a survey of voip over wireless lans · and multimedia (voice and data) transmission raised new...

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International Journal of Civil Engineering and Technology (IJCIET)

Volume 10, Issue 11, November 2019, pp. 214-228, Article ID: IJCIET_10_11_022

Available online at http://www.iaeme.com/ijciet/issues.asp?JType=IJCIET&VType=10&IType=11

ISSN Print: 0976-6308 and ISSN Online: 0976-6316

© IAEME Publication

A SURVEY OF VOIP OVER WIRELESS LANS

Alaa Alaa Abdulrahman

MSc. Project Management Engineering, University of Jordan, Amman, Jordan,

BSc. Computer Engineering, Al- Mansour University Gollage, Baghdad, Iraq

[email protected]

ABSTRACT

Despite of the benefits of VoIP technology, designers still facing some troubles

and challenges to provide the best VoIP quality of service (QoS). Wireless IP network

evolution was mainly developed for data traffic exchange. Using this network for VoIP

and Multimedia (Voice and Data) transmission raised new challenges to use a packet-

switching network for VoIP technology. Sometimes a delay of data transmission, such

as email and website surfing is acceptable for a delay time of ~ 1 sec. But such a delay

on voice and multimedia transmission can cause an insufficient call session, and can

fail the whole VoIP technology.

In this paper we surveyed five scenarios where designers and engineers

consolidate their effort to challenge the technology by providing scenarios that pave

the way for best QoS that make VoIP technology more usable and widely

commercially used.

Keywords: VoIP over LANs; QoS; SIP; H .323; VoLANs

Cite this Article: Alaa Alaa Abdulrahman, A Survey of VoIP over Wireless LANs.

International Journal of Civil Engineering and Technology 10(11), 2019, pp. 214-228.

http://www.iaeme.com/IJCIET/issues.asp?JType=IJCIET&VType=10&IType=11

1. INTRODUCTION

Voice over IP (VoIP) is a group of methodologies that enables us transmitting voice and

multimedia over (packet switching network), by using the internet protocol (IP) and a set of

the other protocols suits for this technology [1]. As on Fig. 1

VoIP is considered as an evolution for a telephony system, it shifted the traditional

analogue telephony communication to an IP associated standards that use the internet

bandwidth as a media to transmit the digital calls [11].

VoIP calls use a compression/decompression (Codec) technology. Calls are progressed by

a similar way of the digital contents transmission technique, in terms of signaling,

channelizing, encoding and decoding. It uses the internet network for deploying VoIP. It’s a

packet switching network based technology rather than a circuit switching one, the internet

media manipulation based on “Time Division Multiplexing “TDM. As on Fig.2

The evolution of this technology paves the way for speed voice and multimedia

communication used by “ Over-The-Top “ OTTs” applications such as (Skype, Hangout and

WhatsApp). This paper has eight sections. Discuss the VoIP network architecture, it consists

A Survey of VoIP over Wireless LANs


of three parties (call origination party, call termination Party and service provider) in section

II, then discuss VoIP protocols and codec in section III, then discuss VoWLAN and the

telecommunication evolution in section IV, after that we clarify some constrains that effect

the call quality in section V, then discuss six scenarios to enhance QoS in section VI, discuss

review of six scenario in VII, and finally discuss conclusion in section VIII.

Figure 1. VoIP session structure

Figure 2. VoIP Packetizing

2. VOIP NETWORK ARCHITECTURE

VoIP is used for communicating; it aims to interconnect long distant parties to exchange voice

and multimedia packets [1].

Figure 3. VoIP Network Architecture.



The VoIP network relies on packet switching network, that is used for the transmitting

data, by using (Internet Protocol) and other protocols more related to VoIP the voice and

multimedia session can be established, terminated and maintained. Beside the protocols

mechanisms, the VoIP network structure consists of group of entities that are essential for

session establishment. It consists of three parties (call origination party, call termination Party

and service provider) as on Fig.3.

2.1. Call Originator Party

It has the components that originate the call & the start the call establishment. The entities are:

The VoIP Customer Premises Equipment (CPE) such as: Analogue Telephone Adapter

(ATA), IP Phone, IP Soft phone, and call manager represent the main CPEs for the caller to

start the call session. These premises start encoding the analogue signal into the digital

packets. CPEs should exist in the call origination and termination parties.

Internet Gateway: such as ADSL modem, wireless router, and cable TV modem. These

component get the digital packets from the premises, and as what is happen for data traffic

exchange they deploy the needed protocols into the packetized voice to move them through

the WAN packet switching network internet gateways connect the LAN with WAN and they

should exist in the call origination and the termination parties.

2.2. Call Termination Party

It has the same premises which are exist on the Call origination party. The VoIP customer

premises and the internet gateways exists on this side and do a reversed function of the

origination party, as shown in Fig. 4 .

Figure 4: VoIP initiation and termination.

2.4. Service Provider

Service provider is connected to the WAN network, and it manages the session connection

and termination between the connected parties. Service provider gives the parties the ability to

interconnect On-net and Off-Net. For Off-net cases it interconnects the caller party to the

PSTN network (Circuit Switching Network).

3. VOIP PROTOCOLS AND CODEC

3.1. VoIP Protocols

VoIP calls based on packet switching networks, these networks require protocols for packets

acknowledgment [1].



VoIP Used protocols are:

H.323

Media Gateway Control Protocol (MGCP)

Session Initiation Protocol (SIP)

H.248 (Megaco)

Real-time Transport Protocol (RTP)

Real-time Transport Control Protocol (RTCP)

Secure Real-time Transport Protocol (SRTP)

Session Description Protocol (SDP)

Inter-Asterisk exchange (IAX)

Jingle XMPP VoIP extensions

Skype protocol

3.2. VoIP Codec

VoIP systems require a set of protocols that facilitate the call setup, call tear down, beside

voice/multimedia decoding and packetizing.

Manipulating the internet media requires one of codecs types that suit the broadband

transmitting such as narrowband, compressed speech [1]. Some popular codes are: G.711,

G.722 and G.729. as table 1

Table. 1. Comparison between the used Codec’s.

Paremeters Maximum Simultaneous VoIP nodes

G.711 G.723.1 G.729

Bit rate (Kbps) 64 6.3 8

Framing interval

(ms)

10 30 10

Payload (Bytes) 80 24 10

Packets/s,Np 100 33 100

4. VOWLAN ( VOIP OVER WIRELESS LAN ) AND THE

TELECOMMUNICATION EVOLUTION

VoIP is connected to the WAN through internet gateways which connect the LAN, which call

originator and termination parties exist, to the WAN. VoIP premises can be connected to the

internet gateways through a wire or wireless. The wireless LAN connection is established

based on IEEE 802.11b/c/g standards

VoIP usage progressed in line with the communication enhancement, starting from Dial-

UP internet connection, to ADSL and 3G/4G. VoIP become the main technology that OTTs

rely on their communication module for long distance connection and wireless connection.

H.323 is the major protocol for long distance and VoIP over Wireless LAN ( VoWLAN ),

IEEE 802.11 standard which has share the same methodology with 3G [9]. For design

considerations WLAN that support voice telephony should secure the call performance. This

includes voice gateways beside the Wi-Fi access. Especially that combine voice and data

degrades the call quality.



5. VOWLAN CONSTRAINTS

The evolution of VoIP over wireless networks accompanied with some constraints that affect

the call quality [6]. These constraints are:

5.1. Latency (Transmission Delay)

Transmitted packets over the Internet and IP network face latency with difference levels,

although this latency is acceptable for some services such as emails, and websites surfing, it

become more annoying on VoIP calls.

To make calls latency more acceptable the delay should be less than 150 milliseconds,

ITU recommend a maximum latency of 150 milliseconds.

5.2. Jitter (Clicks, Pauses and Intelligible Voice )

One of the major constraints that affect the call quality are Jitters, jitter can come as clicks,

pauses and intelligible voice, or latency variation.

Jitters buffers that delayed the voice packets and resend on an acceptable sequence.

Encoding and decoding can cause more latency too.

5.3. Poor Coverage

Poor coverage can be one of the main (or major) factors that affect the established calls;

coverage should be continuous, and consistent. Coverage provides the media that sessions

move one.

5.4. Security Issues

Security still one of the critical factors that affect the VoWLAN and wireless communication

too. Security encryption causes more latency as it adds more encoding to the coded signals

too.

5.5. Media Capacity

Transmission capacity act as one of the factors that determine the number of the concurrent

calls and available channels for media transmutation. The used codes and the transmission

capacity are two factors that impact the call quality. Most of the wireless LAN routers adopt

three or five channels for transmission; as more existing channels give more chance for

session establishment.

Figure 5. VoIP node distance from the access point.



6. VOWLAN PERFORMANCE ENHANCEMENT SCENARIOS

On this section the paper introduce six scenarios to enhance the Quality of service (QoS) and

override the constraints that may happen.

On each scenario a brief that and testing introduction included beside the results and

recommendation to deploy.

6.1. Scenario One: Codec type and Distance Scenario

This scenario figures out the impact of the VoIP performance according to the used codec and

the distance of the VoIP node as shown in Fig. 5 from the access point [5]. The result of this

scenario shows the impact on the following QoS parameters (packet Loss, throughputs and

jitter). Simulator and VoIP nodes CPEs are used on different distances to figure out the

impact.

Packet Loss

It’s a measurement for the number of lost packets not received on the final destination party

compared with the transmitted ones. As a result of the simulation scenario G7.11 suffers

more with the distance variation compared with the other two codec’s.

In the conclusion, the short frame interval of G.711 makes it more fragile for long distance

compared the long frame interval codec’s. High bit rate works better with long frame interval.

As a result using shorter time frame codec will enhance the QoS in case on long

transmission, so for long distance between the VoIP node and the wireless gateway used G

723.1 and G729 for better performance, as shown in the Figures 6,7,and 8.

Figure 6. G.711 and packet loss [5].

Figure 7. G.723.1 and packet loss [5].



Figure 8. G.729 and packet loss [5].

Throughputs

It’s the amount of data that is transmitted over a channel within a certain time, as a result of

simulation G.711 has the highest throughputs at different distances compared with the other

codec’s., as it has the longest time frame, while the other two packets are short.

Jitter

Is a delay on sent packets that can cause voice clicks and delay, as a result the smaller packet

size sent over a time cause more delay, so G.723 has the biggest jitter while the G.711 has the

less jitter.

6.2. Scenario Two: Codec and Concurrent calls through same Bandwidth

This scenario figures the VoIP performance considering the number of concurrent calls that

can use the WLAN channels [6,12]. Network congestion can cause data drop and lack of

throughputs. Jitter degradation become higher as reached packets arrive the destination of

different acceleration as shown in table 2. Note, number of clients represents the number of

concurrent calls.

Considering that WLAN routers use three or five wireless channels for data exchange

session beside different codec and the channel Capacity. This scenario was tested on different

distant levels.

For this test uses 5 VoIP nodes with different codes types and one wireless LAN five

channel routers. As a result G.723 shows the highest number of simultaneous nodes calls.

This shows that the packet size is inversely proportional with the number of conducted

concurrent calls as table 3.

Considering the impact of increasing the number VoIP concurrent calls On jitter and

Latency shows that the connection latency increases as a result of increasing the ongoing

concurrent calls, having a seven concurrent calls through the same Bandwidth cause a voice

degradation by increasing the latency from 2.6ms to 1.45second. Note: The optimal VoIP

latency for this case is 1 ms.

Table 2. Concurrent calls and impact on latency [6].

No of

Clients

Voice

Jitter

Packet

End to

End

Delay

Wireless

LAN

Delay

Voice

Traffic

send)

Voice

Traffic

Recived

Packet

loss

rate

6 0.002sec 1.05sec 0.89sec 5,004 4,798 4.12%

12 0.6 sec 7.0 sec 6.9 sec 35,246 28,562 18.96%

30 1.45 sec 30 sec 25 sec 51,734 36,688 29.08%



Table 3 Bandwidth compared with codec [6].

So as a result there is high increase in jitter and latency as more number of workstation is

added to the network.

6.3. Scenario 3: Coverage Scenario

In order to enhance the coverage of the WLAN to increase the number of ongoing concurrent

calls and shorten the distance between the VoIP node and the access points, we can increase

the number of used APs as shown in Fig. 9. ( WLAN nodes ) [4,3]. For this scenario we use

five WLAN routers ( AP ) and seven VoIP nodes The APs distribution should consider the distance between the adjacent points in order to avoid the channels

override and to reduce the packet loss.

Signal boosters and repeaters are used to strengthen the resend the sent packet which is

originated from the source as shown in Fig.10 , this way increases the throughput and

decreases the lost packets and reduces the latency.

Repeaters are useful for VoIP codec technology that relies on small sized codec such as

G.723 & G.729.

Figure 9 Adjacent Routers.

Figure 10. Wireless Network Environment.

Bit rate Maximum Simultaneous VoIP nodes

G.711 G.723.1 G.729

11 Mbps 5.0 14.8 4

5.5 Mbps 4.5 13.2 4

2 Mbps 3.0 10.5 2.8

1 Mbps 2.5 10.0 2.5



6.4. Scenario 4: Power Consumption

The network coverage acts as one of the factors to sustain healthy VoIP session performance.

Wireless coverage is linked to the power transmission that is needed to connect the VoIP

premises with the LAN internet gateway [7,8].

Telephony system needs a power to transmit the voice signal over the network. For

traditional telephony system which based on circuit switching the power comes from

analogue switches over the telephone cables. This power sustains the call session

establishment/termination going on. In the case of VoWLAN the power for the VoIP CPEs comes from two main sources:

Power Plug: that lightens the VoIP CPEs and enables the premises to transmit the wireless

signal to the Internet Gateway.

Internet Gateway Power: it’s called the PoE (power over Ethernet), which in the wired LAN

network comes from the LAN ports and runs through the RJ 45 cables.

In the case of wireless LANs the wireless router transmits a power to the connected

premises, this power secures the session going on, and help to establish the call initiation and

termination.

The wireless power connection is a subject for IEEE 802.11g. power dBs differs based on

the wireless router type.

The optimum transmission power for WLANs is 39 dBm, this is suited for all wireless

data transmission including the VoIP connection within the WLAN.

In term of IEEE 802.11 standards, VoIP session power consumption differs accordingly.

Based on the VoIP premise structure, power consumption depends on the codec type. As

the LAN gateways send work at standard power of 39 dBm, the other factors that affect the

quality of service is the codec power consumption. The codec acts as one of the factors that

used to design the WLAN in terms of the suitable power. Insufficient power supply means

lower VoIP performance.

To check the power consumption for 802.11 a/b/g. VoIP call sessions conducted for call

duration of 10 min and every IEEE standard scheme were deployed. As a result of testing,

802.11 g standard has the least power consumption because the lower operation frequency

among 802.11 a/b/g. as shown in Fig. 11.

802.11a has shorter transmission duration than 802.11b but higher operation frequency

than 802.11g. The higher frequency of 802.11a is the reason that why it consumes more

power than 802.11g as table.3.

Table 4 IEEE 802.11 a/b/g Standards and The Transmitted Power.

PHY Power (mW/min) Header Size (Msec)

802.11a 40 20

802.11g 39 20

802.11b 45 96,192



Figure 11. Codec and Power Consumption.

6.5. Scenario 5: VoIP Protocols SIP and H323

This scenario compares SIP and H.323 in terms of qualitative and descriptive comparison of

the two protocols. The main issues of comparisons are complexity, extensibility, and

scalability.

This scenario gives a brief idea about 2 protocols and creates a comparative literature of

those protocols on various classes. Actually, they put the description of call control services

[2].

This scenario consists of a WLAN network that has 10 nodes. Three of them are mobile

nodes. Simulation was run and the total simulation time taken is 134 seconds.

There are 3 different channels that have taken for this scenario. Two of them are used for

wireless LAN and last one for backbone.

The results of the simulation are collected and analyzed, the main parameters which

analyzed are: session establishment time, total bytes sent, received and overall throughput.

Session Establishment Time

The initiator session establishment time for SIP application is 125 seconds, whereas H.323

has required 131 seconds to establish the session.

Figure 12. Session Establishment Time on the Transmitter and receiver ends per Protocol [2].

In Fig. 12 shows the sender session establishment time for SIP better than H.323 but in

the receiver session establishment time for SIP and H.323 application is exactly same as the

initiator session establishment time.



Total Bytes Sent

SIP initiator node has sent more numbers of bytes compare to H.323 ones for the same

amount of time. on the receiver end the total bytes are reversed. As shown in Fig. 13. on the receiver end total numbers of send and received bytes are reversed too. As shown in Fig. 14.

Figure 13. Total Bytes on the Transmitter and receiver ends per Protocol [2].

Figure 14. Total Bytes on the Transmitter and receiver ends [2].

Overall Throughput

In terms of throughput for the comparison between the two protocols, the overall throughput

is slightly better for SIP application in same time duration. The maximum throughput is 6.182

Mbps at the duration of 123 ms duration for SIP and H.323 application respectively. As

shown in Fig. 15.

Figure 15. Throughputs per protocol [2].



6.6. Scenario 6: Proactive Codec

Correspondence has seen reliably expanding number of clients and more extensive

transmission capacity necessity of information and sound transmitting advancements that

have always decreased the accessibility of frequency spectrum. While arrangements have

been proposed as far as productive affirmation control and QoS implementation mechanisms

other than upgrades in call flagging protocols, codec adjustment calculations have

additionally seen the light the day. Different codec adjustment instruments have been

proposed till date to take care of issues emerging out of multi rate transmissions and in

addition to blockage issues on WLAN situations. In any case, keeping at the top of the

priority list the bottleneck in WLAN Access Points (Aps) Ongoing imperatives forced by

delay touchy correspondence [10].

In this manner legitimate arrangement of VoIP operators is a need for fruitful VoIP

sending in CRN and this incorporates watchful observing of codec parameters with execution

of versatile enhancement approaches. Keeping up QoS in VoIP has seen fast development

taking after fast organization of VoIP in different systems. Critical work has been done in the

field of cross layer enhancements and versatile methodologies including codec adjustment

algorithms, to give satisfactory QoS to VoIP calls.

Define the proactive codec as a way to select the suitable codec that matches the design

need. In terms, if power supply, distance and bandwidth. Proactive way is responsible to

select the right codec that works for the telecom environment that aims to override the

constraints and improve the QoS

As the result Proactive Codec Configuration in Wireless LAN that losses increments with

expansion in pps even in an uncongested situation as of now saw in . The misfortune is high

for codecs G.711, G.729. In addition to this, it is watched that G.711 brings about the most

elevated medium access delay of 15.445 ms contrasted with other codecs that experience a

latency of around 5 ms. This is as a result of the high bit rate and also the voice payload size

in G.711. Considering the noteworthy impacts of lining delay and transmission delay on the

aggregate end-to-end delay in VoIP correspondence, increment in medium access postpone

even in an un congest.

Table. 5. Parameters for Standard Codec [10].

Codecs Rate(Kbps) Voice Payload

Size (bytes)

Pps

G.711 64 160 50

G.723 6.3 24 33.3

G.729 8 20 50

Figure 16. Variation in packet loss for standard codecs [10].



7. LETRATURE REVIEW

Authors Title Work

M. Alias, O. Lee Loon [5]

S1

Performance of Voice

over IP (VoIP) over a

wireless LAN (WLAN)

for different

audio/voice codecs

it studies the impact of the distance

of the access point from the VoIP

nodes, it studies the impact of the

VoIP nodes that uses different kind

of codes, as a result the codec of

high bit rate is more sensitive for

long distances .

U. R. Alo, and Nweke Henry, [6],

S.Vijay Bhanu,

Dr.RM.Chandrasekaran and

Dr.V.Balakrishnan. [11]

S2

Investigating the

Performance of VOIP

over WLAN in Campus

Network [6], Effective

Bandwidth Utilization

in IEEE802.11 for

VOIP [13]

Study the impact of the concurrent

calls on the VoIP quality, it shows

that the codec’s with lower bit rate

shows better performance and less

packets collusion. On this case

G.723 & G.729 have the best

performance on this case

MONDAL, Amit, et al. [4]

S3

Improving VoIP

Quality for WiFi Users

This paper studies the impact of

network coverage on the VoIP

QoS, it suggest using more

coverage will enhance the QoS. It

figured out the impact with on

different VoIP nodes with different

codec’s.as a result it shows that the

Codec’s with lower bit rates work

better. So G.723 & G.729 work

better than G.711.

Kwan Hong Lee, Sung

Hyuck Lee, Wei Wu[7], M.

Naeem, V.Namboodiri,

R.Pendse [8].

S4

Power Saving in

Wireless VoIP [7],

Energy Implication of

Various VoIP Codecs

in Portable Devices [8]

These two papers studies the Code

power consumption based on the

IEEE 802.11 a/b/g standards. More

consumption happen by using

IEEE 802.11 b.

Sutanu Ghosh [2]

Comparative Study of

Various VoIP

Applications in

802.11A Wireless

Network Scenario

This paper discusses the impact of

the VoIP protocols on VoIP

quality and speed of session

establishment. It shows that SIP

has the highest session

establishment compared with

H323

T.Chakraborty, I. S. Misra

and S. K. Sanyal [10]

S6

Proactive QoS

Enhancement

Technique for Efficient

VoIP Performance over

Wireless LAN and

Cognitive Radio

Network

This paper discusses The proactive

codec way suggest a way to select

the suitable codec that fits the

available telecom environment in

order to override the VoWLAN

effect of: packet loss, delay.



8. CONCLUSION

In this paper, five scenarios studied to improve the voice performance over Internet Protocol

(VoIP), where each scenario uses a special method to improve the quality of service (QoS)

challenges in voice transmission or video. These challenges are: latency, jitter, and energy

consumption ... etc. We found some of the solutions to get rid of some of these challenges.

VoIP uses the Packet-Switching Network; its protocols based on packetizing the voice

signal and send them through the WAN. Same as how the data exchanged through the internet

network. VoIP sessions are based on Coding and Decoding the Voice signals, this makes

them adoptable for Packet-Switching network. SIP and H323 are the more common protocols

used for VoIP sessions. Using VoIP through Wireless LAN is a challenge for engineers to

develop the best VoIP solution that provides better performance and Quality of Services.

Codec: G.711, G.729 and G.723.1 encode and decode the voice signals at different Bit rates.

They respond differently to deviation of: distance, and wireless coverage & have different

power consumption. VoIP nodes that use G7.11 are more sensitive for the distance variation.

These nodes suffer more with far signals as less throughput, more packet loss, high latency

and jitter happen. G.723 shows the highest numbers of simultaneous nodes calls, the packet

size is inversely proportional with the number of conducted concurrent calls. Network

coverage is related to available number of channels that wireless Internet gateways can

provide. More adjacent channels enhance the VoIP performance and QoS. Two types of

power that role the VoIP nodes: the transmitting power which is the main source of the power

supply, and the received power from the Wireless LAN. In terms of power consumption

Codec’s varies in terms of power consumption G.711 has the highest power consumption.

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