VoIP

VoIP

VoIP has been drawing a lot of attention in the past couple years. This section covers the types of applications that are anticipated for VoIP, as well as what network elements are required to make VoIP work and provide similar capabilities to what we're used to from the PSTN.

VoIP Trends and Economics

Although VoIP calling is used for billions of billed minutes each year, it still represents a very small percentage of the market less than 5% overall. According to Telegeography (www.telegeography.com), 40% of VoIP traffic originates in Asia and terminates in North America or Europe; 30% travels between North America and Latin America; one-third of U.S. international VoIP traffic goes to Mexico, with future volume increases predicted for calling to China, Brazil, and India, and the rest moves among the U.S., Asia Pacific, and Western European regions. It is important to closely examine who will be using this and what carriers or operators will be deploying these technologies. Probe Research (www.proberesearch.com) believes that by 2002, 6% of all voice lines will be VoIP. This is still rather minor, given the fact that some have been saying that VoIP would have replaced circuit-switched calling by now. Piper Jaffray (www.piperjaffray.com) reports that minutes of communication services traveling over IP telephony networks will grow from an anticipated 70 billion minutes and 6% of all the PSTN traffic in the year 2003 to over a trillion minutes by the year 2006. In the United States alone, the PSTN is handling some 3.6 trillion minutes of traffic monthly.

Although VoIP has a very important place in telecommunications, it's important to realize that it is not yet taking over the traditional circuit-switched approach to accommodating voice telephony. The exciting future of VoIP lies in advanced and interesting new applications, an environment where voice is but one of the information streams comprising a rich media application. Many expect that sales of VoIP equipment will grow rapidly in the coming months and years. Part of the reason for this growth is that the network-specific cost for VoIP on dedicated networks is quite a bit lower than the cost of calls on circuit-switched networks about US 1.1 cents per minute as compared with US 1.7 cents per minute. Using VoIP to carry telephony traffic greatly reduces the cost of the infrastructure for the provider, but at the expense of possibly not being able to maintain QoS. Potential savings are even greater if VoIP is implemented as an adjunct to data network.

Another factor encouraging customers to examine VoIP is the use of shared networks. Because IP emphasizes logical rather than physical connections, it's easier for multiple carriers to coexist on a single network. This encourages cooperative sharing of interconnected networks, structured as anything from sale of wholesale circuits to real-time capacity exchanges. Also, VoIP can reduce the barriers to entry in this competitive data communications world. New companies can enter the market without the huge fixed costs that are normally associated with the traditional circuit-switched network models. Furthermore, because IP telephony will enable new forms of competition, there will be pressure to better align government-controlled prices with underlying service costs. International VoIP services are already priced well below the official rates and some of VoIP's appeal is that it eliminates the access charges interexchange carriers normally have to pay to interconnect to the local exchange carrier. In the United States, these charges range from US 2 cents to US 5 cents per minute.

Advantages of VoIP

The key benefits of VoIP are cost savings associated with toll calls, enhanced voice services, and creative and innovative new applications. The key concerns related to VoIP are voice quality compared to that in today's PSTN; the cost of QoS to ensure the same quality as in the PSTN; security; the current lack of compelling applications; and regulatory issues, such as whether voice will be allowed on the Internet and whether voice will be treated as an altogether different environment as a converged, integrated application.

Regulations Related to VoIP

It's one thing to approach telephony on the Internet such that the incumbent is protected from competition with other voice telephony services on the Internet. But stating that voice on the Internet should not be allowed would be to cut your own throat. All the exciting new applications on the Internet do involve the use of multimedia applications, and voice is part of that overall stream. So, we have to be very careful about what we're regulating whether it's voice, which is increasingly part of a larger application set, or whether it's traditional voice telephony.

VoIP Applications

VoIP includes any set of enabling technologies and infrastructures that digitize voice signals and transmit them in packetized format. Three major network architectures can be used in support of VoIP applications:

         Voice-over intranets, which could be based on leased lines, Frame Relay, ATM, or VPNs

         Voice-over extranets, which could also be based on leased lines, Frame Relay, ATM, or VPNs

         Voice over the public Internet

The following sections discuss some of the key issues related to VoIP applications.

IP Long-Distance Wholesale

So far, the most compelling business case for VoIP has been in IP long-distance wholesale, where there are clear financial benefits and low barriers to entry. Early pioneers in this area include iBasis, ITXC, and Level 3, which predominantly offer IP services to domestic and international carriers, but also offer services to corporations and other service providers. What the customers gain by doing business in this fashion is a reduction in cost associated with carrying their traffic over expensive toll or international transit links.

In IP long-distance wholesale, the voice service levels must match those of the PSTN. End customers of the international carriers expect to perceive the same voice quality throughout. How can providers guarantee that when it's almost impossible to control QoS over the public Internet? Even in the case of IP backbones, QoS depends on the underlying architecture used. The solution lies in smart management of packet latency, to ensure circuit-like behavior inside the IP network. For example, iBasis developed a proprietary routing algorithm that monitors performance on the Internet; when it detects that congestion levels may affect the quality of the voice, it switches the calls over to the circuit-switched network, thereby ensuring that customers experience the high quality that they expect end-to-end.

The IP long-distance wholesale environment takes advantage of a converged voice/data backbone by using trunking gateways to leverage the PSTN (see Figure 11.11). This allows support and processing of voice calls. The trunking gateways enable connection of the data network to the PSTN, to support long-haul carrying of the switched calls. In addition, switching services can be added to the data networks through the use of softswitches. (The functions and types of softswitches and gateways that make up the new public network are discussed later in this chapter.)

Figure 11.11. A converged long-distance network

These are main issues in selecting providers of IP long-distance wholesale:

         Voice quality versus bandwidth How much bandwidth do you use to ensure the best quality?

         Connecting to the customer How many services need to be supported (voice, data, dialup modem, fax, ISDN, xDSL, cable modem)?

         Maintaining voice quality As bandwidth becomes constrained, how do you maintain the voice quality?

IP Telephony

There are two main approaches to IP telephony. First, there's IP telephony over the Internet. Calls made over the public Internet using IP telephony products provide great cost-efficiencies. But the Internet is a large, unmanaged public network, with no reliable service guarantee, so the low costs come at a trade-off. International long-distance consumer calls are the major application of IP telephony over the Internet. Second, the use of private IP telephony networks is rapidly emerging. In this approach, calls are made over private WANs, using IP telephony protocols. The network owner can control how resources are allocated, thereby providing QoS and a managed network. Many private IP telephony networks are being built. They enable an enterprise to take advantage of its investments in the IP infrastructure. Again, because this is a single-owner network, the QoS issues are much easier to contend with; in fact, a single-owner network makes it possible to contend with QoS issues!

Multinational enterprises spend billions of dollars on international voice services each year, so the savings that IP telephony offers is compelling. The cost benefit of running voice services over a private IP network is on the order of 20% or more savings on international long distance, as compared to using traditional voice services. Private IP transport platforms will be increasingly deployed, therefore, as an enterprisewide telephony option.

Recent deployment of IP local exchange products, coupled with low-bandwidth, high-quality voice compression, creates a solid foundation for extending business telephone service to telecommuters at home or on the road. The efficiencies of IP packet technology, coupled with the ITU G.723.1 voice compression standards at 6.4Kbps, enable road warriors and small office/home office workers to have a complete virtual office over a standard 56Kbps Internet modem connection to the office. The really great feature of this environment is that your current location is your office and your IP phone rings wherever you are. However, this requires an IP local exchange a carrier-class product that resides in the service provider network and provides PBX-like telephony service to multiple business and telecommuter customers. It also requires a softswitch (that is, call-agent software) that's used for purposes of managing call processing functions and administration. Also, end-user services are delivered via IP Ethernet phones or analog telephones that use Ethernet-to-analog adapters.

There are three major categories of IP phones:

         POTS phone The advantage of the POTS phone is high availability and low price. The disadvantage is that it has no feature buttons and the required Ethernet-to-analog adapter is quite costly.

         Soft phone A soft phone is software that runs on the user's PC and graphically resembles a telephone. Its advantage is low price. Its disadvantage is that it relies on the PC sound card, and it can create volume level problems when you switch between it and other applications that use the PC sound card.

         IP Ethernet phone This device looks and works just like a traditional multiline business display phone, and it plugs into an Ethernet RJ-45 jack. It's priced similarly to PBX phones, at US$300 and up. Emerging "IP phone on a chip" technologies promise dramatically lower prices in the near future.

The evolution of IP telephony will involve many different types of applications, including long-distance wholesale voice services; the support of voice applications for campus or enterprise networks in bringing VoIP to the desktop in the form of new advanced applications that involve converged streams (such as video conferencing or multimedia in the establishment of remote virtual offices); Internet smart phones; IP PBXs; IP centrex service; unified messaging; Internet call waiting; and virtual second-line applications.

VoIP Enhanced Services

Another approach to supporting voice services is to look toward enhanced services. There are two categories of enhanced services:

         Transaction-oriented services These services include Click-N-Call applications, interactive chat, Surf-With-Me, videoconferencing, and varieties of financial transactions.

         Productivity-enhancing services These services include worldwide forwarding, multiparty calling, a visual second line, unified messaging, collaboration, access to online directories, visual assistance, CD-quality sound, personal voice response, and video answering machines.

The key to enhanced services is not cost savings, but cost savings are realized through toll bypass, QoS differentiation, the capability to support remote access, and the capability to create new forms of messaging. Because of the cost savings and features available, the use of enhanced services will grow by leaps and bounds over the next several years.

VoIP is part of a larger trend toward innovative voice-enabled Internet applications and network interactive multimedia. This trend includes various facilities to enhance e-commerce, customer service, converged voice and visual applications, new intelligent agents and various forms of bots, and e-calling campaigns. These sorts of advanced services make it possible to gain greater value from the IP investments that have been made, and at the same time, they create interesting new revenue streams with altogether new businesses.

We'll see VoIP applications increasingly used in a number of ways. VoIP applications will be included on Web-based call centers as automatic call-backs from customer service-based phone numbers entered into a Web page; as multiparty conference calls, with voice links and data sharing, initiated also from a Web page; and in the process of reviewing and paying bills. The key is to blend rich, Internet-based content with a voice service. An example of an emerging application that illustrates such innovation is online gaming. InnoMedia and Sega Enterprises are integrating InnoMedia Internet telephony into Sega Dreamcast game consoles to allow game players worldwide to voice chat with each other while playing games. This device can also be used to cost-effectively place calls in more than 200 countries through InnoSphere, InnoMedia's global network. For example, the rate from the United States to Hong Kong will be US2 cents per minute, from the United States to the United Kingdom it will be US5 cents, and from the United States to Japan, Australia, and most of Europe, it will be US9 cents.

Another example of an interesting new VoIP application is Phonecast, a media network of Internet-sourced audio channels for news, entertainment, and shopping, available to telephones. Created by PhoneRun and WorldCom, Phonecast is modeled after television and radio broadcasting, and it allows callers to create a personal radio station and direct it by using simple voice commands. This is the first of a series of innovative content and service partnerships, assembled to form a comprehensive voice-portal product line.

VoIP Service Categories

There are several main VoIP service categories:

         Enterprise-based VoIP In enterprise-based VoIP, whether for the LAN or WAN, specialized equipment is required at the customer site.

         IP telephony service providers These providers are generally involved in toll-bypass operations. They do not require specialized equipment at the customer site, but they may require additional dialing procedures to gain access to the network. Currently, multistage dialing is one of the problems we still face: You have to dial a seven- or eight-digit number to gain access to your ISP, and then you have to dial a string of digits for the authentication code, and then you have to dial the string of digits corresponding to the number you want to reach. Single-stage dialing will remedy this situation in the very near future.

         Converged service providers These companies will bundle together voice, data, and video services.

         Consumer VoIP Consumer VoIP is generally geared toward consumer connections over the public Internet.

VoIP Network Elements

VoIP may seem like rocket science compared to conversations, but the concept is really quite simple: Convert voice into packets for transmission over a company's TCP/IP network. Two characteristics determine the quality of the VoIP transmission: latency and packet loss. Latency is the time it takes to travel from Point A to Point B. The maximum tolerance for voice latency is about 250 milliseconds, and it's recommended that the delay be less than 150 milliseconds. Small amounts of packet loss introduce pops and clicks that you can work around, but large amounts of packet loss render a conversation unintelligible. With too much packet loss, you would sound like you were saying "Da dop yobla bleep op bop," because little packets with much of your conversation would have been lost in congestion and could not be retransmitted while working within the delay requirements of voice. Hence, packet loss with VoIP can cause big chunks of a conversation to be lost. (We will talk about ways to resolve that a little later in this chapter.)

VoIP gateways have allowed IP telephony applications and new, innovative VoIP applications to move into the mainstream. Other features that have helped the development of VoIP are Internet telephony directory, media gateways, and softswitches, as well as telephony signaling protocols.

VoIP Gateways

VoIP gateways bridge the traditional circuit-switched PSTN and the packet-switched Internet. Gateways overcome the addressing problem. A couple years ago, for two VoIP users to communicate, they had to be using the same software, they had to have sound cards and microphones attached to their PCs, and they had to coordinate a common time during which both would be online in order to engage in a VoIP session. Gateways have made all that unnecessary, and now the only requirement is that you know the user's phone number. Phone-to-PC or PC-to-phone operation requires the use of only one gateway. Phone-to-phone operation requires two gateways, one at each end.

VoIP gateway functionality includes packetizing and compressing voice; enhancing voice quality by applying echo cancellation and silence suppression; dual-tone multifrequency (DTMF) signaling support (that is, touch-tone dialing); routing of voice packets; authentication of users; address management; administration of a network of gateways; and the generation of call detail records that are used to create bills and invoices.

To place a call over a VoIP network, the customer dials the number the same way as on a traditional phone. The edge device, the VoIP gateway, communicates the dialed number to the server, where call-agent software that is, a softswitch determines what is the appropriate IP address for that destination call number and returns that IP address to the edge device. The edge device then converts the voice signal to IP format, adds the given address of the destination node, and sends the signal on its way. If enhanced services are required, the softswitch is called back into action to perform the additional functions. (The softswitch is also referred to as a Class 5 agent because it behaves like a local exchange or a Class 5 office.)

There are two primary categories of VoIP gateways:

         Gateways based on existing router or remote access concentrator (RAC) platforms The key providers here include the traditional data networking vendors, such as 3Com, Cisco, Lucent, and Motorola. As incumbent equipment suppliers to ISPs, the data networking vendors are capturing the largest percentage of these sales. They represented the majority of VoIP gateway sales through 2000 because ISPs were buying gateways at a fast rate based on the significant wholesale opportunity available to larger carriers.

         Server-based gateways These are designed from the ground up to support VoIP. Key providers of server-based gateways include telecommunications vendors, as well as companies specifically designed for this business; Clarent, Ericsson, Lucent, NetSpeak, Nortel, Nuera, and VocalTec are among the vendors involved. These gateways will overtake router and RAC solutions as incumbent carriers deploy more server-based gateways with extensive call server and signaling capabilities.

More and more merger and acquisition activities will lead to blended solutions, causing the distinction between the different types of gateways to blur. RAC- and router-based gateways will take on more enhanced call-server characteristics as a result. The market segments for the two categories, then, are composed of the following:

         Enterprise VoIP gateways These gateways are customer premise equipment deployed between a PBX and a WAN device, typically a router, to provide call setup, call routing, and conversion of voice into IP packets and vice versa.

         VoIP routers Voice cards perform packetization and compression functions and are inserted into a router chassis. The router then directs the packets to their ultimate destination.

         IP PBXs An IP PBX is an infrastructure of distributed telephony servers that operates in packet-switched mode and offers the benefits of statistical multiplexing and IP routing. We are still in the early days for IP PBXs, although they are beginning to emerge as a viable alternative. A key concern is reliability. (IP PBXs are discussed in more detail later in this chapter.)

         Service-provider VoIP gateways These are used to aggregate incoming VoIP traffic and route the traffic accordingly. The role is analogous to that of the local exchange. Challenges include the local loop competition among the incumbent carriers, quality concerns, shortage of product, interoperability issues, the lack of hot-swappable and redundant support, and the lack of Network Equipment Building Systems (NEBS) compliance.

         VoIP access concentrators VoIP cards fit into an existing dial access concentrator.

         SS7 gateways SS7 gateways are critical to enabling us to tap into the intelligence services that enhance so much of the telephony activity on the PSTN.

There are many gateway vendors. All gateway vendors share the need for digital signal processors and embedded software solutions that provide for silent suppression, echo cancellation, compression and decompression, DTMF signaling, and packet management. Therefore, another very important part of this equation is the component vendors. Manufacturers of VoIP equipment need to continue to make quality improvements in the underlying technology. This includes addressing interoperability between different gateway vendors' equipment; improving the tradeoffs between cost, function, and quality; and introducing single-stage dialing and the ability to dial from any telephone.

Internet Telephony Directory

An Internet telephony directory is a vital piece of the VoIP puzzle, so this section talks a little bit about the IETF Request for Comment 2916, also known as ENUM services. ENUM services convert telephone numbers into the Internet address information required to support all forms of IP-enabled communication services, including real-time voice, voicemail, fax, remote printing, and unified messaging. In other words, ENUM is a standard for mapping telephone numbers to IP addresses. DNS translates URLs to IP addresses, and EMUM uses the DNS to map a PSTN phone number (based on the E.164 standard) to the appropriate URLs.

ICANN is considering three proposals for the .tel domain. The applicants are NetNumber, which currently runs the Global Internet Telephony Directory (an implementation of ENUM that is used by IP-enabled platforms to convert standard telephone numbers into Internet address information), Number.tel, and Telnic based in the United Kingdom. The ITU is trying to advance an implementation of the IETF ENUM standard under the domain e164.arpa. In this implementation, control of telephone number addressing on the Internet would be distributed to the more than 240 national public network regulatory bodies that administer telephone numbers for the PSTN.

Media Gateways

Media gateways provide seamless interoperability between circuit-switched, or PSTN, networking domains and those of the packet-switched realm (that is, IP, ATM, and Frame Relay networks). They interconnect with the SS7 network and enable the handling of IP services. They're designed to support a variety of telephony signaling protocols. Media gateways are designed to support Class 4, or toll-switch, functions, as well as Class 5, or local exchange, services. They operate in the classic public network environment, where call control is separate from media flow. They support a variety of traffic including data, voice, fax, and multimedia over a data backbone. Enhanced applications of media gateways include network conferencing, network-integrated voice response, fax serving, network, and directory services.

As shown in Figure 11.12, media gateways fit between the access and core layers of the network, and they include several categories: VoIP trunking gateways, VoIP access gateways, and network access service devices. They provide service interconnection or intercarrier call handling. The trunking gateways interface between the PSTN and VoIP networks, terminating trunks associated with SS7 control links. These Time Division Multiplexed trunks carry media from an adjacent switch in the traditional circuit-switched network, and the adjacent switch generally belongs to another service provider. (Depending on the agreements between service providers, these are also referred to as cocarrier trunks or feature group D trunks.) The trunking gateways manage a large number of digital virtual circuits. The access gateways provide traditional analog or ISDN interfaces to the VoIP net works; they are devices that terminate PSTN signaling and media, and they connect to PBXs, as well as to traditional circuit switches, such as the Class 5 and Class 4 offices. With network access servers, you can attach a modem to a telephone circuit and provide data access to the Internet, so that you can attain managed modem service by using cocarrier trunks.

Figure 11.12. VoIP network architecture

VoIP Softswitches

Call-control intelligence is outside the media gateways and VoIP gateways; it is, instead, handled by a softswitch, also referred to as a media gateway controller or call agent. The softswitch implements the service logic. It controls external trunking gateways, access gateways, and remote access servers. Softswitches run on commercial computers and operating systems, and they provide open application programming interfaces.

A softswitch is a software-based, distributed switching and control platform, and it controls the switching and routing of media packets between media gateways, across the packet backbone. Softswitches provide new tools and technologies to build services in a more productive Internet-based service creation environment. Operators are advised to adopt a "service separation" strategy and to distribute applications throughout the network, avoiding the monolithic closed system that is similar to the circuit-switched environment. We can use application servers to partition enhanced telecommunications services and to determine what interface protocol to select for facilitating interoperability between the softswitches and the applications servers.

The softswitch functionally controls the voice or data traffic path by signaling between media gateways that actually transport the traffic (see Chapter 10). The gateway provides the connection between an IP or ATM network and the traditional circuit-switched network, acting a lot like a multiprotocol cross-connect. The softswitch ensures that a call's or a connection's underlying signaling information automatic number identifiers, billing data, and call triggers are communicated between the gateways. Softswitches must reuse intelligent network services through an open and flexible directory interface, so they provide a directory-enabled architecture with access to relational database management systems, and to Lightweight Directory Access Protocol (LDAP) and Transaction Capabilities Applications Part (TCAP) directories. Softswitches also offer programmable back-office features, along with advanced policy-based management of all software components.

The softswitch is a very important element in the new public network. It is what enables the media and trunking gateways to communicate with the underlying infrastructure of the PSTN and thereby to draw on the service logic needed to support telephony activities. In addition, softswitches will be able to reach to new application servers on which new generations of applications have been designed for new versions of enhanced services.

Telephony Signaling Protocols

New generations of signaling and IP telephony control protocols are emerging, and their purpose is to control the communication between the signaling gateway and IP elements. Since the early days of exploring the nature of VoIP and creating devices to enable it, a number of telephony signaling protocols have been considered. Some of the contenders have been H.323, Internet Protocol Device Control (IPDC), Signal Gateway Control Protocol (SGCP), Multimedia Gateway Control Protocol (MGCP), Multimedia Gateway Control (MEGACO), Session Initiation Protocol (SIP), and IP Signaling System 7 (IPS7). Many of those contenders have combined, so this section focuses on the ones that have the strongest presence and potential today.

H.323 The ITU H.323 version 2 specification is based on ISDN standards and limited to point-to-point applications. Version 2 requires multipoint control units (MCUs) to manage multiple sessions. H.323 version 2 provides much of the foundation for exchange of voice and fax messages. The advantage of H.323 is that it is the most mature of the telephony signaling protocols, so many vendors offer it and vendor interoperability is good. On the other hand, H.323 is not as robust as some of the newer entrants, so other protocols on the horizon might eclipse H.323 before too long.

MCGP Bellcore and Level 3 merged their respective SGCP and IPDC specifications into MCGP. In MCGP, softswitches provide the external control and management, so MCGP is becoming a good way to connect an IAD to a gateway.

MEGACO MEGACO is also called H.248 and it is another emerging ITU standard. MEGACO describes how the media gateway should behave and function.

SIP SIP (IETF Request for Comment 2543) is an application-layer control, or signaling protocol, for creating, modifying, and terminating sessions with one or more participants. SIP is used to set up a temporary session, or call, to the server so that the server can execute the necessary enhanced service logic. These sessions may include Internet multimedia conferences, Internet telephony, or multimedia distribution. Linking caller ID to Web page content can link the status of a mobile phone with instant messaging. Members in a session can communicate via multicast or via a mesh of unicast relations, or by a combination of these. This is increasingly popular as the protocol between softswitches and application servers.

LDAP LDAP is the standard directory server technology for the Internet. LDAP enables retrieval of information from multivendor directories. In fact, LDAP 3.0 provides client systems, hubs, switches, routers, and a standard interface to read and write directory information. Directory-oriented services best suited for an LDAP lookup include unified messaging, free phone (that is toll-free number translation), calling name service, and Internet phone number hosting. Remember that as the Internet moves forward, it must connect with the underlying intelligence in the PSTN.

IPS7 The SS7 network acts as the backbone for the advanced intelligent network. SS7 provides access to all the advanced intelligent network features, allows for efficient call setup and teardown, and interconnects thousands of telephony providers under a common signaling network. The capability to communicate with the SS7 network is essential for all service providers. It gives next-generation local exchange carriers access to an existing base of service features, and it ensures that packet-based telephony switching gateways can support key legacy service and signaling features. The interconnection between a legacy circuit switch provider, such as the incumbent local exchange carrier, and a competitive local exchange carrier operated over a packet backbone would include the gateway switch to packetize and digitize the voice coming from the Class 5 office, and the SS7 gateway to provide access into the underlying intelligent network infrastructure. (Chapter 5, "The PSTN," discusses SS7 and next-generation gateway switches in more detail.)

Next-Generation Standards and Interoperability

Next-generation network standards are widely deployed across the globe and are generating billions of dollars in service revenue. Packet-enabled intelligent networks will enhance the revenue stream with new technology to provide intelligent networking services, such as local-number portability, carrier selection, personal numbers, free phone, prepaid call screening, call centers, and voice VPNs. End-to-end, next-generation networks function as seamlessly interoperating wholes; they consist of the legacy-based circuit-switched network, with its underlying SS7 and service logic delivering today's enhanced features, as well as a packet-based network for transport efficiencies that can also be served by new-generation IP servers and enhanced applications, for features we haven't yet thought of.

There are a few key groups to be aware of in the area of standards and interoperability for next-generation networks. There's iNOW!, which stands for Interoperability NOW!, and its members include Ascend, Cisco, Clarent, Dialogic, Natural MicroSystems, and Siemens. These members will interoperate also with Lucent and VocalTec, as well as each other. iNOW! advocates interoperability and certification based on H.323.

The Technical Advisory Committee (TAC), formed by Level 3 Communications, includes 3Com, Alcatel, Ascend, Cisco, Ericsson, Level 3, and others.

The International Softswitch Consortium is focused on enabling softswitch technology and applications on an IP infrastructure. This group advocates interoperability and certification based on H.323, SIP, MGCP, and Real-Time Transfer Protocol (RTP). It is working to develop and promote new standardized interfaces for portable applications, which ride on top of an IP-based softswitch network. The International Softswitch Consortium has more than 68 member companies.

Finally, the Multiservice Switching Forum (MSF) is an open-membership organization committed to developing and promoting implementation agreements for ATM-capable multiservice switching systems. The goal of MSF is to develop multiservice switching with both IP- and ATM-based services, and its founding members are WorldCom, Cisco Systems, Telcordia, AT&T, Alcatel, Lucent, British Telecom, Fujitsu Network Communications, Lucent Technologies, Nortel Networks, Siemens, Telecom Italia, Telia AB, and Qwest.

IP PBXs

IP PBXs are in the very early stages, and they will present some benefits as well as some challenges. Companies can take advantage of IP-based intranets that have been set up between headquarters and remote locations to cost-effectively integrate voice and data traffic. The key strength of IP PBXs is their capability to network over existing IP networks. Because the information is programmed into the phone, phones can be relocated by simply unplugging and moving them. It is also easier to network over existing IP WANs, as long as there is adequate bandwidth to support voice traffic.

Among the challenges to the convergence of IP PBXs is that we expect them to provide high reliability and high availability, which we always require with telephony. Telephony-grade servers are classified as fault tolerant when they achieve 99.99% (that is, four nines) survivability. The standard for most PBX voice systems is 99.999% (that is, five nines), so four nines is quite a bit less than what we're accustomed to. The industry is slowly embracing Windows NT and Windows 2000 for core call processing, but some feel that these products are not reliable enough in their current form. To be fair, research on NT stability and security shows that almost always the problems are a result of poor or improper administrative procedures, not a result of problems in the operating system itself. As NT administrators have gained operational experience, the reliability and security of Windows-based data centers has improved. In summary, customer concerns include security, reliability, survivability, operability, maintainability, and accountability.

Another important issue related to IP PBXs is power distribution. PBXs have internal power distribution plants to support processing memory and internal interface circuit cards. All analog and proprietary digital telephones are line powered by the centralized PBX, using standard unshielded twisted-pair (UTP) wiring. Larger PBXs often have redundant power conversion and distribution elements throughout the cabinet design, and fluctuations in power such as spikes and surges are also regulated by the PBXs. Although there are Ethernet switches that can deliver power to the desktop via Category 5 cabling, they are just being introduced, and standards have not yet been developed for this.

Voice quality is another big issue with IP PBXs. The voice quality delivered over an IP PBX has to match that of the PSTN, so VoIP systems will need to meet stringent technical requirements to manage delay and echo, which are affected by the amount of compression and the type of codec used, as well as by the QoS capabilities of the underlying transport network. Voice quality will become a new performance variable, with various levels available and reflected in the pricing of services.

Another issue related to IP PBXs is network QoS. Voice QoS must remain adequate when it shares the network with bandwidth-intensive data applications. Packet loss must be minimized, and latency must be reduced. We still need to figure out how much voice traffic the data network can accept before voice, data, and video start to degrade.

Features and functionality are other issues. PBXs in general have 400 to 500 features, whereas IP phone systems provide only about 100 features.

There are also issues surrounding distance limitations. Fast Ethernet Category 5 cabling is limited to distances of 330 feet (100 meters), whereas PBXs support analog phone extensions over UTP at up to 2 miles (3.5 kilometers) and proprietary digital phones at up to 1 mile (1.5 kilometers).

Another issue is the lack of management systems. Systems designed to accommodate moves, adds, and changes, as well as troubleshooting, need to be developed.

There are also security questions (for example, Will voice over the LAN demand encryption of voice traffic?) and issues related to legacy voice investments (that is, enterprises generally protect investments in their existing equipment). Finally, we face a lack of a really compelling value proposition. However, PBXs are migrating toward a future in which IP-based packet transport will replace circuit-switched Time Division Multiplexing.

This market is poised for major change in the next several years. PBXs are migrating toward telephony server models, in which a nonproprietary platform will perform the call control and feature provisioning. But these are still the early days. Through integration, we will eventually have a much more cost-efficient platform for network services.

The Future of VoIP

VoIP is very important, and it's part of a larger application set that enables the integration of voice, video, data, and images. It is the early days for VoIP as well. Today, VoIP accounts for only a very small amount of global voice traffic. With VoIP we face issues of interoperability, scalability, and the number of features that can be supported. We face issues of whether the incumbents are motivated to replace all Class 5 exchanges with next-generation telephony. IP QoS is still immature, as these are the early days.