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


									                                                                 (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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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
    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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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

                                                                       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
                                                                                       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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         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
   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
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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                                                                                                                ISSN 1947-5500
                                                                               (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).


   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
                                                                                        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
                                                                                         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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                                                                                                                      ISSN 1947-5500
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                                                                                                                                     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].
                                                                                    [2] Boavida, F., Monteiro, E, “Computer Networks Engineering”, 4th
                                                                                         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

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