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Compelling VoIP Applications
The VoIP technology only becomes useful
when compelling applications meet the needs of customers. Three
such applications are corporations that replace PBX systems, cable
operators offering telephony services using their plant, and video
conferencing. These existing applications are the driving factors
in allowing manufactures to make equipment, service providers
to offer services, and customers to increase their productivity.
Having an established market and a profitable business model allows
VoIP to begin addressing the next generation of applications which
then will allow the market to continue to grow.
Corporate
LAN-based telephony applications offer
attractive business models to consumers today. The most important
applications are the replacement of the traditional PBX, new client
side applications using the PC, and the reduction in maintenance
expenses for wiring changes.
LAN based PBX systems provide superior return on investment to
traditional PBX systems. Although initial equipment costs are
comparable, LAN PBXs typically cost much less than PBXs to install
because they use the existing data infrastructure (Category 5
cabling) rather than separate voice wiring. Administration is
also less burdensome because LAN and server administrators can
manage the system without the need for dedicated telephony technicians.
The most compelling reason businesses consider IP telephony-type
applications is for the integration of applications with voice.
Over the years, a significant amount of work has gone into computer
telephony integration (CTI) in traditional PBXs. These systems
began to offer application-programming interfaces such as Telephony
API (TAPI), Telephony Services API (TSAPI), and Java Telephony
API (JTAPI). This work has resulted in advanced call center functions,
including screen pops for agents and active call routing between
call centers.
Client (end station) products offer LAN telephony services through
the use of a software client on the user's PC, while others offer
telephone instruments that plug into the LAN. When there is a
need to relocate the equipment, it needs only to be unplugged
from one data port and plugged in at the new destination. This
process avoids the wiring changes typically done for convention
phones and thus VoIP clients reduce the cost of ownership. Over
time, these savings add up to the point that a LAN-based telephony
system can offer considerable savings over traditional PBXs.
PacketCable
PacketCable is the telephony architecture
for using VoIP over the Cable TV system. With buried Cable TV
plant passing hundreds of millions of homes worldwide, it is logical
to assume that cable operators desire to offer new services that
make use of their installed system.
The PacketCable standards (1.0, 1.1 and 1.2) were developed by
CableLabs®, the research group of the cable operators and
makes use of the same equipment used for the cable modem services.
The cable modem architecture, known as DOCSIS (Data over Cable
Service Interface Specification), is able to meet the requirements
of transporting VoIP packets. DOCSIS version 1.1 provides Quality
of Service (QoS), security features, and the prioritization of
packet traffic that is necessary for voice communication.
The PacketCable specification also incorporates the Network-based
Call Signaling (NCS) protocol for signaling voice calls over cable
networks. NCS leverages the existing Media Gateway Control Protocol
(MGCP) and the protocol is sometimes referred to as MGCP NCS.
NCS uses network-based call agents to negotiate cable-based IP
telephony calls.
Traditional telephones draw all the power they need from the phone
lines. Part of the reason that the public phone system has evolved
to such a reliable state, is that it is essentially immune from
the effects of power outages. Electrical utilities in most areas
do not offer this degree of unfailing reliability and this reality
is an important issue for the cable television plant.
Normally, head-end and customer-premises cable equipment rely
solely on the local electric company for their power and this
puts users at risk of losing phone service should a power outage
occur. There is not universal agreement among the cable operators
about offering "lifeline" telephony service.
Some Time Warner systems focus their telephony service towards
the "second" phone line in the home. They suggest that
the customer keep the existing primary line and then add the Time
Warner telephony service for a business line, fax line, or children's
phone. AT&T has taken a different approach by enhancing many
of their networks with alternate power sources thereby allowing
lifeline telephony services to their customers.
The PacketCable specifications are available at www.packetcable.com/specifications.html.
Video Conferencing
Video conferencing includes the use of
packetized voice and is an important application for the home
and business. There are several video conferencing standards and
one of the significant standards is known as H.32x where x can
be 0, 1, 2, 3, or 4 representing video conferencing on different
types of communications links. The H.32x model includes establishing
connections between the source and destination devices and this
method connection establishment has been adapted for use in VoIP
gateways.
Table 1. The
ITU-T H.32x Video Conferencing Standards
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H.320
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H.321
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H.322
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H.323
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H.324
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1990
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1995
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1995
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1996
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1996
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Network
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Narrowband switched
digital ISDN
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Broadband ISDN ATM LAN
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QoS packet switched
networks
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Non-QoS networks
(Ethernet)
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The analog phone
system
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Audio
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G.711
G.722
G.728
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G.711
G.722
G.728
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G.711
G.722
G.728
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G.711
G.722
G.728
G.723
G.729
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G.723
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Video
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H.261
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H.261/H.263
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H.261/H.263
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H.261/H.263
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H.261/H.263
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Control
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H.230/H.242
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H.242
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H.230/H.242
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H.245
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H.245
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Multiplexing
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H.221
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H.221
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H.221
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H.225
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H.213
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Comm. Interface
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I.400
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AAL
I.363
AJM I.361
PHY I.400
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I.400&
TCP/IP
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TCP/IP
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V.34 Modem
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A summary of the audio coding recommendations can be found in
the section Voice
Coding Algorithms.
The H.323 recommendation supports the largest number of audio
coding standards. In fact, the audio codecs of the other recommendations
are a subset of the audio codecs outlined in the H.323 recommendation.
In addition, the H.323 is the recommendation that is made for
a non-guaranteed bandwidth, packet switched networks such as the
Internet.
H.323 Video Conferencing
The H.323 recommendation covers the technical
requirements for audio and video communications services in LANs
that do not provide a guaranteed Quality of Service (QoS). The
scope of H.323 does not include the LAN itself or the transport
layer that may be used to connect various LANs. Only elements
needed for interaction with the Switched Circuit Network (SCN)
are within the scope of H.323.
H.323 defines four major components for a network-based communications
system: Terminals, Gateways, Gatekeepers, and Multipoint Control
Units (MCUs). The terminal must support voice transmission with
data and video transfer being an option.
Figure 1. The H.323 Protocol Stack
H.323 uses
the following standards as part of the video conferencing protocol
stack (Figure 1):
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Q.931 QoS - Quality of Service. When the
connection between devices is being established, the end stations
and each data link in the route negotiate to determine what
bandwidth is available, how much delay the application can
tolerate, and how much jitter there will be in the packet
arrival. Once the links agree to the QoS message, they must
guarantee those parameters for the duration of the video conferencing
session.
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H.245 - Control Channel Protocol. Provides
capability negotiation between the two end-points such as
voice compression algorithm to use, conferencing requests,
etc. The H.245 channels transport must be reliable (e.g. TCP,
SPX)
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H.225 RAS - Registration, Admission, and
Status (RAS) Protocol. Used to convey the registration, admissions,
bandwidth change and status messages between IP Telephone
devices and servers called Gatekeepers that provide address
translation and access control to devices.
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RTCP - Real-time Transport Control Protocol
(RTCP). Provides statistics information for monitoring the
quality of service of the voice call.
Control messages (Q.931 signaling, H.245
capability exchange and the RAS protocol) are carried over the
reliable TCP layer. This ensures that important messages get retransmitted
if necessary so they can make it to the other side. Media traffic
is transported over the unreliable UDP layer and includes two
protocols: RTP (Real-Time Protocol) that carries the actual media
and RTCP (Real-Time Control Protocol) that includes periodic status
and control messages. Audio and Video information is carried over
UDP because it need not be retransmitted because if a sound packet
is lost and then transmitted, it would most probably arrive too
late to be used for the real-time conferencing.
More Information
Additional VoIP seminars:
An Introduction to VoIP - An overview of the VoIP technology,
architecture, and the interconnection issues.
Voice Coding Algorithms - A description of the various
methods for digitizing speech.
VoIP Problems - Deployment of VoIP has been slower
than expected because of problems with underlying networks, standardization
issues, and network control devices.
In Summary:
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Corporate applications include replacing the
PBX, integrating phone applications with the corporate data
bases, and reducing the cost of moving phones between areas.
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PacketCable uses the DOCSIS 1.1 cable modem infrastructure
to provide telephony services by cable operators..
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Video Conferencing is an important application
and its control mechanisms are being incorporated into VoIP
gateways.
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