184 Chapter 8: ISDN User Part (ISUP)
`
`ISUP consists of call processing, supplementary services, and maintenance functions. This
`chapter is divided into the following sections, which describe the specific components of
`ISUP:
`
`• Bearers and Signaling
`•
`ISUP and the SS7 Protocol Stack
`•
`ISUP Message Flow
`• Message Timers
`• Circuit Identification Codes
`• Enbloc and Overlap Address Signaling
`
`• Circuit Glare
`• Continuity Test
`•
`ISUP Message Format
`• Detailed Call Walk-Through
`• Circuit Suspend and Resume
`•
`ISUP and Local Number Portability
`•
`ISUP-ISUP Tandem Calls
`•
`Interworking with ISDN
`• Supplementary Services
`• Additional Call Processing Messages
`• Maintenance Messages and Procedures
`
`Bearers and Signaling
`ISUP allows the call control signaling to be separated from the circuit that carries the voice
`stream over interoffice trunks. The circuit that carries the voice portion of the call is known
`within the telephone industry by many different terms . Voice channel, voice circuit, trunk
`member, and bearer all refer to the digital time slot that transports the voice (fax, modem,
`or other voiceband data) part of a call. The term "voice circuit" can be somewhat ambiguous
`in this context because sometimes it is used to ref er to the trunk span that is divided into
`time slots, or to an individual time slot on a span.
`
`The signaling component of the call is, of course, transported over SS7 signaling links . This
`creates two independent paths for call information between nodes: the voice path and the
`signaling path. The signaling mode describes the signaling relation between the two paths.
`Following is a brief review of the associated and quasi-associated signaling modes as they
`relate to ISUP, which we discussed in earlier chapters.
`
`
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`ISUP and the S87 Protocol Stack
`
`185
`
`If the signaling travels on a single linkset that originates and terminates at the same nodes
`as the bearer circuit, the signaling mode is associated. If the signaling travels over two or
`more linksets and at least one intermediate node, the signaling mode is quasi-associated. In
`Figure 8-1,partA shows quasi-associated signaling between SSP A and SSP Band between
`SSP B and SSP C. In part B of Figure 8-1, the same SSP nodes are shown using associated
`signaling. Notice that the signaling links in part B terminate at the same point as the trunks .
`Also, the signaling link is shown as a separate entity in part B to illustrate the signaling
`mode; however, it is typically just another time slot that is dedicated for signaling on a trunk
`span.
`
`Figure 8-1
`
`Signaling Mode Relating to /SUP Trunks
`
`STP A--~ STP B
`
`ISUPTrunks
`
`ISUPTrunks
`
`A.) Quasi-Associated Signaling for ISUP Trunks
`
`ISUPTrunks
`
`ISUPTrunks
`
`B.) Associated Signaling for ISUP Trunks
`
`The signaling mode used for ISUP depends greatly on what SS7 network architecture is
`used. For example, North America uses hierarchical STPs for aggregation of signaling
`traffic. Therefore, most ISUP trunks are signaled using quasi-associated signaling. Using
`this mode, the signaling is routed through the STP before reaching the destination SSP. In
`contrast, while the U.K. uses quasi-associated signaling for some SSPs, they also heavily
`use associated signaling with directly connected signaling links between many SSPs.
`
`ISUP and the SS7 Protocol Stack
`As shown in Figure 8-2, ISUP resides at Level 4 of the SS7 stack with its predecessor, the
`Telephone User Part (TUP). TUP is still used in many countries, but ISUP is supplanting it
`over time. TUP also provides a call setup and release that is similar to ISUP, but it has only
`a subset of the capabilities. TUP is not used in North America because its capabilities are
`not sufficient to support the more complex network requirements.
`
`
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`186 Chapter 8: ISDN User Part (ISUP)
`
`Figure 8-2
`
`!SUP at Level 4 of the SS7 Stack
`
`TCAP
`
`I ~UP TUP
`;::::::=======-====;ii
`
`SCCP
`
`.
`
`MTP Layer3
`
`MTP Layer2
`
`MTP Layer 1
`
`As you can see in Figure 8-2, a connection exists between ISUP and both the SCCP and
`MTP3 levels . ISUP uses the MTP3 transport services to exchange network messages, such
`as those used for call setup and clear down. The connection to SCCP is for the transport of
`end-to-end signaling. While SCCP provides this capability, today ISUP end-to-end signal(cid:173)
`ing is usually transported directly over MTP3 . The "Interworking with ISDN" section of
`this chapter further discusses end-to-end signaling and the two different methods using
`MTP3 and SCCP for transport.
`
`ISU P Standards and Variants
`The ITU-T defines the international ISUP standards in the Q.767 and the national standards
`in the Q. 7 61-Q. 7 64 series of specifications. The ITU-T standards provide a basis from
`which countries or geographical regions can define regional or national versions of the pro(cid:173)
`tocol, which are often referred to as variants. For the U.S. network, the following standards
`provide the primary specifications for the ISUP protocol and its use in local and long dis(cid:173)
`tance networks:
`
`• ANSI Tl.113-ANSI ISUP
`• Telcordia GR-246 Telcordia Technologies Specification of Signaling System No. 7,
`Volume 3. (ISUP)
`
`• Telcordia GR-317 LSSGR-Switching System Generic Requirements for Call
`Control Using the Integrated Services Digital Network User Part (ISDNUP)
`• Telcordia GR-394 LSSGR-Switching System Generic Requirements for lnterex(cid:173)
`change Carrier Interconnection (ICI) Using the Integrated Services Digital Network
`User Part (ISDNUP)
`
`
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`

`ISUP Message Flow
`
`187
`
`In Europe, the following ETSI standards provide the basis for the national ISUP variants:
`
`• ETSI ETS 300-121 Integrated Services Digital Network (ISDN); Application of the
`ISDN User Part (ISUP) of CCITT Signaling System No. 7 for international ISDN
`interconnections
`• ETSI ETS 300-156-x Integrated Services Digital Network (ISDN); Signaling System
`No. 7; ISDN User Part (ISUP) for the international interface
`The ETS 300-121 is version 1, and the ETS 300-156-x (where x represents an individual
`document number) is a suite of specifications that covers ETSI ISUP versions 2-4.
`
`A multitude of different country requirements have created many ISUP variants. A few
`of the several flavors are Swedish ISUP, U .K. ISUP, Japanese ISUP, Turkish ISUP, Korean
`ISUP. Each variant is tailored to the specific national requirements. Although not certain of
`the exact number of variants that are in existence today, the author has encountered over a
`hundred different ISUP variants while developing software for switching platforms.
`
`ISUP Message Flow
`This section provides an introduction to the core set of ISUP messages that are used to set
`up and release a call. The ISUP protocol defines a large set of procedures and messages,
`many of which are used for supplementary services and maintenance procedures. While the
`ITU Q.763 ISUP standard defines nearly fifty messages, a core set of five to six messages
`represent the majority of the ISUP traffic on most SS7 networks . The basic message flow
`that is presented here provides a foundation for the remainder of the chapter. Additional
`messages, message content, and the actions taken at an exchange during message
`processing build upon the foundation presented here.
`
`A basic call can be divided into three distinct phases:
`
`• Setup
`• Conversation (or data exchange for voice-band data calls)
`
`• Release
`ISUP is primarily involved in the set-up and release phases. Further ISUP signaling can take
`place if a supplementary service is invoked during the conversation phase.
`
`In Figure 8-3, part A illustrates the ISUP message flow for a basic call. The call is consid(cid:173)
`ered basic because no supplementary services or protocol interworking are involved. The
`next section, "Call Setup;' explains the figure's message timer values.
`
`
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`188 Chapter 8: ISDN User Part (ISUP)
`
`Figure 8-3
`
`Simple !SUP Message Flow
`
`Setup T
`T
`
`T
`T9*
`j_
`
`Cleardown
`
`T
`T1, T5
`~
`
`1AM
`
`1AM
`
`COT (Optional)
`
`COT (Optional)
`
`REL User Bus
`
`RLC
`
`ACM
`
`ANM
`
`Conversation
`
`REL (Normal Clearing)
`
`RLC
`
`* T9 Not Used in ANSI Networks
`
`A. ) Successful Call Setup
`
`B.) Unsuccessful Call Setup
`
`Call Setup
`A simple basic telephone service call can be established and released using only five ISUP
`messages. In Figure 8-3,partA shows a call between SSP A and SSP B. The Initial Address
`Message (1AM) is the first message sent, which indicates an attempt to set up a call for
`a particular circuit. The 1AM contains information that is necessary to establish the call
`connection-such as the call type, called party number, and information about the bearer
`circuit. When SSP B receives the 1AM, it responds with an Address Complete Message
`(ACM). The ACM indicates that the call to the selected destination can be completed.
`For example, if the destination is a subtending line, the line has been determined to be in
`service and not busy. The Continuity message (COT), shown in the figure, is an optional
`message that is used for continuity testing of the voice path before it is cut through to the
`end users. This chapter's "Continuity Test" section discusses the COT message.
`
`Once the ACM has been sent, ringing is applied to the terminator and ring back is sent to
`the originator. When the terminating set goes off-hook, an Answer Message (ANM) is sent
`to the originator. The call is now active and in the talking state. For an ordinary call that
`does not involve special services, no additional ISUP messages are exchanged until one
`of the parties signals the end of the call by going on-hook.
`
`Call Release
`In Figure 8-3, the call originator at SSP A goes on-hook to end the call. SSP A sends a
`Release message (REL) to SSP B. The REL message signals the far end to release the
`
`
`
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`

`

`Message Timers
`
`189
`
`bearer channel. SSP B responds with a Release Complete message (RLC) to acknowledge
`the REL message. The RLC indicates that the circuit has been released.
`
`If the terminating party goes on-hook first, the call might be suspended instead of being
`released. Suspending a call maintains the bearer connection for a period of time, even
`though the terminator has disconnected. The terminator can go off-hook to resume the call,
`providing that he does so before the expiration of the disconnect timer or a disconnect by
`the originating party. This chapter discusses suspending and resuming a connection in more
`detail in the section titled "Circuit Suspend and Resume."
`
`NOTE
`
`Several different terms are used to identify the two parties who are involved in a telephone
`conversation. For example, the originating party is also known as the calling party, or the
`"A" party. The terminating party, or "B" party, are also synonymous with the called party.
`
`Unsuccessful Call Attempt
`In Figure 8-3, part B shows an unsuccessful call attempt between SSP A and SSP B. After
`receiving the IAM, SSP B checks the status of the destination line and discovers that it is
`busy. Instead of an ACM, a REL message with a cause value of User Busy is sent to SSP A,
`indicating that the call cannot be set up. While this example shows a User Busy condition,
`there are many reasons that a call set-up attempt might be unsuccessful. For example, call
`screening at the terminating exchange might reject the call and therefore prevent it from
`being set up . Such a rejection would result in a REL with a cause code of Call Rejected.
`
`NOTE
`
`Call screening compares the called or calling party number against a defined list of numbers
`to determine whether a call can be set up to its destination.
`
`Message Timers
`Like other SS7 protocol levels, ISUP uses timers as a safeguard to ensure that anticipated
`events occur when they should. All of the timers are associated with ISUP messages and
`are generally set when a message is sent or received to ensure that the next intended action
`occurs. For example, when a REL message is sent, Timer Tl is set to ensure that a RLC is
`received within the Tl time period.
`
`ITU Q.764 defines the ISUP timers and their value ranges. In Figure 8-3, part A includes
`the timers for the messages that are presented for a basic call. The "Continuity Test" section
`
`
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`190 Chapter 8: ISDN User Part (ISUP)
`
`of this chapter discusses the timers associated with the optional COT message. Following
`are the definitions of each of the timers in the figure:
`
`• T7 awaiting address complete timer-Also known as the network protection
`timer. T7 is started when an 1AM is sent, and is canceled when an ACM is received.
`If T7 expires, the circuit is released.
`• TS awaiting continuity timer-Started when an 1AM is received with the Continuity
`Indicator bit set. The timer is stopped when the Continuity Message is received. If TS
`expires, a REL is sent to the originating node.
`• T9 awaiting answer timer-Not used in ANSI networks. T9 is started when an ACM
`is received, and is canceled when an ANM is received. If T9 expires, the circuit is
`released. Although T9 is not specified for ANSI networks, answer timing is usually
`performed at the originating exchange to prevent circuits from being tied up for an
`excessive period of time when the destination does not answer.
`• Tl release complete timer-Tl is started when a REL is sent and canceled when a
`RLC is received. If Tl expires, REL is retransmitted.
`• TS initial release complete timer- T5 is also started when a REL is sent, and is canceled
`when a RLC is received. T5 is a longer duration timer than Tl and is intended to pro(cid:173)
`vide a mechanism to recover a nonresponding circuit for which a release has been ini(cid:173)
`tiated. If TS expires, a RSC is sent and REL is no longer sent for the nonresponding
`circuit. An indication of the problem is also given to the maintenance system.
`We list the timers for the basic call in part A of Figure 8-3 to provide an understanding of
`how ISUP timers are used. There are several other ISUP timers; a complete list can be found
`in Appendix H, "ISUP Timers for ANSI/ETSI/ITU-T Applications."
`
`Circuit Identification Codes
`One of the effects of moving call signaling from CAS to Common Channel Signaling (CCS)
`is that the signaling and voice are now traveling on two separate paths through the network.
`Before the introduction of SS7 signaling, the signaling and voice component of a call were
`always transported on the same physical facility. In the case of robbed-bit signaling, they
`are even transported on the same digital time slot of that facility.
`
`The separation of signaling and voice create the need for a means of associating the two
`entities. ISUP uses a Circuit Identification Code (CIC) to identify each voice circuit. For
`example, each of the 24 channels of a Tl span ( or 30 channels of an El span) has a CIC
`associated with it. When ISUP messages are sent between nodes, they always include the
`CIC to which they pertain. Otherwise, the receiving end would have no way to determine
`the circuit to which the incoming message should be applied. Because the CIC identifies a
`bearer circuit between two nodes, the node at each end of the trunk must define the same
`CIC for the same physical voice channel.
`
`
`
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`

`Circuit Identification Codes
`
`191
`
`TIP
`
`Not defining CI Cs so that they match properly at each end of the connection is a common
`cause of problems that occur when defining and bringing new ISUP trunks into service.
`
`ITU defines a 12-bit CIC, allowing up to 4096 circuits to be defined. ANSI uses a larger
`CIC value of 14 bits, allowing for up to 16,384 circuits.
`
`Figure 8-4 shows an ISUP message from SSP A that is routed through the STP to SSP B.
`For simplicity, only one STP is shown. In the message, CIC 100 identifies the physical
`circuit between SSP A and B to which the message applies . Administrative provisioning at
`each of the nodes associates each time slot of the digital trunk span with a CIC . As shown
`in the figure, Trunk 1, time slot (TS) 1 is defined at each SSP as CIC 100. Trunk 1, time slot 2
`is defined as CIC 101, and so on.
`
`Figure 8-4 C/C Identifies the Specific Voice Circuit
`
`STP
`
`SSPA
`
`SSPB
`Trunk 1
`Trunk 1
`TS 1; CIC 100
`TS 1; CIC 100
`TS 2; CIC 101 1+----'-'"--=,........_-.t TS 2; CIC 101
`TS 3; CIC 102
`Digital 1ime slots
`TS 3; CIC 102
`
`Note: TS ; 1ime Slot of the Trunk Span
`
`DPC to CIC Association
`Since each ISUP message is ultimately transported by MTP, an association must be created
`between the circuit and the SS7 network destination. This association is created through
`provisioning at the SSP, by linking a trunk group to a routeset or DPC .
`
`The CIC must be unique to each DPC that the SSP defines. A CIC can be used again within
`the same SSP, as long as it is not duplicated for the same DPC. This means that you might
`see CIC 0 used many times throughout an SS7 network, and even multiple times at the same
`
`
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`192 Chapter 8: ISDN User Part (ISUP)
`
`SSP. It is the combination of DPC and CIC that uniquely identifies the circuit. Figure 8-5
`shows an example of three SSPs that are interconnected by ISUP trunks. SSP B uses the
`same CIC numbers for identifying trunks to SSP A and SSP C. For example, notice that it
`has two trunks using CIC 25 and two trunks using CIC 26. Since SSP A and SSP C are
`separate destinations, each with their own unique routeset defined at SSP B, the DPC/CIC
`combination still uniquely identifies each circuit. SSP B can, in fact, have many other
`duplicate CIC numbers associated with different DPCs.
`
`Figure 8-5 Combination of DPC/CIC Provide Unique Circuit ID
`
`STPA--STPB
`
`200-10-1
`
`200-10-2
`
`200-10-3
`
`ISUPTrunks
`
`ISUPTrunks
`
`SSPB
`
`Trunk 1
`TS 1 = CIC 25
`TS2=CIC26
`
`Trunk2
`TS 1 =CIC25
`TS2=CIC26
`
`Nole: TS= nme Slot of the Trunk Span
`
`Unidentified Circuit Codes
`When a message is received with a CIC that is not defined at the receiving node, an
`Unequipped Circuit Code (UCIC) message is sent in response. The UCIC message's CIC
`field contains the unidentified code. The UCIC message is used only in national networks.
`
`Enbloc and Overlap Address Signaling
`The Called Party Number (CdPN) is the primary key for routing a call through the network.
`When using ISUP to set up a call, the CdPN can be sent using either enbloc or overlap
`signaling. In North America, enbloc signaling is always used; in Europe, overlap signaling
`is quite common, although both methods are used.
`
`
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`

`Enbloc and Overlap Address Signaling
`
`193
`
`Enbloc Signaling
`The enbloc signaling method transmits the number as a complete entity in a single message.
`When using enbloc signaling, the complete number is sent in the 1AM to set up a call. This
`is much more efficient than overlap signaling, which uses multiple messages to transport
`the number. Enbloc signaling is better suited for use where fixed-length dialing plans are
`used, such as in North America. Figure 8-6 illustrates the use of enbloc signaling.
`
`Figure 8-6 Enbloc Address Signaling
`
`1AM
`
`CDPN=9195522000 - - -
`
`ACM
`
`Overlap Signaling
`Overlap signaling sends portions of the number in separate messages as digits are collected
`from the originator. Using overlap signaling, call setup can begin before all the digits have
`been collected. When using the overlap method, the 1AM contains the first set of digits. The
`Subsequent Address Message (SAM) is used to transport the remaining digits. Figure 8-7
`illustrates the use of overlap signaling. Local exchange A collects digits from the user as
`they are dialed. When enough digits have been collected to identify the next exchange,
`an 1AM is sent to exchange B. When tandem exchange B has collected enough digits to
`identify the next exchange, it sends an 1AM to exchange C; exchange C repeats this process.
`After the 1AM is sent from exchange C to exchange D, the destination exchange is fully
`resolved. Exchange D receives SAMs containing the remaining digits needed to identify the
`individual subscriber line.
`
`When using dialing plans that have variable length numbers, overlap signaling is preferable
`because it decreases post-dial delay. As shown in the preceding example, each succeeding
`call leg is set up as soon as enough digits have been collected to identify the next exchange.
`
`As discussed in Chapter 5, "The Public Switched Telephone Network (PSTN);' interdigit
`timing is performed as digits are collected from a subscriber line. When an exchange uses
`variable length dial plans with enbloc signaling, it must allow interdigit timing to expire
`before attempting to set up the call. The exchange cannot start routing after a specific number
`of digits have been collected because that number is variable. By using overlap signaling,
`the call is set up as far as possible, waiting only for the final digits the subscriber dials.
`Although overlap signaling is less efficient in terms of signaling bandwidth, in this situation
`it is more efficient in terms of call set-up time.
`
`
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`194 Chapter 8: ISDN User Part (ISUP)
`
`Figure 8-7 Overlap Address Signaling
`
`Local Exchange
`
`Tandem Exchange
`
`Tandem Exchange
`
`Local Exchange
`
`~
`
`SAM CDPN=2
`
`SAM CDPN=1
`
`SAM CDPN=2
`
`SAM CDPN=3
`
`SAM CDPN=6
`
`SAM CDPN=1
`
`SAM CDPN=6
`
`SAM CDPN=6
`
`SAM CDPN=O
`
`1AM
`
`CDPN=001212
`
`SAM
`
`CDPN=7
`
`SAM
`
`CDPN=3
`
`SAM
`
`CDPN=6
`
`SAM
`
`CDPN=1
`
`SAM
`
`CDPN=6
`
`SAM
`
`CDPN=6
`
`SAM
`
`CDPN=O
`
`1AM CDPN=001212736
`
`SAM
`
`CDPN=1
`
`SAM
`
`CDPN=6
`
`SAM
`
`CDPN=6
`
`SAM
`
`CDPN=O
`
`ACM
`
`ACM
`
`ACM
`
`Circuit Glare (Dual-Seizure)
`Circuit glare (also known as dual-seizure) occurs when the node at each end of a two-way
`trunk attempts to set up a call over the same bearer at the same time. Using ISUP signaling,
`this occurs when an 1AM for the same CIC is simultaneously sent from each end. Each end
`sends an 1AM to set up a call before it receives the 1AM from the other end. You will recall
`from our discussion of the basic ISUP message flow that once an 1AM is sent, an ACM is
`expected. When an 1AM is received after sending an 1AM for the same CIC , glare has
`occurred.
`
`
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`Circuit Glare (Dual-Seizure)
`
`195
`
`Resolving Glare
`When glare is detected, one node must back down and give control to the other end. This
`allows one call to complete, while the other call must be reattempted on another CIC. There
`are different methods for resolving which end takes control. For normal 64-kb/s connections,
`two methods are commonly used. With the first method, the point code and CIC numbers
`are used to determine which end takes control of the circuit. The node with the higher(cid:173)
`numbered point code takes control of even number CI Cs, and the node with the lower-numbered
`point code takes control of odd numbered CICs. This provides a fair mechanism that allows
`each node to control approximately half of the calls encountering glare. In the United
`States, an example of this use would be two peer End Office exchanges. The second method
`of glare resolution is handled by prior agreement between the two nodes about which end
`will back down when glare occurs. One node is provisioned to always back down, while the
`other node is provisioned to take control. A typical example of this arrangement in the U .S.
`network would be a hand-off between non-peer exchanges, such as an IXC to AT. The meth(cid:173)
`od to use for glare resolution can usually be provisioned at the SSP, typically at the granu(cid:173)
`larity level of the trunk group .
`
`Figure 8-8 illustrates a glare condition when SSP A and B have both sent an 1AM before
`receiving the 1AM from the other end. Assuming that the point code/CIC method of resolv(cid:173)
`ing glare is being used, SSP B takes control of the circuit because the CIC is even numbered
`and SSP B has a numerically higher point code.
`
`Figure 8-8 Glare Condition During Call Setup
`
`200-1-2
`
`200-1-8
`
`CIC 100 1AM
`
`CIC 100 1AM
`
`Avoiding Glare
`When provisioning trunks, glare conditions can be minimized by properly coordinating
`the trunk selection algorithms at each end of a trunk group . A common method is to
`perform trunk selection in ascending order of the trunk member number at one end of the
`trunk group, and in descending order at the other end. This minimizes contention to the
`point of selecting the last available resource between the two ends. Another method is to
`have one end use the "Most Idle" trunk selection while the other end uses the "Least Idle"
`selection. The idea is to have an SSP select a trunk that is least likely to be selected by the
`SSP at the other end of the trunk group.
`
`
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`196 Chapter 8: ISDN User Part (ISUP)
`
`Continuity Test
`Continuity testing verifies the physical bearer facility between two SSPs. When CAS
`signaling is used, a call setup fails if the voice path is faulty. Using ISUP signaling, it is
`possible to set up a call using the signaling network without knowing that the bearer
`connection is impaired or completely broken.
`
`The voice and signaling channels are usually on separate physical facilities, so a means of
`verifying that the voice facility is connected properly between the SSPs is needed. Many
`digital voice transmission systems provide fault detection on bearer facilities, which are
`signaled to the connected switching system using alarm indication bits within the digital
`information frame . However, these bits are not guaranteed to be signaled transparently
`through interconnecting transmission equipment, such as a Digital Access Cross Connect
`system (DACS) or digital multiplexers. Some networks require these alarm indications to
`be passed through without disruption, therefore, reducing the need for continuity testing.
`
`Continuity testing can be considered part of the ISUP maintenance functions . It can be
`invoked to test trunks manually, as part of routine maintenance and troubleshooting proce(cid:173)
`dures. Continuity testing can also be provisioned to take place during normal call setup and
`it has an impact on the flow of call processing. During call processing, the originating
`exchange determines whether a continuity test should be performed. Network guidelines
`vary concerning whether and how often continuity testing is performed. The determination
`is typically based on a percentage of call originations. For example, in the United States,
`the generally accepted practice is to perform continuity testing on 12 percent of ISUP call
`originations (approximately one out of eight calls) . This percentage is based on Telcordia
`recommendations.
`
`Loopback and Transceiver Methods
`The actual circuit testing can be performed using either the loopback or the transceiver
`method. The loopback method is performed on four-wire circuits using a single tone, and
`the transceiver method is used for two-wire circuits using two different tones. The primary
`difference between the two methods is related to the action that takes place at the terminat(cid:173)
`ing end. When using either method, a tone generator is connected to the outgoing circuit at
`the originating exchange. Using the loopback method, the terminating exchange connects
`the transmit path to the receive path, forming a loopback to the originator. The originator
`measures the tone coming back to ensure that it is within the specified parameters. When
`the transceiver method is used, the transmit and receive path are connected to a tone trans(cid:173)
`ceiver that measures the tone coming from the originating exchange and sends a different
`tone back to the originating exchange. The tone frequencies vary between countries. The
`following tones are used for the continuity test in North America:
`
`• 2010 Hz from the originating exchange
`• 1720 Hz from the terminating exchange ( transceiver method only)
`
`
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`Continuity Test
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`197
`
`Another example of the COT tone frequency is 2000 Hz, which is used in the U .K.
`
`Continuity Check Procedure
`The Initial Address Message contains a Continuity Check Indicator as part of the Nature
`of Connection field. When an ISUP trunk circuit is selected for an outgoing call and the
`exchange determines that a continuity check should be performed, the Continuity Check
`Indicator is set to true. A tone generator is connected to the outgoing circuit, and the 1AM
`is sent to the SSP at the far end of the trunk. Timer T25 is started when the tone is applied,
`to ensure that tone is received back within the T25 time period. When the SSP at the far end
`receives the 1AM with the Continuity Check Indicator set to true, it determines whether to
`create a loopback of the transmit and receive path, or to connect a transceiver. The trans(cid:173)
`ceiver receives the incoming tone and generates another tone on the outgoing circuit. The
`determination of whether to use a loopback or transceiver is typically based on provisioned
`data at the receiving exchange. Upon receipt of the 1AM, Timer T8 is started at the termi(cid:173)
`nating exchange, awaiting the receipt of a COT message to indicate that the test passed. The
`terminating exchange does not apply ringing to the called party or send back ACM until the
`COT message has been received with a continuity indicator of continuity check successful
`to indicate that the bearer connection is good.
`
`The originating exchange measures the received tone to ensure that it is within an accept(cid:173)
`able frequency range and decibel level. Next it sends a COT message to the terminating
`exchange to indicate the test results. If the test passes, the call proceeds as normal; if the
`test fails, the CIC is blocked, the circuit connection is cleared, and the originating exchange
`sends a Continuity Check Request (CCR) message to request a retest of the failed circuit.
`While ISUP maintenance monitors the failed circuit's retest, ISUP call processing sets the
`call up on another circuit. Figure 8-9 shows a successful COT check using the loopback
`method.
`
`Figure 8-9
`
`Successful COT Check Using the Loopback Method
`
`COT Check Required
`
`1AM
`
`Send/Receive T
`T
`Tone
`Receive Tone 7 COT Check Success
`
`Within
`Acceptable
`Level
`
`ACM
`
`COT r---
`
`Loopback
`Trans/Rev
`
`T
`TB 1
`
`Call Processing
`Continues
`
`
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`198 Chapter 8: ISDN User Part (ISUP)
`
`ISUP Message Format
`The User Data portion of the MTP3 Signaling Information Field contains the ISUP
`message, identified by a Service Indicator of 5 in the MTP3 SIO field. Each ISUP message
`follows a standard format that includes the following information:
`
`• CIC-The Circuit Identification Code for the circuit to which the message is related.
`• Message Type-The ISUP Message Type for the message (for example, an 1AM,
`ACM, and so on).
`• Mandatory Fixed Part-Required message parameters that are of fixed length.
`• Mandatory Variable Part-Required message parameters that are of variable
`length. Each variable parameter has the following form:
`Length of Parameter
`Parameter Contents
`Because the parameter is not a fixed length, a field is included to specify the actual length.
`
`• Optional Part-Optional fields that can be included in the message, but are not
`mandatory. Each optional parameter has the following form:
`Parameter Name
`Length of Parameter
`Parameter Contents
`Figure 8-10 shows the ISUP message structure, as described here. This message structure
`provides a great deal of flexibility for constructing new messages. Each message type
`defines the mandatory parameters that are necessary for constructing a message. The man(cid:173)
`datory fixed variables do not contain length information because the ISUP standards spec(cid:173)
`ify them to be a fixed length. Because the mandatory variable parameters are of variable
`lengths, pointers immediately follow the mandatory fixed part to point to the beginning of
`each variable parameter. The pointer value is simply the number of octets from the pointer
`field to the variable parameter length field.
`
`In addition to the mandatory fields, each message can include optional fields. The last of
`the pointer fields is a pointer to the optional part. Optional fields allow information to be
`included or omitted as needed on a per-message basis. The optional fields differ based on
`variables such as the call type or the supplementary services involved. For example, the
`Calling Party Number (CgPN) field is an optional parameter of the 1AM, but is usually
`included to provide such services as Caller ID and Call Screening.
`
`
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