If you've never heard of VoIP, get ready to change the way you think about long-distance phone calls. VoIP, or Voice
over Internet Protocol, is a method for taking analog audio signals, like the kind you hear when you talk on the phone, and turning them into digital data that can be transmitted over the Internet.
How is this useful? VoIP can turn a standard Internet connection into a way to place free phone calls. The practical
upshot of this is that by using some of the free VoIP software that is available to make Internet phone calls, you are bypassing
the phone company (and its charges) entirely.
VoIP is a revolutionary technology that has the potential to completely rework the world's phone systems. VoIP providers
like Vonage have already been around for a little while and are growing steadily. Major carriers like BT are already setting up VoIP
calling plans in several markets around the UK. "reinvention of the wheel." In this article, we'll explore the principles
behind VoIP, its applications and the potential of this emerging technology, which will more than likely one day replace the
traditional phone system entirely.
The interesting thing about VoIP is that there is not just one way to place a call. There are three different "flavors"
of VoIP service in common use today:
ATA - The simplest and most common way is through the use of a device called an ATA (analog telephone adaptor).
The ATA allows you to connect a standard phone to your computer or your Internet connection for use with VoIP. The ATA is
an analog-to-digital converter. It takes the analog signal from your traditional phone and converts it into digital data for
transmission over the Internet. Providers like Vonage and AT&T CallVantage are bundling ATAs free with their service.
You simply crack the ATA out of the box, plug the cable from your phone that would normally go in the wall socket into the
ATA, and you're ready to make VoIP calls. Some ATAs may ship with additional software that is loaded onto the host computer
to configure it; but in any case, it is a very straightforward setup.
IP Phones - These specialized phones look just like normal phones with a handset, cradle and buttons. But instead
of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernet connector. IP phones connect directly to your router and have all the hardware and software necessary right onboard to handle the IP call. Soon, Wi-Fi IP phones will be available, allowing subscribing callers to make VoIP calls from any Wi-Fi hot spot.
Computer-to-computer - This is certainly the easiest way to use VoIP. You don't even have to pay for long-distance
calls. There are several companies offering free or very low-cost software that you can use for this type of VoIP. All you
need is the software, a microphone, speakers, a sound card and an Internet connection, preferably a fast one like you would get through a cable or DSL modem. Except for your normal monthly ISP fee, there is usually no charge for computer-to-computer calls, no matter the distance.
If you're interested in trying VoIP, then you should check out some of the free VoIP software available on the Internet.
You should be able to download and set it up in about three to five minutes. Get a friend to download the software, too, and
you can start tinkering with VoIP to get a feel for how it works. One place to look is BTBut chances are good you are already
making VoIP calls any time you place a long-distance call. Phone companies use VoIP to streamline their networks. By routing
thousands of phone calls through a circuit switch and into an IP gateway, they can seriously reduce the bandwidth they're
using for the long haul. Once the call is received by a gateway on the other side of the call, it is decompressed, reassembled
and routed to a local circuit switch.
Although it will take some time, you can be sure that eventually all of the current circuit-switched networks will be replaced
with packet-switching technology (more on packet switching and circuit switching later). IP telephony just makes sense, in terms of both economics and
infrastructure requirements. More and more businesses are installing VoIP systems, and the technology will continue to grow
in popularity as it makes its way into our homes.
The Forrester Research Group predicts that nearly 5 million U.S. households will have VoIP phone service by the end of 2006. Perhaps the biggest draws
to VoIP for the home users that are making the switch are price and flexibility.
With VoIP, you can make a call from anywhere you have broadband connectivity. Since the IP phones or ATAs broadcast their
info over the Internet, they can be administered by the provider anywhere there is a connection. So business travelers can
take their phones or ATAs with them on trips and always have access to their home phone. Another alternative is the softphone.
A softphone is client software that loads the VoIP service onto your desktop or laptop. The Vonage softphone has an interface
on your screen that looks like a traditional telephone. As long as you have a headset/microphone, you can place calls from
your laptop anywhere in the broadband-connected world.
Most VoIP companies provide the features that normal phone companies charge extra for when they are added to your service
plan. VoIP includes:
There are also advanced call-filtering options available from some carriers. These features use caller ID information to
allow you make a choice about how calls from a particular number are handled. You can:
Forward the call to a particular number
Send the call directly to voicemail
Give the caller a busy signal
Play a "not-in-service" message
Send the caller to a funny rejection hotline
With many VoIP services, you can also check voicemail via the Web or attach messages to an e-mail that is sent to your
computer or handheld. Not all VoIP services offer all of the features above. Prices and services vary, so if you're interested,
it's best to do a little shopping.
Now that we've looked at VoIP in a general sense, let's look more closely at the components that make the system work.
In order to understand how VoIP really works and why it's an improvement over the traditional phone system, it helps to first
understand how a traditional phone system works.
The Standard Phone System vs. The VoIP System
Circuit Switching Existing phone systems are driven by a very reliable but somewhat inefficient method
for connecting calls called circuit switching.
Circuit switching is a very basic concept that has been used by telephone networks for more than 100 years. When a call is made between two parties, the connection is maintained for the duration of the call.
Because you are connecting two points in both directions, the connection is called a circuit. This is the foundation
of the Public Switched Telephone Network (PSTN).
Click "Play" to see how circuit switching works.
Here's how a typical telephone call works:
You pick up the receiver and listen for a dial tone. This lets you know that you have a connection to the local office
of your telephone carrier.
You dial the number of the party you wish to talk to.
The call is routed through the switch at your local carrier to the party you are calling.
A connection is made between your telephone and the other party's line using several interconnected switches along the
way.
The phone at the other end rings, and someone answers the call.
The connection opens the circuit.
You talk for a period of time and then hang up the receiver.
When you hang up, the circuit is closed, freeing your line and all the lines in between.
Let's say that you talk for 10 minutes. During this time, the circuit is continuously open between the two phones. In the
early phone system, up until 1960 or so, every call had to have a dedicated wire stretching from one end of the call to the
other for the duration of the call. You would use all those pieces of wire just for your call for the full 10 minutes. You
paid a lot for the call, because you actually owned a 3,000-mile-long copper wire for 10 minutes. somewhat more efficient
and they cost a lot less. Your voice is digitized, and your voice along with thousands of others can be combined onto
a single fiber optic cable for much of the journey (there's still a dedicated piece of copper wire going into your house, though). These calls
are transmitted at a fixed rate of 64 kilobits per second (Kbps) in each direction, for a total transmission rate of 128 Kbps.
Since there are 8 kilobits (Kb) in a kilobyte (KB), this translates to a transmission of 16 KB each second the circuit is
open, and 960 KB every minute it's open. So in a 10-minute conversation, the total transmission is 9,600 KB, which is roughly
equal to 10 megabytes (check out How Bits and Bytes Work to learn about these conversions). If you look at a typical phone conversation, much of this transmitted data is wasted.
While you are talking, the other party is listening, which means that only half of the connection is in use at any given
time. Based on that, we can surmise that we could cut the file in half, down to about 4.7 MB, for efficiency. Plus, a significant
amount of the time in most conversations is dead air -- for seconds at a time, neither party is talking. If we could remove
these silent intervals, the file would be even smaller. Then, instead of sending a continuous stream of bytes (both silent
and noisy), what if we sent just the packets of noisy bytes when you created them? That is the basis of a packet-switched
phone network, the alternative to circuit switching.
Packet Switching Data networks do not use circuit switching. Your Internet connection would be a lot slower if
it maintained a constant connection to the Web page you were viewing at any given time. Instead, data networks simply send and retrieve data as you need it. And, instead of
routing the data over a dedicated line, the data packets flow through a chaotic network along thousands of possible paths.
This is called packet switching.
While circuit switching keeps the connection open and constant, packet switching opens a brief connection -- just long
enough to send a small chunk of data, called a packet, from one system to another. It works like this:
The sending computer chops data into small packets, with an address on each one telling the network devices where to send
them.
Inside of each packet is a payload. The payload is a piece of the e-mail, a music file or whatever type of file
is being transmitted inside the packet.
The sending computer sends the packet to a nearby router and forgets about it. The nearby router send the packet
to another router that is closer to the recipient computer. That router sends the packet along to another, even closer router,
and so on.
When the receiving computer finally gets the packets (which may have all taken completely different paths to get there),
it uses instructions contained within the packets to reassemble the data into its original state.
Packet switching is very efficient. It lets the network route the packets along the least congested and cheapest lines.
It also frees up the two computers communicating with each other so that they can accept information from other computers,
as well
Pros and Cons
Advantages VoIP technology uses the Internet's packet-switching capabilities to provide phone service. VoIP has
several advantages over circuit switching. For example, packet switching allows several telephone calls to occupy the amount
of space occupied by only one in a circuit-switched network. Using PSTN, that 10-minute phone call we talked about earlier
consumed 10 full minutes of transmission time at a cost of 128 Kbps. With VoIP, that same call may have occupied only 3.5
minutes of transmission time at a cost of 64 Kbps, leaving another 64 Kbps free for that 3.5 minutes, plus an additional 128
Kbps for the remaining 6.5 minutes. Based on this simple estimate, another three or four calls could easily fit into the space
used by a single call under the conventional system. And this example doesn't even factor in the use of data compression, which further reduces the size of each call.
Let's say that you and your friend both have service through a VoIP provider. You both have your analog phones hooked up
to the service-provided ATAs. Let's take another look at that typical telephone call, but this time using VoIP over a packet-switched
network:
Click "Play" to see how packet switching works.
You pick up the receiver, which sends a signal to the ATA.
The ATA receives the signal and sends a dial tone. This lets you know that you have a connection to the Internet.
You dial the phone number of the party you wish to talk to. The tones are converted by the ATA into digital data and temporarily
stored.
The phone number data is sent in the form of a request to your VoIP company's call processor. The call processor
checks it to ensure that it is in a valid format.
The call processor determines to whom to map the phone number. In mapping, the phone number is translated to an
IP address (more on this later). The soft switch connects the two devices on either end of the call. On the other end, a signal
is sent to your friend's ATA, telling it to ask the connected phone to ring.
Once your friend picks up the phone, a session is established between your computer and your friend's computer. This means
that each system knows to expect packets of data from the other system. In the middle, the normal Internet infrastructure handles the call as if it were e-mail or a Web page. Each system must use the same protocol to communicate. The systems implement two channels, one for each direction,
as part of the session.
You talk for a period of time. During the conversation, your system and your friend's system transmit packets back and
forth when there is data to be sent. The ATAs at each end translate these packets as they are received and convert them to
the analog audio signal that you hear. Your ATA also keeps the circuit open between itself and your analog phone while it
forwards packets to and from the IP host at the other end.
You finish talking and hang up the receiver.
When you hang up, the circuit is closed between your phone and the ATA.
The ATA sends a signal to the soft switch connecting the call, terminating the session. Probably one of the most compelling
advantages of packet switching is that data networks already understand the technology. By migrating to this technology, telephone
networks immediately gain the ability to communicate the way computers do.
It will still be at least a decade before communications companies can make the full switch over to VoIP. As with all emerging
technologies, there are certain hurdles that have to be overcome.
Hurdles The current Public Switched Telephone Network is a robust and fairly bulletproof system for delivering
phone calls. Phones just work, and we've all come to depend on that. On the other hand, computers, e-mail and other related
devices are still kind of flaky. Let's face it -- few people really panic when their e-mail goes down for 30 minutes. It's
expected from time to time. On the other hand, a half hour of no dial tone can easily send people into a panic. So what the
PSTN may lack in efficiency it more than makes up for in reliability. But the network that makes up the Internet is far more
complex and therefore functions within a far greater margin of error. What this all adds up to is one of the major flaws in
VoIP: reliability.
First of all, VoIP is dependant on wall power. Your current phone runs on phantom power that is provided over the
line from the central office. Even if your power goes out, your phone (unless it is a cordless) still works. With VoIP, no power means no phone. A stable power source must be created for VoIP.
Another consideration is that many other systems in your home may be integrated into the phone line. Digital video recorders, digital subscription TV services and home security systems all use a standard phone line to do their thing. There is currently no way to integrate these products with VoIP. The related
industries are going to have to get together to make this work.
Emergency 999/112/911 calls also become a challenge with VoIP. As stated before, VoIP uses IP-addressed phone numbers,
not NANP phone numbers. There is no way to associate a geographic location with an IP address. So if the caller can't tell
the 999/112/911 operator where he or she is located, then there is no way to know which call center to route the emergency
call to and which EMS should respond. To fix this, perhaps geographical information could somehow be integrated into the packets.
Because VoIP uses an Internet connection, it is susceptible to all the hiccups normally associated with home broadband
services. All of these factors will affect call quality:
Latency
Jitter
Packet loss
Phone conversations can become distorted, garbled or lost because of transmission errors. Some kind of stability in Internet
data transfer needs to be guaranteed before VoIP could truly replace traditional phones.
VoIP is susceptible to worms, viruses and hacking, although this is very rare and VoIP developers are working on VoIP encryption to counter this.
Another issue associated with VoIP is having a phone system dependant on individual PCs of varying specifications and power. A call can be affected by processor drain. Let's say you are chatting away on
your softphone, and you decide to open a program that saps your processor. Quality loss will become immediately evident. In
a worst case scenario, your system could crash in the middle of an important call. In VoIP, all phone calls are subject to
the limitations of normal computer issues.
One of the hurdles that was overcome some time ago was the conversion of the analog audio signal your phone receives
into packets of data. How it is that analog audio is turned into packets for VoIP transmission? The answer is codecs.
Codecs, Soft Switches and Protocols
Codecs A codec, which stands for coder-decoder, converts an audio signal into a compressed digital form
for transmission and then back into an uncompressed audio signal for replay. This is the essence of VoIP. Digital-to-analog
conversion is seen in everything from CD players to cell phones to video game consoles.
Codecs accomplish the conversion by sampling the audio signal several thousand times per second. For instance, a
G.711 codec samples the audio 64,000 times a second. It converts each tiny sample into digitized data and compresses it for transmission.
When the 64,000 samples are reassembled, the pieces of audio missing between each sample are so small that to the human ear,
it sounds like one continuous second of audio signal. There are different sampling rates in VoIP depending on the codec being
used:
64,000 times per second
32,000 times per second
8,000 times per second
A G.729A codec has a sampling rate of 8,000 times per second and is the most commonly used codec in VoIP. It is a compromised balance between
sound quality and efficiency of bandwidth.
Codecs operate by using advanced algorithms that help them sample, sort, compress and packetize audio data. The CS-ACELP
algorithm (CS-ACELP = conjugate-structure algebraic-code-excited linear prediction) is one of the most prevalent algorithms
in VoIP. CS-ACELP helps to organize and streamline the available bandwidth. Annex B is an aspect of CS-ACELP that creates
the transmission rule, which basically states "if no one is talking, don't send any data." As we learned before, the efficiency
created by this rule is one of the greatest ways in which packet switching is superior to circuit switching. It is Annex B
in the CS-ACELP algorithm that is responsible for that aspect of the VoIP call.
So the codec works with the algorithm to convert and sort everything out, but none of that is any good without knowing
where to send the data. In VoIP, that task is handled by soft switches.
Soft Switches E.164 is the name given to the standard for the North American Numbering Plan (NANP). Simply stated, this is the numbering system that phone networks use to know where to route a call based on the numbers
entered into the phone keypad. In that way, a phone number is like an address:
(313) 555-1212
313 = State 555 = City 1212 = Street address
In our example, the switches know to use "313" to route the phone call to the region denoted by the area code. The "555"
prefix sends the call to a central office, and the network routes the call using the last four digits, which are associated
with a specific location. So based on that system, no matter where you are in the world, the number combination "(313) 555"
will always put you in the same central office, which has a switch that knows which phone is associated with "1212."
The challenge with VoIP is that IP-based networks don't read phone numbers based on NANP. They look for IP addresses, which
look more like this:
192.158.10.7
IP addresses correspond to a particular device on the network. It can be a computer, a router, a switch, a gateway or, in this case, a
telephone. To make matters worse, IP addresses are not always static. They are assigned by a DHCP server on the network and
generally change with each new connection. So the challenge with VoIP is figuring out a way to translate NANP phone numbers
to IP addresses and then finding out the current IP address of the requested number. This is the mapping process referred
to earlier and is handled by a central call processor running a soft switch.
IP Mapping The central call processor is a piece of hardware running a specialized database/mapping program
called a soft switch. Think of the user and the phone or computer associated with that user as one package -- man and
machine. That package is called the endpoint. The soft switch connects endpoints.
Soft switches know:
Where the endpoint is on the network
What phone number is associated with that endpoint
The current IP address assigned to that endpoint
So when a call is placed using VoIP, a request is sent to the soft switch asking which endpoint is associated with the
dialed phone number and what that endpoint's current IP address is. The soft switch contains a database of users and phone
numbers. If it doesn't have the information it needs, it hands off the request downstream to other soft switches until it
finds one that can answer the request. Once it finds the user, it locates the current IP address of the device associated
with that user in a similar series of requests. It sends back all the relevant information to the softphone or IP phone, allowing
the exchange of data between the two endpoints.
Soft switches work in tandem with the devices on the network to make VoIP possible. In order for all of these devices to
work together, they must communicate in the same way. This communication is one of the most important aspects that will have
to be refined in order for VoIP to really take off. Currently, there are three protocols used for this communication. In the
next section, we will learn about them.
Protocols As we've seen, on each end of a VoIP call we can have any combination of an analog, soft or IP phone
as acting as a user interface, ATAs or client software working with a codec to handle the digital-to-analog conversion, and
soft switches mapping the calls. So how do you get all of these completely different pieces of hardware and software to communicate
efficiently to pull all of this off? The answer is protocols.
There are several protocols currently used for VoIP. These protocols define ways in which devices like codecs connect to
each other and to the network using VoIP. They also include specifications for audio codecs. The most widely used protocol
is H.323, a standard created by the International Telecommunication Union (ITU). H.323 is a comprehensive and very complex protocol that was originally designed for video conferencing. It
provides specifications for real-time, interactive videoconferencing, data sharing and audio applications such as VoIP. Actually
a suite of protocols, H.323 incorporates many individual protocols that have been developed for specific applications.
As you can see, H.323 is quite a large collection of protocols and specifications. That's what allows it to be used for
so many applications. The problem with H.323 is that it is not specifically tailored to VoIP.
An alternative to H.323 emerged with the development of Session Initiation Protocol (SIP). SIP is a much more streamlined protocol, developed specifically for VoIP applications. Smaller and more efficient
than H.323, SIP takes advantage of existing protocols to handle certain parts of the process. Media Gateway Control Protocol (MGCP) is a third commonly used VoIP protocol that focuses on endpoint control. MGCP is geared toward features like call
waiting. You can learn more about the architecture of these protocols at Protocols.com: Voice Over IP.
One of the challenges facing the worldwide use of VoIP is that these three protocols are not always compatible. VoIP calls
going between several networks may run into a snag if they hit conflicting protocols. Since VoIP is a relatively new technology,
this compatibility issue will continue to be a problem until a governing body creates a standard universal protocol for VoIP.
The overall hurdle facing VoIP is that there are currently no overriding standards. This includes hardware, protocols and
virtually every aspect of the system. In the end, VoIP is a vast improvement over the current phone system in terms of efficiency,
cost and flexibility. Like any emerging technology, VoIP has some challenges to overcome, but it is clear that developers
will keep refining this technology until it eventually replaces the current phone system.
What is VoIP? There has been some confusion amongst communications professionals and customers alike as to the
use of terms like “VoIP”, “Internet Telephony” and “IP Telephony”. For the purposes
of this website, the term “VoIP” is used to define the technology, which underlies BT Hosted IP Telephony. However,
in a more general sense, the following are definitions of IP technologies to aid understanding:
Internet Telephony or Voice over the Internet or Voice on the Net – Mainly used by residential
users with PCs (and perhaps USB handsets). PCs with IP telephony software, internet access via dial up or access via
broadband which allows calls to bypass PSTN tolls. BT Communicator is an example of this type of product.
Voice-over-IP – Traditional digital TDM phones connected to a PBX which itself is connected to
a VoIP gateway or includes an integrated VoIP gateway card, both of which transition Voice data from TDM transport to IP transport,
allowing conveyance over an IP VPN.
IP Telephony (IPT) – Covers the area of IP Phones on the desktop implicit is a telephone feature
equivalence (more than make a call and have a voice conversation), potentially voice/data convergence with IP applications,
including voice, video, and data. The term has been associated with IP PBXs provided by a Carrier, or Service Provider, or
Customer, and often located within an enterprises building. Cisco's Call Manager is an example of an IP Telephony product.
BT IP Telephony (or Hosted IPT) - MM VoIP or VoIP Port are examples of IPT products which are hosted
off-site by a Carrier. BT uses the marketing term "BT Hosted IP Telephony" to describe both MM VoIP, IP Voice and VoIP Port
as these share a common BT-located service platform.
Why are some of the Call Manager features not available with BT MM VoIP? BT only releases
a feature for service when it has been completely tested on its hosted IP Telephony platform. In addition, BT is market
lead and will therefore only develop, test and release a feature where there is such a market demand. If a feature is
required, please contact your Account Manger.
What other IP phones can be supported? Currently BT supports a range of the Cisco IP phones. This situation
is constantly under review so contact your Account Manager for an up to date range.
What quality levels can I expect from Hosted IP Telephony? Extensive testing of the VoIP voice quality has indicated
that the voice quality will be better than that achieved on a good quality mobile call. To minimise the amount of CoS1
bandwidth required, BT strongly advises customers to compress their voice across their WAN. The ITU standard for
voice quality measurement is based on its Mean Opinion Score (MOS) algorithm. MOS is rated on a scale of 1 (noise) to
5 (perfect quality), with “Toll Quality” generally accepted as 4 or better. The CODEC used by MM VoIP
to compress voice across the WAN is G.729a, which produces a maximum theoretical MOS of 3.92. Customer experience has
proved that this is an acceptable level for business use. In addition to MOS, MM VoIP service is monitored against
other quality benchmarks. MM VoIP users should expect to see the following minimum quality standards:
MOS score of 3.7
Jitter of no more than 25 ms
Packet Loss of no more than 0.5%
Round Trip Delay of less than 150ms
Customer experience has proven that these standards are exceeded in most calls made.
How much new equipment needs to be purchased to install VoIP? The customer will be offered
the ability to use their existing WANs (this includes a number of possible infrastructures, ranging from IP VPNs to DIY WANs)
to deliver the voice traffic. The customer will then be able to deliver IP directly to their desktop (i.e. over the LAN),
allowing them to benefit from the 'value-add' of IP applications. It is important that the LAN and WAN infrastructure can
support "Quality of Service" (prioritisation of Voice over Data) and Class of Service (for IP VPNs, reserving a portion of
the available bandwidth for voice at any one time). Depending upon the current LAN equipment, it may be necessary to upgrade
existing LAN infrastructure. The Hosted IPT service is network-based, so customers will need to purchase access to the
service, or buy a bundled services offering. The product may require the customer to purchase new customer premises equipment,
primarily IP phones or PC soft phones. If the customer wants to inter-work with legacy equipment, then a gateway will be
required.
How easy will it be for customers to add on sites/take sites off and take care of moves and changes in their organisation? The
customer as an entity will need to register and configure their individual users when they first purchase the service. The
user can do simple changes like changing speed dials and adjusting ring tones. More complex additions or changes (like changing
DDI numbers) should be carried out through the BT Account Manager and will incur a charge. However, due to the
nature of VoIP, the majority of these additions and changes can be carried out quickly and easily
How will calls be routed? Calls within a customer's network will be routed over their LAN and WAN infrastructure.
A signal is sent from the calling party to the Hosted IPT Call Servers which allows the latter to set up the real-time communications
path. This path routes directly between the calling and the called parties. Calls to and from traditional networks
such as the PSTN will be routed via the BT Hosted IPT platform.
What protection/gateways have been put in place? Additional resilience? The products,
networks and supporting systems are designed to BS7799 standards, with robust access control policy. The platform makes
use of firewalls to guard against unauthorised access to BT's network infrastructure and components. Further, a Security Management
Team has been incorporated who respond to Incident Management Procedures, and the platform uses a comprehensive Backup Policy
to be able to restore from most major incidents. As a final measure, a comprehensive auditing policy is used to identify inappropriate
activity. For more information please see the Hosted VoIP security white paper.
If the network goes down does this mean I lose both my voice and data? One of the key benefits to VoIP is to enable
business customers to consolidate their voice and data network into a single infrastructure. Since the data network now carries
voice as well as data, businesses will need to address how the current capabilities and limitations of their data infrastructure.
In particular, how it offers resilience as the voice will now depend upon the availability of the data network. BT offers
resilient WAN products to ensure that VoIP can achieve the appropriate level of availability. Alternatively, ISDN resilience
options can be discussed.
Does Hosted IPT offer any new applications? Multimedia VoIP allows end users to take advantage of new ways of
working. This includes multimedia collaboration tools, such as “White boarding”, Document Sharing and Video. This
will enable business customers to reduce a lot of their indirect costs of travelling to meetings, and the ability to be able
to react to business changes and drivers more rapidly. Other applications using XML can be created by the customer themselves,
or by BT for a premium.
How much voice and data can I put on it? Any limits? The only limit is the size of the WAN. The underlying
data network will need to be appropriately dimensioned to carry both voice and data traffic. Each voice call equates to about
24Kbps of bandwidth on MPLS, and 40.2Kbps on ATM, including WAN overheads.
How are orders placed? Orders are placed through your BT Account Manager. A Customer
Requirements Form (CRF) needs to be completed, and a project manager assigned. The appropriate LAN/WAN upgrades need
to be ordered in parallel.
Will multimedia Numbers run in parallel with non-geographic numbers? Customers can specify whether they want to
run any of their existing geographic numbers in conjunction with the new Multimedia number range that has been allocated to
this product. After an initial connection fee, the first year is free of any charges. After the first year, there
will be a small quarterly charge per line for doing this.
What BT Network products will be appropriate for BT Hosted IPT services Any products that support Quality-of-Service
can be used for this product. In the UK this is currently Cellstream or Class-of-Service enabled IP-VPNs like BT Equip,
Metro, and IP Clear. Outside the UK, BT MPLS, BT Frame Relay and BT ATM are suitable.
Will BT Hosted IPT customer services inter-work? It is planned to be able to inter-work the BT VoIP services between
business customers, allowing the full capabilities of VoIP and Multimedia to inter-work between users of different VoIP services
on an “Extranet” basis. Special call rates are available for this type of call. Point-to-Point Video
calling on an extranet basis will become available in CY2005.
How will the service appear on the bill? The service will include configuration, subscription
and call charges. These can be integrated into existing BT “One-bills”.
Will a customer bill be separate for voice and data calls? The product will be billed separately
to the access pipes that the customer has purchased. Billing will be integrated with business customers' existing
billing options - this may only be by finally printing a consolidated bill. It will also be possible for business
customers to use their normal payment options.
What are the cost savings? For example on rental? Cost-saving will be at its greatest where organisations
make a lot of on-net calls between sites. Savings can be achieved as only one infrastructure is required, no IP PBX
is required, and no in-house management is needed.
Why should I incorporate both my voice and data? The arrival of VoIP will create the opportunity
for new applications and new ways of working that will affect all key areas of an organisation's activity. - Infrastructure
and maintenance savings. The advantage of incorporating both voice and data is that it saves the customer money in that s/he
only has one communications network to monitor and maintain. Planned upgrades, infrastructure investment, personnel resourcing,
and even facilities management all becomes more straightforward with an integrated network. - The most efficient Use of
Network Resources. In addition, the combination of voice and data on one network makes for more efficient use of the
bandwidth available within the customer's LAN/WAN. - Combining voice and data on an IP network is an applications enabler. The
inevitable globalisation of business will see the activities of a company's employees becoming increasingly dispersed. Voice
and data integration will offer applications that provide more personalisation, making us feel less restricted by spatial
separation. - Customer Care. Future services like Web-based call centres where the customer is required to understand complex
information, neither web nor phone may be suitable in isolation. Combining the two technologies ensures that the customer
sees relevant information yet can ask questions and discuss issues as necessary. - Knowledge Sharing. Incorporated
Voice and Data allows telephones to be closely integrated with directory access applications. This allows workers to fully
utilise corporate information services, and promotes better inter-working between employees. - Customer Relationship Management. Where
help is needed to reassure the customer in an e-commerce purchase, customers want to talk to a real person with information
at their fingertips, not to be redirected to a frequently asked questions page. Giving customers access to call centre agents
from the e-commerce web page could make the difference between winning a sale and losing to a competitor. Also, integrating
voice and data enables companies to deploy IP Contact Centres to further enrich their customer contact.
Does it cost in for smaller companies? This depends on how many sites the customer has
and how many calls between sites he makes. There is a requirement that the customer is able to scale to 100 users within
6 months of taking on the service. This will cost in for such customers.
Why should I use BT - my data router vendor has also offered me a service. BT Hosted IPT is a unique offer:
BT has an end-to-end solution capability
BT has an impressive background in delivering cutting edge technology solutions utilising the expertise in its world-renowned
laboratories at Adastral Park
BT has a proven background in both voice AND data communications
BT offers a strong capability for integrating with existing voice and data platforms
BT offers dedicated national support
BT offers national coverage (subject to individual site survey)
What is BT's position in the VoIP stakes? BT's investment in an IP infrastructure shows that BT is committed to
VoIP. It is also committed to ensuring that VoIP will work with our existing products. ICT is at the key of BT’s
product strategy – both for itself and for its customers. BT is deploying 30,000 IP Telephones for its own use,
and always installs IP Telephony at new sites.
What is BT's USP in this arena? Apart from it’s product range and experience, Effectiveness, Efficiency
& Increased Productivity - by providing the customer with a complete electronic solution which accommodates both large
and small sites.
What is the worst for my business if I don't go VoIP? Loss of business to VoIP technology-enabled
competitors Additional cost and difficulties with controlling a dispersed workforce with traditional technologies Lack
of price competitiveness when compared to VoIP-enabled customers with their associated lower cost bases will lead to a drop
in sales You will miss out on a golden opportunity to enhance, and rationalise your organisation's communications and data
infrastructure/policy for the new millennium.
My business Interfaces directly with customers what can VoIP offer me?
Customer Relationship Management. Where help is needed to reassure the customer
in an e-commerce purchase, customers want to talk to a real person with information at their fingertips, not to be redirected
to a frequently asked questions page. Giving customers access to call centre agents from the e-commerce web page could make
the difference between winning a sale and losing to a competitor.
Customer Care. Services like Web-based call centres where the customer is
required to understand complex information, neither web nor phone may be suitable in isolation. Combining the two technologies
ensures that the customer sees relevant information yet can ask questions and discuss issues as necessary.
Is the VoIP gateway included? No, Customers may have existing LAN maintenance arrangements and choose to purchase
a wires only WAN product. In this can they may be able to insert a Voice card in an existing router with the correct software
build for VoIP. Where they have no arrangement, the router choice will depend on voice signalling requirement and the size
of their connection, (channels to be supported)
Does VoIP Port support DPNSS features? The initial release supported DPNSS sections 1-6, simple call set-up. With
the MGCP enhancement, other features will become available.
Which IP PBX's are supported? At this time Cisco AVVID has been verified to work with BT VoIP Port.
