ATIS 1000081-2018 ATIS Technical Report on a Framework for Display of Verified Caller ID.pdf
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1、 ATIS-1000081 ATIS Standard on ATIS Technical Report on a Framework for Display of Verified Caller ID Alliance for Telecommunications Industry Solutions Approved May 2018 Abstract This technical report provides a framework for signaling verified Caller ID information from the network to a User Equip
2、ment (UE) and displaying the information on the UE in a uniform manner, independent of technology. The main goal is to provide display guidelines that help empower consumers in managing their calls, as per the Robocalling Strike Force recommendations. ATIS-1000081 ii Foreword The Alliance for Teleco
3、mmunication Industry Solutions (ATIS) serves the public through improved understanding between providers, customers, and manufacturers. The Packet Technologies and Systems Committee (PTSC) develops and recommends standards and technical reports related to services, architectures, and signaling, in a
4、ddition to related subjects under consideration in other North American and international standards bodies. PTSC coordinates and develops standards and technical reports relevant to telecommunications networks in the U.S., reviews and prepares contributions on such matters for submission to U.S. Int
5、ernational Telecommunication Union Telecommunication Sector (ITU-T) and U.S. ITU Radiocommunication Sector (ITU-R) Study Groups or other standards organizations, and reviews for acceptability or per contra the positions of other countries in related standards development and takes or recommends appr
6、opriate actions. The SIP Forum is an IP communications industry association that engages in numerous activities that promote and advance SIP-based technology, such as the development of industry recommendations, the SIPit, SIPconnect-IT, and RTCWeb-it interoperability testing events, special worksho
7、ps, educational seminars, and general promotion of SIP in the industry. The SIP Forum is also the producer of the annual SIP Network Operators Conference (SIPNOC), focused on the technical requirements of the service provider community. One of the Forums notable technical activities is the developme
8、nt of the SIPconnect Technical Recommendation a standards-based SIP trunking recommendation for direct IP peering and interoperability between IP Private Branch Exchanges (PBXs) and SIP-based service provider networks. Other important Forum initiatives include work in Video Relay Service (VRS) inter
9、operability, security, Network-to-Network Interoperability (NNI), and SIP and IPv6. Suggestions for improvement of this document are welcome. They should be sent to the Alliance for Telecommunications Industry Solutions, PTSC, 1200 G Street NW, Suite 500, Washington, DC 20005, and/or to the SIP Foru
10、m, 733 Turnpike Street, Suite 192, North Andover, MA, 01845. The mandatory requirements are designated by the word shall and recommendations by the word should. Where both a mandatory requirement and a recommendation are specified for the same criterion, the recommendation represents a goal currentl
11、y identifiable as having distinct compatibility or performance advantages. The word may denotes an optional capability that could augment the standard. The standard is fully functional without the incorporation of this optional capability. The ATIS/SIP Forum IP-NNI Task Force under the ATIS Packet T
12、echnologies and Systems Committee (PTSC) and the SIP Forum Technical Working Group (TWG) was responsible for the development of this document. ATIS-1000081 iii Table of Contents Scope, Purpose, This document is available from the Federal Communications Commission at . 2This document is available fro
13、m the Alliance for Telecommunications Industry Solutions (ATIS) at . ATIS-1000081 2 Ref 2 3GPP TS 22.173, IP Multimedia Core Network Subsystem (IMS) Multimedia Telephony Service and supplementary services.3Ref 3 3GPP TS 24.196, Technical Specification Group Core Network and Terminals; Enhanced Calli
14、ng Name.4Ref 4 IETF RFC 8224: “Authenticated Identity Management in the Session Initiation Protocol (SIP)“.5Ref 5 3GPP TS 24.229: “IP Multimedia call control protocol based on Session Initiation Protocol (SIP) and Session Description Protocol (SDP); Stage 3“.6Ref 6 3GPP TS 29.165: “Inter-IP Multimed
15、ia System (IMS) Network to Network Interface (NNI) (Release 15)“.73 Definitions, Acronyms, i.e., an “Anonymous“ message would be displayed for name and TN. a. The terms of agreement between the service provider and the analytics provider are expected to require preservation of the callers privacy. b
16、. Further clarification and/or safe harbors are needed to determine whether the name and TN could be anonymized while additional call information, such as call category and potential fraud risk, could be delivered to help empower the end user. 5) Where delivery of the final call information display
17、is not delegated to an analytics provider, the service provider is expected to adhere to the application of privacy rules for the calling number and name portions of the display; i.e., an “Anonymous“ message would be displayed for name and TN. 6) In the absence of clarification or any special allowa
18、nces on handling private calls, the guidelines herein call for anonymizing ALL call information if the received calling number is anonymous. The determination of the authentication of the call is NOT “call information” in the traditional sense and should be presented to the user independent of legit
19、imate privacy uses. 7) The end user may subscribe to multiple mitigation services. Order and preference of display may be determined by the service provider. Variations in operating systems and the plethora of available applications make it impractical to set interoperability or prioritization guide
20、lines. Therefore, it is the responsibility of end users to select the mitigation service that best meets their needs. 5.3 Available Call-related Information The data outputs from the network and CVT will be at the center of the message delivered to the user (e.g., warning or other). Making more reli
21、able information available to the CVT algorithms is likely to yield more accurate results for the user. Therefore, it is recommended that attestation levels and identifiers from SHAKEN be made available to the CVT function, according to local policies. 5.4 Preliminary Display Usability & Comprehensi
22、on Studies Over the course of Q2 and Q3 2017, Hiya Inc. conducted several usability studies targeted at the display guidelines of the STIR/SHAKEN protocol. The studies and their results are provided later in this clause. The goals of the study were twofold: 1. To measure the potential impact of a po
23、sitive assurance indication for verified calls (the “green checkmark”). 2. To assess various textual and iconographic display options for caution indicators on suspicious calls. ATIS-1000081 8 Hiya conducted three brief usability studies on different audiences: 1. A user comprehension and influence
24、study on robust caller profiles and certified call markers. 2. A user impact analysis of various phrasings and iconography for suspicious call messaging. 3. A call pickup rate impact analysis of a “certified” checkmark icon against existing Hiya users. 5.4.1 Study #1: Comprehension & Impact of Certi
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