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Education: ATM WHITE PAPERS

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Preamble: Broadband Exchange

The ATM Forum has always been associated with the development of ATM specifications and indeed this is still the mainstay of its charter. However, with ATM being one of a number of technologies being deployed in broadband networks today, there is a strong need for these technologies to combine effectively together. In order to accelerate this process, The ATM Forum initiated the Broadband Exchange to bring together the industry to work towards common goals. Each event explores a different theme associated with broadband technologies and applications. This white paper is the result of presentations and dialogue at Broadband Exchange: Next Generation Networks, Prague, April 2002.

Hosted in conjunction with The ATM Forum’s Technical Committee meeting, the Broadband Exchange demonstrates the increasing level of industry collaboration towards driving forward developments in broadband networking. At the spring 2002 exchange, The ATM Forum and ETSI outlined their joint vision for Next Generation Networks, backed up with viewpoints from leading industry analysts, vendors and service providers.

This white paper serves to document the findings of the meeting. It therefore represents views expressed by participants and speakers at the Broadband Exchange rather than those solely of The ATM Forum.

For more information on the subject of Next Generation Networks and to see the presentation material from the Broadband Exchange in Prague, please visit The ATM Forum website at:

http://www.atmforum.com/pages/meetingsfsbbx-sum.html

Next Generation Networks – The Voice of the Future 

Definition

The term “Next Generation Networks” is seemingly very broad indeed. Almost any development in networking could potentially be placed under this banner. In fact, the definition used by ETSI’s NGN Starter Group places the emphasis on providing operators with a step-by-step manner to create, deploy and manage innovative services. 

“NGN is a concept for defining and deploying networks, which, due to their formal separation into different layers and planes and use of open interfaces, offers service providers and operators a platform which can evolve in a step-by-step manner to create, deploy and manage innovative services”

However, NGN is more commonly associated with voice – a vision for the future of packet-based voice networks as part of the evolution from today’s TDM circuit switched voice. This makes sound commercial and technical sense. For the majority of operators offering a mix of voice and data services, voice is still contributing over 80% of revenues. Besides, data networks have embraced the requirements of packet-based networks from the outset.

Looking towards the collaboration required, The ATM Forum and ETSI TIPHON have recently signed a Memorandum of Understanding (MoU) making the timing of this Broadband Exchange optimal. ETSI TIPHON (which stands for Telephony Internet Protocol Harmonization over Networks) was established in 1997 but took a major step in 2000 to re-focus on long-term multi-protocol interworking issues. 

The Business Case for NGN

Increasing levels of openness and deregulation will force competition between service providers on something other than price.  For the incumbent voice service providers there is always the option of doing nothing. It could be argued that the existing voice networks are essentially free, having been depreciated over the last 20 years or more. So where is the compelling case to replace these established networks? 

Source: Alcatel

Although like-for-like replacement probably does not make commercial sense, these networks are expensive to maintain and upgrade. Products serving the data market have made enormous technology leaps and have achieved economies of scale as demand has grown. Therefore, in simple terms, the market demand drives both innovation and cost reduction. The same can’t be said for TDM equipment and voice switches. It’s been a while since there have been significant technology breakthroughs here or anything approaching a quantum leap! They are also low on innovation in that they do not lend themselves to rapid introduction of new features. They do not follow the type of price/performance curves seen for data products over the last few years. 

Looking then at the networks rather than the equipment, TDM voice networks are typically less than 40% utilized. This level of utilization is costing carriers in wasted resources. Moving to ATM-based voice could raise that instantly up to 65-70%. A converged voice and data network could take that up towards the 100% mark. Moreover, a competitive voice carrier using NGN technology could implement new features in weeks or months. An incumbent operator typically has to wait 1-2 years for new software features to be implemented on their voice switches. TDM technology, the basis for today’s voice networks and some of the wide area data networks, does not provide the flexibility and scalability needed for next generation networks.  The most conservative approach still would incur the cost of enhancing and maintaining the existing infrastructure, and the costs of deploying a new infrastructure solely for voice is an equally unattractive option. These factors are creating a gradual move away from dedicated voice networks to acceptance of a converged packet network that can support multiple services (voice included).

To address this, The ATM Forum and TIPHON bring a unique perspective to IP telephony standardization efforts: a focus on scalability, reliability, and performance requirements for worldwide deployment by major national and international carriers. The work could be seen to deliver on the following:

.                      Convergence: Bringing IP and PSTN/ISDN network architectures together for voice services

.                      Replacement: Allowing for migration to and replacement of PSTN/ISDN by IP

.                      Improvement: Specification of QoS Service Levels, together with a range of new revenue-generating service offerings (as opposed to supplementary services which extend basic voice telephony)

Role of ETSI TIPHON

The collaboration between The ATM Forum and ETSI TIPHON reflects the work that TIPHON has been doing in the field of packet voice and specifically the integration between the new and old world voice networks. As an ETSI project (EP), TIPHON has to reflect the needs of ETSI’s membership and so recognizes the practical realities of the marketplace and the need to provide a mechanism for the support of end-to-end differentiated service over multiple technologies. The original objective of ETSI Project TIPHON was to bring together the interests of existing and evolving communications networks with those of the Internet age, thereby forming a link between the traditional telecommunications and IP worlds. The standards market has evolved so much since TIPHON’s inception, that the question is no longer limited to the simple problem of evolving “old switched networks” to “new packet networks”, but also to the interconnection of networks implementing differing VoIP technologies. Key areas of consideration are:

.                      The service-level interworking between traditional Switched Circuit Networks (SCNs)' especially those served by PSTN, ISDN or GSM networks, and emerging Next Generation Networks (NGN). 

.                      The complex, yet extremely important, area of multi-network interworking, across multiple administrative and technology domains. 

.                      The challenge of providing public communications services in a heterogeneous environment, which TIPHON addresses by defining a generic means of creating services that is independent of any specific underlying network technology – regardless of whether it is circuit or packet based.

The end goal is the widespread deployment of IP-based telephony giving rise to a wave of new applications and services that will fundamentally change the way people use technology to communicate.

Among the many specifications TIPHON has developed are:

.                      • An abstract architecture, which can serve both the migration to, and implementation of, the infrastructure for an all-packet solution. 

.                      • A meta protocol, used to generate protocol mappings and profiles for industry standards associated with any given communications network technology (including H.323, SIP, H.248/MEGACO)

.                      A set of documents related to Quality of Service aiming at improving the quality of VoIP speech, including: QoS general aspects, QoS signaling, network design guide, QoS measurements, and others.

.                      Advanced Test Suites (ATS), Protocol Implementation Conformance Statements (PICS), and Test Purposes (TP) documents for conformance testing of each protocol profile addressed (SIP, H.225, H.245; H.248, etc.).

Like any development, it has a phased approach. Release 3 of TIPHON is the latest and most comprehensive to date in a series of planned releases, which, over time, will offer the means of providing a wide range of multimedia services extending well beyond those available on today’s conventional networks. Release 3 defines mechanisms for the provision of an integrated range of basic telephony services over a mixture of IP, ATM, PSTN, ISDN and GSM wireless transport networks.

TIPHON Release 3 implementations draw together industry protocols such as H.323, SIP, Megaco/H.248 and BICC to enable services to be offered in a coherent way between multiple service providers and across transport networks utilizing differing technologies.

The documents are available on the ETSI web site at http://www.etsi.org/tiphon

Role of The ATM Forum

The ATM Forum VMOA (Voice and Multimedia over ATM) working group has long been associated with the transport of real time data over an ATM network. Having evolved from the original telephony focus, its remit is to represent the needs of a range of real time multimedia applications, voice included. Much of the original work focused on emulation of traditional leased line services as used by TDM multiplexers in private voice networks. Known as Circuit Emulation (CES), it used AAL1 and offered a transparent path across an ATM network that would appear as if it were a leased line. However, to take advantage of the true nature of ATM it was AAL2 that provided the capability for variable bit rate voice services. 

A more recent step in this direction has been the definition by The ATM Forum of the specification for the Loop Emulation Service using AAL2 for Narrowband Services. This fulfills a market need for an efficient transport mechanism to carry voice, voice-band data, fax traffic, ISDN B-channels and D-channels over a broadband subscriber line connection, such as xDSL, HFC or wireless, between customer premises and a Service Node, such as provided by the public switched telephone network. Voice transport will include support for compressed voice and non-compressed voice together with silence removal.

Source: The ATM Forum This specification describes the procedures and signaling required to support the efficient transport of voice band services across an ATM network between two Interworking Functions (IWF) located at a customer’s premises (CP-IWF) and at a service provider’s premises (CO-IWF). It specifies the use of ATM virtual circuits with AAL2 to transport bearer information and signaling. The virtual circuits used may be PVCs, SPVCs, or SVCs. The specification supports the transport of common channel signaling (CCS) information as well as channel associated signaling (CAS) information.

The CO-IWF and CP-IWF described in this specification are functional units, which may be implemented as stand-alone devices, as parts of larger devices, or distributed among several devices. This specification does not dictate the implementation of any one of these configurations. The Service Node shown above may represent a Class 5 PSTN switch delivering public switched telephone services over a narrowband Service Node Interface (SNI), or it may represent a PBX in a private network. The SNI may also be packet-based. The Service Node may connect to the CO-IWF via one or more physical interfaces. Alternatively, the CO-IWF functionality may be present as an integral part of a Service Node, in which case there is no external appearance of the physical interface between CO-IWF and Service Node.

The physical connection between the CP-IWF and the ATM network is typically provided by a DSL, HFC or wireless link. The ATM network may be a full network, a single ATM switching element or simply a direct interconnection between a CO-IWF and a CP-IWF. The ATM virtual circuits through the ATM network between the CP-IWF and the CO-IWF shall be SVCs, PVCs, or SPVCs carrying:

.                      • Bearer traffic and CAS using AAL2, where CAS is carried in the same AAL2 channel as the associated bearer traffic

.                      • Bearer traffic and CCS using AAL2, where CCS for the control of narrowband services is carried in a specific AAL2 channel that does not carry bearer traffic, within the same ATM VCC as the associated bearer traffic

Although initiated in early 2000, the ATM Forum is now revisiting its Next Generation Network Access (NGNA) specification.  This will extend the earlier work on AAL1 and AAL2 voice services into a more generalized model for ATM in the next generation access network.

CPE Considerations

Given that interoperability must exist between the old and new worlds, then legacy voice communications equipment, especially at the CPE level, will need to both connect with other legacy voice handsets over the packet network and also to interact with IP end devices. Where does the integration of the voice and data worlds take place? With such a large installed base there will be no widespread move from conventional analog/digital voice handsets to their IP counterparts. VoIP in the enterprise does indeed promote the use of IP enabled handsets and IP end devices and these will replace some existing voice devices. However, in many cases the use of separate voice and data end devices will co­exist at the desktop requiring that the integration takes place at some point in the network. Examples are gateways; those specifically for use as CPE would be Integrated Access Devices (IADs) or Residential Gateways providing packet voice connection at the customer premises. If not at the customer premises, this gateway function will exist at the Central Office / Local Exchange. Here gateways could provide a connection from a DSLAM into a Class 5 switch (this function being termed an access gateway) or provide a packet voice connection to/from the Class 5 switches (termed a trunking gateway). 

The ATM Forum’s VMOA specifications have made great progress on the IAD concept in recent years. The challenge of integrating voice and data anywhere in the network is one of service differentiation. IP has typically been a poor differentiator of high and low priority packets.  The IAD utilizes ATM AAL-2 in the access network to allow multiple voice channels and an IP data link to share the same physical connection to the network. Rather than rely on IP to handle the voice and data combined, the voice and data connections exist independently as ATM VCs over the physical connection. 

Source: CopperCom

Broadband Access

Clearly, there needs to be a network connection to and from the network user and hence the need for some form of access network.  In the traditional voice world, this was delivered as 64k digital voice using the PSTN and ISDN. However, this limited the physical copper loop to supporting one or two voice channels with no capability to simultaneously transmit data.  The solution was then to use a voice channel for data transmission – the basis for much of the growth in dial access Internet usage. Again, the data traffic would consume a voice channel. In order to support the numbers of channels and the bandwidth, then clearly some form of broadband connection would be required – running over the existing last mile infrastructure to avoid the need for widespread deployment of a new access infrastructure. How are the access networks supporting the NGN vision? What are the implications for how the access networks would need to evolve to support the next generation network model? Right now two clear alternatives have appeared in the fixed line market – xDSL access over the copper loop and cable modems using the broadcast cable TV infrastructure. Future developments are likely to open the way for fiber-based access, with technologies such as Passive Optical Networks (PON) being a possible contender. The requirements for the access network are to offer support for converged voice and data services while being able to provide end-to-end differentiated Quality of Service (QoS) between different applications.

One of the key aspects of the access network in delivering next generation networks would be the ability to carry multiple voice channels on a single physical connection. This is where xDSL technology has a natural advantage with the use of IADs able to carry a converged voice and data stream over the copper loop. Here the voice and data channels can be differentiated in the access network and handled in the appropriate way at the Central Office / Local Exchange. There is much renewed interest in the cable world for the so-called “Triple Play” of offering entertainment, voice and Internet services. However, cable has perhaps lagged behind on the voice over packet capabilities. Many cable companies have neither the mandate nor the ability to offer a PSTN style voice service. Voice can certainly be carried as IP packets using VoIP technology but then the main issue remains of how to offer a differentiated service for voice over the other IP applications.  For all the work in the IETF in this area, much still revolves around a best efforts model. There are industry initiatives aiming to improve the real time IP services support. Founded in 1988 by members of the cable television industry, Cable Television Laboratories, Inc. (CableLabs®) is a non-profit research and development consortium that is dedicated to pursuing new cable telecommunications technologies and to helping its cable operator members integrate those technical advances into their business objectives. PacketCable is a CableLabs®-led initiative aimed at developing interoperable interface specifications for delivering advanced, real-time multimedia services over two-way cable plant. Built on top of the industry's DOCSIS™ 1.1 cable modem infrastructure, PacketCable networks will use Internet protocol (IP) technology to enable a wide range of multimedia services, such as IP telephony, multimedia conferencing, interactive gaming, and general multimedia applications. Working with CableLabs member companies and technology suppliers, the PacketCable project will address issues such as device interoperability and product compliance with the PacketCable specifications.

There are also regional variants. The models in North America and Europe are quite different. UK cable operators, for example, have deployed a twisted pair copper loop in addition to the HFC cabling. While allowing the cable company to gain a residential foothold with a telephony proposition, it does indeed open up the possibility that cable companies in Europe could even offer xDSL based services! 

Other technologies will also compete for the broadband services. Over time the cost of deploying fiber to the curb or fiber to the building/home will drop and therefore add another dimension. From a service capability standpoint, the fiber connection would offer similar services to those available over the copper loop but with the proviso of significantly more bandwidth. For example a data connection could be established using Ethernet Transport Services providing a bridged Ethernet connection over the WAN at speeds of 10Mbps and above.

In fact, the technology available for the access networks to support next generation networks goes beyond just cable, xDSL and fiber. As the world becomes increasingly mobile, a whole range of wireless access technologies will make up the NGN picture including WLL (wireless local loop), wireless LAN (802.11), and mobile wireless (2.5G/3G/4G). However, the same principal applies — the access network must provide sufficient bandwidth for multiple services and a means of differentiating between the service levels required by and delivered to different applications.

Broadband Core

The NGN core network will clearly need to be capable of handling converged services based on IP. Driven by today’s existing data networking requirements that already require a packet based infrastructure, there are a number of different possible implementations that could achieve this. It should be made clear that while the Internet offers a global IP-based infrastructure, the core of next generation networks will not be the Internet but a separate IP infrastructure owned and managed by service providers. 

Originally, the basis for the core of any service provider network was ATM on which the IP overlay had been added. For some years, the deficiencies of operating Layer 3 routing over a Layer 2 switching network have been well known. This led to the development of a number of proprietary IP Switching solutions and the work within the IETF on MPLS. Not restricted to an ATM transport, MPLS is seen by many as the future of the multiservice core. However, it is not without its detractors. Some still favor the ATM core for its ability to guarantee end-to-end QoS. Others favor the “simple is best” approach and argue that a more traditional IP routing solution coupled with enormous amounts of optical bandwidth will suffice. They argue that if optical transport can approach wire speeds then differentiation in the core is not an issue. Whichever solution is chosen, the one remaining fact is that the next generation core network must be able to deliver the requested service level needed for a specific application. ATM does provide these guarantees for QoS. IP and MPLS have made great strides in this direction but even DiffServ still can’t provide the same fundamental guarantee. DiffServ is also a technique for prioritizing aggregate data paths in the core rather than individual streams. One of the areas that ETSI TIPHON has still to address is the end-to-end QoS signaling requirement such that an underlying network could offer the necessary guarantee.  Until then, the only solution to IP and MPLS core infrastructure is to overprovision the networks providing a surplus of bandwidth, thereby ensuring all packets, regardless of service type, meet the most stringent QoS needs. This is an expensive solution and one that most doubt will scale to the needs of global networks.

Next Generation Voice Switching – the “Softswitch” concept

The interoperability between the circuit switched voice and packet voice worlds takes place through gateways. Just as in the voice world there are separate channels for bearer and signaling information, the next generation network defines gateways for both bearer and signaling. Softswitch is the generic name for a new approach to telephony switching that has the potential to address all the shortcomings of traditional local exchange switches. This section explains the concept of softswitching.  The later section on voice migration strategies demonstrates how softswitch solutions can lower the cost of local exchange switching, offer the means to create differentiated local telephony services, and ease the migration of networks to support packet voice end-to-end.

The key components of the softswitch architecture are referred to as the Media Gateway, Media Gateway Controller and Signaling Gateway. Individual implementations can create either monolithic or decomposed gateways depending on how or whether these functions are co-located. Typically these functions have been decomposed, which naturally leads to the need for protocols to exist between the various components. The following looks at each as segregated functions. 

The Media Gateway

The Media Gateway (MG) performs compression/decompression of voice signals, and therefore requires significant processing power to handle this and to interpret a number of different types of compression schemes. 

By decoupling the MG functionality from the other gateway components, manufacturers and network designers are able to distribute functionality and processing power across the network, as needed, in response to particular local demand, for example, according to the number and type of bearer channels terminating in certain geographic locations.

Another benefit to segregating the MG is that customers could deploy “trunking gateways” at the edge of the packet network near each of the ingress carrier trunks. Economically, this would reduce the recurring leased-line circuit charges associated with convergence of this traffic onto a single, centrally located gateway. Given that the specific media support needs may vary, a localized gateway can be configured and provisioned appropriately for the media presented at a particular location. Although there will always be specific considerations with individual networks and carriers, it is generally clear that distributing the MG functionality offers several benefits in “return on investment” and network scalability.


The Media Gateway Controller

The Media Gateway Controller (MGC) is responsible for tracking, allocating and billing for resources, and for the overall management of NGN call resources. Given that it provides centralized control over most services, large networks often place significant processor and memory demand on the MGC. For that reason, the most effective NGN architecture needs to present a single, centrally located, powerful MGC (or several redundant MGCs) in control of many distributed MGs. The MGs would perform real-time processing of voice signals locally, while the MGC would set up and terminate calls, monitor network resources, track billing, handle security and authentication, and perform a number of other critical administrative tasks. Additionally, by forcing the MGs to defer to the MGC for service control, changes to the network could be performed once at the MGC rather than multiple times across the network at each of the MGs.


The Signaling Gateway

The third component is the Signaling Gateway (SG). It terminates SS7 connections, emulates an SS7 Signaling End Point to the SS7 network, and converts SS7 messages into an IP-compatible format. 

In general, the combination of the MG, MGC and SG would cooperate as follows: The SG would translate SS7 signaling information, including call setup and tear-down, from the PSTN to the MGC. The MGC would then notify the appropriate IP device (e.g. an IP telephone or multimedia PC application) or the MG of the call request and provide information for establishing a call. Once the call was connected, the MG would provide ongoing compression/decompression and media translation throughout the duration of the call.

In this configuration, the role of the Signaling Gateway is to convert SS7 from the PSTN to and from equivalent IP protocols within the NGN. Communication between the SG and MGC is nearly always over Stream Control Transmission Protocol or SCTP (IETF RFC 2960). Above SCTP, the SG/MGC pair will rely one of the xUA protocols.


Quality of Service

Perhaps the key factor in determining the success of any real time service over a packet network is one of Quality of Service (QoS). To some degree QoS in legacy voice networks has been taken for granted (but implemented through costly over provisioning of network resources).  In a packet network, it is imperative that QoS can be specified and delivered within defined metrics. The work within ETSI TIPHON looks at both mechanisms for providing guaranteed Quality of Service (QoS) levels on ATM or managed IP Networks, as well as Best Effort communication services over the public Internet. In the former case the inherent QoS capabilities of ATM are used.  For a managed IP network some form of IP QoS requesting (e.g. IntServ, RSVP) would be used. Work in the TIPHON QoS Functional Group is also looking at the impact of various factors on voice quality, including:

.                      • packet size

.                      • packet rate

.                      • allowable loss

.                      • allowable jitter

.                      • delay budget

ETSI TIPHON has also defined a number of QoS classes for packet voice. These are shown, along with their characteristics, in the following table:

Source ETSI TIPHON

An important aspect of the TIPHON Release 3 standards is that they are designed for use in a multi-vendor environment, and therefore functional roles, as well as technology interfaces, have been clearly defined. Security procedures also feature prominently. These standards, however, do not constrain business models, as it is recognized that new services and new ways of offering service will emerge in the Next Generation Network environment. The TIPHON approach does not standardize new services per se. Instead, TIPHON specifies building blocks, or ‘service capabilities’ through which service providers can construct new services. It is also important to recognize that standardization and conformance to a common set of specifications should not inhibit vendor differentiation. As long as individual vendors do not compromise interoperability, then vendor specific extensions are allowable. 

Signaling

SIP and H.323

For packet-based voice networks, the signaling protocols most likely to be used include H.245 (the signaling protocol that belongs to the H.323 protocol suite) and Session Initiation Protocol (SIP). An important aspect is the provision of a set of profiles of commonly used industry protocols that enable existing industry standards such as SIP and H.323 to be used together in a compatible way. As indicated, there will be a need for interoperability between SIP and H.323 signaling for interconnection between different VoIP networks.


H.248 and MEGACO

The ITU-T H.248 is the standard for communication between the MG and MGC. The IETF has specified the MEGACO protocol. These developments are being combined to create an integrated H.248/MEGACO standard by the ITU-T. Further work within the IETF and ITU-T is required in order to complete SIP and H.248 so that they cover NGN requirements. Protocols such as H323, SIP, H248 and BICC will co-exist for the foreseeable future. Profiling and complementing these communication standards to make sure that they will all work together is vital for the success of NGN deployment, and TIPHON is well positioned to support this interoperation.


Signaling between Packet- and Circuit-Switched Networks 

MGCs that control Media Gateways connected to circuit-based or packet-based voice networks must support the termination and processing of the telephony signaling protocols associated with those networks. These protocols sit on top of an IP transport, and since the MGC is likely to have an IP network connection for the transport of MEGACO, this connection can be shared for the transport of the packet voice signaling. For circuit-based networks, the applicable signaling protocols include SS7, ISDN Primary Rate Interface (PRI) signaling, and Channel Associated Signaling (CAS). These protocols are typically transported on circuit-based connections. In the case of message-based protocols such as SS7 and PRI, the messages are carried on a connection oriented data link layer such as LAPD (Link Access Protocol for D-channels) or SS7 MTP2 (Message Transfer Part 2), which in turn occupies one or more DS0 timeslots on a T1 facility.

Source: CopperCom

The MGC function does not typically contain the physical interfaces needed to connect to and terminate these circuit-based signaling protocols. Since the MGC’s native mode of communication with the outside world is IP-based, the obvious solution is to map each of the circuit-based signaling protocols onto an IP-based transport. This is where SIGTRAN, developed in the IETF as a method of carrying circuit-based signaling protocols over IP networks, provides the necessary functions. The key component of this solution is the Simple Control Transmission Protocol (SCTP), which can efficiently carry multiple sessions of various signaling protocols over a single IP connection.

Since circuit-based signaling protocols are carried on the circuit-based transmission facilities that terminate on Media Gateways, the requirement is to implement a SIGTRAN signaling gateway as a functional module within the Media Gateway. This signaling gateway will terminate the lower layers of the SS7 and PRI protocols, and encapsulate the application layer messages for forwarding over the IP connection to the MGC.


Network Management – Provisioning, Assurance and Billing

Source: Alcatel

The traditional viewpoint for a NGN OSS model is evolving from the traditional TMN model to the FAB model specified by the Telemangement Forum (TMF).

Here the key requirements for the OSS are seen to be:

.                      Speed. A pre-packaged solution can be quickly deployed and have the service provider up and running in three to four months.

                        Integration. Open systems frameworks enable the fast, flexible implementation and interoperability of best-of-breed solutions. This then extends to integrated network

                        operations with service assurance to deliver the highest availability and reliability of the network and resultant QoS to customers.

.                      Component-based.  A components-based solution allows service providers to bring new users and functions on line simply.

.                      Configurable. Service providers can tailor the software to define a range of specific functions and introduce new services via configuration.

.                      Flexible and scalable. The increased ability to negotiate multi-technology, multi-vendor and multi-platform environments enables service providers to readily adapt their systems to meet the demands of rapidly evolving technologies, customer needs and service offerings.

.                      Automated. Through automation, service providers can define activities, task definitions, operational groups and their accountability; manage workflows relevant to handling different services; define escalation parameters; and configure service offerings from engineering components. Efficient, automated data flow ensures faster time to market at reduced service fulfilment costs.

Security

TIPHON takes into consideration a broad and diverse set of technical, commercial and legal requirements including quality of service, numbering, billing, mobility, security, emergency services, and lawful interception, as well as providing an innovative approach to communications standardization itself. 

Security is naturally a major consideration.  Being able to survive Denial of Service attacks is just one example. Applied to the typical VoIP situation, a major problem is the filtering of UDP packets in firewalls. Given that VoIP uses UDP there is a real problem that firewalls cannot selectively allow the voice calls. To remedy this, ETSI TIPHON have made a recommendation to ITU-T H.248 that will allow a firewall to temporarily open up on a per call basis to allow IP voice UDP packets.

Another major requirement is for a solution to lawful intercept. A requirement of existing voice services is the ability to lawfully intercept an individual call. While this is perfectly feasible in the circuit switched world, it presents a whole set of challenges in packet networks. The monitoring of IP network usage by a single user and especially for a single application is the end goal.

Voice Migration Strategies

Although the ultimate NGN model evangelizes an entirely packet-based voice network with all legacy voice functions replaced by their NGN counterparts, there is a case to be made for phased migration.  First, for some, the entire packet voice vision will be too radical and even if implemented would take decades. There is also the fact that there is little justification to replace existing voice switches and other TDM hardware that have not yet reached end of life or been fully depreciated. Doing nothing is not necessarily the cheapest option. The business case for a complete packet voice approach may be a lot more difficult to justify than for a phased alternative to conventional voice switching. Breaking the problem up into smaller pieces may indeed be the best commercial solution as each part can proceed under its own economic constraints and timeframe. Even if widespread switch replacement is not planned, there are some immediate benefits that can be achieved with modernizing the control of existing voice switches and their signaling networks. Therefore, most industry experts advocate a step-by-step approach to moving towards a NGN model. Perhaps not surprisingly, there are a number of different views as to where to start first as there are no hard and fast guidelines as to what each step should be. The following examines some of the possible scenarios.

New Entrant Voice Provider

Perhaps the simplest of migration strategies is a scenario where there is nothing from which to migrate. Although much of the attention will be focused on the evolution of existing voice service providers, it is important not to overlook the fact that NGN will open doors to new entrants to offer voice services.

Voice switches are optimized to handle tens of thousands of users and typically have a high entry price point, thereby making a high barrier to market entry for a new service provider. Another problem is the lack of differentiation offered with traditional voice services. Given that the functionality of the voice switches is typically very similar, there is often little scope for innovation in voice services. New features tend to be slow in coming due to lengthy development cycles and regression testing phases for switch software. A softswitch would also cost significantly less to implement and hence open up the possibility for a new entrant to offer voice services.

Existing Switch Control Modernization

Another potential solution is to deploy a softswitch based solution for voice switching as an alternative to or adjunct to existing voice switches. Voice calling features are then implemented on a separate “Feature Server”. The advantage here is that, in contrast to the complex process inherent to develop these features on a voice switch, a separate platform using modular and modern programming tools can be used. New features can be implemented in weeks and months rather than 1-2 years. Here the softswitch could be used with the existing TDM network and therefore not even be dependent on a full migration to packet voice. The migration to packet voice can then proceed at a different pace.

Voice Signaling Network Replacement

Even in the signaling network there are commercial justifications to move away from a dedicated SS7 network. The higher layer functionality would remain but there is justification to migrate the lower level SS7 signaling network, in that it is essentially a leased line network, to a packet-based network.  In fact in some cases the cost justification for the start of a NGN has come from just such an approach.

Class 4 Switch Replacement

Seen by many as the more appropriate place to look to migrate to a packet voice solution, the Class 4 switch is typically reaching capacity in today’s voice networks. It represents a smaller investment for the service provider and with less functionality than a Class 5 switch, means a simpler and therefore lower risk point at which to move to packet voice. A packet-based Class 4 switch replacement can still communicate with the existing Class 5 switches using Inter Machine Trunk (IMT) as well as supporting ATM/IP interfaces. This dual mode capability enables conversion of voice trunks at the Class 5 switch from TDM to ATM/IP packet.

Class 5 Switch Replacement

While a new entrant looking to offer voice services is unlikely to implement a traditional Class 5 switch, it is equally unlikely that the large installed base of Class 5 switches will be replaced in the near future. The typical solution will be to place trunking gateways alongside existing Class 5 switches to provide access to the packet network.  Access Gateways will supplement or replace existing TDM Access Nodes for voice connections to the packet network. At end of life the remainder of TDM exchange equipment can be replaced by gateways.

Conclusions

The ultimate requirement of the next generation network is to handle packetized voice and data in a converged manner. There are numerous technical and commercial justifications as to why separate investment in voice and data infrastructure is not feasible. Packet voice may be applied to “Voice over the Internet” but the primary interest will be in voice carried over business quality packet infrastructure. Next generation networks are not just a PSTN replacement but at a minimum they must provide the equivalent voice quality and reliability of today’s PSTN. 

Voice calls will be made between a mix of conventional handsets and IP enabled devices without the caller or receiver being the slightest bit aware of the manner in which their calls are carried or the nature of the equipment being used.

Users are not going to give up existing voice handsets and equipment and so dedicated CPE must exist to adapt the voice calls to the new infrastructure. Users are also not going to give up existing voice features and so these capabilities must be replicated in the packet voice network. Broadband access, using a variety of technologies, will carry the traffic to the service provider edge and into the core networks.

ATM is re-emerging as the chosen technology for providing this access network aggregation. This is reflected in the work of The ATM Forum VMOA WG to provide a Loop Emulation Service using AAL2 for Narrowband Services. Today 50% of ADSL implementations for residential and SoHo markets are already using ATM to deliver high speed Internet access over the local loop. In the core there will need to be support for QoS for differentiated services and the ability to handle this end-to-end requirement across multiple service provider networks using multiple vendors’ equipment.

The NGN will be the foundation for the creation of a new range of multimedia applications that take full advantage of the characteristics of the broadband network and the “always on” capability.

Attention will then shift from simply doing the same as the PSTN for less cost by using new technology to one of doing much more than was feasible before. Rather than see this as a technical migration of replacing old technology with new, the users that are most ready to embrace the new opportunity will be the first to be migrated to the NGN model. This will then be a commercial migration with the emphasis being placed on new revenue opportunities.

Next generation network infrastructure will both support a new range of multimedia applications and also act as the catalyst for their development. Indeed, ironically, in a world of openness and de-regulation, one of the main challenges to overcome could be to find a way to allow telephony calls, being carried as a series of IP packets, to be legally monitored. In the light of enhanced security concerns on the world stage then this can’t be underestimated.

This package of new standards, profiles and test specifications will allow network operators and service providers, for the first time, to offer an integrated standards-based range of telephony services over multi-protocol packet- and circuit-switched networks. It draws together popular industry protocols such as H.323, SIP, MEGACO/H.248 and BICC, to enable services to be offered in a coherent way between multiple service providers and across transport networks utilizing differing technologies.

Whatever approach is taken to migration, it does mean that the introduction of full packet voice can be phased – targeted towards those users that will benefit from the new broadband services and in a timeframe that meets fiscal goals. With the separation of the MGC and MG functions, it is possible for existing equipment to be replaced by media gateways in a distributed, non-disruptive and transparent manner according to customer requirements and geographic considerations.

The creation of the NGN is no overnight transformation – but it is an evolution that is already underway and gathering pace.

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Terms Used

AAL1 ATM Adaptation Layer 1 AAL2 ATM Adaptation Layer 2 ADSL Asymmetric Digital Subscriber Line ATM Asynchronous Transfer Mode BICC Bearer Independent Call Control CAS Channel Associated Signaling CES Circuit Emulation Service CO Central Office CPE Customer Premises Equipment DSLAM Digital Subscriber Line Access Multiplexer DiffServ Differential Services DSL Digital Subscriber Line ETSI European Telecommunications Standardisation Institute FAB Fulfilment, Assurance, Billing GSM Global System for Mobile Communications HFC Hybrid Fiber Coax IAD Integrated Access Device ISDN Integrated Service Digital Network IETF Internet Engineering Task Force ISUP ISDN User Part ITU International Telecommunications Union IWF Interworking Function MG Media Gateway MGC Media Gateway Controller H.245 Control and procedures for H.323 H.248 Media Gateway Control Protocol H.323 ITU Standard for multimedia communications over packet

networks LAPD Link Access Procedure on D channel LEX Local Exchange MoU Memorandum of Understanding MPLS Multi Protocol Label Switching OSS Operational Support Systems PBX Private Branch Exchange PON Passive Optical Network PRI Primary Rate Interface (ISDN) PSTN Public Switched Telephone Network PVC Permanent Virtual Circuit ROI Return on Investment SCTP Stream Control Transport Protocol SG Signaling Gateway SS7 Signaling System No 7 SVC Switched Virtual Circuit NGN Next Generation Network QoS Quality of Service RFC Request for Comment (used by IETF) SCN Switched Circuit Networks SIGTRAN Signaling Translation SIP Session Initiation Protocol SNI Service Node Interface TDM Time Division Multiplexing/Multiplexor TMF Telemanagement Forum TMN Telecommunications Management Network TIPHON Telecoms Internet Protocol Harmonization over Networks VoIP  Voice over Internet Protocol VMOA Voice and Multimedia over ATM WG Working Group WLL Wireless Local Loop

Contacts / More Information

For more information on the subject of Next Generation Networks and to see the presentation material from the Broadband Exchange in Prague, or register for future events, please visit The ATM Forum website at


http://www.atmforum.com/pages/meetingsfsbbx-sum.html

For more information on ETSI TIPHON:


http://www.etsi.org/tiphon

To find out how you can become a member or to participate further in the work of The ATM Forum, contact The ATM Forum at one of the regional offices below.

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