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Selective calling

Selective calling is a technique in radio communications systems that enables a transmitter to address and alert one or more specific remote receivers by sending a unique signaling sequence, allowing targeted initiation of contact without the need for constant channel monitoring by all users.^[1] This method un-mutes receivers at the intended stations and can facilitate automatic link establishment, reducing interference and improving efficiency in shared frequency environments.^[1] One prominent application is the Selective Calling System (SELCAL) used in aviation, where ground stations transmit a sequence of four preselected audio tones over high-frequency (HF) or very-high-frequency (VHF) radio to alert a specific aircraft, activating a light or chime in the cockpit to notify the crew.^[2] Developed from early 12-tone systems and expanded to 16 tones for up to 10,920 unique codes, SELCAL codes are assigned by the Aviation Spectrum Resources Inc. (ASRI) and included in flight plans, with recent ICAO adoption of SELCAL 32 in 2022 providing backward-compatible expansion to over 200,000 codes to accommodate growing air traffic.^[2] In maritime communications, Digital Selective Calling (DSC) serves as a standardized digital protocol within the Global Maritime Distress and Safety System (GMDSS), functioning like a telephone dial to send predefined messages, including distress alerts, via medium-frequency (MF), HF, or VHF radios using unique Maritime Mobile Service Identities (MMSI).^[3] DSC enables instant transmission of location data via GPS integration and supports call types such as routine, urgency, and safety broadcasts, with mandatory implementation on U.S. vessel radios type-accepted after June 17, 1999, per Federal Communications Commission (FCC) regulations.^[3] More broadly, selective calling protocols, often referred to as selcall, are specified in International Telecommunication Union (ITU) recommendations for HF, VHF, and UHF systems, finding use in ship-to-shore, ship-to-ship, and ground-to-aircraft operations, as well as in land mobile radio for commercial and military purposes.^[1] These systems enhance operational safety and spectrum utilization by minimizing unnecessary listening and enabling precise addressing in multi-user scenarios.^[1]

Overview

Definition and Purpose

Selective calling is a radio communication technique designed to address specific individual receivers or groups of receivers operating on a shared frequency channel. It utilizes in-band signaling, such as subaudible tones transmitted simultaneously with the carrier, or out-of-band methods to selectively activate the squelch circuit in targeted devices, ensuring that only intended transmissions are audible. This mechanism filters out unwanted signals, thereby reducing operator alert fatigue by suppressing irrelevant audio while maintaining the receiver's ability to monitor the channel for activity.^[4] In two-way radio systems, the primary purpose of selective calling is to optimize spectrum usage in environments where multiple users or organizations share the same frequencies, preventing unnecessary audio interruptions and enhancing communication efficiency. By keeping the squelch closed for non-matching signals, it allows operators to continuously monitor the channel for interference or occupancy without being distracted by extraneous transmissions, which supports overall system reliability in both analog and digital formats. This approach aligns with regulatory goals for orderly channel access, as selective calling enables detection of busy conditions prior to transmission.^[4] Federal Communications Commission (FCC) rules in personal and private land mobile radio services emphasize busy channel detection to minimize interference, requiring operators to avoid transmitting on occupied frequencies; selective calling facilitates compliance by permitting silent monitoring of shared channels. Selective calling distinguishes between group calling, which targets multiple units for coordinated operations, and individual calling, which addresses a single unit for direct communication, without specifying the underlying signaling methods.^[4]

Historical Development

The origins of selective calling trace back to the early 1950s, when Motorola developed initial tone-based systems for paging in fire and police radios, utilizing single-tone bursts to alert specific receivers on shared frequencies.^[5] These systems addressed the need for selective activation amid growing radio usage in public safety, marking the shift from basic carrier squelch to more targeted signaling. By the late 1950s, Motorola refined this into continuous tone-coded squelch systems (CTCSS), trademarked as Private Line, which allowed multiple groups to share channels without mutual interference.^[5] The 1960s and 1970s saw expansion into multi-tone methods, with two-tone sequential systems introduced by Motorola as Quick Call II for enhanced paging capacity in professional mobile radio.^[6] In Europe, the ZVEI selective calling standard emerged in the 1970s, developed by the German Association of Radio Users to standardize tone sequences for land mobile communications across the region.^[7] The 1980s brought digital advancements, including Motorola's Digital-Coded Squelch (DCS), trademarked as Digital Private Line, which replaced analog tones with error-corrected binary codes for greater reliability.^[8] Concurrently, Motorola's MDC-1200 protocol enabled data transmission over voice channels, supporting unit identification and status messaging in analog systems.^[6] The 1990s marked a pivot toward integrated digital trunking, exemplified by the European Telecommunications Standards Institute's (ETSI) publication of the TETRA standard in 1995, designed for professional mobile radio with built-in selective calling features.^[9] Entering the 2000s, TETRA gained widespread adoption for public safety networks, while ETSI released the Digital Mobile Radio (DMR) standard in 2005 as a cost-effective digital alternative for business and commercial use.^[10] These developments addressed the limitations of analog methods, particularly as the U.S. Federal Communications Commission (FCC) adopted narrowbanding rules in 1995—refined through the late 1990s—requiring migration to 12.5 kHz channels by 2013 to improve spectrum efficiency.^[11] By the 2020s, analog selective calling has declined significantly, supplanted by digital standards like DMR and TETRA to meet demands for higher capacity and interoperability amid spectrum constraints.^[12]

Group Calling Methods

Continuous Tone-Coded Squelch System (CTCSS)

The Continuous Tone-Coded Squelch System (CTCSS) is an analog signaling method used in frequency modulation (FM) radio communications to control the squelch circuit, allowing only transmissions carrying a specific sub-audible tone to be heard by receivers programmed to that tone. This enables multiple user groups to share the same radio frequency channel without mutual interference, as each group selects a unique tone to "unlock" their receivers. The system transmits a continuous low-frequency tone, typically in the range of 67 to 254 Hz, which is below the typical 300 Hz low-frequency cutoff of voice audio processing in radios, ensuring it remains inaudible during normal conversation.^[5] In operation, the transmitter continuously modulates the carrier with one of several standard tones to encode a group identifier, opening the receiver's squelch only when the matching tone is detected alongside a valid carrier signal. The standard tones are defined in the EIA RS-220A specification, which lists 38 primary frequencies spaced to minimize harmonic interference, such as 67.0 Hz, 82.5 Hz, 123.0 Hz, and up to 254.1 Hz, though some systems support up to 50 tones including non-standard extensions. These tones are frequency-modulated onto the carrier with a typical deviation of ±0.5 to 0.75 kHz in wideband FM systems (5 kHz total deviation), ensuring reliable detection without significantly impacting voice modulation.^[13]^[14]^[15] Implementation involves integrating a tone generator into the FM modulator of the transmitter, often using a stable oscillator like a phase-locked loop (PLL) or crystal-derived circuit to produce the precise frequency. In the receiver, a bandpass filter isolates the sub-audible tone from the incoming signal, followed by a decoder—commonly a PLL-based circuit—that compares the extracted tone against the programmed frequency and activates the squelch gate if it matches within tolerances (typically ±1-2 Hz). A high-pass filter then blocks the tone from the audio output path, preventing it from being heard. Early decoders used mechanical vibrating reed filters, but modern electronic PLL designs offer greater accuracy and resistance to noise.^[16]^[17] CTCSS finds widespread applications in amateur radio for repeater access and shared frequency coordination, in public safety communications to isolate dispatch channels amid urban interference, and in General Mobile Radio Service (GMRS) for family or group privacy on licensed repeaters. It provides a simple means to segment channel usage, such as allowing a local amateur club to use one tone on a community repeater while another group uses a different tone on the same frequency.^[18]^[19]^[20] The system's advantages include its simplicity and low cost, requiring minimal additional hardware in analog radios, which makes it accessible for legacy and budget systems. However, it is susceptible to tone falsing, where voice harmonics—particularly from low-pitched male voices or distorted audio—can mimic a tone frequency and inadvertently open the squelch, leading to unwanted interference. Unlike digital alternatives such as Digital-Coded Squelch (DCS), CTCSS offers no true privacy, as the tones can be detected and stripped by scanning receivers.^[5]^[21]^[17] Variants of CTCSS include Motorola's Private Line (PL) tones, which adhere to the same EIA standards but were trademarked for their proprietary equipment, and General Electric's Channel Guard (CG), an equivalent system with identical tone sets used in early land mobile radios. These branding differences persisted into the 1980s but converged on the universal CTCSS nomenclature for interoperability.^[5]

Digital-Coded Squelch (DCS)

Digital-Coded Squelch (DCS), also known as Digital Private Line (DPL) in Motorola systems or Continuous Digital-Coded Squelch System (CDCSS), serves as a digital enhancement to analog group calling methods like CTCSS, enabling selective squelch activation through sub-audible binary data streams rather than continuous tones.^[8]^[22] In operation, DCS transmits a continuous 23-bit code word at a rate of 134 bits per second using non-return-to-zero level (NRZ-L) encoding and frequency-shift keying (FSK) modulation, with the signal repeated throughout the transmission to maintain receiver synchronization and squelch openness.^[22]^[23] The code word employs a binary Golay (23,12) error-correcting code, capable of detecting and correcting up to three bit errors, ensuring robust performance in noisy environments.^[23] This structure consists of 12 information bits followed by 11 parity bits, where the information bits comprise a 9-bit address code and a fixed 3-bit flag sequence of '001' for synchronization and validation.^[23]^[8] The 9-bit address allows for 512 possible combinations, though only 83 standardized codes are typically used to avoid aliasing (where similar codes might falsely trigger receivers), with each code represented as a 3-digit octal number (e.g., 023).^[8] Some manufacturers extend this to 104 codes, and polarity inversion (normal or inverted) can double the effective options to over 200 unique pairs, further minimizing interference from adjacent groups.^[8] At the end of a transmission, a brief 134 Hz tone may be sent to signal squelch closure, preventing hang-up from residual code.^[22] Implementation involves biphase-like NRZ signaling with a frequency deviation of approximately ±1 kHz (typically 500-800 Hz range) to keep the signal sub-audible below 300 Hz, integrated with the voice audio without significant distortion.^[22] Receivers employ digital correlators or syndrome decoders to validate the Golay code against the programmed address, opening the squelch only upon a match and maintaining it as long as the code repeats correctly.^[23] This digital approach contrasts with CTCSS by providing inherent error correction, though it requires more precise transmitter-receiver alignment to avoid decoding failures from frequency offsets.^[8] DCS finds primary applications in public safety, commercial, and amateur radio systems, such as Motorola's DPL-equipped portables and mobiles, where it enables group-specific channel access without individual addressing.^[8] Many modern radios support hybrid modes, allowing DCS alongside CTCSS for backward compatibility in mixed fleets.^[22] Key advantages include the expanded code space—83 to 104 base codes versus CTCSS's 38-50 tones—substantially reducing the likelihood of accidental squelch opening (falsing) from nearby transmissions, and improved noise immunity through error correction.^[8]^[24] However, DCS incurs higher implementation complexity due to digital processing circuitry and slightly increased power consumption compared to analog CTCSS, potentially impacting battery life in portable units.^[24] Standardization is governed by the Telecommunications Industry Association (TIA) under ANSI/TIA-603, which defines the 83 valid codes and transmission parameters to ensure interoperability across vendors like Motorola (DPL), GE (DCG), and Icom (DTCS).^[8]

Selective Calling Tones (SelCall)

Selective calling tones, commonly known as SelCall, represent a burst-tone signaling protocol designed for selective addressing of individual or group receivers in analog radio systems, allowing targeted activation while muting others on the same frequency. This method transmits short sequences of audio tones prior to the voice message or data, enabling efficient use of shared channels in VHF and UHF bands. Originating in European standards during the 1970s, SelCall has been widely adopted for its simplicity in analog environments, though it predates modern digital alternatives.^[7]^[25] In operation, SelCall employs sequential bursts of 4 or 5 distinct tones, each lasting approximately 70 to 100 milliseconds, within the audible frequency range of 300 to 3000 Hz to ensure compatibility with voice communications. These tones are transmitted as a preamble, followed immediately by an alert tone, voice transmission, or additional data, with pauses between tones typically around 30 to 50 milliseconds to facilitate decoding. The system supports both individual addressing via unique codes and group calls using wildcard patterns, such as repeating a specific tone to target multiple units. For example, in a 5-tone sequence, the first four tones form an identifier, while the fifth may denote a group or function like alert or reset.^[7]^[26]^[27] Key standards include ZVEI1 and ZVEI2, developed in Germany by the Zentralverband Elektrotechnik- und Elektronikindustrie (ZVEI) in the early 1970s at a signaling rate of about 6 tones per second, utilizing 16 possible tones (0-9 and A-F) for coding individual or group IDs. Internationally, the CCIR standards—established by the Comité Consultatif International des Radiocommunications (predecessor to ITU-R)—define variants like CCIR-1 (100 ms tones) and CCIR-2 (70 ms tones), both using 5-tone sequences with 16 frequencies for broad compatibility. The PCC (Polish) variant, known as PCCIR, modifies this to 15 tones with adjusted frequencies for group, reset, and repeat functions, maintaining 100 ms durations. These standards ensure interoperability across European and some international systems, with codes assigned for specific addressing needs.^[7]^[25]^[26] Implementation involves sequential decoding at the receiver using timing gates to verify tone order, duration, and intervals, rejecting false triggers from noise or incomplete sequences. Receivers employ narrowband filters for each tone frequency, followed by logic to match the full code against preprogrammed IDs, commonly integrated into paging systems and dispatch consoles for automated alerting. In practice, this setup allows for revertive signaling, where receivers acknowledge calls, enhancing reliability in operational environments.^[27]^[7] Applications of SelCall are prominent in European public safety sectors, such as fire and rescue services, where it facilitates rapid dispatch to specific units on shared frequencies, and in maritime VHF communications for ship-to-shore or inter-vessel addressing, integrating seamlessly with subsequent voice traffic. For instance, dispatch consoles use SelCall to send status messages or emergency alerts, with receivers providing audible and visual notifications.^[27]^[28]^[26] The primary advantage of SelCall lies in its precise addressing capability, enabling private or group communications on congested channels without continuous tone overhead, thus conserving bandwidth and reducing operator fatigue. However, as an analog system, it is vulnerable to noise and interference, which can distort tones and cause missed or false activations, contributing to its obsolescence in favor of digital protocols in modern networks.^[27]^[29] Evolutionarily, analog SelCall has transitioned in some applications to digital variants, such as ITU-R Recommendation M.493 for maritime digital selective calling (DSC), which retains the addressing concept but uses error-corrected binary codes for improved robustness over VHF and HF.

Extended CTCSS (XTCSS)

Extended CTCSS (XTCSS) represents an evolution of the Continuous Tone-Coded Squelch System (CTCSS), introduced in the late 1990s to provide enhanced group calling capabilities in two-way radio communications. Developed by Motorola, it expands the standard CTCSS framework by supporting 99 analog codes, allowing for greater subdivision of shared channels to minimize interference among user groups.^[30] This system enhances CTCSS for group squelch by offering more selective filtering options while maintaining compatibility with existing analog radio infrastructure. In operation, XTCSS employs sub-audible tones below 300 Hz to activate the receiver's squelch only for matching codes, similar to CTCSS but with an extended set of frequencies that includes additional tones such as 159.8 Hz, 165.5 Hz, and 171.3 Hz beyond the conventional list.^[31] These codes are transmitted in-band within the voice signal, enabling quieter performance by reducing audible tone presence during transmission. The decoder synchronizes with voice activity to process these tones effectively, making it suitable for environments where continuous audio clarity is essential.^[32] Implementation of XTCSS was primarily featured in Motorola's XTS series radios, such as the XTS-5000 model, where service updates allowed for the addition of extended CTCSS tone capabilities to improve signaling reliability.^[32] It finds applications in noisy settings like construction sites, where the expanded code set helps isolate communications and limits tone bleed into the primary audio path, preserving message intelligibility.^[30] Key advantages of XTCSS include its provision of 99 codes compared to the 38 in standard CTCSS, enabling finer control over group access and reduced cross-talk in multi-user scenarios. However, its disadvantages encompass limited adoption, largely overshadowed by the rise of digital-coded squelch (DCS) systems, and the added complexity in decoding the extended tone set, which requires specialized radio hardware.^[30]

Tone Burst

Tone burst, also known as single-tone signaling, is an early method employed in analog radio systems to initiate activation of receivers through a brief, continuous audio tone transmitted at the start of a communication. The tone typically lasts 0.5 to 1.5 seconds and serves to unmute or alert all compatible receivers on the channel, lacking any coding for individual or group specificity.^[33] In operation, the transmitter generates this fixed-frequency tone, often paired with the carrier signal, which the receiver detects to open its squelch and produce an audible or visual alert, remaining active until manually reset or timed out.^[33] Implementation of tone burst requires minimal circuitry: a simple audio oscillator in the transmitter, activated momentarily via a switch or integrated with the push-to-talk function, produces the tone at a predetermined frequency such as 1750 Hz in European systems or within the 2000–2800 Hz range in U.S. applications (e.g., 1950 Hz or 2100 Hz).^[33] On the receiving end, a bandpass filter isolates the tone, followed by a detector circuit and timer to confirm duration and prevent false triggers from incidental audio. This setup was add-on compatible with existing two-way radios, using components like reed relays in early designs or electronic filters in later variants.^[34] Historically, tone burst emerged in the 1950s and gained prominence in the 1960s–1970s for basic paging and repeater access in professional radio networks, including General Electric's Exec I series used in public safety dispatch systems.^[35] Applications included alerting fire departments and emergency services via base station broadcasts, as well as activating mobile units in systems like the GE Exec I for overriding muted consoles in multi-division operations.^[35] Its simplicity made it suitable for shared channels in business and amateur radio, where it reduced constant noise by keeping receivers muted until the tone was received.^[33] The method's primary advantages lie in its reliability and low cost, requiring no complex encoding and offering robust performance in noisy environments due to the distinct tone frequency.^[34] However, it provides no selectivity, causing all units on the frequency to alert simultaneously, which led to inefficiencies in multi-user scenarios and privacy concerns. By the 1980s, tone burst became obsolete, superseded by more selective multi-tone and continuous tone-coded systems like CTCSS for better channel sharing and targeted activation.^[36]

Analog Individual Calling Methods

Dual-Tone Multi-Frequency (DTMF)

Dual-tone multi-frequency (DTMF) signaling serves as an analog individual calling method in radio systems, employing pairs of simultaneous audio tones to encode and transmit unique identifiers for specific receivers, thereby enabling selective activation amid ongoing transmissions.^[37] This approach targets individual units rather than groups, distinguishing it from broader squelch-based methods by allowing precise, addressable communications in two-way radio environments.^[6] In operation, DTMF transmits sequences of 2 to 7 digits as short bursts of tones within the voice-frequency band, typically lasting 50-100 milliseconds per digit with inter-digit pauses to ensure reliable decoding. Each digit is represented by the simultaneous transmission of one frequency from a low group (697 Hz, 770 Hz, 852 Hz, or 941 Hz) and one from a high group (1209 Hz, 1336 Hz, 1477 Hz, or 1633 Hz), creating 16 possible unique combinations corresponding to digits 0-9 and letters A-D.^[38] These tones are generated by radio keypads or encoders and superimposed on the carrier signal, allowing them to pass through standard audio paths without requiring separate modulation.^[39] The code structure leverages these 16 symbols to form alphanumeric sequences that serve as unit identifiers, with systems like Motorola's Touch Call employing 3- to 5-digit codes to uniquely address individual radios or pagers. Decoders in receiving units employ bandpass filters to isolate the low and high frequencies, followed by zero-crossing detection or Goertzel algorithms to identify the precise pair, then count and match the full sequence against preprogrammed IDs before unsquelching the audio.^[40] For automatic number identification (ANI), decoders log the transmitting unit's code upon activation, facilitating dispatch tracking in fleet operations. DTMF finds applications in pager activation, where tone sequences trigger alerts on targeted devices, and in remote control functions for amateur and commercial radios, such as linking repeaters or adjusting settings via encoded commands.^[37] In amateur radio, operators use DTMF to access autopatch systems or control station equipment over the air, while commercial setups employ it for selective paging in industries like public safety and utilities.^[41] A key advantage of DTMF lies in its universality, derived from telephony standards, which simplifies integration with telephone interconnects and enables widespread adoption without proprietary hardware. However, its audible nature can disrupt conversations, and the tones are susceptible to falsing—false activations triggered by speech harmonics or noise mimicking frequency pairs—necessitating robust filtering in decoders.^[6]^[42] Standardization originated with the Bell System in the early 1960s, where AT&T introduced DTMF as "Touch-Tone" dialing on November 18, 1963, to replace rotary pulses with faster, multifrequency signaling over voice lines. This telephony foundation was adapted for radio selective calling in the following decades, as two-way systems incorporated DTMF encoders and decoders to leverage the established tone grid for non-telephonic applications like unit addressing.^[43]^[40]

Two-Tone Sequential

Two-tone sequential is an analog selective calling method that transmits a pair of distinct audio tones in sequence to activate specific receivers, such as pagers or mobile radios, enabling individual or group alerting without disturbing others on the same frequency.^[6] Developed as an improvement over single-tone burst systems for greater selectivity, it was widely adopted in public safety communications during the mid-20th century.^[44] In operation, the system sends the first tone for approximately 1 second, followed by a brief silence of 50 milliseconds to 1 second, and then the second tone for 1-3 seconds, resulting in a total transmission duration of 1-3 seconds per call.^[6] The tones are generated within the audio frequency range of 500-3000 Hz to ensure compatibility with voice channels on VHF and UHF radio systems.^[45] This sequential format allows the receiver to distinguish the intended signal from background noise or unrelated transmissions. The code structure typically supports 40-100 unique combinations, depending on the standard, with pairs selected from predefined tone sets to minimize interference.^[46] For example, the GE Type 99 format uses three groups (A, B, C) of ten frequencies each—such as Group A: 547.5 Hz, 592.5 Hz, 637.5 Hz, 682.5 Hz, 727.5 Hz, 757.5 Hz, 802.5 Hz, 847.5 Hz, 892.5 Hz, 937.5 Hz; Group B: 517.5 Hz, 607.5 Hz, 652.5 Hz, 697.5 Hz, 787.5 Hz, 832.5 Hz, 877.5 Hz, 922.5 Hz, 967.5 Hz; and Group C: 532.5 Hz, 712.5 Hz, 667.5 Hz, 622.5 Hz, 772.5 Hz, 817.5 Hz, 862.5 Hz, 907.5 Hz, 952.5 Hz—allowing up to 99 codes through table-based pairings like 592.5 Hz followed by 607.5 Hz.^[46] Departments often customize codes from these sets to suit operational needs, ensuring unique identifiers for personnel or units.^[47] Implementation involves a transmitter that sequences the tones via an encoder console, modulating them onto the carrier signal before voice dispatch.^[44] Receivers employ dual bandpass filters tuned to the programmed frequencies, coupled with timing logic gates or microprocessors to verify the sequence, silence gap, and duration thresholds, triggering an alert only on a match.^[48] Primarily applied in fire and emergency medical services (EMS) paging from the 1970s to the 1990s, two-tone sequential was a staple for alerting volunteers and responders over shared channels, with Motorola's Quik-Call II serving as a prominent variant integrated into pagers like the Minitor series.^[49] These systems facilitated rapid, tone-and-voice dispatches in rural and urban public safety networks.^[50] Key advantages include low falsing rates, as the dual-tone sequence and precise timing reduce erroneous activations from voice peaks or interference compared to single-tone methods.^[51] However, limitations encompass a restricted number of codes, constraining scalability in large systems, and audible tone alerts that could reveal operational details or cause unnecessary disturbances.^[52] Variants extend the basic two-tone approach, such as Motorola Type 1, which incorporates three tones for enhanced coding capacity, and international standards like those from the CCIR for global interoperability in paging.^[46]

Quik-Call I

Quik-Call I, introduced by Motorola in the 1970s as an upgrade to basic two-tone sequential paging, is an analog selective calling system designed for expanded individual addressing in radio communications. It functions by transmitting two sequential pairs of simultaneous tones, where each pair consists of two specific frequencies sounded together, followed by a short silence interval, enabling precise pager activation without alerting all units on the channel. This method was particularly suited for public safety operations, offering compatibility with existing two-tone infrastructure while providing a larger addressable code space for growing dispatch needs.^[6] The operation involves an initial pair of tones lasting approximately 1 second, a 200 ms silence gap, and a second pair of tones also lasting 1 second, using a selection of 6 to 12 basic frequencies to form the pairs. Code structure relies on predefined combinations of these pairs, allowing up to 24 possible variations per position and yielding a maximum of 576 unique codes (24²), as seen in Motorola's fire dispatch implementations where specific pair assignments enabled targeted alerting for individual units or stations. For example, in systems like those used by U.S. fire departments, a code might assign distinct pairs to different apparatus, ensuring only the intended recipient responds.^[46] Implementation requires advanced pagers or receivers equipped with dual-tuned decoders—often using reed relays or early digital memory—to recognize custom pair frequencies and filter out noise or unintended signals. Upon decoding a match, the device emits an alert tone (typically a warble or beep sequence) to notify the user, followed by the voice dispatch message over the shared channel. These decoders supported programmable codes, allowing agencies to tailor assignments without hardware changes. In the 1980s, Quik-Call I saw widespread adoption in U.S. public safety, such as Los Angeles County Fire Department's dispatch system, where it integrated seamlessly with two-tone setups for hybrid operations.^[6]^[53] Compared to simpler two-tone sequential methods, Quik-Call I's paired-tone approach provided more unique codes, reducing the risk of overlaps in large fleets and improving selectivity in noisy environments. However, as an analog system, it remained vulnerable to interference and was limited to line-of-sight radio propagation, necessitating repeaters for extended coverage and lacking the error correction or acknowledgments of later digital alternatives.^[6]

MDC Signaling

MDC Signaling, also known as Motorola Data Communications, is an analog data-over-voice system developed by Motorola for selective individual calling in two-way radio networks. It embeds low-speed digital data packets within voice transmissions using audio frequency shift keying (AFSK) modulation, allowing radios to exchange identification, status, and alert information without significantly disrupting audio communication.^[6]^[54] The system operates by transmitting short data bursts during pauses in voice activity, typically at the start or end of a transmission. For MDC-1200, the primary variant, data is modulated at 1200 baud using mark and space tones of 1200 Hz and 1800 Hz, respectively, while MDC-600 uses the same tones at a slower 600 baud rate. Each packet carries a 32-bit information payload, typically including a 16-bit message type constant and 16-bit unit ID (with variations for functions like emergency or status using 8-bit function and 8-bit data fields), protected by CRC and encoded into a 112-bit codeword using convolutional encoding and interleaving, preceded by a preamble and synchronization sequence. This enables features such as automatic number identification (ANI) for push-to-talk ID, emergency alerts via dedicated function codes, selective calling to specific units, radio checks, and status messaging, all decoded by the receiving radio without interrupting ongoing voice.^[55]^[56]^[6] Implemented primarily in Motorola's analog radios from the 1980s through the 2000s, such as the Saber and Astro series, MDC signaling integrates via firmware and radio service software for encoding and decoding. Decoders extract the subaudible data bursts, which last about 173 ms for ANI transmissions, ensuring compatibility with standard FM voice channels in fleet dispatch and utility applications. For example, in public safety and commercial fleets, it allows dispatchers to identify calling units and send targeted alerts, evolving briefly from earlier tone-based paging methods by adding structured data capabilities.^[56]^[55] The advantages of MDC include its ability to overlay unit IDs and basic status on existing voice systems without requiring full digital infrastructure, providing backward compatibility and low-cost enhancements for individual calling. However, its low data rates limit throughput, and performance degrades in noisy environments due to susceptibility to bit errors, with falsing rates around 1 in 1 million but potential for inversion in burst errors. The MDC-1200 variant offers faster transmission than MDC-600, supporting broader adoption in high-volume scenarios, though both have largely been supplanted by modern digital protocols.^[55]^[6]

In-Band Miscellaneous Methods

MODAT

MODAT, introduced by Motorola in 1972, was an early in-band signaling system designed for individual unit identification in mobile two-way radios. It functioned as the first commercial mobile data terminal, enabling vehicles to transmit and receive short data messages, such as unit IDs and status updates, to dispatch centers. This system was particularly valued in public safety applications, where officers could send information like license plate details during field operations without verbal communication.^[57]^[58] The operation of MODAT involved transmitting a 7-tone audio sequence immediately before voice transmission to alert and identify the specific unit. Each tone lasted 40 milliseconds and was selected from a set of frequencies approximately between 600 Hz and 2000 Hz, ensuring compatibility with standard radio audio paths. The first tone indicated routine or emergency status, while the subsequent tones encoded the unit identifier. This burst allowed decoders in receiving radios to open the squelch only for the addressed unit, reducing unnecessary activations across the fleet.^[59]^[6] The code structure utilized 10 distinct tones for digits 0-9 (plus a reset tone), supporting a 4-digit unit ID with up to 9000 possible combinations (0000-8999) and fixed timing intervals between tones for reliable decoding. Implementation occurred primarily in Motorola's 1970s-era equipment, such as the MX-300 series portables and MT-500 handhelds, employing simple sequential tone decoders integrated into the radio's audio circuitry. These decoders filtered and timed the incoming tones to match pre-programmed IDs, triggering an alert or opening the audio path.^[59] MODAT found applications in trucking fleets, taxi dispatch operations, and public safety agencies, where it integrated seamlessly with conventional two-way radios to enable selective individual calling without requiring complex infrastructure. Its compact encoding allowed for quick transmission of essential identification data, making it suitable for resource-constrained environments. However, the system's audible tones often interfered with voice communications, and it suffered from high rates of falsing due to environmental noise or similar frequencies, limiting its reliability in noisy settings.^[59]^[57] By the 1990s, MODAT had become obsolete, largely replaced by more robust digital signaling methods like MDC and modern trunked systems that offered greater capacity, lower error rates, and inaudible operation. Unlike group-oriented selective calling tones such as SelCall, MODAT focused exclusively on individual unit addressing.^[59]

Voice Inversion Signaling

Voice inversion signaling is an analog in-band method used to scramble voice transmissions for basic privacy in radio communications, often integrated with selective calling systems to control access on shared channels. The technique inverts the audio spectrum, typically flipping frequencies within the 300-3000 Hz speech band around a carrier frequency of approximately 1650 Hz, so that low frequencies become high and vice versa—for instance, a 300 Hz tone shifts to 3000 Hz, rendering the audio unintelligible without a matching decoder.^[60] This inversion occurs during transmission via a simple analog scrambler in the transmitter, which mixes the audio with the reference carrier and filters the sum frequencies, while the receiver employs a complementary inverter circuit to restore the original spectrum.^[60] Implementation is straightforward and cost-effective, relying on fixed inversion codes selectable via different reference frequencies (e.g., offsets greater than 200 Hz apart), with some systems incorporating tone-controlled activation for added selectivity.^[60] Historically, voice inversion emerged in the 1920s for commercial radio-telephone circuits to provide rudimentary privacy, with significant improvements during World War II for military applications, and it gained prominence in the 1970s through 1990s for commercial and public safety two-way radios.^[60] In these systems, it served as a basic privacy measure on shared frequencies, particularly in law enforcement and business radio networks, where it was frequently paired with calling tones—such as CTCSS or selective signaling—to alert and unblock the receiver's squelch only for authorized users before unscrambling the voice.^[60] Devices from manufacturers like Motorola and Midian exemplified this integration, allowing inversion scramblers to support features like automatic number identification (ANI) and selective calling within analog radios.^[61] The primary advantages of voice inversion include its low cost and simplicity, making it accessible for widespread adoption in analog equipment without requiring complex digital processing.^[60] However, it offers only minimal security, as the scrambled signal can be easily defeated by casual listeners using inexpensive adjustable inverters or even by recognizing speech patterns after brief exposure, with trained listeners achieving significant intelligibility with practice.^[60] Lacking true selectivity beyond basic tone gating, it provides no robust individual addressing and is now largely obsolete, superseded by digital encryption in modern systems, though legacy implementations persist in some niche analog setups.^[60]

Out-of-Band and Digital Methods

Out-of-Band Individual Calling

Out-of-band individual calling in selective calling systems employs a dedicated control channel that operates on a separate frequency from the primary voice channels to transmit identification (ID) codes, alerting specific radio units to an incoming call before directing them to the appropriate voice channel for communication. This process typically uses frequency shift keying (FSK) modulation at rates such as 1200 bits per second for low-speed data or higher rates like 9600 baud in systems such as Enhanced Digital Access Communications System (EDACS), where the control channel continuously broadcasts system status and call assignment information.^[62]^[63] The control channel sends packets containing logical IDs (LIDs) or group IDs (GIDs) to initiate individual calls, enabling the system to dynamically allocate voice channels without interrupting ongoing transmissions on the main paths.^[62] Implementation of these systems often requires radios equipped with dual receivers: one dedicated to continuously monitoring the control channel for alerts and the second to tune to the assigned voice channel upon notification, a configuration common in analog trunked environments. This setup was integral to early trunking platforms like the GE MASTR III base stations used in EDACS, where the Group Equipment Trunking Card (GETC) and site controller manage channel assignments across dedicated frequencies.^[64]^[62] Applications focused on public safety and commercial operations, where the control channel's role in handling signaling reduced interference and clutter on voice channels, allowing for more efficient use of spectrum in multi-user scenarios.^[65] The primary advantages of out-of-band individual calling include cleaner voice transmission paths, as all signaling and coordination occur separately from audio, minimizing disruptions and improving overall system capacity in trunked analog setups. However, disadvantages encompass the need for additional hardware, such as dual receivers in subscriber units and specialized base station components like the GETC, along with increased spectrum demands for the separate control channel. Historically, these methods gained prominence in the 1980s and 1990s through systems like GE's EDACS on MASTR III platforms, serving as precursors to fully digital trunking by demonstrating efficient channel management before the widespread adoption of integrated digital protocols.^[65]

Modern Digital Systems

Modern digital systems for selective calling have evolved to integrate addressing mechanisms directly into digital voice and data streams, enabling efficient, secure targeting of individuals or groups in professional mobile radio environments. Digital Mobile Radio (DMR), standardized by the European Telecommunications Standards Institute (ETSI) in 2005, employs a two-slot Time Division Multiple Access (TDMA) structure operating in 12.5 kHz channels to support selective calling. In DMR Tier II and Tier III configurations, which facilitate conventional and trunked operations respectively, individual and group calls are initiated through Link Control (LC) messages embedded within voice frames. These LC messages include 24-bit source and destination identifiers, along with a 24-bit synchronization pattern and a group/individual bit, distributed across a 360 ms superframe comprising six bursts. This embedding allows for precise addressing without dedicated out-of-band signaling, supporting applications in public safety and utilities where post-2010 deployments have emphasized spectrum efficiency through TDMA doubling of capacity over analog systems.^[66] Terrestrial Trunked Radio (TETRA), initially standardized by ETSI in 1995 and enhanced through Release 2 starting in the early 2000s, including the TETRA Enhanced Data Service (TEDS) standardized in 2007 for higher-speed applications, utilizes a hybrid Frequency Division Multiple Access (FDMA) and TDMA air interface. Selective calling in TETRA relies on 24-bit Short Subscriber Identities (SSI), comprising Individual SSI (ISSI) for personal calls and Group SSI (GSSI) for multicast addressing, carried within the protocol data units (PDUs) in the air interface bursts that form the basis of encrypted signaling and traffic channels. These frames incorporate frame numbering for replay protection and are protected by air-interface encryption using keys such as the Static Cipher Key (SCK) for Class 2 or Derived Cipher Key (DCK) and Common Cipher Key (CCK) for Class 3 operations, ensuring confidentiality in individual and group setups. TETRA's design supports simultaneous voice and data services, including text messaging and location updates, making it suitable for mission-critical public safety networks and utility operations since the early 2010s.^[67]^[68]^[69] Project 25 (P25) Phase II, developed under Telecommunications Industry Association (TIA) standards and widely adopted in North American public safety sectors post-2010, achieves spectrum efficiency via two-slot TDMA in 12.5 kHz channels while maintaining backward compatibility with Phase I FDMA systems. Selective calling incorporates 24-bit source and destination identifiers, alongside 16-bit talkgroup IDs, embedded in the Link Control Word (LCW) of voice frames processed by Improved Multi-Band Excitation (IMBE) or Advanced Multi-Band Excitation (AMBE) vocoders, which encode 88 bits of speech per 20 ms frame. This structure enables Automatic Number Identification (ANI) through the source ID and emergency alerting via a dedicated bit in the LCW, triggering priority handling and notifications. P25 Phase II systems often integrate Global Positioning System (GPS) data for location-aware calling, enhancing coordination in utilities and emergency response, though deployment requires substantial infrastructure investment due to the need for upgraded repeaters and subscriber units.^[70] These systems offer advantages such as enhanced security through integrated encryption—e.g., TETRA's multi-class key management and P25's optional end-to-end encryption—and data-rich capabilities for text alerts and GPS positioning, outperforming analog methods in reliability and capacity. However, disadvantages include higher upfront costs for digital infrastructure and potential interoperability challenges across vendors without strict compliance testing. Adoption in public safety and utilities has grown since 2010, driven by spectrum reallocation pressures and the need for resilient communications, with DMR serving cost-sensitive commercial extensions, TETRA dominating European critical infrastructure, and P25 Phase II leading in U.S. federal and state networks.^[67]^[71]

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