Gateway Decomposition

The different types of gateways described iare specific instances of a generic gateway notion. It is useful to decompose the generic gateway into several functional components. Figure 1 depicts a common view of such a decomposed gateway. Three components are identified: (1) the media gateway (MG) function, (2) the signaling gateway (SG) function, and (3) the media gateway control (MGC) function. We describe these functions next.

Figure 1: Components of a decomposed PSTN-Internet gateway.

Physically, the media gateway function terminates PSTN circuits and connections to IP routers (in relation to which it is a host). It also performs all the necessary transformation to convert bit streams received from the sending network into bit streams particular to the receiving network. The transformation occurs at two levels: transmission and application.

At the transmission level, the MG function converts the bit streams between two different framing schemes. This usually involves multiplexing of bit streams of distinct communication sessions and the reverse operation—demultiplexing. In the PSTN, fixed-size digital channels (each typically carrying a voice conversation) are multiplexed based on the time division multiplexing (TDM) scheme at various hierarchical levels (for example, T1 and E3) and packed into frames for transmission over high-capacity facilities. In the IP networks, bits representing a voice conversation are packetized according to the Real-time Transport Protocol (RTP) profile for audio and video payloads (RFC 1890).

At the application level, the transformation takes place between different media-encoding schemes (see the section on codecs for more information) and is commonly known as transcoding. In IP telephony, two prevalent speech encoding schemes are G.711 and G.729. Operating at a bit rate of 64 kbps, G.711 is used ubiquitously in the digital backbone of the PSTN and sets what is known as the toll-quality voice standard. G.729 operates at a much lower bit rate of 8 kbps but still supports near-toll-quality voice service. For this reason, it is widely used in IP networks where bandwidth is constrained. Note that transcoding is computationally intensive and thus causes delays. In addition, transcoding results in degradation of voice fidelity, in particular when a speech coder uses compression.

Another important task of the MG function is to support the use of the QoS facilities of the IP network. Other tasks include echo cancellation (if required), event detection, signaling generation, usage recording, and support of specialized resources such as conference bridges, fax machines, modem pools, and interactive voice response units.

The SG function receives and sends PSTN signaling (such as SS No. 7 or ISDN access) messages. Depending on the arrangement, it may relay, translate, or terminate the PSTN signaling. It exchanges signaling information with the MGC function over IP, and with the PSTN using the SS No. 7.

The MGC function provides control of the media gateway function, including call and connection control and resource management. To this end, it terminates and originates all the relevant signaling. In addition, the MGC function keeps an inventory of the MG resources (for example, bearer circuits and RTP streams) and instructs the MG to reserve or release resources as required. (Naturally, some sort of local policy will govern the use of resources.) With its central role in call and connection control, the MG function logically also provides support for Internet offloading or advanced services and features (such as freephone or call-forwarding). It has the ability to detect data calls from the PSTN and to direct the data traffic straight to a network access server as well as to launch queries to SCPs for instructions for further call processing.

IP Telephony Gateway

If a call is to be made from a PC or specialized IP phone to a regular PSTN telephone or vice versa, both the PSTN and IP network are involved in making the call. The PSTN may also use an IP network for so-called trunk replacement, or IP trunking, where the long-distance portion of the PSTN voice traffic between two PSTN telephones is carried over an IP network.

When it comes to delivering real-time voice, the PSTN and an IP network are different in a number of ways, as summarized in Table 1. For establishing a call, for example, SS No. 7 has been traditionally used within the PSTN, while H.323 is the most prevalent protocol suite [with Session Initiation Protocol (SIP) as a viable alternative] in the Internet to date. In general, the connection of two dissimilar networks is achieved through some sort of a device—called a gateway—that compensates for the differences in the networks. There is no exception when it comes to interconnecting the PSTN and IP networks for supporting IP telephony. In this case, the interconnecting device is called an IP telephony gateway and will link users of IP telephony with a billion or so PSTN users.

Table 1: Key Differences between Telephony over the PSTN and the Internet

Distinguishing Aspect	PSTN	Internet
Bandwidth allocation for voice transport	A dedicated circuit (e.g., 64 kbps) set up for each two-party communications session.	Best-effort delivery of IP packets.
Numbering and addressing scheme	14-bit point code for network nodes and E.164 for endpoints.	4-byte IP address, domain name system (DNS), e-mail address, uniform resource locator (URL), etc.
Voice representation	Typically analog in the loop and G.711 (either A-or m-law) in the backbone.	G.711, G.723.1, G.729, etc.
Signaling protocols	Signalling System No. 7 Q.931, etc.,	H.323, SIP, etc.
Availability	99.999% (5 min downtime per year).	99% (88 h downtime per year).

Figure 1 illustrates the integration of the PSTN and the Internet (or any IP network) through gateways in support of IP telephony. It distinguishes four types of IP telephony gateways based on the PSTN interfaces and certain specific functions that they support.

Figure 1: PSTN-Internet integration through gateways.

1. 
   
   Trunking gateway. Connects a central office (CO) switch to an IP router. Such a gateway typically has an SS No. 7 signaling interface and manages a large number of 64-kbps digital circuits and Real-Time Transport Protocol (RTP) streams. It is used in the trunk replacement application where the long-distance portion of a call between two telephones is made over the IP network instead of the PSTN. (In IP telephony parlance, such calls are known as phone-to-phone calls.)
2. 
   
   Access gateway. Connects telephones or PBXs to an IP router through an access interface [such as ISDN primary rate interface (PRI)]. It supports calls between two telephones with the IP network as an intermediary transport or between a telephone and a PC. (Again, in IP telephony parlance, calls between a telephone and a PC are also known as PC-to-phone calls or phone-to-PC calls.)
3. 
   
   Network access server. Connects a central office switch to an IP router. (Though previously discussed, it is included for completeness, because this type of gateway can be controlled in exactly the same manner as others.) Such a gateway may have an ISDN interface similar to that of the access gateway.
4. 
   
   Residential gateway. Connects analog phones to an IP router. Such a gateway typically supports a small number (two to four) of analog lines and is located on the customer premises. It brings the Internet interconnection point directly to the curb and maximizes the use of the IP network for calls between two telephones as well as between a telephone and a PC.

Active Networks | QoS

Some implementations of active networks (AN) exist (for example, see www.cccc.com), but no standards projects are currently associated with them. The area of the application of AN is larger than ensuring QoS, but AN is viewed with much interest in the research and development communities as a possible means of ensuring and supporting QoS.

As Calvert et al. (1998) observe, AN means different things to different people. In a sense, this is true, although everyone seems to agree that, in a nutshell, AN is about programmability of network elements (for example, routers) and—to an extent—bypassing, if not totally eliminating, standardized protocols, replacing them with dynamic, created-on-the-fly protocols. Marcus et al. (1998) lament that “existing protocols do not operate well for emerging applications or take advantage of novel network technologies,” citing “IP’s inability to capitalize on sub-networks which offer quality of service . . . guarantees.” While one could argue with this particular example, there is a point in the complaint. It is indisputable, however, as the authors further note, that “Forming a consensus within large groups is a slow process, and is likely to remain slow; therefore, protocol standards will continue to evolve at a slow pace.” The question, of course, is whether this pace is sufficient for the market development, and only the future will bring the answer. The idea behind AN is quite similar to (if not influenced by) the idea that resulted in the creation of Java. The language [Hypertext Markup Language (HTML)] and the protocol [Hypertext Transfer Protocol (HTTP)] that made possible the Internet killer application—the World Wide Web—do not support rapid interaction of the user with the page. Such interaction has been made possible by the invention of the principle by which a program (applet) written in Java is sent to the user’s personal computer (PC) or Internet appliance and then interpreted locally (by a Java interpreter). The user actually sees no difference (unless a silly message on a screen proudly announces that a Java program is being executed). The user simply clicks on an object, and HTTP carries the Java code to the user’s machine. Now, AN proposes pretty much the same mechanism, except that the active code is to be carried not in the application protocol message but in a network layer packet, and this code is to be executed not (or, in general, not only) at the host, but by the network elements themselves. Although many questions can be asked (most cannot yet be answered) regarding the security issues involved with this approach and its exact applications, it is relatively straightforward to see how in principle the QoS-related state of a router can be changed with unprecedented efficiency, and how the network-wide services could potentially be implemented. A specific and somewhat less futuristic application of AN to network management is described in Raz and Shavitt (1999).

The overall architecture for AN is being developed in the Defense Advanced Research Projects Agency (DARPA), the same organization that sponsored the development of what has now become the Internet. Several universities (notably Berkeley University, Columbia University, Georgia Tech, MIT, the University of Arizona, the University of Kansas, the University of Pennsylvania, and Washington University—by no means an exhaustive list), as well as the research facilities of major corporations, have AN projects.

There are two things on which the AN community agrees: (1) Networks must be service independent and (2) end-to-end service programs must be network independent. Do these sound like early IN principles? Exactly! After all, the more things change, the more they stay the same.

Label or Tag Switching | QoS

We now move to label switching or, as called by some implementations, tag switching, which is standardized by the IETF. The technology was initially developed for the purpose of interworking between the IP-based and ATM and frame relay networks, and it was later developed to apply to any network layer protocol (hence the multi- designation). The ATM (B-ISDN) switches follow the PSTN model in establishing and maintaining virtual circuits and virtual paths. The B-ISDN access protocol specifies the QoS, which is then guaranteed by the network.

To get an idea of MPLS, try to answer the following question: If the ATM and IP networks are to interwork, what should the router on the border of the ATM and IP networks do? The most straightforward answer is try to maintain the virtual circuits and virtual paths. To do so, the first router in a chain would need to “understand” the “ATM language” and act (that is, route the packet) based on the connection identifier established by the ATM switch. The next router on the path to the destination does not necessarily have to “understand” the same “ATM language,” but then it needs to understand whatever means the first router uses to identify a connection. The same applies for the rest of the routers on the path to the destination.

This is precisely how the MPLS routers work. They make forwarding decisions based on a fixed-length string called a label to decide how to forward the packet. The labels are meaningful only to the pair of routers sharing a link, and only in one direction—from a sender to the receiver. The receiver, however, chooses the label and negotiates its semantics with the sender by means of the Label Distribution Protocol (LDP). The label can indicate not only where to forward the packet (that is, which port to use), but also the QoS characteristics of the packet that specify its priority and suggest an appropriate treatment.

This approach is very different from the traditional (that is, non-MPLS routing) approach, where a router makes forwarding decisions based on the IP header. In the traditional approach, the routing table must be searched, which takes more time and processing power than a lookup in a label table, which the label-based forwarding requires. Furthermore, the routers that are not capable of analyzing the network layer packet can still perform the label lookup and replacement (a much simpler operation). Another advantage of MPLS is that using the labels (that is, in a sense, maintaining the history of the path) allows the forwarding decisions to be made based on the identity of the router at which the packet enters the network—packets entering the network via different routers are likely to be assigned different labels. Finally, when a packet is to be forced to follow a particular explicit route (rather than be left to the mercy of routing algorithms), the MPLS label can be used to represent the route. RSVP can be extended to complement MPLS by associating labeled paths with the flows. With that, resource reservations associated with an explicit route can be made to guarantee QoS.

We should mention one more important MPLS application: MPLS provides an excellent mechanism for tunneling by stacking the labels and thus supporting nested routing decision making. One important potential application of combining RSVP and MPLS is that the resulting tunnels can be routed away from the points of network failure or congestion. We highly recommend the work of Armitage (2000) as a comprehensive review of the subject.

Another means for ensuring QoS is network-wide enforcement policies, which are rules for control of the network resources and services. In describing these, we follow Kozik et al. (1998). Quality of service is only one aspect of policy-based networks; others are security, authorization, and accounting. These aspects are often inseparable—the accounting function, for example, may determine whether the present level of use has been paid for (by keeping track of the use of the resource). If use has not been paid for, policies can restrict access to the resource or affect QoS by downgrading the level of use.

The architecture of policy-based networks—sometimes also called directory-enabled networks. The architecture actually repeats the IN conceptual pattern in both the way that the policies are stored and the way that they are accessed by network elements (for example, routers, access servers, or telephony gateways). The policies are stored centrally in a policy database by a policy management system. When a network element detects an event that requires policy access (such as a request to provide bandwidth in order to establish an IP telephony call), the network element queries a policy server, which in turn consults the policy database and then either denies the request or carries it through by instructing all concerned entities to perform the actions that would enforce the policy.

Figure 1: The architecture of policy-based networks.

The IETF is addressing the subject of policy-based networks in the Policy Framework (policy) working group.

Fair Queuing and Weighted Fair Queuing | QoS

Using the new queuing schemes, each flow now has its own queue. With the fair queuing policy, the packets are transmitted round-robin in order to guarantee each flow an equal share of the capacity (possibly penalizing flows that have large packets at times of network congestion). Weighted fair queuing—an algorithm that is widely used in today’s advanced QoS-capable routers—assigns each different type of flow its (by no means necessarily identical) share of bandwidth. Figure 1 illustrates the concept: In Figure 1a, with the first-come, first-served queue, airplanes, cars, and elephants move in the same order in which they have arrived (a scheme that would cause plane crashes and annoy the drivers of the cars following elephants!). In Figure 1b, with fair queuing, the queues are formed per each flow (defined here as a formation of planes or cars or a caravan of elephants), but they are preempted so that bigger things have to wait until an equivalent number of smaller things passes (still, a maddening experience for elephants!). In Figure 1c, with weighted fair queuing, the planes are given the right of way, so they move through the queue almost without slowing down and always keeping formation; the planes are followed by cars, and the cars by the caravan of elephants. This property of keeping the packet “formation” eliminates delay variance (called jitter).

Figure 1: Queuing and scheduling in routers. (a) First-come, first-served queuing. (b) Fair queuing. (c) Weighted fair queuing.

In 1992, A. Parekh and R. Gallager of MIT demonstrated that a flow that experiences a service rate slightly higher than the flow’s data rate has a bounded delay. In other words, by requesting that a flow not exceed a certain rate, the network can guarantee that the delay experienced by the flow does not exceed a certain value. (A good example of a similar result is green streets in cities, where stoplights are adjusted so that a car traveling at a certain speed—for example, 25 mph—is guaranteed a green light at about 9 out of 10 intersections.)

The scientists then augmented the weighted fair queuing with the specification of guaranteed delay for each flow. This work resulted in a new architecture for what its creators called integrated services packet networks [compare with the expansion of the integrated services digital network (ISDN)] in Clark et al. (1992). Two types of services—guaranteed (which supports real-time traffic with determined latency and jitter bounds) and controlled-load (which deals with relaxed traffic)—were defined. At that point, the groundwork was laid for the standardization work in the Internet Engineering Task Force (IETF). The protocol that defines integrated services, called the Resource Reservation Setup Protocol (RSVP)—which is not a routing protocol. In a nutshell, RSVP, which was designed with multicasting (that is, sending a message to multiple receivers) in mind, makes bandwidth reservations—from destination to source—in the routers along the spanning tree covering multicast group members. The routers store the necessary state information, which is then maintained by sending specific RSVP messages in both directions.

The integrated services approach has been comprehensive, but apparently far too ambitious to implement widely. One recurring sentiment is that the overhead associated with reservations is far too large; the other is that it is overkill as far as the short-lived flows (of which most of the present Internet traffic consists) are concerned. (The counterargument to the latter is, of course, that the model was not created with the short-lived flows in mind; but then, something needs to be done about the short flows, too.) The third concern (Weiss, 1998) regarding the integrated services approach is that it would make charging those who request a higher QoS difficult. In any event, while the applicability of the RSVP to wide area networks and the Internet is questioned, it is being implemented for smaller enterprise networks. In essence, the integrated services approach has been a top-down one—guaranteeing absolute QoS in the network on a per-flow basis.

A bottom-up alternative technology, where QoS building blocks (which routers can recognize and act on) are defined, is called differentiated services (Kumar et al., 1998; Weiss, 1998). This technology has been actively addressed by the IETF and has resulted in a standard. The concept behind the technology is definition of various classes of services. The service provider establishes with each customer a service level agreement (SLA). Among other things, an SLA specifies how much traffic a user may send within any given class of service. A particular class of service of a packet is encoded in its IP header. The traffic is then policed at the border of the service provider’s network. Once the traffic enters the network, specialized routers provide it with differentiated treatment, but—unlike the case with the integrated services approach—the treatment is based not on a per-flow basis, but solely on the indicated class of service. The overall network is set up so as to meet all SLAs.

Quality of Service (QoS) | Converged Networks

The Glossary of the Telecommunications Terms of the U.S. Federal Standard 10377 (available at www.its.bldrdoc.gov/fs-1037/fs-1037c.htm) defines quality of service (QoS) as:

1. The performance specification of a communications channel or system. . . . 2. A subjective rating of telephone communications quality in which listeners judge transmissions by qualifiers, such as excellent, good, fair, poor, or unsatisfactory.

This definition best expresses both the objective (that is, something based on a computable metrics) and subjective (that is, perception-based) aspects of the QoS concept. Three objectives that drive the need to integrate the Internet and the PSTN relate to QoS: (1) carrying voice across both the IP networks and the PSTN, (2) combining the PSTN transport and IP access to services, and (3) accessing the IP networks over the PSTN lines. The first item is associated with the most perceivable QoS requirements. Nevertheless, the IP network access aspect (and connected to it, issues of supporting VPNs) is equally important, as we will demonstrate.

To begin with, different applications (or, rather, their users) have different perceptions of what the QoS is. Using an application called telemedicine, a doctor may expect a copy of a brain tomogram taken in a remote laboratory. The transmission can be delayed for a few minutes; however, the doctor cannot afford to have any detail of the tomogram compromised (a missing detail could wrongly suggest a tumor or leave it undetected)—the chief QoS requirement here is that no data arrive in error. On the other hand, in an IP telephony application, an occasional error in the signal would cause no problem; however, a long delay (or a variable delay, called jitter) is likely to be unacceptable.

The model of routing for the PSTN is based on the concept of circuits, which are created end to end for the duration of the session. Circuits are mapped into fixed switched physical connections. Thus, any message between two end users in a session always traverses the same physical path for the duration of a session. [For conferencing, such circuits can be bridged (that is, joined) by switches or other devices that have switching fabric; thus, any multicast message will follow the same, predetermined set of physical circuits.] With this routing model, it is possible to determine whether a session that requires certain characteristics (such as bandwidth or loss tolerances) can be established. Once the session is established, it is relatively straightforward to guarantee that the requested characteristics will remain constant for the duration of the session.

One important factor in PSTN routing is the time that it takes to set up a circuit; the call setup time has traditionally been an essential QoS metric in the PSTN. Incidentally, this model, which naturally grew out of telephony, was applied by the ITU-T to the definition of virtual circuit for data communications standards. First, this concept was defined in X.25 and, subsequently, frame relay and asynchronous transfer mode (ATM) networks. ISDN access guarantees certain bandwidth (depending on a particular national standard) to the subscriber. Broadband ISDN (B-ISDN) access, in addition, specifies parameters that are needed by the ATM network. At this very moment, the mechanisms are being developed to back up B-ISDN by Intelligent Network, which could ensure policy-based networkwide QoS enforcement. Overall, in today’s PSTN, the main QoS metric is, as mentioned before, the call blocking rate.

The Internet routing model, on the other hand, has traditionally avoided stressing any built-in mechanism for creation and maintenance of virtual circuits. In the Internet, QoS issues (which also define their respective metrics) include bandwidth availability, latency (that is, end-to-end delay) control, jitter (that is, delay variation) control, and packet loss. Historically, the IP networks have been supporting what is called the best effort (but no guarantees) of packet transmission. In this system, no differentiation among different types of traffic is made, and neither the sequence of packet arrivals nor the arrival itself is guaranteed.

Whatever the end-to-end QoS requirements may be, at the network layer the packets travel (similarly to those of us who take airplanes) from hub to hub (that is, from router to router). Each router queues newly arriving packets for retransmission over the link to the most suitable (according to the routing table) router or destination host. Until very recently, most routers used a first-come, first-served queuing discipline, which is fair to all packets and, for this very reason, cannot make some packets more equal than others.

Overall, for applications such as voice or video over IP, a new network layer model was clearly needed, and such models have been researched and implemented since the 1990s. Two new approaches proposed mechanisms that are now called fair queuing and weighted fair queuing . With fair queuing and weighted fair queuing, routers are no longer required to treat packets equally. The incoming traffic is separated into well-defined flows. (A TCP connection is an example of a flow, although it may be difficult to detect by a router—all TCP connections between the same pair of hosts is a more realistic example; a voice session is another one.)

Internet-Supported PSTN Services

Last year a colleague of ours was called by a reporter from a well-known technical publication and asked to describe the effort of the PSTN/Internet Internetworking (pint) working group in the Internet Engineering Task Force (IETF). Our cautious colleague wisely decided that the best he could do under the circumstances was to read out to the reporter a few selected sentences from the working group charter published by the IETF on its Web page. Specifically, he stressed—prompted by the pint Web page—that the purpose of pint was to “address connection arrangements through which Internet applications can request and enrich PSTN . . . telephony services.” The reporter wrote down what she heard. Later, in accordance with her agreement with our colleague, she sent to him the draft of the article. The article was accurate except for one word: enrich had turned into unleash—this is what the reporter heard over the telephone line (an atypically imperfect telephone line, we presume). When the amazed author called the reporter with the correction, she seemed to be disappointed. (And so are we. We wish the pint charter really did talk about unleashing the telephone services, because this is precisely what the Internet-supported PSTN services are all about!)

In his recent article on Intelligent Network, Scott Bradner correctly observes that the intelligence of IN is strictly in the network, not on its edges. This lack of edge intelligence is precisely what the interworking of the Internet and IN is to change—once and forever!

The basic goal of the IN technology is to allow the user of an Internet host to create, access, and control the PSTN services. A simple example of what the technology can do is the click-to-dial-back service, where you click on an icon displayed on a Web page and the PSTN call is established as the result.

As simple as it sounds (and a crude implementation of this service is just as simple), the unleashing quality of this service alone should not be underestimated. The competition among the long-distance service providers is such that they would do almost anything to get a call flowing through their networks. With Web-based access, their customers are around the world! There are still countries that protect their networks by forbidding the call-back service; people may get angry about such backward and anticompetitive practices, but click-to-dial-back, which technically is not a call-back service, is a way to get even.

Once you start thinking about the possibilities of tweaking this basic concept, you will find that the possibilities for creating new services are virtually unlimited. We will demonstrate a straightforward extension of click-to-dial-back to drastically improve on a PSTN-only service.

The service in question, interestingly enough, came to life as an application of the Web business model (but not the Web technology) to the PSTN. Telephony service providers in Europe started to market free telephone calls to those who would agree to listen to several minutes of audio advertising. The advertisers have so far found the approach ineffective—pure audio is hardly the best means of delivering advertising today. This already bad effect is further worsened by the “push” nature of audio advertising.

Now, with the technology just described, the prospective caller can access the service provider’s page on the Web, where he or she can also subscribe to the service and register a profile stating the preference for the types of products he or she wishes to learn about. Every time a call is to be made, the caller can then be walked through a video presentation of the advertisement on his or her Internet appliance—possibly accompanied by the audio portion over the PSTN line. The caller can control the pace of the advertisement; when it is finished, the caller will be prompted for the number he or she wishes to call and then connected to that number.

There are several early developments in this area (Hubaux et al., 1998). In the relevant architecture, the SCPs and SNs are connected to the Internet (which is fairly easy to achieve because they are almost invariably implemented on the Unix system platform).

Figure 1: The architecture for Internet-supported PSTN service delivery.

It is important to repeat that with this arrangement only SCPs and SNs—but by no means the switches—are connected to the Internet and thus can communicate with other IP hosts.

The service control function can be distributed to Internet hosts as much as the PSTN service provider allows and the owner of a particular IP host (who may be the same PSTN service provider) wishes to handle. In the WebIN, the IP host is actually a Web server, and part of the service control function (called WebSCP) is moved into the Internet (Low, 1997; Low et al., 1996). This arrangement can be used to provide the main features of the PSTN VPN service—the private numbering plan and closed user group in particular (Hubaux et al., 1998). The translation map of the enterprise-significant numbers to the PSTN-significant ones, as well as the specification of the closed groups (including the calling privileges of each group member), are kept in the databases accessible through the Internet. Part of the SCP service logic is executed by the WebSCP. While there is very little interoperability among the legacy IN implementations, integrating them with the Internet immediately establishes a common language for interworking. Even more significant is that the service creation and service management are also moved (via the Internet) to the edge of the network.

As exciting as the opening of PSTN call control to the Internet hosts may be, there is a serious and not completely solved problem associated with it. This problem is security, and it is ubiquitous in the Internet. While trusted relations between the PSTN and IP entities can be established between the enterprise networks, opening IN control fully to anyone on the Internet remains problematic. There may be no need for IN control to be fully opened, except in the cases of some well-understood services.

Potential security issues also prevent (at least for the time being) direct connection of the switching offices to the Internet. If that were done, the SCP itself could be placed on the Internet, and subsequently anything that IN does presently could be done in the Internet. Although there is a single ITU-T standard, it specifies different options. These options reflect implementations that differ not only between different continents (that is, Europe and North America) but also among the network operators in the United States. (Bell Operating Companies use the option corresponding to the Bellcore AIN; some IXCs use proprietary or European versions of IN.) Thus, direct interconnection of the PSTN switches and Internet SCPs, even if secure, would not provide global interoperability. Only interworking of the PSTN service control with the Internet hosts holds the promise (on which it has already started to deliver) of universal, global access to service control of the PSTN.

To conclude this section, we would like to repeat that so far we have taken an intentionally one-sided approach to the role of IN in the integration of PSTN and the Internet. Our only goal was to demonstrate how IN can be used to give more control in creating and executing services to the edge of the network. At this point, it is important to observe an interesting duality: While the PSTN benefits from Internet-based service creation and control, the IP networks greatly benefit from the existing PSTN-based service control for at least three different reasons: (1) efficiency of the access to IP networks; (2) provision of certain combined PSTN-Internet services; and (3) support of the existing PSTN services in the IP telephony environment.