Caller ID
Caller ID, formally known as caller identification service, is a telecommunications feature that provides the recipient of an incoming call with the caller's telephone number and, in some implementations, the caller's name, transmitted via in-band signaling during the ringing phase of the call.[1][2] This service operates as a form of automatic number identification (ANI), leveraging protocols such as Signaling System No. 7 (SS7) in traditional public switched telephone networks (PSTN) or Session Initiation Protocol (SIP) headers in voice over IP (VoIP) systems to convey the originating number.[3] Initially developed in the late 1960s by inventor Theodore George Paraskevakos, who patented a system for transmitting caller information in 1971, Caller ID gained commercial viability in the 1980s through Bell System standards and became widely adopted in the 1990s following regulatory approvals by bodies like the Federal Communications Commission (FCC).[4] The technology enables users to screen calls, reducing unwanted interruptions from telemarketers or unknown parties, and has evolved to include enhanced features like Caller ID with name (CNAM) databases for displaying registered names associated with numbers.[5] However, its effectiveness has been undermined by caller ID spoofing, where malicious actors falsify the transmitted number to impersonate legitimate entities, facilitating scams and robocalls that bypass traditional trust mechanisms.[6] In response, the FCC has mandated authentication frameworks such as STIR/SHAKEN, which cryptographically verify caller identity to combat spoofing, though implementation challenges persist due to legacy network infrastructures and international variances.[7] Early deployment of Caller ID sparked privacy debates, with advocates arguing it infringed on callers' anonymity rights by revealing numbers without consent, particularly affecting victims of domestic abuse or informants seeking to contact authorities discreetly.[8] Over time, concerns shifted toward recipient protection, as spoofing eroded the service's reliability, inverting the privacy dynamic from caller vulnerability to widespread distrust in displayed information.[9] Despite these issues, Caller ID remains a foundational tool in telephony, integral to modern call management apps and services that aggregate reputation data to flag potential threats.[10]Technical Fundamentals
Calling Line Identification
Calling Line Identification (CLI), also referred to as Calling Line Identification Presentation (CLIP) in standards documentation, is a telephony supplementary service that enables the transmission of the calling party's directory number to the called party's terminal equipment during call setup.[11] This service operates within both analog Public Switched Telephone Network (PSTN) and digital systems such as Integrated Services Digital Network (ISDN), allowing the recipient to identify the originator prior to answering.[12] The CLI data typically includes the caller's telephone number in international or national format, with optional sub-addressing for additional routing information.[13] In analog PSTN environments, CLI is conveyed using out-of-band Frequency Shift Keying (FSK) modulation during the brief silent interval between the first and second ringing cadence.[14] The European Telecommunications Standards Institute (ETSI) EN 300 659-1 specifies an asynchronous voice-band FSK protocol for this purpose, encoding the data in binary format with error detection via checksums to ensure reliability over the subscriber line.[15] This method supports transmission rates compatible with standards like V.23 modulation, typically at 1200 bits per second, and includes a channel seizure signal followed by the message payload.[14] In contrast, digital networks employ in-band signaling; for ISDN, CLI is delivered via the Data (D) channel using Digital Subscriber Signalling System No. 1 (DSS1) protocols, integrating seamlessly with call establishment messages.[13] CLI transmission is subject to restrictions through the complementary Calling Line Identification Restriction (CLIR) service, which permits callers to suppress their number presentation on a per-call or permanent basis, resulting in "private" or withheld indicators at the recipient.[16] International interoperability is guided by ITU-T Recommendation E.157, which outlines the delivery of calling party numbers across borders, accounting for numbering plan variations and privacy overrides.[12] Subscriber equipment, such as dedicated CLI receivers or compatible telephones, demodulates the FSK signal in analog cases or interprets signaling messages in digital setups to decode and display the information.[14]Signal Protocols and Transmission
Caller ID signals are transmitted over analog telephone lines using in-band modulation techniques during silent intervals to avoid interference with the ringing cadence. In North America, the primary protocol follows the Telcordia (formerly Bellcore) Technical Reference TR-NWT-000030, which employs frequency-shift keying (FSK) at 1200 bits per second with Bell 202 modulation frequencies: 1200 Hz for the mark (logical 1) and 2200 Hz for the space (logical 0).[17] The transmission occurs after the first ring burst, during the inter-ring silence, ensuring the receiving equipment can decode without disruption.[18] The FSK message structure begins with a channel seizure signal (CSS), consisting of approximately 140 milliseconds of alternating mark and space tones to alert the customer premises equipment (CPE).[17] This is followed by a mark signal of about 80 milliseconds of continuous mark tone, then the data payload framed using a Single Data Link Control Procedure (SDLC)-like format with start and stop bits around each 8-bit ASCII character.[17] The data includes fields such as date (MMDD), time (HHMM), calling number prefixed with "NMBR=", and optionally calling name with "NAME=", terminated by a checksum for error detection.[17] Signal levels are specified at -12 to -18 dBm to ensure reliable detection by CPE decoders. In regions adhering to ETSI standards, such as much of Europe, the protocol defined in EN 300 659-1 supports both FSK and dual-tone multi-frequency (DTMF) transmission over the public switched telephone network (PSTN).[15] FSK variants use V.23 modulation at 1200 baud with frequencies around 1300 Hz (mark) and 2100 Hz (space), transmitted similarly between rings or pre-ring with alert signals. DTMF-based transmission, common in some implementations, sends a sequence of up to 16 DTMF digits representing the caller number before the first ring, often preceded by a start digit or tone pair for synchronization.[19] This method encodes information as standard telephone keypad tones, with each digit lasting 70-100 ms, and is used where FSK compatibility is limited.[20] Transmission reliability depends on line conditions, with FSK offering higher data integrity via checksums compared to DTMF's simpler digit stream, though both require CPE capable of on-hook detection.[21] In digital PSTN segments, signaling system No. 7 (SS7) carries the calling party number (CPN) in the initial address message (IAM) for routing, but the analog leg to the end-user relies on these modulated protocols.[22] Regional variations, such as higher signal levels in ETSI (-9 dBm) versus Bellcore, accommodate differing loop lengths and attenuation.History
Early Development and Introduction
Theodore George Paraskevakos initiated the development of caller identification technology in 1968 while working on systems for automatic number identification in telecommunications. By 1971, he had invented a transmitter-receiver apparatus that encoded and decoded the caller's telephone number during a call, for which he received U.S. Patent 3,727,003 later that year.[4][23] This patent described a method to transmit signaling data between the calling and receiving parties' equipment, enabling the display of the originating number without interrupting the voice path, though practical implementation required advancements in telephone network signaling. Throughout the 1970s, research continued with testing of caller ID prototypes in laboratory and limited field environments, addressing challenges such as signal integrity over analog lines and integration with existing switching equipment. Contributions from institutions like Bell Laboratories advanced related technologies in digital signal processing and line signaling, which facilitated the transmission of identification data in the inter-ring interval or via in-band frequencies.[24][10] These efforts culminated in the late 1980s when regional Bell operating companies began commercial rollouts, with BellSouth launching the first widespread service in December 1988 in Memphis, Tennessee, requiring subscribers to purchase compatible display devices and pay monthly fees.[25] Initial adoption faced regulatory hurdles, including debates over privacy and the potential for anonymous calling, leading to features like caller ID blocking introduced alongside the service. By the early 1990s, deployment expanded across the U.S., supported by tariffs approved by state public utility commissions, though penetration remained limited until equipment costs declined and standards solidified.[23][26]Evolution of Caller ID Standards
The standardization of Caller ID in the United States originated with Bellcore (later Telcordia Technologies), which issued Technical Reference TR-TSY-000030 in September 1987 to define Calling Number Delivery (CND), specifying transmission of the caller's telephone number, date, and time via frequency-shift keying (FSK) at 1200 bits per second during the 3-second silent interval preceding the first ring on analog loops. This standard built on earlier Bell Labs prototypes from the 1970s, enabling customer premises equipment to decode the data without disrupting call setup, and was designed for compatibility with existing plain old telephone service (POTS) infrastructure.[27] Subsequent refinements, such as Generic Requirements GR-30-CORE (initially issued in the late 1980s and updated through the 1990s), addressed voiceband data transmission requirements, including signal levels, error correction via checksums, and integration with automatic number identification (ANI) from central office switches.[27] These documents ensured interoperability among regional carriers post-1984 AT&T divestiture, with commercial deployment accelerating after Federal Communications Commission (FCC) tariff approvals in the mid-1980s.[28] By the early 1990s, standards evolved to support enhanced features like caller name delivery, prompting Bellcore's TR-NWT-000031 for Customer Name Delivery (CND with name via database lookup) and TR-NWT-001188 for associated signaling protocols, which introduced selective transmission options and privacy indicators.[27] These updates standardized FSK formatting with Bell 202 modem compatibility, mandatory message headers, and optional subaddressing, while GR-145 for operations systems interfaces facilitated network provisioning. The protocols emphasized reliability through dual-tone detection of channel seizure and mark signals (approximately 2100 Hz and 1300/2100 Hz bursts), with data packets structured as SDMF (Single Data Message Format) for basic number delivery or MDMF for extended information.[17] This phase marked a shift from experimental services—tested in trials since 1984 in areas like New Jersey—to nationwide rollout, with over 20 million subscribers by the late 1990s, though adoption varied due to equipment costs and privacy concerns.[25] Internationally, parallel developments occurred under the European Telecommunications Standards Institute (ETSI), with EN 300 659-1 (first drafted in the early 1990s and formalized by 1998) defining protocols for PSTN display services over local loops, including V.23 modem-based FSK at 1200/75 baud for asymmetric signaling during the ringing phase or pre-ringing.[15] Unlike U.S. standards, ETSI variants supported multiple formats (e.g., DTMF for legacy systems in some regions) and integrated with ISDN via ETSI EN 300 089 for calling line identity, emphasizing sub-addressing and international numbering plans.[11] Key differences included ETSI's use of inverted FSK polarity and optional reverse polarity detection, tailored for diverse European networks, with adoption driven by directives like the 1997 ONP (Open Network Provision) framework for harmonized services.[29] By the 2000s, these standards influenced global ITU-T recommendations, such as Q.731 Annex A for supplementary services, bridging analog POTS to digital transitions while maintaining backward compatibility.[30]Type II Caller ID
Type II Caller ID, also known as Caller ID on Call Waiting (CIDCW), enables the transmission of calling party identification during an ongoing telephone call, allowing the recipient to view incoming caller details without disconnecting the current conversation.[31] Developed as an extension of standard Caller ID protocols, it uses frequency-shift keying (FSK) modulation akin to the Bell 202 standard but adapted for off-hook conditions following a call-waiting alert tone. This contrasts with Type I Caller ID, which delivers data on-hook between the first and second ring on an idle line.[32] In 1995, Bellcore (now Telcordia Technologies) released specifications for Type II signaling to support CIDCW, addressing the limitations of early Caller ID systems that could not function amid active calls.[33] The protocol involves a Channel Seizure Signal (CSS) to interrupt the conversation briefly, followed by a call-waiting tone and then the FSK-encoded data packet containing the caller's number and optional name.[34] This innovation was codified in Telcordia document TR-NWT-000575, "Calling Identity Delivery on Call Waiting," which outlined generic requirements for network interfaces and customer premises equipment (CPE) compatibility. Deployment required subscribers to have both Call Waiting and Caller ID services activated, with CPE devices capable of decoding the off-hook FSK signal at approximately 600 ohms impedance. The standard's evolution integrated with broader CLASS (Custom Local Area Signaling Services) features, including Calling Name Delivery (CNAM), to provide up to 15-digit numbers and alphanumeric names in Single Data Message Format (SDMF) or Multiple Data Message Format (MDMF).[35] By the late 1990s, Type II became widely adopted in North American networks, enhancing user experience in residential and small business settings by reducing the need to answer unsolicited calls blindly.[27] However, it faced challenges like signal attenuation over long loops and compatibility issues with older analog equipment, prompting refinements in subsequent Telcordia updates such as GR-30-CORE for overall Calling Number Delivery.[32] Despite these advancements, Type II's reliance on in-band analog signaling limited its robustness against spoofing, a vulnerability later addressed by digital authentication frameworks.[36]Recent Advancements in Authentication
In 2020, the U.S. Federal Communications Commission (FCC) mandated the implementation of the STIR/SHAKEN framework, a cryptographic protocol suite designed to authenticate caller identities and combat spoofing in IP-based voice networks. Large voice service providers were required to deploy it by June 30, 2021, enabling carriers to sign calls with digital certificates verifying the originating number's legitimacy.[6] By early 2024, this had resulted in a significant increase in authenticated calls, with industry reports noting a tripling of signed call volumes since the mandate's inception, though adoption varied due to technical challenges in legacy systems.[37] Advancements accelerated in 2025 with FCC rules expanding STIR/SHAKEN's scope, including the establishment of a "Call Authentication Trust Anchor" on August 19, which streamlines certificate issuance and boosts the proportion of signed calls by providing more flexible signing options for providers and enterprises.[38] Effective September 18, 2025, third-party authentication services faced new obligations to verify caller ID data, aiming to reduce reliance on intermediaries and enhance direct provider accountability.[39] Concurrently, the FCC initiated proceedings on October 7, 2025, to explore authentication frameworks for non-IP networks, addressing persistent spoofing vulnerabilities in traditional time-division multiplexing systems that remain unmandated for full STIR/SHAKEN compliance.[40] Emerging integrations with branded calling protocols, which overlay caller name and logo transmission atop STIR/SHAKEN signatures, have gained traction; projections indicate these will authenticate over 90 billion calls globally by 2030, driven by enterprise adoption for verified business communications.[41] Additionally, the phase-out of third-party certificates by June 20, 2025, compelled providers to obtain independent Secure Telephone Identity Policy Administrator (STI-PA) tokens, fostering a more decentralized and robust ecosystem less susceptible to single-point failures in authentication chains.[42] These developments reflect ongoing efforts to extend authentication beyond IP domains, though critics note that incomplete non-IP coverage continues to enable cross-network spoofing.[43]Operation
Standard Caller ID Delivery
In traditional Public Switched Telephone Network (PSTN) systems, standard Caller ID delivery transmits the originating telephone number—and optionally the caller's name—using in-band frequency-shift keying (FSK) signaling over the analog voice path during the silent interval between the first and second ring cadence.[44] This method, originally standardized by Bellcore (now Telcordia Technologies) in the late 1980s for North American networks, employs a 1200 bits-per-second FSK modulation scheme based on the Bell 202 (BEL202) protocol, with a mark frequency of 1200 Hz representing a binary 1 and a space frequency of 2200 Hz representing a binary 0.[44][45] The transmission occurs after the termination of the first ring burst, typically starting between 500 milliseconds and 2000 milliseconds later to ensure the recipient's equipment has time to detect and decode the signal before the second ring begins.[46] The FSK data packet follows a structured format to minimize errors and ensure reliable reception by customer premises equipment (CPE), such as dedicated Caller ID boxes or compatible telephones. It begins with a 250-millisecond channel seizure sequence—a repeating 01010101 bit pattern—to alert the receiver and seize the line for data transmission, followed by a 750-millisecond mark hold signal (continuous 1s) for synchronization.[45] The core message payload, encoded in 7-bit ASCII, includes fields for the date (MMDD format), time (HHMM format), calling number (up to 10 digits for North American numbering), and optionally the caller's name (up to 15 characters) if Caller Name (CNAM) service is provisioned; the packet concludes with two checksum bytes (a sum-check and longitudinal redundancy check) for error detection.[44][45] This delivery relies on the originating central office switch querying the caller's line information from the network database and injecting the FSK burst into the signaling stream toward the terminating switch, which then forwards it to the subscriber's line only if Caller ID service is enabled on both ends.[5] Successful decoding requires compatible CPE with FSK demodulation capabilities, as standard analog telephones lack this functionality; early implementations often used external boxes connected in series with the phone line to capture and display the data on an LCD screen.[44] Transmission power is limited to avoid interfering with ring detection, typically around -12 dBm, and the entire burst lasts under one second to fit within the inter-ring gap of approximately 4-6 seconds in standard ring cadences.[46] While effective for legitimate calls, this analog in-band method provides no inherent cryptographic verification, allowing network operators or intermediaries to alter the transmitted number, though widespread deployment began after regulatory mandates in the U.S. by the mid-1990s required accurate delivery where service was offered.[5]Alternative Signaling Methods
In addition to the standard on-hook Frequency Shift Keying (FSK) delivery used in many analog Public Switched Telephone Network (PSTN) systems, Dual-Tone Multi-Frequency (DTMF) signaling serves as an alternative method for transmitting caller identification data. DTMF encodes each digit of the calling number as a unique pair of audio tones, transmitted sequentially over the line, typically before the first ring in compatible systems. This approach, prevalent in regions such as the Netherlands, supports basic number transmission but lacks the capacity for additional metadata like date, time, or name that FSK enables.[47][48] Type II caller ID represents another variant, designed for off-hook delivery during active calls, such as in call-waiting scenarios. Unlike Type I (on-hook) signaling, Type II begins with a Channel Seizure Signal or Customer Alert Signal to interrupt the ongoing conversation, followed by FSK-modulated data bursts containing the incoming caller's details. Introduced in the mid-1990s to support features like Caller ID with Call Waiting, this method requires compatible customer premises equipment to detect the alerting tones and decode the subsequent FSK without disrupting the primary call.[49][31] In digital telephony environments, such as Integrated Services Digital Network (ISDN), caller ID is delivered via out-of-band signaling protocols rather than in-band audio modulation like FSK or DTMF. For Basic Rate Interface (BRI) or Primary Rate Interface (PRI) lines, the calling party number is embedded in the Q.931 setup message transmitted over the D-channel, a dedicated control channel separate from the bearer (B-channel) used for voice. This digital approach allows for more reliable and higher-bandwidth transmission of identification data, including support for international formats and supplementary services, without interrupting the voice path. ISDN systems thus provide caller ID as an inherent feature of the signaling layer, contrasting with the analog PSTN's requirement for post-switching data injection.[50]Integration with VoIP and Mobile Networks
In Voice over IP (VoIP) systems, Caller ID is integrated through the Session Initiation Protocol (SIP), the predominant signaling protocol for initiating and managing multimedia sessions. The caller's telephone number and identity details are conveyed in SIP message headers, primarily the From header for the displayed caller ID, with additional headers like P-Asserted-Identity (PAI) and Remote-Party-ID (RPID) providing authenticated or privacy-suppressed information depending on network policies.[51] These headers are populated during the SIP INVITE request, enabling VoIP providers to set outbound Caller ID by configuring the From header with a legitimate number.[52] For name display, VoIP services often query databases like CNAM (Caller Name) via protocols integrated into the IP network, though delivery relies on the originating provider's assertions.[53] Mobile networks integrate Caller ID differently based on generation and architecture. In legacy GSM and CDMA networks, which dominate 2G/3G deployments, Caller ID—or Calling Line Identification (CLI)—is delivered out-of-band via SS7 (Signaling System No. 7) protocols during call setup, transmitting the caller's number to the recipient's device before the ringing signal.[54] By the early 2000s, Caller ID became standard on U.S. mobile phones, often bundled in service plans without additional fees.[55] Modern 4G/5G mobile networks employ the IP Multimedia Subsystem (IMS), an IP-based framework that unifies voice and data services, using SIP signaling akin to VoIP for Caller ID delivery—mapping the caller's identity into SIP headers for consistency across packet-switched environments.[56] Interoperability between VoIP and mobile networks requires gateways to convert signaling protocols and preserve Caller ID data. Traditional mobile and PSTN traffic using SS7 ISUP is translated by SS7-to-SIP gateways, which extract the calling party number from SS7 messages and insert it into corresponding SIP headers, such as the From or PAI fields, during protocol conversion.[57] Devices like the Vertex Caller ID converter facilitate compatibility by generating analog-like Caller ID signals for VoIP phones interfacing with legacy systems, supporting SIP but excluding encrypted protocols.[58] In hybrid scenarios, such as VoIP calls over mobile data or IMS-bridged interconnections, challenges arise in maintaining Caller ID integrity, particularly for international routing where format mismatches or privacy regulations can suppress or alter displayed information.[59] Integration via IMS in mobile cores enables seamless VoIP handoff, but gateway misconfigurations can lead to incomplete CLI propagation, as seen in some SIP trunk setups.[60]Authentication Mechanisms
STIR/SHAKEN Framework
The STIR/SHAKEN framework comprises a set of industry-developed protocols and procedures aimed at verifying the authenticity of caller identification information transmitted over IP-based telephone networks, primarily to mitigate caller ID spoofing and associated fraudulent activities.[6] STIR, or Secure Telephone Identity Revisited, defines the structure of a compact, cryptographically signed token called PASSporT that encapsulates asserted caller details, such as the originating telephone number and associated metadata. SHAKEN, or Signature-based Handling of Asserted information using toKENs, specifies the operational processes for generating, signing, and validating these tokens using public key infrastructure (PKI), including digital certificates issued by Policy Administrators to eligible voice service providers (VSPs).[61] Together, these components enable originating VSPs to attest to the legitimacy of calls from their subscribers, with terminating VSPs verifying signatures to assess risk before delivery.[6] The framework emerged from collaborative efforts by the Alliance for Telecommunications Industry Solutions (ATIS) and the Internet Engineering Task Force (IETF), with initial standards published around 2017–2018 to address escalating robocall volumes exceeding 30 billion monthly in the U.S. by 2019. In December 2020, the Federal Communications Commission (FCC) mandated its adoption under 47 CFR Part 64 Subpart HH, requiring all covered providers to implement STIR/SHAKEN in the IP portions of their networks: facilities-based providers with over 100,000 subscriber lines by June 30, 2021, and smaller or non-IP providers by June 30, 2023, with extensions granted for full compliance demonstrations.[61] Non-compliance triggers robocall mitigation plans, and by September 28, 2023, the FCC expanded requirements to include traceback participation and certification filings, while a September 18, 2025, deadline enforced "Third Party Rules" prohibiting reliance on upstream providers' certificates without direct attestation.[6] As of October 2025, implementation covers over 90% of U.S. voice traffic, though interoperability challenges persist in international and legacy TDM networks.[6] Technically, STIR/SHAKEN operates within Session Initiation Protocol (SIP) signaling: the originating VSP queries its PKI certificate, constructs a PASSporT token with hashed caller data, signs it using elliptic curve cryptography (typically ECDSA with SHA-256), and embeds it in the SIP Identity header. Verifying VSPs check the signature against the issuer's public key, validate token integrity, and apply policy-based handling, such as prioritizing attested calls or flagging unauthenticated ones.[62] The framework relies on a consortium of Policy Administrator Authorities (PAAs), such as those operated by iconectiv and Neustar (now TransUnion), to vet and issue certificates, ensuring only authorized entities can sign calls.[63] While effective for domestic IP-to-IP traffic, it excludes end-to-end encryption protocols like those in some VoIP apps and requires gateway conversions for PSTN interconnections, limiting universal applicability.[6]Attestation Levels and Verification
In the STIR/SHAKEN framework, originating voice service providers assign one of three attestation levels—A, B, or C—to authenticated calls, reflecting the degree of confidence in the caller's identity and authorization to use the presented telephone number.[64][65] This assignment occurs during call origination, where the provider embeds the level within a cryptographically signed PASSporT token, which travels with the call signaling through IP networks.[65] The levels enable downstream providers to assess risk without revealing sensitive subscriber data, with A indicating the strongest verification and C the weakest.[64][66]| Attestation Level | Description | Risk Implications |
|---|---|---|
| A (Full) | The originating provider has authenticated the calling party as the subscriber associated with the number and confirmed their authorization to use it, typically through direct customer knowledge or database verification.[64][67][66] | Lowest risk; calls are generally trusted and may receive preferential treatment, such as reduced likelihood of blocking or spam labeling by terminating providers.[64] |
| B (Partial) | The originating provider has identified the calling party but lacks full assurance of their authorization for the specific number, often due to indirect verification or shared infrastructure.[65][64] | Moderate risk; indicates some vetting but prompts caution, potentially leading to additional scrutiny or analytics checks.[64] |
| C (Gateway) | The originating provider cannot authenticate the calling party, commonly for calls entering the U.S. from international gateways or unverified sources where no identity check is feasible.[65][64] | Highest risk; signals minimal trust, often resulting in calls being flagged, blocked, or subjected to heightened robocall mitigation.[64] |