VOIP Overview
VOIP Overview
Introduction
Voice over IP (VOIP), otherwise known as IP telephony, is the delivery of voice information over Internet
Protocol (IP) packet switched networks. This means sending voice information in digital form in discrete packets rather
than in the traditional circuit-committed protocols of the public switched telephone network (PSTN). A major advantage
of VOIP is that it can avoid the tolls charged by ordinary telephone service by utilising fixed charge IP network services
such as broadband. Recent development with SIP (see below) technology and hardware supporting this standard has
resulted in the production of a number of commercially marketed SIP handsets, both wired and wireless networks,
removing the need for a PC or laptop running a software handset, or “softphone”, to connect to VOIP services. A
subscription to a local server from a SIP handset or softphone provides you with all the normal telephony features
including voice and fax, as well as text and even video services.
SIP (Session initiation protocol) is an Internet standard specified by the Internet Engineering Task Force (IETF) in RFC
2543. SIP is used to initiate, manage, and terminate interactive sessions between one or more users on the Internet.
SIP, which borrows heavily from HTTP and the e-mail protocol SMTP, provides scalability, extensibility, flexibility, and
capabilities for creation of new services. SIP is increasingly used for Internet telephony signalling, in gateways, PC
phones, softswitches, and softphones, but it is not limited to Internet telephony and can be used to initiate and manage
any type of session, including video, interactive games, and text chat.
VOIP Servers
There are many different VOIP servers available both commercially and open source. At the time of writing
one of the most popular servers is [ASTERISK], with its ability to be able to handle complex dialling plans and a wide
range of voice, fax, text and video codecs, as well as switchboard, voicemail and operator support. Asterisk provides
support for H.323 clients as well as those using the newer SIP standard. It also provides the proprietary IAX (Inter-
Asterisk eXchange) protocol to enable the interconnection of multiple asterisk servers, with the ability to forward
communications between servers or use one server as backup for another. The downside of Asterisk is the current
lack of IPv6 support.
[SER] (SIP Express Router) has a smaller footprint than Asterisk, supporting only SIP clients and services. It
uses a full scripting language for its configuration, cutting down on the number of individual configuration files and
improving scalability, at the expense of requiring operators to learn a new language. The configuration language
provides full control over routing and aliasing, so it would probably be possible to mimic a large number of the features
of Asterisk using scripts. SER also has inbuilt support for IPv6 as well as IPv4, and can listen on ports under both
protocols concurrently, giving the advantages of IPv6 such as mobility and removing the need for NAT (network
address translation).
Other packages available include [VOCAL] which also supports both IPv4 and IPv6, however no testing was
done with the package prior to writing this report. VOCAL suffers from the disadvantage that the source code
(decompressed) was, last time we checked, a formidable 78.1Mb, as opposed to 9.8Mb for SER or 37Mb for Asterisk.
This did not encourage us to try using the package.
VOIP Clients
As with the server technology the more popular clients, software and hardware, mainly support the IPv4
protocol with support for IPv6. Some good examples of commercially available clients include [WINDOWS
MESSENGER] for windows, [SJPHONE] for Linux and Windows, neither of these being open source. [KPHONE] and
[LINPHONE] are 2 excellent open sources software packages for Linux, both providing an easy to use GUI (Graphical
user interface) to enable an easy setup. LinPhone was the only client found capable of IPv6 support natively without
requirements for patching, however the interface does not supply a long enough field to enter a full IPv6 address
therefore hostnames of IPv6 machines have to be used.
As with the servers, there are many IPv4 client implementations available, both commercial and freely
available, as well as a number of hardware implementations (SIP handsets). Examples of commercially produced
clients include [WINDOWS MESSENGER] for Windows and [SJPHONE] for Linux and Windows. Neither of these
Conclusion
There are many options for configuration of a successful VOIP network. The servers vary in complexity
depending on requirements, from simple home or small office internal phone systems up to full city capacity telecoms
providers with PTSN endpoints. There are also many types of client technology, with the majority of operating systems
are now supported. Up-to-date information can usually be found on the VOIP Wiki at http://www.voip-info.org/, which
covers information on VOIP technologies and news in the industry.
References
[WINDOWS MESSENGER]
http://www.microsoft.com/windows/messenger/ - Click the link in small print for Messenger 5.0 which supports SIP
[KPHONE] KPhone
http://www.wirlab.net/kphone/index.html
[LINPHONE]
http://www.linphone.org
VOIP Wiki
http://www.voip-info.org/