IP Private Branch eXchange of Saint Joseph University, Macao: a Design Case Study

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(IJCSIS) International Journal of Computer Science and Information Security,
Vol. 9, No. 6, 2011

IP Private Branch eXchange of Saint Joseph
University, Macao: a Design Case Study

A. Cotão, R. Whitfield, J. Negreiros
Information Technology Department
University of Saint Joseph
Macau, China

Abstract— To present the specification project of a digital telephone systems because they can significantly lower
telephone system, IP PBX, for the new campus of Saint Joseph telecommunications costs (a significant operating cost) while
University (USJ), Macao, is the main goal of this research. Given business voice network should be integrated with the data one
that the new USJ campus at Green Island, Macao, was projected [3]. The required software will be installed on standard
for this coming September 2012 with the latest technologies
available to achieve energy savings and to contribute somehow to
computer hardware, according to the next items: (A)
the environment sustainability, the available prototype was Understanding of the general benefits and technical trends of
designed using VoIP (Voice Over IP) to follow this novelty trend. small and medium IP PBXs; (B) Configuration of an
It is expected to conclude, as well, that there is a financial reason appropriate hardware and software combination regarding IP
for preferring a VoIP phone system over a conventional one. PBXs; (C) Evaluation of the final prototype; (D) Design and
Choosing this platform eliminates the need for conventional specification proposal as regards a production system suitable
telephone wiring for the new campus, which represents a for being used at USJ new campus.
considerable cost savings and logistics, for instance. Further, Concerning the structure of this writing, section two
good VoIP Open Source software are already available such as summarizes the basic concepts and relationships of PSTN
AsteriskNOW©. At last, the internal USJ connection to the (Public Switch telephone Network), VoIP (Voice Over Internet
Catholic University of Lisbon, Portugal, adds another financial
Protocol), PBX (Private Branch eXchange) and IP PBX for
reason for this project since international calls can be quite
non-technological readers (but who like to understand it). This
expensive. To analyze the setup, test and implementation of a
prototype becomes, hence, the aim of this paper, including the
includes business managers that are not aware of the possibility
explanation of the difficulties, technical lessons and to lower communication costs within their organizations. The
recommendations on the tested IP PBX. following section will describe the specification and design for
the IP PBX server, the IP phones connected to it, the VoIP
Keywords- Voice Over IP (VoIP), Private Branch eXchange gateway to connect the IP PBX to the Public Switch Telephone
(PBX), AsteriskNOW©, PSTN, IP handsets, IP Providers. Network (PSTN) and other VoIP providers. In section four, the
setup and implementation are shown while section five
I. INTRODUCTION illustrates the testing phase, not forgetting the evaluation of its
efficiency and reliability. Inevitable, the last section
USJ is a joint venture result of the Catholic University of recommends the main technical specifications for an IP PBX
Portugal and the Diocese of Macao [1]. It was created in the for the future campus of USJ.
NAPE area of Macao but, with the expansion of new courses
and a significant increase of the student´s number, the II. PSTN, VOIP AND IP PBXS
available facilities rapidly became a serious concern, affecting
the future growth and the overall environment quality. With PSTN stands for Public Switched Telephone Network and is
also known, according to [4], as POTS (Plain Old Telephone
the aim to overcome this issue, a new campus is being
System). Born in 1876 with Alexander Graham Bell, it is a
developed and is under construction at Green Island, Macao.
circuit-switched network where telephones handsets are
This new campus has been designed to use all available new interconnected among themselves through single or multiple
technologies with the purpose of minimize the running costs hub exchanges (cross-connect switches). Including the mobile
as well as to maintain the future environment sustainability. It system, it is the main telecommunication network worldwide
is expected that the new technologies used will take full with 5.4 billion (800 million for fixed line and 4.6 billion for
advantage of solar power to heat, cool and natural lighting as mobile) subscribers.
well as rainwater collecting for re-use. At first, all phones were connected among themselves in a
Historically, some organizations and companies have used meshed network but, with the growing of users, this layout
separate telephone and computer data communications became impractical. Henceforth, a new layout (hierarchical and
networks [2]. VoIP combines both networks to greatly reduce star topologies) was developed to allow this steady growing.
capital and operating costs. USJ wants to adopt this approach Once this tie was established, the human voice is converted
and, thus, the IP PBX technical development becomes the into analogical form and sent through copper twisted pair
ambition. It is believed that VoIP is the future of corporate cables to the central office (CO), where it is converted to digital

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(IJCSIS) International Journal of Computer Science and Information Security,
Vol. 9, No. 6, 2011
signals at the Digital Cross-Connect Switches [5, 6]. These Once again, a key distinction between PSTN and Internet is
digital signals can take different paths depending on, whether it the destiny identification. Within the PSTN, lines, rather than
is a local, national or international call. If it is a local call devices, are identified, that is, your home phone number is the
connected to the same switch, the analogical signal is routed phone line to your place (in fact, you can attach different
directly to the destination number. Otherwise, the very first devices to this line such as faxes and answering machines). By
switch will convert the analogical signal to digital for routing contrast, the Internet identifies each specific device by their IP
afterwards. Later, it is converted back to an analogical form to address.
be sent to the destination call number. IP PBX stands for Internet Protocol Private Branch
VoIP stands for Voice over IP or Voice over Internet eXchange and, basically, is an IT framework which uses the
Protocol. It is another way to make phone calls because, instead common LAN (Local Area Network), Internet, PSTN and
of using the conventional PSTN network, it is done through the VoIP providers for communication purposes (see Figure 1).
Internet infrastructure. VoIP started in 1995 with Vocaltec©
when it released its first Internet phone software. It used the
H.323 protocol and run on any PC with the usual microphones
and speakers. Vocaltec© enjoyed an initial success but, due to
the lack of broadband, it did not survive for long. On 2003, a
major step was given by Skype©. Unlike the previous Internet
experiences, Skype© used its own protocol. Since it held a good
voice quality, it became a major commercial reference for
VoIP. In the first half of 2010, according to Skype© Web site,
users made a total of 6.4 billion calls to landlines and mobile
phones.
A VoIP call usually starts with a typical PC or with an
IP/analogue telephone as long as it is connected to an Analog
Telephone Adapter (ATA). The analog voice is converted to a
successive 0s and 1s and, afterwards, the digital signal is
broken into smaller chunks called packets to be sent to their
destination through the Internet [7, 8]. At the destination, the
process is reversed, the smaller packets are assembled back in a
properly order and converted to an analog signal that is Figure 1. IP PBX and Internet integration.
understandable by the human ear.
VoIP technology uses three protocols: SIP (Session From the view point of the end-user, there are no differences.
Initiation Protocol, an IP signaling procedure to establish, For the organizations, the change relies on the system setup
modify and terminate VoIP calls), RTP (Real-time Transport and phone calls since it is less costly. Basically, an IP PBX
Protocol, a standardized packet format for delivering consists of a server with special hardware for PSTN interfaces,
audio/video over the Internet) and RTCP to monitor SIP phones and VoIP Gateways. Indeed, SIP phone, VoIP
transmission statistics and Quality of Service (QoS) status. phone and IP phone are different names for the same device
The connection between VoIP and PSTN is done through a [10]. It is this device that connects the user with the IP PBX.
communication protocol named ENUM that stands for All the calls received are, then, routed to their destination
telephone numbering mapping [9]. Basically, ENUM translates automatically, according to a set of rules (the dial plan) that
telephone numbers into a format that can be used by the should be previously programmed.
Internet. Bear in mind that PSTN is a circuit switched network An IP PBX server operates similar to a proxy server, that is,
that uses telephone numbers while the Internet structure is a the SIP clients (soft phones or handset IP phones) register at it
packet switched network that uses Uniform Resource and when they wish to make a phone call, they request the IP
Identifiers (URIs) for addressing each device. This ENUM PBX to establish the connection. Internally, the IP PBX has a
protocol enables circuit switched traffic to be carried on a directory with all telephones/users with their corresponding SIP
packet switched network by matching a circuit address address. Therefore, it is possible to connect an internal call
(telephone number) to a network address (URI). Hence, ENUM (located in the same LAN) or route an external call through
links both PSTN and Internet, providing a means for Internet either a VoIP gateway or a VoIP service provider. The
connected phones receive/make calls to the PSTN network. connections with the external parties are done through SIP
Calls between subscribers of the same VoIP provider trunks and, optionally, via a VoIP Gateway, where the SIP
(VoipBuster© or Vonage©, for instance) are usually free. Calls Trunk bonds the IP PBX to an ITSP (Internet Telephony
between subscribers of different VoIP providers should also Service Provider). Peculiarly, the physically moving of a SIP
have no costs associated, as well. On the other hand, when the phone does not affect its relationship to the IP PBX.
calls are originated from VoIP providers and are terminated at
the PSTN, the costs involved equals the interconnection costs
charged by the VoIP provider for the gateway usage to connect III. SYSTEM SPECIFICATION AND DESIGN
the Internet network to the local PSTN. To minimize these According to Figure 2, the IP PBX prototype will run
interconnection charges, as expected, VoIP subscribers use the Asterisk© whose hardware equipments and software packages
Internet network to the nearest PSTN termination point. specifications are as follows: (A) Computer hardware (Intel

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(IJCSIS) International Journal of Computer Science and Information Security,
Vol. 9, No. 6, 2011
Pentium© Dual CPU, RAM of 3GB, NVIDIA© GeForce 9500
GT graphics card, Realtek© RTL8168C network card, 120GB
Maxtor© 6Y120PO ATA HD with a DVD drive); (B) Software
(AsteriskNOW© 1.5 based on Linux CentOS©, MySQL© and
Apache Web server); (C) Connections (local extensions for the
same IP PBX LAN, four remote extensions, links to Macao
PSTN through a VoIP gateway and, as expected, connection to
the Portugal PSTN network through four different VoIP
providers to minimize call costs).
In a more in depth view, the three local extensions will
follow the next pattern: 1001 (ATA Linksys SPA3102), 1002 Figure 3. On the left, the GXP2000© is an IP handset suitable for both small
and large business organizations and it can be connected directly to any LAN
(soft-phone CounterPath’s X-Lite©) and 1005 (Grandstream© (it handles up to four simultaneous VoIP calls). It has a dual 10M/100Mbps
GXP2000 IP Phone) for Macao calls. Regarding external ones, Ethernet port, an intuitive user interface, a large back-lit LCD display with
there will be one VoIP gateway to access the Macao PSTN multiple languages support and privacy protection. On the right, the Siemens©
landline, four VoIP providers to minimize international calls Gigaset C470 IP is a cordless IP phone which allows connecting to any IP
charges to Portuguese nomadic numbers, G9 Telecom© to PBX, via a LAN, as well as to a local PSTN.
receive and make phone calls from and to Portuguese nomadic
phone numbers, SMSDiscount© to connect to the Portuguese
PSTN land lines, VoIPCheap© to connect to the Macanese and
Hong Kong PSTN land lines and, at last, SmartVoIP© to
connect to the Portuguese mobile network (check figure
below)).

Figure 4. On the left, the IP soft-phone CounterPath’s X-Lite© can be used to
make and receive voice/video calls [11]. Its minimum specifications required
to connect and operate with the IP PBX are an audio codec G.711, SIP
protocol and ID/Voicemail caller. On the center, the ATA Linksys© SPA3102
is an adapter with the ability to connect analog telephones and fax machines to
the IP PBX through a computer data network. Curiously, it has also the ability
to bond to any local PSTN. On the right, the ATA Linksys© PAP2T allows the
link of one or two analog phones to the IP PBX in order to create one or two
extra extensions.

IV. SYSTEM SETUP AND IMPLEMENTATION
To start, the Asterisk©NOW IP PBX prototype must be
integrated both to the Macao CTM provider (with a broadband
Figure 2. The IP PBX layout (512Kbps is the minimal recommended Internet connection) and USJ’s LAN. The ATA Linksys©
Internet bandwidth). SPA3102 will allow the link between the PSTN network and
all USJ analog phones. Basically, it will work as a VoIP
Regarding the IP PBX clients specifications, there are different Internet gateway as well as an extension from the IP PBX.
brands, models and types of IP phones available in the market Hence, the bond between the IP PBX and the Macao residential
such as soft-phones, ATA (Analog Telephone Adapter) and PSTN is established through this VoIP gateway. Secondly, the
handset IP phones. With the exception of the first option that ATA Linksys© PAP2T connects two analog handsets but
can be downloaded free of charge, the remaining ones are not working as two different extensions of the IP PBX. Third, the
available in Macao. For the purpose of testing, one IP PBX server and all its local extensions (ATA, handset IP
Grandstream© GXP2000 (standard desktop IP phone handset) phone and soft-phone) will be integrated with the CTM Macao
and one ATA Linksys© SPA3102 (it works as an analog network, underpinned by an ADSL Modem and a broadband
handset extension and, as well, as a VoIP gateway to connect router (see Figure 5). As expected, this router implements NAT
the IP PBX to the Macao PSTN) were purchased from an (Network Address Translation) and firewall functionality in
online USA site. The ATA Linksys© PAP2T and the Siemens© order to protect the local USJ LAN from Internet intrusion. It
Gigaset C470 IP phone (used as remote extensions located in also provides DHCP service to allocate private IP addresses to
Portugal) were purchased from an online Portuguese site (see the local LAN equipment. Fourth, the IP PBX LAN router has
Figures 3 and 4) with the capability to work with more than one to be configured to allow the UDP (User Datagram Protocol)
VoIP provider as long as it is well configured. Moreover, both data packets to pass through it and to be forward to the right IP
are able to work as an extension of the IP PBX prototype. address. With this purpose, several UDP ports were setup: 5004
to 5037 (Real-time Transfer Protocol, RTP), 5039 to 5082
(Session Initiation Protocol, SIP) and 10000 to 20000 (extra
RTP ports). In a brief way, SIP ports are used for signaling the

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connection between two IP phones (the telephone ring) and, Step Action
when the called IP phone answers it, the RTP protocol start 1 Splash screen and choose install.
being used since it is the main responsible to transport the 2 Format hard drive.
3 Accept default disk partitioning.
audio packets [12].
4 Choose the time zone.
5 Input the “root” password.
6 When the installation finishes, remove the bootable CD and
reboot the PC.
7 Configure the firewall setting to “Enabled” and “Permissive”.
8 Choose a static IP instead of DHCP concerning the network
configuration.
9 The IP PBX configuration finishes by rebooting while the
login screen is splashed.

The trunk lines allow the IP PBX to connect with the external
parties, that is, it links the PSTN and VoIP providers. In this
case, the available trunk connects the Macao CTM PSTN
through a VoIP gateway (the ATA Linksys© SPA3102). It can
be used for both outgoing and incoming calls. Alternatively,
the VoIP trunks allow the system to call external parties (VoIP
Figure 5. The overall IP PBX network diagram.
and PSTN telephone numbers in other countries) through local
VoIP providers using the following Internet infrastructure: (A)
Notice that the IP PBX server computer must be configured G9 Telecom© for income/outcome phone calls to Portugal
with a private static IP address to simplify the configuration of nomadic numbers; (B) VoIPCheap© to make land line and
the local VoIP extensions and the forwarding port. This mobile phone calls to Hong Kong and Macau; (C)
configuration is accomplished at the router level and based on SMSDiscount© to make land line calls to Portugal; (D)
the Port Range Forwarding procedure as figure 6 shows. SmartVoIP© will be used for outgoing mobile calls to
Portugal. For instance, if someone is using one of the
extensions connected to the IP PBX and needs to phone a
CTM© number or a Macao mobile one, the call will be
established through the Macao PSTN trunk (or through the
VoIPCheap© trunk, if the PSTN trunk is already being used).
Nevertheless, if he/she wants to call a PSTN land line in
Portugal, this call will be established through the
Figure 6. Example of a port range regarding forward router configuration.
SMSDiscount© trunk.
The Asterisk©NOW download (version 1.5) comes with an Regarding trunks decision, the IP PBX does it through the
integrated distribution that includes Linux distribution (a Outbound Routes which defines the sequence path regarding
stripped down version of CentOS©), MySQL© database, what to do when one external telephone call arrives into the IP
Apache© Web server, PHP© Web programming language, PBX or when someone dials an external phone number
Asterisk© IP PBX server and FreePBX Asterisk© (strategy definition for which trunk should be used to establish
administration packages. The installation sequence is any particular connection).
summarized in Table 1. On the configuration side, this includes With the present prototype, there will be only two trunk
the creation of Asterisk©NOW Extensions, trunks, lines for inbound routes to be configured: the Macao PSTN line
inbound/outbound routes, Follow me, Disa and conference and the Portugal nomadic number. The remaining trunks have
functions, Backup/Restore and DID (Direct Inward Dial). no associated telephone number and cannot receive telephone
Extensions are used for internal calls that only involve the calls (only used for outbound calls). According to Table 2, all
IP PBX. Trunks are used for external calls that are routed the received phone calls from the Macao PSTN trunk line were
through VoIP gateways or VoIP providers [13]. It covers calls set to be forward to the extension 1005 (IP phone
from and to outside parties, that is, PSTN numbers, nomadic Grandstream© GXP2000 handset). Similarly, all incoming calls
numbers (VoIP numbers) or other IP PBX’s. Once again, from Portugal nomadic numbers were routed to the same
extensions are all those numbers regarding soft-phones, ATA’s extension.
or IP phones directly connected to the IP PBX (configured in
the IP PBX itself and in each IP handset). For instance, the TABLE II. CREATION AND CONFIGURATION OF INBOUND ROUTES
©
WITHIN ASTERIX NOW.
common telephone number that people have in their office desk
usually holds a sub-number with three or four digits. It is this Step Action
sub-number that allows he/she to call their office colleagues or 1 Within the Inbound Routes menu, choose the Add Incoming Route
third PSTN parties as long as these extensions are defined in option.
the PBX office. 2 Add the description (for instance, the device model SPA3102) and
the CTM PSTN number.
3 Choose the destination (set destination menu) for incoming phone
TABLE I. INSTALLATION OF ASTERIX©NOW SOFTWARE PACKAGE. calls on this trunk route. Keep in mind that for this trial product, all

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Vol. 9, No. 6, 2011
the incoming calls will be sent to the extension 1005 of the The conference function connects users from different
Grandstream© IP phone GXP2000. places to minimize phone calls cost [14]. There are two ways
4 Click the Submit/Apply button.
to setup this procedure [15]: (A) The participants are informed
The defined dial plan for all outbound routes is summarized in in advance on the date/time of the conference and, previously,
Table 3. Note that Macao, Portugal and Hong Kong the users should call the phone conference number; (B) The
international country codes are 853, 351 are 852, respectively. participants are informed in advance on the date/time of the
conference and, on a pre-defined date/time, the IP PBX
TABLE III. DIAL PLAN FOR THE IP PBX PROTOTYPE SYSTEM. administrator pulls them to the conference through the
Trunk Inbound route Outbound route FreePBX Flash panel (user’s phones will ring and, after they
In case of Macao-PSTN Destination = 008531. answer, the conference call is already setup).
failure, VoIPCheap© will Extension 1005 0085328XXXXXX The Direct Inward Dial (DID) feature redirect a PSTN
be used 008536XXXXXXX
008538XXXXXXX phone number with a single prefix while the last
00853999 two/three/four digits varies. Thus, each block map number
00852XXXXXXXX corresponds to a different extension. To implement DID, it
starts to request a special kind of trunk from the PSTN
G9 Telecom© 003513XXXXXXXX
provider. For this particular line, as each call is started, the
SMSDiscount© 003512XXXXXXXX suffix digits are actually passed to the IP PBX so it can decide
003517XXXXXXXX which route extension to call to. Usually, PSTN telephone
003518XXXXXXXX numbers are obtained in a block of numbers, for instance, from
SmartVoIP© (in case of 003519XXXXXXXX
28831000 until 28831009 (a ten block numbers, in this case).
failure, SMSDiscount© This block of numbers is, then, configured to match the spatial
will be used) extensions defined in the IP PBX. Typically, the first number
28831000 is a direct line for the receptionist while the
In line with Table 4, different dial plans with different VoIP
remaining ones are DIDs (28831001 signifies organization
provider trunks were configured to minimize call costs.
extension 1001, 28831002 means organization extension 1002
TABLE IV. RATES CHARGED BY THE MAIN VOIP PROVIDERS OF USJ. and so on). Hence, any external user that wants to make a
direct phone call to extension 1005, just needs to dial the
Destination VoIP provider (charge rate in Euros/Minute) telephone number 28831005.
G9 VoIPC SMS SmartVoIP©
Telecom© heap© Discount© Naturally, the soft-phones are configured directly in each
Hong Kong 0.100 0.000 0.010 0.000 laptop while the handset IP phone although ATA have to be
(Land Line) configured in a different way, depending on the brand and
Hong Kong 0.250 0.000 0.005 0.000 model. Extension soft-phones and other handsets need to be
(Mobile)
Macao (Land 0.250 0.020 0.030 0.030
register at the IP PBX with a different IP address (a password
Line) must be supplied as part of the registration process). If an
Macao 0.250 0.030 0.030 0.030 extension is turned off or disconnected from the network, for
(Mobile) instance, the IP PBX will divert calls to the voicemail or
Portugal 0.016 0.000 0.000 0.000
(Land Line)
another pre-defined function. Extensions on the same LAN
Portugal 0.106 0.100 0.065 0.060 can also be hard coded with its IP address from the IP PBX.
(Mobile) Yet, outside extensions are different, depending on whether
the IP PBX has a public IP address (or not). In this case study,
The Follow me function is applied whenever the user is not
the DNS (Domain Named Service) is used to obtain the
able to receive the phone call in his/her extension and he/she
required IP address. Even if five IP phones were installed
wants to forward that call to another extension or even to an
external phone number (both PSTN and VoIP number). In this (CounterPath's X-Lite©, Linksys ATA SPA3102©, Linksys©
particular case, the IP PBX was setup on for every weekday ATA PAP2T, Grandstream© GXP2000 and Siemens© Gigaset
(after working hours) and for those days where someone is out C470 IP), Table 5 only shows the main four steps of the first
of the office (it forwards all the calls from the present extension appliance.
to the user´s mobile).
TABLE V. CONFIGURATION OF IP COUNTERPATH’S X-LITE©.
The Disa function is applied whenever he/she needs to do a
costly phone call (international one to New Zealand, for Step Action
instance). To avoid to be personally charged for this 1 Download the CounterPath’s X-Lite© software
(http://www.counterpath.com/x-lite.html).
circumstance, the user calls to a Disa active phone number of
2 On the SIP Accounting Settings menu, add a new account.
the office. If the request password is correct then the IP PBX 3 Fill the following fields: Display name, Extension number,
will setup the line automatically, according to the cheapest Password, Authorization user name and Domain (the IP
defined route (the dial plan of the outbound routes). address of the IP PBX).
Unsurprisingly, the password traditional system is required for 4 Configuration conclusion after the Ready message.
safety purposes.

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(IJCSIS) International Journal of Computer Science and Information Security,
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V. SYSTEM TESTING
True to form, the subsequent pace is to evaluate the IP PBX
prototype efficiency and reliability of the IP PBX, telephone
calls among extensions located in the same/different IP PBX
LAN, inbound/outbound connections (via SIP and PSTN
trunks), voicemail, Follow me, Disa and conferences
capabilities. As shown below, the tests of Asterisk©NOW
includes its start up procedure (see Table 6), IP PBX access
(see Table 7) and Secure Shell (see Table 8), client’s
registration (see Figure 7), PSTN and SIP trunks (see Table 9).

TABLE VI. IP PBX SERVER START UP TEST RESULTS. IT WAS OBSERVED
THAT THE SERVER REQUIRED 29 SECONDS TO SHUTDOWN.

Date Start Time Duration (Seconds) Start Up Errors? Figure 7. Snapshot of the IP phones (IP PBX clients) extensions (1001,
2010.10.17 16:13:00 83 No 1002, 1005, 1007, 1025, 1026 and 1029) registered after the start up
2010.10.17 16:23:00 83 No procedure. No abnormalities were found.
2010.10.17 16:34:30 83 No
2010.10.17 16:44:00 83 No Afterwards, the phone calls between extensions of the
2010.10.17 16:55:00 83 No
same/different IP PBX LAN followed. Figure 8 and 9 shows
TABLE VII. CHECK RESULTS OF THE IP PBX SERVER ACCESS THROUGH
the call status information between extension 1001-1002 and
THE FREEPBX GUI INTERFACE. 1029-1025 and, auspiciously, the quality voice was considered
excellent in both cases, according to the excellent, good,
Date Access Time Login Access to Test reasonable and bad scale. The same relationship was found for
(hh:mm:ss) accepted? IP PBX? Result
2010.10.17 16:15:00 Yes Yes Pass
other calls extensions such as 1002 to 1005 and 1005 to 1001.
2010.10.17 16:25:00 Yes Yes Pass
2010.10.17 16:36:30 Yes Yes Pass
2010.10.17 16:47:00 Yes Yes Pass
2010.10.17 16:58:00 Yes Yes Pass

TABLE VIII. TEST RESULTS OF IP PBX SERVER ACCESS VIA SSH CLIENT Figure 8. Snapshot of the in boundary call (Macao-Macao) between 1001
INTERFACE. and 1002 extensions.

Date Access Time Login Access to Test
(hh:mm:ss) accepted? IP PBX? Result
2010.10.17 16:18:00 Yes Yes Pass
2010.10.17 16:28:00 Yes Yes Pass
2010.10.17 16:39:30 Yes Yes Pass
2010.10.17 16:50:00 Yes Yes Pass Figure 9. Snapshot of the out boundary call (Macao-Maputo, Mozambique)
2010.10.17 17:02:00 Yes Yes Pass between 1029 and 1025 extensions.

TABLE IX. EVALUATION OF PSTN AND SIP TRUNKS. ONCE AGAIN, NO At last, Table 10 and 11 exhibit some trial results of the
IRREGULARITIES WERE FOUND.
inbound and outbound calls using PSTN and SIP trunks. As
Time Trunks Registered well, the voice mail (from extension 1001 to 1002 and 1025),
PSTN G9 VoIP SMS Smart Follow me and Disa (see table 12) tests occurred with no major
Telecom Cheap Discount VoIP problem.
18:56 Yes Yes Yes Yes Yes
19:05 Yes Yes Yes Yes Yes
19:13 Yes Yes Yes Yes Yes TABLE X. APPRAISAL RESULTS OF THE OUTBOUND CONNECTION TO
19:20 Yes Yes Yes Yes Yes MACAO USING PSTN TRUNKS.
19:28 Yes Yes Yes Yes Yes
Start Duration Origin Destination Voice
Time (hh:mm) Quality
(hh:mm) PSTN Place Fix Place
Line
or
Mobile
20:08 02:59 1001 Macao Mobile Macao Good

20:15 05:23 1005 Macao Mobile Macao Good

TABLE XI. ASSESSMENT RESULTS OF THE OUTBOUND CONNECTION TO
OTHER COUNTRIES USING SIP TRUNKS.

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(IJCSIS) International Journal of Computer Science and Information Security,
Vol. 9, No. 6, 2011
Start Duration Origin Destination Voice REFERENCES
Time Quality
[1] USJ (University of Saint Joseph), “About University of Saint Joseph
PSTN Place Fix Place History”, available at
Line http://www.usj.edu.mo/?content_left&col=1&id=15 [accessed Mar 09].
or
[2] Boavida, F., Monteiro, E, “Computer Networks Engineering”, 4th
Mobile
Edition, ISBN 972-722-203-x, FCA-Lidel, 2007, pp. 554.
20:47 02:23 1001 Macao Fix Macao Good
Line [3] Vestias, M., “CISCO Networking”, 4Th Edition, FCA-Lidel, ISBN 978-
972-722-506-4, 2010, 648 p.
20:29 02:51 1002 Macao Mobile Macao Good
[4] Hamdi, M., Verscheure, O., Hubaux, J.-P., Dalgic, I., Wang, P., “Voice
Service Interworking for PSTN and IP Networks”, Communications
Magazine, IEEE, Vol 37 (5), ISSN 0163-6804, 1999, pp. 104-111.
TABLE XII. EXAMINATION RESULTS OF THE DISA FUNCTION. [5] Obara, H., Yasushi, T., “An Efficient Contention Resolution Algorithm
Access Disa Line given Call Test Result for Input Queuing ATM Cross-Connect Switches”, International Journal
of Digital & Analog Cabled Systems, Vol2 (4), pp. 1989, 261-267.
(hh:mm) Active by IP PBX? Success?
? [6] Holtmanns, S., Horn, G., Moeller, W., “Identity Management in Mobile
22:27 No No No Pass Communication Systems”, in Selected Topics in Communication
22:30 Yes Yes Pass Networks and Distributed Systems, Sudip Misra & Isaac Woungang
(Eds), World Scientific Publishing, 2010, pp. 709-730.
The function conference test was done with the extensions [7] Gratz, J., “Voice Over Internet Protocol”, Science & Techology, 6 Minn.
1001 (Macao), 1025 and 1026 (Portugal) and 1029 J.L., 2004, 443 pp.
(Mozambique). According to Figure 10, the overall quality was [8] Prabhakaran, K., “Advanced Link State Protocol”, in Computer and
considered pretty good. Network Technology: Proceedings of the International Conference on
ICCNT 2009, Zhou & Mahadevan (Eds), World Scientific Publishing,,
2009, pp. 89-92.
[9] Neustar, “What Is ENUM?”, Available at http://www.enum.org/
what.html [accessed Oct 10].
[10] Barbeau, M., Boone, P, Kranakis, E., “Wimax/802.16 Broadband
Wireless Netwworks”, in Selected Topics in Communication Networks
and Distributed Systems, Sudip Misra & Isaac Woungang (Eds), World
Scientific Publishing, 2010, pp. 79-111.
Figure 10. Snapshot of the conference calls among 1001, 1025, 1026 and
1029 extensions. [11] Blueface, CounterPath X-Lite Softphone Specifications. [Online]
Available at http://www.blueface.ie/helpandadvice/specification/
xlite.aspx [accessed Sep 10].
VI. FINAL THOUGHTS [12] Sharma, S., “Hello Expired Time Based Greedy Routing Scheme for
Mobile Ad Hoc Networks”, in Computer and Network Technology:
For personal use, a well known VoIP application is Skype©. Proceedings of the International Conference on ICCNT 2009, Zhou &
This application allows audio and video communications at Mahadevan (Eds), World Scientific Publishing,, 2009, pp. 45-49.
very low costs (from Skype© to PSTN telephones) or even at [13] Smarter, “Linksys SPA3102 Voice Gateway with Router - VoIP
no cost at all (from Skype© to Skype©). Therefore, the incentive gateway”, available at http://www.smarter.com/bridges-routers/linksys-
of this project is to assemble, implement and configure a VoIP spa3102-voice-gateway-with-router-voip-gateway/pd--ch-2--pi-
770317.html [accessed Sep 10].
phone system for USJ needs based on Linux© and other FOSS
[14] Hallberg, B., “Networking”, 5th Edition, McGraw-Hill Professional
(Free Open Source Software) technologies [16]. According to Publishing, p. 415, 2009.
the previous results, it seems it is possible to setup an IP PBX [15] Asterisk, “Forum-AsteriskNOW Support”, [Online] Available at
for USJ, including IP phones. http://forums.digium.com/viewforum.php?f=14&sid=feeaa4f3fbe8e9fc1
Regarding future work, the productive equipment depends, 1706bb68efd5cf1&start=2200 [accessed Sep 10].
nowadays, by the end of the construction of the new campus. [16] Chava, K., How, J., “Integration of Open Source and Enterprise IP
Still, two technical lessons should be highlighted from the past: PBXs, Testbeds and Research Infrastructure for the Development of
(A) To have two Internet broadband lines, one for the IP PBX Networks and Communities” (3rd International Conference), ISBN 978-
1-4244-0739-2, 2007, pp. 1-6.
server and another for the remainder Internet data traffic
network; (B) To design carefully the SIP and RTP ports for AUTHORS PROFILE
both protocols work all together without any conflicts. António Cotão holds a Master degree in Information Technology from the
University of Saint Joseph, Macau, China, and, at the moment, he works
Finally and based on the planning projections of all in the logistics and financial department of a Portuguese pharmaceutical
staff/student number of USJ by 2012, it is recommended the (Hovione).
following hardware components: (A) IP PBX Server (2 Intel Richard Whitfield is a full professor at Saint Joseph University, Macau,
Xeon© processor 7500 series, 16GB RAM, Motherboard with China, and, currently, he is one of the responsibles for the construction
standard graphics and dual 100/1000 Ethernet NIC cards, of the new USJ campus. He holds a doctoral degree in Manufacturing
RAID 5 redundancy with 1TB for each HD, UPS and from the University of Melbourne, Australia.
Digium/ATA’s gateway to connect the Macao PSTN. (B) For João Negreiros is an associate professor at University of Saint Joseph from
2011 and he holds a doctoral degree in Information Technology from the
the clients, Polycom©, Cisco© and Linksys© brands are highly New University of Lisbon, Portugal.
recommended appliances. (C) Asterisk©NOW as the main core
software.

11 http://sites.google.com/site/ijcsis/
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