May 2001

Paper A White Paper 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 CONVERGED BROADBAND PACKET NETWORK

INFORMATION COPYRIGHT 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.

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TABLE OF CONTENTS
1 2 Introduction ................................................................................................................... 5 1.1 Scope .................................................................................................................... 5 Transmit Voice ATM? Why Transmit Voice over ATM? ........................................................................................ 5 2.1 2.2 2.3 2.4 3 Low Packetization High Switching Speeds and Low Packetization Delays ................................................ 5 Maximum Network Utilization .................................................................................. 5 Support for Extensive QoS ....................................................................................... 6 Traffic Effective Traffic Management Capabilities ................................................................. 6

Types 2.5 Support for Multiple Data Types ............................................................................... 6 VoA oATM: Technical Implementing VoATM: Technical Challenges ...................................................................... 6 3.1 3.2 3.3 3.4 3.5 3.6 Delay .................................................................................................................... 6 Packetization Delay ................................................................................................ 7 Buffering Delay ...................................................................................................... 7 Encoding Delay ...................................................................................................... 7 Silence Suppression ................................................................................................ 7 Signaling ............................................................................................................... 7

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3.7 Synchronization ..................................................................................................... 8 Transfer Voice ATM: Efficient Transfer of Voice over ATM: Using AAL2 ............................................................... 8 4.1 4.2 Variable Bit Rate (VBR) ATM Traffic Class .................................................................. 8 ATM Traffic Voice Voice compression, Silence Detection/Suppression, Idle Voice Channel Removal .......... 8

5 6 7

Voice Varying ATM 4.3 Multiple Voice Channels with Varying Bandwidth on a Single ATM Connection ............. 8 VoA oATM LES – An Application of VoATM ........................................................................................ 9 LES: A Hughes Software Systems Offering ....................................................................... 11 LES: Conclusion ................................................................................................................... 13

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VOICE OVER ATM – A STEPPING STONE TO CONVERGED BROADBAND PACKET NETWORK

LIST OF FIGURES
F igure 1: LES Reference Model ............................................................................................ 10 F igure 2: Network Configuration Based on LES Reference Model ............................................ 11 CP-IWF F igure 3: LES Stack for CP-IWF ............................................................................................ 13

TABLES LIST OF TABLES
Table 1: Table 2: .......................................................................................................................... 10 .......................................................................................................................... 12

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1

Introduction
In a traditional circuit-switched network, a path

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 transport voice and deliver high quality speech has been recognized. The advantages of ATM are detailed below:

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 periods of silence. Although the presence of a dedicated path guarantees reliable and immediate transmission of voice, the utilization of bandwidth is not efficient. Asynchronous Transfer Mode (ATM) is a multiservice, 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 circuitswitched 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.

Low 2.1 High Switching Speeds and Low 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.

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 presented.

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 considering that most voice and data traffic uses only

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VOICE OVER ATM – A STEPPING STONE TO CONVERGED 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 allows the network to support more users, which is usually twice the number of users that a Time Division Multiplexed (TDM) network can support.

3.1

Delay

To provide an acceptable QoS to subscribers, the delay induced in a network should be minimized. The increase in the network delay deteriorates the quality of voice and disrupts the normal end-to-end interactive conversation. Consider an example. In a 4-wire to 2-wire conversion between a telephone handset and a communication 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 happens 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 milliseconds, the severity of the echo can disrupt normal 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 national networks and less than 100 milliseconds in an international context.

2.3

Support for Extensive QoS

ATM supports extensive QoS, which ensures the reliable transmission of voice traffic across a network.

Traffic 2.4 Effective Traffic 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

Types Support for Multiple Data Types

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 capabilities.

3 Implementing VoATM: Technical Challenges
The networking technologies that transport voice in the form of packets need to overcome some technical challenges. These challenges arise because of the real-time and interactive nature of voice traffic. Some of the challenges that need to be addressed are:

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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-constitute the original voice call. The segmentation and reconstruction of packets is managed by the Segmentation and Reassembly (SAR) function. The reconstruction 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 maintain 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 process. 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 reconstruction. 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 network 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 transmission of packets during periods of silence. This is known as silence suppression.

3.3

Signaling

Signaling includes the efficient utilization of resources for the transfer of control and signaling information. A voice call consists of two parts, the voice samples 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), Channel Associated Signaling (CAS) or Dual Tone MultiFrequency (DTMF) dialed digits. If the voice traffic is sent across an ATM network,

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VOICE OVER ATM – A STEPPING STONE TO CONVERGED 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.

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 Constant Bit Rate (CBR), which requires permanently allocated bandwidth. This leads to inefficient utilization of bandwidth. In ITU-T Recommendation I.363.2 (1997), AAL2 was standardized. The recommendation included a specific mandate to provide efficient VoATM services.

3.4

Synchronization

4.1

F eatures of AAL2

When voice is transported across a network, it is important to synchronize the data that is transmitted from the speaker to the listener. This can be achieved by employing some standard mechanisms between point-to-point applications, such as Adaptive 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 effective for master-slave environment or point-to-point communication. Owing to the availability of the global timing standards, it is easier and more practical to adopt an externally synchronized model in multi-point services. In an externally synchronized model, each node is synchronized to a single external clock source.

In addition to the features supported by AAL1 adaptation protocol, AAL2 supports the following features: 4.1.1 Class ATM Traffic Variable Bit Rate (VBR) ATM Traffic

AAL2 provides for the bandwidth-efficient transmission of low-rate, short and variable packets for delay-sensitive applications. It enables support for both VBR and CBR applications within an ATM network. 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/ Voice Suppression, Idle Voice Channel Removal VBR services enable statistical multiplexing for the higher layer requirements demanded by voice applications, such as compression, silence detection/suppression and idle channel removal. Voice Varying 4.1.3 Multiple Voice Channels with Varying ATM Bandwidth on a Single ATM Connection AAL2 supports multiple user channels on a single ATM virtual circuit. It also supports varying traffic conditions for individual users or channels. The

4 Efficient Transfer of Voice over ATM: Using AAL2
To carry traffic over an ATM network, the application 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

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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’s VTOA specification (afvtoa-113) and the Loop Emulation Service (LES) over AAL2 exploit these user benefits of AAL2 to provide a robust and scalable solution.

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 provides for the interpretation of the voice signaling channels and therefore, network switching mechanism. This facilitates the building of switched private or public voice networks.

5 LES – An Application of VoATM
Based on AAL2 for narrow band services, Loop Emulation Service (LES) has been developed to provide 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 services over a broadband access network to the end user. Data is transmitted over the broadband subscriber line connection xDSL between the customer premises and the service provider-switched telephone network.

5.1 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

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/CPIWF). 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 Office 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.

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.

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VOICE OVER ATM – A STEPPING STONE TO CONVERGED BROADBAND PACKET NETWORK

Reference F igure 1: LES Reference Model

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 packets and converts them to a standards-based format (V5.2/GR303) for delivery to a service node. A typical network configuration based on LES has the components enumerated in the table below. A typical network configuration based on LES Reference Model in depided in figure 2 on the following page.

Customer Telephony Equipment Integrated Access Device (IAD)

TABLE 1 This includes telephones, PBX, fax, modem and key systems. 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 aggregates 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 PSTN

This provides the interface to the Public Switched Telephone Network (PSTN). This is the public telephone network.

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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 carried simultaneously. When the voice calls are not active, the corresponding bandwidth is also available for use by the Internet connection.

tomers reliable, toll-quality multi-line voice services without requiring new wire installations or upgrades and, therefore, utilizes the existing infrastructure and management systems.
G

LES ensures increased revenues for service providers 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 CPIWF and the CO-IWF. This facilitates the evolution towards end-to-end packet voice networking and a more flexible approach to the handling of analog supervisory signaling requirements in international markets. In the MEGACO/H.248 architecture, a Call Agent (CA) provides external network intelligence and control to DSLAMs/gateways, Softswitches and CPE. This protocol achieves a de-coupling of call processing from switching and routing. The CA is responsible for creating connections in gateways and performing signaling interworking with the PSTN. The CA also supports packet voice and data over a single integrated xDSL line.

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 interface. 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-

6 LES: A Hughes Software Systems Offering
LES stacks are a part of the VoATM offerings from Hughes Software Systems. These stacks drastically reduce the developer’s time to market and provide a true Rapid Application Figure 2: Network Configuration Based on Reference LES Reference Model Development (RAD) platform for VoATM-based product deployment. Besides being a modular, scalable,

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VOICE OVER ATM – A STEPPING STONE TO CONVERGED 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 Software 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 ,

for CO-IWF, and their respective System Management modules. The LES stack is depicted in Figure 3. The LES stack entities and their key functions are listed below:

6.1
G G G

System Management Module

System Start-up System Restart Resource Management (AAL2 Channel alloca-

TABLE 2 LAPV5-DL (Layer 2)
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 ISDN D-channel information frame relayed to CP/CO-IWF interface Path setup for a call Communication of telephony events related to subscriber ports Path release at the end of the call Call collision resolution Overload control at CO-IWF side 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.

G

PSTN/PIE (Layer3) Public Switched Telephone Network/PSTN Interworking Entity

G G

G G G G

ELCP (Layer3)Emulated Loop Control Protocol

Channel Allocation Procedure Procedure G Allocation of bearer channel(s) for a call G De-allocation of bearer channels at the end of
G

the call Error handling in message transmission User Port Control Maintaining the functional states of user ports (operational and non-operational) Non-urgent and immediate blocking of user ports Coordinated unblocking of user ports

G

G

G

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FIGURE 3

access network. LES has taken the advantages of 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 generation multi-service, converged broadband packet network.

VoATM and VoIP: The Integration Challenge
With VoATM firmly established in the DSL access network, and Voice over IP (VoIP) emerging as an important force in core networks, the biggest 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 inCP-IWF F igure 3: LES Stack for CP-IWF tion/de-allocation) ability of existing local exchange switches to deal directly with packet voice traffic is a major barrier to packet voice migration. To streamline packet voice migration, a broad consensus has been reached by the telecom industry to separate the call processing function from the physical switching function, and to connect these two functions by using a standard protocol. This has resulted in the emergence of new components called Softswitches, which have been identified to meet the immediate and future market needs of such converged networks.

6.2
G G G

LESLES -V5 Interworking Module

V5 PSTN Restart ELCP Restart Resource Management Interworking (AAL2/V5

Bearer Channel allocation/de-allocation) G V5.1 System Startup Interworking The LES-V5 interworking module is present only in the LES stack for CO-IWF.

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Conclusion

Voice over ATM (VoATM) has emerged as a solution 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 transferring different types of data over a broadband

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VOICE OVER ATM – A STEPPING STONE TO CONVERGED BROADBAND PACKET NETWORK

NOTES NOTES

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NOTES NOTES

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VOICE OVER ATM – A STEPPING STONE TO CONVERGED BROADBAND PACKET NETWORK

The comprehensive set of software building blocks from Hughes Software Systems consists of both frameworks and protocol stacks for the Voice over Packet domain. Frameworks Softswitch Framework Media Gateway Framework Gatekeeper Framework SIP Server Framework Mini Gateway Framework Stacks MEGACO stack MGCP stack SIP stack H.323 stack 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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