codecs with their specifications are described in table 2.
Table 2. Various Codes and their specifications.
Codec Data
Rate
Datagram
Size
A/D
Conversion
Delay
Combined
Bandwidth
(Bidirectional)
G.711u 64.0
kbps
20 ms 1.0 ms 180.80 kbps
G.711a 64.0
kbps
20 ms 1.0 ms 180.80 kbps
G.729 8.0
kbps
20 ms 25.0 ms 68.80 kbps
G.723.1
(MPMLQ)
6.3
kbps
30 ms 67.5 ms 47.80 kbps
G.723.1
(ACELP)
5.3
kbps
30 ms 67.5 ms 45.80 kbps
III. ISSUES AND CHALLENGES OF VOIP
A. Bandwidth:
Bandwidth allocation is a major concern in VoIP as when
it is split off between voice and computer data, it is
necessary for a network to allocate bandwidth for the voice
communication.
B. Securing:
Another big challenge for the VoIP is to secure the voice
communication so that it cannot be overheard or blocked.
The transmission is protected can be guaranteed by using a
double encryption process which uses X.509 for
authentication and 152-bit AES, 3DES or 56-bit DES for
the data flow.
C. Link Failure:
Link failure is the loss of the series of packets during a
period of time, which continue for the few minutes or
seconds resulting into delay after re-establishing the link.
Link failure can occur due to the various reasons. It can
caused by a problem in the equipment, a cable wire being
unplugged or disconnected, change in the configuration of
the transport network or denial of service attack. Routers
are capable enough in detecting a link failure and finding
an alternative route.
D. Packet loss:
Bandwidth limitation and the way the packets move inside
the network are the reasons for the packet loss on the
network. When the IP packets are introduced via a router
over the server, few packets may get lost, resulting in a
period of muteness in the conversation as the multiple
packets containing IP address would move in one direction
or in another direction which leads to clipped-speech effect
that is not satisfactory and acceptable by the users.
E. Network Design:
Network design affects the reliability of the VoIP
communications. If the network is not designed to manage
the combined transmission of voice and data packets, then
the reliability of the system may get ruined. Moreover, the
network design should support the new updated voice
applications which are only possible by using the novel
converged voice and data network.
F. Noise:
As voice communication is sensitive to noise. So, noise is
the main cause due to which the signal reaches the
destination with a lead or lag in the specific time period.
The deviation in the signal characteristics is called jitter.
The lead and lag both degrades the voice quality as lead
causes the negative jitter and lag is responsible for the
positive jitter. Total time taken by the packet to reach the
destination from the source is called packet end-to-end
delay. This delay should be minimum for voice
communication. Perceived voice quality is normally
measured using an arithmetic average of opinion score
defined as mean opinion score (MOS). MOS of a specific
codec is the standard mark provided by a panel of auditors
paying attention to distinct recorded samples [8]. This will
range from 1 (unacceptable) to 5 (excellent) described in

table 3. It will depend on delay and packet loss by the
network.
Table 3. Mean Opinion Score
Quality of Scale Range Listening Effort
Excellent 5 No effort required
Good 4 No appreciable effort
Required
Fair 3 Moderate Effort Required
Poor 2 Considerable effort
Required
Bad 1 No meaning understood
with Reasonable effort
IV. LITERATURE SURVEY
N.Nagarajan et al., (2009) defines the problem generates in
the process of video conferencing and in various
multimedia applications while delivering the video packets
over the WiMAX. A model is formed using a different
competing traffic sources over a point-to-multipoint
topology. To handle the VoIP and flow of video traffic the
performance investigation on the capacity of the WiMAX
device was carried out. The parameters which describe the
QoS such as throughput, average delay and jitter or the
packet loss is examined for the multiple types of service
flows that are defined in the WiMAX systems [9].
Bong-Ho Kim et al., (2009) focuses on a system
which evaluates the performance and capacity of the
mobile WiMAX systems. For the performance evaluation
all the factors from the air link to the application are
necessary. At the beginning they describe a mobile
WiMAX systems briefly which specifically talks about the
OFDMA/TDD systems of IEEE 802.16e and thereafter it
explains subscriber and application profiles which contain
data session attempts, traffic mix ratio, diurnal application
traffic distribution and model of application-traffic. Along
with this it also gives the simulation results of demand
estimation and characteristics of network-traffic.
Eventually, in the last portion it discusses the performance
evaluation of end to end application with the examples of
VoIP and also provides the performance enhancement
method for TCP/IP which can be realized in the mobile
WiMAX MAC or cross layer MAC/IP [10].
Jadhav, S. et al., (2011), Next generation wireless
networks put light on conjunction of numerous Radio
Access Technologies (RATs) supporting good Quality of
Service (QoS) for the various applications like Voice over
IP traffic (VoIP), video conferencing and video streaming.
The voice applications are rising quickly due to their rapid
increase in its popularity and cost. It is very much essential
to develop a suitable model for QoS to accommodate the
demand of giving VoIP service of high quality at any time,
at any cost and from anywhere. For supporting the VoIP
they organized the simulation study to find the quality of
service performance of WiMAX and UMTS. The
simulation results conclude that the WiMAX outperforms
over the UMTS with adequate margins and is suitable
technology, which can support applications of VoIP [11].
Henriques, J., et al., (2012) The momentous
increasing of VoIP have raised various challenges in the
deployment of WiMAX or Long Term Evolution (LTE),
well known as novel broadband wireless access networks
(BWA). To get the successful deployment the practical
assessment of voice traffic evaluating the performance and
quality is essential. In this paper, the capabilities of Mobile
WiMAX (IEEE 802.16e) which supports VoIP traffic
under distinct scenarios and engaging various Quality of
Service (QoS) service classes were implemented. Further,
the paper distinguishes the conditions of heterogeneity
access within a city area by examining both conditions:
Line of Sight (LOS) and Non-Line of Sight (NLOS). By
evaluating the end-user perceived quality (Quality of
Experience) and the network parameters of QoS, the
obtained results show the correct QoS service classes
management on the numerous well served users of VoIP
[12].
Baig, M.T. et al., (2013), this states that as the
users of real time applications such as VoIP is gaining so
much attention so it is essential to attain effective and
efficient service and mobile user must get the continuous
connectivity with parallel node. This paper illustrates the
influence of Vertical Handovers (VHOs) on the
functioning of VoIP. The Results conclude that the existing
protocols such as Realtime Transport Protocol (RTP) and
TCP Friendly Rate Control (TFRC) do not support the
Quality of Service (QoS) requirements during VHOs. A
scheme Adaptive Vertical Handover Rate Control
(AVHRC) is proposed which gets the VoIP’s QoS
requirements during vertical handovers. AVHRC provide
an effective rate control mechanism and obtains the data of
upcoming possible access technology. It also evaluates the
link stability for AVHRC which supports the QoS
requirements. The result summarizes that AVHRC shows
the improvement in terms of packet loss, latency and
throughput as compared to RTP and TFRC for the distinct
mobile scenarios [13].
Ben Salem, A. et al., (2014) Long Term Evolution
which is a latest wireless standard identified by the 3GPP
uses the Voice over Internet Protocol to broadcast the

voice services and data packets. In LTE technology the
traffic scheduling plays a crucial role by authorizing the
shared resources to the users in the most powerful manner.
The main aim of this research is to develop an effective
scheduling algorithm for this standard. It not only gets the
high performance of the system, but also capable of
maintaining fairness. Each user is assigned the requested
resource with respect to the already defined QoS
parameters. The simulation is carried out using MATLAB
and the performance of the scheduling algorithms on the
downlink side is evaluated [14].
V. CONCLUSION
From the exhaustive literature review, it has been
concluded that Mobile WiMAX can not only be used to
fulfill the demand for high internet speed, but can also be
used to provide voice over- IP services. The low-latency
design of mobile WiMAX makes it possible to deliver
VoIP services more effectively and VoIP technologies may
also be used to provide innovative services like voice
chatting, push-to-talk and multimedia chatting. Several
good works have discussed the capacity and performance
of WiMAX networks. But there appears a scope for a
comparative discussion of the performance of a WiMAX
network with respect to the application of VoIP.
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