22nd International Conference on Advanced Information Networking and Applications

Experimental Study of Voice over IP Services over

Broadband Wireless Networks
Edwin W.C. Peh Winston K.G. Seah, Y.H. Chew and Y. Ge
Department of Electrical & Computer Engineering Institute for Infocomm Research
National University of Singapore 21 Heng Mui Keng Terrace
Singapore 117576 Singapore 119613
[email protected] {winston,chewyh,geyu}@i2r.a-star.edu.sg

ABSTRACT interoperable broadband wireless connectivity to fixed, nomadic,

In this paper, a hybrid network consisting of both WiMAX and portable and mobile users. It has a range of up to 50km and allows
WiFi links is set up and used as a testbed for Voice-over-IP users to get broadband access without the need of direct line-of-
(VoIP) performance studies. Relevant metrics are defined and sight to the base station. It is capable of data rate up to 75Mbps.
used to analyse the performance of this hybrid network and its WiMAX can serve as a cheaper alternative to wired network
ability to support VoIP calls. End-to-end delay and packets loss infrastructure due to its long range transmission and large
are measured as a function of number of ongoing VoIP calls. bandwidth capabilities. Thus, WiMAX technology can be used
Based on these measured results, we propose a procedure to for the backbone network. In a typical home network, a WiFi
evaluate the maximum number of ongoing calls can be supported router that connects all the household’s computers can be
by the hybrid network while maintain the call quality at the connected to a WiMAX subscriber station (SS) via Ethernet to
desired level using the ITU-T specified E-model. The provide connectivity to the Internet. This SS is then connected
experimental studies show that this hybrid network is capable of wirelessly to a WiMAX base station (BS) some distance away.
supporting real-time VoIP calls, and that the network is able to Deploying a backbone network using BWA technologies can be
support up to 12 simultaneous G.711-based VoIP calls and more done in weeks, not months, with very limited right-of-way
than 20 simultaneous G.729-based VoIP calls. While it remains a installation requirements. Cost of equipment is reduced with
challenge to ascertain the exact reasons behind the low network significant advances in current BWA systems and the equipment
utilization, the proposed procedure for determining the maximum also provides an increased level of performance. Minimal expense
number of VoIP calls which meet the QoS requirements will be is needed for operators to enter a market quickly to provide a
useful and valuable to service providers with the intention to broadband connection. Furthermore, BWA systems offer security
provide VoIP services over such hybrid networks that interoperate protections such as password-protection and encryption of data.
these two technologies in the near future. There are no known incidents of BWA system security being
compromised. This essentially gives wireless ISPs and DSL/cable
Keywords providers quick entry into new markets and the opportunity to tap
WiMAX, WiFi, broadband wireless access, VoIP, experimental new sources of revenue [5].
performance study, E-model. In recent years, VoIP has emerged as one of the hottest
applications on the Internet. VoIP is a technology that routes
1. INTRODUCTION voice conversations using a broadband Internet connection instead
The IEEE802.11-based WiFi wireless technology [1][2] has of the regular analog phone line. VoIP utilises packet-switching
revolutionized the market for unlicensed client-access radios in a using IP instead of the conventional circuit-switching using
wide variety of applications. This technology is commonly used PSTN. One key benefit of VoIP is the great cost savings.
in wireless local area networks (WLAN). It is developed for use Therefore, this leads to the widespread implementation of VoIP.
in mobile computing devices such as laptops and PDAs. It enables When it first emerged, VoIP was a competitor to the POTS and
wireless access to Internet and supports applications such as VoIP mobile communications service providers. However, service
and online gaming. WiFi is primarily a short-range (typically less providers are now finding ways to earn revenue from this
than 250m) last-hop wireless access technology that requires the application and VoIP is fast becoming one of the mandatory
support of an infrastructure network like a wired backbone. applications to be supported.
Broadband wireless access (BWA) networks have the flexibility This paper presents an experimental study of VoIP performance
to scale and at the same time reduce the total cost of ownership over a hybrid network comprising a WiMAX backbone
compared to typical wired solutions using copper wires and optic infrastructure with WiFi access networks, built using commercial
fibres. Upon these benefits, BWA also provides equivalent off-the-shelf (COTS) equipment. No effort has been made to add
reliability and performance or even exceeding that of wired any adaptation or enhancements to the two protocol stacks to
solutions. As a result, service providers are able to offer further improve the interoperability between WiMAX and WiFi.
competitive services to leased lines, ISDN, DSL and cable An objective method of assessing and measuring the quality of a
modems with the use of unlicensed radio frequency spectrum. VoIP call on a BWA testbed is identified and applied, and the
performance and capabilities of VoIP over this hybrid BWA
The IEEE802.16 WiMAX standard [3][4] is the latest upcoming network infrastructure is assessed. Section 2 provides an overview
BWA technology for the metropolitan area that provides of related works and Section 3 describes the network

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DOI 10.1109/AINA.2008.65
configurations used in this study. Section 4 explains how a VoIP is the packet sending rate that matters and auto-rate adaptation
call quality is assessed, followed by the detailed procedures for does not always lead to capacity improvement when traffic is
running the performance tests Section V. Section VI presents and bursty. An observation is that packet jitter variations are
analyses the network performance results while Section VII significant in current IEEE802.11b networks and, RTS/CTS is not
presents and analyses the VoIP performance results. Lastly, helpful in improving the performance of real-time traffic.
Section VIII concludes this study and proposes some suggestions
for future works.

2. RELATED WORK
2.1 IEEE802.11 Performance Studies
Li, et al. [6] examined the interactions of the IEEE802.11
Medium Access Control (MAC) protocol and ad hoc forwarding
and the effect on capacity for several simple configurations and
traffic patterns. By measuring the throughputs of different packet
sizes, they showed that traffic pattern determines whether an ad
hoc network’s per node capacity will scale to large networks.
They discovered that IEEE802.11 is more efficient for orderly
local traffic patterns, such as a lattice network with only
horizontal flows. They also argued that the locality of traffic
determines whether large ad hoc networks are feasible. The less
local the traffic pattern is, the faster per node capacity degrades
with network size.
Figure 1. Overall Physical Hybrid Network Deployment.
2.2 IEEE802.16 Performance Studies
Hoymann [7] performed an analytical performance evaluation of
the IEEE802.16 standard by taking overall system performance 3. NETWORK DEPLOYMENT
measurements. Simulation results obtained from a software-based The overall physical hybrid network deployment of the
simulator were also compared with the numerical results obtained experimental setup, shown in Figure 1 above, consists of one
from an analytical model implemented in MATLAB. It has been 5.8GHz WiMAX network and two 2.4GHz WiFi networks with
shown that the IEEE 802.16 MAC overhead can be assumed to be all links operating at 54Mbps. L1 and L2 are laptops used to
10% and this can be reduced by optional fragmentation to fill up initiate the tests and collect experimental data for analysis. R1 and
the MAC frame optimally. Next, the simulation results showed R2 are IEEE802.11g routers used for setting up the two separate
maximum MAC throughput and packet delay values close to that WiFi networks. These WiFi networks represent the WLANs and
obtained by theoretical analysis. LANs within buildings. Next, SS1 and SS2 are the IEEE802.16
SSes which are linked to the IEEE802.16 BS in a Point-to-Multi-
2.3 IEEE802.11-IEEE802.16 Interoperability Point (PMP) mode to form a WiMAX network. This WiMAX
network represents the wireless backbone infrastructure. Next, R1
Berlemann, et al. [8] realized the coexistence and interworking of
and R2 are linked to SS1 and SS2 respectively using Ethernet
IEEE 802.16 and IEEE 802.11e. A central coordinating device
(UTP) cables, establishing the connections between the WiMAX
called Base Station Hybrid Coordinator (BSHC) supporting both
network and WiFi networks to create a hybrid network. The
the IEEE802.16 and IEEE802.11e protocols that operate in the
WiMAX network and WiFi networks are also tested separately to
same frequency band was implemented. The interworking of the
get additional results for comparison with that of the hybrid
two protocols is done by inserting the IEEE802.11 transmission
network. These results give a better understanding of how each of
sequences into the MAC frame structure of IEEE802.16. For a
the two networks performs individually and helps to diagnose any
transmission of a 512-byte data packet, the IEEE802.11e MAC
problems encountered by the hybrid network.
protocol has a lower efficiency than the IEEE802.16 protocol
which resulted in longer transmission time. It was also shown that
the overall system capacity based on the relay link throughput 4. VoIP CALL QUALITY ASSESSMENT
decreases as the number of QSTAs (QoS supporting 802.11e While a voice call involves bidirectional data flow, we first study
Stations) increases. a connection as a unidirectional flow that lasts for five minutes,
which is the length of a typical voice call [10]. The ITU-T
2.4 Multimedia Performance Studies specified E-Model [11] is adopted as the means for assessing the
Sun, et al. [9] evaluated the performance of video and voice VoIP call quality (cf: Section 4.1.) The E-model is applied to two
traffic through multi-hop wireless paths and studied the capacity voice codecs, viz., G.711 and G.729, to study their performance
of the mesh network using IEEE802.11b. They set up a UCSB in the hybrid network scenario. In the following, we addressed
MeshNet testbed and selected five nodes to conduct the how the various performance metrics should be maintained so that
performance study. The multimedia performance metrics used are voice quality is ensured.
packet latency, packet loss rate, inter-flow fairness and packet
4.1 The E-Model
jitter. They concluded that the number of hops in the transmission
The E-Model [11] is a computational model that uses the
path restrict the capacity of the network. Data rate and packet size
combined effects of several transmission parameters to predict the
do not influence the number of flows supported by the network. It

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subjective quality of a voice call. A transmission rating factor R is 5. PERFORMANCE TESTS PROCEDURES
calculated by combining all the transmission parameters relevant This section discusses the detailed procedures for both network
for a connection, which can then be used to predict subjective performance and VoIP performance tests. The parameters
user reactions. The factor R is made up of the approximate sum of involved and details on how each performance metric is measured
the impairments caused by these transmission parameters. The and obtained are also given.
mathematical representation of R is shown in the following
formula:
5.1 Network Performance Tests
R = Ro – Is – Id – Ie + A (1) The two network performance tests conducted are: ‘ping’ test
The basic signal-to-noise (SNR) ratio Ro takes into account the using the Fast Pinger tool [16] and UDP throughput test using the
effects of background noise and circuit noise. The simultaneous Iperf tool [17].
impairment factor Is is the sum of all impairments which may
occur more or less simultaneously with the voice transmission. 5.1.1 Ping Test
The delay impairment factor Id consists of all impairments due to The goal of using the Fast Pinger tool is to emulate the behaviour
delay of voice signals such as mouth-to-ear delay [12]. The of a specific codec to be tested. First, the payload size of the ping
equipment impairment factor Ie encompasses the distortion packet has to be determined. From Figure 2, it can be seen that in
impairment caused by low bit-rate codecs and packet loss [13]. addition to the ping payload size matching the voice payload size,
The advantage factor A measures the amount of decrease in the another 12 bytes is needed to represent the RTP header.
factor R that a user is willing to tolerate when using a given
technology over traditional wired telephony. For the purpose of
comparing with PSTN calls, the factor A is set to 0 [11]. The
value of R lies in the range 0 to 100, where R=0 represents an
extremely bad quality and R=100 represents a very high quality.
A traditional PSTN call has a minimum R-value of 70 and this
value serves as the lower limit for a VoIP call to be satisfactory Figure 2. Determination of Packet Size for Ping Test
and acceptable.
Each ping session is conducted for one Round Trip Time (RTT)
4.2 G.711 Codec measurement between the source and the destination. The ping
session runs for five minutes at an interval rate stated in Table 1
G.711 is the conventional codec used by a PSTN call. It uses
for a specific payload size of the codec. (The default parameter
Pulse Code Modulation (PCM) and operates at a high bit-rate of
values for the codecs are highlighted.) Three ping sessions are
64Kbps with a typical voice payload size of 160 bytes sent out at
conducted for each codec at three different payload sizes and the
intervals of 20 ms. The quantization distortion unit (qdu) that
corresponding interval timings. The average RTT measured for
arises from PCM in G.711 causes impairments and this brings the
each session is recorded and halved to obtain the end-to-end delay
R-value of G.711 down to its intrinsic R-value of 93.2, the highest
for that session.
among all other codecs [13]. This is different from the G.711
intrinsic R-value of 94.3 previously reported [14][15]. This is due Table 1. Parameters of G.711 Codec and G.729 Codec
to the revision of the ITU-T Recommendation G.107 in 2000 G.711 G.729
which resulted in the change of this intrinsic R-value for G.711. Bit-rate Payload Interval Bit-rate Payload Interval
(Kbps) (bytes) (ms) (Kbps) (bytes) (ms)
Thus, G.711 has an R-value budget of 23.2 for impairments
before the call quality drops below 70 and becomes unacceptable. 160 + 12 20 20 + 12 20
From Table I.3/G.113 in [13], G.711 has a tolerance of up to 64 240 + 12 30 8 40 +12 40
0.928% packet loss, which equates to an equipment impairment
320 + 12 40 60 + 12 60
factor Ie of 23.2.

4.3 G.729 Codec 5.1.2 UDP Throughput Test

G.729 is a bandwidth efficient codec commonly used for VoIP The Iperf tool is a TCP/UDP bandwidth measurement tool. It
applications. It uses Conjugate Structure Algebraic Code-Excited generates TCP/UDP packets for transmission over the network
Linear Prediction (CS-ACELP) and operates at a low bit-rate of from a client to a server. For each UDP session, UDP packets of
8Kbps with a typical voice payload size of 20 bytes sent out at 1470 bytes are generated and sent from the client to the server. A
intervals of 20 ms. In this case, the qdu is not applicable to low packet size of 1470 bytes allows the packet to fit into an Ethernet
bit-rate codecs. Instead, an equipment impairment factor Ie caused frame as the Ethernet link has a Maximum Transmission Unit
by low bit-rate is given to a specific codec under Table I.1/G.113 (MTU) of 1500 bytes. The UDP buffer size is fixed at the default
in [13]. This Ie value will be deducted from the intrinsic R-value 8KB while the UDP input load is varied from 5Mbps to 30Mbps
of G.711. From this table, G.729 has an Ie value of 11 which leads at intervals of 5Mbps. In total, six UDP sessions have been tested
to its intrinsic value of 82.2. with each session running for 300 seconds (5 minutes). The
Therefore, G.729 has an R-value budget of 12.2 for impairments average UDP throughput calculated at the end of each session is
before the call quality drops below 70 and becomes unacceptable. recorded and used for data analysis. The UDP packet loss statistic
From Table I.2/G.113 in [13], G.729 has a tolerance of up to is also captured to explain the variations in the UDP throughput
3.07% packet loss. for different UDP input loads.

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5.2 VoIP Performance Test End-to-end Delay (ms) against G.729 Payload (bytes)
The rtpplay tool in the set of RTP Tools [18] is used to run VoIP
calls from RTP data files generated from recorded voice 12
conversations for both G.711 and G.729 codecs. This rtpplay tool
10
is run from the laptop to initiate the VoIP call. The laptop at the

End-to-end Delay (ms)

other end will run Ethereal [19] to capture the RTP packet 8 Hybrid
streams. Each instance of the rtpplay tool represents one VoIP WiMAX
6
call. For each codec, ten sets of VoIP performance test runs are Wi-Fi

conducted. The first run starts off with two simultaneous VoIP 4

calls and the simultaneous VoIP calls increase by two for each 2
subsequent run. As a result, the last run ends with 20
simultaneous VoIP calls. Both the RTP end-to-end delay and RTP 0
160 240 320
packet loss are recorded for each run. The R-value for each run is
G.729 Payload (bytes)
then calculated and determined if each VoIP call is valid and
acceptable. Figure 4. End-to-end Delay of G.729 using Ping

6. NETWORK PERFORMANCE RESULTS

6.2 UDP Throughput Test Results
6.1 Ping Test Results The UDP throughput variation with the UDP input load is shown
The end-to-end delay measurements of the Ping test for each of in Figure 5 and the corresponding packet loss variation with the
the codecs evaluated over three different network configurations UDP input load over the three different network configuration is
are shown in Figure 3 and Figure 4. The end-to-end delay of ping reflected in Figure 6.
packets for the WiMAX network is approximately four times Looking at Figure 5 first, the maximum UDP throughput for the
higher than the end-to-end delay for the WiFi networks. hybrid network is around 9Mbps compared to 8Mbps and 7Mbps
for the WiMAX network and the WiFi networks respectively. It
It can also be observed that in most of the test cases, the sum of
was found that the UDP throughput in an IEEE802.11g WLAN
the end-to-end delays of both WiMAX and WiFi networks adds
without RTS/CTS was 13.5Mbps [20] which is twice more than
up to the end-to-end delay value of the hybrid network, with a
the 7Mbps achieved by the WiFi network as shown in Figure 5. It
deviation of about 0.5ms. This shows that the transmission and
is also noticeable that there is a dip in the WiFi curve between
handover of the ping packets across and between the WiFi and
UDP input loads of 15Mbps and 30Mbps. The same observation
End-to-end Delay (ms) against G.711 Payload (bytes) can be made for the WiMAX curve having a slight dip after UDP
input load of 5Mbps.
14

12 UDP Throughput (Mbps) against UDP Input Load (Mbps)

End-to-end Delay (ms)

10 10
Hybrid
8 9
WiMAX
8
UDP Throughput (Mbps)

6 Wi-Fi
7
4 6 Hybrid
5 WiMAX
2
4 Wi-Fi
0 3
160 240 320 2
G.711 Payload (bytes) 1
0
Figure 3. End-to-end Delay of G.711 using Ping 5 10 15 20 25 30
UDP Input Load (Mbps)

WiMAX links in the hybrid network incur minimum or no

Figure 5. UDP Throughput Measurements using Iperf
additional delays.
Next, the end-to-end delay measured for the hybrid network is at In order to explain the phenomenon of the dips, there is a need to
most 12ms for the G.711 codec with a voice payload size of 320 look at the UDP packet loss measurements in Figure 6. Taking the
bytes. An end-to-end delay of only 12ms is 12.5 times lesser than WiFi curve for illustration, the dip in the UDP throughput
the upper bound limit of 150ms for real-time applications as corresponds to an increase in the UDP packet loss. At a UDP
recommended by [12]. Therefore, as the hybrid network is able to input load of 30Mbps, the UDP packet loss drops and this results
meet the one-way delay requirement of 150ms or less, it is then in the increase in the UDP throughput of the WiFi network. For
capable of supporting real-time VoIP call transmission. In our the hybrid network, the UDP throughput saturates at around
study, we only focused on the default voice payload size of 160 9Mbps as the corresponding UDP packet loss increases above
bytes and 20 bytes for G.711 codec and G.729 codec respectively. 23% whereas for the WiMAX network, the UDP throughput
These two cases recorded a smaller end-to-end delay of 10.5ms saturates at around 8Mbps as the corresponding UDP packet loss
which is about 14 times less than 150ms. increases above 50%. When UDP input load increases above the
UDP throughput upper limit for a specific network deployment,

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 Packet Loss (%) against UDP Input Load (Mbps) 7.2 R-value Results
The R-value of the calls using the G.711 codec and G.729 codec
80
are shown in Figure 8 and Figure 9 respectively. From Figure 8,
70 we observe a 41% overall decrease in the R-value for calls using
60 the G.711 codec from 88.7 at two simultaneous calls down to 52.2
Packet Loss (%)

50
Hybrid
at 20 simultaneous calls. It can also be observed that the R-value
40 WiMAX for these calls drops below the acceptable lower limit of 70 as the
30
Wi-Fi number of simultaneous calls increase beyond 12. This indicates
20
that the hybrid network can only support up to 12 simultaneous
calls using G.711 codec.
10

0 Next, from Figure 9, there is an 11% overall decrease in the R-

5 10 15 20 25 30 value for calls using the G.729 codec from 82.08 at two
UDP Input Load (Mbps)
simultaneous calls down to 73 at 20 simultaneous calls. The
Figure 6. UDP Packet Loss Measurements using Iperf decrease in R-value from 79.32 at 18 simultaneous calls down to
73 at 20 simultaneous calls takes up a significant 7.7% of this
overall decrease. Although these calls are still acceptable with an
the UDP packet loss increases and becomes more significantly.
R-value of 73 at 20 simultaneous calls, the decrease in R-value is
Finally, from this UDP throughput test, it is proven that there is a
getting steeper. If more simultaneous calls are added, the R-value
direct relationship between the UDP throughput and the UDP
of the calls will soon drop below 70 and become unacceptable.
packet loss.
R-Value against No. of G.711 Calls
7. VOIP PERFORMANCE RESULTS
100

7.1 Simultaneous VoIP Calls 90

The packet losses of simultaneous VoIP calls for both G.711 and 80
G.729 codecs are shown in Figure 7. The packet loss is calculated Acceptable
70
R-Value

by averaging all the packet losses of the individual calls within Unacceptable
each set of simultaneous-calls test run. It can be observed that 60

G.729 codec is more robust to packet loss than the G.711 codec. 50

The increase in packet loss for G.729 calls is gradual at the start 40
and remains almost constant from 6 to 12 simultaneous calls.
30
Beyond that, the packet loss increases at a significantly faster rate. 2 4 6 8 10 12 14 16 18 20
This packet loss robustness is due to the in-built Packet Loss No. of G.711 Calls

Packet Loss (%) against No. of Calls

Figure 8. R-value of G.711 Calls
1.80
1.60 R-Value against No. of G.729 Calls

1.40
85
Packet Loss (%)

1.20
1.00
80
0.80
0.60 75
0.40 Acceptable
R-Value

0.20 70

0.00
Unacceptable
65
2 4 6 8 10 12 14 16 18 20
No. of Calls 60

G.711 G.729 55
2 4 6 8 10 12 14 16 18 20
Figure 7. Packet Loss of Simultaneous VoIP Calls No. of G.729 Calls

Scheme (PLC) scheme in the G.729 codec [21]. Figure 9. R-value of G.729 Calls

As for the calls using the G.711 codec, the packet loss is about From the VoIP performance test results, it is clear that the hybrid
15% to 110% more than those using the G.729 codec. The 110% network can support more calls using the G.729 codec than with
gap occurs at the point when there are 14 simultaneous calls; the the G.711 codec. Specifically, the hybrid network can support up
packet loss for the calls using the G.711 codec is doubled from to 12 simultaneous VoIP calls using the G.711 codec and 20
case of 12 simultaneous calls. This is the threshold point which (possibly more) simultaneous VoIP calls when the G.729 is used.
indicates that the hybrid network is unable to support more than
12 simultaneous calls when the G.711 codec is used.

838
8. CONCLUSION AND FUTURE WORK [7] C. Hoymann, “Analysis and Performance Evaluation of the
In this paper, we present an experimental study in which a hybrid OFDM-based Metropolitan Area Network IEEE 802.16,”
network consisting of one WiMAX and two WiFi links is set up Computer Networks, vol. 49, no. 3, pp. 341-363, Oct 2005.
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Traffic Performance in Mesh Networks,” Proceedings of the
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