Accelerating the Deployment of VoIP

and VoATM
Overview
The economic advantages of packet voice are driving both the access and core
voice networks away from circuit switching towards packet. The industry
continues to debate whether the future of these packet networks will be based on
pure ATM, pure Internet protocol (IP), IP over asynchronous transfer mode
(ATM), IP over multiprotocol label switching (MPLS), or a combination thereof.
There are advantages to both ATM and IP and reasons for choosing each. This
tutorial will explore the role of next- generation switches which, as they become
widely adopted for both access and core networking, must be able to handle voice
traffic over both IP and ATM networks for future extensibility as the debate
continues and must have the features necessary to interwork with existing public
switched telephone network (PSTN).

Topics
1. Introduction
2. Voice over Packet Architecture
3. Why Voice over IP?
4. Why Voice over ATM?
5. Designer Considerations for Voice over Packet
6. Elements of a Next-Generation Switching Platform
7. Switching Platform/Media Gateway
8. Signalling Gateway
9. The Softswitch/Media Gateway Controller
10. Application Server (AS) and Services
11. Conclusion
Self- Test
Correct Answers

Glossary

1. Introduction
Carriers are moving voice services to packet networks both to reduce upfront and
operational costs and to provide more value-added services in an increasingly
competitive environment. A recent study by a major carrier found that packet
equipment was 70 percent less expensive than traditional voice equipment, and
data access lines were 60 percent to 80 percent cheaper than voice lines.
Maintenance of packet networks was 50 percent less expensive, while
provisioning was 72 percent lower. However, consolidation of voice from the
PSTN onto packet networks has, in the past, proven difficult and therefore has
happened very slowly. International voice-overIP call volumes, which provide
the most compelling business case for packet telephony, are still a drop in the
ocean of international telephony traffic but have experienced phenomenal growth
since 1998, according to a recent report by Washington, D.C.based research
group TeleGeography. According to the "TeleGeography 2001" report, which
contains results of an exclusive survey of major voice-over-packet (VoP)
providers in 1999 and 2000, international Internet telephony traffic volumes
reached 1.7 billion minutes in 1999a growth rate of more than 1,000 percent
from 1998. IDC projected more than 9 billion minutes of voice traffic to travel
over worldwide packet networks in 2000, exceeding 135 billion minutes in 2004.
Service revenue is projected at $1.6 billion in 2000 and $18.7 billion in 2004.
While it is clear that VoP is growing, there is still considerable debate about
whether the underlying network technology will be ATM or IP. At the edge of the
network the choice, driven primarily by the regional Bell operating companies
(RBOCs), is ATM. An ATMdominated access network is clearly in the works
because until recently IP did not provide the quality of service (QoS) guarantees
that are so important for voice. Although QoS protocols such as DiffServ,
resource reservation protocol (RSVP), and MPLS have been implemented, most
of today's IP traffic is actually being carried over ATM. However, in the long term
with the recent success of MPLS it appears that pure IP over lambda may be the
winner. And certainly, IP at the application layer and the desktop is a more than
just a viable near-term situation.
In addition to the challenges in architecting networks with end to end QoS,
service providers must ensure that the rollout of such networks cause no
disruption to their existing voice service revenue, which currently represent
about 80 percent of their overall revenue source. With more than $650 billion of
worldwide revenue generated by traditional voice and fax services and more than
$250 billion installed base of traditional equipment infrastructure in the United
States alone, service providers must deploy next-generation packet switches that
seamlessly interconnect and competitively function as time division multiplexing

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(TDM)based PSTN switches as well as support voice over ATM (VoATM) and
voice over IP (VoIP).
It appears that most carriers, especially the larger incumbent carriers will start
the migration to packet telephony on the trunk side first (Class-4 tandem) and
eventually migrate to the access (Class 5). This migration model is similar to the
migration from analog switches to digital switches, which started in the late
1970s. Carriers first started on the inner network (i.e., tandem) and then moved
outwards to the Class 5.
The architecture for VoP, the reasons for choosing IP or ATM, and considerations
in next-generation system design need to be understood to accelerate VoP
deployments.

2. Voice over Packet Architecture

In principle, two basic technologies are used for building high-capacity networks:
circuit switching and packet switching. In circuit-switched networks, network
resources are reserved all the way from sender to receiver before the start of the
transfer, thereby creating a circuit. The resources are dedicated to the circuit
during the whole transfer. Control signaling and payload data transfers are
separated in circuit-switched networks. Processing of control information and
control signaling such as routing is performed mainly at circuit setup and
termination. Consequently, the transfer of payload data within the circuit does
not contain any overhead in the form of headers or the like. Traditional voice
telephone service is an example of circuit switching.

Circuit-Switched Networks
Carrier-class next-generation switches need to be high-capacity fault-tolerant
TDM and VoP switches. They must be designed to significantly enhance the
economics of providing traditional TDMbased voice and data services as well as
help service providers migrate to a packet-based telecom network (based on VoIP
and VoATM) and generate new competitive services. Service providers deploying
next-generation switches can cap their investment in traditional circuit switches
and migrate to a converged switching infrastructure that allows them to reduce
the number of overlay network platforms and provide profitable voice and data
services over packet networks. See Figure 1.

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Figure 1. A High-Capacity TDM Switch Capable of Packet

Switching

Since most of the core packet networks today are ATMbased, but most likely
migrating to IPbased, the most future-proof investment is in next generation
switches that can be deployed to transport voice on both ATM and IP networks
supporting protocol layers as outlined in Figure 2.
Figure 2.

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3. Why Voice over IP?

Support for voice communications using IP, which is usually called VoIP, has
become especially attractive to consumers given the low-cost, flat-rate pricing of
the public Internet.
VoIP is the ability to make telephone calls and access service over IPbased data
networks with a suitable QoS and superior cost/benefit to PSTNbased calls.
Today, most of the VoIP implementations are carried over ATMbased transport
as shown in the second column of Figure 2.
The benefits of implementing VoIP are mostly consumer-based and can be
divided into the following three categories:

Cost reductionIP is everywhere. It is on our desktops and it is what

the Internet is based on. Many people view the Internet as a "free
transport" for data and voice services. With the introduction of
Net2Phone and other similar "free" services, many people are now
making phone calls over the Internet. In addition, businesses and
individuals have turned to higher-quality commercial products and
services to make voice calls based on IP. The prevalence of IP nodes
and the abundant supply of better IPbased switches and routers
continue to reduce the cost of providing VoIP.

Simplification and consolidationAn integrated infrastructure

that supports all forms of communication could allow more
standardization and could reduce the total equipment complement.
The differences between the traffic patterns of voice and data offer
further opportunities for significant efficiency improvements.
Universal use of IP for all applications, voice and data, holds out the
promise of both reduced complexity and more flexibility.

Advanced applicationsEven though basic telephony and facsimile

are the initial applications for VoIP, the longer-term benefits are
expected to be derived from multimedia and multiservice applications.
For example, Internet commerce solutions can combine World Wide
Web access to information with a voice call button that allows
immediate access to a call center agent from a PC. In addition, voice is
an integral part of conferencing systems that could also include shared
screens, white boards, etc. Combining voice and data features into new
applications will provide the greatest returns over the longer term.

Utilizing an IPbased network for voice traffic can offer advantages to consumers
of reduced costs, simplification, and consolidation due to the proliferation of IP
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based applications and devices at the desktop. These advantages are compelling
for consumers and are driving service providers to consider VoIP
implementations. In contrast, VoP over the ATMbased network offers distinct
advantages directly to service providers and are still much more prevalent today.

4. Why Voice over ATM?

ATM, from the start, was designed to be a multimedia, multiservice technology.
Although ATM has been accepted by service providers for its ability to deliver
high-speed data services, until recently its potential for deploying voice services
was overlooked. With the competitiveness of today's market though, network
operators and service providers have been continuously striving to reduce
operating costs and lift network efficiency and have turned to the ATM network
to achieve these goals.
With hundreds of millions of dollars of ATM equipment infrastructure in the
United States alone, service providers have recognized that significant economies
of scale can be achieved if the data traffic and voice traffic are integrated onto a
single network. In order to achieve this, service providers have started to use the
circuit emulation services (CESs) of ATM switches to carry full or fractional E1/T-1 circuits between end points. These CES mechanisms treat voice as a
constant stream of traffic encoded as a constant bit rate (CBR) stream. In
actuality though, voice is a combination of bursts of speech and silence and this
increases the complexity of VoP.
The ATM Forum and International Telecommunications Union (ITU) came up
with several advanced mechanisms to improve the efficiencies of transporting
voice traffic, including:

ATM trunking using AAL1 for narrowband services

ATM trunking using AAL2 for narrowband services

IP over ATM (AAL5)

Loop emulation service using AAL2

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Table 1 summarizes the benefits of utilizing the different methods for

transporting VoATM.
Table 1.
Standards
CES

Voice
Compression
No

Silence
Removal
No

Channel
Suppression
No

Switched
Concentration
No

BDCES

No

No

Yes

No

ATM
trunking
using AAL1

No

No

Yes

Yes

VoIP over
ATM

Yes

Yes

No

No

AAL2

Yes

Yes

Yes

Yes

5. Design Considerations for Voice over

Packet
Adding voice to packet networks requires an understanding of how to deal with
system level challenges such as interoperability, call control and signaling, voice
encoding, delay, echo, reliability, density, and performance of all the elements
that make up the next-generation switching platform.

6. Elements of a Next-Generation
Switching Platform
The vision for a next-generation switching platform is a distributed architecture
in which media gateway/bearer transport platform, signaling, call control, and
application elements are divided into separate logical network components (see
Figure 3), communicating with one another through the use of intraswitch
protocols such as Megaco, media gateway control protocol (MGCP), and
SCTP/M3UA. This distributed model allows service providers to scale their
network to support hundreds of thousands of subscriber ports per node. In this
concept, voice traffic is directed between the traditional voice network and the
new packet-based networks by the media gateway. The call control is handled by
a softswitch, and the features and services are handled by an application
platform. In reality, the softswitch (or call control platform) may support some of
the more popular services without requiring a separate application platform. An
example of this type of service is 7/10 digit routing, which would be handled
directly by the call control platform. Other examples of where the application
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platform may not be involved are caller name delivery, local number portability
(LNP), and E-800 service. These services are already implemented in the PSTN
using service control points (SCPs). In these cases, the call control platform will
send intelligent network (IN)/transactional capabilities application part (TCAP)
queries over the signaling system 7 (SS7) network to existing SCPs.
Figure 3. Elements of a Next-Generation Switching Platform

Some vendors enable one or more of these logical network elements to be

deployed on the same physical platform. There are some inherent advantages to
this "integrated" model especially with platforms that support up to 100,000
subscriber ports (DS0s) per bearer platform/media gateway, and allow efficient
execution of the softswitch and signaling gateway software. Benefits also include
cost savings and deployment and operation simplicity. In the "integrated" model,
the need for intraswitch protocols such as Megaco and MGCP are not required;
however interswitch protocols such as RTP/UDP/IP (for MG to MG) and BICC
(for SG to SG) are always required for interoperability with the other ends. See
Figure 4 for relevant inter-switch protocols.

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Figure 4. Interoperability: Call Control, Signaling, and Bearer

Platforms

7. Switching Platform/Media Gateway

Sometimes referred to as a media gateway, the switching/bearer transport
platform is hardware that sits at the edge of a network and takes in a packet
and/or circuit containing voice or data traffic and switches it to a voice or data
network. Media gateways come in many different flavors depending on the
breadth of definition. The most popular consist of Class 4 and Class 5
replacement functionality on a voice over digital subscriber line (VoDSL)
gateway. Media gateways are part of the physical transport layer and are
controlled by a call control engine or softswitch (also called a media gateway
controller), which provides instructions to direct voice traffic. Media gateways are
at the heart of the transformation of the voice network, as they are essential to
migrating voice traffic onto a packetized network. As part of packetizing voice
traffic, a media gateway adapts (by using compression and echo cancellation) the
packetized traffic, creates and attaches an IP header and/or ATM header, and
sends the packet through the network according to instructions provided by the
softswitch.
While a media gateway can be physically located almost anywhere within the
network, depending on the network architecture and the features it is intended to
support, all media gateways share certain features including the following:
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ScalabilityA media gateway needs to be able to scale to support

hundreds of thousands of telephone calls (called DS0s, running at 64
Kbps per line) to parallel the scalability of the existing PSTN switches.

Support for several types of access networksNeeded support

includes wireless, fiber, cable, and copper. In addition to electrical
interfaces, a media gateway needs to support a variety of optical
interfaces (including OC3, OC12, OC48, and OC192 speeds).

Carrier-class reliabilityAlso known as five nines (99.999 percent)

reliability (i.e., less than five minutes of downtime per year) and
network equipment building standards (NEBS) certification (the
Telcordia quality rating for meeting environmental stress tests),
reliability is extremely important to service providers because it
enables them to fulfill customer contracts. Most carriers cite reliability
as the impetus to transform their current architecture.

Interworking functionalityMedia gateways are capable of

supporting multiple voice and data interface protocols and
compatibility between them by converting circuit traffic to packet
traffic and vice versa.

InteroperabilityMost networks are a compilation of multivendor

solutions, making interoperability essential for success.

Control supportTo enable communication between the media

gateway and a softswitch. The most common languages (or protocols)
emerging for communication between these devices are MGCP and
Megaco.

SwitchingA media gateway must handle switching and media

processing, based on an ATM, IP, or TDM switching fabric.

Voice transportationThere are 3 transport standards used for

transporting voice traffic: TDM (traditional circuit-switch method),
ATM AAL1/AAL2, and IPbased RTP/RTCP (over ATM or pureIP
transport).

A packetized approach to transmitting voice faces a number of technical

challenges that spring from the real-time or interactive nature of the voice traffic.
Some of the challenges that need to be addressed include the following:

EchoEcho is a phenomenon where a transmitted voice signal gets

reflected back due to unavoidable impedance mismatch and fourwire/two-wire conversion between the telephone handset and the
communication network. Echo can, depending on the severity, disrupt

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the normal flow of conversation and its severity depends on the roundtrip time delay if a round-trip time delay is more than 30 ms the echo
becomes significant making normal conversation difficult.

End-to-end delayVoice traffic is most sensitive to delay and is

mildly sensitive to variations in delay (jitter). It is critical that end-toend delay is minimized to hold interactive communications. Delay can
interfere with the dynamics of voice communication, in the absence of
noticeable echo, whereas in the presence of noticeable echo, increasing
delay makes echo effects worse. When delay reaches above 30 ms, echo
canceller circuits are required to control the echo.

Packetization delay (or cell construction delay)Packetization

delay is the time taken to fill in a complete packet/cell before it is
transmitted. Normal G.711 pulse code modulation (PCM) encoded
voice samples arrive at the rate of 64 Kbps, which means it can take
approximately 6 ms to fill the entire 48-byte payload of an ATM cell.
The problem can be addressed either with partially filled cells or by
multiplexing several voice calls into a single ATM virtual circuit
channel (VCC).

Buffering delaySometimes, due to delay in transit, some cells

might arrive late. If this happens the ATM segmentation and
reassembling (SAR) function provided by the adaptation layer might
have to under run with no voice data to process which results in gaps in
conversation. To prevent this, the receiving SAR function would
accumulate a buffer of information before starting the reconstruction.
In order to ensure that no under runs occur the buffer size should
exceed the maximum predicted delay. The size of the buffer translates
into delay, as each cell must progress through the buffer on arrival at
the emulated circuit's line rate. This implies that the cell delay
variation (CDV) has to be controlled within the ATM network.

Silence suppressionVoice, by its nature, is variable. In fact, a

typical conversation has a speech activity factor of about 42 percent
due to pauses between sentences and words where there is no speech in
either direction. Also, voice communication is half-duplex, which
means that one person is silent while the other speaks. One can take
advantage of these two characteristics to save bandwidth by halting the
transmission of cells during these silent periods. This is known as
silence suppression.

Compression algorithmsG.726 adaptive differential pulse code

modulation (ADPCM) and G.729 adaptive code excited linear
prediction (ACELP) are the two major compression algorithms that are
used. The benefit of compression is efficient use of bandwidth. Most

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voice packets are transmitted today using G.711 encoding that does no
compression and therefore adds further delay.

8. Signaling Gateway
A signaling gateway is hardware and software that provides the connection from
a softswitch and media gateway to the SS7 network. The signaling gateway
receives/sends the call control instructions needed between the SS7 network and
the softswitch, typically through stream control transmission protocol (SCTP)
and MTP Level-3 user adaptation layer (M3UA) protocols. This allows the
softswitch to process and communicate call control instructions to a media
gateway. A signaling gateway can either stand-alone or be integrated with a
softswitch/media gateway. In the traditional circuit-switched telephone network,
a legacy switch provides the interface directly to the SS7 world, essentially acting
as a signaling gateway.

9. The Softswitch/Media Gateway

Controller
A softswitch, also referred to as a "call agent" or "media gateway controller," is
software that provides the call control and signaling for the next-generation
network. The softswitch moves the service intelligence out of the switch into a
database or application server, connects those databases, and ultimately provides
the "brains" or operating system for the next-generation voice network. A
softswitch ensures that a call is routed through the network to the proper
destination and that features from the existing advanced intelligent network
(AIN) such as 1-800 and LNP, as well as new multimedia services, are applied to
calls as appropriate. While the softswitch architecture is similar to the AIN
databases in an SCP, a softswitch provides more robust functionality and is
distinguished by providing control to more than one type of switchincluding
TDM, ATM, IP, etc.while today's AIN controls only TDMbased switches. This
architecture is inherently more flexible and scalable than the architecture of
today's circuit switches.
There is significant debate in the industry about the definition of a softswitch, its
role within the network, how it should interface with other gateways and
softswitches, and how it should interface with the IP and SS7 networks. At the
most basic level, a softswitch must contain call-control features and a signaling
interface to the SS7 network. Call control relates to the setup and teardown of
calls, including service selection ("which services apply to this call?") and call
routing ("where will this call be sent?"). In addition, a softswitch must provide
call authentication ("what calls is this line allowed to make?"), authorization, and
accounting services by accessing information available in the existing SS7
network. The SS7 signaling interface, which allows the softswitch to
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communicate with today's SS7 network, is in some cases distributed to a standalone hardware system called a signaling gateway. Today's softswitches typically
operate on the Sun Solaris operating system and include features such as the
following:

Media independenceto make the software agnostic regarding the

switching fabric (ATM, IP, TDM, etc.)

Interoperabilitywith multiple vendors' media gateway products,

the existing PSTN, and off-the-shelf hardware platforms

Reliabilityto carrier standards (five 9s of reliability)

Support for multiple signaling and control protocols

including emerging and established standards such as ISUP, BICC,
SIP, and MEGACO/H.248

Scalabilityto meet carrier network requirements, supporting

thousands of call attempts, also known as busy hour call attempts
(BHCA) and simultaneous calls

Open application programming interfaces (APIs)or "hooks"

into third-party software applications and services

10. Application Server (AS) and Services

Finally, without services, next-generation switches would not be able to generate
the voice revenue that currently provides 80 percent of overall service provider
revenue. The following Class-4 and Class-5 services need to be supported by
these switches. As stated before, several of these services may be implemented in
the softswitch (call control platform) without the need for an external AS. The
more complex services such three-way conferencing may require the need for an
AS with multimedia support. Until ASs become more capable in terms of
supporting more complex services and providing robust easy-to-use service
creation environments (SCEs), the need to deploy these services outside of the
softswitch environment is less compelling.
Dial tone*
Basic dialing*
Basic 7/10 routing
Announcements
Billing record creation
Call blocking/allow
Call transfer
Call forward
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CF busy
CF no answer
3-way calling
Toll restriction
Outbound restriction
Calling name delivery
Calling number delivery
Int. Dial: 011
Premium rate: 900/976
Toll free: 8xx
Operator: 0/00
E911
LNP
Primary interexchange carrier (PIC)
CALEA
Selective call reject
Selective call accept
Remote call forwarding
Speed dialing 30
Anonymous call reject
Caller ID block
Automatic callback
Automatic recall
Call waiting
Calling identity delivery on call waiting
Customer-originated trace
Distinctive ringing/call waiting
Selective call acceptance
Selective call forwarding
Selective call rejection

11. Conclusion
Voice packet telephony is a reality today, although, as an industry, there still is a
lot of work ahead. The larger incumbent carriers are starting the migration to
packet telephony on the trunk side first (Class-4 tandem) and will eventually
migrate to the access side (Class 5). This migration model is similar to the
migration from analog switches to digital switches, which started in the late
1970s, and offers a proven path for migration to new technologies. The full
migration to packet-based Class 5 systems will happen when the inner network
becomes packet-based and when differentiated Class-5 services become available.
And the services must go beyond currently available PSTNbased services for
packet telephony to become truly compelling. On this journey, the debates over
VoIP and VoATM will continue. While VoATM makes sense today for some
carriers, especially the larger incumbents, VoIP is the longer-term goal especially
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with MPLSbased QoS becoming available. For some carriers, VoIP is the answer
today as the consumer benefits are persuasive.
Service providers looking to deploy VoP will be best served if they choose a
solution that addresses the issues of interoperability, call control and signaling,
voice encoding, delay, echo, reliability, density, and performance of all the
elements that make up the switching platform. And they should look for solutions
that deal with these issues for TDM switching, as well as VoIP and VoATM.

Self-Test
1. A recent study by a major carrier found that packet equipment was 70 percent
more expensive than traditional voice equipment.
a. true
b. false
2. Since most of the core packet networks today are ATMbased, but most likely
migrating to IPbased, it only makes sense to deploy next-generation
switches that can be deployed to transport voice on both ATM and IP
networks supporting protocol layers.
a. true
b. false
3. Support for voice communications using IP, which is usually called
___________, has become especially attractive given the low-cost, flat-rate
pricing of the public Internet.
a. VoATM
b. VoP
c. VoDSL
d. VoIP
4. ___________ from the start, was designed to be a multimedia, multiservice
technology.
a. AAL
b. IP
c. ATM
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d. PSTN
5. In the new architecture, the call control will be handled by a ____________
a. softswitch
b. Megaco
c. MGCP
d. SCTP
6. ____________ are part of the physical transport layer and are controlled by
a call control engine or softswitch (or media gateway controller), which
provides instructions to direct the traffic.
a. Switches
b. Headers
c. Media gateways
d. Platforms
7. A media gateway can be physically located almost anywhere within the
network.
a. true
b. false
8. A ____________ is hardware and software that provides a connection from
a softswitch and media gateway into the SS7 network
a. Megaco
b. header
c. signaling gateway
d. transport layer
9. A softswitch moves the service intelligence out of the switch into a database or
application server, connects to those databases, and ultimately provides the
"brains" or operating system for the next-generation voice network.
a. true

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b. false
10. While VoATM makes sense today for some carriers, especially the larger
incumbents, VoIP is the longer-term goal.
a. true
b. false

Correct Answers
1. A recent study by a major carrier found that packet equipment was 70 percent
more expensive than traditional voice equipment.
a. true
b. false
See Topic 1.
2. Since most of the core packet networks today are ATMbased, but most likely
migrating to IPbased, it only makes sense to deploy next-generation
switches that can be deployed to transport voice on both ATM and IP
networks supporting protocol layers.
a. true
b. false
See Topic 1.
3. Support for voice communications using IP, which is usually called
___________, has become especially attractive given the low-cost, flat-rate
pricing of the public Internet.
a. VoATM
b. VoP
c. VoDSL
d. VoIP
See Topic 3.

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4. ___________ from the start, was designed to be a multimedia, multiservice

technology.
a. AAL
b. IP
c. ATM
d. PSTN
See Topic 4.
5. In the new architecture, the call control will be handled by a
____________.
a. softswitch
b. Megaco
c. MGCP
d. SCTP
See Topic 6.
6. ____________ are part of the physical transport layer and are controlled by
a call control engine or softswitch (or media gateway controller), which
provides instructions to direct the traffic.
a. Switches
b. Headers
c. Media gateways
d. Platforms
See Topic 7.
7. A media gateway can be physically located almost anywhere within the
network.
a. true
b. false
See Topic 7.
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8. A ____________ is hardware and software that provides a connection from

a softswitch and media gateway into the SS7 network.
a. Megaco
b. header
c. signaling gateway
d. transport layer
See Topic 8.
9. A softswitch moves the service intelligence out of the switch into a database or
application server, connects to those databases, and ultimately provides the
"brains" or operating system for the next-generation voice network.
a. true
b. false
See Topic 9.
10. While VoATM makes sense today for some carriers, especially the larger
incumbents, VoIP is the longer-term goal.
a. true
b. false
See Topic 11.

Glossary
ACELP
adaptive code excited linear prediction
AIN
advanced intelligent network
AS
application server
ATM
asynchronous transfer code

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BHCA
busy hour call attempts
CBR
constant bit rate
CDV
cell delay variation
CES
circuit emulation services
IN
intelligent network
IP
Internet protocol
ITU
International Telecommunications Union
LNP
local number portability
MGCP
media gateway control protocol
MPLS
multiprotocol label switching
NEBS
network equipment building standards
PIC
primary interexchange carrier
PSTN
public switched telephone network
QoS
quality of service
RBOC
regional Bell operating companies
RSVP
resource reservation protocol
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The International Engineering Consortium

20/21

SAR
segmenation and reassembling
SEC
service creation environments
SCTP
stream control transmission protocol
SS7
signaling system 7
TCAP
transactional capabilities application part
TDM
time division multiplexing
VCC
virtual circuit channel
VoDSL
voice over digital subscriber line
VoP
voice over packet

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Copyright
The International Engineering Consortium

21/21