Many corporations find it advantageous to operate their own voice networks, and they do so by connecting dedicated links between Private Branch eXchanges (PBXs) for inter-office communication, or by using Virtual Private Networks (VPNs) for voice. Originally, PBXs were connected to the Public Switched Telephone Network (PSTN) for voice services, or they were interconnected using analog tie-lines to transfer voice. When the need for more voice trunks and the technology matured, analog tie-lines were replaced with higher-speed digital facilities capable of accessing sophisticated and feature-rich networks. This chapter analyzes the signaling techniques that traverse analog and digital facilities in corporate and interexchange networks.
This chapter also discusses channel-associated signaling (CAS) systems, such as Bell System MF, Consultative Committee for International Telegraph and Telephone (CCITT) No. 5, R1, and R2, and it reviews how these CAS systems operate.
It also describes access protocols, such as Integrated Services Digital Network (ISDN), Q Signaling (QSIG), and Digital Private Network Signaling System (DPNSS). These protocols deliver PBX signaling through a network to distant PBXs. Private ISDN networks use the PSTN for connectivity and services. QSIG is an inter-PBX signaling system similar to ISDN that enables corporate PBXs to connect, thus creating a private voice network. DPNSS is an ISDN-type protocol that enables PBX connectivity; however, it is not as widely used as ISDN and QSIG.
Before covering signaling methods and standards, it’s important to discuss some basic concepts. These basic concepts are applied in the individual signaling methods further along in the chapter.
Originally, PBXs were connected by simple analog lines that enabled the transmission of voice-band information. Analog systems are not as common today as they used to be, however, and in many cases, they have been replaced by higher-speed digital facilities that cost less than their analog counterparts.
Digital signaling is the most common type of telephony signaling used in today’s corporate and service provider networks. In digital networks, many forms of signaling techniques are used.
One form is robbed-bit signaling. With this method, a bit is “robbed” from designated frames to use for signaling purposes. Robbed-bit signaling inserts the signaling information into the digital voice stream without affecting voice quality. This signaling technique is discussed in more detail in the “CAS” section later in this chapter. In addition to CAS, other digital protocols include R1, R2, ISDN, QSIG, and DPNSS.
This form of signaling relies on direct current (DC) to signal the end switch or office. DC signaling indicates transition state changes by toggling on or off the flow of DC. These end office switches use current detectors to identify changes in state. DC signaling is used in the following two signaling arrangements:
Subscriber Loop. —This is a simple form of DC signaling between the subscriber and the local end office. When a subscriber goes off-hook, DC (-48V) flows across the line or loop between the telephone and the local end office switch. Line cards in the local office are equipped with current detectors to determine when a connection is being requested. When a subscriber goes on-hook, the capacitor in the telephone blocks the flow of current.
Similarly to off-hook, the change in DC signals to the end office switch that the call was terminated. In this case, the same pair of wires is used to provide the voice and signaling path.
recEive and transMit (E&M). —This trunking arrangement uses a form of DC signaling to indicate state changes on trunks or tie-lines. With E&M, two leads—one called “E” and the other called “M”—are dedicated to signaling. You can detect the toggling of E&M leads by applying either ground (earth) or a voltage potential (magneto). This form of signaling is covered in the “E&M Signaling” section later in this chapter.
DC signaling has some limitations. Signaling is limited to the number of states you can represent by DC, for instance. Also, when you use the same pair of wires for voice and signaling, the lines or trunks are kept busy even when the two subscribers are not connected.
In-band signaling uses tones in place of DC. These tones are transmitted over the same facility as voice and, therefore, are within the 0–4kHz voice band. The tones include Single Frequency, Multi-Frequency (MF), and Dual-Tone Multi-Frequency (DTMF), described here:
Single Frequency—This tone is used for interoffice trunks and has two possible states: onhook or idle, and off-hook or busy. The Single Frequency tone is based on a single frequency of 2600 Hz and is used to identify a change in state. Therefore, no tone is present when a connection or circuit is up. When either party hangs up, however, a 2600 Hz tone is sent over the circuit, notifying all interoffice exchanges of the disconnect.
At one time, the Single Frequency tone was used to gain fraudulent long-distance services from service providers. The perpetrator attached a “blue box” to the subscriber line and used it to fool interoffice exchanges into interpreting the 2600 Hz tone as a clear-forward signal. The interoffice switch then accepted the called party number and believed that the local switch would charge for the call. Access to the interoffice switch was accomplished by dialing 0 and fooling the interoffice switch before the operator answered. Service providers eventually curbed this activity by implementing certain protective measures.
MF—This tone is used by interoffice trunks to indicate events, such as seizure, release, answer, and acknowledge, and to transmit information, such as the calling party number. MF signaling uses a combination of pulses specified by frequencies to signal across a network. These frequencies are system-specified and are covered in more detail in the “CAS,” “R1,” and “R2” sections later in this chapter. MF signaling uses the same facilities as the voice path and, therefore, is less efficient than common channel signaling (CCS) systems, such as Signaling System 7 (SS7).
DTMF—This form of addressing is used to transmit telephone number digits from the subscriber to the local office. With the development of DTMF came the replacement of transistor oscillators in telephones with keypads and dual-tone oscillators. DTMF tones identify the numbers 0 through to 9 and the “*” and “#” symbols. When a subscriber presses one of these keys, the oscillator sends two simultaneous tones. Digits are represented by a particular combination of frequencies: one from the low group (697, 770, 852, and 941 Hz) and one from the high group (1290, 1336, 1447, and 1633 Hz). Sixteen possible combinations exist; however, only 12 are implemented on the keypad.
The two most common methods for end-loop signaling are loop-start and ground-start signaling.
Loop-Start Signaling. —This is the simplest and least intelligent of the two signaling protocols. It also is the most common form of subscriber loop signaling. This protocol basically works in the same way as the telephone and the local end office, whereby the creation of a loop initiates a call and the closure of a loop terminates a call. Loop-start signaling is not common for PBX signaling and has one significant drawback, in that glare can occur. Glare occurs when two endpoints try to seize the line at the same time, and it often results in two people being connected unknowingly. The person picking up the phone thinks he has a dial tone, but unbeknownst to him he is connected to someone who called him.
Ground-Start Signaling. —This signaling protocol differs from loop-start signaling, in that it provides positive recognition of connects and disconnects. Current-detection mechanisms are used at each end of the trunk, enabling end office switches to agree on which end is seizing the trunk before it is seized. This form of signaling minimizes the effect of glare and costs the same as loop-start signaling. As such, it is the preferred signaling method for PBXs.
CAS exists in many networks today. CAS systems carry signaling information from the trunk in the trunk itself. CAS systems were originally developed by different equipment vendors and, therefore, exist in many versions or variants. Today’s telecommunication networks require more efficient means for signaling, however, so they are moving to common channel-type systems, such as CCS.
CCS uses a common link to carry signaling information for a number of trunks. This form of signaling is cheaper, has faster connect times, and is more flexible than CAS. The first generation of CCS is known as SS6; the second generation, SS7, is the basis of Chapter 4, “Signaling System 7.”
E&M is a common trunk-signaling technique used on telephony switches and PBXs. The signaling and voice trunks in E&M are separated. In E&M, voice is transmitted over either two or four-wire circuits, with six methods for signaling. E&M signaling methods are referred to as Types I, II, III, IV, and V; they also are known by the British Telecom (BT) standard, SSDC5.
The remainder of this section focuses on four-wire E&M Types I through V. E&M lead conditions for off-hook and on-hook for Types I through V are summarized in Table 3-1.
With the Type I interface, the trunk equipment generates the E signal to the PBX by grounding the E lead (shown in Figure 3-1). The PBX detects the E signal by sensing the increase in current through a resistive load. Similarly, the PBX generates the M signal by sourcing a current to the trunk equipment, which detects it through a resistive load. The numbers 7, 2, 6, and 3 are the pinouts used on an RJ-48c connector.
E&M Type II has two additional leads over Type I: signal battery (SB) and signal ground (SG). In this method, the E lead is paired up with the SG lead, and the M lead is paired up with the SB lead. An on-hook at the PBX end is indicated when the E and M leads are open. Alternatively, an off-hook is indicated when the E lead is grounded and the M lead is providing battery (see Figure 3-2).
E&M Type III is used mostly in older telephone company switching centers. Figure 3-3 shows the Type III setup.
E&M Type IV is similar to E&M Type II; however, from the PBX side, an on-hook occurs when the E and M leads are open, and an off-hook occurs when both leads are at ground (see Figure 3-4).
Under E&M Type V, both the PBX and the switching endpoint supply battery (see Figure 3-5). At the PBX, battery is supplied on the E lead, and at the endpoint it is supplied on the M lead. Type V is the most common method of E&M signaling outside North America.
CAS exists in many varieties that operate over various analog and digital facilities. The analog facilities are either two- or four-wire and the digital facilities are either North American T1 or European E1. This section discusses Bell System MF, CCITT No. 5, R1, and R2 CAS systems.
The main areas of discussion for each CAS system are supervision signaling and address signaling over analog and digital facilities. Bell System uses in-band MF for address signaling. For supervision signaling it uses Single Frequency for analog and a/b bits for digital trunks. CCITT No. 5 was designed for analog trunks and uses different MF signals for supervision and address signaling. In-band tone detection is used to detect and interpret the MF signals.
It is important to cover a few points before proceeding with a discussion of CAS systems. When a call is placed from Exchange A toward Exchange B, Exchange A is considered the outgoing exchange and Exchange B the incoming exchange.
One-way trunks are trunks on which only Exchange A or Exchange B can initiate a call. Exchanges A and B can initiate a call over two-way trunks. Double seizures can occur over twoway trunks when both exchanges try to seize the trunk at the same time, however. When this occurs, mechanisms such as timers are used to detect and resolve such events.
Three groups of signals are present in channel-associated interexchange signaling systems:
Supervision Signals. —These signals represent events that occur on a trunk and can be specific to the CAS variant. Signals include seizure, wink, and answer; they also are referred to as line signals.
Address Signals. —These signals typically represent the digits dialed or called party number and, in some instances, other information. In this chapter, address signals are based on MF signaling and can be system- or variant-specific.
Tones and Announcements. —These include tones such as ringing and busy tones and announcements such as, “The number you have dialed is no longer in service.”
One more concept to cover before moving forward is that of service circuits. Service circuits are used in most exchanges to send and receive address signals and tones, as well as to play announcements. These circuits are typically system-specific; the processor connects a path from the trunk to the appropriate service circuit inside the switch. The pools of service circuits are temporarily used to send and receive tones or to play announcements.
This section introduces the MF signaling systems developed by Bell System in the 1950s. The Bell System is still used today in local networks in the United States and is nearly identical to the R1 signaling system discussed later in this chapter.
With Bell System MF signaling, which you can use on one-way or two-way trunks, supervision and address signaling are signaled link-by-link. Supervision signaling is accomplished through a Single Frequency tone for analog facilities and through robbed-bit signaling for digital facilities. Address information is sent through MF tones.
Supervision signals are continuously sent by endpoint exchanges indicating the state of the trunk. This is known as continuous two-state signaling. States can be different at each endpoint of the trunk. MF signaling is used to indicate on-hook and off-hook states, as listed in Table 3-2.
Supervision signals operate slightly differently for analog and digital trunks.
A Single Frequency 2600 Hz tone is used to indicate trunk state between exchanges over analog facilities. This tone is applied in-band over the trunk and is turned off when a call is in progress or established. Therefore, the state is on-hook or idle when the tone is present and off-hook or in use when the tone is absent. The supervision signals for the Single Frequency method are illustrated in Figure 3-6.
In Figure 3-6, assume that Switch A sends the forward signals and Switch B sends the backward signals. Switch A sends a forward seizure or off-hook signal to Switch B on a chosen trunk. Then, Switch B sends a backward wink or proceed-to-send to Switch A and waits for address signaling or dialed digits. After the digits are sent and the call is answered, Switch B sends a backward answer or off-hook to Switch A, enabling an end-to-end voice path.
In this case, the calling party hangs up first and a clear-forward is sent from Switch A to Switch B. When the called party hangs up, a clear-back signal is sent by Switch B.
Two important aspects of this signaling method need to be discussed:
First, Bell System MF does not have backward signaling for connections that fail during setup. Therefore, the exchange where the call failed must connect an announcement server indicating to the calling party that a problem occurred.
The signaling system then relies on the calling party to release or drop the call so that clear-forward procedures can be initiated.
Second, no release guard-type signal exists, and timers are used after trunks are released. Therefore, after an exchange releases a trunk, it initiates a timer for approximately 1 second. After this timer expires, the exchange assumes that the trunk was released at the other end and is available for use.
The digital trunks most commonly used today are either T1 or E1 facilities (as described in the “Physical Layer—MTP L1” section of Chapter 4). With digital trunks, bits are robbed from specific frames and are used for signaling purposes. This discussion focuses on T1 digital trunks.
T1 has two types of framing formats: Super Frame (SF) and Extended Superframe (ESF). The least significant bits are robbed from frames 6 and 12 for SF and frames 6, 12, 18, and 24 for ESF. These bits are referred to as the Sa and Sb bits for SF, and the Sa, Sb, Sc, and Sd bits for ESF. Robbing these bits has a negligible effect on voice quality.
The SF signaling bits—Sa and Sb—are equal to each other and provide two-state, continuous supervision signaling. Bit values of zero are used to indicate on-hook, and bit values of 1 are used to indicate off-hook.
Address signaling is used to indicate the called and calling number as well as to identify the start and end of the address information. In the Bell System MF method, address signals are a combination of two voice-band frequencies chosen from six different frequencies, as illustrated in Table 3-3.
Table 3-3. Bell System MF Address Signals
Signal | Frequencies in Hz |
|---|---|
Digit 1 | 700 and 900 |
Digit 2 | 700 and 1100 |
Digit 3 | 900 and 1100 |
Digit 4 | 700 and 1300 |
Digit 5 | 900 and 1300 |
Digit 6 | 1100 and 1300 |
Digit 7 | 700 and 1500 |
Digit 8 | 900 and 1500 |
Digit 9 | 1100 and 1500 |
Digit 0 | 1300 and 1500 |
KP (start) | 1100 and 1700 |
ST (end) | 1500 and 1700 |
The address signaling sequence is initiated with a KP or start-of-pulsing signal and terminated with an ST or end-of-pulsing signal. Two important timing intervals exist:
The KP signal’s duration is from 90 to 110 ms, and the ST signal’s duration is from 61 to 75 ms. The silent interval between signals also is from 61 to 75 ms. Figure 3-7 demonstrates supervision and address signaling sequences.
Address signaling uses two other key information digits. The codes in this information (or I bits) indicate the calling number or Automatic Number ID (ANI), as well as operator services (see Table 3-4).
The information codes are sent after the KP signal and before the called party number. I codes 02 and 06 identify that operator assistance is required to proceed with these calls.
The CCITT adopted the CCITT No. 5 signaling system in the 1960s for use in international networks. This signaling system is still used today, usually on long international trunks and, in some cases, over transoceanic and satellite links. This signaling system was designed to operate over analog trunks equipped with Time Assignment Speech Interpolation (TASI). TASI is similar to voice activity detection (VAD), in that it enables unused bandwidth (silences or pauses in speech) to be used for other phone conversations. Link-by-link and in-band signaling are used for both supervision and address signaling.
Supervision signaling is accomplished by two frequencies, sent either individually or in combination. CCITT No. 5 uses compelled supervision signaling, whereby the signaling tone is left on until an acknowledgment is received.
The two in-band frequencies are f1, which equals 2400 Hz, and f2, which equals 2600 Hz. The combination of f1 and f2 produces a composite signal; these signals and frequencies are listed in Table 3-5.
Table 3-5. CCITT No. 5 Supervision Signals
Direction | Signal Type | Frequency |
|---|---|---|
Forward | Seizure | f1 |
Backward | Proceed-to-send | f2 |
Backward | Answer | f1 |
Forward | Acknowledgment | f1 |
Backward | Clear-back | f2 |
Forward | Acknowledgment | f1 |
Forward | Clear-forward | f1 and f2 |
Backward | Release-guard | f1 and f2 |
Backward | Busy-flash | f2 |
Forward | Acknowledgment | f1 |
Forward | Forward-transfer | f2 |
Three new signals are introduced in Table 3-5: Release-guard, Busy-flash, and Forward-transfer.
Release-guard. —This signal is used by the incoming exchange to acknowledge a clear-forward from the outgoing exchange. It also indicates to the outgoing exchange that the trunk is now available for an incoming call.
Busy-flash. —This signal is used by the incoming exchange to indicate to the outgoing exchange that call setup cannot be extended toward the destination.
Forward-transfer. —This signal is used on calls for operator services.
In CCITT No. 5, address signaling is based on the combination of two frequencies, as illustrated in Table 3-6. The address signaling sequence starts with KP1 for national numbers and KP2 for international numbers. Codes 11 and 12 are used to connect international operator services.
Table 3-6. CCITT No. 5 Address Signals
Signal | Frequencies in Hz |
|---|---|
Digit 1 | 700 and 900 |
Digit 2 | 700 and 1100 |
Digit 3 | 900 and 1100 |
Digit 4 | 700 and 1300 |
Digit 5 | 900 and 1300 |
Digit 6 | 1100 and 1300 |
Digit 7 | 700 and 1500 |
Digit 8 | 900 and 1500 |
Digit 9 | 1100 and 1500 |
Digit 0 | 1300 and 1500 |
Code 11 | 700 and 1700 |
Code 12 | 900 and 1700 |
KP1 | 1100 and 1700 |
KP2 | 1300 and 1700 |
ST | 1500 and 1700 |
The CAS system known as R1 is available in the International Telecommunication Union Telecommunication Standardization Sector (ITU-T) Q.310 to Q.332 specifications. This signaling system is almost identical to Bell System MF signaling and, therefore, is not further discussed.
R2 signaling is a CAS system developed in the 1960s that is still in use today in Europe, Latin America, Australia, and Asia. Originally known as multi-frequency code (MFC) signaling, R2 signaling exists in several country versions or variants and in an international version called CCITT-R2.
R2 signaling operates over two- or four-wire analog and digital trunks and does not operate over TASI-equipped trunks or satellite links. R2 signaling is more suitable for relatively short international trunks. One of the differentiating aspects of this system compared to R1 is its register or inter-register signaling.
This section focuses on supervision and inter-register signaling for CCITT-R2 and National R2 signaling systems.
For the purposes of supervision signaling on analog trunks, this section covers operation over four-wire trunks. The transmission path is divided into two parts: a 300- to 3400 Hz voice-band and a 3825Hz narrow-band for signaling. In this method, filters separate the signaling tone from the voice path. This is considered out-of-band signaling, even though signaling is over the same facility.
CCITT-R2 uses the tone-on-idle signaling supervision method; National R2 uses pulse signaling.
This method is commonly used on one-way trunks, is tone-on-idle, and provides two-state signaling. The forward and backward signals and transition states are similar to Bell System MF signaling and are illustrated in Table 3-7.
National R2 signaling has many country variants. Most versions of National R2 use pulse outof-band supervision signals, however. Examples of National R2 supervision signals are illustrated in Table 3-8.
R2 signaling operates over E1 digital facilities (described in the “Physical Layer—MTP L1” section of Chapter 4). E1 has 32 time-slots, numbered TS0 to TS31, whereby TS1–TS15 and TS17–TS31 are used to carry voice encoded with pulse code modulation (PCM), or to carry 64 kbps data.
Sixteen consecutive frames are in the SF format, and they are numbered 0 to 15. TS16 in frame 0 is used for SF alignment, and TS16 in the remaining frames (1–15) is used for trunk signaling. Four status bits are used from TS16 for signaling. They are called a, b, c, and d.
In the case of CCITT-R2 signaling, only the a and b bits are used. The c and d bits are set to 0 and 1, respectively. An idle state is denoted when a and b are equal to 1 and 0. Signaling is continuous. For two-way trunks, the supervision roles for forward and backward signaling vary on a call-by-call basis. Table 3-9 illustrates the R2 supervision signal, transition, and direction used on digital trunks.
The concept of address signaling in R2 is slightly different from that used in the other CAS systems previously discussed. In the case of R2, the exchanges are considered registers, and the signaling between these exchanges is called inter-register signaling. Inter-register signaling uses forward and backward in-band MF signals to transfer called and calling party numbers as well as the calling party category.
In this case, signaling is compelled because the registers in the outgoing and incoming exchanges hold the signal until an acknowledgment is received. The signals consist of two voice-band frequencies and are listed in Table 3-10.
Table 3-10. CCITT-R2 and National R2 Inter-Register Signal Frequencies
Signal | Forward Frequency in Hz | Backward Frequency in Hz |
|---|---|---|
Digit 1 | 1380 and 1500 | 1140 and 1020 |
Digit 2 | 1380 and 1620 | 1140 and 900 |
Digit 3 | 1500 and 1620 | 1020 and 900 |
Digit 4 | 1380 and 1740 | 1140 and 780 |
Digit 5 | 1500 and 1740 | 1020 and 780 |
Digit 6 | 1620 and 1740 | 900 and 780 |
Digit 7 | 1380 and 1860 | 1140 and 660 |
Digit 8 | 1500 and 1860 | 1020 and 660 |
Digit 9 | 1620 and 1860 | 900 and 660 |
Digit 0 | 1740 and 1860 | 780 and 660 |
Not used | 1380 and 1980 | 1140 and 540 |
Not used | 1500 and 1980 | 1020 and 540 |
Not used | 1620 and 1980 | 900 and 540 |
Not used | 1740 and 1980 | 780 and 540 |
End of # | 1860 and 1980 | 660 and 540 |
In R2 signaling, the forward and backward signals can have different meanings depending on which group is used. Three groups of forward signals and two groups of backward signals exist. The forward groups are I, II, and III, and the backward groups are A and B.
Group I. —These forward signals represent the called party number or dialed digits.
Group II. —These forward signals identify the calling party category.
Group III. —These forward signals represent the digits of the calling party number.
Group A. —These backward signals indicate if the signaling ended or if a particular forward signal is required.
Group B. —These backward signals are sent by the terminating switch to acknowledge a forward signal, or to provide call charging and called party information.
The following inter-register group sequence rules are used to identify the signal’s group:
The initial signal received by the incoming exchange is a Group I signal.
Outgoing exchanges consider backward signals as Group A signals.
Group A signals received by outgoing exchanges are used to identify whether the next signal is a Group B signal.
Group B signals always indicate an end-of-signaling sequence.
ISDN has been available to the public since the 1980s. International Telecommunication Union (ITU; formerly CCITT) I series recommendations define the international standards for ISDN. This subscriber or user-based interface protocol provides single access to multiple services.
ISDN signaling is compatible with SS7 and inter-works with the ISDN User Part (ISUP) protocol. This inter-working enables ISDN subscribers to access the same services and intelligence as they can on the SS7 network. ISDN also enables PBXs to connect over the PSTN and create VPNs. This is accomplished by delivering PBX signaling over the network to distant PBXs.
The ISDN suite defines the specifications for access to the network. The following list outlines some ISDN functions and capabilities:
ISDN provides circuit-based (voice and data) communications and packet-based communications to its users.
Many new services can be extended to users.
ISDN includes two access methods: Basic Rate Interface (BRI) and Primary Rate Interface (PRI).
ISDN includes single access for PSTN, Direct-Inward-Dial (DID), Direct-Outward-Dial (DOD), 800, Foreign Exchange (FX), tie-lines, packet-switched data, circuit-switched data, and dedicated data.
ISDN is capable of adding additional channels for high-speed data communications.
ISDN is capable of transmitting voice and data on the same facility.
ISDN uses separate channels for signaling.
ISDN signaling is compatible with SS7.
ISDN enables the creation of VPNs.
The following communication services are available in circuit-switched ISDN networks:
Bearer Services. —Three types of bearer services are available for a call. They include speech, 3.1 kHz audio (for modem data), and 64 kbps digital data.
Bearer services are specified by the calling user in the call setup message and are transferred over the network to the called user. The exchanges within the network also use this information when selecting the appropriate outgoing trunk. In the case of speech, exchanges can use analog or digital trunks for interconnection, whereas 64 kbps digital data requires digital trunks.
Teleservice. —This service enables the calling user to specify the type of data service for 3.1 kHz audio and 64 kbps digital data. The teleservice information (fax, telex, and so on) is transmitted transparently across the network to the called user. The called user processes the information to select the appropriate terminal equipment (TE) function to terminate the incoming call.
Supplementary Services. —The ISDN service offering also provides many supplementary services. These services also are typically found on PBXs and virtual private voice networks. The following are examples of supplementary services:calling line identification (caller ID), closed users groups, call waiting, user-to-user signaling, advice of charge, call forward, and call hold. When a user requests these services, supplementary service messages are sent to the network to invoke the requested processes. In the case of user-to-user signaling, the two ISDN users send signaling information transparently during the call setup and teardown parts of the call.
Before discussing ISDN access methods, it is important to cover the concept of B and D channels:
B Channel. —The B channel is a 64 kbps channel that carries user information streams. No signaling information is carried in the B channel. B-channel user streams include speech encoded at 64 kbps according to ITU G.711, data at or less than 64 kbps, and voice encoded at lower bit rates.
D Channel. —The D channel is used primarily to carry signaling for circuit switching by ISDN networks. D-channel bit rates are different depending on the access method. The D channel also is capable of transmitting user packet data up to 9.6 kbps.
Two types of access methods exist for ISDN:
BRI
PRI
BRI delivers two bi-directional 64 kbps B channels and one bi-directional 16 kbps D channel over standard two-wire telephone lines. Basic rate ISDN service typically is used for residential and small office, home office (SOHO) applications. Each B channel can transmit speech or data; the D channel transmits the signaling or call control messages.
The configuration and reference points for BRI are specified in Figure 3-8.
The reference configuration for ISDN is defined in the ITU specification I.411. The reference points specify the transmission medium, interface, and connectors (if used).
U Reference Point. —The U reference point specifies the transmission characteristics of the local loop. For BRI, this two-wire interface operates at 160 kbps (2B + D + 16 kbps for overhead) over standard copper-twisted wires.
S/T Reference Point. —For basic rate access, this interface provides a four-wire connection to ISDN-compatible terminals or terminal adapters. The interface operates at 144 kbps (2B + D) between the ISDN device and the network termination device. You can connect up to eight ISDN devices to the S/T interface.
R Reference Point. —The R reference point provides connection for non-ISDN devices. Such devices connect to the terminal adapter using interfaces such as RS-232 and V.35.
This reference configuration also specifies the set of functions required to access ISDN networks:
Network Termination 1 (NT1)—Outside the United States, NT1 is on the network side of the defined user-network interface and is considered part of the service provider network. NT1s terminate the two-wire local loop and provide four-wire S/T bus for ISDN terminal equipment (TE).
TE1—TE1s are ISDN-compatible devices that connect directly to the S/T connector on the NT1.
TE2—TE2s are non-ISDN compatible devices that require terminal adapter (TA) interconnection.
TA—TAs provide an ISDN-compliant interface to NT1s and standard interfaces for TE2s. These standard interfaces include RS-232, V.35, RS-449, and X.21.
PRI corresponds to two primary rates: 1.544 Mbps (T1) and 2.048 Mbps (E1). PRIs typically are used in medium to large business applications. PRI is comprised of B channels and one 64 kbps D channel. The interface structure for T1 is 23B + D (North America and Japan). The interface structure for E1 is 30B + D (Europe).
The configuration and references for PRI are specified in Figure 3-9.
The configuration and reference points for PRI are similar to those for BRI. The differences between the two reference models are discussed here.
U Reference Point. —For PRI, the U interface is four-wire and operates at either T1 (1.544 Mbps) or E1 (2.048 Mbps) PRI rates.
T Reference Point. —For PRI, the T interface provides access to the Network Termination 2 (NT2) device.
NT2. —PBX equipment can provide such NT2 functions as Layer 2 (L2) and Layer 3 (L3) protocol handling as well as multiplexing, switching, interface termination, and maintenance. NT2s also can provide connections to ISDN-compatible TE1s and non-ISDN compatible TE2s.
ISDN user-network interface L2 and L3 specifications also are referred to as Digital Subscriber Signaling System No. 1 (DSS1). L2 provides error-free and secure connections for two endpoints across the ISDN reference configuration. L3 provides the mechanism for call establishment, control, and access to services. The L2 protocol for ISDN is Q.920/921, and the L3 protocol is Q.930/931. Q.932 enables general procedures for accessing and controlling supplementary services.
The specifications for L2 are referred to as Link Access Procedures on the D channel (LAPD). This protocol provides the reliable transfer of frames between the local exchange and the TE. The specifications for Q.920 and Q.921 are extensive and are available from the Q series of ITU recommendations.
The specifications for L3 define the messages that pass between the local exchange and the TE. These messages are used for call setup, call supervision, call teardown, and supplementary services. The next section discusses the specifics of ISDN messaging.
The message structure and signaling elements of Q.931 are used in ISDN networks to provide call control capabilities. Q.931 messages are sent from the network to the user and from the user to the network. They are referred to as user-network and network-user messages, as illustrated in Tables 3-11 and 3-12.
Table 3-11. Q.931 Messages and Type Codes
Q.931 Message Type | Message Type Value |
|---|---|
Setup message (SETUP) | 00000101 |
Setup acknowledgment message (SETACK) | 00001101 |
Call proceeding message (CALPRC) | 00000010 |
Progress message (PROG) | 00001111 |
Alerting message (ALERT) | 00000011 |
Connect message (CONN) | 00000101 |
Connect acknowledgment message (CONACK) | 00000111 |
Disconnect message (DISC) | 01000101 |
Release message (RLSE) | 01001101 |
Release complete message (RLCOM) | 01011010 |
Information message (INFO) | 01111011 |
Table 3-12. User to Network—Information Elements in Q.931 Messages
Information Elements | SETUP | CALPC | ALERT | CONN | CONAK | DISC | RLSE | RLCOM | INFO |
|---|---|---|---|---|---|---|---|---|---|
Bearer Capability | M | ||||||||
Called party number | M | ||||||||
Calling party number | O | ||||||||
Called party subaddress | O | ||||||||
Calling party subaddress | O | ||||||||
Cause | M | O | O | ||||||
Channel identification | O | M | O | O | |||||
High-layer compatibility | O | ||||||||
Keypad | O | M | |||||||
Low-layer compatibility | O | ||||||||
Transit network selection | O | ||||||||
User-to-user information | O | O | O | O | O | O |
Some of the most important Q.931 messages are listed in Table 3-11. The message type field in the general format of the Q.931 message is used to determine the type of message being sent.
The information or signaling elements of each message type are listed in Table 3-12. Table 3-12 also indicates the mandatory (M) and optional (O) fields for each network-to-user message.
Table 3-13 indicates the mandatory (M) and optional (O) fields for each network-to-user message.
Table 3-13. Network to User—Information Elements in Q.931 Messages
SETUP | SETACK | CALPC | PROG | ALET | CONN | CONAK | DISC | RLSE | RLCOM | INFO | |
|---|---|---|---|---|---|---|---|---|---|---|---|
Bearer Capability | M | ||||||||||
Called party number | M | ||||||||||
Calling party number | O | ||||||||||
Called party subaddress | O | ||||||||||
Calling party subaddress | O | ||||||||||
Cause | O | M | O | O | |||||||
Channel identification | M | M | O | ||||||||
High-layer compatibility | OO | ||||||||||
Low-layer compatibility | O | ||||||||||
Progress indicator | O | O | M | O | O | ||||||
Signal | M | O | O | M | O | O | O | O | O | O | |
User-to-user information | O | O | O | O | O | O |
This section outlines a typical ISDN call between two users served by the same local exchange. The signaling sequence between User A (TE-A), the local exchange, and User B (TE-B) is illustrated in Figure 3-10.
TE-A initiates the call by sending a SETUP message to TE-B. The SETUP message contains the complete called party number (also known as the en-bloc signal). The local exchange then sends a SETUP message to TE-B and includes in the message the B-channel assignment.
At this point, the local exchange sends a CALPRC message to TE-A indicating that the call setup started. If TE-B accepts the incoming call, an ALERT message is returned. The local exchange then sends an ALERT to TE-A, and if this is a speech call, a ringing tone is applied to the B channel.
When TE-B answers the call, a CONN message is sent to the exchange where the B channels are connected; a CONN message also is sent to TE-A. The local exchange acknowledges TE-B’s CONN with a CONACK; TE-A also can acknowledge the CONN with a CONACK.
Consider the example in which TE-B is the first to initiate a disconnect. TE-B sends a DISC, and the local exchange then sends a DISC to TE-A.
At this point, the local exchange clears the B channel to TE-B and sends an RLSE message to TE-B. Next, TE-B releases the endpoint B channel and sends an RLCOM message. The same release procedure also occurs between TE-A and the local exchange.
QSIG is a peer-to-peer signaling system used in corporate voice networking. Internationally, QSIG is known as Private Signaling System No. 1 (PSS1). This open standard is based on the ITU-T Q.9XX series of recommendations for basic service and supplementary services. Therefore, as well as providing inter-PBX communications, QSIG is compatible with public and private ISDN.
QSIG also has one important mechanism known as Generic Functional Procedures (QSIG GF). This mechanism provides a standard method for transporting features transparently across a network.
The following are attributes of the QSIG global signaling system:
It is a standards-based protocol enabling the interconnection of multivendor equipment.
It enables inter-PBX basic, feature transparency, and supplementary services.
It is interoperable with public and private ISDNs.
It operates in any network configuration (star, mesh, and so on) and is compatible with many PBX-type interfaces.
It does not impose restrictions on private numbering plans.
QSIG is an important signaling system. The remainder of this section covers the following key aspects of QSIG:
Services
Architecture and Reference Points
Protocol Stack
Basic Call Setup and Teardown
QSIG supports a suite of services and features for corporate PBX networks. The three main service groups include basic services, generic functional procedures, and supplementary services.
Basic service (QSIG BC)—This service provides the capabilities to set up, manage, and tear down a call. Similar to an ISDN bearer service, basic services include speech, 3.1 kHz audio, and 64 kbps unrestricted.
QSIG GF—This is a standardized method for transporting nonstandard features, thus providing feature transparency. This mechanism enables the exchange of signaling information for the control of supplementary and additional network features over a corporate network.
Supplementary services—This category includes services and additional network features (ANFs). Supplementary services and ANFs include call completion, call forward, call diversion, call transfer, call waiting, caller ID, and advice of charge.
It is necessary to extend the ISDN reference model to include PBX-to-PBX signaling for corporate networks. To accommodate these two new reference points, “Q” and “C” were identified by the standard, as illustrated in Figure 3-11.
The Q reference point defines the logical signaling between PBXes, and the C reference point identifies the physical interconnection. A corporate network can have dedicated analog or digital channels, or it can have VPN switched connections. Typically, it is assumed that a T1 or E1 digital interface is used to connect to the network. QSIG end-to-end signaling is maintained from PBX to PBX, and ISDN and ISUP inter-working is critical for end-to-end signaling in the ISDN network. As mentioned previously, QSIG is compatible with ISDN; these reference points also are noted in Figure 3-11.
The T reference point defines access to the NT2 device for ISDN PRI. The C reference point is the physical interconnection point to the PBX. It is compatible with many interfaces, including twoand four-wire analog, BRI, PRI, and radio and satellite links. The Q reference point specifies the logical signaling point between two PBXs. This reference point is used to specify signaling-system and related protocols.
The QSIG protocol stack specifies a signaling system at reference point Q and is illustrated in Table 3-14. QSIG has an identical structure to that of ISDN, and at L1 and L2, these protocols can be the same. They differ at L3, however, where QSIG is split into the following three sublayers:
QSIG BC—With this symmetrical protocol, the interfaces and messages for the user and network sides are identical. The messages and sequences of this protocol are more easily understood and demonstrated in the example at the end of this section.
QSIG GF—This defines the procedures for individual or specific services and features. This protocol does not have the capability to control these services, but it does provide the generic layer capabilities to enable them. The protocol provides a connection-oriented and connectionless mechanism between the application entities of different PBXs.
QSIG Supplementary Service and ANF Protocols—These protocols specify the control entities for supplementary services and ANFs. These services and ANFs are defined and detailed in separate specifications. Such organizations as the European Computer Manufacturers Association (ECMA) and the European Telecommunication Standards Institute (ETSI) are developing these protocol standards.
Table 3-14. QSIG Protocol Stack
QSIG Protocol | QSIG Standard | |
|---|---|---|
L1 | None | Based on interface being used |
L2 | None | Identical to ISDN L2 (LAPD) |
L3 | QSIG BC | ECMA 142/143; ETS300[*] 171/172 |
QSIG GF | ECMA 165; ETS300 239 | |
QSIG protocols for supplementary services | Separate specifications, such as call forward (ECMA 173/174, ETS300 256/257) and call transfer (ECMA 177/178, ETS300 260/261) | |
L4—L7 | Application-based service elements | Transparent to the network |
[*] ETS300 is an ETSI-based standard. | ||
The QSIG BC protocol provides the basic capabilities for call setup and teardown. This protocol extends the ISDN access protocol for use in a corporate network or private ISDN. QSIG BC is a symmetrical protocol whereby both the network and user sides of the interface are identical. The message sequence for a basic call is demonstrated in Figure 3-12. The QSIG BC messages are functionally similar to the messages discussed in the “ISDN” section of this chapter.
BT and a group of PBX vendors developed DPNSS in the 1980s. They designed this open standard to provide digital private networks at a time when ISDN and QSIG standards were still being defined.
DPNSS has rich services and feature sets and provided the basis for much of the work on the QSIG protocols. Also, interoperability between DPNSS and the QSIG signaling system is specified as part of the inter-working of both protocols.
The ISDN and QSIG protocols became more popular since they were developed, and DPNSS is not as widely used in today’s private networks. DPNSS specifications are available from BT Plc and are defined in the following four documents:
BTNR 188—DPNSS1
BTNR 188-T—DPNSS1 testing schedule
BTNR 189—Inter-working between DPNSS1 and other signaling systems
BTNR 189-I—Inter-working between DPNSS1 and ISDN signaling systems
The signaling systems discussed in this chapter are wide in scope and exist in many versions. It will take some time before Voice over IP (VoIP) systems fully support all these protocols and their variations. Also, standards bodies such as the Internet Engineering Task Force (IETF) are drafting proposals on ways you can inter-work or backhaul these protocols where appropriate.
As standards are accepted and implemented, the inter-working of telephony signaling systems and VoIP systems will become more like “business as usual.”