Title: Signaling System #7 Second Edition
Author: Travis Russell
Publisher: McGraw-Hill
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Signaling System #7
Early telephone networks were the result of years of evolution, with lit-tle 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 pack-et-switching networks and data communications services the tele-phone 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.
The International Telecommunications Union (ITU) commissioned the then CCITT to study the possibility of an all-digital intelligent net-work. The result was a series of standards known now as Signaling System #7 (SS7). These standards have paved the way for the
Intelli-gent 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 Intelligent Network (IN). Introduction to SS7
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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 enti-ties. These messages travel from one network entity to another, inde-pendent of the actual voice and data they pertain to, in an envelope called a packet.
Common Channel Signaling (CCS) was first introduced in the United States in the 1960s as Common Channel Interoffice Signaling System #6 (SS6). Developed by the International Telecommunications Union—Telecommunications Standards Society (ITU-TS), SS6 used a separate facility for sending signaling information to distant telephone offices.
The first deployment of SS6 in the United States 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.
Signaling System #7 (SS7) was derived from the earlier SS6, which explains the similarities. SS7 provides much more capability than SS6. Where SS6 used fixed-length signal units, SS7 uses variable-length sig-nal units (with a maximum sized length), providing more versatility and flexibility. SS7 also uses high-speed data links (56 kbps). This makes the signaling network much faster than SS6. In international networks, the data links operate at 64 kbps. Study is under way to increase this in the United States to 1.544 Mbps, and internationally to 2.048 Mbps.
As of 1983, SS6 was still being deployed throughout the United States telephone network, even though SS7 was being introduced. As SS7 began deployment in the mid-1980s, SS6 was phased out of the network. SS7 was used in the interoffice network and was not immedi-ately deployed in the local offices until many years later.
In fact, the first usage of SS7 was not for call setup and teardown, but for accessing remote databases. In the 1980s, the 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 Public Switched Telephone Network (PSTN).
When an 800 number is dialed, the telephone company switching equip-ment 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 equip-ment 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 includ-ing 800 numbers, 900 numbers, 911 services, custom callinclud-ing features, caller identification, and many new services yet to be offered.
800 numbers at one time belonged to one service provider. If sub-scribers 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 and accessed via the SS7 network. SS7 is now used to allow 800 num-bers to become transportable and to provide subscrinum-bers the option of keeping their 800 numbers even when they change service providers.
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 cus-tomers wish to change their service from Plain Old Telephone Service (POTS) to ISDN, they would normally be forced to change telephone num-bers. This is because of the way telephone numbers are assigned in switch-ing equipment, with switches assigned ranges of numbers.
With LNP, the telephone number does not change. 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.
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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 informa-tion from their Home Locainforma-tion Registers (HLRs), so cellular sub-scribers 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 net-works to carry IS-41 signaling information through their network. This did not allow them to interconnect through the Public Switched Telephone Network (PSTN) because the X.25 network was not compatible with the PSTN signaling network (SS7). The cellular providers are aggressively changing this situation today, deploying their own SS7 networks.
Today, SS7 has been deployed throughout the Bell Operating Companies 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 tele-phone 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, ATM, ISDN, and cellular. 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 net-work. 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
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
tele-phone 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 com-panies 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 facili-ties, 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.
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pioneers of signaling never dreamed of. CCS is the technology that makes ISDN and SS7 possible.
The concept behind Common Channel Signaling (CCS) SS7 is sim-ple. 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 immedi-ate 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 record-ing 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 Common Channel Signaling (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).
CCS uses existing telephone company resources, so it does not require additional facilities to be installed. 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.
CCS is in wide use today, even though many in the telecommunica-tions business do not understand it. SS7 is the protocol and architec-ture used in this new network and is the topic of this book.
sup-port network management functions or control information between switches and operations systems. The exception is SS7. Because SS7 con-sists 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 supervi-sion. 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 cen-tral 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 data-bases, and provides security against telephone fraud. Personal identi-fication 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 cur-rent. 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.
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DC signaling relies on DC current to signal the distant end. The sim-plest example of DC signaling is used in Plain Old Telephone Service (POTS) between the subscriber and the local end office. When a sub-scriber goes off-hook, DC current from the central office is allowed to flow through the telephone (the switch-hook provides the contact clo-sure 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 send-ing a dialtone. A dialtone signals the subscriber to begin dialsend-ing the telephone number. This can be done using a rotary dial or a Dual-Tone Multi-Frequency (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 cen-tral 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 cir-cuit must remain connected until either party hangs up. The originat-ing office determines which circuits to use by searchoriginat-ing its routoriginat-ing 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.
