May 2001

A White PPaper
aper from

Hughes Software Systems

Plot 31, Electronic City,
Sector 18, Gurgaon 122 015,
INDIA
Tel: +91-124-6346666,
6455555
Fax: +91-124-6455150,
6348931
Website: www.hssworld.com
E-mail: info@hssworld.com
 VOICE OVER ATM – A STEPPING STONE TO CON-
VERGED BROADBAND PACKET NETWORK

COPYRIGHT INFORMATION
INFORMATION

© Copyright Hughes Software Systems, 2001

All information included in this document is under a licence agreement. This publication and its contents
are proprietary to Hughes Software Systems. No part of this publication may be reproduced in any form
or by any means without the written permission of
Hughes Software Systems
Plot 31, Electronic City,
Sector 18, Gurgaon 122 015, INDIA
Tel: +91-124-6346666, 6455555
Fax: +91-124-6455150, 6455155
Website: www.hssworld.com
E-mail: info@hssworld.com

TRADEMARKS
All the brand names and other products or services mentioned in this document are identified by the
trademarks or service marks of their respective owners.

DISCLAIMER
The information in this document is subject to change without notice and should not be construed as
commitment by Hughes Software Systems. Hughes Software Systems assumes no responsibility or makes
no warranties for any errors that may appear in this document and disclaims any implied warranty of
merchantability or fitness for a particular purpose.

2
 TABLE OF CONTENTS
1 Introduction ................................................................................................................... 5
1.1 Scope .................................................................................................................... 5
2 Why TTransmit
ransmit VVoice
oice over A ATM?
TM? ........................................................................................ 5
2.1 High Switching Speeds and LLow
ow PPacketization
acketization Delays ................................................ 5
2.2 Maximum Network Utilization .................................................................................. 5
2.3 Support for Extensive QoS ....................................................................................... 6
2.4 Effective TTraffic
raffic Management Capabilities ................................................................. 6
2.5 Support for Multiple Data TTypes
ypes ............................................................................... 6
3 Implementing VoA
VoATM: TTechnical
oATM: echnical Challenges ...................................................................... 6
3.1 Delay .................................................................................................................... 6
3.2 Packetization Delay ................................................................................................ 7
3.3 Buffering Delay ...................................................................................................... 7
3.4 Encoding Delay ...................................................................................................... 7
3.5 Silence Suppression ................................................................................................ 7
3.6 Signaling ............................................................................................................... 7
3.7 Synchronization ..................................................................................................... 8
4 Efficient TTransfer
ransfer of Voice over A
Voice TM: Using AAL2 ............................................................... 8
ATM:
4.1 Variable Bit Rate (VBR) ATM TTraffic
ATM raffic Class .................................................................. 8
4.2 Voice compression, Silence Detection/Suppression, Idle V oice Channel Removal .......... 8
Voice
4.3 Multiple V oice Channels with V
Voice arying Bandwidth on a Single A
Varying TM Connection ............. 8
ATM
5 LES – An Application of V oA
VoA
oATMTM ........................................................................................ 9
6 LES
LES:: A Hughes Software Systems Offering ....................................................................... 11
7 Conclusion ................................................................................................................... 13

3
 VOICE OVER ATM – A STEPPING STONE TO CON-
VERGED BROADBAND PACKET NETWORK

LIST OF FIGURES

F igure 1: LES Reference Model ............................................................................................ 10

F igure 2: Network Configuration Based on LES Reference Model ............................................ 11
-IWF ............................................................................................ 13
CP-IWF
F igure 3: LES Stack for CP

LIST OF TABLES
TABLES

Table 1: .......................................................................................................................... 10
Table 2: .......................................................................................................................... 12

4
1 Introduction
In a traditional circuit-switched network, a path
is dedicated to the transmission of data for the
duration of a call. This path cannot be used for
another call till one call is terminated. This path
cannot be used for another call even during peri-
ods of silence. Although the presence of a dedi-
cated path guarantees reliable and immediate
transmission of voice, the utilization of bandwidth
is not efficient.
Asynchronous Transfer Mode (ATM) is a multi-
service, high speed, scalable technology that has
been designed to carry voice and all types of data,
such as facsimile and multimedia.
Voice over ATM (VoATM) provides a platform for
the transport of plain old telephone service (POTS)
over a digital, broadband access network. VoATM
can support toll quality voice, a Quality of Service
(QoS) parameter associated with traditional circuit-
switched networks. In addition, VoATM provides
efficient use of bandwidth, high network efficiency
and therefore, low operational costs.
The convergence of data and voice traffic on a
single network yields significant economic benefits,
but, at the same time, it brings forth the technical
challenges of their co-existence.
1.1 Scope
This paper details the transmission of voice over
ATM and the challenges that need to be addressed
to implement this transmission. It highlights how
the characteristics of ATM Adaptation Layer 2 (AAL2)
are best suited to the efficient transfer of voice over
ATM. It also describes Loop Emulation Service (LES),
which is an application that uses ATM with AAL2,
and leverages the inherent operational efficiencies
of AAL2. In addition, the features of the LES stack
offered by Hughes Software Systems are also pre-
sented.
2 Why Transmit Voice over
ATM?
The WAN environment, today, is made up of a
variety of services fulfilling the requirements of a
multiplicity of applications. Each service provides
unique benefits for certain communication needs.
However, ATM, being a multimedia, multi-service
technology, has the potential to support all network
applications more efficiently than any other single
internetworking protocol. This leads to improved
performance and lower communication costs for
both the service provider and the user. With the
standards already evolved, ATM’s capacity to trans-
port voice and deliver high quality speech has been
recognized. The advantages of ATM are detailed
below:
2.1 High Switching Speeds and LLow
ow
Packetization Delays
ATM uses a 53-byte fixed length packet for voice
transmission. Of these 53 bytes, 48 bytes are used
for the payload and 5 bytes are used for the ATM
header. The packet format is based on data, voice
and video payload requirements.
In a fixed-length packet, the interpretation of
headers and trailers is not required. Therefore, the
fixed length of the packet enables high switching
speeds. In addition, a fixed-length packet improves
bandwidth utilization and therefore, reduces delay
variation. For timing-sensitive traffic, this eliminates
the possibilities of echoes in voice communications
and blips in videoconferencing sessions.
2.2 Maximum Network Utilization
ATM provides virtual networking, and therefore,
statistical multiplexing of traffic over any network
resource is possible. Virtual networking attempts to
utilize as much of the network as possible, thus
conserving relatively limited WAN fiber. Therefore,
network capacity is shared. This is important con-
sidering that most voice and data traffic uses only
5
 VOICE OVER ATM – A STEPPING STONE TO CON-
VERGED BROADBAND PACKET NETWORK
35 percent of the available bandwidth; the rest is
wasted during idle, quiet times.
Statistical multiplexing allocates resources and
bandwidth to a user only when required. This al-
lows the network to support more users, which is
usually twice the number of users that a Time Divi-
sion Multiplexed (TDM) network can support.
2.3 Support for Extensive QoS
ATM supports extensive QoS, which ensures the
reliable transmission of voice traffic across a net-
work.
2.4 Effective TTraffic
raffic Management
Capabilities
ATM switches have been designed with effective
traffic management capabilities. For example, call
admission control, usage parameter control and
traffic shaping are supported.
2.5 Support for Multiple Data TTypes
ypes
ATM supports the transfer of voice, data and
video signals. ATM can easily inter-network with
Public- Switched Telephone Network (PSTN). In
addition, different ATM Adaptation Layers (AALs)
provide support for different service classes capa-
bilities.
3 Implementing VoATM:
Technical Challenges
The networking technologies that transport voice
in the form of packets need to overcome some tech-
nical challenges. These challenges arise because
of the real-time and interactive nature of voice traf-
fic.
Some of the challenges that need to be ad-
dressed are:
6
3.1 Delay
To provide an acceptable QoS to subscribers,
the delay induced in a network should be mini-
mized. The increase in the network delay deterio-
rates the quality of voice and disrupts the normal
end-to-end interactive conversation.
Consider an example. In a 4-wire to 2-wire con-
version between a telephone handset and a com-
munication network, the transmitted voice signal is
reflected back. This is because of an impedance
mismatch between the handset and the network.
The reflected voice signal results in an echo, which
is transmitted to the ear piece of the handset. If the
round-trip delay in the network is minimal, this echo
is not detected. However, if there is increased round
trip delay across the network, which usually hap-
pens in long distance networks, the effect of the
echo is more pronounced. In such cases, the echo
can disrupt the normal flow of conversation. The
severity of the echo depends on the round-trip time
delay. If the round-trip delay is more than 30 milli-
seconds, the severity of the echo can disrupt nor-
mal conversation. In such cases, echo cancellor
circuits need to be added to the network.
The echo cancellor circuits allow the round-trip
delay in a network to increase to 150 milliseconds
without further degrading the voice quality. These
devices are expensive and complex and are most
suitable when the delay in a network is constant. In
a packetized voice network, the delay is not always
constant.
In a network spanning satellite links, the delay
can increase beyond 150 milliseconds. This results
in stilted conversations and clashing, in which both
parties try to talk at once. For improved toll quality
in such networks, the end-to-end network delay for
voice should be less than 25 milliseconds in na-
tional networks and less than 100 milliseconds in
an international context.
 The delay in ATM networks occurs due to the
following reasons:
3.1.1 Packetization Delay
This is the time taken to fill a packet before it is
transmitted. It is also called cell construction delay.
Normally, Pulse Code Modulation (PCM)-encoded
voice samples are transmitted at the rate of 64 Kbps.
This means that it takes around 6 milliseconds to
fill the entire 48-byte payload of an ATM packet.
This delay is directly proportional to the level of
voice compression employed; greater compression
implies greater delay. The delay depends on the
level of voice compression because the length of
an ATM packet is fixed. The delay in packetization
can be reduced either by partially filled packets or
by multiplexing several voice calls into a single ATM
Virtual Circuit Channel (VCC).
3.1.2 Buffering Delay
At the origin, voice traffic must be broken down
into small packets for transmission. These packets
are then reconstructed at the destination to re-con-
stitute the original voice call. The segmentation and
reconstruction of packets is managed by the Seg-
mentation and Reassembly (SAR) function. The re-
construction of packets must be done in real-time
to avoid distortion. Therefore, the packets carrying
voice traffic should be transmitted from the origin
to the destination in real time. This need to main-
tain a real-time delivery of voice traffic across the
network results in the buffering delay.
If there is a delay in the transmission of packets,
the SAR function might not have any data to pro-
cess. This is known as an under-run. An under-run
results in gaps in conversation. To prevent these
gaps, the receiving SAR function accumulates a
buffer of information before starting the reconstruc-
tion. To prevent an under-run, the buffer size must
exceed the maximum predicted delay. The size of
the buffer translates into delay. Each packet that
arrives at the emulated circuit’s line rate traverses
through the buffer. Therefore, to minimize the net-
work delay, the Cell Delay Variation needs to be
tightly controlled. The Cell Delay Variation has a
major impact on total network delay.
3.1.3 Encoding Delay
This delay occurs as a result of the encoding of
an analogue signal to a digital form. LD-CELP (Low
Delay Code Excited Linear Prediction), a derivative
of CELP is the most commonly used compression
algorithm used on any voice signal. This provides
a toll quality voice at 16kbps with low encoding/
decoding delay.
3.2 Silence Suppression
The characteristics of voice can be used to an
advantage for optimum use of available bandwidth.
Voice in its inherent nature is variable. When
there is no speech, a normal flow of conversation
consists of pauses between sentences and periods
of silence. Voice communication is half-duplex i.e.
one person is silent while the other speaks. These
two characteristics of voice can be used for saving
bandwidth. This is done by halting the transmis-
sion of packets during periods of silence. This is
known as silence suppression.
3.3 Signaling
Signaling includes the efficient utilization of re-
sources for the transfer of control and signaling
information.
A voice call consists of two parts, the voice sam-
ples and the signaling information, such as the
dialed number, the on-hook/off-hook status of the
call, and other routing and control information. The
signaling information can be encoded and can be
sent as Common Channel Signaling (CCS), Chan-
nel Associated Signaling (CAS) or Dual Tone Multi-
Frequency (DTMF) dialed digits.
If the voice traffic is sent across an ATM network,
7
 VOICE OVER ATM – A STEPPING STONE TO CON-
VERGED BROADBAND PACKET NETWORK
the signaling information can be transferred from
end to end. However, in CCS, where traffic from
one site requires switching to be delivered to two
or more end points, the signaling channels must
be terminated and interpreted at the ATM switch so
that the correct information is passed to the correct
end point.
The off-hook/on-hook signals can be used to
allocate active voice channels to active VCCs within
the network. This ensures that bandwidth is only
allocated to active calls and, therefore, a large
number of users can vie for the available resources.
3.4 Synchronization
When voice is transported across a network, it is
important to synchronize the data that is transmit-
ted from the speaker to the listener. This can be
achieved by employing some standard mechanisms
between point-to-point applications, such as Adap-
tive Clocking and Synchronous Residual Time
Stamping (SRTS). These adjust the clock rate at one
end of the circuit based on the clock rate of the
other end. However, these mechanisms are effec-
tive for master-slave environment or point-to-point
communication. Owing to the availability of the
global timing standards, it is easier and more prac-
tical to adopt an externally synchronized model in
multi-point services. In an externally synchronized
model, each node is synchronized to a single ex-
ternal clock source.
4 Efficient Transfer of Voice
over ATM: Using AAL2
To carry traffic over an ATM network, the appli-
cation must be adapted to ATM transport. This is
called ATM Adaptation Layer (AAL), and different
types of AALs have been standardized by ITU-T:
AAL1, AAL2, and AAL5.
In the development of ATM standards, AAL1 was
used to replace TDM circuits at fixed rates such as
1.536 Mbps (T1) or 2.048 Mbps (E1). It further
8
replaced the traditional digital voice circuits or 64K
circuits and provided a means to convey voice on
ATM backbones instead of TDM infrastructures.
However, AAL1 was not an optimal solution for
VoATM. This is because AAL1 supports the Con-
stant Bit Rate (CBR), which requires permanently
allocated bandwidth. This leads to inefficient utili-
zation of bandwidth.
In ITU-T Recommendation I.363.2 (1997), AAL2
was standardized. The recommendation included
a specific mandate to provide efficient VoATM serv-
ices.
4.1 F eatures of AAL2
In addition to the features supported by AAL1
adaptation protocol, AAL2 supports the following
features:
4.1.1 Variable Bit Rate (VBR) ATM TTraffic
ATM raffic
Class
AAL2 provides for the bandwidth-efficient trans-
mission of low-rate, short and variable packets for
delay-sensitive applications. It enables support for
both VBR and CBR applications within an ATM net-
work. AAL2’s VBR and CBR capabilities mean that
network administrators can take traffic variations
into account when designing an ATM network and
optimize the network to match traffic conditions.
4.1.2 Voice compression, Silence Detection/
Suppression, Idle Voice Channel Removal
Voice
VBR services enable statistical multiplexing for
the higher layer requirements demanded by voice
applications, such as compression, silence detec-
tion/suppression and idle channel removal.
4.1.3 Multiple V oice Channels with V
Voice arying
Varying
Bandwidth on a Single ATM Connection
ATM
AAL2 supports multiple user channels on a sin-
gle ATM virtual circuit. It also supports varying traf-
fic conditions for individual users or channels. The
structure of AAL2 supports the packing of short
length packets into one or more ATM packets. The
structure of AAL2 also incorporates a mechanism
to recover from transmission errors.
ATM Forum
The ATM forum has defined an advanced
mechanism for the transport of voice as a VBR
compressed stream. This is described in the
af-vtoa-0113.000 specification entitled ATM
Trunking using AAL2 for Narrowband Services,
which was completed in February 1999. This
approach combines the suppression of silence
in a conversation, with compression, and the
ability to multiplex multiple voice signals into
a single VCC. The multiplexing overcomes the
packetisation delay issues resulting from the
use of low bit rate voice encoding. It also pro-
vides for the interpretation of the voice signaling
channels and therefore, network switching
mechanism. This facilitates the building of
switched private or public voice networks.
4.2 User Benefits of AAL2
AAL2 for narrow band services can be used as a
part of ATM trunking. The major benefit of using
AAL2 is that it enables the saving of bandwidth.
This can be achieved by:
G compressing voice. When voice is compressed,
lesser bandwidth is allocated per call;
G releasing bandwidth when the voice application
does not need it. Bandwidth can be released when
the talker is silent or when the call is completed;
and
G routing and switching narrowband calls on a per
call basis. This leads to further improvements in
performance and efficiency.
ATM Forum
The ATM forum’s VTOA specification (af-
vtoa-113) and the Loop Emulation Service
(LES) over AAL2 exploit these user benefits of
AAL2 to provide a robust and scalable solu-
tion.
5 LES – An Application of
VoATM
Based on AAL2 for narrow band services, Loop
Emulation Service (LES) has been developed to pro-
vide an effective transport mechanism for voice,
voice-band data and fax traffic. The essential goal
of LES is to provide a fully functional transport
mechanism for Class 5 switch subscriber line serv-
ices over a broadband access network to the end
user. Data is transmitted over the broadband sub-
scriber line connection xDSL between the customer
premises and the service provider-switched tel-
ephone network.
5.1 LES Reference Model
A reference model of Loop Emulation Service
(LES), which uses AAL2, is depicted in Figure 1 on
the next page. The reference model consists of three
main components:
G AAL2-enabled Customer Premises Equipment/
Customer Premises Interworking function (CPE/CP-
IWF). The CPE prioritizes the voice packets over data
calls. This ensures toll quality voice delivery and
sends the packets over the Digital Subscribers Link
(DSL) line.
G ATM Virtual Circuits, which run through the ATM
network between the CP-IWF and the Central Of-
fice Interworking Function (CO-IWF), are Switched
Virtual Circuit (SVCs) or Permanent Virtual Circuit
(PVCs). These circuits carry:
❍ bearer traffic and Channel Associated
Signaling (CAS) using AAL2. CAS is carried in the
same AAL2 channel as the associated bearer traffic.
9
 VOICE OVER ATM – A STEPPING STONE TO CON-
VERGED BROADBAND PACKET NETWORK

F igure 1: LES Reference Model

Reference ❍ bearer traffic and CCS using AAL2. CCS for
the control of narrow band services is carried in a
specific AAL2 channel that does not carry bearer
traffic within the same ATM VCC as the associated
bearer traffic.
G Voice gateway, which depacketizes voice pack-

ets and converts them to a standards-based for-

mat (V5.2/GR303) for delivery to a service node.
A typical network configuration based on LES
has the components enumerated in the table be-
low. A typical network configuration based on LES
Reference Model in depided in figure 2 on the fol-
lowing page.

TABLE 1
Customer Telephony Equipment This includes telephones, PBX, fax, modem and key systems.

Integrated Access Device (IAD) This device controls upstream traffic to ensure proper QoS for
voice and data. It serves as the interface between the DSL
network service and the customer’s voice and data equipment.
The packetization of voice traffic takes place on this device.

DSL line This transports data and sends packetizsed voice to the
nearest carrier facility by using existing twisted-pair copper
loops. These loops must be able to support the distance and
qual ity requirements for DSL service to be offered.

Digital Subscriber Loop Access This provides ATM packet multiplexing. DSLAM Multiplexer
(DSLAM) terminates multiple DSL lines and aggre-
gates traffic from them.

ATM Switch This receives the traffic from the DSLAM and separates the
data from the voice packets. Data is sent to a data network,
for example Internet, and voice packets are sent to the Voice
gate way.

Class 5 Switch This is a telephony switch, which provides the dial tone, call
routing and other services. It also generates records used for
billing.

V5.2/GR303 This provides the interface to the Public Switched Telephone

Network (PSTN).
PSTN This is the public telephone network.

10
 LES uses the ATM AAL2 network to create an
extension cord between voice ports on the CP-IWF,
which is typically an Integrated Access Device (IAD),
and the corresponding voice ports on the Class 5
switch. The Class 5 switch terminates on the V5.2/
GR303-based Voice gateway. The interface between
the IAD and the first ATM-based edge switch or
DSLAM can be E1/T1 or xDSL. In either case, the
use of AAL2 allows multiple voice circuits to be car-
ried simultaneously. When the voice calls are not
active, the corresponding bandwidth is also avail-
able for use by the Internet connection.
5.2 Benefits of LES
LES leverages the inherent operational
efficiencies of AAL2 and SVCs.
G LES allows over-subscription on the V5.2-based
interface between the CO-IWF or a voice gateway
and the Class 5 switch. Using either PVCs or SVCs,
a large number of end terminals can share a lesser
number of available timeslots on the V5.2 inter-
face. The use of SVCs allows this process to be
even more efficient due to the inherent operational
efficiencies.
G LES offers residential and small business cus-
Figure 2: Network Configuration Based on
LES R eference Model
Reference
tomers reliable, toll-quality multi-line voice serv-
ices without requiring new wire installations or up-
grades and, therefore, utilizes the existing infra-
structure and management systems.
G LES ensures increased revenues for service pro-
viders by way of packaged voice and data services.
ATM Forum
An additional signaling option for LES
based on the use of MEGACO protocol has
been proposed to the ATM forum. The use of
H.248 signaling provides improved control
over the AAL2 media stream between the CP-
IWF and the CO-IWF. This facilitates the evo-
lution towards end-to-end packet voice net-
working and a more flexible approach to the
handling of analog supervisory signaling re-
quirements in international markets.
In the MEGACO/H.248 architecture, a Call
Agent (CA) provides external network intelli-
gence and control to DSLAMs/gateways,
Softswitches and CPE. This protocol achieves
a de-coupling of call processing from switch-
ing and routing. The CA is responsible for
creating connections in gateways and per-
forming signaling interworking
with the PSTN. The CA also sup-
ports packet voice and data over
a single integrated xDSL line.
6 LES: A Hughes
Software Systems
Offering
LES stacks are a part of the VoATM
offerings from Hughes Software Sys-
tems. These stacks drastically reduce
the developer’s time to market and
provide a true Rapid Application
Development (RAD) platform for
VoATM-based product deployment.
Besides being a modular, scalable,
11
 VOICE OVER ATM – A STEPPING STONE TO CON-
VERGED BROADBAND PACKET NETWORK
portable stack, an LES stack can be customised for
National specifications. It also supports Hot standby,
offers debugging and trace functionality and many
more features. The LES stack from Hughes Soft-
ware Systems leverages the inherent operational
efficiencies of AAL2 and SVCs and conforms to the
ATM forum’s Voice and Multimedia over ATM
(VMoA) standards.
The LES stack for narrow band services consists
of the signaling stack for CP-IWF, the signaling stack
TABLE 2
LAPV5-DL (Layer 2)
PSTN/PIE (Layer3)
Public Switched Telephone Network/PSTN
Interworking Entity
ELCP (Layer3)Emulated Loop Control Protocol
12
for CO-IWF, and their respective System Manage-
ment modules. The LES stack is depicted in Fig-
ure 3.
The LES stack entities and their key functions are
listed below:
6.1 System Management Module
G System Start-up
G System Restart
G Resource Management (AAL2 Channel alloca-
G LAPV5 based on LAPD (ETS 300 125 or Link
Access Protocol V5 – Data Link Q.920 and
Q.921 of ITU-T)
G Only point-to-point link access procedures are
applicable
G ISDN D-channel information frame relayed to
CP/CO-IWF interface
G Path setup for a call
G Communication of telephony events related to
subscriber ports
G Path release at the end of the call
G Call collision resolution
G Overload control at CO-IWF side
G Error handling in Message transmission PSTN is
present in the LES stack for CP-IWF and PIE is
present in the LES stack for CO-IWF.
Channel Allocation PProcedure
rocedure
G Allocation of bearer channel(s) for a call
G De-allocation of bearer channels at the end of
the call
G Error handling in message transmission User
Port Control
G Maintaining the functional states of user ports
(operational and non-operational)
G Non-urgent and immediate blocking of user
ports
G Coordinated unblocking of user ports
 access network. LES has taken the advantages of
FIGURE 3
high speed ATM all the way to the residence and
small home/office, thus eliminating the ‘last mile’
bottleneck. It is paving the way for the next genera-
tion multi-service, converged broadband packet
network.

VoATM and VoIP: The

Integration Challenge
With VoATM firmly established in the DSL ac-
cess network, and Voice over IP (VoIP) emerging
as an important force in core networks, the big-
gest challenge that emerges is the integration of
the two networks. Packetized voice from the
VoATM access network is converted to circuit form
and then re-packetized again for transmission
over the VoP network. This leads to incremental
delay. This incremental delay degrades the quality
of voice and hence, impacts the QoS. The in-
F igure 3: LES Stack for CP-IWF
CP-IWF ability of existing local exchange switches to deal
directly with packet voice traffic is a major bar-
tion/de-allocation)
rier to packet voice migration.
To streamline packet voice migration, a broad
6.2 LES
LES--V5 Interworking Module
G
consensus has been reached by the telecom in-
V5 PSTN Restart
G
dustry to separate the call processing function from
ELCP Restart
G
the physical switching function, and to connect
Resource Management Interworking (AAL2/V5
these two functions by using a standard protocol.
Bearer Channel allocation/de-allocation)
G V5.1 System Startup Interworking
This has resulted in the emergence of new com-
ponents called Softswitches, which have been iden-
The LES-V5 interworking module is present only
tified to meet the immediate and future market
in the LES stack for CO-IWF.
needs of such converged networks.

7 Conclusion
Voice over ATM (VoATM) has emerged as a so-
lution to service providers and network operators,
especially in the WAN environment, owing to its
low operational costs, multi-service capability and
higher network efficiency.
LES, an application of VoATM which uses AAL2,
provides a powerful and effective means for trans-
ferring different types of data over a broadband

13
 VOICE OVER ATM – A STEPPING STONE TO CON-
VERGED BROADBAND PACKET NETWORK

NOTES
NOTES

14
NOTES
NOTES

15
 VOICE OVER ATM – A STEPPING STONE TO CON-
VERGED BROADBAND PACKET NETWORK

The comprehensive set of software building

blocks from Hughes Software Systems con-
sists of both frameworks and protocol stacks
for the Voice over Packet domain.

Frameworks Stacks
Softswitch Framework MEGACO stack
Media Gateway Framework MGCP stack
Gatekeeper Framework SIP stack
SIP Server Framework H.323 stack
Mini Gateway Framework SIGTRAN stack

Hughes Software Systems is a key

supplier of communication
technologies for Voice over Packet,
Intelligent Networks and High-speed
Mobile Networks, and is fully focussed
on the needs of its customers to build
Next Generation Networks.

For clarifications, answers to queries or more information, please contact

Hughes Software Systems
Plot 31, Electronic City, Sector 18, Gurgaon 122 015, INDIA
Tel: +91-124-6346666, 6455555; Fax: +91-124-6455150, 6455155
Website: www.hssworld.com; E-mail: info@hssworld.com
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