Wireless
Voice over IP - Voice over Wireless LAN
(VoWLAN)
A
VoIP-enabled telephone uses standard TCP/IP
protocols to send digitized voice calls
across a conventional, IP-routed infrastructure
(like the Internet.) A special signaling
protocol is used to set up the end-to-end
call. VoWLAN is the implementation of VoIP
over a wireless network.
What is VoIP Telephony
?
VoIP (also
called "IP Telephony") is transmission
of a telephone call using digitized voice
data carried by Internet Protocol (IP).
This can be accomplished using the services
of either a wired Ethernet or a Wi-Fi
wireless LAN. Alternatively, one could
make a phone call using the Public Switched
Telephone Network (PSTN)
with a conventional telephone.
In the PSTN,
intelligence and control belong to
the central office and the network itself.
With VoIP, intelligence is in the phone
set or other edge device. This allows
the VoIP user access to much more sophisticated
and productive features, beyond those that
can be offered in the centralized PSTN environment.
Some of these capabilities include the following:
Location
Awareness: Since the VoIP user goes through a login
and registration process when they attach
their phone to the network they can move
to different locations and still be reached
through the same telephone number.
Presence
Management: In the PSTN a called party is either "on
hook" or "off hook". With VoIP the availibility
of a called party takes on a variety of additional
possible characteristics. For example,
if someone is in a meeting they may be able
to receive text messages, but not talk on
the phone. The VoIP features that releate
to various called party status responsoses
are implemented is called Presence Management.
Call
Forking: A user
can register at more than one location
so an incoming call will ring, for example,
at their desk, at home, and on their cell
phone.
Unified
Messaging: Access
to all messaging formats from a single
mobile terminal including: voice mail,
fax, pager, and text messaging. The user
has the ability to manage their data and
information services under a single umbrella.
User
Profile Management: Each user has a unique profile file that
identifies their membership in groups, their
availability times, the types of media they
can receive, and (in a billing environment)
how their account should be billed.
Why Move From PSTN
Telephony to VoIP? - The VoIP Value Proposition
Voice-over-IP telephony
as an alternative to standard PSTN PBX systems
provides a number of advantages and is evolving
to be the implementation-of-choice for all
new telephone systems. Some areas of benefit
to consider include the following:
Application
Advantage: The natural integration between VoIP telephony
and Ethernet/Wi-Fi networking allows for the
implementation of software applications that
increase user productivity and improve the
customer experience. The most basic VoIP applications
integrate database functions with telephone
functions to present customer records on-screen
at the time an incoming call is received. Easy
voice conferencing with accompanying multimedia
(slide shows, video, shared virtual whiteboarding)
improve in-house productivity. Telecommuting
and the virtual team enjoys a seamless mechanism
for collaboration and communication. A cohesive
reporting and management structure lets the
system manager have the information they need
for capacity planning and troubleshooting.
Return-on-Investment
(ROI) Advantage: Because the infrastructure
for VoIP telephony is the same as that used
for computer data networking there is the
need to install and manage only one communications
system (the "network") instead
of wiring for phones and also creating a
computer data system.
The "Converged
Network" Advantage: The historical separation of voice
networks (the phone company), video networks
(cable and satellite systems), and data (the
Internet) is gradually disappearing. This "convergence" of
voice, video, and data systems can be seen
in 3G cellular
where VoIP combines with web browsing on PDA's and
phones, and camera phones provide picture
and movie capabilities. Today's investment
in VoIP technology becomes a foundation for
tomorrow's converged network services.
Understanding Voice-over-IP: "How Does It
Work?"
To the user of a VoIP
telephone the experience of placing a call
can be identical to the PSTN alternative: You
pick up the phone, hear a dial tone, dial the
number, the called party's phone rings, and
you talk. Of course, the experience may be
enhanced by picking a desired called party
from an on-screen list, a sales lead database,
or other integrated function.
Behind the scenes
a number of things must happen to make a call
possible. This is true for both PSTN telephony
as well as VoIP systems.
Voice-over-IP
PSTN
How does the network know where each
called party is located?
Each station registers its location with
a server when it attaches to the network
Stations are hard-wired with circuits
going back to the telephone company central
office
How does a calling party discover the
location of a called party?
The called party's location is resolved
(using DNS) into an Internet domain name
Central office switch equipment connects
to the numeric identity of the called party's
phone number
What communication standards are used
for the registration and location services?
Signaling messages may be sent in accordance
with the IETF Session Initiation Protocol
(SIP) standards or the ITU-T H.323 standards
Signaling and inter-switch communication
is performed in accordance with the ITU-T
Signaling System Number 7 (SS7) communication
standards used by telephone companies worldwide
VoIP Signaling Messages
In both VoIP and PSTN telephony
there are signaling messages used
to communicate information concering the connection
between a calling party, the switched network
system (either packet-switched IP or telephone
switches between telephone company central
offices), and the called party. These signaling
messages include requests and replies that
perform the tasks necessary to establish, manage,
and terminate a call:
"ESTABLISH CALL FORWARDING"
or "END CALL FORWARDING"
"CONNECT TO THE LAST CALLING
NUMBER"
"SETUP A CALL" to which the
response should be "CALL SETUP IS PROCEEDING"
"CALL ACCEPTED" (now the parties
can talk) or "CALLED PARTY IS NOT AVAILABLE"
(perhaps they're on vacation) or "CALLED
PARTY BUSY" (they're using their phone -
busy signal) or "CALL REJECTED" (the calling
party's number has been blocked)
"INITIATE 3-WAY CALLING",
"INDICATE THAT A CALL IS WAITING", "TRANSMIT
CALLER ID INFORMATION"
"CALL TERMINATED"
The PSTN uses a set of rules
(protocols) referred to as Signaling System
Number 7 or "SS7" which defines, among
other things, the signaling messages. VoIP
initially used a set of rules called H.323 and now also uses an alternative set of signaling
rules called Session Initiation Protocol
or "SIP". Special servers can translate between
the protocols used in SS7, H.323, and SIP allowing
calls to be placed across systems utilizing
different signaling methods.
Identifying the Calling and Called Parties
Connect802, in San
Ramon, California, is in Area Code 925. Our
local central office exchange is 552, and that
makes our phone number 925-552-0802. This structure
for a PSTN end-station identifier is defined
in an ITU-T standard called E.164. E.164 telephone
numbers are physically mapped to individual
telephones because the circuits are wired from
the telephone company central office, to the
customer’s
premises, and to the telephones themselves.
When you call 1-925-552-0802 you reach Connect802’s
main switchboard.
With VoIP each
calling set must register its identity with
the network. The identity can be an E.164
numeric value, an email address, or a URL.
H.323 records this information in a server
called a Registration,
Admission, and Status (RAS) server.
As a side note, don’t
confuse the acronym RAS as used here in the
VoIP environment with the same acronym used
in the Microsoft Windows’ networking
world where it stands for “Remote Access
Server” (the two are completely unrelated.)
SIP calls the server a SIP
Proxy or sometimes
the term Registration
Server is
used. The terminology differs from H.323 to
SIP, but the functionality is the same.
H.323 represents information using a binary
code called ASN.1 (Abstract Syntax Notation.1)
whereas SIP uses plain ASCII text. The difference
is transparent to the user of the VoIP system
but, as a decision maker choosing a VoIP product
you should know that H.323 is a more closed
system, requiring special software development
for the addition of new features. SIP, on the
other hand, was designed to work easily with
web-enabled devices and is the signaling protocol
used in 3G cellular phone systems. All other
things being equal, we recommend going with
SIP as opposed to H.323.
The switches in the network fabric
provide the intelligence, management,
and control and, hence, the available
features and capabilities of the network
The network is simply a communication
link (ie: the Internet) and intelligence
(along with features and capabilities)
are a function of the edge devices
using the network
The
scale of the network is dependent on
the hardware used by the Telco central
office to provide connectivity for users
Scale is independent of hardware from
the standpoint that features are a
function of the user's edge devices
and not inherent in the network fabric
itself
Bandwidth must be allocated for all
devices (ie: separate wiring)
Bandwidth is shared (ie: IP routing
of packets across the Internet)
A telephone number refers to a specific
physical location
The telephone number is mapped to
an IP address and is not location-dependent
Telephone numbers are just that, a
string of numbers
The Telco network (PSTN) is designed
to manage analog voice circuits
The Internet is designed to manage
packet-switched connections
VoIP
Products and Services are available through
Connect802 Technology Partners
utilizing
the
Suite Spot Predictive Site Survey for
Accurate Wireless VoIP Design
Click
on a logo to
go to the Connect802
Technology
Partner page
TeleSym
develops Voice-over-Internet-Protocol (VoIP)
systems for the mobile enterprise. The company's
SymPhone system delivers secure, high quality
Internet phone calls to Windows-based PC,
laptops, Pocket PCs, and mobile handheld
devices over any broadband IP network
Definitions and Descriptions
of VoIP that you might find on the Web:
VoIP technology
allows people to make phones calls
that travel over the Internet rather
than solely across wires owned by long-distance
phone companies. Such calls can be made
from telephone systems that tap into the
Internet, and from PCs.
U.S. government
reports estimate that the United
States, Canada, Guam, Bermuda and Trinidad
will run out of 10-digit numbers by the
year 2025, driven by demand for cell phones,
faxes and other devices. The coming
crunch has led at least one industry
organization to draw up a plan for a 12-digit
future that could add some 640 billion
new numbers to the pool.
VoIP's efficiencies
come through its use of packet-switching
technology, which breaks up communications
into small bits that are dispersed
to find the fastest path across the network
and recombined at the end point. Traditional
telephony, by contrast, is "circuit-switched," creating
a dedicated channel for the duration
of the call.
VoIP
technology offers an alternative
to POTS. In
order to understand the changes that VoIP
technology makes to POTS, one should have
a good grasp of POTS. POTS have traditionally
been highly regulated by both state public
utility commissions and the FCC. With
the passage of the Telecommunications Act
of 1996, the FCC relaxed many of its POTS
regulations which were based on specifically
defined geographic calling areas. By
using a mobile technology such as VoIP,
users do not have a defined calling area,
nor are they required to identify the source
of the call. These differences have
sparked the current regulatory debate. While
POTS and VoIP exist as separate technologies,
in the future VoIP may dominate the
telephone industry.
Academic technologists
are concerning themselves with wireless
delivery of VoIP, and a new Institute
for Electrical and Electronics Engineers
(IEEE) standard designed for longer wireless
links, colloquially known as 802.16 or WiMax.
The first, wireless VoIP, works much like
traditional VoIP, only transmits voice packets
over the Internet Protocol on a wireless
network. This technology generally requires
more bandwidth than a wireless network is
capable of providing, but still seems to
be drawing considerable interest.
Because current
802.11a and 802.11g wireless standards
can only send signals up to 300 feet, some
experts hail the WiMAX (802.16) standard
as the "next
big thing" in wireless telecommunications,
and note that the technology's ability
to send wireless signals at 70 mpbs
for up to 30 or 40 miles could henceforth
revolutionize wireless network design.
Most users implementing
VOIP these days are primarily concerned
about voice quality, latency and interoperability.
All are fundamental quality-of-service
considerations that companies need to
deal with before they can even begin justifying
the move to VOIP. But some security organizations
are cautioning users about the dangers
of unsecured VOIP services. For instance,
in an August 2001 paper on its Web site,
the Bethesda, Md.-based SANS Institute warned
of privacy- and authentication-related
issues stemming from VOIP services and urged
users to apply the same precautions they've
used to protect their data services.
Voice over
Internet Protocol (VoIP) is a rapidly emerging
technology for voice communication that uses
the ubiquity of IP-based networks to deploy
VoIP client devices—such as desktop IP phones,
mobile VoIP-enabled handheld devices, and
VoIP gateways—in an increasing
number of businesses and homes around
the world. (Microsoft Windows Embedded
Developer Center)