The Second International Conference on Next Generation Mobile Applications, Services, and Technologies

Deploying Open Source IP Telephony in Rural Environments

Lambros Lambrinos
Dept. of Communication and Internet Studies
Cyprus University of Technology
Limassol, Cyprus
lambros.lambrinos@cut.ac.cy

Abstract convergence: speech and video are converted to data

In the current era of advanced telecommunications
and the Voice-over-IP revolution, areas lacking
modern telephony facilities are still commonplace. In
this paper we propose an architecture that utilises
open source software and other technologies to
provide enhanced telephony services in a rural
community. Using a wireless network infrastructure we
interconnect our SIP-based telephony exchange with
the IP phones and the adaptors installed at various
locations enabling seamless internal communication.
Connectivity to the normal telephone network is
facilitated through a satellite link and a number of
analogue lines. A pilot implementation has provided
some interesting findings and pinpointed some issues.
We discuss potential ways of resolving those issues and
enhancing our infrastructure even further.
1. Introduction
The deployment of low-cost wireless voice network
infrastructures can enhance the standard of living of
many citizens worldwide. This statement may initially
appear that it is exaggerating but people who live in
rural areas do not always enjoy the advanced
telecommunications facilities available in urban areas.
Due to the high infrastructure development costs,
telecommunications companies do not offer all their
services to remote areas. Some areas in the world do
not even have a basic telephone connection and long-
range wireless links are used to provide rudimentary
access (e.g. through a single location servicing a whole
village) to the Public Switched Telephone Network.
During the last few years the landscape in the
telecommunications sector is changing dramatically;
the large and traditional telecom operators are no
longer monopolising the market and face fierce
competition from alternative carriers and cable
operators. The major reason for this is data
which is “transported” using the same networks as
email, web browsing data, file transfer data etc. Voice-
over-IP (VoIP) is the term associated with the
accomplishment of voice conversations through IP-
based networks.
Another significant technological change is the
deployment of wireless IP networks (wifi). Off-the-
shelf components for the creation of a wireless network
are nowadays readily available. This allows for a quick
and flexible deployment of wireless networks opening
up the road for applications such as VoIP, Instant
Messaging (IM) and the provision of internet access to
mobile users.
In an attempt to resolve the problem of non-
availability of advanced telecommunication facilities in
remote areas, we take advantage of the features
associated with VoIP together with the flexibility of
wireless networks to design a low-cost wireless voice
infrastructure. The low-cost factor stems from the use
of freely available open-source software in the core
parts of our system. Based on our design we built a
test-bed that would enable us to identify problems
related to the actual usage of the system from a
technological as well as a usage perspective.
The paper is organized as follows: an overview of
the major characteristics of relevant technologies is
followed by a description of the overall architecture of
our solution and its components. We then present some
details on our test-bed implementation and its
evaluation illustrating our user experiences and the
problems identified. The final section consists of our
conclusions and plans for future work.
2. Background
A number of different technologies are relevant to
our work for this paper so we give a brief introduction
to them emphasising on their major aspects.

978-0-7695-3333-9 /08 $25.00 © 2008 IEEE 623

DOI 10.1109/NGMAST.2008.99
2.1 Open source software
The concept of open-source software is not new but
lately, the initiative is gathering momentum as
increasing numbers of users and system administrators
become less tolerant to proprietary systems that are
often expensive and not always as flexible and
extensible as required. The rationale behind the success
of open source software is that everyone is allowed to
access and modify the source code of a program. Users
get the opportunity to test pre-releases of the software
helping in identifying bugs in the code. User
suggestions for new features are always encouraged;
once incorporated in the system, they are there for
everyone to utilise. Although predominantly operating
on Linux platforms, interest from end-users as well as
companies has resulted in many open source software
packages becoming compatible with multiple operating
systems.
Due to the openness of the development process and
the rapid evolvement of some systems, users have
traditionally worried about the issue of support. The
community spirit and efforts that drive the
development of open source software also drive its
support; users and developers write documentation and
provide help through forums, wikis etc. Adding to that,
companies and consultants who specialise in the
product, and in many cases contribute to its
development, are beginning to offer paid support
options. The result is that many operations on the
internet today rely on open-source solutions. Perhaps
the most prime example is the LAMP system whose
name is derived from the software packages it
incorporates: Linux, the Apache web server, the Mysql
database and PHP.
2.2 Voice over IP
Data convergence has revolutionised the telecoms
world and enriched internet-based communication
between remote users. Voice and video over IP
networks provide for a different communication
experience and at low costs.
Initial implementations of VoIP technology were
initially based on proprietary protocols which
gradually gave way to more open interoperability
standards such as H.323 [1] and Megaco [2]. Until
recently, the H.323 protocol from the ITU-T was the
most widely adopted. H.323 offers a complete solution
for call setup and control but it may be inflexible in
some scenarios.
The Session Initiation Protocol (SIP) [3] from the
IETF follows a more flexible approach allowing many
potential applications. SIP is an application-layer
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signaling protocol that is used to setup and terminate
multimedia sessions such as VoIP calls.
Communication using SIP is text-based and consists of
requests and responses.
In its simplest form, SIP can operate under a peer-
to-peer model where a user sends a direct invite to
another user in order to setup a call. As the protocol
was becoming more widely adopted, both in respect of
the categories of applications it was utilized in (e.g.
VoIP, instant messaging, videoconferencing etc.) and
the number of users, SIP servers became a vital part of
the support infrastructure. A SIP server primarily acts
as a registrar holding location information and
preferences for its registered users which is used
during call establishment. SIP servers may also act as
proxy and redirect servers taking intelligent request
routing decisions based on information they have about
the destination of the request.
Following the standardization of SIP, many open
source and proprietary implementations of the protocol
became available both on the client as well as the
server side. Equipment vendors also implement SIP in
their products and a range of hardware devices are
available. While there is still a large number of
deployments based on H.323, SIP is establishing itself
as the signaling protocol for current and future
multimedia communications.
It is important to note that due to the enormous
research and commercial interest in VoIP, other
proprietary offerings (not necessarily SIP compliant)
also appeared with Skype [4] being a prime and highly
successful example. The disadvantage of such free
offerings, is that the user has absolutely no control over
the service which may become unavailable at any time;
a major outage of Skype’s service [5] illustrates our
point.
2.3 Wireless networks
Wireless networks offer rapid deployment solutions
in situations where wired networks are not easily
implementable. In one mode, a “local” wireless
network (widely known as a “hotspot”) can be created
to provide connectivity to roaming users. They use
their own devices (e.g. laptops, PDAs) to connect to
the access point and access the internet. In another
mode, point-to-point links are established between two
separate locations whose distance may vary between a
couple hundred meters and a few kilometres. Multiple
point-to-point links may originate from the same
access point (point-to-multipoint). Finally, mesh
networks are used to extend connectivity even further;
a receiver also serves as an access point for other
receivers.
 Wireless connectivity is based on the IEEE 802.11
family of protocols which keeps expanding with recent
additions including the 802.11n draft that offers
theoretical speeds of 300 Mbps and 802.11e that
attempts to define enhancements for quality of service
for delay-sensitive applications. Most user equipment
supports the 802.11b/g standards and operates over 2.4
Ghz with connections upto 54 Mbps whereas on the 5
Ghz spectrum, 802.11a offers speeds of upto 54 Mbps
but at shorter ranges. It is important to note that the
actual goodput (i.e. the amount of useful data) is lower
due to IP protocol header overheads. This is highly
significant in a VoIP environment where data packets
are small; test-bed experiments indicate that the
maximum number of simultaneous calls over a
802.11b link is 15 for 64 kb/s constant bit rate traffic
[6].
In addition to the 802.11 protocols, the Wimax
(802.16) standard is also being used for data
communication links. However, it is more targeted
towards long range (up to 50km), high speed (up to 70
Mbps) backhaul links and links to provide increased
bandwidth in cellular applications.
3. System architecture
We utilised the three technological concepts
described in the previous section to design a wireless
voice platform. The general architecture of the system
is shown in Figure 1 and the major components are
outlined in the sections below.
3.1 IP network
Our IP network infrastructure is based on two levels
of connectivity and a combination of wired and
wireless connections. At the first level, we encounter
the wireless backhaul connections which are formed
based on a point-to-multipoint connectivity mode. The
access distribution point (AP) is attached to a sector
antenna focused on the intended coverage area. At the
other end, outdoor Customer Premise Equipment
(CPE) units point towards the AP in order to establish
a connection. Each CPE is connected to a switch and
the second level of connectivity starts at this point;
devices such as IP phones are connected to the switch
using ethernet cables.
To accommodate mobile device users, it is possible
to create small hotspots using standard wireless
routers; these routers can be connected to the CPE
switch in order to build a “neighbourhood” hotspot.
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3.2 SIP server
SIP servers form the core of VoIP systems. The
most well known, open source SIP servers are SER [7]
and OpenSER [8]. They can act as SIP registrar, proxy
and redirect server and can handle thousands of SIP
registrations. Through various add-on modules, they
can offer other services such as voicemail and PSTN
access. On the other hand, Asterisk [9] is an open-
source software system that provides advanced Private
Branch eXchange (PBX) functionality. It offers VoIP
support using multiple protocols (SIP, H.323 and IAX)
and also allows seamless interconnection with the
PSTN using specialized hardware. Moreover, it can
readily facilitate multiparty conferencing for
simultaneous interaction between multiple participants.
OpenSER and SER provide much better
implementations of advanced SIP features when
compared to Asterisk and also scale much better in
terms of the number of SIP registrations. Asterisk
however, offers excellent telephony features and in
many system architectures SER or OpenSER are
deployed [10] in combination with Asterisk in order to
get the best of both worlds.
The large PBX system vendors are also offering IP-
based solutions either as a pure IP system or as a
hybrid solution that consists of an older technology
system along with an add-on to provide IP
functionality. For our case, proprietary systems are too
expensive and sometimes inflexible.
3.3 Endpoints
From a subscriber’s point of view, a number of
options exist for connecting to the system to utilise the
services offered. The simplest, cheapest and most user
friendly solution for non-technology savvy people is
the Analogue Terminal Adapter (ATA); by attaching to
the ATA, a standard analogue telephone can be
connected to the VoIP network. Enhanced call
handling features, directory, missed calls etc. are
offered by dedicated devices called IP phones which
attach directly to the IP network. IP phones also come
in a comparatively expensive wireless version; slightly
larger than mobile phones, wifi phones attach to the IP
network through a hotspot’s access point. Running
special software, mobile phones and PDAs with wifi
capabilities may operate as a wireless IP phone
allowing the user to make calls bypassing the
expensive cellular networks (toll bypass).
Last but not least, are software phones (softphones)
which were the first form of user endpoints in VoIP-
 Figure 1. System architecture
based communication. Softphones run on a personal
computer and a headset with microphone is required
for their operation. They are a good solution for mobile
users with laptops.
4. Testbed implementation and evaluation
In order to assess our design from a technological
and a user point of view, we built a test-bed in a small
village community in Cyprus that had PSTN
connectivity through analogue lines and no DSL-type
internet access.
4.1 Test-bed deployment
Before the deployment of the telephony services the
underlying network had to be in place. For our
backhaul connections we chose to deploy an
IEEE802.11a (5.4 Ghz) network. This was in an
attempt to avoid creating any interference within the
2.4Ghz frequency band which we intend to use for
hotspots; a small hotspot was setup to test the wifi IP
phones. It was also important to comply with local
rules and regulations regarding the strength of the
signal transmission from the access points: 100
milliwatts in the 2.4GHz band and 1Watt in the 5 GHz
band.
In this initial test-bed we only used Asterisk as our
SIP server and PBX as it is relatively easy to setup and
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administer and we did not need any scalability. Our
PBX was running on a standard Pentium 4 server
machine and was connected to the Public Switched
Telephone Network (PSTN) using two ways: 4
analogue lines through an internal card and a satellite
link. Using the satellite link, the server was registered
with a commercial VoIP service provider for low-cost
call termination worldwide. If all the analogue lines
were busy, the calls were going through the satellite
link. International calls were routed automatically via
the satellite link.
For the client side, we used a combination of ATAs,
IP phones from different manufacturers and a wireless
IP phone. We grouped clients together so that those
nearby (up to the maximum permitted length for
Ethernet cable which is 100 meters) were connected to
the same switch and served by the same CPE. To
provide power to the phones we used Power-over-
ethernet switches in conjunction with a UPS in order to
keep the service running for a while in case of a power
cut.
Users were able to use modern PBX facilities such
as call transfer, call parking, do-not-disturb, user
hunting etc. Features on the IP phones included call
lists, contact directory, message waiting indicator etc.
Moreover users were able to call into the system and
listen to their voicemail or access it remotely using a
web browser.
4.2 Evaluation
In general, the system was quite stable and it had no
problems coping simultaneously with 4 calls over the
analogue lines, 1 call over the satellite link and 2-3
internal calls.
During the trial we interchanged the ATAs and the
IP phones between users to gauge their reaction. For
almost all users, the small learning curve with the
phone feature set and the PBX facilities was well
justified; analogue phones are a thing of the past for
them!
Regarding the wireless backhaul links, we have not
had any capacity issues but we are well aware of the
limitations; additional access points will be deployed if
required. We had an issue with the low (512kbps
download and 256kbps upload) capacity of the satellite
link; this was expected though and confirmed by
attempting to push two calls on the link which resulted
in highly problematic conversations; a single call limit
was therefore imposed. The users were initially
shocked by the audio delay (1 to 2 seconds ping times)
over the satellite link; they eventually managed to
adapt their conversations.
A potential issue arises with emergency calls as on
an outbound call over the analogue lines we can not
present any information to identify the caller and hence
their location. The emergency services will only know
that the call came from the area supported by the
system. This is similar to the issues faced with mobile
phone users although their location may be
approximated in a densely covered area. The only issue
with inbound calls was that the caller had to know the
user’s extension.
The most distinct advantage of our architecture is its
expandability; new locations can be readily added
based on a wireless connection. It is important to note
that the satellite link acts as a backup communications
channel and that results in an independent and more
resilient infrastructure.
5. Conclusion and future work
In this paper we presented the architectural design
of a system that provides ubiquitous access to IP
telephony services based on an open-source
implementation of the SIP signaling protocol. Using a
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real testbed, we have demonstrated that the
combination of advanced technologies can have a huge
impact in situations where an application does not
initially seem feasible or economically viable: open
source software provides us with free, expandable and
customisable solutions; wifi allows for rapid
deployment of IP networks and VoIP offers advanced
telephony features over an IP infrastructure.
Overall, we are satisfied with the outcome of the
early stages of our work. Our setup is targeted for a
more permanent solution whereas similar technologies
have been applied in emergency scenarios [11]. In the
next stages we will concentrate on potential solutions
to the issues we faced. We will try to identify a
mechanism for user billing and assess the sustainability
of the system from a business point of view. Moreover,
we will attempt to deploy another test-bed in a nearby
community in order to link the two together for further
tests. Once our system expands, we will use
SER/OpenSER as the SIP front end and interface it
with Asterisk that will take the role of a media,
voicemail and PSTN gateway.
References
[1] ITU-T Recommendation H.323, "Packet-based
communications systems"
[2] C.Groves et.al, "Gateway Contol Protocol Version 1",
RFC 3525, IETF, June 2003
[3] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston,
J. Peterson, R. Sparks, M. Handley, E. Schooler: "SIP:
Session Initiation Protocol", RFC 3261, IETF, June 2002
[4] Skype web page: http://www.skype.com
[5]See:
http://heartbeat.skype.com/2007/08/what_happened_on_augu
st_16.html
[6] S.Sangho and H.Schulzrinne, “Experimental
Measurement of the Capacity for VoIP Traffic in IEEE
802.11 WLANs”, in proc. INFOCOM 2007
[7] SER web page: http://www.iptel.org
[8] OpenSER web page: http://www.openser.org
[9] Asterisk web page: http://www.asterisk.org
[10] L.Lambrinos and P.Kirstein, "Integrating Voice over IP
services in IPv4 and IPv6 networks", in proc ICCGI07,
Gouadeloupe, March 2007
[11] D.Sisalem et. al, "VDSat: Nomadic Satellite-based VoIP
infrastructure", in proc. ISWCS 2005, Siena, Italy, 2005