Voice over Internet Protocol (VoIP) technologies have been with us for quite some time now, which means the hardware and software have had time to be developed and most of the bugs ironed out. There are also recent convergent technologies and multimedia extensions that can be added to complement most of the VoIP solutions on the market today, this includes instant messaging, live collaboration, and multi-user conferencing.
These convergent technologies -- combined with the relatively low cost and increasing speed of Internet bandwidth and the number of enterprises taking up dedicated VPN tunnels to branch/head offices -- mean there has never been a more perfect time for VoIP to flourish. The traditional PABX as we know it is in its death throes, definitely heading the way of the old manual switched network. No more rooms of operators sitting there plugging wires in and out of manual switch boards to connect calls; this is the digital era, and most systems eventually in one form or the other will be converted kicking and screaming whether they like it or not. Voice communications are at that stage now. VoIP deployment in some respects looks almost as tough a decision to make as a move to IPv6, but eventually it has to be done. Will you be a leader or a follower?
This review will concentrate on the technology itself and how it has developed to where it is today, as well as looking at what some of the major vendors behind VoIP products currently have on offer. There is an overwhelming plethora of options, from downloading software and installing it on a PC that has multimedia capabilities (such as CoolTalk or Microsoft NetMeeting), to rolling out massive enterprise-wide telephony infrastructure changes including special routers, switches, cabling, gateways, power solutions, PSTN/digital interconnects, and custom IP handsets, not to mention the associated redundancy infrastructure, servers, and embedded software such as voice mail, call logging, and system management.
This write-up will focus on the mid level, which is where most organisations would look at starting. These solutions allow for future scalability and expansion as well as servicing the needs of most SMEs.
There's an awful lot of jargon and technical language involved in VoIP, so if you find yourself getting lost, check out the glossary.
Will I save money?
Probably the most attractive feature of VoIP technology to the bean counters is the ability to bypass toll (STD and international) calls. This saves money by routing calls away from the public switched telephone network (PSTN) and running them over a WAN connection instead. This saves monthly telephone bills, and is particularly applicable if the company runs branch offices as it allows them to interconnect all offices via the Internet or an IP-based VPN. With most offices now running some form of broadband, the connection is already there to use. This also suits companies with a lot of teleworkers, who can use consumer broadband or small office data solutions to access the enterprise's VoIP gateways and network remotely.
Although these cost savings may be attractive, this is just the tip of the iceberg when it comes to VoIP technology itself. While many vendors initially came to market in the early days touting toll bypass savings as a major benefit, they are now realising the technology brings a wide range of expanded possibilities that result in potential productivity gains.
Especially prevalent these days are organisations developing specialised applications to run on "smart" handsets (some handsets really are virtual thin clients). This enables those developers to leverage into vertical markets such as medical, government, and hospitality. Other benefits include internal management of the company telephony system, more accurate call logging/management, voice mail, integration with existing directory services, and the ability to manage the phone system using the organisation's data network systems policies and procedures. Workers can also collaborate remotely on projects, and handle conference situations using technologies such as simultaneous presentation, video/voice conference, and instant messaging.
VoIP at a glance
The key components to a basic VoIP installation are:
- Gateway. This is the hub of the network that replaces the
traditional PABX rack. These can be wall mounted or rack mounted and
take on a multitude of sizes and form factors. Most have the ability to
take one or more blades, which enable the interconnects with the
outside world and normal telephone system. Most also incorporate some
level of autosensing directory service and device registration
capabilities, along with the voice mail system and call control/logging
- Routing and switching. This is mostly done with standard network routers and switches; some gateways have routing built in.
- Cabling. Good old "bluey" RJ45 cabling and termination is all that is required.
- Handsets. An overwhelming array of handsets is available
from all VoIP vendors. These "transparently" replace the traditional
handset, but are far in advance of regular dumb handsets. Smart VoIP
handsets incorporate their own CPUs, memory, storage (in some cases),
as well as some of them having integrated, routers, firewalls,
switches, management software, and even calculators.
- Soft phones. These can replace or augment the deployment
by transferring all functions of the handsets and their displays onto a
client PC. As long as it's equipped with microphone and speakers or a
headset, a PC can totally replace the desktop handset. This is very
popular in call centres.
Important issues to look out for include:
Getting started with VoIP
- Security. Right from the get-go ensure that your plans
revolve around security, security, security. It is bad enough having
data leaks from sniffers picking up e-mail transmissions or hackers
gaining unauthorised access to your servers, but to have voice calls
intercepted is beyond the pale. Ensure your VoIP data traffic security
is as good if not better than your normal data traffic security.
Encryption is one of the keys currently accepted to protect calls; most
if not all vendors offer encryption of some level across their systems.
- Load testing. Ensure your existing network will cope
with the added load of a VoIP deployment, or plan to add another
network for the VoIP system.
- Location. Make sure the placement of the PSTN
interconnects or gateways will benefit your company financially by
reducing the cost of telephone bills (using toll bypass). This may even
include the necessity to set up PSTN gateways in several remote offices
to take advantage of the reduced interstate or overseas call rates. If
this is the case, a robust VoIP routing system needs to be developed so
that the calls are routed correctly through the right gateways with a
sufficient amount of available bandwidth to maintain the quality of the
- Redundancy. Always have a backup plan; redundancy is not
a luxury or option when it comes to telephony systems. Any company
worth its salt when implementing a VoIP solution must ensure that there
are alternative routes and options available should the worst happen
and the IP network or links stop working. Consider having PSTN gateways
as a backup at all offices, even if these only have a low ratio of
ports per user available, it would still allow calls to be received and
placed should the main Internet or VPN link go down.
It would be wise for most businesses to audit their current network
infrastructure, paying particular attention to the quality and
condition of the cabling and switching as well as the provision of the
extra network points to each location a phone is required. Most vendors
offer phones with at least a two-port integrated network switch so most
desks that already have networked PCs on them are covered.
However, consider the areas where a phone may be needed but there are
no PCs. Keep security in mind here too; while it may be good to offer
clients a phone to use in waiting areas or hallways, remember that you
are also offering potential attackers a port into your network should a
VoIP phone be placed there. The VoIP handsets in some cases are
powerful tools which can themselves be used for mischief on the network
in the wrong hands so think and plan carefully.
While you are auditing your network, don't forget to factor in the
added overhead of the planned VoIP deployment to the network system. If
the existing infrastructure is running at 85 percent of capacity now,
adding VoIP to this network may not be advisable without overhauling
the whole network. And while the network is being overhauled, you may
also want to consider adding the ability to run power over Ethernet
(PoE) on most if not all network ports -- this enables provision for
devices to come in the future as well as to power the VoIP handsets.
As well as considering the network and its quality and capabilities, it
may be a better idea to allow for a new network cabling/switching
rollout dedicated solely to the VoIP system or to boost the data
network infrastructure. While this would definitely be costly in the
short term, there would be significant benefits if the system is
managed effectively as well as reducing potential stress on the
existing data network and enabling the VoIP system to "stand alone" and
just interconnect with the rest of the data network at strategic
points. This not only provides added redundancy, but also allows for
future scalability of each network independent of each other as well as
the many obvious security benefits.
If there is no possibility of physically running new cable to a
location or you have a lot of employees mobile around the
office/factory, consider wireless phones running 802.11x. This is a
whole new can of worms that I will stay away from in this particular
Managing your network
Managing a VoIP network is very similar to a normal data network system. You manage
- a group of systems (IP handsets),
- on a network system (standard IP based),
- with gateways (between the IP network and the digital/anaglogue communications network)
- and switches with management tools, which facilitate the use of
system servers/appliances (for management, logs and voice mail
system/messages etc) as well as having
- directory service capabilities and call routing in their basic
form, being able to provide names or numbers for addresses that are
assigned to systems on the network segment they are
The beauty of these systems is their ease of management. Being IP
based, adding handsets and controlling their individual properties can
be as simple as a click of the mouse. Same with the ability to manage
numbers and calls, even simple name selection and routing calls between
offices. This is particularly applicable if the workforce changes
locations regularly or hot desks.
Another benefit -- providing people use it -- is the ability to show
who is in/out of the office and even simply who is on/off the phone,
including any number of in-betweens such as in meetings, busy, at
lunch, etc. Simply by clicking their status on their IP console, the
user can flag to anyone who checks on the entire system their current
availability status. This enables people wishing to contact them to
leave messages, schedule a call as soon as that person become
available, or to seek alternate assistance from someone else if the
matter is urgent. Bandwidth issues
One of the main criticisms of VoIP is the amount of bandwidth required for each conversation, and the knockers are sometimes quite justified to pick on this point. Not only must there be adequate bandwidth, the quality of the link must be maintained throughout each call to ensure the users are not affected. At the end of the day if a user is transferring a 3GB file from a workstation to a server, they will not really notice if it takes eight seconds longer one time than the last time due to network traffic impacting on the transfer time; it is all transparent and handled in the very nature of the TCP transmissions. However the same problem with a VoIP call would definitely be noticeable and unacceptable to the participants of that call because it is real-time. And real-time is the key. While VoIP may use TCP packets to set up and establish the call parameters, it mostly uses UDP packets to send/receive the voice traffic. Ensure that your firewall supports forwarding of UDP traffic. The two issues here are bandwidth and quality of service (QOS).
While bandwidth is now abundant and cheap, particularly on a local area network (LAN), moving what could potentially be large continuous amounts of data without interruption from one point to another within a given period of time may not be easy, especially over a wide area network (WAN) link. The average PSTN call runs at 64Kbps. That 64Kbps channel needs to be open and unaffected for the duration of the call. Naturally, not many VoIP installations could afford that type of sustained traffic on the network, particularly large deployments, therefore the dreaded technology C-word must be used: "compression".
Go on, compress me
Of course, with compression comes loss of audio quality, say the knockers. The most commonly standard used with VoIP is H.323, which incorporates the G.723 codec. This can take a 64Kbps stream of data and squash it down to a mere 5.5Kbps or so. Before you get too excited, you also need to take into account the overheads that it takes to transmit that data, and in some situations these could be quite high. For VoIP to work effectively over WAN links, there needs to be low jitter, low packet loss, a relatively high-speed connection between the endpoints, and less than 200ms delay.
Long pauses, unexpected dropouts, or any other strange phenomena not usually associated with land line telephone calls are unacceptable in a VoIP deployment. The service has to be as good as normal landline telephone services. Jitter buffers in the technology help to reduce the effect that jitter can have on the connection, but ultimately the connection is only as healthy as the network it is running over.
This takes us back to the compression protocols -- surely if something is removed through the compression then the quality can't be the same. Interestingly, where the most savings come in the G.723.1 standard are in the pauses between words. Believe it or not, up to 50 percent of a telephone conversation is silence. Please don't even mention music on hold; most vendors have some very interesting ways of dealing with it.
However, when the data is decompressed at the other end, if silence was inserted between the gaps it would sound odd because usually there is some background noise or even the usual reassuring line noise. Various developers deal with the situation in different ways. Some introduce a "generated" hiss or line noise, so that the user of the system does not think that the line has dropped every time the speaker pauses for breath. Another solution is to randomly sample some background noise from each end of the phone conversation link and inject that back into the blank gaps in the conversation.
What's out there? Avaya
We looked at VoIP systems from Avaya, Cisco, Nortel, and Zultys. Each vendor participating in this review was asked to provide either a demo site at their premises or a live site where their VoIP products were in operation. The Test Lab then visited these sites, spoke with the vendors and their engineers about the products being used as well as other products in the vendors range which could also be deployed.
We visited Avaya's premises in St Kilda Road, Melbourne, and had a look
at the equipment and systems there. Avaya's sales teams are split in
two -- one for SME clients and one for enterprise clients. The
company's equipment reflects this also, with dedicated SME and
Enterprise products with a little bit of interoperability built in
between the two ranges.
Avaya has VoIP solutions supporting from two to 30,000 users, beginning
at AU$4995. Avaya solutions are mostly proprietary, the company having
come from its heritage of Western Electric, AT&T, Bell Labs, and
Lucent. As of result, some of the enterprise equipment supports Avaya's
existing PABX infrastructure and can even share the modular cards.
On the enterprise end of the scale there are the Media Servers: the
S8700 which has the potential to scale up to 30,000 endpoints, the
S8500 supports 3000 users, and the S8300 which can run up to 450 users.
Then there ae the Media Gateways: from the MCC1 which is Avaya's legacy
PABX system, the G650 which has 14 card slots and is a 19in rackmount
unit perfect for migration from the old to the new. There's the G350
Media Gateway which is suited to a small standalone office, and the
G700 which is designed to integrate with a medium-sized office or
multi-storey deployment (one on each level) or even run remote branch
offices and provide a level of redundancy.
Enterprise handsets start with the 4601 that retails for AU$199 and
moves up to the 4629SW colour touch-screen model that sells for around
AU$1200. Wireless Handsets are also available ranging in price from
AU$595 to AU$895.
SME deployments run on the IP Office family of products. These scale
from the IP401 (six extensions and four lines) through to the IP412
Office (360 extensions and 120 lines). The IP Office products are most
often recommended for between four and 100 extensions. They are pretty
much a VoIP solution in a box, with limited internal expansion
capabilities but with a modular external expansion system that can be
taken from one main terminal to another as the company grows.
There is also a range of softphones, including PDA-based applications
that convert wireless enabled PDAs into cordless phones. There is also
a softphone that takes advantage of systems equipped with speakers and
a microphone or headsets. This can operate in several modes from pure
IP to teleconnect, which allows users to telecommute from their home to
the office and have all the same functionality via their PC as they
would in the office. It is also a boon for businesses that use hot
desks or call centres.
of the Avaya technology is based on proprietary protocols, however
Avaya offers a SIP proxy server that can interface with the SIP
standard for that equipment.
Security is also well covered. Avaya by default encrypts
handset-to-handset and handset-to-gateway data traffic. Even the key
tones (DTMF) are encrypted. Why? You wouldn't want to use phone banking
with your IP phone and have the tones you are plugging in, such as
account number and PIN sniffed from the network, would you?
One of the most impressive demonstrations Avaya rolled out for us was
the ability for their system to spilt a call to simultaneously ring
both a mobile phone (as in a GSM or 3G mobile), as well as a desktop IP
phone. The user can answer their mobile phone, then pick up their
desktop phone to continue with the conversation. While it all sounds
very simple when written down, in fact it's technically very complex to
Avaya developers have also created an application that can take CCTV
security/camera feeds and display them on the colour displays of the
handsets. This displays just one unique "outside the square" solution
and use of this converging VoIP technologies and the amazing building
blocks that having "smarts" in phone systems like this can provide.
Link this with a panic button on the phone and the image from the CCTV
could be sent directly to a security team member's phone wherever they
are in the building or campus so that they can assess the situation and
react as quickly as possible.
Current customers include Inchcape Motors and MYOB. Cisco Systems
Little introduction is required for Cisco Systems. Cisco has been
involved with VoIP development and deployments for a great number of
years now. In the past, Cisco's designs and features had to take into
account relatively limited bandwidth, but these days Cisco's VoIP
products offer a great deal of features and functionality.
Cisco has somewhat left behind the traditional "voice" part of VoIP and
now roll that up into "IP communications". This encompasses solutions
that extend from information systems (Web and browsable
information/advertising on handsets), through to soft video and voice
communications on products such as TV/Monitor displays, and impressive
Unlike Avaya and Nortel -- who have their roots in the telephony and PABX type side of the
equation -- Cisco Systems originated on networking side of the divide
and therefore isn't as interested in the gap between legacy devices and
VoIP technologies. This, combined with Cisco's prodigious development
and acquisition budgets, has assisted the company to develop and refine
many of the systems it has available today.
We visited Cisco in its North Sydney office and had a brief tour of the
demonstration labs there. Two products reside at the heart of any Cisco
VoIP solution, either the CallManager Express or the CallManager.
CallManager Express is telephony router based on Cisco's IOS operating
system and scales from two to 100 users. CallManager scales from 100 to
30,000 users in a single cluster.
Cisco's data communications heritage ensures its systems are built on
the grounds of data security, scalability, and redundancy (including
geographical redundancy between two physically separate sites).
An estimated price for a converged enterprise-level (400 to 500 users)
solution from Cisco -- including gateway, handsets, applications, and
redundancy -- can be had for around AU$1000 per user.
has developed an extension to the H.323 protocol called Skinny Client
Control Protocol (SCCP) or H.323+ that is used in its systems. Some
devices now also support the SIP protocol.
Video and content delivery convergence across devices are currently a
strong focus for Cisco in this field. from integrated video solutions
(voice, video, data), through to rich-media collaboration, audio
conferencing, instant messaging, video conferencing, and Web
Cisco's main business focus for its IP communications division is on
head offices with remote sites of at least 100 users, generally
averaging around 300 to 500 users. An example of a Cisco deployment in
Australia is Woodside Petroleum with 2500 users at a single site. Nortel
Nortel has been around for over 100 years, and has made the transition
from a communications services and infrastructure company into data
networking solutions. This includes the natural progression to VoIP and
related converged solutions.
Nortel, which is similar in many ways to Avaya,
has two main levels of systems and deployment as well as support and
integration for its legacy PABX user base. The first solution is suited
to the SME market with the BCM 200 and 400 series, supporting up to 150
users and starting at less than AU$10,000. The enterprise-level
equipment is the CSE1000 which can scale from 200 to 15,000 users.
An innovative and vendor-independent overlay for any VoIP system is the
Nortel MCS5100, which provides advanced multimedia support and
capabilities to VoIP networks, and is fully SIP compatible.
Nortel, like Cisco and Avaya, is mainly a proprietary VoIP system based on the company's MCDM and
Unistim protocols. However, the signalling server supports the H.323
and SIP standards.
Nortel is targeting sites with 100 or more users as well as its
existing installed base of PABX and key system users. Some of the high
profile users of Nortel VoIP systems include, ANZ, JB Hi-Fi, and the
University of Ballarat. Zultys
Zultys is the baby of the group at only three years old. Despite this,
the products appear very well designed and manufactured with a lot of
thought and effort going into their usability and functionality. Zultys
says it has very aggressive product development and rollout plans, at
least one product to market per month.
Where other systems may have many overlapping features and specific
functions that not all users may want or need, Zultys' approach is to
give the users and operators what they need while still maintaining a
very easy upgrade path allowing customisation to be added almost on the
fly. The key benefit of this is the simplicity of physically deploying
and using the system.
The Zultys product range shows a surprising level of functionality
while maintaining a very open standards based platform. All Zultys'
products are based on the SIP protocol as well as other open
architectures, for exanple each phone and gateway runs on embedded
Linux with Power PC processors.
The system is based on one of several Media Exchanges: the MX250 (up to
250 users), MX1200 (between 25 to 1200 users), and the MX25. Each Media
Exchange can run standalone or in a redundant array; they also support
a battery backup module of –48V DC, similar to most traditional
telephony systems. The Media Exchanges can also be clustered up to 16
units; 16 MX250s clustered around the world would support up to 4000
For the purposes of this review and the Australian market in particular
the MX250 would be the Media Exchange of choice. It supports three
expansion cards to give it connectivity into the organisation's
existing telephone lines. And it also has two hard disk drives running
RAID 1 (mirrored data). These can be removed by the administrator, as
can the innovative fan tray/clock battery backup module. An entry level
MX250 costs around AU$8000 for five users, however most deployments
range between AU$15,000 and AU$18,000 for 25 or so users.
are three Zultys handsets, the ZIP2, ZIP 4x4, and ZIP 4x5, these are
priced AU$150, AU$550, and AU$640 respectively. Understandably the 4x4
and 4x5 are the power phones whereas the ZIP2 is the entry level. They
support 128-bit AES encryption, which can be enabled by the user and
shows up on the phone's display. The 4x5 handset even has Bluetooth
integrated so users with Bluetooth headsets can connect and use these
The current target markets for the Zultys team here in Australia is
multi-site deployments of 150 users or less. However the solutions are
available from five to 16,000 users. Zuyltys recently deployed a system
for 4Logik, running an MX250 with around 80 users, 20 or so using ZIP
4x4 handsets and the rest running softphones. Final Words
As far as we're concerned, PABX systems are dead, so you should learn to live with it. Start training your IT staff now on telephony conventions. Bandwidth is getting faster, cheaper, and the connecting infrastructure is improving. Normal telephone public copper cabling, switches, and exchanges will not become redundant for a long long time, but hybrid or gateway VoIP and their related enhancements are the way to go. Companies are going to have to make to move sooner or later, and those who adopt now can only surely benefit from the increased productivity gains and experience that they have with a VoIP solution.
It may take a while for the IT department to come to terms with the technology, especially if in the past they have not had to manage the company communications systems. However, given time it is relatively simple to learn, certainly no harder than adding a new piece of equipment or network attached appliance. Some vendors claim VoIP can seamless be integrated with your existing network cabling and infrastructure -- just plug in the gateway and the phones to your network and Bob's your uncle.
Ah the number of times we've heard that marketing hype in the IT industry...
I apologise in advance to all the over-worked company technicians and technical departments for having them lumped with learning and implementing another system to support, but once it is in place they will also benefit from its features, don't forget that you will have the ability to route voice messages just like e-mail messages so long live the technical query handball!
Glossary of terms
- DTMF (dual tone multi-frequency). The system used by touch
tone phones, where each key is converted to a predefined unique
frequency or tone enabling the user to enter codes via their
telephone's keypad to interface with electronic systems such as
voicemail or electronic banking.
- Forklift upgrade. Applied to legacy telephony systems too
old or not able to be upgraded within an organisation. Therefore a
forklift mentality is used: out with the old in with the new.
Unfortunately in some VoIP deployments this may be the most practical
-- if not cost effective -- solution.
- G.723. Part of the H.323 protocol designed for voice/audio in particular and supports real-time encoding/decoding.
- Gateway. The box that controls the links between IP and
the normal telephony network, in many cases also controls the phone
links and software such as voicemail.
- Greenfields. A VoIP deployment where there is no other
legacy VoIP and in most cases no existing telephony solution at all.
For example, in a brand new development that is under construction, the
business may make a conscious decision from the word go to deploy VoIP.
If you are currently planning to expand and build, this is definitely
something worthy of consideration.
- H.323. A standard approved by the International
Telecommunication Union (ITU) that dictates how video and audio streams
across IP devices, in theory enabling two different devices to
- Handset. The thing on your desk with the buttons and the banana-shaped device you hold to your ear.
- IM (instant messaging). Enables the users to establish a
private text-based chat over the Internet. Most IM systems have the
ability for users of the system to indicate whether or not a they are
available for instant messaging at any given time.
- ISDN (integrated services digital network). A digital telephone technology standard that supports voice, video, and data calls at 64kbps.
- Jitter. The slight variation of a transmission signal in
time or phase that can introduce errors and loss of synchronisation.
More jitter will be encountered with longer cables, cables with higher
attenuation, and signals at higher data rates. Also, called phase
jitter, timing distortion, or intersymbol interference (see www.femf.org/education/Summit2000syll/ottglossary.htm)
- PABX/PBX (private automatic branch exchange). The main box that controls how calls are handled when coming into the site and out of the site.
- PoE (power over Ethernet). The ability to run relatively
low DC voltages over Ethernet cable to power devices such as wireless
access points and VoIP handsets.
- POTS (plain old telephone system). The slowest, oldest
form of analogue telephone system still in use, mainly defined and
hindered by its limited bandwidth of less than 50Kbps in most cases.
- PSTN (public switched telephone network). The worldwide network of POTS carrying analogue voice data.
- SIP (session initiated protocol). A standards-based
protocol which is used for setting up IP-based communications like VoIP
or IM sessions; controls the initiation of call setup, features,
routing, and authentication between the endpoints.
- Sniffing. Sniffers exist on the one hand to allow
network administrators to monitor the quality of their network but on
the other hand to allow hackers to intercept network traffic for their
own use. Once the data has been sniffed and dumped to a file, the data
can then be pieced together again. A very popular tool used to convert
from unencrypted G.7x dumped data traffic to audio files is called
VOMIT (voice over misconfigured Internet telephones) and can be found
- Soft phone. An application that enables suitably enabled
PCs or handhelds (generally those with a speaker and microphone) to act
as an IP phone. Most of these applications have a graphical
representation of a normal telephone handset to ease users' migration
from one to the other.
- Toll bypass. Some VoIP calls can be placed over the
Internet bypassing the PSTN. These cost less than a call placed
directly via the PSTN, effectively bypassing the traditional phone
system altogether. While some savings can be made by this method, it
certainly should not be the main reason why a business deploys VoIP.
- VPN (virtual private network). A secure link or tunnel between two systems over a WAN that effectively simulates a LAN link.
- WAN (wide area network). A network connecting the local
area networks of several remote sites together via IP-based VPNs,
leased lines, or other connections.
This article was first published in Technology & Business magazine.
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