Voice-over-IP update: Evaluating the products

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 functionality.
• 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:

• 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 call.
• 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.

Getting started with VoIP

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 review.

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 managing/controlling.

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?
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.

Avaya

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.

Most 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 achieve.

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 roaming solutions.

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.

Cisco 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 collaboration.

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 users.

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.

There 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 devices.

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 interoperate seamlessly.
• 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 at vomit.xtdnet.nl.
• 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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