Voice over IP - WLAN, 3G and LTE issues

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Voice over IP - WLAN,3G and LTE issues
Baran Kiziltan,Majid Khan and Francesco M.Velotti
—The aim of this paper is to give a basic
introduction on
VoIP- WLAN,3G and LTE issues
thoroughly describe QoS,problems in different wireless
network techniques and scenarios to get the best quality
in real-time.The attention is focused on the Quality of
Service in three different wireless networks,so,there will
be proposed some solutions to improve these issues.
Index Terms
—WLAN,3G,LTE,Voice over IP
OICE over Internet Protocol (VoIP),also known
as IP telephony or Internet telephony,is a set
of protocols to transport voice traffic over IP-based
packet-switched networks with acceptable quality of
service (QoS) and reasonable cost.
Wireless Local Area Networks (WLANs) have
become a part of everyday technology.This has now
been deployed around the world.Voice over WLAN
(VoWLAN) has been emerging as an infrastructure to
provide low-cost wireless voice services.However,since
the performance characteristics of wireless networks are
much worse than wire line counterparts,and the IEEE
802.11-based WLAN was not originally designed to
support delay-sensitive voice traffic.
Third generation (3G) packet switched
UMTS/WCDMA networks with High Speed Downlink
Packet Access (HSPDA) is being installed worldwide.
With the introduction of HSDPA in 3G networks,packed
switched wireless systems will allow dynamic resource
sharing therefore more efficient use of bandwidth and
improved network efficiency will be possible.Since
voice applications are real-time,they are intolerant
of lengthy delays,packet losses and jitter (delay
variation).All these problems degrade the quality of the
voice transmitted.These QoS issues over 3G wireless
networks will be assessed with respect to network load,
packet switching,buffer length and packet segmentation
under certain protocols,such as Adaptive Multi-Rate
Speech Codec,HC (Header Compression),RTP (Real
Time Protocol) and RLC (Radio Link Control Layer).
The challenges for achieving this include typical
VoIP related QoS (Quality of Service) problems,such
as delay,delay variation (i.e.jitter),packet loss and
additional overhead brought by the VoIP protocol stack
The QoS problems for VoIP over LTE will be
analyzed by comparing physical layer techniques and
try to obtain the best one in terms of VoIP quality,and
compared with some simulation or data.
When talking about quality on WLAN it is useful to
distinguish between scenarios.If your WLAN access
point keeps crashing then you could say QoS for you is
poor.To get best access point (AP),one can describe
the IEEE family 802.11 to find what technique should
be used.
The paper is organized as follow.Section II we focus
our attention on issues that afflict VoIP in WLAN
networks.Section III we consider QoS over 3G net-
works.Section IV we analyze QoS and physical layer
techniques to improve VoIP quality over LTE networks.
Finally,Section V we present our conclusion and possi-
ble future implementations.
The Wireless Local Area Network (WLAN) becomes
popular to support high-data-rate Internet access for
users in proximity of an access point (AP).The main
advantages of WLAN is simplicity,flexibility and
cost effectiveness.VoWLAN applications use the
infrastructure based on WLANs.There is a variety
of standards defined in the IEEE 802.1 [2].The most
deployed standard is 802.11b,whereas 802.11g is
receiving acceptance because of the high rate and
backward compatibility with 802.11b.
WLANs are only specified at the physical layer and
part of the data link layer.Reason why security and QoS
on WLAN is hard is because of all IP routing,session
control etc is outside the scope of WLAN,since both se-
curity and QoS are clearly needed end to end then higher
layer solution needs to interface o the WLAN capabilities
VoIP is real time applications and WLAN is not basically
Fig.1.VoIP over WLAN [3]
made for real time application because the QoS which is
big issue with WLAN is important and main portion of
VoIP applications.Challenges VoWLAN:there are many
challenge in VoWLAN like Quality of Services (QoS),
security etc QoS in VoWLAN consist of these three thing
which is discussed below.
A.Packet Loss
The total number of packet transmit over the network
is not receive to the end point or destination,so it means
the some data or packet loos or not received by the desti-
nation.There are two main sources of packet losses:one
is network packet losses,mainly due to network conges-
tion (router buffer overflow),link failures and rerouting,
transmission errors,etc;and the other is discarded packet
losses for packets experienced excessive delay.
The time taken by a packet to reach from a source
to destination,delay can be occurred from different
sources like delay at source,delay at receiver,delay
in network.Delay at source and receiver is due to
coding like changing analog to digital and digital to
analog and packetization,while network delay is due
to transmission,queuing and propagation.
The variation of time between packet transmit from
source to reach destination,means one packet reach in
100 ms and one reach in 125 ms to handle this problem
jet-buffer is used at receiving end and it has two type
static jet-buffer which hardware base and dynamic base
which is software base and can be handle by administra-
tor.but should take care about jet-buffer because some
time it is also becoming reason for delay like memory
over-flows etc.The following are the some measurement
and recommendation of ITU-T G.114 for a VoIP call for
the three attribute which is define in tab.I [5].“Factors
such as packet delay,jitter,packet loss and network
latency can noticeably affect the quality of UDP- based
Packet delay
Packet loss



services such as VoIP and video streaming.Contrary to
TCP-based services such as HTTP,SMTP,etc,a steady
stream of data packets is crucial for VoIP connections,
where even slight connectivity problems can cause noise
or echo”.“Quality of any service depends on the traffic
flow as well as the network of terminating partners.
Following are some issues should be considered to
provide better-quality service.Number of calls managed
simultaneously by the network The alternate way to
transfer the call to it desired destination in case of any
fault/failure occurred in the network CODECs for coding
and encoding purposes.Overall setup of the network”
D.Original IEEE 802.11 MAC layer
The original IEEE802.11 has no idea QoS especially
for voice data application have no sensitivity about Delay
jitter.The basic MAC layer use distributed coordina-
tion function (DCF) and Point coordination function
(PCF ) to share medium with station both have several
limitation [2].DCF relies on CSMA/CA and optional
802.11 RTS/CTS to share the medium between station.
The problem in DCF is that if many station want to
communicate at the same time there is always a collision
occur and it is based on collision avoidance means it has
to wait the medium to be free which produce delay and
if collision occur it is waste of the bandwidth and make
communication slow.some problem in DCF:

there is no QoS guaranty and priority between data
traffic like voice and data;

if a station sense medium and it is free and
get medium to communicate no other can’t
communicate until it didn’t let free the medium
if a station has slow bit rate it will capture the
medium for along time.
PCF is the other coordination which is define by ba-
sic IEEE802.11.It is optional.PCF is used only in
infrastructure mod in which all station are connected
by one center object called Access Point (AP).PCF
define two frame Contention Free Period (CFP) and
Contention Period (CP).In the CP DCP is used.To
give the right of communicate over the medium the CFP
send Contention-Free-Poll (CF-Poll) to station at time
one packet each.The AP is coordinator.PCF has a little
bit QoS management but have no idea of the different
class of traffic.
E.IEEE 802.11e
This standard define enhancement in the original
IEEE802.11 Mac layer DCF and ECF with new coordi-
nation function Hybrid Coordination Function (HCF).It
proposed priority and class based traffic means the voice
and multimedia application data class will have high
priority during transmission compare with other data like
email data class in a shared wireless medium etc.There
are two method to access the channel to communicate
like original IEEE802.11 MAC.HCF Controlled Chan-
nel Access (HCCA) and Enhanced Distributed Channel
Access (EDCA)[6].With the EDCA the data which
have high priority will have have chance to send early
then the low priority data and station having EDCA
implemented will have to wait less to send data.It work
mostly like PCF.In PCF scenario the interval between
to beacon frame is divide into two period CFP and an
CP,the HCCA is allowing the CFP to initiated almost
any time during CP.This kind of CFP is called Access
Phase (CAP) in 802.11e.The AP will initiate CAP any
time which it want and can receive frame from other
contention-free manner.The CAC is a method which
will decide whether a new connection will be allow
to established or not,it will be decide on the basis
of capacity of WLAN means if the new connection is
allowed what will be MOS or quality of over all call
which would be specified.So the CAC will maintain the
over all quality of Voice of VoWLAN.For infrastructure
mode of VoWLAN the CAC can be implemented in AP.
Codec is used to convert voice signal to digitally encoded
version compress it on the sender end and then reverse
the processes on the receiver end.These codec are
standardized by International Telecommunication (ITU-
T).There are many codec technique which is used in
VoIP for Encoding and Decoding.Some coding and it
different result are mentioned in below table fig.2 which
has been calculated with different IEEE 802.11 standard
with sample period 20,and voice activity detection active
The above data int the table is calculated by [9]
Connect 802 VoIP Bandwidth Provisioning Calculator
:from the table we got different result from different
code which different bit rate per kbps and with have
different WLAN IEEE802.11 like a,b,g and got different
MOS and found how much simultaneously connection or
calls can be established at time on per AP.Among we
observe that Codec G.711 have high MOS rate and with
reasonable simultaneously calls at time.But it should
Fig.2.VoIP over WLAN
be care about the bandwidth of connection is ok for
requirement of codec selected like G.711 require at least
128 bit for both way communication.MOS is Inter-
national Telecommunications Union Telecommunication
Standardization sector (ITU-T) approved which gives
a numerical indication of the perceived quality of the
media received after being transmitted and eventually
compressed using codec.The WLAN are working on
radio wave which are open which can eavesdrops and
some one can manage to use it illegally like crack the
secret key.
F.IEEE 802.11i
The IEEE802.11e enhance the security issue of orig-
inal WLAN and put forward the WAP2,it using Ad-
vanced Encryption Standard (AES) block cipher.The
WEP and WAP were using RC4 stream cipher.The
IEEE802.11e replace the issue of Authentication and
privacy issue with more detail and security adjustment
[9].Different VLAN can be used to separate Voice traffic
and data traffic:it will solve the space problem and
voice device can be protected from external network.
Separate VLAN will have private addresses which will
hide phone device from directly connected to public
network;QoS trust boundary extension to voice devices-
QoS trust boundaries can be extended to voice devices
without extending these trust boundaries and,in turn,
QoS features to PCs and other data devices;protection
from malicious network attacks-Subnet access control,
can provide protection for voice devices from malicious
internal and external network attacks such as worms,
denial of service (DoS) attacks,and attempts by data
devices to gain access to priority queues;ease of man-
agement and configuration-Separate VLANs for voice
and data devices at the access layer provide ease of
management and simplified QoS configuration.
Traditionally,real-time services (e.g.voice) are
transported over dedicated channels because of their
Fig.3.Transport of speech in IP
delay sensitivity while data is transported over shared
channels because of its transmitted in short,uneven
spurts.In order to carry voice on IP networks,
appropriate protocols must be used.The main protocols
are Real Time Protocol (RTP),User Datagram Protocol
(UDP) and Internet Protocol (IP) [11].In Fig.1,the
voice frames are generated in the application layer,
encoded and encapsulated within payload of an RTP
SDU.The RTP PDU is encapsulated into an UDP SDU,
which is delivered to the IP layer.
Adaptive Multi-Rate Speech Codec
(AMR) is a codec
with 8 narrow-band speech encoding modes with bit
rates between 4.75 and 12.2 kbps.If the data rate
is 12.2 kbps,the AMR codec generates packets of
244 bits which represent voice frames of 20 ms [12].
Since the AMR codec encodes and decodes digital
speech data with an optimum power and bandwidth
consumption,the Internet Engineering Task Force
(IETF) has approved the RTP payload format for AMR.
Real Time Protocol
(RTP) is an end to end transport
protocol,used to transport multimedia traffic in IP
networks,supporting unicast and multicast traffic.In the
case of VoIP service,it is implemented together with
UDP/IP [11].Since RTP does not provide any reliability
mechanisms and other layers should be implemented.
AMR and RTP the main performance parameters for
VoIP quality that are described earlier,can be measured
by the RTP protocol.RTP
AMR and RTP The main performance parameters for
VoIP quality that are described earlier,can be measured
by the RTP protocol.
User Datagram Protocol
(UDP) as a transport layer
protocol for VoIP over Internet Protocol (IP),UDP is
used to avoid any retransmission delays.On the other
hand,it provides no reliability on datagram delivery.
The UDP header size is standardized in 8 bytes and 20
Fig.4.FER for several loads and channel error for Simulation 1[11]
bytes for IPv4 or 40 bytes for IPv6.
Header Compression
(HC) in 3G networks it is
important to use bandwidth efficiently.On the other
hand,large headers of the protocols used when voice
data is sent over the wireless network where a high bit
error rate (BER) due to fading and mobility is present.
Robust Header Compression (ROHC) protocol has been
developed for this problem.The effective compression
makes use of the fact that majority of the fields in
the combined IP,UDP and RTP header either remain
constant or introduce constant change throughout a
A.QoS Analysis
One main parameter for assessing packet loss is FER
(Frame Error Rate).Although packet loss is undesired
some loss can be tolerated since error-concealment
techniques can be used.Buffer length can also cause
packet loss due to discarding of delayed packets.On
the other hand buffer length also may also increase the
delay where for acceptable conversational quality,the
maximum end-to-end delay should be around 250-300
ms [13].Therefore buffer length takes an important role
short buffering time will risk buffer underflows causing
jitter,and long buffering time causes long delay and
buffer overflows.Too short buffering time may also
cause increased packet loss due to loss of segmented
packets.Simulation with parameters specified for 2 dif-
ferent simulations can be seen in tab.II.
From the first simulation it can be seen that for
different error probabilities,ranging from 1% to 10%,
packet loss is directly related to the load on the wireless
network.With the increase number of network users,
applied packet switching technique is not feasible.
Therefore it can be said that delay and delay jitter
mainly depends on both Round Robin switching
Value (Simulation 1)
Value (Simulation 2)
Simulation runs
of speech
One user
Channel error
AMR source
data rate
AMR voice
frame duration
Call duration
Voice on/off
Silence Descriptor (SID)
(mean duration 3s)
(160 ms intervals)
AMR voice
packet payload
Protocol Stack
= 40
= 60
Robust HC
Robust HC
Compression (HC)
RLC mode
Maximum number
of MAC-hs
Number of
parallel processes
Packet scheduling
Delay budget
Predefined jitter
buffer (FIFO algorithm)
Fig.5.Mean packet delay for several loads and channel for
Simulation 1[11]
technique and Hybrid-ARQ mechanism where the
main features of MAC-hs (Medium Access Control-
high speed) protocol of HSDPA are retransmission
of erroneous packets which is handled by H-ARQ
and sequential delivery of the packets to the upper
layer [14].This reasoning can also be seen in the PDF
of delay jitter for 5 fixed users on the network in figure 7.
In simulation 2,a predetermined buffer is imple-
mented;therefore average network delay is constant
for different error probabilities.On the other hand as
Fig.6.PDF of the mean packet delay jitter for several channel error
for Simulation 1[11]
Fig.7.Simulation Results for Simulation 2[1]
packet loss ratio increases on the wireless channel,total
packet loss rate increases.The reason for occurrence of
erroneous packets in loss-free simulation is due to the
packet segmentation at RLC (Radio Link Control Layer),
where packets larger than one TTI (Time Transfer Inter-
val) are segmented over several TTIs,introducing longer
transmission delays and packet drops.
There are two important conditions must be met to
ensure an adequate VoIP quality:
1) delay from sender to receiver must be as low as
2) packet loss must be between 1% to 3%.
So,in LTE,end-to-end Quality of Service is based
on two parameters that formalize these two conditions.
First,Layer 2 Packet Delay Budget is specified for
every connection and for every User Equipment (UE).
Second,Layer 2 Packet Loss Ratio is defined in order
to guarantee the above specification.Hence,if a VoIP
connection has a L2PDB of 100 ms and a L2PLR
of 2% it mens that the QoS level for a subscriber is
In wireless networks,like LTE,the principal cause
of issues is the path between the radio base-station and
the UE.In fact,there are many new physical layer
techniques made to try to avoid the bit errors and the
delay,for example:Hybrid automatic repeat request
(HARQ) or advanced channel coding.Given that LTE is
strongly dependent on HARQ,reducing the bit errors,
the delay over the connection link will be reduced as
LTE Hybrid ARQ
is a physical layer technique to
increase robustness against transmission errors,and to
increase capacity.It is part of the MAC layer but the
soft-combining operation is handled by the physical
layer.In this technique,the erroneously received packet
is stored in a buffer memory and later combined
with the retransmission to obtain a single,combined
packet which is more reliable than its constituents.
Decoding of the error-correcting code operates an the
combined signal.If the decoding fails,a retransmission
is requested.[17]
There are four kind of HARQ schemes,the first one is
and it is based on the use of Cyclic
Redundancy Check (CRC),the second one is
Type I CC
because it is the same of the first one,but it uses a
Chase combining
technique,the third one is
Type II Full
(Incremental Redundancy) and it gradually decreases
coding rate in each transmission by sending additional
redundancy bits [18] and the last one is
Type III Partial
and as the previous,it gradually decreases coding
rate by sending additional redundancy bits,but each bit
maintains self-decodability in each retransmission [18].
is a scheme that,when the receiver
finds an error,it discards erroneous packets and sends
a retransmission request to the transmitter.The entire
packet is retransmitted.The packets are combined based
on either the weighted SNR?s of individual bits,in
which case the technique is termed Chase combining
Type II Full IR
is a scheme,where retransmission
requests consist only of parity bits.The receiver
combines additional parity bits from retransmission with
bits of the first transmission resulting in lower rates,
before FEC decoding is attempted [20].
Type III Partial IR
is a schemes,in which individually
transmitted packets are self-decodable and each packet
differs in coded bits from the previous transmission.
In Type III ARQ,packets are only combined after
decoding has been attempted on the individual packet
Fig.8.Packet error ratio in function of SNR for a modulation
64QAM with gray coding
coding rate.[18]
Given that,we analyzed simulation results of [18]
showed in 8 and we can say that to achieve an acceptable
quality of service,based on our previous parameters,the
best choice it will be HARQ Type III Partial IR.
In this paper we focused on QoS issues of VoIP over
WLAN,3G and LTE and tried to analyze these issues
by comparing different studies and proposing new ideas
for future work.As a result,conclusion for this paper
can be discussed in 3 parts.
VoIP over WLAN is real time application and very
sensitive to delay,packet loss and jitter but on the other
hand security is also an issue.This is because WLAN is
mainly on physical and MAC layer where the security
is handled in the upper layer.However this kind of
problem can be overcome if IEEE802.11e standard
is implemented where “Call Admission Control” will
monitor the voice quality.In addition,by adding G.711
coding technique,high quality multiple simultaneous
calls will be possible.On the other hand for more secure
VoIP applications IEEE802.11i should be implemented
and appropriate coding technique such as G.711 should
be simulated for future work.
VoIP over 3G,end-to-end QoS analysis of two similar
simulations under same protocols shows that current 3G
networks offer an adequate level of quality for VoIP
services.However,to improve this some further analysis
can be carried out.

As the number of user increases packet switching in
HSPDA becomes more important.Therefore more
capable Expo-Linear packet switching technique
can be simulated.This technique calculates the user
priority not only based on ranking users according
to their instantaneous channel quality,relative to
their own average channel conditions but also the
delay bound.Therefore,it is able to meet the
different QoS requirements of real time users [15].

MAC-hs protocol enables retransmissions which
causes decrease on QoS.On the other hand intro-
ducing predetermined TTIs also causes segmenta-
tion problems related to RLC creating delay and
packet loss.Therefore an adaptive TTI and buffer-
ing should be simulated for future work.Only
then MAC-hs protocols retransmission can be fully
effective since RLC does not guarantee delivery.

Current Packet Loss Concealment techniques are
effective only for small numbers of consecutive lost
packets,for example a total of 20-30 milliseconds
of speech,and for low packet loss rates.Therefore
a further study on intelligent PLC where a learning
technique can overcome packet loss issues.
VoIP over LTE,QoS analysis is mainly based on variants
of H-ARQ to improve
over a low packet error
rate which is usually

for voice and

data transmissions.The trade-offs in this assessment
was between memory usage and SNR.Standard H-ARQ
needs almost no memory but provides very little SNR
improvement on the other hand Type-II Full Incremental
Redundancy requires high memory but provides more
improvement compared to the standard H-
ARQ.Therefore Type-III Partial IR where the retransmit-
ted packet can be chase combined with previous packets
to increase the diversity gain,is the main candidate for
future work.
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Voice over IP

WLAN, 3G and L
TE issues


What are the Quality of Service
for voice communications over different wireless technologies?





Packet Loss


Jitter (Delay Variation)