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WIRELESS NETWORKS, CHALMERS 2011 2

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.

B. 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.

C. JitterThe 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]. Factorssuch as packet delay, jitter, packet loss and network

latency can noticeably affect the quality of UDP- based

TABLE I

QOS

Packet delay Packet loss Jitter

150 ms 1% 25 ms

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 codingand encoding purposes. Overall setup of the network

[6].

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 optional802.11 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 cantcommunicate until it didnt 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 CFPsend Contention-Free-Poll (CF-Poll) to station at time

one packet each The AP is coordinator PCF has a little

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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 basisof 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 itdifferent 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

[8].

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 weobserve 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. TheWEP 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.

III. VOIP OVER 3GTraditionally, real-time services (e.g. voice) are

transported over dedicated channels because of their

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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 protocolsare 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 of244 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 reliabilitymechanisms 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 otherhand, 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

session.

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 roleshort 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

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TABLE II

SIMULATION PARAMETERS

Parameter Value (Simulation 1) Value (Simulation 2)

Simulation runs 10000 6min30s

of speech

Load Variable One user

Channel error Variable Variable

probability

AMR source 12.2kbps 12.2kbps

data rate

AMR voice 20ms 20ms

frame duration

Call duration 120s 390s

Sil ence Voice on/off Sil ence Descri pt or (SID)

p er io ds ( mean d ur at io n 3 s) ( 16 0 ms i nt er val s)

AMR voice 244bits 244bits

packet payload

size

Protocol Stack RTP + UDP + IP v4 RTP + UDP + IP v6

size = 40 byte = 60 byte

Header Robust HC Robust HC

Compression (HC)RLC mode Unacknowledged Unacknowledged

Mode Mode

Maximum number 3 None

of MAC-hs

retransmissions

Number of 4 None

MAC-hs H- ARQ

parallel processes

Packet scheduling Round-Robin None

algorithm

Delay budget 100ms 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.

IV. VOIP OVER LTE

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

possible;

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

satisfactory. [16]

In wireless networks, like LTE, the principal cause

of issues is the path between the radio base-station and

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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 Eb/No over a low packet errorrate which is usually 103 for voice and 106 for

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

than 10 dB 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.

REFERENCES

[1] Renaud Cuny, Ari Lakaniemi, VoIP in 3G Networks: An End-

to-End Quality of Service Analysis, Nokia Research Center.

[2] INTERNATIONAL JOURNAL OF COMMUNICATION SYS-

TEMS Int. J. Commun. Syst. 2006; 19:491?508.

[3] http://uwanted.blogspot.com/2006/09/wireless-voip.html

[4] Recommendation of ITU-T G.114

[5] 1998 - 2011 Paessler AG.[6] www.advancedvoip.com

[7] IInt. J. Commun. Syst. 2006; 19:491?508 Published on-

line in Wiley InterScience (www.interscience.wiley.com). DOI:

10.1002/dac.801

[8] http://www.ozvoip.com/voip-codecs/

[9] http://www.connect802.com/voipbandwidth.php

[10] Voice over WLAN Campus Test Architecture Cisco

[11] Leonardo Ramon N. Sousa, Marcone L. Carvalho, Emanuel B.

Rodrigues, Leonardo Sampaio and Francisco R. P. Cavalcanti,

Quality of Service Evaluation of VoIP over HSDPA, Wireless

Telecommunications Research Group - GTEL, Department of

Teleinformatics Engineering - DETI, Federal University of

Ceara, 2006.

[12] 3GPP, Mandatory speech codec speech processing func-

tions; amr speech codec; error concealment of lost frames,

3rd GenerationPartnership Project, 1999. [Online]. Available:

http://www 3gpp org

[13] IETF Differentiated Services (DiffServ) Working Group,

http://www.ieft.org/html.charters/diffserv-charter.html.

[14] Robert Bestak, Performance Analysis of MAC-hs Protocol,

Czech Technical University in Prague, Department of Telecom-

munications Engineering, 2005.

[15] Matthias Malkowski, Andreas Kemper, Xiaohua Wang, Perfor-

mance of Scheduling Algorithms for HSDPA, CommunicationNetworks, RWTH Aachen University.

[16] Capacity Enhancement of VoIP over LTE by Stochastic Adap-

tive Modulation and Coding, K.Homayounfar and B. Rohani,

10-10 Cendex Center , Singapore.

[17] E. Dahlman. 3G Evolution: HSPA and LTE for Mobile Broad-

band. Academic Press, 2008.

[18] Kian Chung Beh, Angela Doufexi, Simon Armour, PER-

FORMANCE EVALUATION OF HYBRID ARQ SCHEMES

OF 3GPP LTE OFDMA SYSTEM, The 18th Annual IEEE

International Symposium on Personal, Indoor and Mobile Radio

Communications (PIMRC?07).

[19] D. Chase, Code combining; A maximum likelihood decoding

approach for combining an arbitrary number of noisy packets,

IEEE Trans. Commun., vol. 33, pp. 385 to 393, May 1985.[20] S. Kallel, Analysis of Type II Hybrid ARQ Schemes with code

combining, IEEE Trans. on Commun., vol. 38, No. 8, Aug.

1990.

[21] Kingsley Oteng-Amoako, Jinhong Yuan, Saeid Nooshabadi,

Selective Hybrid-ARQ turbo schemes with various Combining

methods in Fading Channels, Dept. of Electrical Eng. and

Telecomm, University of NSW, Sydney 2052, Australia.

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Group-13 Voice over IP - WLAN, 3G and LTE issues

Question:

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

Answer:

- Delay- Packet Loss- Jitter (Delay Variation)

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