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Performance Analysis of VoIP Services over


WiFi-based systems
David Villacı́s, Freddy R. Acosta, Student Member, IEEE, and Román A. Lara Cueva, Member, IEEE
Department of Electrical and Electronic, Escuela Politécnica del Ejército
Quito-Ecuador, UIO 171-5-231B
Email: {fdvillacis, fracosta, ralara}@espe.edu.ec

Abstract—In this paper we present a performance analysis world, and are comparatively cheaper and more effectives than
about the capabilities of Alix board for handling VoIP over WiMAX systems.
WiFi based systems with an embedded Asterisk server. First,
we study the characteristics of all components of the Alix board Nowadays the IEEE 802.11 has to deal with Voice over
to determine the possible options for operating systems that can Internet Protocol (VoIP) technologies. Therefore a network
run within this board using an Asterisk server. In addition we that support VoIP services with acceptable quality in relation
analyse the performance of SIP and IAX protocols combined with to the higher number of users as possible is relevant. Currently,
GSM, G.711 U/A, Speex, and G.726 codecs in WiFi based systems
in conformance with IEEE 802.11b and IEEE 802.11g standards. there are several middlewares running over evaluation boards
Results were obtained using a SIPP’s tool which determines for satisfying phone calls, one of them is an Asterisk-based
the boundaries reached by the board respect SIP protocol. embedded system over Alix board. This board has low power
Performance analysis was tested on metrics related to CPU consumption, high flexibility to work and low cost, near to
percentage of use and RAM memory consumption depending 100 USD.
on which protocols and codecs were used. The main results
show the feasibility of implementing VoIP over small processors There are several models of Alix boards having similar
with low consumption of RAM memory. Finally, regarding CPU characteristics. In this work an Alix 2D2 board was used, this
use percentage the best performance in a real environment was has a 500 MHz processor, 256 MB of RAM, two USB ports,
achieved when we used IEEE 8011.b standard, with IAX protocol
and GSM codec, reaching up to 110 simultaneous phone calls with two miniPCI slots, one serial connector, two RJ45 jacks and
29% CPU resources and 9,4% Memory to process simultaneous one slot for a compact flash that works as a hard drive. Our
calls were achieved considering that phone calls have coexisted research group primarily has been using this board as a long
with other type of internet services. distance communication platform in conformance with IEEE
Index Terms—Alix, VoIP, WiFi, SIPP, Asterisk, IEEE 802.11, 802.11 standard.
IAX, SIP The operating system used in this work was Voyage, a low
level Linux, in where Asterisk is integrated to make calls
I. I NTRODUCTION between places on the same network. However, it is necessary
WiFi systems in conformance with IEEE 802.11 standard to determine how many calls could this board support, with
are growing every year, based on low cost, and simple this question solved, Alix Board could be considered as
structure. These systems have been deployed as an alternative an efficient way for making calls, since they are a cheap,
to give internet services on developing countries, essentially, expandable and a popular technology that integrates many
rural regions have been more attracted for deploying these features in a tiny board. Moreover, it is also necessary to
kind of systems where carriers do not have the interest to determine which protocol on VoIP calls works better with this
satisfy these services. For those reasons determining if several board: SIP or IAX protocols, and also to evaluate all the codecs
internet services such as phone calls can be run over these available on the default installation. Finally, a parameter to
WiFi systems, would be necessary and also attractive for future determine how much influence the http request has on the
research, specially dedicated in improving the capacity in WiFi performance is introduced. This new petition to the server
systems [1]–[6]. Another alternative applicable on developing gives us an idea of how much the processor, and calls placed
countries is IEEE 802.16 standard, widely known as WiMAX, are affected.
this system includes mandatory QoS guarantees for a wide The rest of the paper is organized as follows: Section 2
set of real-time applications, this is the case of phone calls presents a review of the related works and the background
[7]–[12], the main differences between WiFi and WiMAX in where studies related to VoIP in WiFi systems have been
is that WiFi systems have been deployed mostly around the determined. In Section 3 we present the experimental test
environment, set up and metrics to be analysed. Section 4
Manuscript received February 20, 2013; revised April 15, 2013; accepted deals with the summary of results obtained and their analysis,
May 22, 2013.
F.D. Villacis, F.R. Acosta and R.A. Lara are with the Department of Section 5 concludes the paper with future work directions.
Electrical and Electronic Engineering, Escuela Politécnica del Ejercito, Quito-
Ecuador, 171-5-231B e-mail: (see http://wicom.espe.edu.ec/contactos.html).
978-1-4799-0367-2/13/$31.00 c 2013 IEEE
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II. R ELATED W ORKS ports, one serial port, a slot for a Compact Flash board, which
The interest of measuring the capacity of WiFi systems works as a hard drive, 1 or 2 mini-PCI slots and , 2 USB ports.
based on IEEE 802.11 is always growing. We found some The power supply is 7-18V DC . With the mini-PCI and USB
interesting results related to determine the maximum number plugged, the power consumption is about 15W. It can operate
of simultaneous calls, in [13] for example an analysis of si- from 0 to 50 degrees Celsius. There are 10 models of Alix
multaneous VoIP calls capacity over an Ad-Hoc network using boards that are separated into four families: 1Dx, 2Dx, 3Dx
G.711, G.726, and G.729 codecs is presented , determining and 6Ex. Each one with different characteristics.
that the maximum number of phone calls is 7 operating at 11 About operating system, we found out voyage one was the
Mbps in up to four hops. only model to carry Asterisk as packaged when we test all
In [14] the QoS to support VoIP in wireless networks the operating systems listed on the seller’s website [20]. For
in conformance with IEEE 802.11 standard using G.711, this work, Voyage One 0.5.2 release was used. Voyage one
G.726, and G.729 codecs is analysed, the results analysis provides server software, with VoIP features through Asterisk
show how packet loss is the metric with major influence on and Zaptel, with virtual private networks VPNs.
quality degradation for the considered scenarios, but VoIP is One computer starts to run the SIPP tool to make a fixed
a critical delay service requiring fast routing algorithms, the number of simultaneous calls. Another computer with the
tests determine that G.711 supports up to 12 VoIP phone calls softphone, checks the quality of the call and determines if the
better than G.726, meanwhile G.729 could support until 10 Alix board can support the specific number of calls. Based
simultaneous phone calls. on this call from the softphone to the Asterisk’s information
An experimental measurements on a wireless network have number, we increase or decrease the number of calls on the
shown that throughput gain is obtained at a cost of delay, SIPP tool until an appropriate number of simultaneous calls
setting a maximum of up to 6 pairs of VoIP calls [15]. can be processed. On the same computer, we connected a serial
Experimental study conducted with an increasing number of cable in order to monitor the %CPU and RAM used by the
VoIP users, increase the level of traffic up to saturation of the Alix board.
infrastructure network and these samples were obtained as a When performing the experiments, we divided them into
result showing that the network is capable of supporting up to two groups. The first group consist of VoIP over IEEE 802.11b
16 VoIP stations (STAs), where the Access Point (AP) acts as standard and the second one by using IEEE 802.11g standard.
a bottleneck for all traffic destined for wireless STAs. It was
also determined that large peak loads occur during periods of A. First Group - IEEE 802.11b
double-talk effectively reducing the capacity of the network 1) First Stage - SIP protocol: The first stage was based
[16]. on SIP protocol implementation over WiFi-based system in
In [17], an study to determine the performance of a board conformance with IEEE 802.11b, measuring the percentage
similar to the Alix board was conducted, it proposed the of CPU and memory use. The program SIPP generated traffic
development of middleware software and hardware that allows based on the SIP protocol. It manages the characteristics with
a version of Asterisk 1.6 to run on a modified version of the the following command: sipp -s 4444 -sn uac -i 192.168.2.3
Blackfin IPx architecture allowing wideband audio telephony 192.168.2.10 -d 150000 -r 4 -rp 2 -t un -l 10. Where -s
and sampling rates of 16 KHz using Analog Telephone Adap- extension, -sn create scenarios, -d Length of call, -l maximum
tors (ATA) devices. The ATA device was tested with Asterisk limit call, -xf load extensions .xml.
giving us an idea of how much benchmarking the CPU does, • The first test performed was with or without http request
but does not determine the maximum number of phone calls. via web browser at a rate of 1 petition per second.
At present there are not reference studies to determine dif- • The second test consisted of measuring one call between
ferences between IEEE 802.11b and IEEE 802.11g standards X-pro softphone with SIP protocol. In this experiment,
running SIP or IAX protocols, and the http request. we configured the same codecs at both ends.
An on-line calculator can be found on the web [18] where • On the third test, we worked on different codecs at each
we can set parameters as standard wireless, however, since not end.
all of the free codecs are available, it only gives us an idea of • The fourth test contrasts the results of the second experi-
how much a network can support. ment. We decided to use the same softphone, but this time
dialling to extension 500 on the Asterisk server (extension
III. E XPERIMENTAL T EST ENVIRONMENT 500 is the number where the server provides information
The Alix family is very broad, containing approximately about itself).
ten models, however, the main models are 2Dx and 3Dx. 2) Second Stage - IAX protocol: This stage was based
Both have the capability to work with two miniPCI’s on the on IAX protocol implementation over WiFi-based system in
same board. 2Dx is designed for indoor use, while 3Dx can conformance with IEEE 802.11b, measuring the percentage of
be placed outdoors because it has the appropriate accessories CPU and memory use on each.
[19]. • Test with an http request (rate 1 petition each 5-10
The Alix board has an embedded system with a 500 MHz seconds).
processor (64 KB Instruction cache, 64 KB L1 data cache and • Call between softphones with the same codecs at both
128 KB L2), 256 MB RAM and 400 MHz, 2 or 3 Ethernet ends.
3
TABLE II
SIP P ROTOCOL WITH ACTIVE CHANNELS REQUEST EACH SECOND TO A
• Work with different codec at both ends. GUI A STERISK VIA WEB BROWSER
• Make a call between the softphone and extension 500. Codec Simultaneous %CPU %Memory
Calls Min Max
G.711 u/a 16 15.3 28.1 6.9
B. Second Group - IEEE 802.11g GSM 10 5.95 9.12 9.33
On the second group IEEE 802.11g we developed two tests:
one with SIP protocol without http request, and another with TABLE III
SIP protocol with an http request at a rate of 1 petition per S OFTPHONE TO EXTENSION 500 (SIP PROTOCOL)
second. We did not develop a test for IAX due to the lack of Codec %CPU %Memory
a call simulator. However, this can be developed with more Min Max
people and computers dialing at the same time. G.711 u/a 1.3 2.3 6.2
GSM 0.3 1.3 6.2
Concerning to SIP protocol, in order to generate call traffic Speex 0.7 1.7 6.1
with SIPP, it is necessary to create an extension in the file (Q=3,C=2)
/etc/asterisk/extensions.conf to allow recordings that come pre- Speex 0.3 1.7 6.2
(Q=0,C=1)
loaded on the system to respond to a call for an average call Speex 1 2 6.2
duration (ACD) of 2:30 minutes. (Q=10,C=10)
When wondering about the quality of calls, another user G.729 1 2.3 6.2
was created. It works in parallel to the calls generated by the GSM-G.711-u 5.6 8 6.2
Speex 41.3 48.8 6.1
SIPP. This call was measured by the perception of the quality (Q=3,C=2)-GSM
when making a call to extension 500 at the same time the Speex 37.3 44.9 6
SIPP generates the calls. With IAX protocol, we did not have (Q=3,C=2)-G.711-U
a call generator, we use the same methodology, making calls
with one computer and using a second computer to measure
the quality with a unique call to extension 500. On this test Table III shows the amount of %CPU and %RAM memory
with IAX protocol, we used a softphone that lets us dial calls used when one call is placed. When speex codec is used the
without restriction of the lines, like Zoiper. percentage of the processor in use increases significantly.
Meanwhile, a special situation is showed, the percentage of
RAM memory used does not present a relevant increment, it
IV. R ESULTS A NALYSIS
presents an average of 6,1% of use.
In this section we present and analyse the main results
obtained for each group. When we use X-PRO softphones with SIP protocol, the
results showed in Table IV, determine a little improve just in
A. First Group Analysis the case of using speex (Q=3, C=2) code, another interesting
1) First Stage - SIP protocol: In this stage when SIP result shows that when G.729 codec was used we do not have
protocol was implemented, Table I shows a comparison response from the asterisk server.
between simultaneous calls without http request on IEEE TABLE IV
802.11b. By looking at table I the maximum number of C ALL BETWEEN X-PRO SOFTPHONES (SIP PROTOCOL)
simultaneous calls reached are 18 with G.711 and 25 with Codec %CPU %Memory
GSM, if we focus on percentage CPU and memory, it is Min Max
possible to identify a poor use of the processor remaining at G.711 u/a 1.3 2.3 6.2
GSM 0.7 1.3 6.2
66% and 84% using G.711 and GSM, respectively, to work
Speex 36 39 6.1
on other activities. Also, the poor use of the RAM memory, (Q=3,C=2)
which is actually positive because of the small memory RAM Speex 29.3 30.3 6.2
this board has, average 7%. (Q=0,C=1)
Speex 89.1 96.2 6.2
(Q=10,C=10)
Table II shows simultaneous calls using SIP Protocol with G.729 No response from the
active channels requesting each second where the maximum asterisk server
number of simultaneous calls were reduced considerably
in GSM, the use %CPU is also reduced, but the %RAM
Memory was increased. In fig.1 we can observe that G.711 presents a better perfor-
mance in the test with http request each second, because GSM
TABLE I decrease 60% of the maximum amount of simultaneous phone
S IMULTANEOUS CALLS USING SIP P ROTOCOL WITHOUT HTTP REQUEST -
G.711/GSM CODECS
calls while GSM has a better performance without http request
each second where G.711 decreased 11.12% of the maximum
Codec Simultaneous %CPU %Memory
Calls Min Max amount of simultaneous phone calls.
G.711 u/a 18 28 34 7.1 2) Second Stage - IAX protocol: In this stage, when IAX
GSM 25 12 16 6.9 protocol was used with the IEEE 802.11b standard, we can
G.726 The codec is not functional with
any softphone codec determine according with the results showed in Table V, that
the maximum number of simultaneous calls were increased
4

25
Simultaneous phone calls with SIP protocol We also measure one call on IAX protocol, these results
G711
GSM
are shown in Tables VI and VII. In Table VI, we observe, that
the speex codec does not have a set of message to answer
20 and a high percentage of CPU was used in comparison with
G.711 and GSM in a SIP and IAX environment.
Simultaneous phone calls

15
TABLE VI
S OFTPHONE TO EXTENSION 500 (IAX PROTOCOL)
Codec %CPU %Memory
10
Min Max
G.711 u/a 1.7 5.3 6.2
GSM 0.7 1.7 6.2
Speex 36.5 39.9 6.1
5 (Q=3,C=2,BR=32000)
Speex 28.6 32.5 6.2
(Q=0,C=1,BR=32000)
0 Speex 91 99.9 6.3
test with http test without http
(Q=10,C=10,
BR=32000)
Speex 37.2 40 6
Fig. 1. Maximun amount of simultaneous phone calls with SIP protocol (Q=3,C=2,BR=8000)
Speex 37 40.7 6
(Q=-1,C=1,BR=8000)
Speex 37 40.3 6
(Q=10,C=10,BR=8000)
in comparison when SIP protocol was implemented, among 6
and 11 times, the %CPU presents an increment in mean of 5
and 4 times, by using G.711 and GSM codecs, respectively, In Table VII we can determine that the obtained data
meanwhile the percentage of memory does not increase is similar to the calls with the same codecs at both ends.
significantly. The speed codec shows lower percentage of use on the
processor with a maximum of 0.3%, as shown in Table III,
TABLE V
IAX P ROTOCOL WITH ACTIVE CHANNELS REQUEST EACH SECOND TO A the reprocessing increases the amount of CPU use, however,
GUI A STERISK VIA WEB BROWSER . when IAX protocol is used the reprocessing is smaller when
Codec Simultaneous %CPU %Memory SIP protocol is used.
Calls Min Max
G.711 u/a 66 98 99 8.8 TABLE VII
GSM 110 28.2 30.4 9.4 C ALL BETWEEN IAXCOMM SOFTPHONES (IAX PROTOCOL)
Codec %CPU %Memory
Min Max
G.711 u/a 0.3 1.3 6
In fig. 2 we can see the comparison between SIP and IAX GSM 0 0.7 6.1
protocols according to the maximum number of phone calls, Speex 0 0.7 6.1
%CPU and %Memory. Thet IAX protocol with GSM codec (Q=3,C=2,BR=32000)
Speex 0 0.3 6
has the best performance, according to 110 simultaneous calls (Q=-1,C=1,BR=32000)
reached, and a optimal %CPU and %Memory used. Speex 0 0.7 6.2
(Q=10,C=10
,BR=32000)
SIP and IAX protocol with http request GSM-G.711-u 6.3 9.3 6.2
120
G711 µ/a (SIP) Speex 34.3 38.8 6
GSM (SIP) (Q=3,C=2,BR=32000)-
G711 µ/a (IAX)
100
GSM (IAX) GSM
Speex 28.3 37.3 6.1
(Q=3,C=2,BR=32000)-
80
G.711-u
Speex 34 39 6
(Q=3,C=2,BR=8000)-
60 GSM
Speex 28.3 37 6
(Q=3,C=2,BR=8000)-
40 G.711

20

B. Second Group Analysis


0
Simultaneous calls MIN CPU (%) MAX CPU (%) In the second group of tests with the IEEE 802.11g standard,
we could determine that Asterisk server can run only when
SIP protocol was implemented, instead of IAX protocol which
Fig. 2. A comparison between SIP and IAX protocol with http request does not allow us to obtain results.
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In this experiment the http requests were not considered, calls with 29% CPU resources, and 9,4% Memory to process
Table VIII shows a full activity of the processor, reaching simultaneous calls. It optimizes the rate of 11 Mbps that
99.9%. In this experiment, the processor does not have IEEE 802.11b present, instead of 54 Mbps presented in IEEE
capacity for other activities due the high traffic generated by 802.11g, but it was not possibly to implement this protocol.
the calls. This is compared to the IEEE 802.11b standard For that reason SIP protocol must be implemented when IEEE
where we have 66% for deploying other activities. 802.11g is used, the same number of simultaneous phone calls
was obtained but with a considerably difference, the %CPU
TABLE VIII
SIP P ROTOCOL WITHOUT HTTP REQUEST. resources is used at limit with a 9,4% Memory to process
simultaneous calls.
Codec Simultaneous %CPU %Memory
Calls Min Max Each call consumes about 5 to 11 MHz depending on codec
G.711 u/a 46 92.8 99.9 8.5 and protocol implementation, Voyage 0.5.2 ONE is a system
GSM 131 93.1 99.9 12.5 that maintain the optimal level of RAM in all test less than
12%.
Nowadays, people and industries are migrating to VoIP
In fig. 3, we could appreciate that the CPU percentage,
technology in order to bring economic solutions related to
where the IEEE 802.11g standard almost reaches 100% CPU
each requirement. These requirements help to further research
use, while IEEE 802.11b standard left 66% of the CPU without
and develop tools to help those purposes, for that reasons
activity.
Alix board with Asterisk server embedded systems present
an effective alternative to solve communication problems in
Performance of the Alix board using SIP protocol
140 developing countries, mainly in rural zones.
G711/IEEE 802.11g
GSM/IEEE 802.11g Future work will include testing Voyage One 0.7.5 in order
G711/IEEE 802.11b
120 GSM/IEEE 802.11b to determine the behavior of RAM memory. Last year miniPCI
Wi-MAX appears, it would be interesting to implement this
100 module over Alix board and develop some experiments related
to VoIP to analyse the possibility of implementing this service
80
over WiMAX. Another idea is to work with the security
on the WiFi-based network to test the performance of the
60
cryptographic accelerator include in the features of the board.
40
VI. ACKNOWLEDGMENTS
20
The authors gratefully acknowledge the contribution of
Escuela Politecnica del Ejercito (ESPE) for the economical
0
Simultaneous calls MIN CPU (%) MAX CPU (%) Memory (%)
support in the development of this project through the Wireless
Communications Research Group (WiCOM).

Fig. 3. Benchmark between IEEE 802.11b and IEEE 802.11g at the Alix
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[9] Jadhav, S.; Haibo Zhang; Zhiyi Huang, Performance Evaluation of Román Lara received the Electronic and Telecom-
Quality of VoIP in WiMAX and UMTS, Parallel and Distributed munications Engineering Degree in 2001 from Es-
Computing, Applications and Technologies (PDCAT), 2011 12th In- cuela Politécnica Nacional (EPN), Ecuador, a master
ternational Conference on , vol., no., pp.375,380, 20-22 Oct. 2011 degree in Wireless Systems and Related Technolo-
[10] Bakar, R.; Ibrahim, M.; Ali, D.M., Performance measurement of VoIP gies in 2005 from Politecnico di Torino, Italy, and
over WiMAX 4G network, Signal Processing and its Applications a master degree in Telecommunication Networks for
(CSPA), 2012 IEEE 8th International Colloquium on , vol., no., Developing Countries in 2010 from Rey Juan Carlos
pp.539,544, 23-25 March 2012 University (URJC), Spain. In 2002, he joined the
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802.16j transparent mode , Communications Conference (COLCOM), Politécnica del Ejército (ESPE), and since 2005, he
2012 IEEE Colombian, vol., no., pp.1-6, 16-18 May 2012, ISBN 978- has been an Associate Professor at ESPE. His main
1-4673-1267-7 research interests include digital signal processing, smart cities, and wireless
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[19] http://mini-box.com
[20] http://linux.voyage.hk

David Villacis received the Electronic, Data and


Network Communications Engineering Degree in
2011 from Escuela Politécnica del Ejército (ESPE),
Ecuador. In 2010, he joined the Wireless Communi-
cations Research Group of the Electrical and Elec-
tronic Engineering Department of ESPE. In 2012, he
join to Weatherford ILL. His main research interests
include developing new wireless systems and the
benchmark for communications systems.

Freddy Acosta received the Electronic and


Telecommunications Engineering Degree in 2002
from Escuela Politécnica del Ejército (ESPE),
Ecuador, a master degree in Telecommunication
Networks for Developing Countries in 2012 from
Rey Juan Carlos University (URJC), Spain. In 2004,
he joined the Electrical Engineering Department of
ESPE, and since 2008, he has been an Associate Pro-
fessor at ESPE. His main research interests include
alternative energy systems for telecommunications,
wireless systems, VoIP systems and TDT systems.

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