ATIS 1000064-2014 Transitional Circuit Switched Packet Switched Services.pdf

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1、 AMERICAN NATIONAL STANDARD FOR TELECOMMUNICATIONS ATIS-1000064 Transitional Circuit Switched/Packet Switched Services As a leading technology and solutions development organization, ATIS brings together the top global ICT companies to advance the industrys most-pressing business priorities. Through

2、 ATIS committees and forums, nearly 200 companies address cloud services, device solutions, emergency services, M2M communications, cyber security, ehealth, network evolution, quality of service, billing support, operations, and more. These priorities follow a fast-track development lifecycle from d

3、esign and innovation through solutions that include standards, specifications, requirements, business use cases, software toolkits, and interoperability testing. ATIS is accredited by the American National Standards Institute (ANSI). ATIS is the North American Organizational Partner for the 3rd Gene

4、ration Partnership Project (3GPP), a founding Partner of oneM2M, a member and major U.S. contributor to the International Telecommunication Union (ITU) Radio and Telecommunications sectors, and a member of the Inter-American Telecommunication Commission (CITEL). For more information, visit . AMERICA

5、N NATIONAL STANDARD Approval of an American National Standard requires review by ANSI that the requirements for due process, consensus, and other criteria for approval have been met by the standards developer. Consensus is established when, in the judgment of the ANSI Board of Standards Review, subs

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8、 National Standards Institute does not develop standards and will in no circumstances give an interpretation of any American National Standard. Moreover, no person shall have the right or authority to issue an interpretation of an American National Standard in the name of the American National Stand

9、ards Institute. Requests for interpretations should be addressed to the secretariat or sponsor whose name appears on the title page of this standard. CAUTION NOTICE: This American National Standard may be revised or withdrawn at any time. The procedures of the American National Standards Institute r

10、equire that action be taken periodically to reaffirm, revise, or withdraw this standard. Purchasers of American National Standards may receive current information on all standards by calling or writing the American National Standards Institute. Notice of Disclaimer do-it-yourself tools for the layma

11、n are plentiful in todays market. As a result, Caller ID-based services have become more susceptible to such fraud. Congress is considering legislation to protect consumers against such fraud (i.e., caller ID spoofing). Telecom service providers are investigating verification solutions to increase t

12、he reliability and authenticity of the delivered calling numbers, but those solutions are outside the scope of this TR. It should be noted that there are legitimate reasons for changing the original CPN in some cases. However, from a purely operational perspective, the capability to change the origi

13、nal CPN may still lead to incorrect service operation and fraud. 5.2.1 Anonymous Call Rejection (ACR) Anonymous Call Rejection allows customers to reject calls from lines that have a privacy feature preventing the delivery of their CPN to the customer. Calls with a CPN that is anonymous are routed t

14、o a network announcement and the customer does not receive alerting for the call. The announcement informs the calling party that the customer does not accept anonymous calls and the caller would have to unblock his/her number to reach the customer. The NANP activation/deactivation codes for ACR are

15、 *77/*87, respectively. ACR is described in GR-567-CORE. 5.2.1.1 All-IP Network Implementation There is an equivalent mechanism to ACR described for SIP users in RFC 5079, Rejecting Anonymous Requests in SIP. This RFC introduced the 433 (Anonymity Disallowed) response code to specifically indicate t

16、he reason for rejection. This allows for proper automated machine handling; not just human interpretation. 5.2.1.2 PSTN-IP Interworking This feature relies on the delivery of the presentation status of the calling party number to the called party. ATIS-1000679 attends to the SIP-ISUP requirements ha

17、ndling the translation of the necessary presentation status indicators. However, feature-specific elements necessary for interworking are currently not supported, as explained below. Interworking with an IP ACR Subscriber On calls from an SS7 caller to the SIP ACR subscriber, the ISUP Gateway will m

18、ap the IAM Calling Party Number parameter to the SIP P-Asserted-Identity header fields according to ATIS-1000679, Section 7.1.3, where the privacy header content will be determined from the APRI value of the Calling Party Number. ATIS-1000064 14 If the Calling Party Number was omitted/not available,

19、 or if the APRI value is “presentation allowed,” the incoming call/session is offered to the ACR subscriber. If the APRI value is “presentation restricted,” the user agent serving the ACR subscriber is expected to refuse to fulfill the request by sending the 433 response code defined in RFC 5079 for

20、 this purpose. The mapping for the 433 response code is not included in the list of 4xx codes recognized by the ATIS-1000679 IWF. It is likely that this message would be rejected at the PSTN Gateway. Interworking with a PSTN ACR Subscriber On calls from SIP callers to the PSTN ACR subscriber, ATIS-1

21、000679 IWF will map the SIP P-Asserted-Identity header fields to ISUP Calling Line Identification parameters. The SS7 switch serving the ACR subscriber is expected to follow procedures in GR-567-CORE and the call will not be terminated to the ACR subscriber. In addition, the switch plays an audio an

22、nouncement indicating that the called party does not accept anonymous calls. Since the announcement is played from the switch serving the ACR subscriber, there is no signaling interworking back to the caller. 5.2.1.3 Identified Gaps A SIP 433 response code to an anonymous PSTN caller is expected to

23、cause release treatment on the PSTN side, causing confusion on the part of the PSTN caller, potential reattempted calls that will result in repeated failures. To avoid inconsistent implementations of ACR between the PSTN and IP, it is strongly recommended that an audio announcement be generated BEFO

24、RE sending the 433 response code. This way, anonymous callers would receive a clear indication of why their calls are not being completed. 5.2.2 Automatic Callback (AC) and Automatic Recall (AR) GR-215-CORE defines Automatic Callback as: An outgoing call management feature that allows customers to p

25、erform an activation procedure to set up a call to the last station that the customer called without the customer having to redial the telephone number. If the called party is busy when AC is activated, call setup is performed automatically when the called station becomes idle. The NANP activation/d

26、eactivation codes for AC are *66/*86, respectively. GR-227-CORE defines Automatic Recall as: An incoming call management feature that allows a customer to perform an activation procedure to automatically set up a call to the last incoming number. The AR customer does not need to know the telephone n

27、umber or the calling party of the last incoming call. The NANP activation/deactivation codes for AR are *69/*89, respectively. ATIS-1000611 defines AC and AR in the context of a Multi-Location Business Group, noting that: AC and AR are very similar MBG features. The AC feature attempts to call the n

28、umber associated with the most recent outgoing call made by the customer. The AR feature attempts to call the number associated with the most recent incoming call received by the customer. 5.2.2.1 All-IP Network Implementation Comparable AC/AR monitoring of the busy/idle status of the terminating pa

29、rty could be accomplished through SUBSCRIBE and NOTIFY requests as described in RFC3842 and RFC 6665, Session Initiation Protocol (SIP)-Specific Event Notification. The calling user agent sends a SUBSCRIBE requesting to be notified when the terminating user agent becomes available. The terminating u

30、ser agent would send a NOTIFY response when it becomes available. If the calling user agent is available, a new INVITE is sent and the session is established. In addition, RFC 6910 defines “Completion of Calls for SIP,” which uses SUBSCRIBE-NOTIFY to allow the caller of a failed call to have the cal

31、l completed without having to make a new call attempt. RFC 5359 ATIS-1000064 15 provides a call flow example using the name “Automatic Redial”. The feature closely resembles the PSTN AC/AR services and offers more functionality that is suitable for SIP implementation. NOTE: Currently, there is no SI

32、P EVENT type describing voice media explicitly. It is possible to assume EVENT: available for all media types, including voice, thus supporting the use of SUBSCRIBE-NOTIFY to implement AC and AR. 5.2.2.2 PSTN-IP Interworking In the PSTN, the monitoring of the busy/idle status of the line to be calle

33、d may be carried out by (1) the originating switch launching periodic TCAP queries to the terminating end requesting the target status, or (2) the originating switch requesting that the terminating switch perform terminating scanning and inform it only when the called party becomes available. Compar

34、ing the capabilities available in the SIP protocol to those necessary to deliver an equivalent AC experience to its end users, the SUBSCRIBE and NOTIFY methods appear to satisfy most of these capabilities effectively. SUBSCRIBE-NOTIFY method is expected to be the prevalent method for implementing an

35、 event-based service, such as AC, between two SIP users. This TR is not the appropriate vehicle for feature requirements, therefore the following text is provided on an informational basis, highlighting potential areas for investigation. A Service Transactions Gateway will potentially perform the TC

36、AP-SIP mapping and provide the necessary monitoring and notification. Standardizing the translation function (TCAP Components into SIP headers and vice versa), managing transaction IDs, busy/idle scanning requests, and handling errors would be some of the key responsibilities of the Service Transact

37、ions Gateway. 5.2.2.3 Identified Gaps Interworking an AC request between SIP and SS7 is not defined in standards today. Informative interworking guidelines need to be added to ATIS-1000051. AC services based on GR-215-CORE include an initial TCAP Query requesting information about the services, feat

38、ures, and restrictions on the terminating line prior to initiation of scanning for the line to become idle. The functionality provided by this initial Query is not available in NGN functionality provided by the SUBSCRIBE-NOTIFY method. SIP extensions, service-specific TCAP-to-SIP interworking, or AC

39、 use of TCAP encapsulation as described in ATIS-100051 may need to be standardized. 5.2.3 Customer Originated Trace (COT) Customer Originated Trace allows the recipient of harassing calls to initiate a trace of the last call presented to the COT customers line. COT information (i.e., CPN, date, time

40、 associated with the call, and the date and time of the trace activation) is sent to a designated retention location. The information is therefore potentially available for further use by the service provider and/or law enforcement. If the captured incoming number is invalid (i.e., unavailable or pa

41、rtial), the customer is informed that a trace cannot be done and an announcement is given to indicate why. In some offerings, the announcement includes the contact information for a local Law Enforcement Agency. The NANP activation code for COT is *57. COT is described in GR-216-CORE. The interface

42、that delivers the trace information to the retention/processing location is outside the scope of this TR. This TR assumes the agency (Law Enforcement Agency or other) that receives and retains the COT data resides locally, within the same network (PSTN or IP) as the COT customer. 5.2.3.1 All IP Impl

43、ementation Commercial VoIP offerings of COT service already exist. While there are no standard descriptions of COT in an IP environment, RFC 5503 introduces SIP extensions that support COT. ATIS-1000064 16 5.2.3.2 PSTN-IP Interworking For this feature to be successfully activated, the CPN of the cal

44、ler must be available (anonymous CPNs are captured for the Trace log but are not revealed to the COT subscriber). SS7 to SIP Interworking Scenario On calls from an SS7 caller to a SIP COT subscriber, the ISUP Gateway will map the IAM Calling Party Number parameter to the SIP P-Asserted-Identity head

45、er fields according to ATIS-1000679. Therefore, the required Calling Party Number and privacy indication will be delivered to the terminating end and will be available for logging as a COT is activated by the called party. SIP to SS7 Interworking Scenario On calls from SIP callers to the SS7 COT sub

46、scriber, the ISUP Gateway will map the SIP P-Asserted-Identity header fields to ISUP Calling Line Identification parameters according to ATIS-1000679. Therefore, the required Calling Party Number and privacy indication will be delivered to the terminating end and will be available for logging as a C

47、OT is activated by the called party. 5.2.3.3 Identified Gaps RFC 5503 SIP uses an INVITE to deliver the COT data to the designated agency, except as noted in the excerpt from RFC 5503 below, for a subset of cases. If the INVITE is forwarded within the IP network, the COT request will complete as int

48、ended. However, if the INVITE traverses a PSTN gateway (i.e., the COT customer and the designated agency are served by different networks one SIP-based and the other SS7 the COT data and call content will be lost and the request will fail. “To initiate a customer-originated-trace from an untrusted U

49、ser Agent Client (UAC), an additional header is defined for the INVITE request.The entity addressed by the Request-URI performs the service-provider specific functions of recording and reporting the caller identity in the P-DCS-Trace-Party-ID for law enforcement action. It then forwards the call to either an announcement server or to the service providers business office to collect further information about the complaint. A trusted UAC does not use this header, as it initiates this action locally.” 5.2.4 Calling Number Delivery (CND)