Chapter 1 - Signaling System 7

	Chapter 1 Signaling System #7

		Early telephone networks were the result of years of evolution, with little thought about future technology. Based around analog equipment, the telephone network of the early telephone company was not well suited for services such as data and video. Many individual technology service providers began popping up during the 1960s, providing packet-switching networks and data communications services the telephone companies were just not equipped to provide.

		The international telephone network was facing the same problems. In many countries, just getting telephone service was a feat in itself. As international bodies began investigating alternative technologies for providing telephone service to the masses (such as cellular), the need for an all-digital network became apparent. Thus arose the beginnings of an all-digital network with intelligence.

		Why intelligence? To understand the answer to this question, you must first understand the mechanics of a telephone call. When a subscriber picks up a telephone receiver, an electrical signal is sent over a wire to a telephone switch. The telephone switch detects electrical current on this wire and interprets this ''signal" as a request for dialtone.

		But let's say the subscriber wants to transmit data over this same line using packet switching rather than an analog modem. The information sent to the telephone switch has to define the transmission as digital, as well as data and not voice, before the switch can determine how to handle the call. This is only one portion of the call.

		To transmit the data to another network, the switch must determine first how the data is to be routed (to what destination), and what circuits to use to reach the destination. After this has been determined, some form of request must be sent to the telephone switch on the other end of the circuit to request a connection. This continues all the way through the network, with the same requirements at each leg of the call. Telephone switches need the ability to signal one another and share information regarding the type of transmission, how the transmission is to be routed (call destination), and what the contents of the transmission are (audio, video, data, and so on).

		If there is to be special handling or routing for a call, the telephone switches involved in routing the call must be able to obtain these instructions. Rather than store routing instructions for every single telephone number in the world within each and every telephone switch, each network is responsible for their own network database. The telephone switches then need the ability to connect and communicate with these databases to obtain the special instructions.

		This is a high level view of what signaling networks really do. They allow telephone switches (and now packet switches) to communicate directly with one another, and share information needed to process any type of a call autonomously. Signaling System #7 (SS7) was originally designed for the analog telephone network, but has continually undergone enhancements and changes to accommodate the ever-changing world of telecommunications. Today, SS7 is used for data, video, voice, audio, and even Voice over IP (VoIP) networks.

		The International Telecommunications Union (ITU) commissioned the then CCITT to study the possibility of an all-digital intelligent network. The result was a series of standards known now in the United States as Signaling System #7 (SS7). These standards have paved the way for the Intelligent Network (IN) and, with it, a variety of services, many yet to be unveiled.

		This book outlines the technologies related to the SS7 protocols and details how the protocols work within the IN.

		Introduction to SS7

		The ITU-TS (once known as the CCITT) developed a digital signaling standard in the mid-60s called Common Channel Interoffice Signaling System #6 (CCIS6) that would revolutionize the telephone industry. Based upon a proprietary, high-speed data communications network, CCIS6 later evolved into C7 (known as Signaling System #7 (SS7) in the U.S.), which has now become the signaling standard for the entire world.

		The secret to its success lies in the message structure of the protocol and the network topology. The protocol uses messages, much like X.25 and other message-based protocols, to request services from other entities. These messages travel from one network entity to another, independent of the actual voice and data they pertain to, in an envelope called a packet.

		CCS was first introduced in the United States in the 1960s as CCIS6. Developed by the International Telecommunications Union Telecommunications Standards Society (ITU-TS), CCIS6 used a separate facility for sending signaling information to distant telephone offices.

		The first deployment of CCIS6 in the U.S. used 2.4-kbps data links. These were later upgraded to 4.8 kbps. Messages were sent in the form of data packets and were used to request connections on voice trunks between two central offices. This became the first use of packet switching in the Public Switched Telephone Network (PSTN). The packets were assembled by placing 12 signal units of 28 bits each into a data block. This is similar to the method used in SS7 today.

		SS7 was derived from the earlier CCIS6, which explains the similarities. SS7 provides much more capability than CCIS6. Where CCIS6 used fixed-length signal units, SS7 uses variable-length signal units (with a maximum sized length), providing more versatility and flexibility. SS7 also uses higher speed data links (56 kbps). This makes the signaling network much faster than CCIS6. In international networks, the data links operate at 64 kbps. Recently, high speed links (HSL) operating at 1.544 Mbps have been deployed in the US, conforming to a Telcordia (formerly Bellcore) standard. TCP/IP has also been introduced as a transport for SS7 providing 100 Mbps facilities.

		As of 1983, CCIS6 was still being deployed throughout the U.S. telephone network, even though SS7 was being introduced. As SS7 began deployment in the mid-1980s, CCIS6 was phased out of the network. SS7 was used in the interoffice network and was not immediately deployed in the local offices until many years later.

		In fact, the first usage of SS7 in the U.S. was not for call setup and teardown, but for accessing remote databases. The opposite is true of Europe and other International communities, where C7 is still used today for call setup and teardown, but the concept of centralized databases for custom call routing is still new. In the 1980s, the U.S. telephone companies offered a new service called Wide Area Telephone Service (WATS), which used a common 800 area code regardless of the destination of the call. This posed a problem for telephone-switching equipment, which uses the area code to determine how to route a call through the PSTN.

		To overcome this problem, a second number was assigned to every 800 number. This second number is used by the switching equipment to actually route the call through the voice network. But the number had to be placed in a centralized database where all central offices could access it. This database became a popular commodity for all telephone companies and still exists today.

		When an 800 number is dialed, the telephone company switching equipment uses a data communications link to access this remote database and look up the actual routing number. The access is in the form of a message packet, which queries the network for the number. The database then responds with a response message packet, providing the routing telephone number as well as billing information for the 800 number. The switching equipment can then route the call using conventional signaling methods.

		SS7 provides that data communications link between switching equipment and telephone company databases. Shortly after the 800 number implementation, the SS7 network was expanded to provide other services, including call setup and teardown. Still, the database access capability has proven to be the biggest advantage behind SS7 and is widely used today to provide routing and billing information for all telephone services including 800 numbers, 900 numbers, 911 services, custom calling features, caller identification, and many new services yet to be offered.

		800 numbers at one time belonged to one service provider. If subscribers wanted to change service providers, they had to surrender their 800 number. This was due to the location of the routing information. All routing information for 800 numbers is located in a central database within the carrier's network and accessed via the SS7 network. SS7 is now used to allow 800 numbers to become transportable and to provide subscribers the option of keeping their 800 numbers even when they change service providers.

		When someone dials an 800 number today, the telephone switch sends a query to the network database to first determine to which carrier the 800 number belongs. After the switch receives this information, it can direct a query to the network of the carrier owning the 800 number and translate the 800 number into an actual routing number. This concept was later extended to support number portability.

		Without SS7, number portability would be impossible. Local Number Portability (LNP) is a service mandated by the FCC in 1996 which requires telephone companies to support the porting of a telephone number. If customers wish to change their service from Plain Old Telephone Service (POTS) to ISDN, they would normally be forced to change telephone numbers. This is because of the way telephone numbers are assigned in switching equipment, with switches assigned ranges of numbers.

 

The same is true if a subscriber decides to change to a new carrier offering local service in their area. Changing telephone companies would require surrendering your old telephone number and receiving a new number from the new carrier. With number portability, subscribers can keep their old numbers and still change carriers.

		This requires the use of a database to determine which switch in the network is assigned the number, very similar to the way 800 numbers are routed. Future implementations of LNP will support subscribers moving from one location to another without changing their telephone number (even if they move to a new area code). This obsoletes the former numbering plan and the way calls are routed through the telephone network.

		In addition to database access, the SS7 protocol provides the means for switching equipment to communicate with other switching equipment at remote sites. For example, if a caller dials a number which is busy, the caller may elect to invoke a feature such as automatic callback. When the called party becomes available, the network will ring the caller's phone. When the caller answers, the called party phone is then rung. This feature relies on the capabilities of SS7 to send messages from one switch to another switch, allowing the two systems to invoke features within each switch without setting up a circuit between the two systems.

		Cellular networks use many features requiring switching equipment to communicate with each other over a data communications network. Seamless roaming is one such feature of the cellular network that relies on the SS7 protocol.

		Cellular providers use the SS7 network to share subscriber information from their Home Location Registers (HLRs), so cellular subscribers no longer have to register with other service providers when they travel to other areas. Cellular providers can access each other's databases and share the subscriber information so that subscribers can roam seamlessly from one network to another.

		Before deploying SS7, cellular providers were dependent on X.25 networks to carry IS-41 signaling information through their network. This did not allow them to interconnect through the PSTN because the X.25 network was not compatible with the PSTN signaling network (SS7). Some of the largest SS7 networks deployed today are owned by wireless carriers.

		Today, SS7 has been deployed throughout the Bell Operating Companies (BOCs) network, and is being deployed by almost all independent telephone companies and interexchange carriers as well.

 

This makes SS7 the world's largest data communications network, linking telephone companies, cellular service providers, and long distance carriers together into one large information-sharing network.

		SS7 supports many new features and applications. Because of its ability to transfer all types of digital information, this new network is being used to deliver many sophisticated services to the customer premises such as custom calling features and intelligent routing. Many new applications are still under development. The SS7 network interconnects thousands of telephone company providers all over the world into one common signaling network.

		New technology will continue to place demands on the signaling network. SS7 continues to evolve and become more sophisticated as new features are added. While the network is sophisticated enough to work on its own with very little interaction from maintenance personnel, when problems do arise, knowledge of the protocols and the processes that take place between network entities is critical.

		Yet to fully understand what SS7 is about, one must understand the conventional signaling methods used prior to SS7 in telephone networks. The following discussion explains the signaling methods used prior to SS7.

		Introduction to Telephony Signaling

		Ever since the beginning of the telephone, signaling has been an integral part of telephone communications. The first telephone devices depended on the receiving party standing next to the receiver. Early telephones did not have ringers like today's telephones, and used crude speakers to project the caller's voice into the room. If the party being called was not within close proximity of the speaker, he or she would have no indication of an incoming call.

		Later, after the formation of the Bell Telephone Company, Alexander Graham Bell's faithful assistant Watson invented the telephone ringer. This new signaling method served one purpose: to alert the called party of an incoming call. When the called party lifted the receiver, another form of signaling used DC battery and ground to indicate the called party had answered the telephone and completed the circuit. Although not having an immediate impact, this method became important when the first telephone exchange was created. By lifting the receiver and allowing DC current to flow through the phone and back through the return of the circuit, a lamp would be lit on the exchange operator's switchboard. This signaled the operator when someone needed telephone service, and was often accompanied by a buzzer.

 

Signaling has evolved over the decades to include significantly more information than these early methods could. Consider the typical long distance telephone call today. When a caller dials the area code and prefix of the telephone number, the local exchange must determine how to route the call. In addition, billing information must be passed to a central database. If the caller is using a contemporary digital facility (such as T-1 or ISDN), information regarding the digitization of the line must also be provided.

		Early signaling methods were limited because they used the same circuit for both signaling and voice. They were also analog and had a limited number of states, or values, which could be represented. The circuit would be busy from the time the caller started dialing until the caller went "on-hook." To compound the problem, the telephone companies were quickly running out of facilities and were in desperate need of additional facilities.

		Many telephone companies in metropolitan areas such as Los Angeles were facing substantial investments to add new facilities to support the millions of customers that were creating an enormous amount of traffic. The telephone companies had to find a way to consolidate their facilities, making more economical use of what they had. In addition, they needed a service that would vastly improve their network's capability and support the many new services being demanded by subscribers.

		Europe had already begun the process of digitizing the network in the early 1960s. One of the first steps was to remove signaling from the voice network and place the signaling on a network all its own. This way, the call setup and teardown procedures required with every call could be faster than the previous methods, and voice and data circuits could be reserved for use when a connection was possible, rather than maintaining the connection even when the destination was busy. Common Channel Signaling (CCS) paved the way for services the early pioneers of signaling never dreamed of. CCS is the technology that makes ISDN and SS7 possible.

		The concept behind CCS is simple. Rather than use voice trunks for signaling, they are used only when a connection is established. For instance, when a call is placed to a distant party using conventional signaling, the signaling for that call begins from the time the caller lifts the receiver and goes off-hook until the caller goes back on-hook. After the end office has received the dialed digits, an outgoing trunk to the destination end office is seized, based on a routing table entry and the digits dialed.

		The voice circuit remains busy even if the distant party never answers the call until the calling party hangs up. Meanwhile, other subscribers are tying up other voice circuits by placing calls of their own. This is not good utilization of voice circuits and it placed immediate limitations on the networks. But if the signaling could be placed over a different network and the voice circuit used only when the called party answered, the voice circuit would remain available for a longer period of time. This meant the availability of voice circuits would be higher and the need for additional circuits would decrease.

		When a caller is to receive an intercept recording ("all circuits are busy"), the same trunk used for the voice is also used for the recording. The recording is sent by the distant office. Busy tones and other service tones are sent over the trunk by the distant office to the caller. With SS7, the caller's local office can provide these tones and recordings at the command of the distant office. These commands are received via the signaling network. The voice trunk is left unconnected (although in some implementations, one side of the trunk is connected for transmitting tones and recordings).

		The procedure for tearing down a circuit is much faster in CCS than in conventional signaling, and is not as error prone. Even if voice circuits do get connected, with the speed of the signaling network, circuits can be disconnected and quickly connected again for a new call. While a call is in progress, information regarding the call can be sent through the SS7 network (for instance, information from a database requested during an interactive multimedia call).

		When signaling information is placed on existing digital transmission facilities (such as DS0 or DS1), it uses a fraction of the circuits required for in-band signaling (discussed later in this chapter). One digital data link can carry the signaling information for thousands of trunks and maintain thousands of telephone calls. When TCP/IP is used as the transport for signaling, this number increases significantly.

		CCS is in wide use today, even though many in the telecommunications business do not understand it. SS7 is the protocol and architecture used in this new network and is the topic of this book.

		There are many methods used for signaling, with only a few used in the telephone network. None of the methods described in this section can support network management functions or control information between switches and operations systems. The exception is SS7. Because SS7 consists of a data network using data messages, SS7 can meet the demands both now and in the future of the evolving telephone network.

		Signaling takes place in two parts of the telephone network between the subscriber and the local end office, and from switching office to switching office within the telephone company network. The signaling requirements are similar, though interoffice signaling can be more demanding.

		There are two basic functions of signaling: addressing and supervision. With the earlier methods of signaling, supervision was simple. If current existed from one end to the other, the circuit was good. For addressing, dialed digits would be passed through the network in the same fashion as they were originated, either in pulses or tones. Only the destination address could be provided.

		But as the telephone network grew more sophisticated, the signaling methods grew as well. Signaling between the subscriber and the central office now includes the calling party number, which is forwarded to the called party and displayed before the phone is even answered. Interoffice signaling now includes information obtained from regional databases, pertaining to the type of service a subscriber may have or billing information.

		Calling card validation is another important function of these databases, and provides security against telephone fraud. Personal identification numbers are kept in a subscriber database and verified every time a call is placed using a calling card.

		Previous to SS7, signaling was accomplished over the same facility as the voice call. This was accomplished in many cases using DC current. There are many disadvantages to DC signaling, which is what led to the development of SS7.

		In addition to DC signaling, many companies used Single Frequency (SF) signaling. This was also accomplished over the same facility as the voice. This method of signaling used tones above the voice frequencies, but still within the 4-kHz bandwidth of the facility to set up and tear down circuits. Other methods have been used in addition to DC and SF signaling, depending on the type of facility. But with all of the various signaling methods, none can offer the features and versatility of SS7.

		Conventional Signaling

		Conventional signaling relies on many different types of mechanisms, depending mostly on the location within the network. Dual-Tone Multi-Frequency (DTMF) is used between the subscriber and the end office. SF is used between telephone company offices. Following is an example of how conventional signaling is used to process a call.

		DC signaling relies on DC current to signal the distant end. The simplest example of DC signaling is used in Plain Old Telephone Service (POTS) between the subscriber and the local end office. When a subscriber goes off-hook, DC current from the central office is allowed to flow through the telephone (the switch-hook provides the contact closure between the two-wire interface) and back to the central office. The central office switch uses a DC current detector to determine when a connection is being requested.

		The central office acknowledges receipt of the loop current by sending a dialtone. A dialtone signals the subscriber to begin dialing the telephone number. This can be done using a rotary dial or a DTMF dial. Rotary dials use a relay to interrupt the current creating pulses (10 pulses per second). The central office switch counts each series of pulse "bursts" to determine the number dialed.

		When DTMF is used, the dial creates a frequency tone generated by mixing two frequencies together (hence the name dual-tone). The central office switch "hears" these tones and translates them into dialed digits.

		After the telephone number has been dialed, the central office switch must determine how to connect to the destination. This may involve more than two central offices. A facility (or circuit) must be connected between every telephone company office involved in the call. This circuit must remain connected until either party hangs up. The originating office determines which circuits to use by searching its routing tables to see which office it must route the call through to reach the final destination. That office, in turn, will search its routing tables to determine the next office to be added to the call.

		Once the circuits are all connected, the distant party can be alerted by sending a generator (80 V AC at 20 Hz) out to the telephone. This activates a ringer inside the telephone. At the same time, the distant telephone company switch sends a ringback tone to the originator to alert the caller that the called party's phone is being rung. When the distant party answers, the ringback tone is interrupted and the circuits now carry the voice of both callers.

		If the called party is busy, the same facilities are used so that the far end office can send a busy tone back to the originator. This means those facilities cannot be used for any other calls and are being tied up to send the busy tone.

		The limitations of DC signaling are somewhat obvious. For example, the telephone number of the originator cannot be sent to the called party (at least not without long delays in setup). Signaling is limited to seizing circuits, call supervision, and disconnect. Because DC signaling uses the voice trunk, the trunks are kept busy even when the two parties are never connected.

 

DC Signaling

		As mentioned previously, DC signaling relies on direct current to signal distant offices. This is a very limited signaling method, because of the minimum number of states that can be represented by voltages and current. When a subscriber lifts the receiver of a phone, current is allowed to flow through the phone and back to the central office. Current detectors on the line cards in the central office switching equipment detect the current and provide the subscriber with a dialtone.

		Other types of trunks use similar techniques. E&M signaling is another form of DC signaling. These trunks use a separate pair of wires for signaling. The two wires are labeled as E and M, or ''ear" and "mouth." These, of course, are not what the letters really stand for, but this designation is often used to describe their function.

		The M lead is used to send 48 V DC or ground to the distant switch (implementation dependent). The M lead of one switch must be connected to the E lead of the distant switch. When the distant switch detects current on its E lead, it closes a relay contact and allows the current to flow back to the sending switch through its M lead.

		When the sending switch detects the current flow on its E lead, the connection is considered established and transmission can begin on the separate voice pairs. This type of trunk is often used between two PBXs, and is often referred to as tie lines.

		In-band Signaling

		In-band signaling is used when DC signaling is not possible, for example, in tandem offices. In-band signaling uses tones in place of DC current. These tones may be Single Frequency (SF) tones, Multi-Frequency (MF) tones, or Dual-Tone Multi-Frequency (DTMF). The tones are transmitted with the voice. Because these tones must be transmitted over the same facility as the voice, they must be within the voice band (0 to 4 kHz). There is the possibility of false signaling when voice frequencies duplicate signaling tones. The tones are designed for minimal occurrence of this, but this is not 100 percent fault tolerant. Signal delays and other mechanisms are used to prevent the possibility of voice frequencies from imitating SF signals.

		SF signaling is used for interoffice trunks. Two possible states exist: on-hook (idle line) or off-hook (busy line). To maintain a connection, no tone is sent while the circuit is up. When either party hangs up, a disconnect is signaled to all interconnecting offices by sending a tone (2.6 kHz) over the circuit. Detectors at each end of the circuits detect the tone and drop the circuit.

 

SF signaling has become the most popular of all the in-band methods, and the most widely used of all signaling methods. SF is still in use today in some parts of the telephone network. However, as deployment of the SS7 network spreads, SF is no longer needed.

		Multi-Frequency (MF) is much like Dual Tone Multi-Frequency (DTMF), and is used to send dialed digits through the telephone network to the destination end office. Because voice transmission is blocked until a connection to the called party is established, there is no need for mechanisms that prevent the possibility of voice imitating signaling tones.

		MF is also an interoffice signaling method used to send the dialed digits from the originating office to the destination office.

		Out-of-band Signaling

		Out-of-band signaling has not shared the popularity and widespread usage of SF signaling. Out-of-band signaling was designed for analog carrier systems, which do not use the full 4 kHz bandwidth of the voice circuit. These carriers use up to 3.5 kHz, and can send tones in the 3.7 kHz band without worrying about false signaling. In other words, some frequencies within the 4 kHz bandwidth are left as "buffers," and not used for anything. This is where the signaling took place, using tones. It is called out-of-band signaling because the signaling takes place outside of the voice frequency bands; however, it is still sent on the same facility as the voice. Out-of-band signaling is an analog technology, and is of no advantage today.

		SS7 is often referred to as out-of-band signaling. This is not an accurate label for SS7 because the messages sent in SS7 are not sent in the frequency band outside of the voice band. From a purist perspective, SS7 is best described as common channel signaling because it is sent on a completely separate facility from the voice.

		Digital Signaling

		As digital trunks became more popular, signaling methods evolved that greatly enhanced the reliability of the network. One technique used in digital trunks (such as DS1) is the use of signaling bits. A signaling bit can be inserted into the digital voice bit stream, without sacrificing voice quality. One bit is "robbed" out of designated frames and dedicated to signaling (robbed-bit signaling). The digitized voice does not suffer from this technique since the loss of one bit does not alter the voice signal enough to be detectable by the human ear.

		Because of its digital nature, digital signaling is much more cost effective than SF. SF requires expensive tone equipment both for sending and detection, whereas digital signaling can be detected by any digital device loaded in the switching equipment and can create any kind of signaling information. This has fueled the efforts to make carriers digital rather than analog.

		Digital signaling has another fundamental difference. It does not use messages as SS7 and other message-based protocols do. This limits the type of signaling it can provide.

		Common Channel Signaling

		As discussed earlier, Common Channel Signaling (CCS) uses a digital facility, but places the signaling information in a time slot or channel separate from the voice and data it is related to. This allows signaling information to be consolidated and sent through its own network apart from the voice network. It is this method that is used in ISDN and SS7 today.

		In addition, this method of signaling is capable of sending and receiving messages, supporting an unlimited number of signaling values. Even information retrieved from a remote database can be transferred from one entity to another using CCS.

		We mentioned earlier that SS7 is sometimes referred to as out-of-band signaling. When one compares the methods used to transmit SS7 with conventional out-of-band signaling, it becomes clear why SS7 is best defined as common channel signaling. Of course, out-of-band signaling is also an analog method, whereas common channel signaling is digital.

		Introduction to Telephony

		The telecommunications industry has undergone many organizational changes as well as service changes. With the current trend to merge data communications with telephony, many new professionals have entered the industry without the opportunity to learn the nuances of the industry and its players.

		This section provides a fundamental outline of the telephony industry and its players. Of major importance is understanding how the telephone network is structured, and how it has evolved over the years.

		Bell System Hierarchy

		The Bell System network really can be divided into several distinct functions: signaling, data, video, audio, and voice switching. The signaling network is the SS7 network. Data transmission is routed away from the telephone switches onto separate data networks, either owned and maintained by the telephone company or leased through a data provider. The video network is used for transmission signals from television studios out to their transmitters. This is usually over DS4/5 facilities. Audio transmission is radio broadcasts, also transmitted from the studio out to the transmitters. The switching network is the portion used for the transfer of voice from one subscriber to another, through the telephone service providers' network.

		It is important to understand this separation of networks because what is happening today with the convergent networks has a big impact on how telephone companies manage these different transmissions. Internet traffic, for example, is now "offloaded" from the switch network right at the local telephone office and immediately routed to an Internet Service Provider (ISP) through dedicated circuits leased from the ISP by the local telephone company. This allows them to remove any and all Internet-related traffic from the backbone switching network, saving the telephone company millions in equipment and facility costs. This also opens new opportunities for data network providers.

		There exists a hierarchy within the switching network, to ensure efficient use of trunk facilities and to provide alternate circuits in the event of failures or congestion. Before the divestiture of the Bell System in 1984, the hierarchy was much different than today. The hierarchy provided five levels of switching offices, with the class five office being the end office or local office (see Figure 1.1).

		The class five office was capable of connecting to other end offices within its calling area, but relied on the class four office to connect to offices outside of its calling area. The calling area was not defined by area codes, but was a geographical service area drawn by the Bell System. Service access areas have since been reallocated as Local Access Transport Areas (LATAs) by the Justice Department as a result of the Bell System divestiture.

		The class four office allowed the Bell System to aggregate its facilities and used high-capacity trunks to interconnect to other class four offices. In this way, the class five office did not require high-capacity facilities and handed off the bulk of its calls to the class four office. This also prevented the necessity for the class five office to have trunks to every other class five office in the service area. The class four office was also known as the toll center.

		As seen in Figure 1.1, the class four office provided two paths for a telephone connection. The interconnection of these various offices depended mostly on distance. There were many occurrences of class five offices connecting directly to class one offices.

 

		Figure 1.1 This figure illustrates the Bell System Switching Hierarchy prior to the divestiture of the Bell System. The priority of the routes is indicated by the numbers.

		The toll office searched its trunks for an available trunk as low in the hierarchy as possible. If one was not available, it would search for a trunk to a primary center in the destination calling area. If there were no available trunks to the primary center, then the last choice would be an overflow trunk to its own primary center.

		The class three office, or primary center, was part of the toll network. This office connected to class two offices, or sectional offices, but also provided a path to other class three offices and class four offices. This office served as an overflow switching center in the event that other routes lower in the hierarchy were not available.

		The class two office was also known as the sectional center, and provided access to the regional center. Only two routes were available at this level, one to its peer in the destination calling area and one to the regional switching center, or class one office.

		The class one office was known as the regional center, and was used for toll calls. The regional center also provided access to the long distance network. A typical toll call required an average of three trunks. The maximum number of trunks allowed in a connection was nine.

		Postdivestiture Switching Hierarchy

		In the mid-1980s, technology allowed many of the functions just described to be combined. As switching equipment was improved, systems were given the capability to act as local switches, tandems, and even toll switches. In addition to better routing functionality, these switches were also given the ability to record billing records and perform alternate routing in the event of congestion or failures.

		The new hierarchy consists of fewer levels, consolidating many of the functions of the previous hierarchy into two layers. Long distance access is accomplished through a Point-of-Presence (POP) office. The long distance carrier will also have its own multilevel hierarchy, which may be several layers as well.

		After divestiture of the Bell System, the calling areas were also redefined and changed to Local Access Transport Areas (LATAs). Within each LATA is a simple hierarchy, with three levels. With newer advanced switching equipment, many of the functions once found in the higher layers of the hierarchy can now be combined and located in the end office (see Figure 1.2).

		This of course assumes we are describing the traditional circuit-switched network. Packet switched networks are much different. A hierarchy still exists in the packet switched world, but there are fewer layers to deal with, and of course, no "switching" within the network. Everything is transmitted over the same facilities. We will talk about packet switched networks later on.

		Local Access Transport Areas (LATAs)

		After divestiture, the telephone companies' service areas (or exchanges, as they were sometimes called) were redrawn by the Justice Department so that telephone companies would have evenly divided service areas with equal revenue potential. These areas, called Local Access Transport Areas (LATAs), were divided according to census information regarding the demographics of each LATA. Each Regional Bell Operating Company (RBOC) and each independent telephone company received a service area that would provide it a fair and equal market. There were originally 146 LATAs, but as changes take place in the networks, the number of LATAs is growing.

		Figure 1.2 This figure illustrates a much flatter switching hierarchy, used after the divestiture of the Bell System.

 

		Other considerations had to be made regarding LATAs. Telephone companies already had a significant amount of equipment and capital invested in the previous service areas, which meant reassigning service areas could have a major financial impact. The service areas were divided into LATAs, which are much smaller than the original service areas defined by the telephone companies themselves. Each telephone company could have more than one LATA for which it provides service. This allowed telephone companies to maintain their original investment, but forced them to divide areas into smaller chunks.

		The difference was made between LATAs. The local operating companies and independents were not allowed to carry traffic from one LATA to another. Instead, they were required to use a long distance carrier such as AT&T or MCI to provide them that service. This ensured that the local telephone companies did not interfere with long distance competition, and provided the long distance companies a fair and profitable boundary for which they could compete with other carriers.

		In the mid-1980s, the telephone companies were forced to provide the long distance carriers equal access into the telephone network. This was accomplished in the current switching hierarchy by establishing a POP, which serves as an interface to all interexchange carriers into the LATA. Every LATA must have one POP. The telephone companies collected access fees from the long distance carriers for this interface into their network, to offset the cost of equipment and ensure a revenue stream from long distance traffic.

		The local telephone companies further divided each LATA into a local market and a toll market. The toll market is within the LATA (intraLATA) but considered by the telephone company as a long distance call, because of the distance from one city to another or the distance between central offices handling the call. These toll calls are currently very expensive, and have recently been open for competition. These are the only long distance calls local telephone companies are allowed to provide. Many states have allowed long distance carriers to compete in this market, opening up the LATA to competition.

 

In 1996, the government approved the Telecommunications Act of 1996. This piece of legislation has changed the face of the telecommunications industry in many ways, and impacts both users of the telephone network and the telephone companies themselves. One part of this legislation allows long distance companies to provide local telephone service in their markets. They must pass specific criteria defined by the Telecommunications Act before the FCC can grant them permission to provide local service. This reverses the legislation put into place by Judge Green when divestiture of the Bell System reshaped the nation's telephone industry and limited long distance companies from offering local service.

		At the same time, the Telecommunications Act of 1996 allows local telephone companies to offer long distance (interLATA) service in their market areas. They must demonstrate that they have allowed competition in their market areas, and pass criteria set in the Telecommunications Act before offering such service. The result of this sweeping new legislation is yet to be seen, but many anticipate local telephone companies will partner and merge with long distance companies taking advantage of one another's markets.

		There are over 300 LATAs presently defined throughout the United States. Throughout these LATAs are hundreds of telephone companies competing for revenues. Many of the smaller independent companies are investing capital to get connected to the SS7 network so that they can offer the same types of advanced calling features the larger service providers offer. These independent companies have joined forces forming telephone associations across the nation. This allows them to represent their industry in standards committees, and voice their concerns to the government as a collective body rather than a long voice. These associations also pool their resources and build their own networks using monies from the member companies to pay for the cost of the network. This is how many small independent companies are getting involved in SS7.

		Who Are the Players?

		A number of companies offer telephone services of some type today. To fully understand their business, we need to understand their roles in the network. Let's define what these companies do so we can understand the services they provide. Some of these companies by the way offer their services to other telephone companies and do not have any subscribers connecting to their network.

		Regional Bell Operating Company (RBOC)

		These are the companies resulting from the divestiture of the Bell Operating Company (BOC). Only a couple of these companies are left now due to mergers and acquisitions. Southwestern Bell has acquired Southern New England Telephone (SNET), Ameritech, Pacific Bell, and continues to work on others. Bell Atlantic has acquired GTE and is working on acquiring others at the time this book was published. The RBOCs provide local telephone service, but as they open up their territories for competition (meeting the requirements of the Telecommunications Act of 1996), they are now also allowed to provide long distance service (in some areas).

		Local Exchange Carrier (LEC)

		This is where it gets fairly fuzzy. These are typically the independent telephone companies offering local service where the RBOCs did not provide service. An example of a LEC is General Telephone (GTE). There are many other LECs throughout the U.S. that are considerably smaller in size. They continue to operate today as small independent telephone companies, typically in the smaller rural areas of America.

		Incumbent Local Exchange Carrier (ILEC)

		The ILEC is a local telephone company, usually one of the independents, not affiliated with any of the RBOCs. They fall under the same guidelines as the RBOCs, and have been given territories like the RBOCs.

		Competitive Local Exchange Carrier (CLEC)

		The CLEC is a new entrant into a service area. These are usually small startups offering local telephone service in a few LATAs. They typically start as resellers of long distance service, and then begin investing in their own telephone network as they build their customer base. They target specific LATAs to offer service in and then expand out. Hundreds of CLECs exist in the U.S. today, and many more are starting. With the new packet switching networks rolling out, the number of CLECs is increasing sharply, mostly due to the decrease in network deployment costs associated with packet switching.

		Interexchange Carrier (IXC)

		These are the long distance carriers providing long distance service between LATAs. They started out as long distance service providers in specific areas, but now many of them have expanded their markets by offering local telephone service. AT&T, Frontier, and Sprint are all examples of IXCs.

		Data Competitive Local Exchange Carrier (CLEC)

		You don't hear much about data CLECs, but this is big business. These companies take the Internet and data traffic from the local telephone companies and route it through their own packet switched networks. The RBOCs are big customers of data CLECs because they do not want to manage this data traffic in their own networks. Internet Service Providers (ISPs) are also big users of these networks because it saves them from having to build huge nationwide networks to provide service in every city. They can use a data CLEC in every city, route the traffic through these networks to one national backbone network, and save the equipment and facility costs associated with nationwide network buildouts.

		Hub Providers

		Another emerging business is the hub provider business. These are carriers who provide facilities and interconnection to other telephone companies. For small telephone companies entering into a new market, this is usually the most economical way to get started because there is no capital investment for them to make. They simply lease the services they need from the hub provider.

		A hub provider provides switching and circuits to other networks for an access fee. They will also provide database services such as calling name (CNAM) and 800 applications. They collect fees from the networks they connect to based on the amount of traffic they send into other networks, as well as the amount of traffic they terminate in their own networks. Several hub providers provide SS7 services in the U.S. today, but the concept is still new in Europe and the rest of the world.

		Many of the RBOCs have entered into the hub provider business themselves, recognizing this as a new source of revenue. The market has become very competitive for hub providers, forcing a reduction in access fees and other fees collected.

		Hierarchy of the Synchronization Network

		All digital networks rely on timing mechanisms to maintain integrity of the data transmission. Because all digital transmission is multiplexed and based on time division, accurate timing is critical. This is especially true when DSOA links are used in SS7. DSOA links must have accurate timing sources in order for them to synchronize and carry signaling traffic.

		Digital facilities must have reliable, accurate clock sources to determine proper bit timing. These clocks must be synchronized with the same source, and are deployed throughout the telephone network. To maintain timing in the telephone network, a separate synchronization network has been defined (see Figure 1.3).

		The source for a clock signal is referred to as the Primary Reference Source (PRS). These clock sources reside in the various regions of the telephone network. They are highly accurate clocks, usually cesium beam or rubidium-based clocks. These clocks must be resynchronized and continuously verified using a universal time source. Loran-C and the Global Positioning System (GPS) are currently used by many companies to check the accuracy of their clocks. The distribution of clock signals is implemented at different levels, referred to as strata. The highest stratum obtainable is stratum 1, which is the primary clock source (primary meaning it is referenced directly from LORAN or GPS sources).

		Figure 1.3 To maintain synchronized timing throughout the digital network, the Bell System Operating Companies (BOCs) use this timing network.

		Clock signals are distributed in a primary/secondary relationship to all other levels. This means that central office switching equipment at stratum 2 distributes its clock signal to equipment at stratum 3. Equipment that sits downstream of the clock signal will not receive as accurate a clock signal as those connected directly to the source (stratum 1).

		The PRS distributes clocking signals to toll offices. The toll office is considered stratum 2, and must redistribute the clock to end offices within its LATA. Each LATA must have at least one stratum 2 clock. Whenever a clock is redistributed, it loses some accuracy. Yet the clock signal is accurate enough to operate throughout the network reliably, despite the loss of accuracy. End offices are considered stratum 3.

		The end office will distribute clock signals to other users of digital transmission facilities, such as private branch exchanges (PBXs) and channel banks. These are considered to be at stratum 4. In some cases, devices at this level can redistribute the clock signal to other adjunct equipment. These devices are considered to be at the lowest level of the hierarchy, stratum 5.

		Within a central office, clocks are distributed through a building integrated timing system (BITS). BITS is a distribution system for clock signals, and is distributed throughout the office to switching equipment in that office. BITS is critical to the proper operation of DS0A links in the SS7 network. Failure of this clock signal will result in the failure of the signaling links.

		Digital Signaling Hierarchy

		The telephone network also has a digital hierarchy for all digital transmission facilities. This digital hierarchy is a means for expressing the capacity of these various facilities. Typically, the highest level of the hierarchy is an aggregate of the levels below it. Thus, the lowest digital signal in the hierarchy is multiplied to establish the next level, which is multiplied to obtain the next level, and so forth. Table 1.1 illustrates the correlations between levels in the hierarchy.

		The SS7 network uses DS0As for signaling links. This is a 56- or 64-kbps data link, capable of sending voice or data. The DS0A in the U.S. is always 56 kbps, because the telephone company uses 8 kbps for control information. This control information is used by the transmission equipment to maintain the integrity of the data link. Some studies have been under way regarding the use of 64 kbps, but the multiplexers used throughout the telephone network today do not support 64 kbps links.

		In order for equipment to use the DS0A, special digital interfaces that are capable of sending and receiving signals at this level must be installed. If a DS1 is used, a device called a channel bank must be used as the interface. The channel bank divides the 24 time slots of the DS1 into 24 separate DS0As, which can then be distributed to their proper destinations. Other types of multiplexers/demultiplexers exist that perform the same task, depending on the location in the network.

		Currently, the DS0A level is the most commonly used in the SS7 network. New high-speed networks are driving the need for high speed facilities to be adapted for use in the signaling network. Asynchronous Transfer Mode (ATM) and Broadband ISDN (BISDN) require signaling speeds beyond that of DS0A. As ATM is deployed in the public networks, signaling traffic is migrating to these new facilities. The current digital facilities are being replaced and the signaling network is becoming integrated with the new broadband network. The first step is to support DS1 speeds over an ATM interface. This is currently deployed in some larger networks and is gaining acceptance in some smaller networks as well. Originally, it was thought that ATM would be the technology of choice for telephone companies migrating to broadband networks. However, TCP/IP has most recently entered into the picture as a more efficient and cost effective technology for signaling, and in some cases, voice and data. Of course, TCP/IP can be packetized and sent over ATM backbone networks, which seems to be the choice for many network operators today.

TABLE 1.1 North American Hierarchy
Digital signal destination	Bandwidth	Channels (DSOs)	Carrier designation
				DS0	64 kbps	1 channel	None
DS1	1.544 Mbps	24 channels	T-1
DS1C	3.152 Mbps	48 channels	T-1c
DS2	6.312 Mbps	96 channels	T-2
DS3	44.736 Mbps	672 channels	T-3
DS4	274.176 Mbps	4032 channels	T-4

 

		As seen in Table 1.1, the DS1 facility provides a total bandwidth of 1.544 Mbps. This should be more than adequate for almost any signaling requirements. New TCP/IP facilities provide 100 Mbps of bandwidth and are being deployed around the world as the next generation network choice. The existing network will require new hardware as well as software upgrades to support these new technologies. The SS7 protocol will undergo many changes as well to support these new high-speed data links.

		The digital hierarchy is quickly being replaced with a newer and faster technology. Fiber optics is quickly replacing copper facilities throughout the telephone network. Fiber optics has the capability to transmit at much higher data rates than copper, and is critical to the success of technologies such as broadband and ATM.

		Synchronous Optical NETwork (SONET) is currently found in telephone company networks worldwide and has become the transmission medium of choice. SONET provides data rates up to 2.4 Gbps, and will support broadband ISDN, TCP/IP, and ATM. SONET is also being used to link LANs through the Public Switched Telephone Network (PSTN).

		As seen in Table 1.2, SONET is also divided into different levels of service, each level being an aggregate of the levels below it. There are two designations used for these levels: the electrical signal itself and the optical signal. They are terms used for different reasons. Electrical signals are directly related to the optical signals and, therefore, can almost be used synonymously in most discussions. In this book, we will always refer to the optical signal.

		When compared to the digital signal hierarchy, there is a stark difference (see Table 1.3). Even at the lowest level of the optical hierarchy, 28 DS1s can be supported on one facility. This represents a significant cost savings to telephone companies. At OC-1, 672 time slots are supported for voice, data, or even signaling.

 

TABLE 1.2 SONET Digital Hierarchy
Electrical signal	Optical signal	Data rate (Mbps)	ITU designation
STS-1	OC-1	51.84
STS-3	OC-3	155.52	STM-1
STS-9	OC-9	466.56	STM-3
STS-12	OC-12	622.08	STM-4
STS-18	OC-18	933.12	STM-6
STS-24	OC-24	1244.16	STM-8
STS-36	OC-36	1866.24	STM-12
STS-48	OC-48	2488.32	STM-16

TABLE 1.3 Optical and Digital Compared
Digital signal		Optical signal
DS0	(64 Kbps)	OC-1	(51.84 Mbps)
DS1	(1.544 Mbps)	OC-3	(155.52 Mbps)
DS1C	(3.152 Mbps)	OC-9	(466.56 Mbps)
DS2	(6.312 Mbps)	OC-12	(622.08 Mbps)
DS3	(44.736 Mbps)	OC-18	(933.12 Mbps)
DS4	(274.176 Mbps)	OC-24	(1244.16 Mbps)
		OC-36	(1866.24 Mbps)
		OC-48	(2488.32 Mbps)

		A single SONET facility is not dedicated entirely to one application. One channel may be used for signaling, while the remainder carry voice, data, and video. This practice allows telephone companies to use existing transmission facilities between offices, rather than deploying a special link just for SS7.

		Current Trends in Telecommunications Technology

		Today's telecommunications industry has changed dramatically. Data communications and voice networking have been merged to provide a variety of services that leave even the most educated somewhat confused and baffled. These services all revolve around the backbone of the new Intelligent Network (IN), SS7.

		Because of SS7, these new technologies can be supported in the PSTN rather than having to have a separate network for each type of service (as previously done to support packet switching in the '70s and '80s). In fact, all data and voice communications will be simplified to the point that the subscriber does nothing but dial a number and get connected. The signaling network will handle the rest.

		The goal of the telephone network is to provide seamless service to all subscribers, regardless of the information being sent through the network. As previously discussed, the IN will provide this capability. But before the IN is fully deployed, there are many different pieces that must first be put into place.

		The telephone network of today will not support the types of services that subscribers are asking for. If there is a need for high-speed data, a special circuit must be installed from the customer premises to the other end of the circuit. If video is to be transmitted through the telephone network, special high-capacity circuits must be installed from the studio through the telephone network to another high-capacity circuit at the transmitter.

		The ultimate goal is to provide one network capable of transferring all kinds of information regardless of the bandwidth necessary and sending it through the network just as if placing a telephone call. To support this level of service, the network must be changed.

		One such change is referred to as network convergence. Again, TCP/IP has quickly become the technology of choice for these networks. There are several reasons for this development. First, TCP/IP is widely available. Virtually every major corporation using e-mail of some type is connected to a TCP/IP network, usually their own.

		Second, TCP/IP is a proven technology. Originally developed for use by the Defense Department for their data networks, TCP/IP has quickly migrated into the private sector and gained widespread popularity because of the Internet. However, TCP/IP is not well suited for voice transmission and must undergo changes to ensure voice transmission without noticeable delays in the transmission.

		This change is underway today through many of the standards organizations. In the meantime, many telephone companies have relied on ATM to provide quality of service for voice transmission. A number of new companies are emerging who use TCP/IP as their primary network for all traffic. These companies are driving a new and fast-growing market.

		As a result, a number of new equipment vendors have found themselves in the telephony business. They are building a new generation of switches that are based on computer platforms rather than the conventional switching platforms used in older legacy networks. This new breed of switches enables small companies to enter the telephone business and deploy their own networks quickly and a lot cheaper than conventional telephone companies.

		The IN is also evolving to support TCP/IP. The same vendors who developed the legacy computer systems used in the IN are quickly adding TCP/IP interfaces to support access to their products through both the Internet and intranets. SS7 itself is being adapted to meet the demands of this fast growing-market.

		Technology is changing the network and IN plays a new role. To understand the role SS7 plays in each of these technologies, the rest of this chapter will provide some overview of current services being offered by telephone companies and independents. Some of these services are still under development and may not be offered for general availability for some years to come.

		Introduction to the Intelligent Network

		The 1990s will become known as the years of the Clinton administration's information highway. The fact is that the information highway was already under way before this time. The IN has been under development for many years, with the goal of allowing all types of information to pass through the telephone network without special circuits or long installation cycles.

		The concept of being able to access all kinds of information is not new. The network to support this type of service is new. The IN provides the backbone to support and define these services.

		As the need for new features and services becomes more important to customers, the need to deliver those services and features in an economical way becomes equally important. The problem facing telephone companies today is being able to provide these features and services quickly and efficiently. Ordering an 800 line for two weeks' usage is now easy and can be implemented within hours instead of days.

		The IN makes it easier, because now, when subscribers order new services, technicians do not have to be dispatched to add programming to the switching equipment and to cross-connect the circuits.

		In the IN, everything is controlled or configured by workstations with user-friendly software interfaces. Telephone service representatives can create new services and tailor a subscriber's service from the terminal while talking with the customer. The changes are immediately implemented in the switches. Circuits are cross-connected using digital cross-connect systems, which are also controlled by the workstation. Customers today can order high-speed communications, video, audio, and digital voice facilities on an as-needed basis.

		Networks were not always equipped to handle such demand. But switch manufacturers have added new features to switching equipment that allow services to be added to subscriber lines by simple commands at a terminal. Some new products even allow customers to order services and features by dialing a sequence of codes on their telephones. Soon, ordering an 800 line will be as simple and as fast as ordering a pizza.

		Welcome to the age of the Intelligent Network (IN). The IN is just what its name implies: intelligent. Services and features can be changed or deployed using simple procedures through a terminal, rather than through expensive programming changes made by certified technicians. All the customer needs is the facility (trunks) to utilize the new services. With TCP/IP, this is no longer a concern. The customer needs only enough bandwidth to handle the traffic.

		Imagine a small business with fewer than 100 employees. By building a TCP/IP network and using computers equipped for Internet telephony, this company can interconnect all of its employees over the company's computer network. Now imagine if that same company were to extend its computer network out to the local telephone company. Suddenly, everyone connected to the local telephone company could access this small business through the same computer network.

		If the local telephone company were to extend this same network to other telephone companies, the small business would suddenly find itself connected to a huge computer network, with access to thousands of computers equipped to send not only data but voice as well. This is an over-simplified explanation of computer telephony, but the basic concept has changed the telephone industry overnight.

		Still, computer networks do not provide the telephone services we depend on daily. This is where the IN comes into play. The IN can be deployed into this same computer network and used to deliver services and features to computer telephony devices.

		As more and more customers line up to deploy ATM and TCP/IP, the IN will become as commonplace as Touch ToneTM dials. Yet, little is understood about the IN. To understand what the IN is about, let us first examine the architecture of such a network. The IN consists of a series of intelligent nodes, each capable of processing at various levels, and each capable of communicating with one another over data links.

		The IN relies on the SS7 network, which forms its backbone. SS7 provides the basic infrastructure needed for the Service Switching Point (SSP), which provides the local access as well as an ISDN interface, for the Signaling Transfer Point (STP), which provides packet switching of message-based signaling protocols for use in the IN, and for the Service Control Point (SCP), which provides access to the IN database. The SCP is connected to a Service Management System (SMS), which provides a human interface to the database, as well as the capability to update the database when needed. The SMS uses a command-line interface or a GUI interface and a man-to-machine language to build services and manage the network. The SMS can also be used in some applications as a central control point for updating multiple databases and controlling the updates to those databases from a central authority.

		One additional node used in the IN that is not seen in the SS7 architecture is the Intelligent Peripheral (IP). The IP provides resource management of devices such as voice response units, voice announcers, and DTMF sensors for caller-activated services. The IP is accessed by the Service Control Point (SCP) when services demand its interaction. IPs provide the IN with the functionality to allow customers to define their network needs themselves, without the use of telephone company personnel.

		When a call is placed in the IN, a request for call-handling instructions is sent to the SCP using the Transaction Capabilities Application Part (TCAP) protocol. The database provides the instructions for handling the call based upon the customized service instructions the subscriber has programmed, and sends them to the end office switch. The end office switch then communicates to the IP using the ISDN protocol to attain the use of resources such as recordings and other devices. The call setup and teardown is handled using conventional SS7 protocols.

		Advanced Intelligent Networks (AINs) in Figure 1.4 provide many components not found in the earlier versions of INs. One of the key components is the Service Creation Environment (SCE). In the AIN standard, SCE defines the look and feel of the software used to program end office switches to provide a new service. This look and feel defined in the AIN standard provides a graphical user interface (GUI), which uses icons, for building customized services. AIN administrators can then tailor services to meet the customers' specific needs by clicking on network capability icons rather than programming via commands on a command line.

		Eventually, this terminal and the SCE will be extended to the customer premises, allowing large and small companies with special network needs to tailor their services on an as-needed basis, without telephone company assistance. This concept has already intrigued many large companies with large volumes of inbound and outbound calls, especially those in the telemarketing industry whose network needs vary from week to week.

		The first offerings of an IN were in the early '80s when AT&T announced a centralized database for all 800 numbers. End offices wishing to handle 800 calls would have to access this database through the SS7 network. Control of the network was provided by operations systems (OSs), which used intelligent workstations to provide maintenance and administration for the network databases and other network nodes.

 

		Figure 1.4 The Advanced Intelligent Network (AIN) relies on SS7 for interconnection of network switches.

		In 1988, AT&T asked vendors to participate in defining the standards for the evolving IN. The end result was the release of an evolutionary path defined in a series of releases, now known as the Advanced Intelligent Network (AIN). The AIN defines how features will be invoked by the telephone company or the subscriber.

		The AIN deployed in today's network is known as AIN Release 0.2, and is presently deployed in several local exchanges. AIN Release 0.2 offers a variety of services to the consumer, such as redirecting the destination of calls on a per-line basis. This can be altered by dialing a code through a phone and entering in the destination telephone number. This feature would be of huge value to marketing firms with large inbound call volumes. Eventually, the feature will be deployed by the subscriber through a terminal located at the customer premises and connected to the telephone company's signaling network.

 

Other features currently available include call screening (Do Not Disturb), selective call acceptance, calling name delivery, and spoken caller identification. The many services and features offered to customers are based on databases linked to the SS7 network through nodes called Service Control Points (SCPs). Local end offices and other networks can access these databases by sending database query messages through the SS7 network to the SCP. The SCP replies to the query after accessing the information from the appropriate database and sending the requested information in an SS7 message format through the SS7 network to the requesting end office. Based on the information received, the end office (or service node) is then able to create the requested services. INs use the Transaction Capabilities Application Part (TCAP) protocol for sending database queries to the SCP.

		The SCP provides the call-handling instructions and service instructions to the end office so that it knows how to handle calls for a specific subscriber. The following features are just a sampling of the features of an IN. As the IN evolves, new services will become available. Features include

		Find Me Service

		Follow Me Service

		Computer Security Service

		Call Pickup Service

		Store Locator Service

		Call Routing Service

		Multilocation Extension Dialing

		Name Delivery

		Outgoing Call Restriction

		Find Me Service

		This service allows calls to be forwarded to another location. The difference between this feature and today's call forwarding feature is the ability to screen unwanted calls from forwarding. Only authorized callers are forwarded to the new location.

		Follow Me Service

		Similar to call forwarding, this allows a number to be forwarded on a time schedule. The subscriber determines the time forwarding is to take place when the feature is invoked. Destinations can include cellular telephones or Personal Communications Services (PCS) handsets.

		Computer Security Service

		This feature prevents unauthorized callers from accessing a computer via modem. Only callers with the authorized access code or calling from an authorized number can access the computer. The SS7 network delivers the calling party number to the destination end office. This number is then checked in a database located with a SCP, and, if authorized, is allowed to connect with the modem.

		Call Pickup Service

		When a call is placed to a number and is unanswered, the called party can be paged via radio pager. The called party can then dial a code from any telephone at any location and immediately be connected with the waiting caller. Some manufacturers of PCS devices have already developed two-way pagers that connect the caller with the party being paged. The pager is a two-way transceiver capable of receiving calls (pages) and connecting the caller with the paged party (similar to the voice pagers used a few years back, but this pager allows two-way conversation).

		Store Locator Service

		Businesses can advertise one number, and callers are automatically transferred to the nearest location based on their own telephone number. The telephone company provides the routing service based on the prefix of the calling party number. This allows businesses to advertise nationwide for all locations without special ads based on geography. The calling party number is matched in a routing database located at a SCP. The SCP provides the end office with the routing instructions based on the calling party number.

		Call Routing Service

		This allows businesses to reroute calls when congestion occurs or after business hours. It is an excellent feature for telemarketing and reservation centers with multiple locations. ACD switches can be interfaced to the end office by SS7 data links. This allows the ACD to send network management messages via SS7 protocols to a database, located by a SCP. Calls are then rerouted around the call center based on the routing instructions in the database.

		Multilocation Extension Dialing

		This allows the usage of abbreviated extension numbers to reach personnel regardless of their location and without PBX equipment. Very similar to PCS offerings, the subscribers receive personal numbers which can be used to reach them no matter where they go.

		Name Delivery

		As a call rings the telephone, the caller's name is displayed on a digital display. This is offered to residential and business customers alike. The digital display is built into a digital phone or is an adjunct to any standard telephone. This is somewhat different than the controversial Automatic Number Identification (ANI) feature, which displays the caller's telephone number. ANI allows telemarketing companies to store calling party numbers in their databases, which are later sold to other telemarketing companies. Name delivery delivers the name only, retrieved from a line subscriber database.

		Outgoing Call Restriction

		This feature allows the restriction of specific numbers or prefixes and area codes, allowing customers to restrict long distance calls and service numbers such as 900 and 976 numbers from being dialed on their phones. Presently, the telephone companies can offer restriction of 900 and 976 numbers, but they cannot provide restriction of specific area codes, prefixes, or individual telephone numbers. This feature allows those restrictions to be programmed by the subscriber.

		It is important to note that AIN does not define the features and services, but how those features and services will be deployed by the customer. The features and services are defined by the service providers themselves, and may or may not be consistent from one telephone company to another. They are limited, however, to the capabilities of the switching equipment used in their end offices.

		AIN is currently offered in two releases, Release 0.1 and Release 0.2, with new releases on the way. Currently, Release 0.2 has been deployed, with plans for future releases to follow.

		Many telephone companies are reluctant to deploy AIN because of talk of a newer network called Information Network Architecture (INA). This new technology is still in development, but many feel it will succeed AIN. Yet others view INA as a subset of AIN. At any rate, INA provides better utilities for managing the new broadband services being deployed by major telephone companies. It is very likely that there will be two architectures: AIN for the voice network and INA for managing the broadband network.

		The AIN is certainly not a new concept. As we have already seen, this concept evolved from an earlier introduction to make 800 services easier to customize and maintain. The ability to create new services quickly and efficiently was one of the early driving forces behind the IN. In fact, 800 services today can be customized by the customer according to their immediate needs with a simple phone call. AIN has not been deployed for local telephone service, however, and that is the undertaking today.

		Personal Communications Services (PCS) and Centrex also benefit from the IN. PCS will be based on the ability to tailor subscriber services in real time. This ability cannot be supported without the IN. Centrex popularity is on the rise. With the IN, customers can tailor their specific service requirements within hours instead of days. With features such as networked voicemail and Automatic Call Distribution (ACD), Centrex will become a powerful competitor to the PBX.

		Presently, Centrex is difficult to market against the PBX because of the limited feature set offered with Centrex. With PBX systems, proprietary phones can offer a multitude of high-tech features that greatly enhance the systems' capabilities. Centrex does not use electronic telephones, but standard analog telephones. While Centrex can offer simple features such as forwarding, conferencing, and speed dialing, it cannot match the features of a PBX.

		When Centrex can be offered with IN support, subscribers can take advantage of the telephone network to provide them with seamless end-to-end call-handling capability. Callers can be routed to noncongested calling centers dependent on traffic and/or time-of-day. Voice-mail can be linked so that users can reach their voicemail from any phone at any company location, without dialing into the system from an outside line. Even station detailed message recording (SMDR), the feature that provides the calling records of every extension in the system, can be bridged to include all extensions in the corporation, rather than just those within a specific office handled by one carrier. The IN will allow Centrex to provide many of the same features previously seen only in the PBX environment.

		The IN is under implementation even as this book is being completed. Yet, there is a lot of work to be defined before it can be called complete. It may take many years before the IN reaches its full potential, but the framework has been defined and the infrastructure is being laid.

		The Integrated Services Digital Network (ISDN)

		The Integrated Services Digital Network (ISDN) (see Figure 1.5) was first offered to the public in the 1980s, as the telephone service providers began deploying their SS7 networks. With ISDN, many new services could be extended to the customer premises. By using ISDN, subscribers can consolidate all their trunks to one DS1 facility. The ISDN protocols provide circuit allocation within the ISDN.

		When additional bandwidth is needed for high-speed data communications, the protocol is capable of allocating additional channels within the DS1 to carry the call. When the call is terminated, the channels are released and made available for other calls. This is often referred to as dynamic bandwidth allocation and is one of the principal features of ISDN. Besides transmitting voice, ISDN is also capable of transmitting data using the same facilities as the voice.

 

		Figure 1.5 ISDN and BISDN extend the services of the IN out to the subscriber premises. They both act as interfaces to the signaling network (SS7) without providing direct links.

		ISDN signaling uses a separate channel and is compatible with SS7. The signaling information is handed off to the SS7 network and transferred to the distant end using the SS7 ISDN User Part (ISUP) protocol. The ISUP protocol was developed for all call setup and teardown, and replaced the SS7 Telephone User Part (TUP) protocol in ANSI networks.

		The term ISDN was originally used to refer to the entire IN, including SS7. This later evolved to reference only the subscriber interface. Originally, the creators of SS7 thought of extending the SS7 network all the way to the subscriber. This was abandoned, however, over concern for security and network fraud. The solution was to create an intelligent interface, compatible with SS7, which could offer the same services and intelligence as the SS7 network. It was this that spurred the creation of the ISDN protocol.

		Perhaps the most important application for ISDN is the concept of connecting PBXs within a private network. When SS6 was first deployed, there was the thought that the network signaling could be extended to the local PBX. This would allow the PBX to send its signaling information directly to a central office switch using the same message packet-switching protocol used by SS7 today.

		This concept was quickly dropped, however, due to security issues. Instead, a separate access protocol was developed. With a specialized access protocol, signaling could be extended through the Public Switched Telephone Network (PSTN) to distant PBXs, without sacrificing security of the PSTN. ISDN was created as the access protocol to deliver PBX signaling through the network to distant PBXs, allowing large companies with multiple PBXs to bridge their switches together transparently. The United Kingdom already uses the Digital Private Network Signaling System (DPNSS), an ISDN protocol designed to extend PBX signaling through the SS7 network to distant PBXs.

		ISDN offers many services to the subscriber. The basic levels of service are defined as:

		Transport elements. Allow information to be transported through the telephone service provider's network and its switches, routers, multiplexers, and other network equipment transparently, without alteration to the original data

		Control elements. Support real-time operations of transport capabilities (connection establishment and database queries)

		Network management elements. Provide procedures and capabilities to administer, maintain, and operate the communications infrastructure. Includes provisioning of transmission facilities, fault management, congestion control, and administration of databases and routing tables.

		Communications applications environment. Provides a development environment for programmers from which applications can be developed, using the other three elements

		Transport. Provides the lower three layers of OSI, providing allocation of bandwidth, routing, relaying, and error detection/correction

		To understand how ISDN can be of significant benefit to PBX networks, consider this example. Many large corporations own several PBXs at different locations. Tie lines are often used to tie the PBXs together. This allows users to access extensions in any other company location by dialing an access number (to access the tie line) and dialing the extension. Many digital PBXs allow callers to dial extension numbers of remote extensions without dialing an access code. Automatic routing features provide software to determine which trunk the call must be routed to.

		Another advantage of tying PBXs together is that long distance calls can be routed over tie lines to a PBX in the calling area of the dialed number. The call is then routed to a trunk in the remote PBX, as a local call. This can save corporations thousands of dollars in long distance charges. The problem is that the remote PBX does not know what class of service the calling phone has been assigned. The class of service is a software feature that determines what numbers a phone is allowed to dial. This programming takes place in the PBX terminating the telephone.

		ISDN allows this information to be passed along to the remote PBX. In addition, other information regarding the privileges of a telephone and even features can be passed from one PBX to another. This allows corporations to create their own proprietary network, without expensive facilities between PBX locations.

		ISDN never got a good start and has not shared the popularity of other service offerings, mostly because it was somewhat premature in its release. Many telephone service providers tried to market this service to residence subscribers, but failed because of the prohibitive cost and lack of intrinsic value. To truly take advantage of ISDN capabilities, the signaling information must be able to travel from the originating end to the distant end. Without SS7, this is not possible. When ISDN was first introduced, SS7 was not yet fully deployed, leaving ''islands" of ISDN service that could not be connected with other ISDN networks.

		Now with SS7 deployed in all the Bell Operating Companies' (BOCs) networks as well as those of many of the independent telephone companies, ISDN can finally be used to its full potential. Unfortunately, the telephone companies still do not understand how to market ISDN and often cite the many features ISDN provides, rather than real applications.

		One feature often used in marketing ISDN is automatic number identification (ANI). For a short period there was a lot of interest in ANI, as telemarketing companies scrambled for databases that could provide names and addresses of callers triggered by the calling party information supplied by the ISDN network. All interest in ANI was quickly lost as the states began legislation blocking ANI, or at least forcing telephone companies to provide ANI blocking at no charge. Without the guarantee that at least a majority of the calls would provide ANI information, and without any other substantial advantages in the way of features, ISDN quickly faded into the background.

 

The problem was that most areas could not provide the calling party information. This was due to the fact that many central offices had not yet been upgraded to digital switches and were not capable of passing this information through the SS7 network. By the time networks caught up with ISDN and the SS7 network bridged all the "islands," digital switch vendors began offering features in their central office switches that would support not only the delivery of the calling parties telephone number, but their name as well, without having an ISDN interface. Calling Name (CNAM) continues to be a popular feature today, possibly because of the SS7 network.

		Another plague of ISDN has been the failure of manufacturers to follow a standard. Unfortunately, many manufacturers created ISDN terminals and devices that were proprietary and did not share the concept of open interconnection, making interoperability difficult, if not impossible.

		This was later addressed by the North American ISDN User's Forum (NIUF) in June 1988. The NIUF comprises industry vendors and service providers. This forum agreed on a set of features and how they would be deployed in the ISDN network and ratified a new ISDN standard, now known as National ISDN #1. This new standard is an agreement among ISDN vendors here in the U.S. to standardize the equipment interfaces, interoperability, and features, guaranteeing ISDN's success in North America.

		Still, ISDN continues to be marketed for its features rather than for true applications. What ISDN really offers to any business with a PBX is the ability to consolidate its trunking requirements to one or more digital spans, or T-1s. Using the ISDN protocols, they can take advantage of end-to-end digital communications for both voice and data. This allows owners of PBX equipment to rid themselves of costly dedicated lines for data communications, facsimile, and video conferencing. ISDN can provide all these services and more, with a common facility, eliminating the need for special circuits. In addition, these same facilities can be assigned dynamically, on an as-needed basis.

		ISDN may also become popular as an access to the new broadband network. Small and medium-sized companies wishing to connect their LANs can use narrowband ISDN to connect to the ATM network. This interface is being considered by many service providers as the midpriced interface to the broadband network.

		Larger corporations with higher bandwidth requirements will find broadband ISDN (BISDN) more attractive, with enough bandwidth to support Ethernet, Token Ring, and even Fiber Distributed Data Interface (FDDI). BISDN will provide bandwidth up to 155 Mbps and higher, and will most likely be the interface of choice to the new ATM network currently under development.

		It is this dynamic capability that makes ISDN attractive to businesses. But ISDN can be used in residential applications as well. This is a tougher market, since consumers look for obvious advantages and features to offset cost. This is the market ISDN has had a tough time breaking into, primarily because of the prohibitive cost for the average consumer. But as the cost of ISDN decreases and becomes comparable to Plain Old Telephone Service (POTS), ISDN will become as commonplace as the telephone itself. This is especially true in metropolitan areas where telecommuting has become a way of life for many living in congested areas. ISDN provides high-speed data communications and seamless connectivity to the office LAN. As the information highway evolves, ISDN should gain in popularity.

		ISDN is also an excellent medium for telemetering, something many utilities do not like to talk about because of its impact on the workforce. Telemetering has already been defined and is waiting on the digital capability of ISDN before it becomes a reality. With ISDN at the residence, utilities can attach interfaces to the ISDN circuit and use its channels during idle periods to send meter information to a collection center. This allows utilities to read meters in real time, rather than waiting until the end of a billing cycle. Telemetering also allows them to monitor usage in geographic areas during peak periods, providing invaluable information for studies and preparation for heavy loads.

		There are two classes of ISDN service: Basic Rate (BRI) and Primary Rate (PRI). BRI service provides two 64-kbps bearer channels (B channels) and one 16-kbps signaling channel (D channel). This service is designed for residential and small business usage. PRI offers 23 64-kbps B channels and one 64-kbps D channel. PRI is designed for larger businesses with large call volumes. Many PBX manufacturers already provide ISDN-compatible trunking interfaces for their equipment, making ISDN a good choice for companies who need end-to-end voice and data communications.

		ISDN cannot be successful by itself. As we have already seen, without SS7, ISDN remains a local digital service providing a limited number of features and applications. With the addition of SS7, ISDN can become an extension of the telephone network to the customer premises, offering true end-to-end voice and data communications with no boundaries.

		The ISDN standards can be found in the ITU-TS "I" series. The signaling standards are defined in publication Q.921 (defines the Link Access Procedure D Channel) and the Q.931 publication (defines the ISDN call control procedures at layer 3).

		The Cellular Network and SS7

		Cellular networks have evolved into two different networks. The European cellular network uses Groupe Special Mobile (GSM). The GSM network is described next and in Figure 1.6. In the U.S., cellular networks are still primarily analog (although cellular companies are quickly deploying all-digital technology to replace the analog network), but the addition of IS-41 for signaling has increased the efficiency of these networks. Both networks are very similar, the difference being in the procedures used for call handling and the entities which exist in these networks.

		Previous to the deployment of SS7, cellular networks in North America used X.25 as the transport for signaling information. The signaling information in itself is somewhat new and is made possible by an application called IS-41. The IS-41 application is not particular as to what transport it uses, so X.25 was sufficient for carrying this information within one network.

		Figure 1.6 The GSM network is the standard for cellular networks outside of the U.S. Many U.S. cellular providers are now looking at GSM for new PCS networks.

 

Many North American cellular networks have already deployed SS7, replacing their X.25 networks. Cellular networks use the Transaction Capabilities Application Part (TCAP), which supports the access of remote databases. The ISDN User Part (ISUP) is now being implemented in cellular networks for the connection to the wireline network.

		European cellular networks have always relied on SS7 for their signaling requirements and have enjoyed a much more robust and feature-rich network because of SS7. GSM is deployed in several networks in North America as well, but has not shared the same popularity as in Europe. It is possible now to purchase a GSM cellular telephone in the U.S. Provided you have a service agreement that supports usage in Europe, you can use the same telephone when you travel.

		The GSM Network

		The GSM cellular network comprises two different segments: the radio segment and the switching segment. The radio segment consists of the cellular telephone itself, or transceiver, and the antenna system used to receive and aggregate signals within a geographical location. This antenna is referred to as the cell site.

		Cell sites use different types of antennas, depending on the coverage required. Omnidirectional antennas provide coverage in a circular pattern, radiating in all directions from the cell site. Directional antennas can be used to cover a specific sector, and normally cover an area of 120 degrees.

		All cell sites consist of the antenna and the radio transceiver, as well as interface equipment. This equipment is referred to as the base transceiver station (BTS) and the base station controller (BSC). The combination of both the BTS and the BSC is called the base station subsystem (BSS).

		The purpose of the base transceiver station is to communicate with the cellular telephone. This is actually the ratio transceiver, capable of transmitting and receiving within the 900-Mhz frequency band. When a caller is connected with a cell site, the BTS measures the strength of a supervisory audio tone (SAT), which is sent at a much higher frequency than the actual voice transmission. The SAT is sent at regular intervals by cellular telephones and is received by multiple cell sites. Each cell site reports the strength of the signal it receives to the Mobile Switching Center (MSC), which then determines which cell site will take possession of the call. If a call is in progress and the cellular phone moves into a new cell site coverage area, the call is handed off by signaling messages through the MSC to the new cell site.

 

The hand-off is controlled by the Mobile Switching Center (MSC), which communicates with all cell sites within its geographic area. The MSC does not communicate directly with the BTS but communicates with the BSC, which serves as an interface between the radio segment and the switching segment. The BSC uses digital facilities to communicate with the MSC.

		The interface between the MSC and the BSC is called the A-bis interface. The A-bis interface is a 64-kbps digital link, and uses three protocols to transport signaling information to the MSC:

		Link Access Procedure on the D Channel (LAPD)

		Base Transceiver Station Management (BTSM)

		A-bis Operations and Maintenance (ABOM)

		Direct Transfer Application Part (DTAP)

		The LAPD protocol is used as the layer 2 transport protocol and provides the node-to-node communications necessary to send packets through the network. The Base Transceiver Station Management (BTSM) protocol is used for managing the radio equipment resident at the base station, as well as the interface between the base station and the MSC. Data and other signaling information are sent from the base station equipment via the Direct Transfer Application Part (DTAP).

		In addition to the Public Switched Telephone Network (PSTN), the MSC must also interface to other entities within the cellular network. These entities include

		Home Location Register (HLR)

		Visitor Location Register (VLR)

		Operation and Maintenance Center (OMC)

		The HLR is a database used to store the subscriber information for all subscribers within the home service area of the service provider. In European GSM networks, the HLR is linked to other service areas so that subscriber information may be shared between networks. Networks in North America can now offer the same seamless roaming through interconnection agreements with other cellular providers. It has only been the last four years or so that this has been possible in North America.

		The VLR is used to store information about visiting subscribers who are not in their home service area. This is where the roaming number information gets stored, so that subscribers may use their cellular phones while in another city. This is also linked to other networks, so that this information may be shared. While subscribers are in this network, the information regarding their service will remain in the VLR. This information is retrieved from the home HLR by the network.

		SS7 protocols are used throughout the cellular network to provide the signaling information required to establish circuit connections and disconnect circuit connections, as well as share database information from one entity to another. In addition to the Message Transfer Part (MTP) and the Signaling Connection Control Part (SCCP), the following protocols are used from the MSC to other entities within the network:

		Mobile Application Part (MAP)

		Base Station Subsystem Mobile Application Part (BSSMAP)

		Direct Transfer Application Part (DTAP)

		Transaction Capabilities Application Part (TCAP)

		Cellular Entities

		The HLR is a database used to store information regarding the users of the cellular network. When a cellular telephone is purchased, the phone must be activated before it can be used in the network. The purpose of the activation is to program its serial number into the HLR database.

		Whenever the cellular telephone is activated (powered on), the serial number is transmitted to the closest cell site. This information is used to identify the location of a cellular station, so that incoming calls can be routed to the appropriate cell site for reception by the destination cellular phone. The cellular phone continues to transmit its identification at regular intervals, so the network always knows the whereabouts of all active cellular phones. This is in addition to the SAT, used by the MSC to measure the signal strength of cellular telephones as they move from one cell area to another.

		The VLR is a database used to store information regarding cellular telephones being used in the coverage area that are not normally registered to operate in this home area.

		With the advent of IS-41 (a protocol used by the MSC to connect to other cellular providers), roaming can now be seamless. No advance provisions are necessary the user just carries the cellular telephone from one area to the next. This technology is widely deployed in many cellular networks today. IS-41 provides the signaling protocols necessary for cellular providers to share database information.

 

The Operation and Maintenance Center is used to access the Equipment Identification Register (EIR) and the Authentication Center (AC). The EIR stores the identification serial number of all cellular telephones activated within the coverage area. The AC stores a security key embedded into all cellular phones. This code is transmitted along with the serial number when the phone is activated, and prevents unauthorized phones from being used in the network.

		So far, we have discussed only the cellular network. This network must connect to the Public Switched Telephone Network (PSTN) at some point. The signaling information used to request service and connect calls within the public network is sent through the SS7 network. The Mobile Switching Center (MSC) connects to the SS7 network via a Signaling Transfer Point (STP) and the ISUP, TCAP, and MTP protocols. The SS7 network is instrumental in connecting all the cellular providers together and allowing their various databases to be shared with one another.

		The IS-41 Network

		In the U.S., moving from one calling area to another no longer requires prior negotiations with the service providers. When the calling area belongs to another service provider, access to subscriber records is accessed using IS-41, and the subscriber is able to use their cellular phone as if they were in their own home network. To support seamless roaming, cellular providers have negotiated interconnect agreements, providing access to each others' databases.

		To provide for seamless roaming between calling areas, the EIA/TIA developed the IS-41 protocol. IS-41 is really an application entity, which relies on the Transaction Capabilities Application Part (TCAP) and the Signaling Connection Control Part (SCCP) protocols to travel through the network. Both of these protocols are particular to SS7 networks.

		Before SS7 was developed in the cellular network, X.25 has provided the carrier services for data messages traveling from one HLR to another, or from a VLR to an HLR. With SS7 services and the new application entity, registration of a cellular subscriber in a new area is automatic and transparent to the user. The IS-41 provides the messages and transactions necessary to register and cancel registration in various database, while TCAP and SCCP provide the routing and transport of these messages.

		IS-41 was developed by the Electronic Industries Association (EIA) and endorsed by the Cellular Telecommunication Industry Association (CTIA). The standard is divided into a series of recommendations presented by the Telecommunication Industry Association (TIA) 45 subcommittee (TR45). The same subcommittee is responsible for Personal Communications Services (PCS) standards.

		The principal differences between IS-41 and GSM lie in the protocols used to communicate between the various entities and the frequencies of the telephone units themselves. The network topology is virtually the same (see Figure 1.7).

		IS-41 aligns with the ANSI version of SS7, using the TCAP protocol from the SS7 protocol stack to communicate with databases and other network entities. These are the same databases found in the GSM network, the HLR, VLR, and EIR. The ISUP and MTP are also used in the IS-41 network to connect cellular calls to the public switched telephone network, and to connect cellular circuits from the MSC to the base stations.

		The IS-41 network uses and MSC, as does the GSM network, for connecting to other networks and databases. Little has changed functionally within the MSC other than the fact that the data stored in the HLR can now be shared with other MSCs across the SS7 network. This requires a transport protocol (TCAP) to move the data through the SS7 network. The HLR and VLR can be colocated with the MSC.

		The biggest advantage to IS-41 is the passing of cellular telephone information needed when cellular subscribers wander from one service provider's calling area to another calling area. Previously, users had to call ahead and arrange for a special roaming number. Callers then had to call the roaming number when the user was in a different area. With IS-41, the information stored in each service provider's network is shared by passing signaling information between cellular networks.

		Figure 1.7 A typical cellular network in the United States

 

		In essence, the cellular networks were much like ISDN was several years ago. Every service provider could provide service within its own network, but was not connected with other service providers. This created a bunch of individual "islands" of service. IS-41 is the bridge that allows service providers to share database information with other networks and eliminate the need for setting up roaming in advance. Subscribers can now move from calling area to calling area without worrying about coverage.

		The PCS Network

		Personal Communications Services (PCS) is a new type of wireless communication based on the same philosophy as cellular, but with significant differences (see Figure 1.8). The most apparent difference is in the distance between the base station and the handset.

		PCS is really a combination of the IN and cellular networks, with a different topology. The PCS network relies heavily on the IN for delivery of custom features on a demand basis. The PCS network offers many new features not currently available in the cellular network. These features include

Available mode

		Screen

		Private

		Unavailable

 

 

 

		Figure 1.8 The PCS network being proposed by many cellular providers today.

		Available Mode

		This allows all calls through except for a minimal number of telephone numbers, which can be blocked from reaching the PCS subscriber. Relies on delivery of the calling party number, which is checked against a database. The subscriber updates the database through the dialpad of a PCS handset.

		Screen

		The name of the caller appears on the display of the PCS handset, allowing PCS subscribers to screen their own calls. Unanswered calls are forwarded to voice-mail or another number. The forwarding destination is determined by the subscriber.

		Private

		All calls can be forwarded, except for a limited list of numbers that are allowed to ring through. The list is maintained by the subscriber and can be changed at any time through a special set of codes, which are input by the subscriber through the dialpad of the handset.

		Unavailable

		No calls are allowed to get through to the subscriber. In the cellular network, a cell site is deployed to cover a wide area (typically four to eight miles). This distance does not permit calls to be of the same quality as terrestrial lines. In PCS networks, the area of coverage is much smaller (only a radius of about a quarter of a mile). This allows calls to be of much higher quality than in the cellular network. In addition, PCS handsets use digital technology, which is quieter than analog.

		The disadvantage to this type of deployment is in the cost of the network. To cover such a small area in a metropolitan area requires many more antennas and transceivers than in the cellular network. The positioning of these antennas also is critical. Large towers, such as those currently used for cellular, are not acceptable in most neighborhoods, which is where many of these PCS antennas have to be located.

		The base station communicates to other networks (whether cellular or the PSTN) through the PCS Switching Center (PSC). The PSC connects directly to the SS7 network with a link to a STP. PCS networks rely heavily on the SS7 network for internetworking, as well as for database access.

		The most important feature will be the database access. The problem in cellular networks is that the database access takes place through the mobile switching center, and the databases are a noncompatible mixture of mainframes and minicomputers deployed in proprietary networks. In the case of PCS, the database is located in the SS7 network itself. The SS7 Service Control Point acts as the interface point to these databases. These databases also store data for the PSTN, such as calling card information and subscriber data, and also are capable of providing information to all networks regarding PCS telephones.

		This is an improvement over cellular, in which each network has its own database. In theory, the database in PCS will be centrally located, so all service providers can share it equally, but, in practice, each service provider will deploy its own database services and charge other service providers an access fee for sending queries to their database.

		The purpose of having quick access to the telephone network databases is to allow subscribers of PCS services to define their own call-handling instructions and tailor their services to meet their immediate needs. For example, a subscriber may want to be left undisturbed during an important meeting, yet is expecting an important phone call from the president of a new client company. The subscriber wants that call to get through during the meeting. The telephone number of the person calling could be entered into the database with a service that would route all calls to a recording and voice-mail, except that from the president of the new client company, whose call would be routed to the PCS subscriber immediately.

		The PCS radio spectrum is much higher than the 900-MHz cellular network. In 1993, the Federal Communications Commission (FCC) allocated 160 MHz of bandwidth in the 1850- to 2200-MHz range. The FCC divided this range into licensed frequencies and unlicensed frequencies. The licensed frequencies have been further divided into seven separate bands.

		The service areas have been drawn according to Rand McNally's boundaries for Major Trading Areas (MTAs) and Basic Trading Areas (BTAs), although this is under much debate. There are 51 MTAs and 492 BTAs in the U.S. Much of the debate is related to the fairness of trading areas versus the Local Access Transport Areas (LATAs) created by the Justice Department during the divestiture of the Bell System. Many service providers argue that it would be unfair to use different boundaries for PCS than those issued for local telephone companies. Others argue that the LATAs are a better approach because they represent a more fair and equitable market.

		To add to the subscribers' confusion in this market, consumers are now faced with the decision of whether to buy 49-MHz cordless telephones, 900-MHz cellular telephones, or the new PCS communications devices. Many vendors are confusing the issue even further by describing new cellular features under the auspices of PCS.

 

Many features are being offered today that emulate PCS in the cellular market, such as one-number dialing (the ability to use one telephone number to reach a subscriber no matter where that subscriber is). However, there is more to PCS than just simple features. It represents a whole new network architecture, very different from the current cellular network.

		Many vendors have already alluded to the possibility of GSM technology for use in their new PCS networks. The only difference between other digital technologies lies in the digitizing method used at the air interface. GSM may prove to be the best solution for cellular providers here in the U.S. who are looking for a proven technology for their new wireless networks.

		Video and the Telephone Network

		During the early to mid-90s, there was a lot of activity between cable television companies and telephone companies. That interest has faded somewhat due to the lack of interest from subscribers. There is much speculation surrounding the mergers taking place between the cable television companies and the Bell Operating Companies (BOCs). Some speculate that the cable providers will begin competing for dialtone in other market areas. Others believe this provides yet another service that the Bell companies can provide to their subscribers: video on demand (VOD). But many have missed the underlying motivation for making these mergers today. The cable companies own lots of coaxial cable that is already installed to nearly every American home. In fact, statistics show cable companies already pass 90 percent of the homes in the U.S.

		Along with that coaxial cable is a healthy fiber optic backbone, capable of handling lots of high-speed data. While the cable network may be inadequate for broadband data services, it certainly can handle the demands of most telecommuters today. In fact, with a bandwidth of at least 10 Mbps, coaxial could be the answer for many of those who must reach LANs from their homes.

		But whether or not the telephone companies want to use this installed base of coaxial for data or video remains to be seen. One thing is certain: the merger of these networks into the Public Switched Telephone Network (PSTN) will place new requirements on SS7. This will be especially true if video dialtone becomes a reality.

		Video dialtone will be offered by the local telephone companies within this decade. The philosophy is that subscribers want convenience. Most subscribers will be willing to pay for that convenience. (Domino's Pizza proved that theory.) If the local telephone companies can provide pay-per-view movies over the telephone line for the cost of a long distance telephone call, they are guaranteed success or so the theory goes. In order for this to be successful, the network must be able to support it.

		The telephone companies will also find it necessary to add new entities to their networks. Cable service providers have already begun defining the specifications for video servers. A video server is nothing more than a large disk array, providing digital storage for scores of movies. These video servers can then be located anywhere in the telephone company network and, through signaling, triggered to download any movie to the appropriate telephone circuit.

		That circuit may very well be a coaxial service or digital service such as ISDN to a subscriber's home. Once again, the key to this process will be signaling to the video server to download the movies to the proper circuit and to initiate billing.

		On the voice networking side, little is required. With the advent of SONET between end offices, there is enough bandwidth to pump video signals through the voice network. However, a higher-layer protocol must manage the video transmission over the SONET network. Currently, the ATM protocol will support high-speed data and video over the SONET network.

		The ATM protocol provides the capability of packetizing video and any other form of information and moving it through the network in a switched fashion, just like placing a telephone call. ATM also supports the bandwidth requirements of video dialtone, currently offering 155 Mbps of bandwidth and migrating to a stealthy 600 Mbps by the end of this decade.

		At the subscriber level, there will need to be some additional hardware. Vendors are already releasing television set top boxes which will connect to the telephone line and allow movies down-loaded from the telephone company to be stored in memory and played back at the viewer's convenience. These playback boxes also provide rewind, fast-forward, and pause functions just as a VCR does, but will only allow a movie to be played in its entirety one time.

		Video dialtone will also bring home shopping options a little closer to convenience than they already are. While viewing the shop-at-home channels, purchases can be made by choosing a selection key on the television set top box remote control. This will allow the subscriber's information, such as credit card number and shipping address, to be downloaded to the shopping network without picking up the telephone. At the touch of a button, one will be able to purchase groceries, appliances, automobiles, and jewelry without ever leaving the armchair.

 

As far as SS7 is concerned, this new addition to the telephone network will be nothing more than one more facility. For multimedia applications, voice and video are combined over the same facility. The interactive portion (subscriber data downloading to home shopping networks) will rely on the signaling network for database access and control information.

		Another application is multimedia. Multimedia allows video to be combined with graphics and a user interface that allows the user to make selections by choosing icons on the television screen. When an icon is selected, a database is accessed to retrieve specific information. The medical profession is a big supporter of this type of service. Distance learning and teleconferencing are also likely candidates. However, the medical profession has already begun using this technology for patient diagnosis.

		For example, a doctor is sitting in his office in Fargo, North Dakota, diagnosing a patient with possible heart disease. For a second opinion, the doctor dials a specialist in New York City. By using a video camera connected to the telephone and a computer connected to the same interface, the doctor can video the entire office visit live to the specialist in New York City. The specialist can ask pertinent questions regarding the patient and view the patient through video as if the specialist were in the office with the patient.

		The patient's medical records can be transmitted in seconds to the specialist, while the office examination is under way using the computer. The specialist can even add comments to the file and send it back to the patient's doctor. Sound futuristic? This is the impetus behind the Information Highway and it is already under trial in several major cities today.

		So why don't all of us have this service now? Early trials were successful when the service was offered for free. However, when subscribers were asked to pay a monthly fee for the video access, they quickly lost interest. Many would still rather visit their local video store and rent the movies they want to see when they want to see them.

		Cable companies, however, now have a new driver. Internet access has reached an all time high. Subscribers are hungry for faster access to data and Web sites, and cable companies can deliver this access. Telephone companies have also found new ways of delivering high speed data access to the home through the use of Digital Subscriber Line (DSL) technology, which uses existing telephone wiring for high speed access. This technology has become serious competition to cable connections because existing wiring in the home can be used, eliminating the need to rewire subscribers' homes.

 

SS7 is in the background during all of this activity, quietly setting up connections and accessing databases. Information needed by the telephone companies to complete the video conference is provided by SS7 by accessing a Service Control Point (SCP) where all the information is stored and sending it to the requesting end office. SS7 will provide the connection control and billing services needed for such services.

		Broadband Data Communications

		Broadband (see Figure 1.9) is formally defined as any data communications with a data rate from 45 up to 600 Mbps. Broadband standards are still being defined today, with ATM receiving the bulk of attention. SS7 does not carry the actual user data, but provides the services necessary to connect the end-to-end telephone company facilities required for data transfer between two end points. The specific requirements of SS7 have been defined (see the

		Figure 1.9 This drawing depicts a typical broadband network. The subscriber side of the network may consist of these components or others. The SS7 network is used to control the connection of circuits between exchanges (end-to-end).

 

chapter on Broadband ISUP) and are being used in some networks where switched ATM is now available.

		ATM is a transport protocol that relies on upper-layer protocols for functions above and beyond layer 3. The ATM technology was created to support broadband ISDN, but has found favor with a number of other networking protocols as well. Frame relay, with speeds up to 155 Mbps, will most likely serve as a good solution for those requiring a PVC service. These work much like a dedicated special circuit and are not typically switched. However, there are plans to provide switched frame relay, making it a viable service for switched data networks.

		TCP/IP has also found new popularity as new telephone companies build entire networks based on this protocol for both voice and data. TCP/IP is not a good protocol for voice transmission today because it does not support real-time transmission (such as video and voice) and introduces a lot of delay in the transmission. For this reason, many of the companies deploying TCP/IP for real-time applications are using ATM as the transport, which supports realtime applications.

		Narrowband ISDN (1.554 Mbps) is being suggested by many service providers as a cheaper access to broadband services, while broadband ISDN (155 Mbps) is being targeted at the meganet-works that need much more bandwidth.

		Broadband ISDN (BISDN) will provide fast data transmission for services such as video dialtone. With the bandwidth of BISDN, video and audio can be transmitted simultaneously on the same facility. This will prove to be an important feature for the information highway.

		A point that often gets lost among all the marketing hype over ATM is the purpose of this technology. Talk of a new telephone network is nothing new. Telephone company officials have been planning for many years to upgrade their old analog networks to support the services and applications under much demand. Subscribers are no longer complacent with slow data transmission and expensive special circuits for sending data and other information through the network.

		ATM was developed to eliminate the need for time division multiplexed (TDM) circuits used in telephone networks between switching centers. The concept is to support one common network, rather than several distinct networks serving specialized applications such as television broadcast and voice transmission. Many vendors have placed ATM switches in the hands of the consumer (philosophically), thinking that this will meet the needs of every business and become a necessity in office buildings everywhere. Some have even been so bold as to claim that ATM will replace the local area network.

 

		In reality, ATM is found at customer premises where nothing else will deliver the bandwidth required for video and data services. ATM is too expensive for the average business to deploy in place of a LAN, and certainly constitutes overkill for most daily business activities. But for the university needing the 600 Mbps of bandwidth and the hospital using the telephone network to send highresolution medical images, ATM and broadband ISDN can be an integral part of their networks.

		Narrowband ISDN is still used as the customer interface for many subscribers, whereas ATM has become the transport mechanism within the telephone network to the destination. Digital Subscriber Line (DSL) technology is now being deployed as a better (and more affordable) alternative to narrowband ISDN. One of the attractions of DSL is that existing twisted copper wire can be used.

		ATM requires a transmission facility capable of carrying the bandwidth (600 Mbps and higher). Synchronous Optical NETwork (SONET) is being deployed as the physical medium for ATM.

		The Information Highway

		No book on telecommunications networking would be complete without some discussion on the Information Highway. The Information Highway is a coined phrase from the Clinton administration describing the technologies required to provide information services to all citizens. The requirements for the Information Highway are unclear. The only criterion with which politicians seem to agree is the ability to access a wealth of information using a variety of media, all accessible through the PSTN.

		This Information Highway in itself will require a major investment in the telephone network infrastructure. Certainly, this task could be achieved today if private networks were acceptable and if information was somewhat centralized. This is not the philosophy of the present administration, and they have set their agendas to transform the telephone network into an all-purpose information network, capable of providing access to telephones, databases, and video sources anywhere in the world.

		This message was carried to the United Nations (UN) in an effort to encourage all nations to jump onto the Information Highway, resulting in a worldwide network. This is certainly achievable maybe not within the time frame the Clinton administration would like, but certainly within the next decade.

 

The challenge is not how to access information from a variety of locations. This can be done today through packet switching. The challenge is how to provide this access through the PSTN, to anyone anytime, and to support all forms of media, whether it be audio, video, high-resolution graphics, or data.

		Today's telephone network is not equipped to handle much more than voice and some data transmission. The telephone companies have embarked on a major upgrade of the existing infrastructure. This upgrade is being implemented in phases.

		The first phase is to upgrade the carrier facilities. Telephone switching offices typically use DS1 or DS3 circuits for interoffice trunks. These facilities carry voice, data, and signaling traffic between exchanges. Video and high-speed data require more bandwidth than what any of these facilities provide and, for that reason, fiber optic facilities are now replacing the previous DS1s and DS3s.

		This fiber optic technology, called SONET, provides the bandwidth necessary to meet almost all applications. Yet another technology will be necessary for the switching and routing functions to carry the information from originator to destination, across the backbone network. Asynchronous Transfer Mode (ATM) has been developed for this reason.

		ATM provides the mechanism for transporting information in all forms of media from one exchange to another, and eventually to the subscriber. ATM development has been slow, as telephone companies battle with the cost of existing infrastructure and the cost of replacing that infrastructure with new technology.

		Already there has been much talk about the information highway, ATM, video on demand (VOD), and high-speed data available to every household by the end of the Clinton administration. But this development is not happening as fast as most would like. There are many issues to be resolved and much work to be completed on the standards that will deliver these services.

		In the last few years, an existing technology has suddenly gained popularity for voice transmission. Voice over IP (VoIP) has been pushed into the limelight by what began as a means for bypassing the existing telephone network (and associated long distance costs) by making telephone calls over the public Internet. Many technical issues are associated with using the public Internet, delay being the most prominent. Nonetheless, the industry has recognized the many benefits of TCP/IP as a means for transmitting voice as well as data.

		If there is any technology that will allow us to realize the Information Highway, it is TCP/IP. Already being deployed nationwide by a number of carriers, these networks will allow subscribers to send and receive packetized voice, data, audio, and video on computer terminals operating with software that supports telephone applications as well as audio and video. The following section describes packet telephony in more detail.

		Convergent Networks and Packet Telephony

		The Internet has spawned a completely new telephone network. VoIP has become a reality, with several companies currently providing VoIP services as an alternative to local telephone companies. Some of these companies are providing their network to other telephone companies, routing packetized voice over the public Internet. (This is not the trend for the majority by the way, as the public Internet lacks the robustness and reliability needed for telephone services.)

		TCP/IP brings many benefits to telephone service providers today. For new companies deploying new networks, packet switched technology is much more affordable (and more efficient) than the older legacy switching systems. The issue of voice over IP and the inability of IP to support real-time applications such as voice and video remain to be addressed, but there are other benefits to this approach of networking.

		A surprising number of new telephone companies are popping up all over the world, building their entire voice and data networks with TCP/IP. Rather than use expensive TDM-based switches, they are deploying next-generation computer platforms designed specifically for voice and data transmission in carrier networks. This new generation of switches are fully programmable, much more scalable, and of course much cheaper, allowing new startups to deploy many switches in markets they would not otherwise be able to afford.

		Many existing carriers are looking at TCP/IP for a different reason. They have realized that in many parts of their network, TCP/IP will save them money. Because many networks rely on databases to deliver services to their customers, and because these databases are deployed on computer platforms that support TCP/IP, carriers are finding that connecting SS7 networks to these databases via TCP/IP makes good sense.

		Even Signal Transfer Points (STPs) can be connected via TCP/IP today, reducing the cost of facilities within the SS7 network, as well as increasing the bandwidth of these switches to 100 Mbps per link (rather than 56/64 kbps supported by DS0A links).

		Using TCP/IP as a transport for SS7 provides huge benefits to carriers. SS7 is nothing more than packet data, and TCP/IP is a packet network protocol. Sending SS7 traffic over TCP/IP can be much cheaper because the facilities are less expensive than conventional channelized facilities. Another advantage to TCP/IP is the fact that every network supports it for data transmission (and Internet access). TCP/IP is everywhere, so finding a network to connect to is much easier than finding an existing ATM network.

		Some issues still exist, however. The IP protocol does not support many of the protocol functions we will discuss later in this book. Specifically, network management (a layer three protocol function) is lost when using IP. The industry is actively working to solve this problem through a variety of solutions. Changes are being made to the IP protocol to support this type of function, and vendors are developing overlay protocols that emulate the functions of SS7 layer three in an IP network (such as Tekelec's TALI).

		Carriers are already deploying TCP/IP into their networks today as a transport for SS7, as well as data. By moving data traffic away from the existing interoffice facilities onto TCP/IP networks, carriers save money they would normally have to spend to expand their switching equipment to support the growing number of callers making long duration data calls.

		A number of companies are working on standards to support this new networking trend. This has impacted the telephone industry more than any other technology and warrants discussion in this book. The impact on SS7 will be significant, but by no means will packet telephony make SS7 obsolete. If anything, packet telephony will allow telephone carriers to use the full potential of SS7.

		The Standards

		A number of standards are now being defined, all for different applications within the network. For the purposes of this book, we will only discuss the standards that are applicable to signaling (and related to SS7).

		The ITU, as well as TIA, ANSI, and Internet Engineering Task Force (IETF) are all actively involved in defining how signaling will be carried through the network. New signaling methods are also being examined for the various components of the network. Let's look at what the two largest standards groups are working on.

		ITU-TS H.323

		H.323 is really a suite of standards defining a variety of media types through packet networks. These standards cover everything from voice to data and facsimile and even video. They define how the signals are to be converted from analog to digital and what signaling is to be used.

 

		Figure 1.10 The protocol stack for H.323 applications

		In Figure 1.10, we see the various recommendations from the ITU for the applications to be supported. For the most part, these recommendations come from the H.323 suite of recommendations.

		The ITU has defined several entities for the converged packet telephony network. The four principal components are the terminal, gateway, gatekeeper, and signaling gateway. Each of these components has distinct functions, and various protocols are being defined to operate between the various components.

		Terminals

		These are the actual telephone devices themselves. They can be part of a computer terminal, or some other form of device with the responsibility of supporting voice transmission over the packet switched network. The voice is digitized and transmitted over the network using the Real Time Protocol (RTP), which is defined in TCP/IP. Think of terminals as computers on a LAN, even though in actuality the standards are referring to a logical function rather than a physical entity.

		Gateways

		The gateway is used to connect to other networks. It provides protocol conversion (say from an ANSI protocol to an ITU version of the same protocol) and uses Q.931 and Q.2931 (both defined in the ISDN recommendations) for sending call control and signaling between terminals and gateways.

 

Gatekeepers

		Gatekeepers are deployed and assigned to zones, much like a tandem telephone switch. The gatekeeper is the central point for all calls within its own zone and interconnects with gatekeepers in other zones using Q.931/Q.2931 protocols. Gatekeepers communicate to the PSTN network through a signaling gateway. (See Figure 1.11)

		Signaling Gateway

		The signaling gateway acts as a bridge between the PSTN and the packet telephony network. Standard SS7 messages are received from the PSTN and converted to TCP/IP-based messages. The upper levels of SS7 can remain the same as they are in the PSTN (ISUP, SCCP/TCAP), but the Message Transfer Part (MTP) is not usable in packet telephony and must be eliminated. Unfortunately, this means the many functions of MTP must be emulated (or replaced by something else) in the packet network. A great deal of work is being done today within the standards groups to define the methods to be used for SS7 type signaling within packet networks.

		Figure 1.11 The role of the gatekeeper and the gateway, and how they tie networks together.

 

Internet Engineering Task Force (IETF)

		The IETF has defined similar components as the ITU, but use entirely different terminology.

		Media Gateway (MG)

		The MG is synonymous with the terminal. It supports digitization and transmission of voice over the packet network. Many vendors are using Q.2931 for call control between media gateways and media gateway controllers. RTP is being used for the voice transmission over IP. Proprietary protocols are also being defined by various vendors for consideration by the IETF.

		Media Gateway Controller (MGC)

		The MGC is much like the gateways and the gatekeepers defined by the ITU. They are used to connect to other networks and provide interworking between dislike networks. The MGC interfaces with the signaling network, receiving information about the call from the media gateway (again using the Q.2931 protocol).

		Several competing protocols are being introduced as possible standards for these components to use for call control between one another, but none have been actually adopted as yet. It is important to understand that these protocols really do not replace SS7 because they will not work in the PSTN world.

		Signaling Gateway (SG)

		This component is the same as in the ITU world and is used to bridge the PSTN networks with the packet networks. The signaling gateway uses mapping tables to map SS7 point codes to the appropriate IP address. The gateway must be able to maintain current network status information as part of these routing tables, to ensure proper routing of messages during failure modes.

		There are presently two options for sending SS7 messages over IP networks. Tekelec has submitted the Transport Adaptation Layer Interface (TALI) for this purpose and has gained the support of numerous other vendors who have incorporated this interface into their own products. TALI uses TCP and adds an overlay layer to emulate SS7 MTP functions. TALI also provides additional features not yet supported in M2UA and M3UA, such as automatic CIC registration and MGC registration. These two features allow MGCs to be added to the network without operators having to perform translations in the network switches. TALI will not be replaced by M2UA or M3UA, but will continue to be supported as an augment to these protocols for those vendors wishing to continue supporting the interface on their platforms.

		M2UA, M3UA, and SCTP are presently being defined as replacements for MTP by the IETF, providing the same services as MTP but in a packet network. SCTP is a peer protocol to UDP and TCP, providing connection-oriented type services to guarantee delivery of messages. The M2UA and M3UA protocols provide replacements to SS7 MTP, providing the same services as MTP in packet networks. These will be discussed in more detail later.

		New Architecture

		There are numerous proposals for the new converged networks. The data world tends to follow the architecture of data networks, whereas the telephony world leans towards architectures more similar to the existing circuit-switched networks. There are pros and cons for both.

		One approach is to implement signaling gateway functions within each media gateway controller (we will use the IETF nomenclature for simplicity here). This means that each MGC is responsible for routing SS7 messages from the PSTN to IP entities within the packet network. In order to perform this function, routing tables must be maintained within each MGC mapping the SS7 addresses (point codes) to IP addresses on the packet side.

		In the TCP/IP environment, this is a very typical approach (routers maintain their own routing tables through the use of OSPF, RIP, and other routing protocols). However, these protocols are not fast enough for the function of SS7. For example, if a media gateway fails, the routing protocols may take several seconds before the status of the failed node can be propagated through the network. This doesn't seem like much time to us mere mortals, but this is a long time to the SS7 network.

		As we examine the SS7 MTP protocol, you will find that the SS7 network sends continuous messages to every node in the network. The purpose of these messages is to determine whether the node is still capable of handling traffic. If a node fails, the network knows about the failure immediately and notification of adjacent nodes is already taking place. The SS7 MTP protocol is much more proactive than TCP/IP routing protocols, which is why there is still much work to be done in TCP/IP to support these functions.

		Another issue with the previous configuration is the need to administer every MGC whenever new components are added into the network. This is resource-intensive and one of the reasons SS7 was developed the way it was. The intent of the IN is to place network intelligence in a more centralized location, keeping devices at the edge of the network as simple as possible. This not only lowers the cost of the entities at the edge of the network, it also simplifies network management.

		Figure 1.12 A centralized approach for network intelligence, supporting cheaper non-intelligent devices at the network edge.

		The best approach for converging networks is to centralize the intelligence of the network as shown in Figure 1.12. By centralizing the SS7 to IP conversion, you minimize the administration within the network. You also provide a central point of contact for network monitoring, billing, and Advanced Intelligent Network (AIN) applications.

		This architecture also lowers the cost of network equipment. By moving the intelligence up in the network, you maintain simplicity in the edge equipment. Because there are more of these devices than databases and STPs, this is where the cost should be minimized.

		If legacy equipment must be interfaced, some vendors offer front ends. These devices are more like scaled-down signaling gateways, providing conversion between SS7 and TCP/IP. This is much more economical than developing modules to incorporate into their product and provides a faster path to market than if development was required.

 

Now that we have discussed the technologies used today to form the world's most advanced communications network, let us look at the various organizations responsible for developing these standards.

		Standards Organizations

		Standards such as SS7 and TCP/IP do not happen quickly. They are the result of years of research and development conducted by standards committees. These organizations are usually composed of government agencies or industry representatives from manufacturers and service providers.

		To understand the various standards available today, one must first understand the purpose of new standards and how they are developed. There are two different types of standards: de jure and de facto.

		The de jure standard is formed by committee. These standards take many years to develop because the processes used in committees are long and bureaucratic. Nonetheless, many of the standards used today are the result of standards committees.

		A de facto standard is the result of a manufacturer or service provider monopolizing a market. A good example of a de facto standard is the DOS personal computer. IBM was instrumental in saturating the market with their PCs, but, more importantly, with encouraging third-party vendors to use their architecture and build IBM PC clones, using the same disk operating system. The result of this IBM marketing strategy is still felt by its competitors today. There are so many DOS PC computers in the market that introducing a new platform requiring a different operating system is a high risk. Yet there are no standards in existence that define the use of DOS in all personal computers.

		De jure standards and de facto standards can be voluntary or regulatory standards. Voluntary standards are adopted by companies on a voluntary basis. There are no rules that say all manufacturers and service providers must comply with a voluntary standard. However, the advantages are many. Voluntary standards help ensure that everyone developing networking products builds their products for interconnectivity. Without this interconnectivity and interoperability, only a few equipment manufacturers would win the market those with the largest install base.

		Interoperability is another issue in data communications. Interoperability is the ability for equipment to communicate with equipment from different vendors in a network environment. Often, vendors will implement varying protocol versions in their equipment and are noncompliant with the standards. When this occurs, other equipment cannot communicate with the noncompliant system because it uses a proprietary interface, forcing subscribers to purchase all their equipment from the same manufacturer.

		Voluntary standards help ensure the interoperability of all networking equipment. With voluntary standards, all those participating in the technology can have a voice in the final ''product." The organizations responsible for creating these voluntary standards are usually made up of industry representatives. These representatives work for the same companies who build the equipment. As the technologies evolve, companies participating in the development of the standards can also get a sneak preview of what the final standards will consist of, and can be first to market with product that is compliant with the standards.

		Regulatory standards are created by government agencies and must be conformed to by the industry. These standards do not hold any major advantage to the service provider or the manufacturer, but are in place in most cases to protect the consumer.

		Regulatory standards are monitored by government agencies such as the Federal Communications Commission (FCC). These agencies ensure the protection of the public and other network users by enforcing standards covering safety, interconnectivity, and, in some cases, health (i.e., radiation emission from computer terminals and cellular phones).

		SS7 networks use standards from a variety of organizations and standards committees. Some of the standards used in SS7 networks were developed for other applications as well, not specifically for SS7. The following organizations have written standards directly related to SS7:

		International Telecommunications Union Telecommunications Standardization Sector (ITU-TS)

		American National Standards Institute (ANSI)

		Bell Communications Research (Bellcore) Telcordia (formerly Bellcore)

		In addition to these organizations, many other standards have been written that affect SS7. The following standards organizations have contributed to the SS7 network with standards not written specifically for SS7 but used by equipment in the network:

 

Electronic Industries Association (EIA)

		ATM Forum

		Federal Communications Commission (FCC)

		Underwriters Laboratories (UL)

		Canadian Standards Association (CSA)

		International Organization for Standardization (ISO)

		Internet Engineering Task Force (IETF)

		These organizations are responsible for the standards which govern the quality of cables, quality standards for manufacturing practices, electrical specifications, and interfaces used to interconnect telecommunications equipment. In addition to these standards organizations, many new forums have evolved. As the industry begins to understand the importance of standards bodies and compliance to these standards, new industry forums evolve composed of vendors and service providers in the industry with a vested interest in the technology.

		These forums are often commissioned by the ITU-TS and ANSI to develop new standards in their behalf (such as ATM) or work on issues with existing standards (such as ISDN). In many cases, these forums can develop standards much faster than the standards committees themselves, saving the committees years in development time.

		Currently, there are just a few of these forums which are relevant to the telecommunications industry. They are described here because their work involves portions of the SS7 network. They are

		Network Operations Forum (NOF)

		ATM Forum

		The rest of this section will look at all of these organizations in greater detail. The purpose of describing these organizations is to provide a better understanding of who the players are and what significance they carry. The major organizations are described in greater detail than some of the less significant ones.

		International Telecommunications Union Telecommunications Standardization Sector (ITU TS)

		Formerly known as the CCITT, this organization is a part of the ITU, which is a United Nations (UN) Treaty organization. The purpose of the ITU-TS is to provide standards that will allow end-to-end compatibility between international networks, regardless of the countries of origin. The standards are voluntary standards, but many countries require full compliance to connect to their networks.

		The members of the ITU-TS are government representatives from the various nations. The representative for the U.S. is the Department of State. In addition to government agencies, manufacturers and service providers carry some influence as well. Membership is limited to four categories:

		Administrations of a country's public telephone and telegraph companies

		Recognized private operating agencies

		Scientific and industrial organizations

		Standards organizations

		The ITU was reorganized into three sectors: the Radiocommunication Sector (ITU-RS), the Telecommunication Development Sector (ITU-D), and the Telecommunication Standardization Sector (ITU-TS). The ITU-TS is the sector responsible for defining SS7 (or C7, as it is known internationally) standards and other related standards.

		The ITU-TS standards for SS7 have been embraced by every country that is deploying SS7, yet not every country's network is the same. One would think that, with international standards in place, interconnectivity would not be an issue. Yet every country creates its own standards to meet the requirements within its own networks.

		Because of this independence within individual countries, the SS7 network hierarchy consists of an international network and many national networks. The national networks are based on the ITU-TS standards, but modified for usage within individual countries. The U.S. uses the ANSI standards as its national standard. The ANSI standards are based on the ITU standards, but with several differences. The differences between the ANSI standards and the ITU-TS standards are mainly in the addressing (point codes) and in network management procedures. Telcordia has also published a set of standards, based on and endorsed by ANSI. The ANSI standards and the Bellcore standards are virtually the same except for the additions and modifications added by Telcordia to ensure network reliability and diversity.

		The ITU-TS C7 standards were first defined in 1980. These are referred to as the Yellow Book. ITU-TS standards were once published every four years in a set of documents, which are color-coded to indicate the year in which they were published. The color code is as follows:

		1980 Yellow Book

		1984 Red Book

		1988 Blue Book

		1992 White Book

		The ITU has changed the way it releases updated standards today. Rather than wait four years, they have adopted the model used by ANSI, and release new updates and standards as soon as contributions have been adopted and agreed upon by the members.

		The C7 standards can be found in the ITU-TS documents numbered Q.701 through Q.741. The following list identifies all those documents which are SS7 standards or related to SS7:

		Q.700-Q.709 Message Transfer Part (MTP)

		Q.710 PBX Application

		Q.711-Q.716 Signaling Connection Control Part (SCCP)

		Q.721-Q.725 Telephone User Part (TUP)

		Q.730 ISDN Supplementary Services

		Q.741 Data User Part (DUP)

		Q.761-Q.766 ISDN User Part (ISUP)

		Q.771-Q.775 Transaction Capabilities Application Part (TCAP)

		Q.791-Q.795 Monitoring, Operations, and Maintenance

		Q.780-Q.783 Test Specifications

		The ITU-TS recently announced a change in the documentation structure, which will affect all new publications from the ITU-TS. There should be no effect, however, on existing SS7 standard publications. These will not be changed.

		It should also be noted here that the ITU sponsors the world's largest telecommunications trade show every four years in Geneva. This trade show (as well as the associated conferences) have become the highlight of the industry, with companies from all over the world assembling to discuss the latest technologies and products available.

		American National Standards Institute (ANSI)

		The ANSI (see Figure 1.13) is responsible for approving standards from other standards organizations for use in the U.S. There are many organizations considered accredited standards bodies by ANSI, including Telcordia and the EIA. ANSI is divided into committees. The ANSI Accredited Standards Committee T1 is responsible for standards associated within the telecommunications industry.

 

		Figure 1.13 The ANSI organizational chart.

		The T1 committee's responsibilities include developing standards for interconnection and interoperability of telecommunications networks. The T1 committee is divided into seven Technical Subcommittees which receive their direction from the T1 Advisory Group (T1AG).

		The T1AG meets on a bimonthly basis and is responsible for the establishment and administration of procedures for the committee's activities. As illustrated in Figure 1.14, the Advisory Group reports to the T1 committee.

		The secretariat provides support functions to the subcommittees and interacts with ANSI to get approval and publication of the standards created by the subcommittees. Duties of the secretariat include scheduling committee meetings and approving memberships.

 

Following is a description of each of the seven subcommittees along with their missions, according to the T1 Committee Procedures Manual.

		Technical Subcommittee T1E1

		This subcommittee defines standards for network interfaces, concentrating on physical layer interfaces. This includes the electrical, optical, and magnetic specifications of data communications interfaces, as well as telephony interfaces.

		It consists of four working groups, all which work closely with each other and with other groups within the T1 Committee:

		T1E1.1 Analog Access

		T1E1.2 Wideband Access

		T1E1.3 Connectors and Wiring Arrangements

		T1E1.4 DSL Access

		Technical Subcommittee T1M1

		This subcommittee's primary focus is on the processes and procedures used in the operations, maintenance, and administration of telecommunications networks. This includes the testing of facilities, measurements, routine maintenance, and traffic routing plans. Their standards focus on the engineering and planning functions, network resources, and support systems. There are four working groups in this subcommittee:

		T1M1.1 Internetwork Planning and Engineering

		T1M1.2 Internetwork Operations

		T1M1.3 Testing and Operations Support Systems and Equipment

		T1M1.5 OAM&P Architecture, Interfaces, and Protocols

		Technical Subcommittee T1P1

		This subcommittee provides support services and program management for the rest of the T1 subcommittees. It provides high-level descriptions, high-level overviews and architectures, scheduling for interactive sessions between subcommittees, publishing support of standards, and reference models, and it determines T1 endorsement of programs.

		Technical Subcommittee T1Q1

		This subcommittee focuses on performance issues of network traffic, switching, transmission, maintenance, availability, reliability, and restoration. Its performance specifications are standards used from carrier to carrier and from carrier to customer interfaces. There are four working groups:

		T1Q1.1 4-kHz Voice and Voiceband Data

		T1Q1.2 Survivability

		T1Q1.3 Digital Packet and ISDN

		T1Q1.5 Wideband Program

		Technical Subcommittee T1S1

		The T1S1 subcommittee is directly involved with SS7 signaling standards as well as ISDN and other related services, architectures, and signaling. It reviews international standards and makes decisions on how those standards will be implemented in the U.S. In addition, it works closely with the ITU-TS in developing standards for the international community. There are four working groups in this subcommittee:

		T1S1.1 Architecture and Services

		T1S1.2 Switching and Signaling Protocols

		T1S1.3 Common Channel Signaling

		T1S1.5 Broadband ISDN

		Technical Subcommittee T1X1

		The members of this subcommittee define the standards used to define the hierarchy of digital networks and synchronization networks. This subcommittee defines standards used for internetworking, focusing on the functions necessary to interconnect at the network transport level. There are four working groups:

		T1X1.1 Synchronization Interfaces

		T1X1.4 Metallic Hierarchical Interfaces

		T1X1.5 Optical Hierarchical Interfaces

		T1X1.6 Tributary Analysis

		Technical Subcommittee T1Y1

		This subcommittee defines standards not covered by any of the other subcommittees and serves as a sort of miscellaneous standards committee. Specialized video and audio services, including broadcast services, teleconferencing, and graphics, as well as specialized voice and data processing, are within this group's jurisdiction. There are three working groups:

 

T1Y1.1 Specialized Video and Audio Services

		T1Y1.2 Specialized Voice and Data Processing

		T1Y1.4 Environmental Standards for Exchange and Interexchange Carrier Networks

		The ANSI publications regarding SS7 define the function of the protocols. Telcordia has published numerous other documents detailing the specific requirements of all SS7 entities and management procedures. Telcordia protocol chapters and ANSI publications for SS7 are numbered as follows:

		T1.110 Signaling System 7 (SS7), General

		T1.111 Message Transfer Part (MTP)

		T1.112 Signaling Connection Control Part (SCCP)

		T1.113 ISDN User Part (ISUP)

		T1.114 Transaction Capabilities Application Part (TCAP)

		T1.115 Monitoring and Measurements

		T1.116 Operations, Maintenance, and Administration Part (OMAP)

		An ANSI catalog is available for all ANSI publications by contacting the ANSI organization in New York City.

		Bell Communications Research (Bellcore) Telcordia (Formerly Bellcore)

		Telcordia was the research and development arm for the seven Bell Operating Companies. These seven companies were divided from the Bell System in 1984 as part of the divestiture. AT&T was separated from the local exchanges, taking with it Bell Laboratories. The new Bell Operating Companies (BOCs) then founded what was then named Bellcore to replace the services once provided them by Bell Laboratories. The seven operating companies were

		Ameritech

		Bell Atlantic

		BellSouth Telecommunications

		NYNEX

		Pacific Telesis

 

Southwestern Bell

		US West

		Through the many mergers and acquisitions over the last few years, this list has dwindled down to just a few company names. Today, the RBOCs are

		Bell Atlantic

		BellSouth

		Southwestern Bell (SBC)

		In addition to these regional operating companies, Telcordia also provided services to the following "nonowners":

		Cincinnati Bell, Inc.

		Southern New England Telephone Co.

		Centel Corporation

		General Telephone Company and its local telephone companies

		Sprint and its local telephone companies

		Canadian local telephone companies

		The old Bellcore was recently sold to a company called SAI, which promptly changed the name to Telcordia. Although the company still provides services to the regional operating companies, it has broadened its scope to cover the entire industry through a variety of services.

		Telcordia publishes standards and recommendations for all kinds of telecommunications services, maintenance and operations procedures, and network architecture. These documents are available for a fee to any individual through the Telcordia organization.

		Telcordia changed their documentation numbering system a few years ago, so you will find some older documents may not match their newer counterparts. What is described next is the former identity used to communicate document phases. The three levels described below have since been collapsed into one identity, known as the Generic Requirement (GR). All new Telcordia documents will use this new convention. The former identity is defined because there are still many documents in circulation using the older identification.

 

	Bellcore documents were developed and published in three phases. The first phase of publication was the Framework Advisory (FA). Any document beginning with "FA" is a Framework Advisory and is still in draft format. These documents were published and submitted to the industry for comments and suggestions.

		The second phase was known as the Technical Advisory (TA). These documents were preliminary publications and were still subject to minor changes. They were submitted to the industry for final comment and suggestions before reaching final publication.

		The third phase of publication was the Technical Reference (TR), which was the final phase of the document. Technical References represent final released versions of a document. All of these publications were numbered with the following convention:

		TR-NWT-000082

		The preceding document number is just one example of the document numbering of Telcordia publications. The "TR" indicates that this is a technical reference, the "NWT" indicates this is a network-related publication, and the six-digit number identifies the unique publication. As previously mentioned, Telcordia has recently implemented a new document numbering scheme, which alleviates the three levels of numbers and identifies all new documents as Generic Requirements (GR).

		In addition to these types, Telcordia also publishes documents on technology findings and services offered through their regional companies. These are referred to as Science and Technology (ST) publications and Special Reports (SRs).

		SRs are issued for a variety of reasons today. Vendors who deal with the RBOCs are sometimes required to have their systems tested by a third party (such as Telcordia) and provide the results of the testing through a publication. Telcordia publishes their test results through serialized SRs, which are tightly controlled documents (in this case).

		They also issue SRs to the general industry when there is an event that the industry needs to know about. For example, a major network outage is usually investigated, with the results published in an SR for the industry. This is one means of educating the entire industry of potential network problems so the same event will not repeat itself.

		A publication regarding Telcordia requirements for interconnectivity within networks is called a FR and specifies design requirements, as well as the functionality of specific network elements. These are classified as

 

LATA Switching Systems Generic Requirements (LSSGR) (FR-NWT-000064)

		Operator Services Systems Generic Requirements (OSSGR) (FR-NWT-000271)

		Operations Technology Generic Requirements (OTGR) (FR-NWT-000439)

		Reliability and Quality Generic Requirements (RQGR) (FR-NWT-000796)

		Transport Systems Generic Requirements (TSGR) (FR-NWT-000440)

		The Telcordia publication GR-246-CORE is a multi-volume series that closely matches the ANSI publications and defines the SS7 protocol. Telcordia has added many modifications and procedures enhancing the security and reliability of SS7 networks. Originally, the Telcordia version identified the Bell System implementation of SS7 and identified specific procedures and functions required in these networks.

		The chapter numbering is the same as in ANSI publications, allowing cross-referencing between publications. The Bellcore version identifies the Bell System implementation of SS7 and identifies specific procedures and functions required in the Bellcore networks. The protocol descriptions are identical to the ANSI publications, with the exception of some Bellcore implementations. A catalog of Bellcore/Telcordia documents is available through Bell Communications Research at www.telcordia.com.

		Electronic Industries Association (EIA)

		The EIA develops standards focused on the physical interfaces used in the data communications industry. Probably the best example of an EIA standard is the RS-232C. This standard was originally created for interfacing modems to computer systems, but it proved so simple and versatile that it quickly became the de facto standard for any application requiring a serial interface.

		The EIA has developed many other standards besides the RS-232C, including some faster interfaces designed to replace the RS-232C. The EIA is also responsible for many wireless standards, including the cellular standard called the Interim Standard-41 (IS-41). This standard defines the interfaces between cellular network entities as well as the communications protocols used at these interfaces. The IS-41 standard was developed by the EIA/TIA Sub-committee TR-45.2, Cellular System Operation.

		ATM Forum

		The ATM Forum is a voluntary organization consisting of industry and public sector representatives. Their mission is to assist the ITU-TS in developing a standard for international use in ATM networks. The ITU-TS has been actively working on this standard, but was not expected to complete the standards until the year 2000. The ATM Forum was formed to help expedite the process of development, in hopes of finishing this standard much sooner.

		Much of the work accomplished by the ATM Forum has been submitted and accepted (with modification) by the ITU for inclusion in the final ATM standards. It is important to understand, however, that the ATM Forum does not write standards. They write implementation agreements, which allow vendors to agree on certain aspects of the technology and begin development on products prior to the final standards being completed.

		Formed in November 1991, the ATM Forum is now over 400 members strong, with representatives from a wide spectrum of interests. Data communications and telephone companies, service providers, and private networking types have joined together in this consortium to ensure a standard which meets the needs of all interested industries.

		Federal Communications Commission (FCC)

		The FCC was created as part of the Communications Act of 1934 and is responsible for the regulation of the airwaves, as well as for the regulation of the telecommunications industry. There are five commissioners in the FCC, all appointed by the president of the U.S. Each commissioner serves a term of five years. One commissioner is appointed by the president to serve as chairperson of the FCC.

		There are four operating bureaus within the FCC organization, with a total staff of 1700 personnel. Each operating bureau serves a specific function: mass media, common carrier, field operations, and private radio. The common carrier bureau regulates all aspects of the telecommunications industry, including paging, electronic message service, point-to-point microwave, cellular radio, and satellite communications.

		The FCC regulates the access of interconnect companies and determines the type of interfaces to be used. They do not necessarily create standards, but they enforce regulations regarding the interconnection of networks and network devices. They also issue licenses for radiotelephone circuits and assign frequencies for their operation. The FCC allocated new frequencies for the PCSs being deployed in the U.S. by many major wireless service providers. Rather than create new frequencies, the FCC reallocated frequencies previously used by microwave.

		The FCC also supervises charges and practices of the common carriers, approves applications for mergers, and determines how the carriers will maintain accounting for their operations. The FCC also governs the type of services that a service provider can provide.

		A good example of how the FCC regulates the telecommunications industry is the registered jack program. All manufacturers of telephone equipment, or any equipment which connects to the telephone network, must have their equipment registered with the FCC. The FCC then determines which type of interface [registered jack (RJ)] the equipment must use to connect to the network.

		The former Bell System sometimes gets credit for these interfaces, since they usually are the ones who install and maintain them. But the FCC determines how these interfaces will be used and who must use them. The RJ-11, used to connect single-line telephones to the central office line, is found in every home across the United States today. Most of us refer to them as modular jacks.

		The Network Reliability Council (NRC) was formed to monitor network outages and seek resolutions through the industry vendors and service providers. This council works closely with the key corporations in resolving key issues which can be related to service outages in the nation's SS7 networks. This information is published and shared with all service providers to prevent future outages from occurring due to common failures. The NRC was formed by the FCC after several network outages in the SS7 network shut down telephone service in several major cities in 1991.

		Today, all outages must be reported by the service providers to the NRC. This information is then disseminated among service providers as an information-sharing mechanism. The FCC hopes that sharing information regarding software deficiencies from the various vendors can help deter many network outages.

		Underwriters Laboratories (UL)

		The UL is a not-for-profit organization which began in 1894. The UL uses a suite of tests and gives approval to any equipment which passes the minimal requirements set by the test suite.

		UL approval is not required here in the U.S., but is sought after by most manufacturers of electrical or electronic equipment. The UL label certainly influences buying decisions, and in most building codes it is a requirement for electrical equipment.

		Canadian Standards Association (CSA)

		The CSA is Canada's equivalent of the UL. CSA approves all electrical and electronic products for sale in Canada. Manufacturers in the U.S. who plan to sell their equipment in Canada usually have the equipment approved by the CSA. If the equipment is sold in the U.S. as well, the equipment must be UL and CSA approved.

		As with the UL, CSA approval is not a requirement. However, in the telecommunications industry, most buyers will require CSA approval before purchasing equipment. This is their guarantee that the equipment they are buying went through some level of testing for electrical compliance.

		International Standards Organization (ISO)

		The ISO is an international standards organization responsible for many data communications standards, including the Open Systems Interconnection (OSI) model. The OSI model was developed after SS7 and was not adopted as a standard until 1984. However, layering was well understood and practiced during the early to late '70s, which is when the work was being done on SS7 protocols. The OSI model is still used today to define the functions of the various levels within a protocol stack.

		The ISO is composed of other standards bodies from various countries, mostly government agencies responsible for setting communications standards within their own governments. The U.S. representative is ANSI.

		The ISO does not limit itself to just data communications standards. They have created many other types of standards as well. One of their most recent contributions to industry is the ISO 9000 quality standards. The ISO 9000 defines processes to be used in manufacturing to ensure quality production. Again, these are not mandatory standards, but are essential for companies selling products in Europe because most European buyers now require ISO 9000 compliance.

		Internet Engineering Task Force (IETF)

		This organization has the responsibility for defining all standards related to the Internet. The standards are published in the form of Recommendations for Comments (RFCs). Anyone can submit RFCs; however, they do not become adopted standards without approval from the IETF working committees. Thousands of RFCs cover hundreds of protocols used in TCP/IP networks. The IETF also validates through testing that the proposed RFCs will actually work within the TCP/IP networks.

		IETF has three different areas of concern: applications, transport, and security. Signaling is being addressed through the transport working groups. Several groups are involved in defining transport protocols: Sigtran, SIP, and Megaco.

		Sigtran is defining the protocols to be used with SS7. This includes M2UA, M3UA, SCTP, and TALI. At the time of writing, Sigtran was near completion with the SCTP RFCs. The M2UA and M3UA protocols are still some time from completion.

		The SIP working group is responsible for the Session Initiation Protocol (SIP) to be used between MGCs within a network. This protocol will allow call setups from gateway to gateway when MGC domains have to be traversed.

		The Megaco working group is responsible for defining the protocol to be used between the MG and the MGC. This started as the Media Gateway Controller Protocol (MGCP), but has since been adapted and renamed as Megaco.

		Other Agencies

		In addition to the standards organizations, there are other agencies that have made a significant impact on the telecommunications network. These agencies are responsible for ensuring reliability in our telecommunications network. Following are the most prominent agencies:

		Network Reliability Council (NRC)

		First commissioned by the FCC in 1992, this council was chartered to investigate network outages and report them to all network providers, as well as to vendors. They were placed into existence only after numerous network outages caused telephone service to be out for extended periods of time, costing many corporations millions of dollars (including Wall Street in New York) and even necessitating the closing of an airport.

		Their charter was to have expired in 1994, but they received a new charter by the FCC and are still tasked with the investigation of network outages. The commission is made up of CEOs from leading carriers and manufacturers, and provides reports regarding the reliability of the nation's network, as well as explanations for outages and how they can be prevented in the future.

 

Network Operations Forum (NOF)

		The NOF was originally formed in 1984, over concerns as to who would track and clear trouble reports that crossed network boundaries. They have since expanded their operations to include definitions for interoperability testing and interworking issues. They also meet with manufacturing companies to resolve issues regarding the reliability of network equipment.