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7.1.1 Test purposes for Basic call, Successful
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Test case number SS_bcall_001 Test case group BCALL/successful Reference [4] SELECTION EXPRESSION Test purpose Basic call normal call clearing from the called user. Ensure that call establishment is performed correctly. In the active call state ensure the property of speech. The call is released from the called user. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 100 Trying 180 Ringing 200 OK INVITE ACK Communication BYE 200 OK BYE Comments Establish a communication from network A to Network B Check: Ensure the property of speech. Check: Are the media streams terminated after the 200 OK BYE was sent? Repeat this test in reverse direction. Repeat this test with all chosen end devices. Test case number SS_bcall_002 Test case group BCALL/successful Reference [4] SELECTION EXPRESSION Test purpose Basic call normal call clearing from the calling user. Ensure that call establishment is performed correctly. In the active call state ensure the property of speech. The call is released from the calling user. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 100 Trying 180 Ringing 200 OK INVITE ACK Communication BYE 200 OK BYE Comments Establish a communication from network A to Network B Check: Ensure the property of speech. Check: Are the media streams terminated after the 200 OK BYE was sent? Repeat this test in reverse direction. Repeat this test with all chosen end devices. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 16 Test case number SS_bcall_003 Test case group BCALL/successful Reference 8/ [1] SELECTION EXPRESSION Test purpose Request line in the INVITE. Ensure that the Request line in the INVITE contains in the user part the telephone number of the destination user equipment formatted as a 'tel' URI in the global number format and the host portion is set to the host name of the interconnected network. The user URI parameter is present set to 'phone'. Configuration SIP Parameter INVITE Request line Address of user B @ network B;user=phone Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Establish a communication from network A to Network B Check: The user part is in the format of a tel URI in global number format. Check: The host portion is set to the host name of the interconnected network. Check: The user parameter is set to phone. Repeat this test in reverse direction. Repeat this test with all chosen end devices. Test case number SS_bcall_004 Test case group BCALL/successful Reference 5.10/ [2] Testspec Reference SELECTION EXPRESSION SE 1 Test purpose P-Charging-Vector header in the INVITE. Ensure that the P-Charging-Vector header is present in the INVITE establishes a communication between a user of network A and a user of network B and the 'icid-value' and the 'orig-ioi' parameter is present. Configuration SIP Parameter INVITE P-Charging-Vector: icid-value; orig-ioi Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Establish a communication from network A to Network B Check: The P-Charging-Vector header contains the icid-value parameter. Check: The P-Charging-Vector header contains the orig-ioi parameter. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 17 Test case number SS_bcall_005 Test case group BCALL/successful Reference 5.10/ [2] Testspec Reference SELECTION EXPRESSION SE 2 Test purpose P-Charging-Vector header in the INVITE. Ensure that the P-Charging-Vector header is present in the INVITE establishes a communication between a user of network A and a user of network B and the 'icid-value' or the 'orig-ioi' parameter is present. Configuration SIP Parameter INVITE P-Charging-Vector: icid-value; orig-ioi Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Establish a communication from network A to Network B Check: The P-Charging-Vector header contains the icid-value parameter (optional). Check: The P-Charging-Vector header contains the orig-ioi parameter (optional). Repeat this test in reverse direction. Test case number SS_bcall_006 Test case group BCALL/successful Reference 8/ [21] SELECTION EXPRESSION [Network A] SE 3 Test purpose P-Early-Media header support indication in the initial INVITE request. Ensure that the support of the P-Early. Media header is indicated in the initial INVITE request. A P-Early-Media header is present set to 'supported'. Configuration SIP Parameter INVITE P-Early-Media: supported SDP Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Establish a communication from network A to Network B Check: Is a P-Early-Media header present in the INVITE request? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 18 Test case number SS_bcall_007 Test case group BCALL/successful Reference 8/ [21] SELECTION EXPRESSION [Network A] SE 3 AND [Network B] SE3 Test purpose P-Early-Media header supported in early dialogue. Ensure that an early dialogue is established by sending a 183 Session Progress or 180 Ringing from Network B and the P-Early-Media header is present authorizes early media. Configuration SIP Parameter INVITE P-Early-Media: supported SDP 183 P-Early-Media: [any value authorizes early media] SDP OR 180 P-Early-Media: [any value authorizes early media] SDP Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE CASE A 183 Session Progress CASE B 180 Ringing Apply post test routine Comments Establish a communication from network A to Network B Check: Is a 183 or 180 send from Network B to establish an early dialogue? Check: Is an SDP present in the 183 as a SDP answer? Check: A bearer transmission is possible in backward directions. NOTE: The absence of the direction parameter of an 'a' line represents the default value 'sendrecv' NOTE: The presence of the P-Early-Media header in the INVITE request indicates the support of "early media Authorization" in the originating Network. NOTE: The presence of the P-Early-Media header in the 183 or 180 indicates the support of the P-Early-Media header and authorizes the media in the early dialogue Repeat this test in reverse direction. Repeat this test by a call setup to an announcement application. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 19 Test case number SS_bcall_009 Test case group BCALL/successful Reference 8/ [21] SELECTION EXPRESSION [Network A] SE 3 AND [Network B] SE 3 AND SE 25 AND SE 30 Test purpose P-Early-Media header supported early dialogue with 181. Ensure that an early dialogue is established by sending a 181 Call Is Being Forwarded from Network B and the P-Early-Media header is present authorizes early media. The Call is forwarded in network B. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes SIP Parameter INVITE P-Early-Media: supported SDP 181 P-Early-Media: [any valu authorizes early media] Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Call Is Being Forwarded Apply post test routine Comments Establish a communication from network A to Network B Check: Is a 181 sent from Network B to establish an early dialogue? Check: Is an SDP present in the 181 as a SDP answer? NOTE: The presence of the P-Early-Media header in the INVITE request indicates the support of "early media Authorization" in the originating Network. NOTE: The presence of the P-Early-Media header in the 181 indicates the support of the P-Early-Media header and authorizes the media in the early dialogue Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 20 Test case number SS_bcall_010 Test case group BCALL/successful Reference 8/ [21] SELECTION EXPRESSION [Network A] SE 3 AND [Network B] SE 3 AND SE 35 Test purpose P-Early-Media header supported early dialogue with 182. Ensure that an early dialogue is established by sending a 182 Queued from Network B and the P-Early-Media header is present authorizes early media. The Call is a waiting call in network B. Configuration SIP Parameter INVITE P-Early-Media: supported SDP 182 P-Early-Media: [any value authorizes early media] Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 182 Call Is Being Forwarded Apply post test routine Comments Establish a communication from network A to Network B Check: Is a 181 sent from Network B to establish an early dialogue? Check: Is an SDP present in the 182 as a SDP answer? NOTE: The presence of the P-Early-Media header in the INVITE request indicates the support of "early media Authorization" in the originating Network. NOTE: The presence of the P-Early-Media header in the 182 indicates the support of the P-Early-Media header and authorizes the media in the early dialogue Repeat this test in reverse direction. Test case number SS_bcall_011 Test case group BCALL/successful Reference 5.10/ [2] SELECTION EXPRESSION Test purpose Record-route header in the INVITE. Ensure that if the Record-Route header is present in the INVITE establishes a communication between a user of network A and a user of network B the topmost header is set to the IBCF of network A. Configuration SIP Parameter INVITE Record-Route: <Address of IBCF in network A> Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Establish a communication from network A to Network B Check: If present the topmost Record-Route header or entry contains the address of the IBCF of network A. Repeat this test in reverse direction. Repeat this test with all chosen end devices. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 21 Test case number SS_bcall_012 Test case group BCALL/successful Reference 5.10/ [2] SELECTION EXPRESSION Test purpose Via header in the INVITE. Ensure that the Via header is present in the INVITE establishes a communication between a user of network A and a user of network B and the topmost header is set to the IBCF of network A and contains a branch parameter. Configuration SIP Parameter INVITE Via: <Address of IBCF in network A>; branch=[any value] Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Establish a communication from network A to Network B Check: The topmost Via header contains the Address of IBCF in network A and a branch parameter. Repeat this test in reverse direction. Repeat this test with all chosen end devices. Test case number SS_bcall_013 Test case group BCALL/successful Reference 5.10/ [2] SELECTION EXPRESSION Test purpose Record-Route header in the 180 Ringing. Ensure if a Record-Route header was present in the initial INVITE that the Record-Route header is present in the 180 Ringing provisional response as the first response from network B upon a connection establish setup from network A. Configuration SIP Parameter INVITE Record-Route 180: Record-Route Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Apply post test routine Comments Establish a communication from network A to Network B Check: If the Record-Route header is present is in the 180 Ringing. The Record-Route header is optional. Repeat this test in reverse direction. Repeat this test with all chosen end devices. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 22 Test case number SS_bcall_014 Test case group BCALL/successful Reference 5.10/ [2] SELECTION EXPRESSION Test purpose Route header in the BYE of the originating user. Ensure that if a Record-Route header was present in the initial INVITE the Route header may be present in the BYE request sent from the originating user equipment in network A the topmost Route header or entry is set to the IBCF of network B. Configuration SIP Parameter BYE: Route: <Address of IBCF in network B>;lr, …. Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists BYE 200 OK BYE Apply post test routine Comments Establish a communication from network A to Network B Check: Is the Route header present in the BYE, the topmost header or entry is set to the address of the IBCF of network B. Repeat this test in reverse direction. Repeat this test with all chosen end devices. Test case number SS_bcall_015 Test case group BCALL/successful Reference 5.10/ [2] SELECTION EXPRESSION Test purpose Route header in the BYE of the terminating user. Ensure that if a Record-Route header was present in the initial INVITE the Route header may be present in the BYE request sent from the terminating user equipment in network B the topmost Route header or entry is set to the IBCF of network A. Configuration SIP Parameter BYE: Route: <Address of IBCF in network A>;lr, …. Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists BYE 200 OK BYE Apply post test routine Comments Establish a communication from network A to Network B Check: If the Route header present in the BYE, the topmost header or entry is set to the address of the IBCF of network A. Repeat this test in reverse direction. Repeat this test with all chosen end devices. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 23 Test case number SS_bcall_016 Test case group BCALL/successful Reference 5.10/ [2] SELECTION EXPRESSION Test purpose Route header in the ACK. Ensure that if a Record-Route header was present in the initial INVITE the Route header may be present in ACK from network A upon a connection establishment from network A is completed the topmost Route header or entry is set to the IBCF of network B. Configuration SIP Parameter ACK: Route: <Address of IBCF in network B>;lr, …. Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK Apply post test routine Comments Establish a communication from network A to Network B Check: Is the Route header present in the ACK, the topmost header or entry is set to the address of the IBCF of network B. Repeat this test in reverse direction. Repeat this test with all chosen end devices. Test case number SS_bcall_017 Test case group BCALL/successful Reference [4] and [5] SELECTION EXPRESSION Test purpose Handling of SDP parameters in the INVITE. Ensure that call establishment and the correct handling of the SDP parameters of the INVITE message defined as: TYPE_SDP is performed correctly. Ensure that in the active call state the voice/data transfer on the media channels is performed correctly (e.g. testing QoS parameters). In case when the parameter in the SDP rtpmap:<dynamic-PT> is used the codecs in table 7.1.1-1 applies. Configuration SIP Parameter INVITE: Content-Type: application/sdp m=audio <Port number> RTP/AVP TYPE_SDP= PIXIT (table 7.1.1-1) or m= Image <Port number> Udptl or Tcptl TYPE_SDP= PIXIT (table 7.1.1-1) a=TYPE_SDP= PIXIT (table 1) b=TYPE_SDP= PIXIT (table 1) Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Establish a communication from network A to Network B Check: Is the preferred codec set to TYPE_SDP? Check: If present: is the a line set to TYPE_SDP? Check: If present: is the b line set to TYPE_SDP? Check: Is the codec list consistent with the attribute(s) (bandwidth) regarding the media description? Repeat this test in reverse direction. Repeat this test with all chosen end devices. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 24 Test case number SS_bcall_018 Test case group BCALL/successful Reference [4] and [5] SELECTION EXPRESSION Test purpose The SDP answer is sent in the 200 OK. Ensure that the call establishment performed correctly. The initial INVITE contains a SDP with the offer 1 according table 7.1.1-1. Ensure that answer related to the SDP offer is contained in the 200 OK INVITE message. Ensure that in the confirmed state the voice transfer on the media and B- channels is performed correctly. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (SDP1) 180 Ringing 200 OK INVITE (SDP2) ACK Apply post test routine Comments Establish a communication from network A to Network B Check: Is the SDP answer contained in the 200 OK INVITE. Repeat this test in reverse direction. Repeat this test with all chosen end devices. Test case number SS_bcall_018 Test case group BCALL/successful Reference [4] and [5] SELECTION EXPRESSION Test purpose First response 200 OK INVITE. Ensure that call establishment and the correctly if the called user answers with a 200 OK message. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 200 OK INVITE ACK Apply post test routine Comments Establish a communication from network A to Network B Check: Is it possible to confirm a session without early dialogue? Repeat this test in reverse direction. Repeat this test with all chosen end devices. Table 7.1.1-1 TYPE_SDP m= line b= line a= line VA <media> <transport> <fmt-list> <modifier>:<bandwidth- value> (see note) rtpmap:<dynamic-PT> <encoding name>/<clock rate>[/encoding parameters> VA_01 Audio RTP/AVP 0 N/A or up to 64 kbit/s N/A or rtpmap 0 PCMU/8000 VA_02 Audio RTP/AVP Dynamic PT N/A or up to 64 kbit/s rtpmap:<dynamic-PT> PCMU/8000 VA_03 Audio RTP/AVP 8 N/A or up to 64 kbit/s N/A or rtpmap 8 PCMA/8000 VA_04 Audio RTP/AVP Dynamic PT N/A or up to 64 kbit/s rtpmap:<dynamic-PT> PCMA/8000 VA_05 audio RTP/AVP Dynamic PT N/A or up to 64 kbit/s rtpmap:<dynamic-PT> CLEARMODE NOTE: <bandwidth value> for <modifier> of AS is evaluated to be B kbit/s. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 25 Test case number SS_bcall_020 Test case group BCALL/successful Reference [4] and [5] SELECTION EXPRESSION [Network A] SE 43 AND [Network B] SE 43 Test purpose Fax transmission using the G.711 codec. Ensure that a Fax transmission is possible from Network A to Network B and the relevant codec is the G.711 codec. Ensure in the active call state the property of Fax transmission. Configuration SIP Parameter INVITE: SDP m=audio <Port> RTP/AVP 8/0 180/200 OK INVITE: SDP m=audio <Port> RTP/AVP 8 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (SDP1) 180 Ringing 200 OK INVITE (SDP2) ACK Apply post test routine Comments Establish a communication from network A to Network B Check: Is the SDP answer contained in the 200 OK INVITE. Check: If Fax transmission is successful? Repeat this test in reverse direction. Test case number SS_bcall_021 Test case group BCALL/successful Reference [5] and [22] SELECTION EXPRESSION [Network A] SE 44 AND [Network A] SE 44 Test purpose Fax transmission using the V.152 codec. Ensure that a Fax transmission is possible from Network A to Network B and the relevant codec is the V.152 codec. Ensure in the active call state the property of Fax transmission. Configuration SIP Parameter INVITE: SDP m=audio <Port> RTP/AVP 8 <dynamic-PT> a=rtpmap <dynamic-PT> PCMA/8000 a=gpmd; vbd=yes 180/200 OK INVITE: SDP m=audio <Port> RTP/AVP <dynamic-PT> a=rtpmap <dynamic-PT> PCMA/8000 a=gpmd; vbd=yes Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (SDP1) 180 Ringing 200 OK INVITE (SDP2) ACK Apply post test routine Comments Establish a communication from network A to Network B Check: Contains the SDP offer in the initial INVITE a voice band data codec. Check: Contains the SDP answer in the 180 or 200 OK INVITE a voice band data codec. Check: If Fax transmission is successful? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 26 Test case number SS_bcall_022 Test case group BCALL/successful Reference [5] and [23] SELECTION EXPRESSION [Network A] SE 45 AND [Network B] SE 45 Test purpose Fax transmission using the T.38 in an audio m-line codec. Ensure that a Fax transmission is possible from Network A to Network B and the relevant codec is the T.38 in an 'audio' m-line codec. Ensure in the active call state the property of Fax transmission. Configuration SIP Parameter INVITE: SDP m=audio <Port> RTP/AVP 8 OR <dynamic-PT> a=rtpmap 8 OR <dynamic-PT> PCMA/8000 m=image <Port> udptl t38 180/200 OK INVITE: SDP m=image <Port> udptl t38 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (SDP1) 180 Ringing 200 OK INVITE (SDP2) ACK Apply post test routine Comments Establish a communication from network A to Network B Check: Contains the SDP offer in the initial INVITE a T.38 codec in an 'audio' line. Check: Contains the SDP answer in the 180 or 200 OK INVITE a T.38 codec in an 'audio' line. Check: If Fax transmission is successful? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 27 Test case number SS_bcall_023 Test case group BCALL/successful Reference 4.9, N/ [2] SELECTION EXPRESSION [Network A] SE 47 AND [Network A] SE 4 AND [Network B] SE 4 Test purpose Overlap sending, the Multiple INVITE method is used. Ensure that call establishment using overlap sending is performed correctly. Ensure that in the confirmed state the voice transfer on the media and B-channels is performed correctly. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(CSq 1) INVITE(CSq 2) 484 Address Incomplete(CSq 1) ACK INVITE(CSq 3) 484 Address Incomplete(CSq 2) ACK ……. INVITE(CSq 4) 484 Address Incomplete(CSq 3) ACK 180 Ringing(CSq 4) Apply post test routine Comments Establish a communication from ISDN to SIP using the overlap operation in ISDN Check: All INVITE requests contain the same Call ID and From header values. SIP answers with 180 Ringing. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 28 Test case number SS_bcall_024 Test case group BCALL/successful Reference 4.9, N/ [2] SELECTION EXPRESSION [Network A] SE 47 AND [Network A] SE 4 AND [Network B] SE 5 Test purpose Overlap sending, the in-Dialogue method is used Ensure that call establishment using overlap sending is performed correctly. Ensure that in the confirmed state the voice transfer on the media and B-channels is performed correctly. Configuration SIP Parameter INVITE 2: Supported: 100rel 183: Require: 100rel INFO: Content-Type: application/x-session-info SubsequentDigit: <additional digits> Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(CSq 1) 1 484 Address Incomplete(CSq 1) ACK INVITE(CSq 2) 2 183 Session Progress(CSq 2) PRACK 200 OK PRACK INFO 200 OK INFO …….. INFO 200 OK INFO 180 Ringing(CSq 2) Apply post test routine Comments Establish a communication from ISDN to SIP using the overlap operation in ISDN Check: All INVITE requests contains the same Call ID and From header values. Check: The 183 session Progress that establishes an early dialogue contains a Require header set to 100rel. Check: All INFO requests contain the Content-Type header set to 'application/x- session-info'. Check: All INFO requests contains the 'SubsequentDigit:' MIME body containing the additional digits. The UE B answers with 180 Ringing response after the INVITE was received. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 29 Test case number SS_bcall_025 Test case group BCALL/successful Reference 5.1.1.1.2/ [25] SELECTION EXPRESSION [Network A] (SE 46 OR SE 47) AND [Network A] SE 6 Test purpose PSTN XML BearerCapability element in the INVITE. User A is located in network A and an ISDN end device is used. Ensure that the INVITE request contains a PSTN XML MIME body and a BearerCapability element as indicated in table 7.1.1-2 is present. Configuration User A is an ISDN access either in the PSTN or the SIP - ISDN interworking according [10] applies SIP Parameter INVITE: Content-Type: application/vnd.etsi.pstn+xml Content-Disposition: signal;handling=optional <?xml version="1.0" encoding="utf-8"?> PSTN BearerCapability BCoctet3 CodingStandard>00< InformationTransferCabability>ITC_value< < BCoctet4 TransferMode>00< InformationTransferRate>10000< BCoctet5 Layer1Identification>01< UserInfoLayer1Protocol>00011< Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Check: Is a PSTN XML MIME body contained in the INVITE request? Check: If the BearerCapability element is present? Check: If the InformationTransferCabability element is set as indicated in table 7.1.1-1? Check: Is the InformationTransferCabability element value consistent with the codec list in the SDP? Check: Is the InformationTransferCabability element value consistent with the bandwidth information in the SDP? Repeat this test in reverse direction. Table 7.1.1-2: PSTN XML BearerCapability ITC_value BC Information transfer capability XML InformationTransferCabability ITC_VA_1 Speech '00000' ITC_VA_2 3,1 kHz audio '10000' ITC_VA_3 unrestricted digital information '01000' ETSI ETSI TS 101 585 V1.2.1 (2014-04) 30 Test case number SS_bcall_026 Test case group BCALL/successful Reference 5.1.1.1.2/ [25] SELECTION EXPRESSION [Network A] (SE 46 OR SE 47) AND [Network A] SE 6 Test purpose PSTN XML HighLayerCapability element in the INVITE. User A is located in network A and an ISDN end device is used. Ensure that the INVITE request contains a PSTN XML MIME body and a HighLayerCapability element is present. Configuration User A is an ISDN access either in the PSTN or the SIP - ISDN interworking according [10] applies SIP Parameter INVITE: Content-Type: application/vnd.etsi.pstn+xml Content-Disposition: signal;handling=optional <?xml version="1.0" encoding="utf-8"?> PSTN HighLayerCompatibility HLOctet3 CodingStandard>00< Interpretation>100< PresentationMethod>01< HLOctet4 HighLayerCharacteristics>[any value]< Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Check: Is a PSTN XML MIME body contained in the INVITE request? Check: If the HighLayerCapability element is present? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 31 Test case number SS_bcall_027 Test case group BCALL/successful Reference 5.1.1.1.2/ [25] SELECTION EXPRESSION [Network A] (SE 46 OR SE 47) AND [Network A] SE 6 Test purpose PSTN XML ProgressIndicator element in the INVITE. User A is located in network A and an ISDN end device is used. Ensure that the INVITE request contains a PSTN XML MIME body and at least one ProgressIndicator element is present. Configuration User A is an ISDN access either in the PSTN or the SIP - ISDN interworking according [10] applies SIP Parameter INVITE: Content-Type: application/vnd.etsi.pstn+xml Content-Disposition: signal;handling=optional <?xml version="1.0" encoding="utf-8"?> PSTN ProgressIndicator ProgressOctet3 CodingStandard>00< Location>yyyy< ProgressOctet4 ProgressDescription>0000110< ProgressIndicator ProgressOctet3 CodingStandard>00< Location>0000< ProgressOctet4 ProgressDescription>[any value]< Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Check: Is a PSTN XML MIME body contained in the INVITE request? Check: Is a ProgressIndicator element present and the [10] ProgressDescription element is set to '0000110'? Check: Is optional a second ProgressIndicator element present and the ProgressDescription element is set to any value not #2 and not #8? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 32 Test case number SS_bcall_028 Test case group BCALL/successful Reference 5.1.2.2/ [25] SELECTION EXPRESSION [Network B] (SE 46 OR SE 47) AND [Network B] SE 6 Test purpose PSTN XML ProgressIndicator element in the 180. User B is located in network B and an ISDN end device is used. Ensure that the 180 Ringing response contains a PSTN XML MIME body and at least one ProgressIndicator element is present. Configuration User B is an ISDN access either in the PSTN or the SIP - ISDN interworking according [10] applies SIP Parameter 180: Content-Type: application/vnd.etsi.pstn+xml Content-Disposition: signal;handling=optional <?xml version="1.0" encoding="utf-8"?> PSTN ProgressIndicator ProgressOctet3 CodingStandard>00< Location>yyyy< ProgressOctet4 ProgressDescription>0000111< ProgressIndicator ProgressOctet3 CodingStandard>00< Location>0000< ProgressOctet4 ProgressDescription>[any value]< Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Apply post test routine Comments Check: Is a PSTN XML MIME body contained in the 180 Ringing response? Check: Is a ProgressIndicator element present and the ProgressDescription element is set to '0000110'? Check: Is optional a second ProgressIndicator element present and the ProgressDescription element is set to any value not #2 and not #8? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 33 Test case number SS_bcall_029 Test case group BCALL/successful Reference 5.1.2.3/ [25] SELECTION EXPRESSION [Network B] (SE 46 OR SE 47) AND [Network B] SE 6 Test purpose PSTN XML ProgressIndicator element in the 200. User B is located in network B and an ISDN end device is used. Ensure that the 200 OK INVITE response contains a PSTN XML MIME body and at least one ProgressIndicator element is present. Configuration User B is an ISDN access either in the PSTN or the SIP - ISDN interworking according [10] applies SIP Parameter 200: Content-Type: application/vnd.etsi.pstn+xml Content-Disposition: signal;handling=optional <?xml version="1.0" encoding="utf-8"?> PSTN ProgressIndicator ProgressOctet3 CodingStandard>00< Location>yyyy< ProgressOctet4 ProgressDescription>0000111< Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK Apply post test routine Comments Check: Is a PSTN XML MIME body contained in the 200 OK INVITE response? Check: Is a ProgressIndicator element present and the ProgressDescription element is set to '0000110'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 34 Test case number SS_bcall_030 Test case group BCALL/successful Reference 5.1.1.1.2/ [25] SELECTION EXPRESSION [Network A] (SE 46 OR SE 47) AND [Network A] SE 6 Test purpose PSTN XML BearerCapability Fallback connection type element in the INVITE. User A is located in network A and an ISDN end device is used. Ensure that the INVITE request contains a PSTN XML MIME body and one BearerCapability element is present the InformationTransferCabability element is set to '00000' and one InformationTransferCabability element is set to '10001'. Configuration User A is an ISDN access either in the PSTN or the SIP - ISDN interworking according [10] applies SIP Parameter INVITE: Content-Type: application/vnd.etsi.pstn+xml Content-Disposition: signal;handling=optional <?xml version="1.0" encoding="utf-8"?> PSTN BearerCapability BCoctet3 CodingStandard>00< InformationTransferCabability>00000< BearerCapability BCoctet3 CodingStandard>00< InformationTransferCabability>10001< Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Check: Is a PSTN XML MIME body contained in the INVITE request? Check: If the first BearerCapability InformationTransferCabability element is set as indicated to '00000'? Check: If the second BearerCapability InformationTransferCabability element is set as indicated to '10001'? Check: Is the InformationTransferCabability element value consistent with the codec list in the SDP? Check: Is the InformationTransferCabability element value consistent with the bandwidth information in the SDP? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 35 Test case number SS_bcall_031 Test case group BCALL/successful Reference 5.1.2.3/ [25] SELECTION EXPRESSION [Network B] (SE 46 OR SE 47) AND [Network B] SE 6 Test purpose Fall back does not occur. User B is located in network B and an ISDN end device is used. The Fallback connection type was requested in the initial INVITE request. Ensure that the 200 OK INVITE response contains a PSTN XML MIME body and a BearerCapability element is present the InformationTransferCabability element set to '10001'. Configuration User B is an ISDN access either in the PSTN or the SIP - ISDN interworking according [10] applies SIP Parameter 200: Content-Type: application/vnd.etsi.pstn+xml Content-Disposition: signal;handling=optional <?xml version="1.0" encoding="utf-8"?> PSTN BearerCapability BCoctet3 CodingStandard>00< InformationTransferCabability>10001< Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK Apply post test routine Comments Check: Is a PSTN XML MIME body contained in the 200 OK INVITE response? Check: Is a BearerCapability element present, the InformationTransferCabability element set to '10001'? Check: Is the InformationTransferCabability element value consistent with the codec list in the SDP? Check: Is the InformationTransferCabability element value consistent with the bandwidth information in the SDP? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 36 Test case number SS_bcall_032 Test case group BCALL/successful Reference 5.1.2.3/ [25] SELECTION EXPRESSION [Network B] (SE 46 OR SE 47) AND [Network B] SE 6 Test purpose Fall back occurs. User B is located in network B and an ISDN end device is used. The Fallback connection type was requested in the initial INVITE request. Ensure that the 200 OK INVITE response contains a PSTN XML MIME body and a BearerCapability element is present the InformationTransferCabability element set to '00000'. A PSTN XML MIME ProgressIndicator body is present, the ProgressDescription is set to '0000101'. Configuration User B is an ISDN access either in the PSTN or the SIP - ISDN interworking according [10] applies SIP Parameter 200: Content-Type: application/vnd.etsi.pstn+xml Content-Disposition: signal;handling=optional <?xml version="1.0" encoding="utf-8"?> PSTN BearerCapability BCoctet3 CodingStandard>00< InformationTransferCabability>00000< ProgressIndicator ProgressOctet4 ProgressDescription>0000101< Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK Apply post test routine Comments Check: Is a PSTN XML MIME body contained in the 200 OK INVITE response? Check: Is a BearerCapability element present, the InformationTransferCabability element set to '00000'? Check: Is a ProgressIndicator element present, the ProgressDescription is set to '0000101'? Check: Is the InformationTransferCabability element value consistent with the codec list in the SDP? Check: Is the InformationTransferCabability element value consistent with the bandwidth information in the SDP? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 37 Test case number SS_bcall_032A Test case group BCALL/successful Reference 5.1.2.3/ [26] SELECTION EXPRESSION Test purpose Telephony events transmission Ensure that the ability of transmission of Telephony events can be performed by the originating user. The Telephony transmission can be done by: • either indicated in the SDP offer in the RTP stream • or SIP INFO/NOTIFY Method for DTMF tone generation Configuration SIP Parameter INVITE: CASE A m=audio <Port> RTP/AVP <dynamic-PT> a=rtpmap <dynamic-PT> telephone-event/8000 a=rtpmap <dynamic-PT> 0-15 CASE B Accept: application/dtmf CASE C Accept: application/dtmf NOTIFY CASE B Content-Type: application/dtmf 'x' CASE C Content-Type: application/dtmf-relay Signal=x Duration=y Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK CASE A RTP DTMF events CASE B INFO 200 OK INFO CASE C INFO 200 OK INFO Apply post test routine Comments Establish a communication from network A to Network B Check: Case A is the dynamic payload type 'telephone-event' present in the SDP offer? Check: Case A is the dynamic payload type 'telephone-event' covered in the RTP stream if the Telephone event occurs? Check: Case B is the Content-Type header field in the INFO request conveying the DTMF signal set to 'application/dtmf'? Check: Case B contains the MIME body of the INFO request covering the TMF signal the events regarding the used content type? Check: Case C is the Content-Type header field in the INFO request conveying the DTMF signal set to 'application/dtmf-relay'? Check: Case C contains the MIME body of the INFO request covering the TMF signal the events and duration regarding the used content type? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 38 Test case number SS_bcall_033 Test case group BCALL/successful Reference 7.1/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 Test purpose SIP-I support, Basic call, IAM present in the INVITE request. Ensure that when a call initiated in the PSTN or the PLMN and the ISUP - SIP-I interworking is applicable in the originating network, an ISUP IAM is encapsulated in the initial INVITE request. Ensure that all the mandatory parameters in the IAM are present and the values are valid and the Transmission medium requirement parameter is consistent with the SDP. Configuration SIP Parameter INVITE: Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Nature of connection indicators Forward call indicators Calling party's category Transmission medium requirement Called party number Calling party number (optional) Optional forward call indicators (optional) Hop counter (optional) User service information (optional) Access transport (optional) --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(IAM) 100 Trying Apply post test routine Comments Establish a communication from network A to Network B Check: Is an ISUP IAM encapsulated in the INVITE request? Check: Are all the mandatory ISUP parameters present in the IAM and are the values valid? Check: Are the values of the optional parameters in the encapsulated IAM valid? Check: Is the 'm' line with corresponding attributes in the SDP consistent with the Transmission medium requirement parameter? Check: Is the Transmission medium requirement value consistent with the bandwidth information in the SDP? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 39 Test case number SS_bcall_034 Test case group BCALL/successful Reference 7.2.1/ [24] SELECTION EXPRESSION [Network A] SE 4 AND SE 17 AND SE 47 Test purpose SIP-I support, Basic call, overlap signalling. Ensure that when overlap signalling applies in the ISUP -SIP-I interworking in the originating network, several INVITE requests with the same Cal-ID and From tag are sent from Network A to Network B. Ensure that the original IAM is encapsulated in any INVITE request. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(1) 484 Address Incomplete(1) ACK INVITE(2) 484 Address Incomplete(2) ACK INVITE(3) 484 Address Incomplete(3) ACK . . INVITE(4) 180 Ringing(4) Apply post test routine Comments Establish a communication from network A to Network B using the overlap procedure in Network A Check: Are the INVITE requests sent with the same From tag and the Call-ID? Check: After the 180 applies, are all previous INVITE transactions are terminated with a 484 final response? Check: Is the encapsulated IAM present in the initial INVITE request also encapsulated in any following INVITE request required for the call setup? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 40 Test case number SS_bcall_035 Test case group BCALL/successful Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 Test purpose SIP-I support, Basic call, ACM present in the 180 response. Ensure that on receipt of a 180 Ringing provisional response and an SIP-I - ISUP interworking is applicable in the terminating network the Backward call indicators parameter in the encapsulated ACM is present and the values are valid. Ensure that the values of the optional parameters in the encapsulated ACM are valid. Configuration SIP Parameter 180: Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM Backward call indicators --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 100 Trying 180 Ringing(ACM) Apply post test routine Comments Establish a communication from network A to Network B Check: Is an ISUP ACM message encapsulated in the 180 Ringing provisional response? Check: Is the mandatory Backward call indicators parameter present in the encapsulated ISUP ACM and are the values valid? Check: Are the values of optional parameters in the encapsulated ISUP ACM valid? Check: If an SDP answer is present in the 180, are the codec and the bandwidth information in the 'a' attributes consistent with Transmission medium requirement in the encapsulated IAM of the INVITE request? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 41 Test case number SS_bcall_036 Test case group BCALL/successful Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 Test purpose SIP-I support. Basic call, early ACM present in the 183 response. Ensure that on receipt of a 183 Session Progress provisional response and an SIP-I - ISUP interworking is applicable in the terminating network the Backward call indicators parameter in the encapsulated ACM is present and the value of the Called party's status indicator is set to 'no indication'. Ensure that the values of the optional parameters in the encapsulated ACM are valid. Configuration Select a proper destination that sends an early ACM in the PSTN/PLMN e.g. announcement SIP Parameter 183: Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM Backward call indicators Called party's status indicator= no indication --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 100 Trying 183 Session Progress(ACM) Apply post test routine Comments Establish a communication from network A to Network B Check: Is an ISUP ACM message encapsulated in the 183 Session Progress provisional response? Check: Is the mandatory Backward call indicators parameter present in the encapsulated ISUP ACM and are the values valid? Check: Is the Called party's status indicator in the encapsulated ISUP ACM set to 'no indication'? Check: Are the values of optional parameters in the encapsulated ISUP ACM valid? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 42 Test case number SS_bcall_037 Test case group BCALL/successful Reference 6.6/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 Test purpose SIP-I support. Basic call, CPG present in a 180 response. Ensure that on receipt of a 180 Ringing provisional response and an SIP-I - ISUP interworking is applicable in the terminating network the Event indicator in the encapsulated CPG is present and set to 'ALERTING'. Ensure that the values of the optional parameters in the encapsulated CPG are valid. Configuration Select a proper destination that sends at first an early ACM and after then a CPG 'ALERTING' in the PSTN/PLMN (e.g. PBX). SIP Parameter 180: Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Event indicator = ALERTING --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 100 Trying 183 Session Progress(ACM) 180 Ringing(CPG) Apply post test routine Comments Establish a communication from network A to Network B Check: Is an ISUP CPG message encapsulated in the 180 Ringing provisional response? Check: Is the mandatory Event indicator present in the encapsulated ISUP CPG set to 'ALERTING'? Check: Are the values of optional parameters in the encapsulated ISUP CPG valid? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 43 Test case number SS_bcall_038 Test case group BCALL/successful Reference 6.7/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 Test purpose SIP-I support. Basic call, ANM present in a 200 OK INVITE response. Ensure that on receipt of a 200 OK INVITE final response and an SIP-I - ISUP interworking is applicable in the terminating network the ISUP ANM is encapsulated in the 200 OK. Ensure that the values of the optional parameters in the encapsulated ANM are valid. Configuration SIP Parameter 180: Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 100 Trying 180 Ringing(ACM) 200 OK INVITE(ANM) ACK Apply post test routine Comments Establish a confirmed communication from network A to Network B Check: Is an ISUP ANM encapsulated in the 200 OK INVITE? Check: Are the values of optional parameters in the encapsulated ISUP ANM valid? Check: Ensure the property of speech. Check: Are the codec and the bandwidth information in the 'a' attributes consistent with Transmission medium requirement in the encapsulated IAM of the INVITE request? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 44 Test case number SS_bcall_039 Test case group BCALL/successful Reference 5.4.3.4, 6.11.2/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 Test purpose SIP-I support. Basic call, REL present in a BYE request sent from the originating network. Ensure that a ISUP REL message is encapsulated in a BYE request sent in the release procedure initiated from the originating user when ISUP - SIP-I interworking is applicable in the originating network. Ensure the validity of the cause indicator in the encapsulated REL. Ensure that the ISUP RLC is encapsulated in the 200 OK BYE. Configuration SIP Parameter BYE: Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL Cause value: --[any boundary name]-- 200 OK BYE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required RLC --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 100 Trying 180 Ringing 200 OK INVITE ACK Communication BYE(REL) 200 OK BYE(RLC) Comments Establish a confirmed communication from network A to Network B The originating user terminates the communication Check: Is the ISUP REL encapsulated in the BYE request? Check: Are the cause indicators in the encapsulated ISUP REL valid? Check: If a Reason header is present in the BYE request, is the 'cause' value of Reason header equal to the 'Cause value' in the encapsulated REL? Check: Is the ISUP RLC encapsulated in the 200 OK BYE? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 45 Test case number SS_bcall_040 Test case group BCALL/successful Reference 5.4.3.4, 6.11.2/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 Test purpose SIP-I support. Basic call, REL present in a BYE request sent from the terminating network. Ensure that a ISUP REL message is encapsulated in a BYE request sent in the release procedure initiated from the terminating user when SIP-I - ISUP interworking is applicable in the terminating network. Ensure the validity of the cause indicator in the encapsulated REL. Ensure that the ISUP RLC is encapsulated in the 200 OK BYE. Configuration SIP Parameter BYE: Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL Cause value: --[any boundary name]-- 200 OK BYE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required RLC --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 100 Trying 180 Ringing 200 OK INVITE ACK Communication BYE(REL) 200 OK BYE(RLC) Comments Establish a confirmed communication from network A to Network B The terminating user terminates the communication Check: Is the ISUP REL encapsulated in the BYE request? Check: Are the cause indicators in the encapsulated ISUP REL valid? Check: If a Reason header is present in the BYE request, is the 'cause' value of Reason header equal to the 'Cause value' in the encapsulated REL? Check: Is the ISUP RLC encapsulated in the 200 OK BYE? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 46
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7.1.2 Codec negotiation
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Test case number SS_codec_001 Test case group BCALL/Codec_Negotiation Reference [3], [4] and [5] SELECTION EXPRESSION Test purpose Session update requested by the calling user. During the session, the calling user decides to change the characteristics of the media session. This is accomplished by sending a re-INVITE or UPDATE containing a new media description. This re-INVITE or UPDATE references the existing dialog so that the other party knows that it is to modify an existing session instead of establishing a new session. The other party sends a 200 (OK) to accept the change. The requestor responds to the 200 (OK) with an ACK. In case when the parameter in the SDP rtpmap:<dynamic-PT> is used the codecs in table 7.1.2-1 applies. Configuration SIP Parameter SDP1: codec x chosen from table 7.1.2-1 SDP3: codec y chosen from table 7.1.2-1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists (SDP 1) CASE A INVITE(SDP3) 200 OK INVITE(SDP4) ACK CASE B UPDATE(SDP3) [5] 200 OK UPDATE(SDP4) Apply post test routine Comments Establish a communication from network A to Network B using SDP1 chosen from the table 7.1.2-1 Check: The calling user changes the media description using INVITE request containing SDP 3 codec chosen from table 7.1.2-1 different to SDP1. Check: Is the codec list consistent with the attribute(s) (bandwidth) regarding the media description? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 47 Test case number SS_codec_002 Test case group BCALL/Codec_Negotiation Reference [3], [4] and [5] SELECTION EXPRESSION Test purpose Session update requested by the called user. During the session, the called user decides to change the characteristics of the media session. This is accomplished by sending a re-INVITE containing a new media description. This re- INVITE references the existing dialog so that the other party knows that it is to modify an existing session instead of establishing a new session. The other party sends a 200 (OK) to accept the change. The requestor responds to the 200 (OK) with an ACK. In case when the parameter in the SDP rtpmap:<dynamic-PT> is used the codecs in table 7.1.2-1 applies. Configuration SIP Parameter SDP1: codec x chosen from table 7.1.2-1 SDP2: codec y chosen from table 7.1.2-1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists (SDP 1) CASE A INVITE(SDP3) 200 OK INVITE(SDP4) ACK CASE B UPDATE(SDP3) 200 OK UPDATE(SDP4) Apply post test routine Comments Establish a connection from SIP UE 1 to SIP UE 2 using SDP1 chosen from the table 7.1.2-1 Check: The called user changes the media description using INVITE request containing SDP 2 codec chosen from table 7.1.2-1 different to SDP1. Check: Is the codec list consistent with the attribute(s) (bandwidth) regarding the media description? Repeat this test in reverse direction. Test case number SS_codec_003 Test case group BCALL/Codec_Negotiation Reference [3], [4] and [5] SELECTION EXPRESSION Test purpose The SDP answer is contained in a 200 OK final response. Ensure that the call establishment performed correctly. • The initial INVITE contains a SDP with the offer 1. • Ensure that answer related to the SDP offer is contained in the 200 OK INVITE message. Ensure that in the confirmed call state the voice transfer on the media channels is performed correctly. Configuration SIP Parameter INVITE: SDP offer 200: SDP answer Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(SDP1) 180 Ringing 200 OK INVITE(SDP2) ACK Apply post test routine Comments Establish a communication from network A to Network B Check: Is the SDP offer contained in the initial INVITE request? Check: Is the SDP answer contained in the 200 OK INVITE final response? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 48 Table 7.1.2-1 VARIABLE PT Encoding Media type Clock rate Channels Supported in network A Supported in network B VA_01 0 PCMU A 8 000 1 VA_02 3 GSM A 8 000 1 VA_03 4 G723 A 8 000 1 VA_04 5 DVI4 A 8 000 1 VA_05 6 DVI4 A 16 000 1 VA_06 7 LPC A 8 000 1 VA_07 8 PCMA A 8 000 1 VA_08 9 G722 A 8 000 1 VA_09 10 L16 A 44 100 2 VA_10 11 L16 A 44 100 1 VA_13 12 QCELP A 8 000 1 VA_12 13 CN A 8 000 1 VA_13 14 MPA A 90 000 VA_14 15 G728 A 18 000 1 VA_15 16 DVI4 A 11 025 1 VA_16 17 DVI4 A 22 050 1 VA_17 18 G729 A 8 000 1 VA_18 Dyn G726-40 A 8 000 1 VA_19 Dyn G726-32 A 8 000 1 VA_20 Dyn G726-24 A 8 000 1 VA_21 Dyn G726-16 A 8 000 1 VA_22 Dyn G729D A 8 000 1 VA_23 Dyn G729E A 8 000 1 VA_24 Dyn GSM-EFR A 8 000 1 VA_25 25 CelB V 90 000 VA_26 26 JPEG V 90 000 VA_27 28 Nv V 90 000 VA_28 31 H261 V 90 000 VA_29 32 MPV V 90 000 VA_30 33 MP2T V 90 000 VA_31 34 H263 V 90 000 VA_32 Dyn H263-1998 V 90 000 VA_33 Dyn AMR A 8 000 1 VA_34 Dyn AMR-WB A 16 000 1 VA_35 Dyn telephone- event A 8 000 1 ETSI ETSI TS 101 585 V1.2.1 (2014-04) 49
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7.1.3 Resource Reservation
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Test case number SS_resource_001 Test case group BCALL/Resource_Reservation Reference [3], [4], [5] and [6] SELECTION EXPRESSION ([Network A] SE 50 AND [Network B] SE 50) AND SE 7 Test purpose Resource reservation successful, segmented status. Ensure that the network is able to reserve resources for quality of service when requested from the initiating user. • In the INVIT the UE requests to establish QoS preconditions for all the media streams. • In the 183 Session Progress the UAS supports the QoS preconditions and requests that UAC sends a confirmation when the QoS preconditions are met. • The UPDATE includes in the SDP the information about the successful QoS bidirectional mode, due to the successful bidirectional PDP context established. • 200 OK UPDATE the SDP contains an indication that the UE successfully reserved the QoS in the send and receive directions. Configuration SIP Parameter INVITE: Supported: 100rel precondition SDP1: m=audio 3456 RTP/AVP 8 a=curr:qos local none a=curr:qos remote none a=des:qos mandatory local sendrecv a=des:qos none remote sendrecv 183 Session Progress: Supported: 100rel precondition SDP2: m=audio 6544 RTP/AVP 8 a=curr:qos local none a=curr:qos remote none a=des:qos mandatory local sendrecv a=des:qos mandatory remote sendrecv UPDATE SDP3: m=audio 3456 RTP/AVP 8 a=curr:qos local sendrecv a=curr:qos remote none a=des:qos mandatory local sendrecv a=des:qos mandatory remote sendrecv 200 OK UPDATE SDP4: a=curr:qos local sendrecv a=curr:qos remote sendrecv a=des:qos mandatory local sendrecv a=des:qos mandatory remote sendrecv Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(SDP1) 183 Session Progress(SDP2) PRACK 200 OK PRACK Resource reservation UPDATE(SDP3) 200 OK UPDATE(SDP4) Apply post test routine Comments Establish a communication from network A to Network B Check: Is the quality of service for the current state local and remote set to 'none' indicated in the SDP in the INVITE? Check: Is the quality of service for the desired state local and remote set to 'mandatory' and 'sendrecv' in the 183? Check: Is the quality of service for the current state local set to 'sendrecv' indicated in the SDP in the UPDATE? Check: Is the quality of service for the current state local and remote set to 'sendrecv' indicated in the SDP in the 200 OK UPDATE? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 50
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7.1.4 Test purposes for SIP-SIP, Basic call, Unsuccessful
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Test case number SS_unsucc_001 Test case group BCALL/unsuccessful Reference [4] SELECTION EXPRESSION Test purpose Called number is not allocated in the assumed network. Ensure that, when calling to unallocated number, the network initiate call clearing to the calling user with a 404 Not Found message. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 404 Not Found ACK Comments Establish a communication from network A to Network B, called user number is not allocated in Network B Check: Is a 404 Not Found sent from Network B to Network A? Repeat this test in reverse direction. Repeat this test with all chosen end devices. Test case number SS_unsucc_002 Test case group BCALL/unsuccessful Reference [4] SELECTION EXPRESSION Test purpose The network B is unable to process the request. Ensure that the call will be released if the Service unavailable. The network initiates call clearing to the calling user with a 503 Service unavailable message. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 503 Service unavailable ACK Comments Establish a communication from network A to Network B, Network B is unable to process the request. Check: Is a 503 Service unavailable sent from Network B to Network A? Repeat this test in reverse direction. Repeat this test with all chosen end devices. Test case number SS_unsucc_003 Test case group BCALL/unsuccessful Reference [4] SELECTION EXPRESSION Test purpose The called user is network determined busy. Ensure that, when the called user is busy, the network initiates call clearing to the calling user with a 486 Busy Here message. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 486 Busy Here ACK Comments Establish a communication from network A to Network B, user B is network determined user busy. Check: Is a 486 Busy Here sent from Network B to Network A? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 51 Test case number SS_unsucc_004 Test case group BCALL/unsuccessful Reference [4] SELECTION EXPRESSION Test purpose The called user is user determined busy. Ensure that, when the called user is busy, the user initiates call clearing to the calling user with a 486 Busy Here message. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 486 Busy Here ACK Comments Establish a communication from network A to Network B, user B is user determined user busy. Check: Is a 486 Busy Here sent from Network B to Network A Repeat this test in reverse direction. Test case number SS_unsucc_005 Test case group BCALL/unsuccessful Reference [4] SELECTION EXPRESSION Test purpose The called user is not available on the called number. Ensure that when the number is changed, the network initiate call clearing to the calling user with a 410 Gone message. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 410 Gone ACK Comments Establish a communication from network A to Network B, user B is not allocated in Network B. Check: Is a 410 Gone sent from Network B to Network A? Repeat this test in reverse direction. Test case number SS_unsucc_006 Test case group BCALL/unsuccessful Reference [4] SELECTION EXPRESSION Test purpose The number of the called user is incomplete. Ensure that the call will be released when the called number is incomplete. The network initiates call clearing to the calling user with 484 Not Found message. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 484 Address Incomplete ACK Comments Establish a communication from network A to Network B, the called number is incomplete. Check: Is a 484 Address Incomplete sent from Network B to Network A? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 52 Test case number SS_unsucc_007 Test case group BCALL/unsuccessful Reference [3], [4] and [5] SELECTION EXPRESSION Test purpose Session update requested by the calling user is unsuccessful, existing session remains unchanged. During the session, the calling user decides to change the characteristics of the media session. This is accomplished by sending a re-INVITE containing a new media description. This re-INVITE references the existing dialog so that the other party knows that it is to modify an existing session instead of establishing a new session. Ensure that if the other party does not accept the change, he sends an error response such as 488 Not Acceptable Here, which also receives an ACK. The session remains unchanged. Configuration SIP Parameter INVITE: codec not supported in Network B Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK Communication INVITE 488 Not Acceptable Here ACK Apply post test routine Comments Establish a communication from network A to Network B. User A in Network A attempts to change the session by sending a SDP offer to the UE in Network B. Network B does not support the codec sent in the offer. Check: Is a 488 Not Acceptable Here sent from Network B to Network A? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 53 Test case number SS_unsucc_008 Test case group BCALL/unsuccessful Reference [3], [4] and [5] SELECTION EXPRESSION Test purpose Session update requested by the called user is unsuccessful, existing session remains unchanged. During the session, the called user decides to change the characteristics of the media session. This is accomplished by sending a re-INVITE containing a new media description. This re-INVITE references the existing dialog so that the other party knows that it is to modify an existing session instead of establishing a new session. Ensure that if the other party does not accept the change, he sends an error response such as 488 Not Acceptable Here, which also receives an ACK. The session remains unchanged. The 488 Not Acceptable Here may be sent by a simulation equipment. Configuration SIP Parameter INVITE: codec not supported in Network A Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK Communication INVITE 488 Not Acceptable Here ACK Apply post test routine Comments Establish a communication from network A to Network B. User B in Network B attempts to change the session by sending a SDP offer to the UE in Network A Network A does not support the codec sent in the offer. Check: Is a 488 Not Acceptable Here sent from Network B to Network A? Repeat this test in reverse direction. Test case number SS_unsucc_009 Test case group BCALL/unsuccessful Reference [4] SELECTION EXPRESSION Test purpose Call clearing due to no answer from the called user initiated by the calling user. Ensure that when there is no answer from the called user, the calling user initiates call clearing to the called user with CANCEL or BYE Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing CANCEL/BYE 200 OK CANCEL/BYE 487 Request Terminated ACK Comments Check: Is a CANCEL or BYE request is sent from the originating user? Check: Is a 487 Request Terminating send from the terminating user? Check: Are the media streams terminated after the 200 OK CANCEL/BYE was sent? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 54 Test case number SS_unsucc_010 Test case group BCALL/unsuccessful Reference [3], [4] and [5] SELECTION EXPRESSION Test purpose Codec not supported by the called user. The initial INVITE contains a SDP with codes that does not support by the called user. Ensure that, when the called user does not accept the Media session, the called user initiate call clearing to the calling user with 488 Not Acceptable Here, which also receives an ACK. Configuration SIP Parameter INVITE: codec not supported at user (Network B) Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE CASE A 488 Not Acceptable Here ACK CASE B 606 Not Acceptable ACK Comments Establish a call setup from network A to Network B. User B in Network B does not support the codec offered in the SDP received from Network A. Check: Is a 488 Not Acceptable Here sent from Network B to Network A. Repeat this test in reverse direction. Test case number SS_unsucc_011 Test case group BCALL/unsuccessful Reference [4] SELECTION EXPRESSION Test purpose Call clearing due to no answer from the called user initiated by the originating network. Ensure that when there is no answer from the called user, the originating network initiate the call clearing after timeout of SIP timer C and sends a CANCEL or BYE to the called user. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Start timer C Timeout timer C CANCEL/BYE 200 OK CANCEL/BYE 487 Request Terminated ACK Comments Check: Is a CANCEL or BYE request sent by the originating network? Check: Is a 487 Request Terminating send from the terminating user? Check: Are the media streams terminated after the 200 OK CANCEL/BYE was sent? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 55 Test case number SS_unsucc_011A Test case group BCALL/unsuccessful Reference [27] SELECTION EXPRESSION Test purpose Negotiation of session timer. Ensure that the interconnected networks are able to negotiate the session time to refresh the session. If the session refresh duration is to short for one of the involved entities, a 422 Session Interval Too Small unsuccessful final response is sent in backward direction to update the session duration time. A new INVITE is sent and a Min-SE header present proposes a longer session duration. Configuration The session time in Network B is smaller as the session time used in Network A Comment This test case is only applicable if the session refresh time is different in Network A and Network B. This situation is also load dependant. SIP Parameter INVITE 1: Supported: timer Session-Expires: x 422: Min-SE. x + y INVITE 2 Session-Expires: x + y Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 1 422 Session Interval Too Small ACK INVITE 2 180 Ringing Apply post test routine Comments Establish a communication setup from Network A to Network B Check: Is the supported header in the initial INVITE set to 'timer' Check: Is a 422 Session Interval Too Small send from the terminating Network? Check: Is the Session-Expires header in the second initial INVITE request sent from Network A set to the value indicated in the 422 final response? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 56 Test case number SS_unsucc_012 Test case group BCALL/unsuccessful Reference 6.11.2/ [24] SELECTION EXPRESSION [Network B] SE 17 Test purpose SIP-I support. Called number is not allocated in the PSTN/PLMN network. Ensure that, when calling to an unallocated number in the PSTN/PLMN part of network B and ISUP - SIP-I interworking applies in Network B, the network initiate call clearing to the calling user with a 404 Not Found message. A ISUP REL message is encapsulated and the Cause value indicator is set to '1'. Configuration The called user number is not assigned to the PSTN/PLMN part in Network B SIP Parameter 404: Reason: Q.850;cause=1 (optional) Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL Cause value: 1 --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 404 Not Found(REL) ACK Comments Establish a communication from network A to Network B, called user number is not allocated in the PSTN/PLMN part of Network B Check: Is a 404 Not Found sent from Network B to Network A? Check: is a ISUP REL encapsulated and the Cause value indicator is set to '1'? Check: If a Reason header is present, is the cause value equal to the value in the Cause value of the encapsulated ISUP REL? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 57 Test case number SS_unsucc_013 Test case group BCALL/unsuccessful Reference 6.11.2/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 Test purpose SIP-I support. The called user is busy. Ensure that, when the called user in the PSTN/PLMN part of Network B and ISUP - SIP-I interworking applies in Network B is busy, the network initiates call clearing to the calling user with a 486 Busy Here message. A ISUP REL message is encapsulated and the Cause value indicator is set to '17'. Configuration The called user is busy in the PSTN/PLMN part in Network B SIP Parameter 486: Reason: Q.850;cause=17 (optional) Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL Cause value: 17 --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 486 Busy Here(REL) ACK Comments Establish a communication from network A to Network B, user B in the PSTN/PLMN part of Network B is busy. Check: Is a 486 Busy Here sent from Network B to Network A? Check: Is a ISUP REL encapsulated and the Cause value indicator is set to '17'? Check: If a Reason header is present, is the cause value equal to the value in the Cause value of the encapsulated ISUP REL? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 58 Test case number SS_unsucc_014 Test case group BCALL/unsuccessful Reference 6.11.2/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 Test purpose SIP-I support. The called user rejects the call. Ensure that, when the called user in the PSTN/PLMN part of Network B and ISUP - SIP-I interworking applies in Network B rejects the communication setup, the network initiates call clearing to the calling user with a 480 Temporarily Unavailable final response. A ISUP REL message is encapsulated and the Cause value indicator is set to '21'. Configuration SIP Parameter 480: Reason: Q.850;cause=21 (optional) Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL Cause value: 21 --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 480 Temporarily Unavailable (REL) ACK Comments Establish a communication from network A to Network B, user B in the PSTN/PLMN part of network B rejects the communication setup. Check: Is a 480 Temporarily Unavailable sent from Network B to Network A? Check: is a ISUP REL encapsulated and the Cause value indicator is set to '21'? Check: If a Reason header is present, is the cause value equal to the value in the Cause value of the encapsulated ISUP REL? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 59 Test case number SS_unsucc_015 Test case group BCALL/unsuccessful Reference 7.7.1/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 Test purpose SIP-I support. Call clearing due to no answer from the called user initiated by the calling user. Ensure when the early dialogue is not confirmed by the called user, the calling user located in the PSTN/PLMN part of Network A and ISUP - SIP-I interworking applies in Network A initiates call clearing to the called user with CANCEL or BYE. An ISUP REL message is encapsulated in the BYE request and the Cause value indicator is set to '16'. Configuration SIP Parameter 480: Reason: Q.850;cause=16 (optional) Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL Cause value: 16 --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing CASE A CANCEL 200 OK CANCEL 487 Request Terminated ACK CASE B BYE(REL) 200 OK BYE(RLC) 487 Request Terminated ACK Comments Establish a communication from network A to Network B, user B does not confirm the communication. The originating user in the PSTN/PLMN part of Network A terminates the early dialogue. Check: Is a CANCEL or BYE request is sent from the originating network? Check: Is a ISUP REL encapsulated in a BYE request? Check: Is the Cause value of the encapsulated REL set to '16'? Check: If a Reason header is present, is the cause value equal to the value in the Cause value of the encapsulated ISUP REL? Check: Is a 487 Request Terminating send from the terminating user? Check: Are the media streams terminated after the 200 OK CANCEL/BYE was sent? NOTE: A ISUP REL is not encapsulated in a CANCEL request. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 60 Test case number SS_unsucc_016 Test case group BCALL/unsuccessful Reference 7.7.1/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 Test purpose SIP-I support. Call clearing due to no answer from the called user initiated by the originating network. Ensure when the early dialogue is not confirmed by the called user, the originating network initiate the call clearing after timeout of ISUP timer T9 if the calling user is located in the PSTN/PLMN part of Network A and ISUP - SIP-I interworking applies in Network A and the originating network sends a CANCEL or BYE to the called user. An ISUP REL message is encapsulated in the BYE request and the Cause value indicator is set to '19'. Configuration SIP Parameter 480: Reason: Q.850;cause=19 (optional) Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL Cause value: 19 --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Start timer T9 Timeout T9 CASE A CANCEL 200 OK CANCEL 487 Request Terminated ACK CASE B BYE(REL) 200 OK BYE(RLC) 487 Request Terminated ACK Comments Establish a communication from network A to Network B, user B does not answer the communication setup. The ISUP timer T9 in the PSTN/PLMN expires Check: Is a CANCEL or BYE request is sent by the originating network? Check: Is a ISUP REL encapsulated in a BYE request? Check: Is the Cause value of the encapsulated REL set to '19'? Check: If a Reason header is present, is the cause value equal to the value in the Cause value of the encapsulated ISUP REL? Check: Is a 487 Request Terminating send from the terminating user? Check: Are the media streams terminated after the 200 OK CANCEL/BYE was sent? NOTE: A ISUP REL is not encapsulated in a CANCEL request. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 61
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7.1.5 Test purposes for Supplementary services
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101 585
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7.1.5.1 Test purposes for OIP
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Test case number SS_oip_001 Test case group SIP-SIP/Service/OIP Reference 5.2.6.3/ [2] SELECTION EXPRESSION Test purpose No P-Preferred-Identity received. The terminating user receives the default public user identity of the originating user. In case the preconditions are fulfilled to provide the terminating UE with originating identification information without preventing the presentation, ensure that no identity information in the P-Preferred-Identity header is provided by the originating UE, the terminating user receives a P-Asserted-Identity based on the default public user identity associated with the originating UE identifies the originator of the session. Configuration SIP Parameter INVITE P-Asserted-Identity= default public user identity Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Check: Is the P-Asserted-Identity set to the default public user identity? Check: Is optional a second P-Asserted-Identity header present as a 'tel' URI with a public user identity? Check: Is the user parameter set to phone? Repeat this test in reverse direction. Repeat this test with all relevant end devices. Test case number SS_oip_002 Test case group SIP-SIP/Service/OIP Reference 5.2.6.3/ [2] SELECTION EXPRESSION Test purpose P-Preferred-Identity received, no match with the set of registered public identities. The terminating user receives the default public user identity of the originating user. In case the preconditions are fulfilled to provide the terminating UE with originating identification information without preventing the presentation, ensure that an identity information in the P-Preferred-Identity header is provided by the originating UE, does not match with the set of registered public identities of the originating UE the terminating user receives a P-Asserted-Identity based on the default public user identity associated with the originating UE identifies the originator of the session. Configuration SIP Parameter INVITE P-Asserted-Identity= default public user identity Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Check: Is the P-Asserted-Identity set to the default public user identity? Check: Is optional a second P-Asserted-Identity header present as a 'tel' URI with a public user identity? Check: If the user parameter is set to phone? Check: Is the P-Preferred-Identity header not present? Repeat this test in reverse direction. Repeat this test with all relevant end devices. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 62 Test case number SS_oip_003 Test case group SIP-SIP/Service/OIP Reference 5.2.6.3/ [2] SELECTION EXPRESSION Test purpose P-Preferred-Identity received, match with the set of registered public identities. The terminating user receives the registered public user identity of the originating user. In case the preconditions are fulfilled to provide the terminating UE with originating identification information without preventing the presentation, ensure that an identity information in the P-Preferred-Identity header is provided by the originating UE, matches with the set of registered public identities of the originating UE the terminating user receives a P-Asserted-Identity based on the information provided by the originating UE identifies the originator of the session. Configuration SIP Parameter INVITE P-Asserted-Identity= matched public user identity Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Check: Is the P-Asserted-Identity set to the identified public user identity? Check: Is optional a second P-Asserted-Identity header present as a 'tel' URI with a public user identity? Check: Is the user parameter is set to phone? Check: Is the P-Preferred-Identity header not present? Repeat this test in reverse direction. Repeat this test with all relevant end devices. Test case number SS_oip_004 Test case group SIP-SIP/Service/OIP Reference 4.5.2.4/ [7] SELECTION EXPRESSION SE 18 AND NOT SE 19 Test purpose No Special arrangement exists. The special arrangement does not exist (screening of user provided information). The network compares the information in the From header with the set of registered public identities of the originating user If is no match is found, the AS sets the From header to the SIP URI that includes the registered default public user identity. Configuration Special arrangement for the originating user does not exist SIP Parameter INVITE From=default public user identity P-Asserted-Header=[any registered public user identity] Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Check: Is the From header URI set to the value of the P-Asserted-Identity URI? Check: Is the P-Asserted-Identity set to any registered public user identity? Check: Is the user parameter set to phone? Repeat this test in reverse direction. Repeat this test with all relevant end devices. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 63 Test case number SS_oip_005 Test case group SIP-SIP/Service/OIP Reference 4.5.2.4/ [7] SELECTION EXPRESSION SE 18 AND SE 19 Test purpose Special arrangement exists. The special arrangement exists (no screening of user provided information). The network does not attempt to match the information in the From header with the set of registered public identities of the originating user. The From header field is transparently transported to the terminating user. Configuration Special arrangement for the originating user exists SIP Parameter INVITE From= original value P-Asserted-Header=[any registered public user identity] Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Check: Is the From header URI set to original value sent by the user? Check: Is the P-Asserted-Identity set to any registered public user identity? Check: Is the user parameter set to phone? Repeat this test in reverse direction. Repeat this test with all relevant end devices. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 64 Test case number SS_oip_006 Test case group SIP-SIP/Service/OIP Reference 7.1.3/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 52 Test purpose SIP-I support. ISUP Calling party number presentation allowed in the encapsulated IAM. Ensure when BICC/ISUP - SIP-I interworking applies in the originating network the BICC/ISUP IAM is encapsulated in the INVITE request. The P-Asserted- Identity header field is derived from the Calling party number in the encapsulated IAM. The 'Presentation restriction' indicator in the encapsulated IAM is set to 'allowed' no Privacy value 'id' is present in the INVITE request. Configuration SIP Parameter INVITE P-Asserted-Identity=[derived from the ISUP calling party number] Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Calling party number Screening indicator Network provided or user provided, verified and passed Presentation restriction [24] allowed Address signal --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(IAM) Apply post test routine Comments Check: Is a BICC/ISUP IAM encapsulated in the in the INVITE request? Check: Is the Calling party number present in the encapsulated IAM and the screening indicator is set to 'Network provided' or 'user provided, verified and passed' and the Presentation restriction indicator is set to 'allowed'? Check: Is the P-Asserted-Identity header field derived from the Calling party number in the encapsulated IAM? Check: Is the value 'id' not present in the Privacy header field (if included)? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 65 Test case number SS_oip_007 Test case group SIP-SIP/Service/OIP Reference 7.1.3/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 52 Test purpose SIP-I support. ISUP Additional Calling party number presentation allowed in the encapsulated IAM. Ensure when BICC/ISUP - SIP-I interworking applies in the originating network the BICC/ISUP IAM is encapsulated in the INVITE request. The From field is derived from the Additional Calling party number in the encapsulated IAM. The 'Presentation restriction' indicator in the encapsulated IAM is set to 'allowed' no Privacy value 'id' is present in the INVITE request. Configuration The originating user in the PSTN/PLMN part of Network A is subscribed to the 'no screening option' SIP Parameter INVITE From=[derived from the ISUP Additional calling party number] P-Asserted-Identity=[derived from the ISUP calling party number] Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Calling party number Screening indicator Network Provided Presentation restriction allowed Address signal Generic number Number Qualifier Indicator Additional calling party number Screening indicator user provided, not verified Presentation restriction allowed Address signal --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(IAM) Apply post test routine Comments Check: Is a BICC/ISUP IAM encapsulated in the in the INVITE request? Check: Is the Calling party number present in the encapsulated IAM and the screening indicator is set to 'Network Provided' and the Presentation restriction indicator is set to 'allowed'? Check: Is the P-Asserted-Identity header field derived from the Calling party number in the encapsulated IAM? Check: Is a Generic number parameter, Number Qualifier Indicator set to Additional calling party number present and the screening indicator is set to 'user provided, not verified' and the Presentation restriction indicator is set to 'allowed'? Check: Is the From header field derived from the Additional calling party number in the encapsulated IAM? Check: Is the value 'id' not present in the Privacy header field (if included)? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 66
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7.1.5.2 Test purposes for OIR
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Test case number SS_oir_001 Test case group SIP-SIP/Service/OIR Reference 4.3.2, 4.5.2.4/ [7] SELECTION EXPRESSION SE 20 Test purpose Terminating user does not receive the identity of the originating user. In case the preconditions are fulfilled not to provide the terminating UE with originating identification information (e.g. permanent mode ), ensure that the P- Asserted-Identity still contains identity information and the privacy is set to 'id' or 'header' or 'user'. The terminating user does not receive the identity of the originating user. As a network option, the From header is set to an anonymous User Identity. Configuration Originating user subscribes to the OIR service SIP Parameter INVITE P-Asserted-Identity: Privacy:id OR header OR user From: <sip:anonymous@anonymous.invalid> (optional) Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Check: Is the P-Asserted-Identity is present? Check: Is the Privacy header set to 'id' or 'header' or 'user'? Check: Is optional the From header set to an anonymous User Identity? Repeat this test in reverse direction. Repeat this test with all chosen end devices. Test case number SS_oir_002 Test case group SIP-SIP/Service/OIR Reference 4.3.2, 4.5.2.4/ [7] SELECTION EXPRESSION SE 20 AND SE 25 Test purpose Communication forwarding unconditional, served user subscribes OIR. The user A and user C are in network B and user C is provided with OIP. The user B is in network A and is provided with CFU "diverting number is released to the diverted-to user" = Yes. In case the served user subscribes Originating Identification Restriction (e.g. permanent mode), ensure that when user A calls user B, the call is forwarded unconditional to user C, user C is not informed of the forwarding number. The diverted-to user receives no identity of the diverting user neither in a History-Info header nor in the To header. Configuration Diverting user subscribes to the OIR service SIP Parameter INVITE1: no history entry present INVITE2: History-Info header: <sip:userB@networkA?Privacy=history >;index=1, <sip: userC@networkB;cause=302 >;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE1 CFU is performed in Network A INVITE2 Apply post test routine Comments Check: No History-Info header is received in the INVITE from Network B. Check: Is the Privacy value history is escaped in the hi-targed-to-uri of the diverting user in Network A? Repeat this test in reverse direction. Repeat this test with all chosen end devices. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 67 Test case number SS_oir_003 Test case group SIP-SIP/Service/OIR Reference 7.1.3/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 52 Test purpose SIP-I support. ISUP Calling party number presentation restricted in the encapsulated IAM. Ensure when BICC/ISUP - SIP-I interworking applies in the originating network the BICC/ISUP IAM is encapsulated in the INVITE request. The P-Asserted-Identity header field is derived from the Calling party number in the encapsulated IAM. The 'Presentation restriction' indicator in the encapsulated IAM is set to 'restricted' the value 'id' is present in the Privacy header of the INVITE request. Configuration SIP Parameter INVITE P-Asserted-Identity=[derived from the ISUP calling party number] Privacy: id Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Calling party number Screening indicator Network provided or user provided, verified and passed Presentation restriction restricted Address signal --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(IAM) Apply post test routine Comments Check: Is a BICC/ISUP IAM encapsulated in the in the INVITE request? Check: Is the Calling party number present in the encapsulated IAM and the screening indicator is set to 'Network provided' or 'user provided, verified and passed' and the Presentation restriction indicator is set to 'restricted'? Check: Is the P-Asserted-Identity header field derived from the Calling party number in the encapsulated IAM? Check: Is the value 'id' present in the Privacy header field? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 68 Test case number SS_oir_004 Test case group SIP-SIP/Service/OIR Reference 7.1.3/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 52 Test purpose SIP-I support. ISUP Additional Calling party number presentation restricted in the encapsulated IAM. Ensure when BICC/ISUP - SIP-I interworking applies in the originating network the BICC/ISUP IAM is encapsulated in the INVITE request. The From field is derived from the Additional Calling party number in the encapsulated IAM. The 'Presentation restriction' indicator in the Generic number parameter is set to 'allowed' no Privacy value 'id' is present in the INVITE request. Configuration The originating user in the PSTN/PLMN part of Network A is subscribed to the 'no screening option' SIP Parameter INVITE P-Asserted-Identity=[derived from the ISUP calling party number] From=[derived from the ISUP Additional calling party number] Privacy: id Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Calling party number Screening indicator Network Provided Presentation restriction restricted Address signal Generic number Number Qualifier Indicator Additional calling party number Screening indicator user provided, not verified Presentation restriction restricted Address signal --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(IAM) Apply post test routine Comments Check: Is a BICC/ISUP IAM encapsulated in the in the INVITE request? Check: Is the Calling party number present in the encapsulated IAM and the screening indicator is set to 'Network Provided' and the Presentation restriction indicator is set to 'restricted'? Check: Is the P-Asserted-Identity header field derived from the Calling party number in the encapsulated IAM? Check: Is a Generic number parameter, Number Qualifier Indicator set to Additional calling party number present and the screening indicator is set to 'user provided, not verified' and the Presentation restriction indicator is set to 'restricted'? Check: Is the From header field derived from the Additional calling party number in the encapsulated IAM? Check: Is the value 'id' present in the Privacy header field? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 69
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7.1.5.3 Test purposes for TIP
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Test case number SS_tip_001 Test case group SIP-SIP/Service/TIP Reference 5.2.6.4/ [8] SELECTION EXPRESSION Test purpose Originating user receives the identity of the terminating user. Ensure in case the preconditions are fulfilled to provide the originating UE with terminating identification information without preventing the presentation , the originating UE receives in a 1xx or 200 SIP response a P-Asserted-Identity header field with a valid public user identity of the terminating UE. Configuration SIP Parameter 18x/200 OK INVITE P-Asserted-Identity: Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE CASE A 180 Ringing CASE B 183 Session Progress CASE C 200 OK INVITE(P-Asserted-Identity) Apply post test routine Comments Check: Is the P-Asserted-Identity is present in a 180 Ringing or 183 Session Progress or in a 200 OK INVITE? Repeat this test in reverse direction. Repeat this test with all relevant end devices. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 70 Test case number SS_tip_002 Test case group SIP-SIP/Service/TIP Reference 4.5.2.9/ [8] SELECTION EXPRESSION SE 21 AND SE 22 AND [Network B] SE 48 Test purpose Second identity provided in UPDATE. Ensure that, when the option tag "from-change" in the Supported header field is provided by the originating UE in the INVITE request and the terminating UE receives the from-change tag, The terminating user sends a 'from-change' tag in the supported header in the 200 OK INVITE a second identity is provided in the UPDATE request sent by the terminated user in the From header after the ACK was received. Configuration Special arrangement for the terminating user exists SIP Parameter INVITE Supported: from-change 200 OK INVITE Supported: from-change P-Asserted-Identity: UPDATE From: (second user identity) Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE(P-Asserted-Identity) ACK UPDATE (From) 200 OK UPDATE Apply post test routine Comments Check: Is the 'from-change' tag present in the Supported header of the initial INVITE request? Check: Is the P-Asserted-Identity present in a 180 Ringing or 183 Session Progress or in a 200 OK INVITE? Check: Is the 'from-change' tag present in the supported header of the provisional (18x) or final (200 OK) response? Check: Is an UPDATE request sent by the terminating user containing a From header field set to the value send by the terminating user? Repeat this test in reverse direction. Repeat this test with all chosen end devices. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 71 Test case number SS_tip_003 Test case group SIP-SIP/Service/TIP Reference 4.5.2.9/ [8] SELECTION EXPRESSION SE 21 AND SE 22 AND [Network B] SE 48 Test purpose Second identity not provided. Ensure that, when the option tag "from-change" in the Supported header field is provided by the originating UE in the INVITE request, the terminating user does not receive the from-change tag in the initial INVITE, no from-change tag is sent in the 200 OK INVITE response, an UPDATE containing a second identity is sent and the From header is set to the default public user identity of the terminating user. Configuration Special arrangement for the terminating user does not exist SIP Parameter INVITE Supported: from-change 200 OK INVITE P-Asserted-Identity: UPDATE From: (default public user identity) Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE(P-Asserted-Identity) ACK UPDATE (From) 200 OK UPDATE Apply post test routine Comments Check: Is the 'from-change' tag present in the Supported header of the initial INVITE request? Check: Is the P-Asserted-Identity present in the 200 OK INVITE? Check: Is the 'from-change' tag present in the supported header of the provisional (18x) or final (200 OK) response? Check: Is an UPDATE request sent by the terminating user containing a From header field set to the public user identity of the terminating user? Repeat this test in reverse direction. Repeat this test with all relevant end devices. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 72 Test case number SS_tip_004 Test case group SIP-SIP/Service/TIP Reference 6.7/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 53 Test purpose SIP-I support. The Connected number presentation allowed is present in the encapsulated 200 OK. Ensure that on receipt of a 200 OK INVITE to establish a confirmed dialogue an ANM is encapsulated if SIP-I - BICC/ISUP interworking is applicable in Network B. The Address presentation restriction indicator is set to 'allowed'. The screening indicator is set to Network provided or user provided, verified and passed. Configuration SIP Parameter 200 OK INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Connected number Screening indicator Network provided or user provided, verified and passed Address presentation restriction allowed Address signal --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(IAM) 180 Ringing(ACM) 200 OK INVITE(ANM) ACK Apply post test routine Comments Check: Is the BICC/ISUP ANM encapsulated in the 200 OK INVITE final response? Check: Is the Screening indicator in the encapsulated ANM set to 'Network provided' or 'user provided, verified and passed'? Check: Is the Address presentation restriction indicator in the encapsulated ANM set to allowed? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 73 Test case number SS_tip_005 Test case group SIP-SIP/Service/TIP Reference 6.7/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 53 Test purpose SIP-I support. The additional connected number restricted is present in the encapsulated 200 OK. Ensure that on receipt of a 200 OK INVITE to establish a confirmed dialogue an ANM is encapsulated if SIP-I - BICC/ISUP interworking is applicable in Network B. A Generic number parameter is present the Number qualifier indicator set to 'additional connected number' the Screening indicator is set to 'user provided, not verified' and the Address Presentation Restricted is set to 'allowed'. A Connected number parameter is present the Screening indicator is set to 'Network provided' and the Address Presentation Restricted indicator is set to 'allowed'. Configuration The terminating user in the PSTN/PLMN part of Network B is subscribed to the COLP 'no screening option' SIP Parameter 200 OK INVITE P-Asserted-Identity=[derived from the ISUP Connected number] Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Connected number Screening indicator Network provided or user provided, verified and passed Presentation restriction allowed Address signal Generic number Number Qualifier Indicator Additional calling party number Screening indicator user provided, not verified Address Presentation Restricted allowed Address signal --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(IAM) 180 Ringing(ACM) 200 OK INVITE(ANM) ACK Apply post test routine Comments Check: Is the BICC/ISUP ANM encapsulated in the 200 OK INVITE final response? Check: Is a Generic number parameter present in the encapsulated ANM? Check: Is the Number Qualifier Indicator of the Generic number set to 'additional connected number'? Check: Is the Screening indicator of the Generic number set to 'user provided, not verified'? Check: Is the Address presentation restriction indicator in the Generic number set to 'allowed'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 74
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7.1.5.4 Test purposes for TIR
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Test case number SS_tir_001 Test case group SIP-SIP/Service/TIR Reference 4.5.2.9/ [8] SELECTION EXPRESSION SE 23 Test purpose Originating user does not receive the identity of the terminating user. Ensure that, when the preconditions are fulfilled to prevent the presentation of the terminating user identity at the originating user, the originating UE receives, in any non-100 SIP response (e.g. 180, 183, 200), a Privacy header field is set to "id" and no P-Asserted-Identity header field is present. Configuration The terminating user subscribes to the 'TIR' service SIP Parameter 18x/200 OK INVITE P-Asserted-Identity: Privacy: id Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE CASE A 180 Ringing CASE B 183 Session Progress CASE C 200 OK INVITE(P-Asserted-Identity) Apply post test routine Comments Check: Is the P-Asserted-Identity is present in the provisional (18x) or final (200 OK) response? Check: Is the Privacy header in the provisional (18x) or final (200 OK) response is set to 'id'? Repeat this test in reverse direction. Repeat this test with all chosen end devices. Test case number SS_tir_001A Test case group SIP-SIP/Service/TIR Reference 4.5.2.6.2.2/ [9] SELECTION EXPRESSION SE 23 Test purpose CDIV occurs. Originating user does not receive the identity of the served user. Ensure that, when Call diversion occurs, the identity of the CDIV served user is restricted when the CDIV served user is subscribed to the TIR service and requires to prevent the presentation of his/here identity. The hi-entry of the History-Info header in the 181 identifying the served user contains an escaped 'Privacy' header set to 'history'. Configuration The served user subscribes to the 'TIR' service SIP Parameter 181 History-Info header: <sip:userB@networkB?Privacy=history>;index=1, <sip: userC@networkB;cause=[any] >;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 181 Being Forwarded INVITE Apply post test routine Comments Check: Is the History-Info header present in the 181 sent to the originating user? Check: Is the Privacy header is escaped in the hi-entry identify the served user set to 'history'? Repeat this test in reverse direction. Repeat this test with all chosen end devices. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 75 Test case number SS_tir_001B Test case group SIP-SIP/Service/TIR Reference 4.5.2.7/ [9] SELECTION EXPRESSION SE 23 Test purpose CDIV occurs. Originating user does not receive the identity of the served user. Ensure that, when Call diversion occurs, the identity of the diverted-to user is restricted when the diverted-to user is subscribed to the TIR service and requires to prevent the presentation of his/here identity. The hi-entry of the History-Info header in the 180 or 200 OK INVITE identifying the diverted-to user contains an escaped 'Privacy' header set to 'history'. Configuration The diverted-to user subscribes to the 'TIR' service SIP Parameter 180/200 OK History-Info header: <sip:userB@networkB>;index=1, <sip: userC@networkB;cause=[any]?Privacy=history>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(1) INVITE(2) 180 Ringing(2) 180 Ringing(1) 200 OK INVITE(2) ACK 200 OK INVITE(1) ACK Apply post test routine Comments Check: Is the History-Info header present in the 180 or 200 OK sent to the originating user? Check: Is the Privacy header is escaped in the hi-entry identify the diverted-to user set to 'history'? Repeat this test in reverse direction. Repeat this test with all chosen end devices. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 76 Test case number SS_tir_002 Test case group SIP-SIP/Service/TIR Reference 6.7/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 53 Test purpose SIP-I support. The Connected number presentation allowed is present in the encapsulated 200 OK. Ensure that on receipt of a 200 OK INVITE to establish a confirmed dialogue an ANM is encapsulated if SIP-I - BICC/ISUP interworking is applicable in Network B. The Address presentation restriction indicator is set to 'restricted'. The screening indicator is set to 'Network provided' or 'user provided, verified and passed'. Configuration SIP Parameter 200 OK INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Connected number Screening indicator Network provided or user provided, verified and passed Address presentation restriction restricted Address signal --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(IAM) 180 Ringing(ACM) 200 OK INVITE(ANM) ACK Apply post test routine Comments Check: Is the BICC/ISUP ANM encapsulated in the 200 OK INVITE final response? Check: Is the Screening indicator in the encapsulated ANM set to 'Network provided' or 'user provided, verified and passed'? Check: Is the Address presentation restriction indicator in the encapsulated ANM set to allowed? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 77 Test case number SS_tir_003 Test case group SIP-SIP/Service/TIR Reference 6.7/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 53 Test purpose SIP-I support. The additional connected number restricted is present in the encapsulated 200 OK. Ensure that on receipt of a 200 OK INVITE to establish a confirmed dialogue an ANM is encapsulated if SIP-I - BICC/ISUP interworking is applicable in Network B. A Generic number parameter is present the Number qualifier indicator set to 'additional connected number' the Screening indicator is set to 'user provided, not verified' and the Address Presentation Restricted is set to 'restricted'. A Connected number parameter is present the Screening indicator is set to 'Network provided' and the Address Presentation Restricted indicator is set to 'restricted'. Configuration The terminating user in the PSTN/PLMN part of Network B is subscribed to the COLP 'no screening option' SIP Parameter 200 OK INVITE P-Asserted-Identity=[derived from the ISUP Connected number] Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Connected number Screening indicator Network provided or user provided, verified and passed Presentation restriction restricted Address signal Generic number Number Qualifier Indicator Additional calling party number Screening indicator user provided, not verified Address Presentation Restricted restricted Address signal --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(IAM) 180 Ringing(ACM) 200 OK INVITE(ANM) ACK Apply post test routine Comments Check: Is the BICC/ISUP ANM encapsulated in the 200 OK INVITE final response? Check: Is a Generic number parameter present in the encapsulated ANM? Check: Is the Number Qualifier Indicator of the Generic number set to 'additional connected number'? Check: Is the Screening indicator of the Generic number set to 'user provided, not verified'? Check: Is the Address presentation restriction indicator in the Generic number set to 'allowed'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 78
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7.1.5.5 Communication Hold (HOLD)
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Test case number SS_hold_001 Test case group SIP-SIP/Service/HOLD Reference 4.5.2.1/ [17] SELECTION EXPRESSION SE 24 Test purpose Hold the session the media stream was previously set to sendrecv. Ensure that the UE A requesting hold of the session sends an INVITE or UPDATE request to hold the session. Hold is done containing the SDP with the attribute "a=sendonly". The UE A after requesting the hold session receives 200 OK final response containing the SDP with the attribute "a=recvonly". Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists CASE A INVITE(sendonly) 200 OK INVITE (recvonly) ACK CASE B UPDATE(sendonly) 200 OK UPDATE (recvonly) Apply post test routine Comments Check: Is the user in network A able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 79 Test case number SS_hold_002 Test case group SIP-SIP/Service/HOLD Reference 4.5.2.1/ [17] SELECTION EXPRESSION SE 24 Test purpose Hold the session the media stream was previously set to recvonly. Ensure that the UE B requesting hold of the session stops sending media and sends an INVITE or UPDATE request to hold the session. Hold is done containing the SDP with the attribute "a=sendonly". The UE A after requesting to hold the held session sends an INVITE or UPDATE request containing the SDP with the attribute "a=inactive". Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists CASE A INVITE (sendonly) 200 OK INVITE (recvonly) ACK INVITE (inactive) 200 OK INVITE (inactive) ACK CASE B INVITE (sendonly) 200 OK INVITE (recvonly) ACK UPDATE(inactive) 200 OK UPDATE (inactive) CASE C UPDATE (sendonly) 200 OK UPDATE (recvonly) INVITE (inactive) 200 OK INVITE (inactive) ACK CASE D UPDATE (sendonly) 200 OK UPDATE (recvonly) UPDATE(inactive) 200 OK UPDATE (inactive) Apply post test routine Comments Check: Is the user in network B able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Check: Is the user in network A able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 80 Test case number SS_hold_003 Test case group SIP-SIP/Service/HOLD Reference 4.5.2.1/ [17] SELECTION EXPRESSION SE 24 Test purpose Resume the session the media stream was previously set to sendonly. Ensure that the UE A is requested to resume the session with user B the UE-A starts sending media and sends an INVITE or UPDATE request to resume the session with the attribute "a=sendrecv in the SDP. The UE A after requesting to resume the held session receives 200 OK final response and optionally the attribute "a=sendrecv in the SDP. The a=sendrecv attribute is the default value therefore the attribute can be omitted. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists CASE A INVITE (sendonly) 200 OK INVITE (recvonly) ACK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK CASE B UPDATE (sendonly) 200 OK UPDATE (recvonly) UPDATE (sendrecv) 200 OK UPDATE (sendrecv) Apply post test routine Comments Check: Is the user in network A able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Check: Is the user in network A able to retrieve the session by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? The absence of the 'sendrecv' attribute is the default value. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 81 Test case number SS_hold_004 Test case group SIP-SIP/Service/HOLD Reference 4.5.2.1/ [17] SELECTION EXPRESSION SE 24 Test purpose Resume the session the media stream was previously set to inactive. The Session is in the "inactive" state. Ensure that the UE A is requesting to resume the session with user B the UE-A sends an INVITE or UPDATE to resume the session with the attribute "a=recvonly in the SDP. The UE A after requesting to resume the held session receives 200 OK final response with the attribute "a=sendonly in the SDP. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists CASE A INVITE(sendonly) 200 OK INVITE (recvonly) ACK INVITE(inactive) 200 OK INVITE (inactive) ACK INVITE (recvonly) 200 OK INVITE (sendonly) ACK CASE B INVITE(sendonly) 200 OK INVITE (recvonly) ACK UPDATE(inactive) 200 OK UPDATE (inactive) INVITE (recvonly) 200 OK (sendonly) ACK CASE C UPDATE (sendonly) 200 OK UPDATE (recvonly) INVITE(inactive) 200 OK INVITE (inactive) ACK UPDATE (recvonly) 200 OK UPDATE (sendonly) CASE D UPDATE (sendonly) 200 OK UPDATE (recvonly) UPDATE(inactive) 200 OK UPDATE (inactive) UPDATE (recvonly) 200 OK UPDATE (sendonly) Apply post test routine Comments Check: Is the user in network B able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Check: Is the user in network A able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Check: Is the user in network A able to retrieve the session by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 82 Test case number SS_hold_005 Test case group SIP-SIP/Service/HOLD Reference 4.5.2.1/ [17] SELECTION EXPRESSION SE 24 Test purpose Hold the session the media stream was previously set to sendrecv. Ensure that the UE B sends an INVITE or UPDATE request to hold the session. Hold is done containing the SDP with the attribute "a=sendonly". The UE A sends a 200 OK final response containing the SDP with the attribute "a=recvonly" and stops sending media. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists CASE A INVITE(sendonly) 200 OK INVITE(recvonly) ACK CASE B UPDATE(sendonly) 200 OK UPDATE (recvonly) Apply post test routine Comments Check: Is the user in network B able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 83 Test case number SS_hold_006 Test case group SIP-SIP/Service/HOLD Reference 4.5.2.1/ [17] SELECTION EXPRESSION SE 24 Test purpose Hold the session the media stream was previously set to sendonly. The Session is in the held state done by UE-A. Ensure that the UE B sends an INVITE or UPDATE request to hold the session. Hold is done containing the SDP with the attribute "a=inactive". The UE A after receiving the hold request sends 200 OK final response containing the SDP with the attribute "a=inactive" and stops sending media. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists CASE A INVITE(sendonly) 200 OK INVITE (recvonly) ACK INVITE (inactive) 200 OK INVITE (inactive) ACK CASE B INVITE(sendonly) 200 OK INVITE (recvonly) ACK UPDATE (inactive) 200 OK UPDATE (inactive) CASE C UPDATE (sendonly) 200 OK UPDATE (recvonly) INVITE (inactive) 200 OK INVITE (inactive) ACK CASE D UPDATE (sendonly) 200 OK UPDATE (recvonly) UPDATE (inactive) 200 OK UPDATE (inactive) Apply post test routine Comments Check: Is the user in network A able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Check: Is the user in network B able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 84 Test case number SS_hold_007 Test case group SIP-SIP/Service/HOLD Reference 4.5.2.1/ [17] SELECTION EXPRESSION SE 24 Test purpose Resume the session the media stream was previously set to recvonly. Ensure that the UE B sends an INVITE or UPDATE request requesting to resume the session with user A, the UE-B starts sending media. Resume is done containing the SDP with the attribute "a=sendrecv". The UE A after receiving the Resume of the session sends 200 OK final response containing the SDP with the attribute "a=sendrecv". The a=sendrecv attribute is the default value therefore the attribute can be omitted. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists CASE A INVITE (sendonly) 200 OK INVITE(recvonly) ACK INVITE(sendrecv) 200 OK INVITE(sendrecv) ACK CASE B UPDATE (sendonly) 200 OK UPDATE (recvonly) UPDATE (sendrecv) 200 OK UPDATE (sendrecv) Apply post test routine Comments Check: Is the user in network B able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Check: Is the user in network B able to retrieve the session by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 85 Test case number SS_hold_008 Test case group SIP-SIP/Service/HOLD Reference 4.5.2.1/ [17] SELECTION EXPRESSION SE 24 Test purpose Resume the session the media stream was previously set to inactive. The Session is in the "inactive" state. Ensure that the UE B sends an INVITE or UPDATE request requesting to resume the session with user A, the UE-A starts sending media. Resume is done containing the SDP with the attribute "a=recvonly". The UE A after receiving the Resume of the session sends 200 OK final response containing the SDP with the attribute "a=sendonly". Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists CASE A INVITE (sendonly) 200 OK INVITE (recvonly) ACK INVITE (inactive) 200 OK INVITE (inactive) ACK INVITE (recvonly) 200 OK INVITE (sendonly) ACK CASE B INVITE (sendonly) 200 OK INVITE (recvonly) ACK UPDATE (inactive) 200 OK UPDATE (inactive) UPDATE (recvonly) 200 OK UPDATE (sendonly) CASE C UPDATE (sendonly) 200 OK UPDATE (recvonly) INVITE (inactive) 200 OK INVITE (inactive) ACK INVITE (recvonly) 200 OK INVITE (sendonly) ACK CASE D UPDATE (sendonly) 200 OK UPDATE (recvonly) UPDATE (inactive) 200 OK UPDATE (inactive) UPDATE (recvonly) 200 OK UPDATE (sendonly) Apply post test routine Comments Check: Is the user in network A able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Check: Is the user in network B able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Check: Is the user in network B able to retrieve the session by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 86 Test case number SS_hold_009 Test case group SIP-SIP/Service/HOLD Reference 4.5.2.1/ [17] SELECTION EXPRESSION SE 24 Test purpose Resume the session on both sides the media stream was previously set to inactive. The Session is in the "inactive" state. Ensure that the UE A is requesting to resume the session with user B, the UE-A starts sending media and sends an INVITE or UPDATE request to resume the session with the attribute "a=sendonly in the SDP. The UE A after requests to resume the session receives 200 OK final response containing the SDP with the attribute "a=recvonly. The UE B after requests to resume the session sends an INVITE or UPDATE request containing the SDP with the attribute "a=sendrecv". The a=sendrecv attribute is the default value therefore the attribute can be omitted. Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists CASE A INVITE(sendonly) 200 OK INVITE (recvonly) ACK INVITE(inactive) 200 OK INVITE (inactive) ACK INVITE(sendonly) 200 OK INVITE (recvonly) ACK INVITE(sendrecv) 200 OK INVITE (sendrecv) ACK CASE B INVITE(sendonly) 200 OK INVITE (recvonly) ACK UPDATE (inactive) 200 OK UPDATE (inactive) INVITE(sendonly) 200 OK INVITE (recvonly) ACK UPDATE (sendrecv) 200 OK UPDATE (sendrecv) CASE C UPDATE (sendonly) 200 OK UPDATE (recvonly) INVITE(inactive) 200 OK INVITE (inactive) ACK UPDATE (sendonly) 200 OK UPDATE (recvonly) ACK INVITE(sendrecv) 200 OK INVITE (sendrecv) ACK CASE D UPDATE (sendonly) 200 OK UPDATE (recvonly) UPDATE (inactive) 200 OK UPDATE (inactive) UPDATE (sendonly) 200 OK UPDATE (recvonly) UPDATE (sendrecv) 200 OK UPDATE (sendrecv) Apply post test routine ETSI ETSI TS 101 585 V1.2.1 (2014-04) 87 Comments Check: Is the user in network A able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Check: Is the user in network B able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Check: Is the user in network A able to retrieve the session by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Check: Is the user in network B able to retrieve the session by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? The absence of the 'sendrecv' attribute is the default value. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 88 Test case number SS_hold_010 Test case group SIP-SIP/Service/HOLD Reference 4.5.2.1/ [17] SELECTION EXPRESSION SE 24 Test purpose Resume the session on both sides the media stream was previously set to inactive. The Session is in the "inactive" state. Ensure that the UE B sends an INVITE or UPDATE request to resume the session with user A, the UE-B starts sending media. Resume is done containing the SDP with the attribute "a=sendonly". The UE A after receiving the Resume of the session sends 200 OK final response containing the SDP with the attribute "a=recvonly". The UE A after requests to resume the session sends an INVITE or UPDATE request containing the SDP with the attribute "a=sendrecv. The UE B after receiving the Resume of the session sends 200 OK final response containing the SDP with the attribute "a=sendrecv". The a=sendrecv attribute is the default value therefore the attribute can be omitted. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists CASE A INVITE(sendonly) 200 OK INVITE (recvonly) ACK INVITE(inactive) 200 OK INVITE (inactive) ACK INVITE(sendonly) 200 OK INVITE (recvonly) ACK INVITE(sendrecv) 200 OK INVITE (sendrecv) ACK CASE B INVITE(sendonly) 200 OK INVITE (recvonly) ACK UPDATE (inactive) 200 OK UPDATE (inactive) INVITE(sendonly) 200 OK INVITE (recvonly) ACK UPDATE (sendrecv) 200 OK UPDATE (sendrecv) CASE C UPDATE (sendonly) 200 OK UPDATE (recvonly) INVITE(inactive) 200 OK INVITE (inactive) ACK UPDATE (sendonly) 200 OK UPDATE (recvonly) INVITE(sendrecv) 200 OK INVITE (sendrecv) ACK CASE D UPDATE (sendonly) 200 OK UPDATE (recvonly) UPDATE (inactive) 200 OK UPDATE (inactive) UPDATE (sendonly) 200 OK UPDATE (recvonly) UPDATE (sendrecv) 200 OK UPDATE (sendrecv) Apply post test routine ETSI ETSI TS 101 585 V1.2.1 (2014-04) 89 Comments Check: Is the user in network B able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Check: Is the user in network A able to set the session on hold by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Check: Is the user in network B able to retrieve the session by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? Check: Is the user in network A able to retrieve the session by sending an INVITE or UPDATE request and the version parameter in the SDP 'o' line is incremented? The absence of the 'sendrecv' attribute is the default value. Repeat this test in reverse direction. Test case number SS_hold_011 Test case group SIP-SIP/Service/HOLD Reference B.10/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 54 Test purpose SIP-I support. Hold requested by the calling user. Ensure that when an INVITE request updates a confirmed session a CPG is encapsulated if ISUP - SIP-I interworking is applicable in Network A. The Generic Notification Indicator parameter is present set to 'hold'. The 'a' attribute is set to 'sendonly' present in the SDP. In the 200 OK INVITE the 'a' attribute is set to 'recvonly' present in the SDP. Configuration SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] a=sendonly --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic notification remote hold --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists CASE A INVITE(sendonly, CPG hold) 200 OK INVITE (recvonly) ACK Apply post test routine Comments Establish a session from Network A to Network B The user in the PSTN/PLMN part of Network A places the session on hold. Check: Is a CPG encapsulated in the INVITE request? Check: Is a Generic notification parameter present the Notification indicator set to 'remote hold'? Check: Is the 'a' attribute in the SDP set to 'sendonly'? Check: Is the Version parameter in the SDP incremented? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 90 Test case number SS_hold_012 Test case group SIP-SIP/Service/HOLD Reference B.10/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 54 Test purpose SIP-I support. Hold requested by the called user. Ensure that when an INVITE request updates a confirmed session a CPG is encapsulated if SIP-I - ISUP interworking is applicable in Network B. The Generic Notification Indicator parameter is present set to 'hold'. The 'a' attribute is set to 'sendonly' present in the SDP. In the 200 OK INVITE the 'a' attribute is set to 'recvonly' present in the SDP. Configuration SIP Parameter INVITE: Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] a=sendonly --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic notification remote hold --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists CASE A INVITE(sendonly, CPG hold) 200 OK INVITE (recvonly) ACK Apply post test routine Comments Establish a session from Network A to Network B The user in the PSTN/PLMN part of Network B places the session on hold. Check: Is a CPG encapsulated in the INVITE request? Check: Is a Generic notification parameter present the Notification indicator set to 'remote hold'? Check: Is the 'a' attribute in the SDP set to 'sendonly'? Check: Is the Version parameter in the SDP incremented? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 91 Test case number SS_hold_013 Test case group SIP-SIP/Service/HOLD Reference B.10/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 54 Test purpose SIP-I support. Hold requested by the originating user, Hold by the terminating user. Retrieve requested by the originating user. Ensure the hold and retrieve procedure when ISUP - SIP-I interworking applies in the Network A: • Originating user in Network A places the session on hold. • Terminating user in Network B places the session on hold. • Originating user in Network A retrieves the session. • Terminating user in Network B retrieves the session. Verify the Generic notification parameter in the encapsulated CPG present in the INVITE request from the Network A. Configuration SIP Parameter INVITE: Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic notification remote hold or remote retrieval --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists CASE A INVITE(sendonly, CPG hold) 200 OK INVITE (recvonly) ACK INVITE(inactive) 200 OK INVITE (inactive) ACK INVITE(sendonly, CPG retrieval) 200 OK INVITE (recvonly) ACK INVITE(sendrecv) 200 OK INVITE (sendrecv) ACK Apply post test routine Comments Establish a session from Network A to Network B The user in the PSTN/PLMN part of Network A places the session on hold. Check: Is a CPG encapsulated in the INVITE request? Check: Is a Generic notification parameter present the Notification indicator set to 'remote hold'? Check: Is the 'a' attribute in the SDP set to 'sendonly'? Check: Is the Version parameter in the SDP incremented? The user in Network B places the session on hold Check: Is the 'a' attribute in the SDP set to 'inactive'? Check: Is the Version parameter in the SDP incremented? The user in Network A retrieves the session Check: Is a CPG encapsulated in the INVITE request? Check: Is a Generic notification parameter present the Notification indicator set to 'remote retrieval'? Check: Is the 'a' attribute in the SDP set to 'sendonly'? Check: Is the Version parameter in the SDP incremented? The user in Network B retrieves the session Check: Is the 'a' attribute in the SDP set to 'sendrecv'? Check: Is the Version parameter in the SDP incremented? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 92 Test case number SS_hold_014 Test case group SIP-SIP/Service/HOLD Reference B.10/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 54 Test purpose SIP-I support. Hold requested by the originating user, Hold by the terminating user. Retrieve requested by the terminating user. Ensure the hold and retrieve procedure when ISUP - SIP-I interworking applies in the Network A: • Originating user in Network A places the session on hold. • Terminating user in Network B places the session on hold. • Terminating user in Network B retrieves the session. • Originating user in Network A retrieves the session. Verify the Generic notification parameter in the encapsulated CPG present in the INVITE request from the Network A. Configuration SIP Parameter INVITE: Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic notification remote hold or remote retrieval --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists INVITE(sendonly, CPG hold) 200 OK INVITE (recvonly) ACK INVITE(inactive) 200 OK INVITE (inactive) ACK INVITE(recvonly) 200 OK INVITE (sendonly) ACK INVITE(sendrecv, CPG retrieval) 200 OK INVITE (sendrecv) ACK Apply post test routine Comments Establish a session from Network A to Network B The user in the PSTN/PLMN part of Network A places the session on hold. Check: Is a CPG encapsulated in the INVITE request? Check: Is a Generic notification parameter present the Notification indicator set to 'remote hold'? Check: Is the 'a' attribute in the SDP set to 'sendonly'? Check: Is the Version parameter in the SDP incremented? The user in Network B places the session on hold Check: Is the 'a' attribute in the SDP set to 'inactive'? Check: Is the Version parameter in the SDP incremented? The user in Network B retrieves the session Check: Is the 'a' attribute in the SDP set to 'recvonly'? Check: Is the Version parameter in the SDP incremented? The user in Network A retrieves the session Check: Is a CPG encapsulated in the INVITE request? Check: Is a Generic notification parameter present the Notification indicator set to 'remote retrieval'? Check: Is the 'a' attribute in the SDP set to 'sendrecv'? Check: Is the Version parameter in the SDP incremented? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 93 Test case number SS_hold_015 Test case group SIP-SIP/Service/HOLD Reference B.10/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 54 Test purpose SIP-I support. Hold requested by the terminating user, Hold by the originating user. Retrieve requested by the originating user. Ensure the hold and retrieve procedure when ISUP - SIP-I interworking applies in the Network A: • Terminating user in Network B places the session on hold. • Originating user in Network A places the session on hold. • Originating user in Network A retrieves the session. • Terminating user in Network B retrieves the session. Verify the Generic notification parameter in the encapsulated CPG present in the INVITE request from the Network A. Configuration SIP Parameter INVITE: Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic notification remote hold or remote retrieval --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists INVITE(sendonly) 200 OK INVITE (recvonly) ACK INVITE(inactive, CPG hold) 200 OK INVITE (inactive) ACK INVITE(recvonly, CPG retrieval) 200 OK INVITE (sendonly) ACK INVITE(sendrecv) 200 OK INVITE (sendrecv) ACK Apply post test routine Comments Establish a session from Network A to Network B The user in Network B places the session on hold. Check: Is the 'a' attribute in the SDP set to 'sendonly'? Check: Is the Version parameter in the SDP incremented? The user in Network A places the session on hold Check: Is a CPG encapsulated in the INVITE request? Check: Is a Generic notification parameter present the Notification indicator set to 'remote hold'? Check: Is the 'a' attribute in the SDP set to 'inactive'? Check: Is the Version parameter in the SDP incremented? The user in Network A retrieves the session Check: Is a CPG encapsulated in the INVITE request? Check: Is a Generic notification parameter present the Notification indicator set to 'remote retrieval'? Check: Is the 'a' attribute in the SDP set to 'recvonly'? Check: Is the Version parameter in the SDP incremented? The user in Network B retrieves the session Check: Is the 'a' attribute in the SDP set to 'sendrecv'? Check: Is the Version parameter in the SDP incremented? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 94 Test case number SS_hold_016 Test case group SIP-SIP/Service/HOLD Reference B.10/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 54 Test purpose SIP-I support. Hold requested by the terminating user, Hold by the originating user. Retrieve requested by the terminating user. Ensure the hold and retrieve procedure when ISUP - SIP-I interworking applies in the Network A: • Terminating user in Network B places the session on hold. • Originating user in Network A places the session on hold. • Terminating user in Network B retrieves the session. • Originating user in Network A retrieves the session. Verify the Generic notification parameter in the encapsulated CPG present in the INVITE request from the Network A. Configuration SIP Parameter INVITE: Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic notification remote hold or remote retrieval --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists INVITE(sendonly) 200 OK INVITE (recvonly) ACK INVITE(inactive, CPG hold) 200 OK INVITE (inactive) ACK INVITE(sendonly) 200 OK INVITE (recvonly) ACK INVITE(sendrecv, CPG retrieval) 200 OK INVITE (sendrecv) ACK Apply post test routine Comments Establish a session from Network A to Network B The user in Network B places the session on hold. Check: Is the 'a' attribute in the SDP set to 'sendonly'? Check: Is the Version parameter in the SDP incremented? The user in Network A places the session on hold Check: Is a CPG encapsulated in the INVITE request? Check: Is a Generic notification parameter present the Notification indicator set to 'remote hold'? Check: Is the 'a' attribute in the SDP set to 'inactive'? Check: Is the Version parameter in the SDP incremented? The user in Network B retrieves the session Check: Is the 'a' attribute in the SDP set to 'sendonly'? Check: Is the Version parameter in the SDP incremented? The user in Network A retrieves the session Check: Is a CPG encapsulated in the INVITE request? Check: Is a Generic notification parameter present the Notification indicator set to 'remote retrieval'? Check: Is the 'a' attribute in the SDP set to 'sendrecv'? Check: Is the Version parameter in the SDP incremented? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 95
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7f110eeff44cca5c59b5fc967e08482a
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101 585
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7.1.5.6 Communication Diversion (CDIV)
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7f110eeff44cca5c59b5fc967e08482a
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101 585
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7.1.5.6.1 Communication Forwarding Unconditional (CFU)
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Test case number SS_cfu_001 Test case group SIP-SIP/Service/CFU Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 25 Test purpose Communication forwarding unconditional, basic rules. The user A and user C are in Network A. The user B is in network B and is provided with CFU. Ensure that when user A calls user B, the call is forwarded unconditional to user C. In the active call state, ensure the property of speech. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) 200 OK INVITE(Call-ID C-B) ACK(Call-ID B-C) 200 OK INVITE(Call-ID B-A) ACK(Call-ID A-B) Communication Apply post test routine Comments Check: CDIV unconditional is successful. Check: In the active call state, ensure the property of speech. Check: Is the P-Asserted-Identity present set to the identity of the originating user? Repeat this test in reverse direction. Test case number SS_cfu_002 Test case group SIP-SIP/Service/CFU Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 25 AND SE 30 Test purpose Communication forwarding unconditional, no notification. The user A and user C are in Network A. The user B is in network B and is provided with CFU, subscription option: Originating user receives notification that his communication has been diverted = No. Ensure that when user A calls user B, the call is forwarded unconditional to user C, the originating user is not notified. Configuration Subscription options: Originating user receives notification that his communication has been diverted = No SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) Apply post test routine Comments Check: No notification regarding call forwarding in network B is received at the interconnection interface. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 96 Test case number SS_cfu_003 Test case group SIP-SIP/Service/CFU Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 25 AND SE 30 Test purpose Communication forwarding unconditional, originating user is notified. URI of the diverted-to user not received. The user A and user C are in network A. The user B is in network B and is provided with CFU "Originating user receives notification that his communication has been diverted" = Yes and "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = No and. "Served user allows the presentation of his/her URI to originating user in diversion notification" = No. Ensure that when user A calls user B, the call is forwarded unconditional to user C, user A is notified of call diversion and not informed of the diverted-to number and served user number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = No • Served user allows the presentation of his/her URI to originating user in diversion notification = No SIP Parameter 181 Being Forwarded History-Info: <sip:userB@networkB?Privacy=history>;index=1, <sip: userC@networkA;cause=302?Privacy=history>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) Apply post test routine Comments Check: A 181 Being Forwarded and a History-Info header is received at the interconnection interface in both entries in the History-Info header a Privacy header is escaped value 'history'. Check: Is the cause parameter in the last entry is set to '302' NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 97 Test case number SS_cfu_004 Test case group SIP-SIP/Service/CFU Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 25 AND SE 30 Test purpose Communication forwarding unconditional, originating user is notified. URI from the diverted-to user received. The user A and user C are in network 1. The user B is in network N2 and is provided with CFU "Originating user receives notification that his communication has been diverted" = Yes and "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = Yes. Ensure that when user A calls user B, the call is forwarded unconditional to user C, user A is notified of call diversion and informed of the diverted-to number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = Yes SIP Parameter 181 Being Forwarded History-Info: <sip:userB@networkB>;index=1, <sip: userC@networkA;cause=302>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) Apply post test routine Comments Check: A 181 Being Forwarded is received at the interconnection interface Check: A History-Info header is contained in the 181 with the URI of the diverted-to user. Check: Is the cause parameter in the last entry is set to '302'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 98 Test case number SS_cfu_005 Test case group SIP-SIP/Service/CFU Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 25 AND SE 30 Test purpose Communication forwarding unconditional, diverted-to user does not receive the URI of the served user. The user A and user C are in network A. The user B is in network B and is provided with CFU "Served user allows the presentation of his/her URI to the diverted-to user"= No. Ensure that when user A calls user B, the call is forwarded unconditional to user C, user C is not informed of the forwarding number. Configuration Subscription options: • Served user allows the presentation of his/her URI to the diverted-to user = No SIP Parameter INVITE: Request line contains ';cause=302' History-Info header: <sip:userB@networkB?Privacy=history>;index=1, <sip: userC@networkA;cause=302>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C) Apply post test routine Comments Check: A History-Info header is received in the INVITE contains the URI of user B (served user) at the interconnection interface and a Privacy header is escaped set to 'history'. Check: Is the 'cause' parameter present in the Request line sent to user C (diverted-to user) set to '302'? Check: Is the cause parameter in the last entry is set to '302'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 99 Test case number SS_cfu_006 Test case group SIP-SIP/Service/CFU Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 25 AND SE 30 Test purpose Communication forwarding unconditional, diverted-to user receives the URI of the served user. The user A and user C are in network A. The user B is in network B and is provided with CFU "Served user allows the presentation of his/her URI to diverted-to user" = Yes. Ensure that when user A calls user B, the call is forwarded unconditional to user C, user C is informed of the forwarding number. Configuration Subscription options: • Served user allows the presentation of his/her URI to diverted-to user = Yes SIP Parameter INVITE: Request line contains ';cause=302' History-Info header: <sip:userB@networkB>;index=1, <sip: userC@networkA;cause=302>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C) Apply post test routine Comments Check: A History-Info header is received in the INVITE contains the URI of user B (served user) at the interconnection interface. Check: Is the 'cause' parameter present in the Request line sent to user C (diverted-to user) set to '302'? Check: Is the cause parameter in the last entry is set to '302'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 100 Test case number SS_cfu_007 Test case group SIP-SIP/Service/CFU Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 25 AND SE 30 Test purpose Communication forwarding unconditional, full notification. The user A and user C are in network A. The user B is in network B and is provided with CFU Originating user receives notification that his communication has been diverted = Yes and "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = Yes, and "Served user allows the presentation of his/her URI to diverted-to user" = Yes. Ensure that when user A calls user B, the call is forwarded unconditional to user C, user A is notified of call diversion and informed of the diverted-to number and user C is informed of the forwarding number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = Yes • Served user allows the presentation of his/her URI to diverted-to user = Yes SIP Parameter INVITE: Request line contains ';cause=302' History-Info header: <sip:userB@networkB>;index=1, <sip: userC@networkA;cause=302>;index=1.1 181 Being Forwarded History-Info header: <sip:userB@networkB>;index=1, <sip: userC@networkA;cause=408>;index=1.1 200 OK INVITE History-Info header: <sip:userB@networkB>;index=1, <sip: userC@networkA;cause=486>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) 200 OK INVITE(Call-ID C-B) ACK(Call-ID C-B) 200 OK INVITE(Call-ID B-A) ACK(Call-ID A-B) Communication Apply post test routine Comments Check: A History-Info header is received in the INVITE at the interconnection interface sent to user C containing the URI identifying the served user. Check: A History-Info header is received in the 181 Being Forwarded at the interconnection interface sent to user A containing the URI identifying the diverted-to user. Check: Is the 'cause' parameter present in the Request line sent to user C (diverted-to user) set to '302'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 101 Test case number SS_cfu_008 Test case group SIP-SIP/Service/CFU Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 25 Test purpose Communication forwarding unconditional, unsuccessful UDUB. The user A and user C are in network A. The user B is in network B and is provided with CFU. Ensure that when user A calls user B, the call is forwarded unconditional to user C user C is user determined user busy. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C) 486 Busy Here(Call-ID C-B) ACK(Call-ID B-C) 486 Busy Here(Call-ID A-B) ACK(Call-ID A-B) Comments Check: The dialogue is terminated by receiving a 486 Busy Here. Repeat this test in reverse direction. Test case number SS_cfu_009 Test case group SIP-SIP/Service/CFU Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 25 Test purpose Communication forwarding unconditional, unsuccessful NDUB. The user A and user C are in network A. The user B is in network B. Ensure that when user A calls user B, the call is forwarded unconditional to user C and user C is network determined user busy. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C) 486 Busy Here(Call-ID C-B) ACK(Call-ID B-C) 486 Busy Here(Call-ID A-B) ACK(Call-ID A-B) Comments Check: The dialogue is terminated by receiving a 486 Busy Here. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 102 Test case number SS_cfu_010 Test case group SIP-SIP/Service/CFU Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 25 AND SE 30 AND [Network A] SE 9 Test purpose Communication forwarding unconditional, interaction with a not trusted network. The user A and user C are in network A. The user B is in network B and is provided with CFU Originating user receives notification that his communication has been diverted = Yes "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = No, "diverting number is released to the diverted-to user" = No. Ensure that when user A calls user B, the call is forwarded unconditional to user C, user A is notified of call diversion and not informed of the diverted-to number and user C is not informed of the forwarding number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = No • Served user allows the presentation of his/her URI to originating user in diversion notification = No • Served user allows the presentation of his/her URI to the diverted-to user = No SIP Parameter INVITE: no History-Info header 181 Being Forwarded no History-Info header Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) Apply post test routine Comments Check: No History-Info header is received in the INVITE at the interconnection interface. Check: No History-Info header is received in the 181 Being Forwarded at the interconnection interface (if sent). Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 103 Test case number SS_cfu_011 Test case group SIP-SIP/Service/CFU Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFU performed in Network B, Notification subscription options is set to presentation not allowed. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFU, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number. Ensure that when user A calls user B, the call is forwarded unconditional to user C, user A is not notified about call diversion. The notification information is present in the encapsulated ACM contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = no SIP Parameter 183 Session Progress Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM Backward call indicator Called party's status indicator no indication Redirection number Address signal (Diverted-to user) Call diversion information Notification subscription options presentation not allowed Redirecting reason unconditional Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C, IAM) 183 Session Progress (Call-ID B-A, ACM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 183 Session Progress received at the interconnection interface? Check: Is an ACM encapsulated in the 183? Check: Is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator set to 'presentation not allowed'? Check: Is the Redirecting reason set to 'unconditional'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 104 Test case number SS_cfu_012 Test case group SIP-SIP/Service/CFU Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFU performed in Network B, Notification subscription options is set to presentation allowed without redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFU, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number. Ensure that when user A calls user B, the call is forwarded unconditional to user C, user A is notified of call diversion and informed of the diverted-to number. The notification information is present in the encapsulated ACM contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number SIP Parameter 183 Session Progress Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM Backward call indicator Called party's status indicator no indication Redirection number Address signal (Diverted-to user) Call diversion information Notification subscription options presentation allowed without redirection number Redirecting reason unconditional Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C, IAM) 183 Session Progress (Call-ID B-A, ACM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: 183 Session Progress is received at the interconnection interface. Check: Is an ACM encapsulated in the 183? Check: Is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator set to 'presentation allowed without redirection number'? Check: Is the Redirecting reason set to 'unconditional'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 105 Test case number SS_cfu_013 Test case group SIP-SIP/Service/CFU Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFU performed in Network B, Notification subscription options is set to presentation allowed with redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFU, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, with diverted-to user number. Ensure that when user A calls user B, the call is forwarded unconditional to user C, user A is notified of call diversion and informed of the diverted-to number. The notification information is present in the encapsulated ACM contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, with diverted-to user number SIP Parameter 183 Session Progress Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM Backward call indicator Called party's status indicator no indication Redirection number Address signal (Diverted-to user) Call diversion information Notification subscription options presentation allowed with redirection number Redirecting reason unconditional Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C, IAM) 183 Session Progress (Call-ID B-A, ACM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: 183 Session Progress is received at the interconnection interface. Check: Is an ACM encapsulated in the 183? Check: Is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator set to 'presentation allowed with redirection number'? Check Is the Redirecting reason set to 'unconditional'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 106 Test case number SS_cfu_014 Test case group SIP-SIP/Service/CFU Reference 6.7/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 53 Test purpose SIP-I support. CFU performed in Network B, Restriction of the Redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFU, Diverted-to user is subscribed to the COLR service in Permanent mode. Ensure that when user A calls user B, the call is forwarded unconditional to user C, a Redirection number restriction parameter is present set to 'Presentation restricted' in the encapsulated ANM contained in the 200 OK INVITE if ISUP/BICC- SIP-I interworking is applicable in Network A. Configuration Subscription options: • Connected user subscribed to COLR, Permanent = yes SIP Parameter 200 OK Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Redirection number restriction Presentation restricted --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B), IAM CFU is performed INVITE(Call-ID B-C) 180 Ringing (Call-ID C-B, ACM) 180 Ringing (Call-ID B-A) 200 OK INVITE (Call-ID C-B, ANM) ACK (Call-ID B-C) 200 OK INVITE (Call-ID B-A) ACK (Call-ID A-B) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 200 OK INVITE received at the interconnection interface? Check: Is an ANM encapsulated in the 200 OK? Check: Is the ISUP/BICC Redirection number restriction set to 'Presentation restricted'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 107 Test case number SS_cfu_015 Test case group SIP-SIP/Service/CFU Reference 6.7/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 53 Test purpose SIP-I support. CFU performed in Network B, No restriction of the Redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFU, Diverted-to user is not subscribed to the COLR service. Ensure that when user A calls user B, the call is forwarded unconditional to user C, if a Redirection number restriction parameter is present it is set to 'Presentation allowed' in the encapsulated ANM contained in the 200 OK INVITE if ISUP/BICC- SIP-I interworking is applicable in Network A. Configuration Subscription options: • Connected user subscribed to COLR = no SIP Parameter 200 OK Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Redirection number restriction Presentation allowed or Redirection number restriction not present --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B), IAM CFU is performed INVITE(Call-ID B-C) 180 Ringing (Call-ID C-B, ACM) 180 Ringing (Call-ID B-A) 200 OK INVITE (Call-ID C-B, ANM) ACK (Call-ID B-C) 200 OK INVITE (Call-ID B-A) ACK (Call-ID A-B) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 200 OK INVITE received at the interconnection interface? Check: Is an ANM encapsulated in the 200 OK? Check: Is the ISUP/BICC Redirection number restriction present set to 'Presentation allowed' or is the parameter absent? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 108 Test case number SS_cfu_016 Test case group SIP-SIP/Service/CFU Reference 7.1/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFU performed in Network B, Notification of diverted-to user Redirecting number 'presentation allowed'. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFU, Served user releases his/her number to diverted-to user = Release diverting number information. Ensure that when user A calls user B, the call is forwarded unconditional to user C, user C is notified of call diversion and informed of the diverting number. The notification information is present in the encapsulated IAM contained in the Redirecting number 'presentation allowed' and Redirection information if ISUP/BICC - SIP-I interworking is applicable in Network B. Configuration Subscription options: • Served user releases his/her number to diverted-to user = Release diverting number information SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Redirecting number Address presentation restricted indicator presentation allowed Address signal (Diverting user) Original called number Address presentation restricted indicator presentation allowed Address signal Redirection information Original Redirection Reason unknown Redirecting indicator Redirection counter Redirecting reason unconditional --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C, IAM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a INVITE request received at the interconnection interface? Check: Is an IAM encapsulated in the INVITE? Check: Is the Redirecting number present and the Address presentation restricted indicator is set to 'presentation allowed'? Check: Is the Original called number present and the Address presentation restricted indicator is set to 'presentation allowed'? Check: Is the Redirection number present? Check: Is Redirection information present and the Redirecting reason is set to 'unconditional'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 109 Test case number SS_cfu_017 Test case group SIP-SIP/Service/CFU Reference 7.1/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFU performed in Network B, Notification of diverted-to user Redirecting number 'presentation restricted'. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFU, Served user releases his/her number to diverted-to user = Release diverting number information. Ensure that when user A calls user B, the call is forwarded unconditional to user C, user C is notified of call diversion and informed of the diverting number. The notification information is present in the encapsulated IAM contained in the Redirecting number 'presentation restricted' and Redirection information if ISUP/BICC - SIP-I interworking is applicable in Network B. Configuration Subscription options: • Served user releases his/her number to diverted-to user = Do not release diverting numberinformation SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Redirecting number Address presentation restricted indicator presentation restricted Address signal (Diverting user) Original called number Address presentation restricted indicator presentation restricted Address signal Redirection information Original Redirection Reason unknown Redirecting indicator Redirection counter Redirecting reason unconditional --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFU is performed INVITE(Call-ID B-C, IAM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a INVITE request received at the interconnection interface? Check: Is an IAM encapsulated in the INVITE? Check: Is the Redirecting number present and the Address presentation restricted indicator is set to 'presentation restricted'? Check: Is the Original called number present and the Address presentation restricted indicator is set to 'presentation restricted'? Check: Is the Redirection number present? Check: Is Redirection information present and the Redirecting reason is set to 'unconditional'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 110
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7f110eeff44cca5c59b5fc967e08482a
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101 585
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7.1.5.6.2 Communication Forwarding Busy (CFB)
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Test case number SS_cfb_001 Test case group SIP-SIP/Service/CFB Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 26 Test purpose Communication forwarding busy, basic rules. The user A and user C are in Network A. The user B is in network B and is provided with CFB. Ensure that when user A calls user B, the call is forwarded busy to user C. In the active call state, ensure the property of speech. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) 200 OK INVITE(Call-ID C-B) ACK(Call-ID B-C) 200 OK INVITE(Call-ID B-A) ACK(Call-ID A-B) Communication Apply post test routine Comments Check: CDIV busy is successful. Check: In the active call state, ensure the property of speech. Check: Is the P-Asserted-Identity present set to the identity of the originating user? Repeat this test in reverse direction. Test case number SS_cfb_002 Test case group SIP-SIP/Service/CFB Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 26 AND SE 30 Test purpose Communication forwarding busy, no notification. The user A and user C are in Network A. The user B is in network B and is provided with CFB, subscription option: "Originating user receives notification that his communication has been diverted" = No. Ensure that when user A calls user B, the call is forwarded busy to user C, originating user is not notified. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = No SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) Apply post test routine Comments Check: No notification regarding call forwarding in network B is received at the interconnection interface. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 111 Test case number SS_cfb_003 Test case group SIP-SIP/Service/CFB Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 26 AND SE 30 Test purpose Communication forwarding busy, originating user is notified. URI from the served user not received. The user A and user C are in network A. The user B is in network B and is provided with CFB "Originating user receives notification that his communication has been diverted" = Yes "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = No and. "Served user allows the presentation of his/her URI to originating user in diversion notification" = No. Ensure that when user A calls user B, the call is forwarded busy to user C, user A is notified of call diversion and not informed of the diverted-to number and served user number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = No • Served user allows the presentation of his/her URI to originating user in diversion notification = No SIP Parameter 181 Being Forwarded <sip:userB@networkB?Privacy=history&Reason=SIP;cause=486>;index=1, <sip: userC@networkA;cause=486?Privacy=history>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) Apply post test routine Comments Check: A 181 Being Forwarded and a History-Info header is received at the interconnection interface in both entries in the History-Info header a Privacy header is escaped value 'history'. Check: Is the cause parameter in the last entry set to '486'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 112 Test case number SS_cfb_004 Test case group SIP-SIP/Service/CFB Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 26 AND SE 30 Test purpose Communication forwarding busy, originating user is notified. URI from the diverted-to user received. The user A and user C are in network A. The user B is in network B and is provided with CFB "Originating user receives notification that his communication has been diverted" = Yes "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = Yes. Ensure that when user A calls user B, the call is forwarded busy to user C, user A is notified of call diversion and informed of the diverted-to number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = Yes SIP Parameter 181 Being Forwarded <sip:userB@networkB?Reason=SIP; cause=486>;index=1, <sip: userC@networkA;cause=486>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) Apply post test routine Comments Check: A 181 Being Forwarded is received at interconnection interface. Check: A History-Info header is contained in the 181 with the URI of the diverted-to user. Check: Is the cause parameter in the last entry set to '486'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 113 Test case number SS_cfb_005 Test case group SIP-SIP/Service/CFB Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 26 AND SE 30 Test purpose Communication forwarding busy, diverted-to user does not receive the URI of the served user. The user A and user C are in network C. The user B is in network B and is provided with CFB "Served user allows the presentation of his/her URI to the diverted-to user" = No. Ensure that when user A calls user B, the call is forwarded busy to user C, user C is not informed of the forwarding number. Configuration Subscription options: • Served user allows the presentation of his/her URI to the diverted-to user = No SIP Parameter INVITE: Request line contains ';cause=486' History-Info header: <sip:userB@networkB?Privacy=history&Reason=SIP;cause=486>;index=1, <sip: userC@networkA;cause=486>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C) Apply post test routine Comments Check: A History-Info header received in the INVITE contains the URI of user B (served user) at the interconnection interface and a Privacy header is escaped set to 'history'. Check: Is the 'cause' parameter present in the Request line sent to user C (diverted-to user) set to '486'? Check: Is the cause parameter in the last entry set to '486'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 114 Test case number SS_cfb_006 Test case group SIP-SIP/Service/CFB Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 26 AND SE 30 Test purpose Communication forwarding busy, diverted-to user receives the URI of the served user. The user A and user C are in network C. The user B is in network B and is provided with CFB "Served user allows the presentation of his/her URI to the diverted-to user" = Yes. Ensure that when user A calls user B, the call is forwarded busy to user C, user C is informed of the forwarding number. Configuration Subscription options: • Served user allows the presentation of his/her URI to the diverted-to user = Yes SIP Parameter INVITE: Request line contains ';cause=486' History-Info header: <sip:userB@networkB?Reason=SIP;cause=486>;index=1, <sip: userC@networkA;cause=486>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C) Apply post test routine Comments Check: A History-Info header received in the INVITE contains the URI of user B (served user) at the interconnection interface. Check: Is the 'cause' parameter present in the Request line sent to user C (diverted-to user) set to '486'? Check: Is the cause parameter in the last entry set to '486'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 115 Test case number SS_cfb_007 Test case group SIP-SIP/Service/CFB Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 26 AND SE 30 Test purpose Communication forwarding busy, full notification. The user A and user C are in network A. The user B is in network B and is provided with CFB "Originating user receives notification that his communication has been diverted" = Yes "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = Yes, "diverting number is released to the diverted-to user" = Yes. Ensure that when user A calls user B, the call is forwarded busy to user C, user A is notified of call diversion and informed of the diverted-to number and user C is informed of the forwarding number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = Yes, • diverting number is released to the diverted-to user = Yes SIP Parameter INVITE: Request line contains ';cause=486' History-Info header: <sip:userB@networkB&Reason=SIP;cause=486>;index=1, <sip: userC@networkA;cause=486>;index=1.1 181 Being Forwarded History-Info header: <sip:userB@networkB&Reason=SIP;cause=486>;index=1, <sip: userC@networkA;cause=486>;index=1.1 200 OK INVITE History-Info header: <sip:userB@networkB&Reason=SIP;cause=486>;index=1, <sip: userC@networkA;cause=486>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) 200 OK INVITE(Call-ID C-B) ACK(Call-ID C-B) 200 OK INVITE(Call-ID B-A) ACK(Call-ID A-B) Communication Apply post test routine Comments Check: A History-Info header is received in the INVITE at the interconnection interface sent to user C containing the URI identifying the served user. Check: A History-Info header is received in the 181 Being Forwarded at the interconnection interface sent to user A containing the URI identifying the diverted-to user. Check: Is the 'cause' parameter present in the Request line sent to user C (diverted-to user) set to '486'? Check: Is the cause parameter in the last entry set to '486'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 116 Test case number SS_cfb_008 Test case group SIP-SIP/Service/CFB Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 26 Test purpose Communication forwarding busy, unsuccessful UDUB. The user A and user C are in network A. The user B is in network B and is provided with CFB. Ensure that when user A calls user B, the call is forwarded busy to user C and user C is user determined user busy. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C) 486 Busy Here(Call-ID C-B) ACK(Call-ID B-C) 486 Busy Here(Call-ID A-B) ACK(Call-ID A-B) Comments Check: The dialogue is terminated by receiving a 486 Busy Here. Repeat this test in reverse direction. Test case number SS_cfb_009 Test case group SIP-SIP/Service/CFB Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 26 Test purpose Communication forwarding busy, unsuccessful NDUB. The user A and user C are in network A. The user B is in network B and is provided with CFB. Ensure that when user A calls user B, the call is forwarded busy to user C and user C is network determined user busy. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C) 486 Busy Here(Call-ID C-B) ACK(Call-ID B-C) 486 Busy Here(Call-ID A-B) ACK(Call-ID A-B) Comments Check: A 181 Being Forwarded is received at network 1 originating access. Check: The dialogue is terminated by receiving a 486 Busy Here. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 117 Test case number SS_cfb_010 Test case group SIP-SIP/Service/CFB Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 26 AND SE 30 AND [Network A] SE 9 Test purpose Communication forwarding busy, interaction with a not trusted network. The user A and user C are in network A. The user B is in network B and is provided with CFB "Originating user receives notification that his communication has been diverted" = Yes "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = No, "diverting number is released to the diverted-to user" = No. Ensure that when user A calls user B, the call is forwarded busy to user C, user A is notified of call diversion and not informed of the diverted-to number and user C is not informed of the forwarding number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = No • Served user allows the presentation of his/her URI to originating user in diversion notification = No • Served user allows the presentation of his/her URI to the diverted-to user = No SIP Parameter INVITE: no History-Info header 181 Being Forwarded no History-Info header Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) Apply post test routine Comments Check: No History-Info header is received in the INVITE at the interconnection interface. Check: No History-Info header is received in the 181 Being Forwarded at the interconnection interface (if sent). Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 118 Test case number SS_cfb_011 Test case group SIP-SIP/Service/CFB Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFB performed in Network B, Notification subscription options is set to presentation not allowed. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFB, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number. Ensure that when user A calls user B, the call is forwarded on busy user to user C, user A is not notified about call diversion. The notification information is present in the encapsulated ACM contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = no SIP Parameter 183 Session Progress Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM Backward call indicator Called party's status indicator no indication Redirection number Address signal (Diverted-to user) Call diversion information Notification subscription options presentation not allowed Redirecting reason User Busy Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C, IAM) 183 Session Progress (Call-ID B-A, ACM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 183 Session Progress received at the interconnection interface? Check: Is an ACM encapsulated in the 183? Check: Is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator set to 'presentation not allowed'? Check: Is the Redirecting reason set to User Busy'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 119 Test case number SS_cfb_012 Test case group SIP-SIP/Service/CFB Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFB performed in Network B, Notification subscription options is set to presentation allowed without redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFB, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number. Ensure that when user A calls user B, the call is forwarded on busy user to user C, user A is notified of call diversion and informed of the diverted-to number. The notification information is present in the encapsulated ACM contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number SIP Parameter 183 Session Progress Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM Backward call indicator Called party's status indicator no indication Redirection number Address signal (Diverted-to user) Call diversion information Notification subscription options presentation allowed without redirection number Redirecting reason User Busy Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C, IAM) 183 Session Progress (Call-ID B-A, ACM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: 183 Session Progress is received at the interconnection interface. Check: Is an ACM encapsulated in the 183? Check: Is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator is set to 'presentation allowed without redirection number'? Check: Is the Redirecting reason set to 'User Busy'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 120 Test case number SS_cfb_013 Test case group SIP-SIP/Service/CFB Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFB performed in Network B, Notification subscription options is set to presentation allowed with redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFB, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, with diverted-to user number. Ensure that when user A calls user B, the call is forwarded on busy user to user C, user A is notified of call diversion and informed of the diverted-to number. The notification information is present in the encapsulated ACM contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, with diverted-to user number SIP Parameter 183 Session Progress Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM Backward call indicator Called party's status indicator no indication Redirection number (Diverted-to user) Address signal Call diversion information Notification subscription options presentation allowed with redirection number Redirecting reason User Busy Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C, IAM) 183 Session Progress (Call-ID B-A, ACM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: 183 Session Progress is received at the interconnection interface. Check: Is an ACM encapsulated in the 183? Check: Is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator is set to 'presentation allowed with redirection number'? Check: Is the Redirecting reason set to 'User Busy'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 121 Test case number SS_cfb_014 Test case group SIP-SIP/Service/CFB Reference 6.7/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 53 Test purpose SIP-I support. CFB performed in Network B, Restriction of the Redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFB, Diverted-to user is subscribed to the COLR service in Permanent mode. Ensure that when user A calls user B, the call is forwarded on busy user to user C, a Redirection number restriction parameter is present set to 'Presentation restricted' in the encapsulated ANM contained in the 200 OK INVITE if ISUP/BICC- SIP-I interworking is applicable in Network A. Configuration Subscription options: • Connected user subscribed to COLR, Permanent = yes SIP Parameter 200 OK Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Redirection number restriction Presentation restricted --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B), IAM CFB is performed INVITE(Call-ID B-C) 180 Ringing (Call-ID C-B, ACM) 180 Ringing (Call-ID B-A) 200 OK INVITE (Call-ID C-B, ANM) ACK (Call-ID B-C) 200 OK INVITE (Call-ID B-A) ACK (Call-ID A-B) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 200 OK INVITE received at the interconnection interface? Check: Is an ANM encapsulated in the 200 OK? Check: Is the ISUP/BICC Redirection number restriction set to 'Presentation restricted'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 122 Test case number SS_cfb_015 Test case group SIP-SIP/Service/CFB Reference 6.7/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 53 Test purpose SIP-I support. CFB performed in Network B, No restriction of the Redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFB, Diverted-to user is not subscribed to the COLR service. Ensure that when user A calls user B, the call is forwarded on busy user to user C, if a Redirection number restriction parameter is present it is set to 'Presentation allowed' in the encapsulated ANM contained in the 200 OK INVITE if ISUP/BICC- SIP-I interworking is applicable in Network A. Configuration Subscription options: • Connected user subscribed to COLR = no SIP Parameter 200 OK Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Redirection number restriction Presentation allowed or Redirection number restriction not present --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B), IAM CFB is performed INVITE(Call-ID B-C) 180 Ringing (Call-ID C-B, ACM) 180 Ringing (Call-ID B-A) 200 OK INVITE (Call-ID C-B, ANM) ACK (Call-ID B-C) 200 OK INVITE (Call-ID B-A) ACK (Call-ID A-B) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 200 OK INVITE received at the interconnection interface? Check: Is an ANM encapsulated in the 200 OK? Check: Is the ISUP/BICC Redirection number restriction present set to 'Presentation allowed' or is the parameter absent? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 123 Test case number SS_cfb_016 Test case group SIP-SIP/Service/CFB Reference 7.1/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFB performed in Network B, Notification of diverted-to user Redirecting number 'presentation allowed'. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFB, Served user releases his/her number to diverted-to user = Release diverting number information. Ensure that when user A calls user B, the call is forwarded on busy user to user C, user C is notified of call diversion and informed of the diverting number. The notification information is present in the encapsulated IAM contained in the Redirecting number 'presentation allowed' and Redirection information if ISUP/BICC - SIP-I interworking is applicable in Network B. Configuration Subscription options: • Served user releases his/her number to diverted-to user = Release diverting number information SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Redirecting number Address presentation restricted indicator presentation allowed Address signal (Diverting user) Original called number Address presentation restricted indicator presentation allowed Address signal Redirection information Original Redirection Reason unknown Redirecting indicator Redirection counter Redirecting reason User Busy --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C, IAM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a INVITE request received at the interconnection interface? Check: Is an IAM encapsulated in the INVITE? Check: Is the Redirecting number present and the Address presentation restricted indicator is set to 'presentation allowed'? Check: Is the Original called number present and the Address presentation restricted indicator is set to 'presentation allowed'? Check: Is the Redirection number present? Check: Is Redirection information present and the Redirecting reason is set to 'User Busy'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 124 Test case number SS_cfb_017 Test case group SIP-SIP/Service/CFB Reference 7.1/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFB performed in Network B, Notification of diverted-to user Redirecting number 'presentation restricted'. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFB, Served user releases his/her number to diverted-to user = Release diverting number information. Ensure that when user A calls user B, the call is forwarded on busy user to user C, user C is notified of call diversion and informed of the diverting number. The notification information is present in the encapsulated IAM contained in the Redirecting number 'presentation restricted' and Redirection information if ISUP/BICC - SIP-I interworking is applicable in Network B. Configuration Subscription options: • Served user releases his/her number to diverted-to user = Do not release diverting numberinformation SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Redirecting number Address presentation restricted indicator presentation restricted Address signal (Diverting user) Original called number Address presentation restricted indicator presentation restricted Address signal Redirection information Original Redirection Reason unknown Redirecting indicator Redirection counter Redirecting reason User Busy --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFB is performed INVITE(Call-ID B-C, IAM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a INVITE request received at the interconnection interface? Check: Is an IAM encapsulated in the INVITE? Check: Is the Redirecting number present and the Address presentation restricted indicator is set to 'presentation restricted'? Check: Is the Original called number present and the Address presentation restricted indicator is set to 'presentation restricted'? Check: Is the Redirection number present? Check: Is Redirection information present and the Redirecting reason is set to 'User Busy'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 125
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7f110eeff44cca5c59b5fc967e08482a
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101 585
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7.1.5.6.3 Communication Forwarding No Reply (CFNR)
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Test case number SS_cfnr_001 Test case group SIP-SIP/Service/CFNR Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 27 Test purpose Communication forwarding no reply, basic rules. The user A and user C are in Network A. The user B is in network B and is provided with CFNR. Ensure that when user A calls user B, the call is forwarded no reply to user C. In the active call state, ensure the property of speech. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing(Call-ID B-A) CFB is performed INVITE(Call-ID B-C) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) 200 OK INVITE(Call-ID C-B) ACK(Call-ID B-C) 200 OK INVITE(Call-ID B-A) ACK(Call-ID A-B) Communication Apply post test routine Comments Check: CDIV no reply is successful. Check: In the active call state, ensure the property of speech. Check: Is the P-Asserted-Identity present set to the identity of the originating user? Repeat this test in reverse direction. Test case number SS_cfnr_002 Test case group SIP-SIP/Service/CFNR Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 27 AND SE 30 Test purpose Communication forwarding no reply, no notification. The user A and user C are in Network A. The user B is in network B and is provided with CFNR, subscription option: "Originating user receives notification that his communication has been diverted" = No. Ensure that when user A calls user B, the call is forwarded no reply to user C, originating user is not notified. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = No SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing(Call-ID B-A) CFB is performed INVITE(Call-ID B-C) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) Apply post test routine Comments Check: No notification regarding call forwarding in network B is received at the interconnection interface. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 126 Test case number SS_cfnr_003 Test case group SIP-SIP/Service/CFNR Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 27 AND SE 30 Test purpose Communication forwarding no reply, originating user is notified. URI from the served user not received. The user A and user C are in network A. The user B is in network B and is provided with CFNR "Originating user receives notification that his communication has been diverted" = Yes "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = No and. "Served user allows the presentation of his/her URI to originating user in diversion notification" = No. Ensure that when user A calls user B, the call is forwarded no reply to user C, user A is notified of call diversion and not informed of the diverted-to number and served user number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = No • Served user allows the presentation of his/her URI to originating user in diversion notification = No SIP Parameter 181 Being Forwarded <sip:userB@networkB?Privacy=history>;index=1, <sip: userC@networkA;cause=408?Privacy=history>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing(Call-ID B-A) CFB is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) Apply post test routine Comments Check: A 181 Being Forwarded and a History-Info header is received at the interconnection interface in both entries in the History-Info header a Privacy header is escaped value 'history'. Check: Is the cause parameter in the last entry set to '408'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 127 Test case number SS_cfnr_004 Test case group SIP-SIP/Service/CFNR Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 27 AND SE 30 Test purpose Communication forwarding no reply, originating user is notified. URI from the diverted-to user received. The user A and user C are in network A. The user B is in network B and is provided with CFNR "Originating user receives notification that his communication has been diverted" = Yes and "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = Yes. Ensure that when user A calls user B, the call is forwarded no reply to user C, user A is notified of call diversion and informed of the diverted-to number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification =Yes SIP Parameter 181 Being Forwarded <sip:userB@networkB>;index=1, <sip: userC@networkA;cause=408>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing(Call-ID B-A) CFB is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) Apply post test routine Comments Check: A 181 Being Forwarded is received at the interconnection interface. Check: A History-Info header is contained in the 181 with the URI of the diverted-to user. Check: Is the cause parameter in the last entry is set to '408'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 128 Test case number SS_cfnr_005 Test case group SIP-SIP/Service/CFNR Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 27 AND SE 30 Test purpose Communication forwarding no reply, diverted-to user does not receive the URI of the served user. The user A and user C are in network A. The user B is in network B and is provided with "Served user allows the presentation of his/her URI to the diverted- to user" = No. Ensure that when user A calls user B, the call is forwarded no reply to user C, user C is not informed of the forwarding number. Configuration Subscription options: • Served user allows the presentation of his/her URI to the diverted-to user = No SIP Parameter INVITE Request line contains ';cause=408' History-Info header: <sip:userB@networkB?Privacy=history>;index=1, <sip: userC@network1;cause=408>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing(Call-ID B-A) CFB is performed INVITE(Call-ID B-C) Apply post test routine Comments Check: A History-Info header received in the INVITE contains the URI of user B (served user) at the interconnection interface and a Privacy header is escaped set to 'history'. Check: Is the 'cause' parameter present in the Request line sent to user C (diverted-to user) set to '408'? Check: Is the cause parameter in the last entry is set to '408'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. Test case number SS_cfnr_006 Test case group SIP-SIP/Service/CFNR Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 27 AND SE 30 Test purpose Communication forwarding no reply, diverted-to user receives the URI of the diverted-to user. The user A and user C are in network A. The user B is in network B and is provided with "Served user allows the presentation of his/her URI to the diverted- to user" = Yes. Ensure that when user A calls user B, the call is forwarded no reply to user C, user C is informed of the forwarding number. Configuration Subscription options: • Served user allows the presentation of his/her URI to the diverted-to user = Yes SIP Parameter INVITE Request line contains ';cause=408' History-Info header: <sip:userB@networkB>;index=1, <sip: userC@network1;cause=408>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing(Call-ID B-A) CFB is performed INVITE(Call-ID B-C) Apply post test routine Comments ETSI ETSI TS 101 585 V1.2.1 (2014-04) 129 Test case number SS_cfnr_007 Test case group SIP-SIP/Service/CFNR Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 27 AND SE 30 Test purpose Communication forwarding no reply, full notification. The user A and user C are in network A. The user B is in network B and is provided with CFNR "Originating user receives notification that his communication has been diverted" = Yes, "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = Yes, "diverting number is released to the diverted-to user" = Yes. Ensure that when user A calls user B, the call is forwarded no reply to user C, user A is notified of call diversion and informed of the diverted-to number and user C is informed of the forwarding number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = Yes • diverting number is released to the diverted-to user = Yes SIP Parameter INVITE: Request line contains ';cause=408' History-Info header: <sip:userB@networkB&Reason=SIP;cause=408>;index=1, <sip: userC@networkA;cause=486>;index=1.1 181 Being Forwarded History-Info header: <sip:userB@network>;index=1, <sip: userC@networkA;cause=408>;index=1.1 200 OK INVITE History-Info header: <sip:userB@networkB>;index=1, <sip: userC@networkA;cause=408>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing(Call-ID B-A) CFB is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) 200 OK INVITE(Call-ID C-B) ACK(Call-ID C-B) 200 OK INVITE(Call-ID B-A) ACK(Call-ID A-B) Apply post test routine Comments Check: A History-Info header is received in the INVITE at the interconnection interface sent to user C containing the URI identifying the served user. Check: A History-Info header is received in the 181 Being Forwarded at the interconnection interface sent to user A containing the URI identifying the diverted-to user. Check: Is the 'cause' parameter present in the Request line sent to user C (diverted-to user) set to '408'? Check: Is the cause parameter in the last entry is set to '408'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 130 Test case number SS_cfnr_008 Test case group SIP-SIP/Service/CFNR Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 27 Test purpose Communication forwarding no reply, unsuccessful UDUB. The user A and user C are in network A. The user B is in network B and is provided with CFNR. Ensure that when user A calls user B, the call is forwarded no reply to user C and user C is user determined user busy. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing(Call-ID B-A) CFB is performed INVITE(Call-ID B-C) 486 Busy Here(Call-ID C-B) ACK(Call-ID B-C) 486 Busy Here(Call-ID A-B) ACK(Call-ID A-B) Comments Check: The dialogue is terminated by receiving a 486 Busy Here. Repeat this test in reverse direction. Test case number SS_cfnr_009 Test case group SIP-SIP/Service/CFNR Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 27 Test purpose Communication forwarding no reply, unsuccessful NDUB. The user A and user C are in network A. The user B is in network B and is provided with CFNR. Ensure that when user A calls user B, the call is forwarded no reply to user C and user C is network determined user busy. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing(Call-ID B-A) CFB is performed INVITE(Call-ID B-C) 486 Busy Here(Call-ID C-B) ACK(Call-ID B-C) 486 Busy Here(Call-ID A-B) ACK(Call-ID A-B) Comments Check: The dialogue is terminated by receiving a 486 Busy Here. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 131 Test case number SS_cfnr_010 Test case group SIP-SIP/Service/CFNR Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 27 AND SE 30 AND [Network A] is SE 9 Test purpose Communication forwarding no reply, interaction with a not trusted network. The user A and user C are in network A. The user B is in network B and is provided with CFNR "Originating user receives notification that his communication has been diverted" = Yes "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = No, "diverting number is released to the diverted-to user" = No. Ensure that when user A calls user B, the call is forwarded no reply to user C, user A is notified of call diversion and not informed of the diverted-to number and user C is not informed of the forwarding number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = No • Served user allows the presentation of his/her URI to originating user in diversion notification = No • Served user allows the presentation of his/her URI to the diverted-to user = No SIP Parameter INVITE: no History-Info header 181 Being Forwarded no History-Info header Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing(Call-ID B-A) CFB is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) Apply post test routine Comments Check: No History-Info header is received in the INVITE at the interconnection interface. Check: No History-Info header is received in the 181 Being Forwarded at the interconnection interface (if sent). Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 132 Test case number SS_cfnr_011 Test case group SIP-SIP/Service/CFNR Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFNR performed in Network B, Notification subscription options is set to presentation not allowed. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFNR, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number. Ensure that when user A calls user B, the call is forwarded on no reply to user C, user A is not notified about call diversion. The notification information is present in the encapsulated CPG contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = no SIP Parameter 183 Session Progress Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Event indicator Alerting or Progress Redirection number Address signal (Diverted-to user) Call diversion information Notification subscription options presentation not allowed Redirecting reason No reply Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing (Call-ID B-A, ACM) CFNR is performed INVITE(Call-ID B-C, IAM) 183 Session Progress (Call-ID B-A, CPG) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 183 Session Progress received at the interconnection interface? Check: Is an CPG encapsulated in the 183? Check: Is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator set to 'presentation not allowed'? Check: Is the Redirecting reason set to 'No reply'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 133 Test case number SS_cfnr_012 Test case group SIP-SIP/Service/CFNR Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFNR performed in Network B, Notification subscription options is set to presentation allowed without redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFNR, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number. Ensure that when user A calls user B, the call is forwarded on no reply to user C, user A is notified of call diversion and informed of the diverted-to number. The notification information is present in the encapsulated CPG contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number SIP Parameter 183 Session Progress Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Event indicator Alerting or Progress Redirection number Address signal (Diverted-to user) Call diversion information Notification subscription options presentation allowed without redirection number Redirecting reason No reply Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing (Call-ID B-A, ACM) CFNR is performed INVITE(Call-ID B-C, IAM) 183 Session Progress (Call-ID B-A, CPG) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: 183 Session Progress is received at the interconnection interface. Check: Is an CPG encapsulated in the 183? Check: is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator is set to 'presentation allowed without redirection number'? Check: Is the Redirecting reason set to 'No reply'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 134 Test case number SS_cfnr_013 Test case group SIP-SIP/Service/CFNR Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFNR performed in Network B, Notification subscription options is set to presentation allowed with redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFNR, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, with diverted-to user number. Ensure that when user A calls user B, the call is forwarded on no reply to user C, user A is notified of call diversion and informed of the diverted-to number. The notification information is present in the encapsulated CPG contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, with diverted-to user number SIP Parameter 183 Session Progress Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Event indicator Alerting or Progress Redirection number Address signal (Diverted-to user) Call diversion information Notification subscription options presentation allowed with redirection number Redirecting reason No reply Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing (Call-ID B-A, ACM) CFNR is performed INVITE(Call-ID B-C, IAM) 183 Session Progress (Call-ID B-A, CPG) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: 183 Session Progress is received at the interconnection interface. Check: Is an CPG encapsulated in the 183? Check: Is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator is set to 'presentation allowed with redirection number'? Check: Is the Redirecting reason set to 'No reply'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 135 Test case number SS_cfnr_014 Test case group SIP-SIP/Service/CFNR Reference 6.7/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 53 Test purpose SIP-I support. CFNR performed in Network B, Restriction of the Redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFNR, Diverted-to user is subscribed to the COLR service in Permanent mode. Ensure that when user A calls user B, the call is forwarded on no reply to user C, a Redirection number restriction parameter is present set to 'Presentation restricted' in the encapsulated ANM contained in the 200 OK INVITE if ISUP/BICC- SIP-I interworking is applicable in Network A. Configuration Subscription options: • Connected user subscribed to COLR, Permanent = yes SIP Parameter 200 OK Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Redirection number restriction Presentation restricted --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B), IAM 180 Ringing (Call-ID B-A, ACM) CFNR is performed INVITE(Call-ID B-C) 180 Ringing (Call-ID C-B, ACM) 180 Ringing (Call-ID B-A) 200 OK INVITE (Call-ID C-B, ANM) ACK (Call-ID B-C) 200 OK INVITE (Call-ID B-A) ACK (Call-ID A-B) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 200 OK INVITE received at the interconnection interface? Check: Is an ANM encapsulated in the 200 OK? Check: Is the ISUP/BICC Redirection number restriction set to 'Presentation restricted'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 136 Test case number SS_cfnr_015 Test case group SIP-SIP/Service/CFNR Reference 6.7/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 53 Test purpose SIP-I support. CFNR performed in Network B, No restriction of the Redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFNR, Diverted-to user is not subscribed to the COLR service. Ensure that when user A calls user B, the call is forwarded on no reply to user C, if a Redirection number restriction parameter is present it is set to 'Presentation allowed' in the encapsulated ANM contained in the 200 OK INVITE if ISUP/BICC- SIP-I interworking is applicable in Network A. Configuration Subscription options: • Connected user subscribed to COLR = no SIP Parameter 200 OK Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Redirection number restriction Presentation allowed or Redirection number restriction not present --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B), IAM 180 Ringing (Call-ID B-A) CFNR is performed INVITE(Call-ID B-C) 180 Ringing (Call-ID C-B, ACM) 180 Ringing (Call-ID B-A) 200 OK INVITE (Call-ID C-B, ANM) ACK (Call-ID B-C) 200 OK INVITE (Call-ID B-A) ACK (Call-ID A-B) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 200 OK INVITE received at the interconnection interface? Check: Is an ANM encapsulated in the 200 OK? Check: Is the ISUP/BICC Redirection number restriction present set to 'Presentation allowed' or is the parameter absent? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 137 Test case number SS_cfnr_016 Test case group SIP-SIP/Service/CFNR Reference 7.1/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFNR performed in Network B, Notification of diverted-to user Redirecting number 'presentation allowed'. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFNR, Served user releases his/her number to diverted-to user = Release diverting number information. Ensure that when user A calls user B, the call is forwarded on no reply to user C, user C is notified of call diversion and informed of the diverting number. The notification information is present in the encapsulated IAM contained in the Redirecting number 'presentation allowed' and Redirection information if ISUP/BICC - SIP-I interworking is applicable in Network B. Configuration Subscription options: • Served user releases his/her number to diverted-to user = Release diverting number information SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Redirecting number Address presentation restricted indicator presentation allowed Address signal (Diverting user) Original called number Address presentation restricted indicator presentation allowed Address signal Redirection information Original Redirection Reason unknown Redirecting indicator Redirection counter Redirecting reason No reply --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing (Call-ID B-A, ACM) CFNR is performed INVITE(Call-ID B-C, IAM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a INVITE request received at the interconnection interface? Check: Is an IAM encapsulated in the INVITE? Check: Is the Redirecting number present and the Address presentation restricted indicator is set to 'presentation allowed'? Check: Is the Original called number present and the Address presentation restricted indicator is set to 'presentation allowed'? Check: Is the Redirection number present? Check: Is Redirection information present and the Redirecting reason is set to 'No reply'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 138 Test case number SS_cfnr_017 Test case group SIP-SIP/Service/CFNR Reference 7.1/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFNR performed in Network B, Notification of diverted-to user Redirecting number 'presentation restricted'. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFNR, Served user releases his/her number to diverted-to user = Release diverting number information. Ensure that when user A calls user B, the call is forwarded on no reply to user C, user C is notified of call diversion and informed of the diverting number. The notification information is present in the encapsulated IAM contained in the Redirecting number 'presentation restricted' and Redirection information if ISUP/BICC - SIP-I interworking is applicable in Network B. Configuration Subscription options: • Served user releases his/her number to diverted-to user = Do not release diverting numberinformation SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Redirecting number Address presentation restricted indicator presentation restricted Address signal (Diverting user) Original called number Address presentation restricted indicator presentation restricted Address signal Redirection information Original Redirection Reason unknown Redirecting indicator Redirection counter Redirecting reason No reply --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing (Call-ID B-A, ACM) CFNR is performed INVITE(Call-ID B-C, IAM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a INVITE request received at the interconnection interface? Check: Is an IAM encapsulated in the INVITE? Check: Is the Redirecting number present and the Address presentation restricted indicator is set to 'presentation restricted'? Check: Is the Original called number present and the Address presentation restricted indicator is set to 'presentation restricted'? Check: Is the Redirection number present? Check: Is Redirection information present and the Redirecting reason is set to 'No reply'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 139
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7f110eeff44cca5c59b5fc967e08482a
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101 585
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7.1.5.6.4 Communication Forwarding Not Logged in (CFNL)
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Test case number SS_cfnl_001 Test case group SIP-SIP/Service/CFNL Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 28 Test purpose Communication forwarding not logged in, basic rules. The user A and user C are in Network A. The user B is in network B and is provided with CFNL. Ensure that when user A calls user B, the call is forwarded not logged in to user C. In the active call state, ensure the property of speech. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed INVITE(Call-ID B-C) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) 200 OK INVITE(Call-ID C-B) ACK(Call-ID B-C) 200 OK INVITE(Call-ID B-A) ACK(Call-ID A-B) Communication Apply post test routine Comments Check: The CDIV not logged in is successful. Check: In the active call state, ensure the property of speech. Check: Is the P-Asserted-Identity present set to the identity of the originating user? Repeat this test in reverse direction. Test case number SS_cfnl_002 Test case group SIP-SIP/Service/CFNL Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 28 AND SE 30 Test purpose Communication forwarding not logged in, no notification. The user A and user C are in Network A. The user B is in network B and is provided with CFNL, subscription option: "Originating user receives notification that his communication has been diverted" = No. Ensure that when user A calls user B, the call is forwarded not logged in to user C, originating user is not notified. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = No SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed INVITE(Call-ID B-C) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) Apply post test routine Comments Check: No notification regarding call forwarding in network B is received at interconnection interface. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 140 Test case number SS_cfnl_003 Test case group SIP-SIP/Service/CFNL Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 28 AND SE 30 Test purpose Communication forwarding not logged in, originating user is notified. URI of the diverted-to user not received. The user A and user C are in network A. The user B is in network B and is provided with CFNL "Originating user receives notification that his communication has been diverted" = Yes and "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = No and. "Served user allows the presentation of his/her URI to originating user in diversion notification" = No. Ensure that when user A calls user B, the call is forwarded not logged in to user C, user A is notified of call diversion and not informed of the diverted-to number and the served user number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = No • Served user allows the presentation of his/her URI to originating user in diversion notification = No SIP Parameter 181 Being Forwarded <sip:userB@networkB?Privacy=history>;index=1, <sip: userC@networkA;cause=404?Privacy=history>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) Apply post test routine Comments Check: A 181 Being Forwarded and a History-Info header is received at the interconnection interface in both entries in the History-Info header a Privacy header is escaped value 'history'. Check: Is the cause parameter in the last entry is set to '404'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 141 Test case number SS_cfnl_004 Test case group SIP-SIP/Service/CFNL Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 28 AND SE 30 Test purpose Communication forwarding not logged in, originating user is notified. URI from the diverted-to user received. The user A and user C are in network A. The user B is in network B and is provided with CFNL "Originating user receives notification that his communication has been diverted" = Yes and "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = Yes. Ensure that when user A calls user B, the call is forwarded not logged in to user C, user A is notified of call diversion and informed of the diverted-to number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = Yes SIP Parameter 181 Being Forwarded <sip:userB@networkB>;index=1, <sip: userC@networkA;cause=404>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) Apply post test routine Comments Check: A 181 Being Forwarded is received at interconnection interface. Check: A History-Info header is contained in the 181 with the URI of the served user and the URI of the diverted-to user. Check: Is the cause parameter in the last entry is set to '404'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 142 Test case number SS_cfnl_005 Test case group SIP-SIP/Service/CFNL Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 28 AND SE 30 Test purpose Communication forwarding not logged in, diverted-to user does not receive the URI of the diverted-to user. The user A and user C are in network A. The user B is in network B and is provided with CFNL "Served user allows the presentation of his/her URI to diverted-to user" = No. Ensure that when user A calls user B, the call is forwarded not logged in to user C, user C is not informed of the forwarding number. Configuration Subscription options: • Served user allows the presentation of his/her URI to diverted-to user = No SIP Parameter INVITE Request line contains ';cause=404' History-Info header: <sip:userB@networkB?Privacy=history>;index=1, <sip: userC@network1;cause=404>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed INVITE(Call-ID B-C) Apply post test routine Comments Check: A History-Info header is received in the INVITE contains the URI of user B (served user) at the interconnection interface and a Privacy header is escaped set to 'history'. Check: Is the 'cause' parameter present in the Request line sent to user C (diverted-to user) set to '404'? Check: Is the cause parameter in the last entry is set to '404'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 143 Test case number SS_cfnl_006 Test case group SIP-SIP/Service/CFNL Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 28 AND SE 30 Test purpose Communication forwarding not logged in, diverted-to user receives the URI of the served user. The user A and user C are in network A. The user B is in network B and is provided with CFNL "Served user allows the presentation of his/her URI to diverted-to user" = Yes. Ensure that when user A calls user B, the call is forwarded not logged in to user C, user C is informed of the forwarding number. Configuration Subscription options: • Served user allows the presentation of his/her URI to diverted-to user = Yes SIP Parameter INVITE Request line contains ';cause=404' History-Info header: <sip:userB@networkB>;index=1, <sip: userC@networkA;cause=404>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed INVITE(Call-ID B-C) Apply post test routine Comments Check: A History-Info header is received in the INVITE contains the URI of user B (served user) at the interconnection interface. Check: Is the 'cause' parameter present in the Request line sent to user C (diverted-to user) set to '404'? Check: Is the cause parameter in the last entry is set to '404'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 144 Test case number SS_cfnl_007 Test case group SIP-SIP/Service/CFNL Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 28 AND SE 30 Test purpose Communication forwarding not logged in, full notification. The user A and user C are in network A. The user B is in network B and is provided with CFNL "Originating user receives notification that his communication has been diverted" = Yes, "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = Yes, "diverting number is released to the diverted-to user" = Yes. Ensure that when user A calls user B, the call is forwarded not logged in to user C, user A is notified of call diversion and informed of the diverted-to number and user C is informed of the forwarding number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = Yes • diverting number is released to the diverted-to user = Yes SIP Parameter INVITE: Request line contains ';cause=404' History-Info header: <sip:userB@networkB&Reason=SIP;cause=404>;index=1, <sip: userC@networkA;cause=404>;index=1.1 181 Being Forwarded History-Info header: <sip:userB@network>;index=1, <sip: userC@networkA;cause=404>;index=1.1 200 OK INVITE History-Info header: <sip:userB@networkB>;index=1, <sip: userC@networkA;cause=404>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) 200 OK INVITE(Call-ID C-B) ACK(Call-ID C-B) 200 OK INVITE(Call-ID B-A) ACK(Call-ID A-B) Apply post test routine Comments Check: A History-Info header is received in the INVITE at the interconnection interface sent to user C containing the URI identifying the served user. Check: A History-Info header is received in the 181 Being Forwarded at the interconnection interface sent to user A containing the URI identifying the diverted-to user. Check: Is the 'cause' parameter present in the Request line sent to user C (diverted-to user) set to '404'? Check: Is the cause parameter in the last entry is set to '404'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 145 Test case number SS_cfnl_008 Test case group SIP-SIP/Service/CFNL Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 28 Test purpose Communication forwarding not logged in, unsuccessful UDUB. The user A and user C are in network A. The user B is in network B and is provided with CFNL. Ensure that when user A calls user B, the call is forwarded not logged in to user C and user C is user determined user busy. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed 486 Busy Here(Call-ID C-B) ACK(Call-ID B-C) 486 Busy Here(Call-ID A-B) ACK(Call-ID A-B) Comments Check: The dialogue is terminated by receiving a 486 Busy Here. Repeat this test in reverse direction. Test case number SS_cfnl_009 Test case group 4.5.2.6/ [9] Reference ES 183 004 SELECTION EXPRESSION SE 28 Test purpose Communication forwarding not logged in, unsuccessful NDUB. The user A and user C are in network A. The user B is in network B and is provided with CFNL. Ensure that when user A calls user B, the call is forwarded not logged in to user C and user C is busy. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed 486 Busy Here(Call-ID C-B) ACK(Call-ID B-C) 486 Busy Here(Call-ID A-B) ACK(Call-ID A-B) Comments Check: The dialogue is terminated by receiving a 486 Busy Here. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 146 Test case number SS_cfnl_010 Test case group SIP-SIP/Service/CFNL Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 28 AND SE 30 AND [Network A] SE 9 Test purpose Communication forwarding not logged in, interaction with a not trusted network. The user A and user C are in network A. The user B is in network B and is provided with CFNL "Originating user receives notification that his communication has been diverted" = Yes "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = No, "diverting number is released to the diverted-to user" = No. Ensure that when user A calls user B, the call is forwarded not logged in to user C, user A is notified of call diversion and not informed of the diverted-to number and user C is not informed of the forwarding number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = No • Served user allows the presentation of his/her URI to originating user in diversion notification = No • Served user allows the presentation of his/her URI to the diverted-to user = No SIP Parameter INVITE: no History-Info header 181 Being Forwarded no History-Info header Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) Apply post test routine Comments Check: No History-Info header is received in the INVITE at the interconnection interface. Check: No History-Info header is received in the 181 Being Forwarded at the interconnection interface (if sent). Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 147 Test case number SS_cfnl_011 Test case group SIP-SIP/Service/CFNL Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFNL performed in Network B, Notification subscription options is set to presentation not allowed. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFNL, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number. Ensure that when user A calls user B, the call is forwarded on Mobile subscriber not reachable to user C, user A is not notified about call diversion. The notification information is present in the encapsulated ACM contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = no SIP Parameter 183 Session Progress Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM Backward call indicator Called party's status indicator no indication Redirection number Address signal (Diverted-to user) Call diversion information Notification subscription options presentation not allowed Redirecting reason Mobile subscriber not reachable Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed INVITE(Call-ID B-C, IAM) 183 Session Progress (Call-ID B-A, ACM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 183 Session Progress received at the interconnection interface? Check: Is an ACM encapsulated in the 183? Check: Is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator set to 'presentation not allowed'? Check: Is the Redirecting reason set to 'Mobile subscriber not reachable'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 148 Test case number SS_cfnl_012 Test case group SIP-SIP/Service/CFNL Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFNL performed in Network B, Notification subscription options is set to presentation allowed without redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFNL, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number. Ensure that when user A calls user B, the call is forwarded on Mobile subscriber not reachable to user C, user A is notified of call diversion and informed of the diverted-to number. The notification information is present in the encapsulated ACM contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number SIP Parameter 183 Session Progress Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM Backward call indicator Called party's status indicator no indication Redirection number Address signal (Diverted-to user) Call diversion information Notification subscription options presentation allowed without redirection number Redirecting reason Mobile subscriber not reachable Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed INVITE(Call-ID B-C, IAM) 183 Session Progress (Call-ID B-A, ACM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: 183 Session Progress is received at the interconnection interface. Check: Is an ACM encapsulated in the 183? Check: Is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator is set to 'presentation allowed without redirection number'? Check: Is the Redirecting reason set to 'Mobile subscriber not reachable'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 149 Test case number SS_cfnl_013 Test case group SIP-SIP/Service/CFNL Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFNL performed in Network B, Notification subscription options is set to presentation allowed with redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFNL, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, with diverted-to user number. Ensure that when user A calls user B, the call is forwarded on Mobile subscriber not reachable to user C, user A is notified of call diversion and informed of the diverted-to number. The notification information is present in the encapsulated ACM contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, with diverted-to user number SIP Parameter 183 Session Progress Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM Backward call indicator Called party's status indicator no indication Redirection number Address signal (Diverted-to user) Call diversion information Notification subscription options presentation allowed with redirection number Redirecting reason Mobile subscriber not reachable Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed INVITE(Call-ID B-C, IAM) 183 Session Progress (Call-ID B-A, ACM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: 183 Session Progress is received at the interconnection interface Check: Is an ACM encapsulated in the 183? Check: Is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator is set to 'presentation allowed with redirection number'? Check: Is the Redirecting reason set to 'Mobile subscriber not reachable'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 150 Test case number SS_cfnl_014 Test case group SIP-SIP/Service/CFNL Reference 6.7/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 53 Test purpose SIP-I support. CFNL performed in Network B, Restriction of the Redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFNL, Diverted-to user is subscribed to the COLR service in Permanent mode. Ensure that when user A calls user B, the call is forwarded not logged in to user C, a Redirection number restriction parameter is present set to 'Presentation restricted' in the encapsulated ANM contained in the 200 OK INVITE if ISUP/BICC- SIP-I interworking is applicable in Network A. Configuration Subscription options: • Connected user subscribed to COLR, Permanent = yes SIP Parameter 200 OK Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Redirection number restriction Presentation restricted --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B), IAM CFNL is performed INVITE(Call-ID B-C) 180 Ringing (Call-ID C-B, ACM) 180 Ringing (Call-ID B-A) 200 OK INVITE (Call-ID C-B, ANM) ACK (Call-ID B-C) 200 OK INVITE (Call-ID B-A) ACK (Call-ID A-B) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 200 OK INVITE received at the interconnection interface Check: Is an ANM encapsulated in the 200 OK? Check: Is the ISUP/BICC Redirection number restriction set to 'Presentation restricted'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 151 Test case number SS_cfnl_015 Test case group SIP-SIP/Service/CFNL Reference 6.7/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 53 Test purpose SIP-I support. CFNL performed in Network B, No restriction of the Redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFNL, Diverted-to user is not subscribed to the COLR service. Ensure that when user A calls user B, the call is forwarded not logged in to user C, if a Redirection number restriction parameter is present it is set to 'Presentation allowed' in the encapsulated ANM contained in the 200 OK INVITE if ISUP/BICC- SIP-I interworking is applicable in Network A. Configuration Subscription options: • Connected user subscribed to COLR = no SIP Parameter 200 OK Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Redirection number restriction Presentation allowed or Redirection number restriction not present --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B), IAM CFNL is performed INVITE(Call-ID B-C) 180 Ringing (Call-ID C-B, ACM) 180 Ringing (Call-ID B-A) 200 OK INVITE (Call-ID C-B, ANM) ACK (Call-ID B-C) 200 OK INVITE (Call-ID B-A) ACK (Call-ID A-B) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 200 OK INVITE received at the interconnection interface? Check: Is an ANM encapsulated in the 200 OK? Check: Is the ISUP/BICC Redirection number restriction present set to 'Presentation allowed' or is the parameter absent? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 152 Test case number SS_cfnl_016 Test case group SIP-SIP/Service/CFNL Reference 7.1/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFNL performed in Network B, Notification of diverted-to user Redirecting number 'presentation allowed'. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFNL, Served user releases his/her number to diverted-to user = Release diverting number information. Ensure that when user A calls user B, the call is forwarded on Mobile subscriber not reachable to user C, user C is notified of call diversion and informed of the diverting number. The notification information is present in the encapsulated IAM contained in the Redirecting number 'presentation allowed' and Redirection information if ISUP/BICC - SIP-I interworking is applicable in Network B. Configuration Subscription options: • Served user releases his/her number to diverted-to user = Release diverting number information SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Redirecting number Address presentation restricted indicator presentation allowed Address signal (Diverting user) Original called number Address presentation restricted indicator presentation allowed Address signal Redirection information Original Redirection Reason unknown Redirecting indicator Redirection counter Redirecting reason Mobile subscriber not reachable --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed INVITE(Call-ID B-C, IAM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a INVITE request received at the interconnection interface? Check: Is an IAM encapsulated in the INVITE? Check: Is the Redirecting number present and the Address presentation restricted indicator is set to 'presentation allowed'? Check: Is the Original called number present and the Address presentation restricted indicator is set to 'presentation allowed'? Check: Is the Redirection number present? Check: Is Redirection information present and the Redirecting reason is set to 'Mobile subscriber not reachable'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 153 Test case number SS_cfnl_017 Test case group SIP-SIP/Service/CFNL Reference 7.1/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CFNL performed in Network B, Notification of diverted-to user Redirecting number 'presentation restricted'. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CFNL, Served user releases his/her number to diverted-to user = Release diverting number information. Ensure that when user A calls user B, the call is forwarded on Mobile subscriber not reachable to user C, user C is notified of call diversion and informed of the diverting number. The notification information is present in the encapsulated IAM contained in the Redirecting number 'presentation restricted' and Redirection information if ISUP/BICC - SIP-I interworking is applicable in Network B. Configuration Subscription options: • Served user releases his/her number to diverted-to user = Do not release diverting numberinformation SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Redirecting number Address presentation restricted indicator presentation restricted Address signal (Diverting user) Original called number Address presentation restricted indicator presentation restricted Address signal Redirection information Original Redirection Reason unknown Redirecting indicator Redirection counter Redirecting reason Mobile subscriber not reachable --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CFNL is performed INVITE(Call-ID B-C, IAM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a INVITE request received at the interconnection interface? Check: Is an IAM encapsulated in the INVITE? Check: Is the Redirecting number present and the Address presentation restricted indicator is set to 'presentation restricted'? Check: Is the Original called number present and the Address presentation restricted indicator is set to 'presentation restricted'? Check: Is the Redirection number present? Check: Is Redirection information present and the Redirecting reason is set to 'Mobile subscriber not reachable'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 154
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7f110eeff44cca5c59b5fc967e08482a
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101 585
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7.1.5.6.5 Communication Deflection
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Test case number SS_cd_001 Test case group SIP-SIP/Service/CD Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 29 Test purpose Communication deflection during alerting, basic rules. The user A and user C are in Network A. The user B is in network B and is provided with CDa. Ensure that when user A calls user B, the call is deflected during alerting to user C. In the active call state, ensure the property of speech. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CDa is performed 180 Ringing(Call-ID B-A) INVITE(Call-ID B-C) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) 200 OK INVITE(Call-ID C-B) ACK(Call-ID B-C) 200 OK INVITE(Call-ID B-A) ACK(Call-ID A-B) Communication Apply post test routine Comments Check: CDa is successful. Check: In the active call state, ensure the property of speech. Check: Is the P-Asserted-Identity present set to the identity of the originating user? Repeat this test in reverse direction. Test case number SS_cd_002 Test case group SIP-SIP/Service/CD Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 29 Test purpose Communication deflection immediate, basic rules. The user A and user C are located in Network A. The user B is located in network B and is provided with CDi. Ensure that when user A calls user B which deflects immediately the communication towards user C (i.e. before alerting starts), the call is forwarded to user C. In the active call state, ensure the property of speech. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CDi is performed INVITE(Call-ID B-C) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) 200 OK INVITE(Call-ID C-B) ACK(Call-ID B-C) 200 OK INVITE(Call-ID B-A) ACK(Call-ID A-B) Communication Apply post test routine Comments Check: CDi is successful. Check: In the active call state, ensure the property of speech. Check: Is the P-Asserted-Identity present set to the identity of the originating user? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 155 Test case number SS_cd_003 Test case group SIP-SIP/Service/CD Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 29 AND SE 30 Test purpose Communication Deflection immediate response, no notification. The user A and user C are located in Network A. The user B is located in network B and is provided with CDi, subscription option: Originating user receives notification that his communication has been diverted = No. Ensure that when user A calls user B which deflects immediately the communication towards user C (i.e. before alerting starts), the call is forwarded to user C. Ensure that User A does not receive a 181 Call Is Being Forwarded message. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = No SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CDi is performed INVITE(Call-ID B-C) 180 Ringing(Call-ID C-B) 180 Ringing(Call-ID B-A) Apply post test routine Comments Check: No notification regarding call forwarding in network B is received at the interconnection interface. Check: Is the cause parameter in the last entry is set to '480'. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 156 Test case number SS_cd_004 Test case group SIP-SIP/Service/CD Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 29 AND SE 30 Test purpose Communication Deflection immediate response, originating user is notified. URI of the diverted-to user not received. The user A and user C are located in network A. The user B is located in network B and is provided with CDi "Originating user receives notification that his communication has been diverted" = Yes and "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = No and. "Served user allows the presentation of his/her URI to originating user in diversion notification" = No. Ensure that when user A calls user B which deflects immediately the communication towards user C (i.e. before alerting starts), the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message, user A is notified of call diversion and not informed of the diverted-to number and served user number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Originating user receives notification that his communication has been diverted = No • Served user allows the presentation of his/her URI to originating user in diversion notification = No SIP Parameter 181 Being Forwarded History-Info: <sip:userB@networkB?Privacy=history&Reason=SIP;cause=302>;index=1, <sip: userC@networkA;cause=480?Privacy=history>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CDi is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) Apply post test routine Comments Check: A 181 Being Forwarded and a History-Info header is received at the interconnection interface in both entries in the History-Info header a Privacy header is escaped value 'history'. Check: Is the cause parameter in the last entry is set to '480'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 157 Test case number SS_cd_005 Test case group SIP-SIP/Service/CD Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 29 AND SE 30 Test purpose Communication Deflection immediate response, originating user is notified. URI from the diverted-to user received. The user A and user C are in network A. The user B is in network B and is provided with CDi "Originating user receives notification that his communication has been diverted" = Yes and "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = Yes. Ensure that when user A calls user B which deflects immediately the communication towards user C (i.e. before alerting starts), the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message, user A is notified of call diversion and informed of the diverted-to number. Configuration Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of diverted to URI to originating user in diversion notification = Yes SIP Parameter 181 Being Forwarded History-Info: <sip:userB@networkB?Reason=SIP;cause=302>;index=1, <sip: userC@networkA;cause=480>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CDi is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) Apply post test routine Comments Check: A 181 Being Forwarded is received at the interconnection interface. Check: A History-Info header is contained in the 181 with the URI of the diverted-to user. Check: Is the cause parameter in the last entry is set to '480'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 158 Test case number SS_cd_006 Test case group SIP-SIP/Service/CD Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 29 AND SE 30 Test purpose Communication Deflection immediate response, diverted-to user does not receive the URI of the served user. The user A and user C are in network A. The user B is in network B and is provided with CDi "Served user allows the presentation of his/her URI to the diverted-to user" = No. Ensure that when user A calls user B which deflects immediately the communication towards user C (i.e. before alerting starts), the call is forwarded to user C, user C is not informed of the forwarding number. Configuration Subscription options: • Served user allows the presentation of his/her URI to diverted-to user = No SIP Parameter INVITE Request line contains ';cause=480' History-Info: <sip:userB@networkB?Privacy=history&Reason=SIP;cause=302>;index=1, <sip: userC@networkA;cause=480>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CDi is performed INVITE(Call-ID B-C) Apply post test routine Comments Check: A History-Info header is received in the INVITE contains the URI of user B (served user) at the interconnection interface and a Privacy header is escaped set to 'history'. Check: Is the 'cause' parameter present in the Request line sent to user C (diverted-to user) set to '480'. Check: Is the cause parameter in the last entry is set to '480'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 159 Test case number SS_cd_007 Test case group SIP-SIP/Service/CD Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 29 AND SE 30 Test purpose Communication Deflection immediate response, diverted-to user receives the URI of the served user. The user A and user C are in network A. The user B is in network B and is provided with CDi "Served user allows the presentation of his/her URI to diverted-to user" = Yes. Ensure that when user A calls user B which deflects immediately the communication towards user C (i.e. before alerting starts), the call is forwarded to user C, user C is informed of the forwarding number. Configuration Subscription options: • Served user allows the presentation of his/her URI to diverted-to user = Yes SIP Parameter INVITE Request line contains ';cause=480' History-Info: <sip:userB@networkB?Reason=SIP;cause=302>;index=1, <sip: userC@networkA;cause=480>;index=1.1 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CDi is performed INVITE(Call-ID B-C) Apply post test routine Comments Check: A History-Info header is received in the INVITE contains the URI of user B (served user) at the interconnection interface. Check: Is the 'cause' parameter present in the Request line sent to user C (diverted-to user) set to '480'? Check: Is the cause parameter in the last entry is set to '480'? NOTE: The history entries can be accumulated in "one" History-Info header or each history entry is present in one single History-Info header. Repeat this test in reverse direction. Test case number SS_cd_008 Test case group SIP-SIP/Service/CD Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 29 Test purpose Communication Deflection immediate response, unsuccessful UDUB. The user A and user C are in network A. The user B is in network B and is provided with CDi. Ensure that when user A calls user B, the call is deflected immediate to user C user C is user determined user busy. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CDi is performed INVITE(Call-ID B-C) 486 Busy Here(Call-ID C-B) ACK(Call-ID B-C) 486 Busy Here(Call-ID B-A) ACK(Call-ID A-B) Apply post test routine Comments Check: The dialogue is terminated by receiving a 486 Busy Here. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 160 Test case number SS_cd_009 Test case group SIP-SIP/Service/CD Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 29 Test purpose Communication Deflection immediate response, unsuccessful NDUB. The user A and user C are in network A. The user B is in network B. Ensure that when user A calls user B, the call is deflected immediate to user C and user C is network determined user busy. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CDi is performed INVITE(Call-ID B-C) 486 Busy Here(Call-ID C-B) ACK(Call-ID B-C) 486 Busy Here(Call-ID B-A) ACK(Call-ID A-B) Apply post test routine Comments Check: The dialogue is terminated by receiving a 486 Busy Here. Repeat this test in reverse direction Test case number SS_cd_010 Test case group SIP-SIP/Service/CD Reference 4.5.2.6/ [9] SELECTION EXPRESSION SE 29 AND SE 30 AND [Network A] SE 9 Test purpose Communication Deflection immediate response, interaction with a not trusted network. The user A and user C are in network A. The user B is in network B and is provided with CDi "Originating user receives notification that his communication has been diverted" = Yes "Served user allows the presentation of forwarded to URI to originating user in diversion notification" = No, "diverting number is released to the diverted-to user" = No. Ensure that when user A calls user B, the call is deflected immediate response to user C, user A is notified of call diversion and not informed of the diverted-to number and user C is not informed of the forwarding number. Configuration SIP Parameter Subscription options: • Originating user receives notification that his communication has been diverted = Yes • Served user allows the presentation of forwarded to URI to originating user in diversion notification = No • Served user allows the presentation of his/her URI to originating user in diversion notification = No • Served user allows the presentation of his/her URI to the diverted-to user = No SIP Parameter INVITE: no History-Info header 181 Being Forwarded no History-Info header Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) CDi is performed INVITE(Call-ID B-C) 181 Being Forwarded(Call-ID B-A) Apply post test routine Comments Check: No History-Info header is received in the INVITE at the interconnection interface. Check: No History-Info header is received in the 181 Being Forwarded at the interconnection interface. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 161 Test case number SS_cd_011 Test case group SIP-SIP/Service/CD Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CD performed in Network B, Notification subscription options is set to presentation not allowed. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CDi or CDa, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number. Ensure that when user A calls user B, the call is deflected to user C, user A is not notified about call diversion. The notification information is present in the encapsulated ACM or CPG contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = no SIP Parameter 183 /180 Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM/CPG Redirection number Address signal (Diverted-to user) Call diversion information Notification subscription options presentation not allowed Redirecting reason Deflection immediate or Deflection during alerting Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing (Call-ID B-A, ACM) in case CDa CD is performed INVITE(Call-ID B-C, IAM) 183 / 180 (Call-ID B-A, ACM/CPG) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 183 Session Progress received at the interconnection interface? Check: Is an ACM encapsulated in the 183? Check: Is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator set to 'presentation not allowed'? Check: Is the Redirecting reason set to 'Deflection immediate' or 'Deflection during alerting'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 162 Test case number SS_cd_012 Test case group SIP-SIP/Service/CD Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CD performed in Network B, Notification subscription options is set to presentation allowed without redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CDi or CDa, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number. Ensure that when user A calls user B, the call is deflected to user C, user A is notified of call diversion and informed of the diverted-to number. The notification information is present in the encapsulated ACM or CPG contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, without diverted-to user number SIP Parameter 183 /180 Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM/CPG Redirection number Address signal (Diverted-to user) Call diversion information Notification subscription options presentation allowed without redirection number Redirecting reason Deflection immediate or Deflection during alerting Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing (Call-ID B-A) in case CDa CD is performed INVITE(Call-ID B-C, IAM) 183 / 180 (Call-ID B-A, ACM/CPG) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: 183 Session Progress is received at the interconnection interface. Check: Is an ACM encapsulated in the 183? Check: Is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator is set to 'presentation allowed without redirection number'? Check: Is the Redirecting reason set to 'Deflection immediate' or 'Deflection during alerting'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 163 Test case number SS_cd_013 Test case group SIP-SIP/Service/CD Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CD performed in Network B, Notification subscription options is set to presentation allowed with redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CDi or CDa, Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, with diverted-to user number. Ensure that when user A calls user B, the call is deflected to user C, user A is notified of call diversion and informed of the diverted-to number. The notification information is present in the encapsulated ACM or CPG contained in the Redirection number and Call diversion information if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration Subscription options: • Calling user receives notification that his call has been diverted (forwarded or deflected) = yes, with diverted-to user number SIP Parameter 183 /180 Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM/CPG Redirection number Address signal (Diverted-to user) Call diversion information Notification subscription options presentation allowed with redirection number Redirecting reason Deflection immediate or Deflection during alerting Generic notification call is diverting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing (Call-ID B-A) in case CDa CD is performed INVITE(Call-ID B-C, IAM) 183 / 180 (Call-ID B-A, ACM/CPG) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: 183 Session Progress is received at the interconnection interface. Check: Is an ACM encapsulated in the 183? Check: Is the Called party's status indicator set to 'no indication'? Check: Is the Redirection number present? Check: Is Notification subscription options indicator is set to 'presentation allowed with redirection number'? Check: Is the Redirecting reason set to 'Deflection immediate' or 'Deflection during alerting'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 164 Test case number SS_cd_014 Test case group SIP-SIP/Service/CD Reference 6.7/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 53 Test purpose SIP-I support. CD performed in Network B, Restriction of the Redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CDi or CDa, Diverted-to user is subscribed to the COLR service in Permanent mode. Ensure that when user A calls user B, the call is deflected to user C, a Redirection number restriction parameter is present set to 'Presentation restricted' in the encapsulated ANM contained in the 200 OK INVITE if ISUP/BICC- SIP-I interworking is applicable in Network A. Configuration Subscription options: • Connected user subscribed to COLR, Permanent = yes SIP Parameter 200 OK Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Redirection number restriction Presentation restricted [24] --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B), IAM 180 Ringing (Call-ID B-A) in case CDa CD is performed INVITE(Call-ID B-C) 180 Ringing (Call-ID C-B, ACM) 180 Ringing (Call-ID B-A) 200 OK INVITE (Call-ID C-B, ANM) ACK (Call-ID B-C) 200 OK INVITE (Call-ID B-A) ACK (Call-ID A-B) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 200 OK INVITE received at the interconnection interface? Check: Is an ANM encapsulated in the 200 OK? Check: Is the ISUP/BICC Redirection number restriction set to 'Presentation restricted'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 165 Test case number SS_cd_015 Test case group SIP-SIP/Service/CD Reference 6.7/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 53 Test purpose SIP-I support. CD performed in Network B, No restriction of the Redirection number. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CDi or CDa, Diverted-to user is not subscribed to the COLR service. Ensure that when user A calls user B, the call is deflected to user C, if a Redirection number restriction parameter is present it is set to 'Presentation allowed' in the encapsulated ANM contained in the 200 OK INVITE if ISUP/BICC- SIP-I interworking is applicable in Network A. Configuration Subscription options: • Connected user subscribed to COLR = no SIP Parameter 200 OK Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Redirection number restriction Presentation allowed or Redirection number restriction not present --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B), IAM 180 Ringing (Call-ID B-A) in case CDa CD is performed INVITE(Call-ID B-C) 180 Ringing (Call-ID C-B, ACM) 180 Ringing (Call-ID B-A) 200 OK INVITE (Call-ID C-B, ANM) ACK (Call-ID B-C) 200 OK INVITE (Call-ID B-A) ACK (Call-ID A-B) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a 200 OK INVITE received at the interconnection interface? Check: Is an ANM encapsulated in the 200 OK? Check: Is the ISUP/BICC Redirection number restriction present set to 'Presentation allowed' or is the parameter absent? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 166 Test case number SS_cd_016 Test case group SIP-SIP/Service/CD Reference 7.1/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CD performed in Network B, Notification of diverted-to user Redirecting number 'presentation allowed'. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CDi or CDa, Served user releases his/her number to diverted-to user = Release diverting number information. Ensure that when user A calls user B, the call is deflected to user C, user C is notified of call diversion and informed of the diverting number. The notification information is present in the encapsulated IAM contained in the Redirecting number 'presentation allowed' and Redirection information if ISUP/BICC - SIP-I interworking is applicable in Network B. Configuration Subscription options: • Served user releases his/her number to diverted-to user = Release diverting number information SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Redirecting number Address presentation restricted indicator presentation allowed Address signal (Diverting user) Original called number Address presentation restricted indicator presentation allowed Address signal Redirection information Original Redirection Reason unknown Redirecting indicator Redirection counter Redirecting reason Deflection immediate or Deflection during alerting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing (Call-ID B-A) in case CDa CD is performed INVITE(Call-ID B-C, IAM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a INVITE request received at the interconnection interface? Check: Is an IAM encapsulated in the INVITE? Check: Is the Redirecting number present and the Address presentation restricted indicator is set to 'presentation allowed'? Check: Is the Original called number present and the Address presentation restricted indicator is set to 'presentation allowed'? Check: Is the Redirection number present? Check: Is Redirection information present and the Redirecting reason is set to 'Deflection immediate' or 'Deflection during alerting'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 167 Test case number SS_cd_017 Test case group SIP-SIP/Service/CD Reference 7.1/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 55 Test purpose SIP-I support. CD performed in Network B, Notification of diverted-to user Redirecting number 'presentation restricted'. The user A and user C are in Network A. The user B is in the PSTN/PLMN part of Network B and is provided with CDi or CDa, Served user releases his/her number to diverted-to user = Release diverting number information. Ensure that when user A calls user B, the call is deflected to user C, user C is notified of call diversion and informed of the diverting number. The notification information is present in the encapsulated IAM contained in the Redirecting number 'presentation restricted' and Redirection information if ISUP/BICC - SIP-I interworking is applicable in Network B. Configuration Subscription options: • Served user releases his/her number to diverted-to user = Do not release diverting numberinformation SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Redirecting number Address presentation restricted indicator presentation restricted Address signal (Diverting user) Original called number Address presentation restricted indicator presentation restricted Address signal Redirection information Original Redirection Reason unknown Redirecting indicator Redirection counter Redirecting reason Deflection immediate or Deflection during alerting --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(Call-ID A-B) 180 Ringing (Call-ID B-A) in case CDa CD is performed INVITE(Call-ID B-C, IAM) Apply post test routine Comments Originating user in Network A establishes a call to user in Network B. Network B performs the diversion to a user in Network A Check: Is a INVITE request received at the interconnection interface? Check: Is an IAM encapsulated in the INVITE? Check: Is the Redirecting number present and the Address presentation restricted indicator is set to 'presentation restricted'? Check: Is the Original called number present and the Address presentation restricted indicator is set to 'presentation restricted'? Check: Is the Redirection number present? Check: Is Redirection information present and the Redirecting reason is set to 'Deflection immediate' or 'Deflection during alerting'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 168
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7f110eeff44cca5c59b5fc967e08482a
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101 585
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7.1.5.7 Conference (CONF)
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Test case number SS_conf_001 Test case group SIP-SIP/Service/CONF Reference 4.5.2/ [10] SELECTION EXPRESSION ([Network A] SE 11 AND [Network B] SE 11) AND SE 31 Test purpose 3 Party establishment using the REFER method. User B1 and user B2 are located in network B, user A is located in network A. A confirmed session from user A to user B1 is set on hold; a confirmed session from user A to user B2 is set on hold. • Ensure that when user A refers to user B1 to invite to the conference, the user B1 sends a NOTIFY to user A indicating 'Tying'. The user B1 sends an INVITE request to the conference focus in network A. Is the request is confirmed, user B1 sends a NOTIFY indicating '200 OK'. User A terminates the original dialogue. • Ensure that when user A refers to user B2 to invite to the conference, the user B2 sends a NOTIFY to user A indicating 'Tying'. The user B2 sends an INVITE request to the conference focus in network A. Is the request is confirmed, user B2 sends a NOTIFY indicating '200 OK'. User A terminates the original dialogue. Configuration SIP Parameter REFER(user B1) Refer-To: <uri of conference focus;method=INVITE > NOTIFY(B1, 100) Content-Type: message/sipfrag SIP/2.0 100 INVITE: Request URI: uri of conference focus From: user B1 NOTIFY(B1, 200) Content-Type: message/sipfrag SIP/2.0 200 OK REFER(user B2) Refer-To: <uri of conference focus;method=INVITE > NOTIFY(B2, 100) Content-Type: message/sipfrag SIP/2.0 100 INVITE: Request URI: uri of conference focus From: user B2 NOTIFY(B2, 200) Content-Type: message/sipfrag SIP/2.0 200 OK ETSI ETSI TS 101 585 V1.2.1 (2014-04) 169 Message flow SIP (Network A) Interconnection Interface SIP (Network B) Establish a confirmed session to user B1 from Network A to Network B and put it on hold Establish a confirmed session to user B2 from Network A to Network B and put it on hold User A establishes a 3PTY conversation REFER(user B1) 202 Accepted NOTIFY(B1, 100) 200 OK NOTIFY INVITE(focus, user B1) 200 INVITE ACK NOTIFY(B1, 200) 200 OK NOTIFY BYE(user B1) 200 OK BYE REFER(user B2) 202 Accepted NOTIFY(100) 200 OK NOTIFY INVITE(focus, user B2) 200 INVITE ACK NOTIFY(B2, 200) 200 OK NOTIFY BYE(user B2) 200 OK BYE Apply post test routine Comments User A establishes a 3PTY conversation after the confirmed communication to user B1 and B2 are set on hold Check: The Refer-To header in the REFER method sent to user B1 and B2 contains the URI of the conference focus and is the method parameter set to 'INVITE'. Check: The NOTIFY after the REFER request contains the 'SIP/2.0 100' message body. Check: The INVITE request is sent by user B1 and user B2 to the conference focus the Request URI is used from the Refer-To header of the received REFER request. Check: The NOTIFY after the REFER request contains the 'SIP/2.0 200 OK' message body. Check: The original session is terminated by user A. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 170 Test case number SS_conf_002 Test case group SIP-SIP/Service/CONF Reference 4.5.2/ [10], 4.7.2.9.7/ [20] SELECTION EXPRESSION [Network A] SE 12 AND SE 31 Test purpose 3 Party establishment using reINVITE performed by the AS in network A. User B1 and user B2 are located in network B, user A is located in network A. A confirmed session from user A to user B1 is set on hold; a confirmed session from user A to user B2 is set on hold. • Ensure that user A can invite user B1 to the conference by sending a reINVITE request. • Ensure that user A can invite user B2 to the conference by sending a reINVITE request. Configuration SIP Parameter INVITE <B1> From: <userA> To: <userB1> Call-ID: A-B1 P-Asserted-Identity: <userA> SDP: a=sendrecv INVITE <B2> From: <userA> Call-ID: A-B2 To: <userB2> P-Asserted-Identity: <userA> SDP: a=sendrecv Message flow SIP (Network A) Interconnection Interface SIP (Network B) Establish a confirmed session to user B1 from Network A to Network B and put it on hold Establish a confirmed session to user B2 from Network A to Network B and put it on hold User A establishes a 3PTY conversation INVITE(Call-ID A-B1) 200 INVITE ACK INVITE(Call-ID A-B2) 200 INVITE ACK Apply post test routine Comments User A establishes a 3PTY conversation after the confirmed communication to user B1 and B2 are set on hold Check: An INVITE is sent to user B1 and user B2 indicating a new IP address in the 'c' line of the SDP. Check: The 'a' line indicates 'sendrecv'. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 171 Test case number SS_conf_003 Test case group SIP-SIP/Service/CONF Reference 5.4/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 56 Test purpose SIP-I/ISUP interworking. Served user establishes a 3 Party communication. Served User A is located in Network A and ISUP/BICC - SIP-I interworking applies in Network A. User A establishes a confirmed communication with a User B1 in Network B and sets it on HOLD. User A establishes a confirmed communication with a User B2 in Network B. Ensure that when User A establishes a 3 PTY communication: • an INFO request is sent to User B1 in Network B and a ISUP/BICC CPG is encapsulated the Generic Notification is set to 'conference established'; • an INFO request is sent to User B2 in Network B and a ISUP/BICC CPG is encapsulated the Generic Notification is set to 'conference established'. Configuration ISUP/BICC interworking applies in Network A User in Network A is subscribed to the 3PTY supplementary service SIP Parameter INFO <B1> Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic Notification Conference established INFO <B2> Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic Notification Conference established Message flow SIP (Network A) Interconnection Interface SIP (Network B) Establish a confirmed session from User A in Network A to user B1 in Network B and put it on hold Establish a confirmed session from User A in Network A to user B2 in Network B INFO(Call-ID A-B1, CPG) 200 INFO INFO(Call-ID A-B2, CPG) 200 INFO Apply post test routine Comments User A establishes confirmed communication to user B1 in Network B and sets it on hold User A establishes a confirmed communication to user B2 in Network B User A invokes the 3PTY communication Check: Is an INFO request sent to user B1 and user B2 in Network B? Check: Is a ISUP/BICC CPG message encapsulated in the INFO request to both remote users in Network B? Check: Is the Generic Notification parameter in the encapsulated CPG in both INFO set to 'Conference established'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 172 Test case number SS_conf_004 Test case group SIP-SIP/Service/CONF Reference 5.4/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 56 Test purpose SIP-I/ISUP interworking. Served user disconnects one of the remote users. Served User A is located in Network A and ISUP/BICC - SIP-I interworking applies in Network A. User A establishes a confirmed communication with a User B1 in Network B and sets it on HOLD. User A establishes a confirmed communication with a User B2 in Network B. User A invokes 3PTY conversation. Ensure that when User A disconnects the previous active user: • a BYE request is sent to User B1 in Network B; • an INFO request is sent to User B2 in Network B and a ISUP/BICC CPG is encapsulated the Generic Notification is set to 'Conference disconnected'. Configuration ISUP/BICC interworking applies in Network A User in Network A is subscribed to the 3PTY supplementary service SIP Parameter INFO <B2> Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic Notification Conference disconnected Message flow SIP (Network A) Interconnection Interface SIP (Network B) Establish a confirmed session from User A in Network A to user B1 in Network B and put it on hold Establish a confirmed session from User A in Network A to user B2 in Network B User A establishes a 3PTY conversation BYE(Call-ID A-B1, REL) 200 INFO INFO(Call-ID A-B2, CPG) 200 INFO Apply post test routine Comments User A establishes a 3PTY conversation with user B1 and user B2 located in Network B User A disconnects the communication with user B1 in Network B (previous on hold) Check: Is a BYE request is sent to user B1 in Network B? Check: Is a ISUP/BICC CPG message encapsulated in the INFO request to user B2 in Network B? Check: Is the Generic Notification parameter in the encapsulated CPG in the INFO sent to user B2 set to 'Conference disconnected'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 173 Test case number SS_conf_005 Test case group SIP-SIP/Service/CONF Reference 5.4/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 56 Test purpose SIP-I/ISUP interworking. Served user splits the 3 Party communication. Served User A is located in Network A and ISUP/BICC - SIP-I interworking applies in Network A. User A establishes a confirmed communication with a User B1 in Network B and sets it on HOLD. User A establishes a confirmed communication with a User B2 in Network B. User A invokes 3PTY conversation Ensure that when User A splits the 3 PTY communication: • an INFO request is sent to User B1 in Network B and a ISUP/BICC CPG is encapsulated the Generic Notification is set to 'Conference disconnected'; • an INFO request is sent to User B2 in Network B and a ISUP/BICC CPG is encapsulated the Generic Notification is set to 'Conference disconnected'. Configuration ISUP/BICC interworking applies in Network A User in Network A is subscribed to the 3PTY supplementary service SIP Parameter INFO <B1> Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic Notification Conference disconnected INFO <B2> Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic Notification Conference disconnected Message flow SIP (Network A) Interconnection Interface SIP (Network B) Establish a confirmed session from User A in Network A to user B1 in Network B and put it on hold Establish a confirmed session from User A in Network A to user B2 in Network B User A establishes a 3PTY conversation INFO(Call-ID A-B1, CPG) 200 INFO INFO(Call-ID A-B2, CPG) 200 INFO Apply post test routine Comments User A establishes confirmed communication to user B1 in Network B and sets it on hold User A establishes a confirmed communication to user B2 in Network B Check: Is an INFO request sent to user B1 and user B2 in Network B? Check: Is a ISUP/BICC CPG message encapsulated in the INFO request to both remote users in Network B? Check: Is the Generic Notification parameter in the encapsulated CPG in both INFO set to 'Conference established'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 174 Test case number SS_conf_006 Test case group SIP-SIP/Service/CONF Reference 5.4/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 56 Test purpose SIP-I/ISUP interworking. Establishment of aCONF conversation. Served User A is located in Network A and ISUP/BICC - SIP-I interworking applies in Network A. User A establishes a confirmed communication with a User B1 in Network B and invokes the CONF communication. Ensure that when User A invokes the CONF communication: • an INFO request is sent to User B1 in Network B and a ISUP/BICC CPG is encapsulated the Generic Notification is set to 'conference established' when the conference is invoked. User A establishes a confirmed communication with a User B2 in Network B. Ensure when User A adds the user B2 to the established conference: • an INFO request is sent to User B1 in Network B and a ISUP/BICC CPG is encapsulated the Generic Notification is set to 'Other party; • an INFO request is sent to User B2 in Network B and a ISUP/BICC CPG is encapsulated the Generic Notification is set to 'conference established' when the user is added to the conference. Configuration ISUP/BICC interworking applies in Network A User in Network A is subscribed to the 3PTY supplementary service SIP Parameter INFO1 <B1> Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic Notification conference established INFO2 <B1> Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic Notification Other party added INFO3 <B2> Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic Notification conference established Message flow SIP (Network A) Interconnection Interface SIP (Network B) Establish a confirmed session from User A in Network A to user B1 in Network B User A establishes a CONF conversation INFO1(Call-ID A-B1, CPG) 200 INFO Establish a confirmed session from User A in Network A to user B2 in Network B and add to the conference INFO2(Call-ID A-B2, CPG) 200 INFO INFO3(Call-ID A-B2, CPG) 200 INFO Apply post test routine ETSI ETSI TS 101 585 V1.2.1 (2014-04) 175 Comments User A establishes confirmed communication to user B1 in Network B and invoke the CONF communication Check: Is an INFO request sent to user B1 and in Network B and Is a ISUP/BICC CPG message encapsulated in the INFO request and the Generic Notification is set to 'conference established'? User A establishes a confirmed communication to user B2 in Network B and add it to the conference. Check: Is an INFO request sent to user B2 Network B and a ISUP/BICC CPG message encapsulated the Generic Notification is set to 'conference established'? Check: Is an INFO request sent to user B1 Network B and a ISUP/BICC CPG message encapsulated the Generic Notification is set to 'Other party added'? Repeat this test in reverse direction. Test case number SS_conf_007 Test case group SIP-SIP/Service/CONF Reference 5.4/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 56 Test purpose SIP-I/ISUP interworking. Isolation and Reattachment of one party of the conference. Served User A is located in Network A and ISUP/BICC - SIP-I interworking applies in Network A. User A invokes a CONF communication with user B1 and user B2 in Network B. Ensure that when User A isolates one remote party (B1) from the CONF communication: • an INFO request is sent to User B1 in Network B and the Generic Notification is set to 'isolated' in the encapsulated ISUP/BICCCPG; • an INFO request is sent to User B2 in Network B and the Generic Notification is set to 'Other party isolated' in the encapsulated ISUP/BICCCPG. Ensure that when User A reattaches one remote party (B1) to the CONF communication: • an INFO request is sent to User B1 in Network B and the Generic Notification is set to 'reattached' in the encapsulated ISUP/BICCCPG; • an INFO request is sent to User B2 in Network B and the Generic Notification is set to 'Other party reattached' in the encapsulated ISUP/BICCCPG. Configuration ISUP/BICC interworking applies in Network A User in Network A is subscribed to the 3PTY supplementary service SIP Parameter INFO1 <B1> CPG Generic Notification= isolated INFO2 <B1> CPG Generic Notification= Other party isolated INFO3 <B2> CPG Generic Notification= reattached INFO4 <B2> CPG Generic Notification= Other party reattached ETSI ETSI TS 101 585 V1.2.1 (2014-04) 176 Message flow SIP (Network A) Interconnection Interface SIP (Network B) Establish a CONF communication with User B1 and User B2 in Network B User A isolates User B1 from the CONF conversation INFO1(Call-ID A-B1, CPG) 200 INFO INFO3(Call-ID A-B2, CPG) 200 INFO User A reattaches User B1 to the CONF conversation INFO2(Call-ID A-B2, CPG) 200 INFO INFO4(Call-ID A-B2, CPG) 200 INFO Apply post test routine Comments User A Invokes a CONF conversation with User B1 and User b2 in Network B User A splits user B1 in Network B from the CONF conversation Check: Is an INFO request sent to user B1 and the Generic notification is set to 'isolated' in the encapsulated CPG? Check: Is an INFO request sent to user B2 and the Generic notification is set to 'Other party isolated' in the encapsulated CPG? User A reattaches user B1 in Network B to the CONF conversation Check: Is an INFO request sent to user B1 and the Generic notification is set to 'reattached' in the encapsulated CPG? Check: Is an INFO request sent to user B2 and the Generic notification is set to 'Other party reattached' in the encapsulated CPG? Repeat this test in reverse direction ETSI ETSI TS 101 585 V1.2.1 (2014-04) 177 Test case number SS_conf_008 Test case group SIP-SIP/Service/CONF Reference 5.4/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 56 Test purpose SIP-I/ISUP interworking. Splitting and Adding of a party. Served User A is located in Network A and ISUP/BICC - SIP-I interworking applies in Network A. User A invokes a CONF communication with user B1 and user B2 in Network B. Ensure that when User A split one remote party (B1) from the CONF communication: • an INFO request is sent to User B1 in Network B and the Generic Notification is set to 'conference disconnected' in the encapsulated ISUP/BICCCPG; • an INFO request is sent to User B2 in Network B and the Generic Notification is set to 'Other party split' in the encapsulated ISUP/BICCCPG. Ensure that when User A adds one remote party (B1) to the CONF communication: • an INFO request is sent to User B1 in Network B and the Generic Notification is set to 'Conference established' in the encapsulated ISUP/BICCCPG; • an INFO request is sent to User B2 in Network B and the Generic Notification is set to 'Other party added' in the encapsulated ISUP/BICCCPG. Configuration ISUP/BICC interworking applies in Network A User in Network A is subscribed to the 3PTY supplementary service SIP Parameter INFO1 <B1> CPG Generic Notification= conference disconnected INFO2 <B1> CPG Generic Notification=Other party split INFO3 <B2> CPG Generic Notification=Conference established INFO4 <B2> CPG Generic Notification= Other party added Message flow SIP (Network A) Interconnection Interface SIP (Network B) Establish a CONF communication with User B1 and User B2 in Network B User A isolates User B1 from the CONF conversation INFO1(Call-ID A-B1, CPG) 200 INFO INFO3(Call-ID A-B2, CPG) 200 INFO User A reattaches User B1 to the CONF conversation INFO2(Call-ID A-B2, CPG) 200 INFO INFO4(Call-ID A-B2, CPG) 200 INFO Apply post test routine Comments User A Invokes a CONF conversation with User B1 and User b2 in Network B User A splits user B1 in Network B from the CONF conversation. Check: Is an INFO request sent to user B1 and the Generic notification is set to 'conference disconnected' in the encapsulated CPG? Check: Is an INFO request sent to user B2 and the Generic notification is set to 'Other party split' in the encapsulated CPG? User A adds user B1 in Network B to the CONF conversation. Check: Is an INFO request sent to user B1 and the Generic notification is set to 'Conference established' in the encapsulated CPG? Check: Is an INFO request sent to user B2 and the Generic notification is set to 'Other party added' in the encapsulated CPG? Repeat this test in reverse direction ETSI ETSI TS 101 585 V1.2.1 (2014-04) 178 7.1.5.8 Anonymous Communication Rejection (ACR) and Communication Barring (CB) Test case number SS_acr-cb_001 Test case group SIP-SIP/Service/ACR-CB Reference 4.5.2.6/ [12] SELECTION EXPRESSION SE 32 Test purpose Call Barring performed in network B for user B. User A is located in network A and user B is located in network B and is subscribed to the Incoming Call Barring service. Ensure that a communication from user A is rejected in network B by sending a 603 Decline due to the Call Barring service of user B. Configuration User B is subscribed to the incoming Call Barring service (e.g. user A in a black list) SIP Parameter INVITE P-Asserted-Identity: <URI of user A> Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 603 (Decline) ACK Comments Check: Is the P-Asserted-Identity present? Check: Is the communication rejected by sending a 603 (Decline) final response sent to user A? Repeat this test in reverse direction. Test case number SS_acr-cb_002 Test case group SIP-SIP/Service/ACR-CB Reference 4.5.2.6/ [12] SELECTION EXPRESSION SE 33 Test purpose ACR performed in network B for user B. User A is located in network A and user B is located in network B and is subscribed to the Anonymous Communication rejection service. Ensure that an anonymous communication from user A is rejected in network B by sending a 403 Anonymity Disallowed final response due to the Anonymous Communication Rejection service of user B. Configuration User B is subscribed to the Anonymous Communication Rejection service SIP Parameter INVITE P-Asserted-Identity: <URI of user A> Privacy: id Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 433 (Anonymity Disallowed) ACK Comments Check: Is the P-Asserted-Identity present? Check: Is the Privacy header set to 'id'? Check: Is the communication rejected by sending a 433 (Anonymity Disallowed) final response sent to user A? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 179 Test case number SS_acr-cb_003 Test case group SIP-SIP/Service/ACR-CB Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 57 Test purpose SIP-I interworking. ACR performed in network B for user B. User A is located in network A and user B is in the PSTN/PLMN part of Network B and is subscribed to the Anonymous Communication rejection service. Ensure that an anonymous communication from user A is rejected in network B by sending a 603 Decline final response due to the Anonymous Communication Rejection service of user B. A ISUP/BICC REL is present in the 603 the Cause indicator value is set to '21' if SIP-I - ISUP/BICC interworking is applicable in Network B. Configuration User B is subscribed to the Anonymous Call Rejection service SIP Parameter INVITE P-Asserted-Identity: <URI of user A> Privacy: id 433 Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL: Cause indicator Cause = 21 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 603 Decline (REL) ACK Comments Check: Is the P-Asserted-Identity present? Check: Is the Privacy header set to 'id'? Check: Is the communication rejected by sending a 603 Decline final response sent to user A? Check: Is an ISUP/BICC REL is present in the 603 and the cause value is set to '21'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 180
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7f110eeff44cca5c59b5fc967e08482a
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101 585
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7.1.5.9 Closed User Group (CUG)
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Test case number SS_cug_001 Test case group SIP-SIP/Service/CUG Reference 4.5.2.4/ [13] SELECTION EXPRESSION SE 34 Test purpose Originating user +OA to terminating user no CUG. An originating user in a CUG Outgoing Access allowed calls to a user not in a CUG. The session establishment is successful. Configuration Originating user: CUG, outgoing access allowed SIP Parameter INVITE: Content-Type: application/vnd.etsi.cug+xml Content-Disposition: signal;handling= ...... ……. <…cug> <…networkIndicator>01</… networkIndicator <…networkIndicator>23</… networkIndicator <….cugInterlockBinaryCode>0F03</…cugInterlockBinaryCode> <…CugCommunicationIndicator>10</…cugCommunicationIndicator> <…:cug> Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Apply post test routine Comments Check: Is the Content-Type in The INVITE set to application/vnd.etsi.cug+xml? Check: Contains the XML body in the INVITE a 'cug' element? Check: Contains the XML body in the INVITE a 'networkIndicator' element as a 'cug' child element? Check: Contains the XML body in the INVITE a 'cugInterlockBinaryCode' element as a 'cug' child element? Check: Contains the XML body in the INVITE a 'cugCommunicationIndicator' element set to '10' as a 'cug' child element? Check: Is the session setup not rejected? Repeat this test in reverse direction. NOTE: The networkIndicator element value and the cugInterlockBinaryCode element value are examples. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 181 Test case number SS_cug_002 Test case group SIP-SIP/Service/CUG Reference 4.5.2.4, 4.5.2.10/ [13] SELECTION EXPRESSION SE 34 Test purpose Originating user -OA to terminating user no CUG. An originating user in a CUG Outgoing Access not allowed calls to a user not in a CUG. The session establishment is not successful, a 403 (Forbidden) response is sent. Configuration Originating user: CUG, outgoing access not allowed SIP Parameter INVITE: Content-Type: application/vnd.etsi.cug+xml Content-Disposition: signal;handling= required ……. <…:cug> <…networkIndicator>01</…networkIndicator <…networkIndicator>23</…networkIndicator <….ugInterlockBinaryCode>0F03</…cugInterlockBinaryCode> <…cugCommunicationIndicator>11</…cugCommunicationIndicator> <…cug> Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 403 (Forbidden) ACK Comments Check: Is the Content-Type in The INVITE set to application/vnd.etsi.cug+xml? Check: Is the handling parameter in the Content-Disposition header set to required? Check: Contains the XML body in the INVITE a 'cug' element? Check: Contains the XML body in the INVITE a 'networkIndicator' element as a 'cug' child element? Check: Contains the XML body in the INVITE a 'cugInterlockBinaryCode' element as a 'cug' child element? Check: Contains the XML body in the INVITE a 'cugCommunicationIndicator' element set to '11' as a 'cug' child element? Check: Is the session setup rejected? A 403 (Forbidden) final response is sent by the terminating network? Repeat this test in reverse direction. NOTE: The networkIndicator element value and the cugInterlockBinaryCode element value are examples. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 182 Test case number SS_cug_003 Test case group SIP-SIP/Service/CUG Reference 4.5.2.4, 4.5.2.10/ [13] SELECTION EXPRESSION SE 34 Test purpose Originating user -OA to terminating user -IA. An originating user in a CUG Outgoing Access not allowed calls to a user in the same CUG Incoming Access not allowed. The session establishment is successful. Configuration Originating user: CUG, outgoing access not allowed Terminating user: CUG incoming access not allowed User in network A and user in network B are in the same CUG SIP Parameter INVITE: Content-Type: application/vnd.etsi.cug+xml Content-Disposition: signal;handling= required ……. <…cug> <…networkIndicator>01</…networkIndicator <…networkIndicator>23</…networkIndicator <…cugInterlockBinaryCode>0F03</…cugInterlockBinaryCode> <…cugCommunicationIndicator>11</…cugCommunicationIndicator> <…cug> Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Apply post test routine Comments Check: Is the Content-Type in The INVITE set to application/vnd.etsi.cug+xml? Check: Is the handling parameter in the Content-Disposition header set to required? Check: Contains the XML body in the INVITE a 'cug' element? Check: Contains the XML body in the INVITE a 'networkIndicator' element as a 'cug' child element? Check: Contains the XML body in the INVITE a 'cugInterlockBinaryCode' element as a 'cug' child element? Check: Contains the XML body in the INVITE a 'cugCommunicationIndicator' element set to '11' as a 'cug' child element? Check: Is the session setup not rejected? Repeat this test in reverse direction. NOTE: The networkIndicator element value and the cugInterlockBinaryCode element value are examples. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 183 Test case number SS_cug_004 Test case group SIP-SIP/Service/CUG Reference 4.5.2.4, 4.5.2.10/ [13] SELECTION EXPRESSION SE 34 Test purpose Originating user in a CUG to terminating user -IA. An originating user in a CUG calls to a user in a different CUG Incoming Access not allowed. The session establishment is not successful, a 403 (Forbidden) response is sent. Configuration User in network A and user in network B are not in the same CUG Terminating user: CUG incoming access not allowed SIP Parameter INVITE: Content-Type: application/vnd.etsi.cug+xml Content-Disposition: signal;handling= ..... ……. <…cug> <…networkIndicator>01</…networkIndicator <…networkIndicator>23</…networkIndicator <…cugInterlockBinaryCode>0F03</…cugInterlockBinaryCode> <…cugCommunicationIndicator>..</…cugCommunicationIndicator> <…cug> Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 403 (Forbidden) ACK Comments Check: Is the Content-Type in The INVITE set to application/vnd.etsi.cug+xml? Check: Contains the XML body in the INVITE a 'cug' element? Check: Contains the XML body in the INVITE a 'networkIndicator' element as a 'cug' child element? Check: Contains the XML body in the INVITE a 'cugInterlockBinaryCode' element as a 'cug' child element? Check: Contains the XML body in the INVITE a 'cugCommunicationIndicator' element set to '10' or '11'as a 'cug' child element? Check: Is the session setup rejected? A 403 (Forbidden) final response is sent by the terminating network? Repeat this test in reverse direction. NOTE: The networkIndicator element value and the cugInterlockBinaryCode element value are examples. Test case number SS_cug_005 Test case group SIP-SIP/Service/CUG Reference 4.5.2.10/ [13] SELECTION EXPRESSION SE 34 Test purpose Originating user no CUG to terminating user +IA. An originating user not in a CUG calls to a user in a CUG Incoming Access allowed. The session establishment is successful. Configuration Terminating user: CUG incoming access allowed SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Apply post test routine Comments Check: Is the session setup not rejected? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 184 Test case number SS_cug_006 Test case group SIP-SIP/Service/CUG Reference 4.5.2.10/ [13] SELECTION EXPRESSION [Network A] SE 34 AND NOT [Network B] SE 34 Test purpose Originating user no CUG to terminating user -IA. An originating user not in a CUG calls to a user in a CUG Incoming Access not allowed. The session establishment is not successful, a 403 (Forbidden) response is sent. Configuration User in Network B in a CUG incoming access not allowed SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 403 (Forbidden) ACK Comments Check: Is the session setup rejected? A 403 (Forbidden) final response is sent by the terminating network. Repeat this test in reverse direction. Test case number SS_cug_007 Test case group SIP-SIP/Service/CUG Reference 4.5.2.4/ [13] SELECTION EXPRESSION SE 34 Test purpose Originating user -OA, network B does not support CUG. An originating user in a CUG Outgoing Access not allowed calls to a user in network B. Network B does not support CUG. The session establishment is not successful, a 4xx unsuccessful final response is sent. Configuration SIP Parameter INVITE: Content-Type: application/vnd.etsi.cug+xml Content-Disposition: signal;handling= required ……. <…cug> <…networkIndicator>01</…networkIndicator <…networkIndicator>23</…networkIndicator <…cugInterlockBinaryCode>0F03</…cugInterlockBinaryCode> <…cugCommunicationIndicator>11</…cugCommunicationIndicator> <…cug> Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 4xx/501 Not Implemented ACK Comments Check: Is the Content-Type in The INVITE set to application/vnd.etsi.cug+xml? Check: Is the handling parameter in the Content-Disposition header set to required? Check: Contains the XML body in the INVITE a 'cug' element? Check: Contains the XML body in the INVITE a 'networkIndicator' element as a 'cug' child element? Check: Contains the XML body in the INVITE a 'cugInterlockBinaryCode' element as a 'cug' child element? Check: Contains the XML body in the INVITE a 'cugCommunicationIndicator' element set to '11' as a 'cug' child element? Check: Is the session setup rejected by sending an unsuccessful final response? Repeat this test in reverse direction. NOTE: The networkIndicator element value and the cugInterlockBinaryCode element value are examples. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 185 Test case number SS_cug_007A Test case group SIP-SIP/Service/CUG Reference 4.5.2.4/ [13] SELECTION EXPRESSION SE 34 Test purpose Originating user CUG-OA to terminating CUG user +ICB An originating user in a CUG outgoing access not allowed calls to a user in the same CUG Incoming communication barred. The session establishment is not successful, a 603 (Decline) response is sent. Configuration User in Network B in a CUG incoming Communication Barring SIP Parameter INVITE: Content-Type: application/vnd.etsi.cug+xml Content-Disposition: signal;handling= required ……. <…cug> <…networkIndicator>01</…networkIndicator <…networkIndicator>23</…networkIndicator <…cugInterlockBinaryCode>0F03</…cugInterlockBinaryCode> <…cugCommunicationIndicator>11</…cugCommunicationIndicator> <…cug> Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 603 Decline ACK Comments Check: Is the Content-Type in The INVITE set to application/vnd.etsi.cug+xml? Check: Is the handling parameter in the Content-Disposition header set to required? Check: Contains the XML body in the INVITE a 'cug' element? Check: Contains the XML body in the INVITE a 'networkIndicator' element as a 'cug' child element? Check: Contains the XML body in the INVITE a 'cugInterlockBinaryCode' element as a 'cug' child element? Check: Contains the XML body in the INVITE a 'cugCommunicationIndicator' element set to '11' as a 'cug' child element? Check: Is the session setup rejected by sending a 603 Decline unsuccessful final response? Repeat this test in reverse direction. NOTE: The networkIndicator element value and the cugInterlockBinaryCode element value are examples. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 186 Test case number SS_cug_008 Test case group SIP-SIP/Service/CUG Reference 7.1/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 58 Test purpose SIP-I/ISUP interworking. CUG call with outgoing access allowed. User A is located in the PSTN part of Network A and ISUP/BICC interworking applies in Network A. ensure that when user A is in a CUG 'outgoing access allowed' calls user B in Network B. The call is successful. There is a Optional forward call indicator the CUG Call Indicator Outgoing access allowed present in the encapsulated IAM sent to Network B. Configuration • User in PSTN/PLMN part of Network A in a CUG outgoing access allowed SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Optional Forward call indicator CUG Call Indicator Outgoing access allowed CUG interlock code --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Comments User A in the PSTN part of Network A calls user B in Network B Check: Is an IAM encapsulated in the INVITE request sent from Network A to Network B? Check: Is the Optional forward call indicator present, the CUG Call Indicator is set to 'Outgoing access allowed'? Check: Is the CUG interlock code parameter present in the encapsulated IAM? NOTE: CUG outgoing access allowed can appear like a basic call. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 187 Test case number SS_cug_009 Test case group SIP-SIP/Service/CUG Reference 7.1/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 58 Test purpose SIP-I/ISUP interworking. CUG call with outgoing access not allowed. User A is located in the PSTN part of Network A and ISUP/BICC interworking applies in Network A. ensure that when user A is in a CUG 'outgoing access allowed' calls user B in Network B. The call is successful. There is a Optional forward call indicator the CUG Call Indicator Outgoing access not allowed present in the encapsulated IAM sent to Network B. Configuration • User in PSTN/PLMN part of Network A in a CUG outgoing access not allowed SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Optional Forward call indicator CUG Call Indicator Outgoing access not allowed CUG interlock code --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Comments User A in the PSTN part of Network A calls user B in Network B Check: Is an IAM encapsulated in the INVITE request sent from Network A to Network B? Check: Is the Optional forward call indicator present, the CUG Call Indicator is set to 'Outgoing access not allowed'? Check: Is the CUG interlock code parameter present in the encapsulated IAM? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 188 Test case number SS_cug_010 Test case group SIP-SIP/Service/CUG Reference 7.1/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47 AND SE 58) AND ([Network B] SE 17 AND SE 47 AND SE 58) Test purpose SIP-I/ISUP interworking. CUG call with outgoing access not allowed (both user in the same CUG). User A in a CUG is located in the PSTN part of Network A and ISUP/BICC interworking applies in Network A. User B is located in the PSTN/PLMN part and SIP-I - ISUP/BICC interworking applies in the same CUG. Ensure that when user A is in a CUG 'outgoing access not allowed' calls user B in Network B. The call is successful. There is a Optional forward call indicator the CUG Call Indicator Outgoing access not allowed present in the encapsulated IAM sent to Network B. Configuration • User in PSTN/PLMN part of Network A in a CUG outgoing access not allowed • User in PSTN/PLMN part of Network B in a CUG • User A and User B are in the same CUG SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Optional Forward call indicator CUG Call Indicator Outgoing access not allowed CUG interlock code --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Comments User A in the PSTN part of Network A calls user B in the PST/PLMN part of Network B Check: Is an IAM encapsulated in the INVITE request sent from Network A to Network B? Check: Is the Optional forward call indicator present, the CUG Call Indicator is set to 'Outgoing access not allowed'? Check: Is the CUG interlock code parameter present in the encapsulated IAM? Check: Is the call setup successful? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 189 Test case number SS_cug_011 Test case group SIP-SIP/Service/CUG Reference 7.1/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47 AND SE 58) AND ([Network B] SE 17 AND SE 47 AND SE 58) Test purpose SIP-I/ISUP interworking. CUG call to a CUG user incoming access not allowed (both user in the same CUG). User A in a CUG is located in the PSTN part of Network A and ISUP/BICC interworking applies in Network A. User B is located in the PSTN/PLMN part and SIP-I - ISUP/BICC interworking applies in the same CUG. Ensure that when user A is in a CUG 'outgoing access not allowed' calls CUG user B in Network B. The call is successful. There is a Optional forward call indicator the CUG Call Indicator Outgoing access not allowed present in the encapsulated IAM sent to Network B. Configuration • User in PSTN/PLMN part of Network A in a CUG outgoing access not allowed • User in PSTN/PLMN part of Network B in a CUG incoming access not allowed • User A and User B are in the same CUG SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Optional Forward call indicator CUG Call Indicator Outgoing access not allowed CUG interlock code --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Comments User A in the PSTN/PLMN part of Network A calls user B in Network B User B in the PSTN/PLMN part of Network B. Check: Is an IAM encapsulated in the INVITE request sent from Network A to Network B? Check: Is the Optional forward call indicator present, the CUG Call Indicator is set to 'Outgoing access not allowed'? Check: Is the CUG interlock code parameter present in the encapsulated IAM? Check: Is the call setup successful? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 190 Test case number SS_cug_012 Test case group SIP-SIP/Service/CUG Reference 7.1/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47 AND SE 58) AND ([Network B] SE 17 AND SE 47 AND SE 58) Test purpose SIP-I/ISUP interworking. CUG call to a CUG user incoming access not allowed (both user in different CUG). User A in a CUG is located in the PSTN part of Network A and ISUP/BICC interworking applies in Network A. User B is located in the PSTN/PLMN part and SIP-I - ISUP/BICC interworking applies in different CUG. Ensure that when user A is in a CUG 'outgoing access not allowed' calls CUG user B in Network B. There is a Optional forward call indicator the CUG Call Indicator Outgoing access not allowed present in the encapsulated IAM sent to Network B. The call is rejected with a 500 (Server Internal error) final response. A ISUP/BICC REL is encapsulated and the Cause value is set to '87'. Configuration • User in PSTN/PLMN part of Network A in a CUG outgoing access not allowed • User in PSTN/PLMN part of Network B in a CUG incoming access not allowed • User A and User B are in different CUG SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Optional Forward call indicator CUG Call Indicator Outgoing access not allowed CUG interlock code --[any boundary name]-- 500 Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL Cause indicators Cause value 87 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 500 Server Internal error(REL) ACK Comments User A in the PSTN/PLMN part of Network A calls user B in Network B User B in the PSTN/PLMN part of Network B. Check: Is an IAM encapsulated in the INVITE request sent from Network A to Network B? Check: Is the Optional forward call indicator present, the CUG Call Indicator is set to 'Outgoing access not allowed'? Check: Is the CUG interlock code parameter present in the encapsulated IAM? Check: Is the call rejected with a 500 final response and a ISUP/BICC REL is encapsulated and the cause value is set to 87? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 191 Test case number SS_cug_013 Test case group SIP-SIP/Service/CUG Reference 7.1/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 58 Test purpose SIP-I/ISUP interworking. Call to a CUG user incoming access not allowed. User A is located in Network A. User B in a CUG Incoming access not allowed is located in the PSTN/PLMN part and SIP-I - ISUP/BICC interworking applies. Ensure that when user A calls user B in Network B. The call is rejected with a 500 (Server Internal error) final response. A ISUP/BICC REL is encapsulated and the Cause value is set to '87'. Configuration • User in PSTN/PLMN part of Network B in a CUG incoming access not allowed SIP Parameter 500 Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL Cause indicators Cause value 87 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 500 Server Internal error(REL) ACK Comments User A in Network A calls user B in Network B User B in the PSTN/PLMN part of Network B. Check: Is the call rejected with a 500 final response and a ISUP/BICC REL is encapsulated and the cause value is set to 87? Repeat this test in reverse direction. Test case number SS_cug_014 Test case group SIP-SIP/Service/CUG Reference 7.1/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 58 Test purpose SIP-I/ISUP interworking. Call to a CUG user incoming access allowed. User A is located in Network A. User B is located in the PSTN/PLMN part and SIP-I - ISUP/BICC interworking applied. Ensure that when user A calls CUG user B Incoming access allowed in Network B. The call is successful. Configuration • User in PSTN/PLMN part of Network B in a CUG incoming access allowed SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Comments User A in Network A calls user B in Network B User B in the PSTN/PLMN part of Network B. Check: Is the call setup successful? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 192
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7f110eeff44cca5c59b5fc967e08482a
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101 585
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7.1.5.10 Communication Waiting (CW)
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Test case number SS_cw_001 Test case group SIP-SIP/Service/CW Reference 4.5.5.2/ [15] SELECTION EXPRESSION SE 35 Test purpose Call Waiting indication in 180 response. User A is located in network A, user B is located in network B and subscribed to the communication Waiting service. Ensure that when user A calls user B, user A receives the 'communication Waiting indication' in the 180 Ringing provisional response if the user B is NDUB or UDUB. Configuration User B subscribed to the CW service SIP Parameter 180: Alert-Info: <urn:alert:service:call-waiting> Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Apply post test routine Comments Check: Is an Alert-Info header present in the 180 Ringing Response and is the value set to '<urn:alert:service:call-waiting>'? Repeat this test in reverse direction. Test case number SS_cw_002 Test case group SIP-SIP/Service/CW Reference 4.5.5.2/ [15] SELECTION EXPRESSION SE 35 AND SE 36 Test purpose Call rejected after timeout TAS-CW. User A is located in network A, user B is located in network B and subscribed to the communication Waiting service. Ensure that when user A calls user B, user A receives the 'communication Waiting indication' in the 180 Ringing provisional response if the user B is NDUB or UDUB. After timeout TAS-CW network B sends a 480 (Temporarily unavailable) response toward user A and the Reason header field is set to '19'. Configuration SIP Parameter 180: Alert-Info: <urn:alert:service:call-waiting> 480: Reason: Q.850 ;cause=19 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Timeout TAS-CW 480 (Temporarily unavailable) ACK Comments Check: Is an Alert-Info header present in the 180 Ringing Response and is the value set to '<urn:alert:service:call-waiting>'? Check: Is a Reason header present in the 480 Response and is the protocol is set to 'Q.850' and the cause parameter set to '19'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 193 Test case number SS_cw_003 Test case group SIP-SIP/Service/CW Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 59 Test purpose SIP-I support. Call Waiting indication in 180 with encapsulated ACM. User A is located in network A, user B is located in the PSTN/PLMN part of network B and subscribed to the Call Waiting service. Ensure that when user A calls user B, an encapsulated ISUP/BICC ACM Generic notification 'call is a waiting call' is present in the 180 Ringing provisional response if the user B is NDUB. Configuration User B subscribed to the CW service SIP Parameter 180 Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM Backward call indicator Called party's status indicator subscriber free Generic notification Notification indicator call is a waiting call Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Apply post test routine Comments Check: Is an ISUP/BICC ACM present in the 180 provisional response and the Generic notification is set to 'call is a waiting cal'? Repeat this test in reverse direction. Test case number SS_cw_004 Test case group SIP-SIP/Service/CW Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 59 Test purpose SIP-I support. Call Waiting indication in 180 with encapsulated CPG. User A is located in network A, user B is located in the PSTN/PLMN part of network B and subscribed to the Call Waiting service. Ensure that when user A calls user B, an encapsulated ISUP/BICC CPG Generic notification 'call is a waiting call' is present in the 180 Ringing provisional response if the user B is NDUB. Configuration User B subscribed to the CW service SIP Parameter 180 Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Event information Event indicator ALERTING Generic notification Notification indicator call is a waiting call Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 183 Session Progress (ACM) 180 Ringing (CPG) Apply post test routine Comments Check: Is an ISUP/BICC CPG present in the 180 provisional response and the Generic notification is set to 'call is a waiting cal'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 194
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7f110eeff44cca5c59b5fc967e08482a
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101 585
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7.1.5.11 Explicit Communication Transfer (ECT)
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Test case number SS_ect_001 Test case group SIP-SIP/Service/ECT Reference 4.5.2/ [11] SELECTION EXPRESSION [Network A] SE 37 AND [Network A] SE 11 AND [Network A] SE 49 Test purpose Blind/assured transfer using the REFER method. User A is located in network A, user B and user C are located in network B. User A invokes ECT to transfer a session with user B to user C. • Ensure that a REFER request is sent from network A to network B in the dialogue with user B. The URI in the Refer-To header is set to the address of the ECT AS in network A and the method parameter is set to 'INVITE'. • Ensure that an INVITE request is sent from network B to network A and the Request URI is set to the address of the ECT AS in network A. • Ensure that an INVITE request is sent from network A to network B and the Request URI is set to the address of user C. Configuration SIP Parameter REFER: Request URI address of user B Refer-To: <URI of ECT-AS>; method=invite INVITE1 Request URI address of ECT-AS INVITE2: Request URI address of user C Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session is established between user A and user B A confirmed session is established between user A and user C User A invokes ECT to transfer the session to user C REFER 202 Accepted NOTIFY (100) 200 OK NOTIFY CASE Blind transfer BYE (A-B) 200 OK BYE INVITE1 (ECT-AS) INVITE2 (user C) 200 OK INVITE ACK 200 OK INVITE ACK NOTIFY (200) 200 OK NOTIFY CASE Assured transfer BYE (A-B) 200 OK BYE Apply post test routine ETSI ETSI TS 101 585 V1.2.1 (2014-04) 195 Comments Check: Is a REFER request is sent network B, the Refer-To header is set to the URI of the ECT-AS in network A and a method parameter is present set to 'INVITE'? Check: Is a NOTIFY request sent to network A containing sipfrag body set to 'SIP/2.0 100 Trying' and if Blind transfer is applicable the session from user A to user B is terminated by user A? Check: Is an INVITE request sent to network A the Request line is set to the address of the ECT-AS in network A? Check: Is an INVITE request is sent to network B the Request is set to the address of user C? Check: When the session from user B to user C is confirmed a NOTIFY request is sent to network A containing sipfrag body set to 'SIP/2.0 200 OK' and if Assured transfer is applicable the session from user A to user B is terminated by user A? Check: Ensure the property of speech between user B and user C. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 196 Test case number SS_ect_002 Test case group SIP-SIP/Service/ECT Reference 4.5.2/ [11] SELECTION EXPRESSION [Network A] SE37 AND [Network A] SE 11 AND [Network A] SE 50 Test purpose Consultative transfer using the REFER method. User A is located in network A, user B and user C are located in network B. User A invokes ECT to transfer a session with user B to user C. • Ensure that a REFER request is sent from network A to network B in the dialogue with user B. The URI in the Refer-To header is set to the address of the ECT AS in network A and the method parameter is set to 'INVITE'. • Ensure that an INVITE request is sent from network B to network A and the Request URI is set to the address of the ECT AS in network A. • Ensure that an INVITE request is sent from network A to network B and the Request URI is set to the address of user C and a Replaces header is present containing the session identifiers of the session A - C. Configuration SIP Parameter REFER:Request URI address of user B Refer-To: <URI of ECT-AS>; method=invite INVITE1 Request URI address of ECT-AS INVITE2: Request URI address of user C Require: replaces Replaces: <session A-C> Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session is established between user A and user B A confirmed session is established between user A and user C User A invokes ECT to transfer the session to user C REFER 202 Accepted NOTIFY (100) 200 OK NOTIFY INVITE1 (ECT-AS) INVITE2 (user C) 200 OK INVITE ACK 200 OK INVITE ACK NOTIFY (200) 200 OK NOTIFY BYE (A-B) 200 OK BYE BYE (A-C) 200 OK BYE Apply post test routine Comments Check: Is a REFER request is sent network B, the Refer-To header is set to the URI of the ECT-AS in network A and a method parameter is present set to 'INVITE'? Check: Is an INVITE request sent to network A the Request line is set to the address of the ECT-AS in network A? Check: Is an INVITE request is sent to network B the Request is set to the address of user C and a Replaces header is present contains the session identifiers of the session A-C? Check: Is the session A - B and the session A - C terminated? Check: Ensure the property of speech between user B and user C. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 197 Test case number SS_ect_003 Test case group SIP-SIP/Service/ECT Reference 4.5.2/ [11], 4.7.2.9.7/ [20] SELECTION EXPRESSION [Network A] SE37 AND NOT [Network A] SE 12 AND [Network A] SE 49 Test purpose Blind/assured transfer using the 3pcc method. User A is located in network A, user B an user C are located in network B User A invokes ECT to transfer a session with user B to user C. • Ensure that the network A establishes a session to user C. • Ensure that the network A sends a reINVITE to update the session between user A and user B (SDP: IP address, port and codec). Configuration SIP Parameter INVITE1 Request URI address of user C INVITE2: Request URI address of user B SDP c=IN IP4/6 [new IP address] m=audio [new port] RTP/AVP [new codec list] a=[new attributes] Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session is established between user A and user B User A invokes ECT to transfer the session to user C INVITE1 (user C) 180 Ringing 200 OK INVITE ACK INVITE2 (user B) 200 OK INVITE ACK Apply post test routine Comments Check: Is an initial INVITE is sent from network A to user C to establish a dialogue between network A and user C? Check: Is a reINVITE is sent from network A to user B update the session parameter in the SDP? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 198 Test case number SS_ect_004 Test case group SIP-SIP/Service/ECT Reference 4.5.2/ [11], 4.7.2.9.7/ [20] SELECTION EXPRESSION [Network A] SE37 AND [Network A] SE 12 AND [Network A] SE 50 Test purpose Consultative transfer using the 3pcc method. User A is located in network A, user B and user C are located in network B User A invokes ECT to transfer a session with user B to user C. • Ensure that the network A sends a reINVITE to update the session between user A and user B (SDP: IP address, port and codec). • Ensure that the network A sends a reINVITE to update the session between user A and user C (SDP: IP address, port and codec). Configuration SIP Parameter INVITE1: Request URI address of user C SDP c=IN IP4/6 [new IP address] m=audio [new port] RTP/AVP [new codec list] a=[new attributes] INVITE2: Request URI address of user B SDP c=IN IP4/6 [new IP address] m=audio [new port] RTP/AVP [new codec list] a=[new attributes] Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session is established between user A and user B A confirmed session is established between user A and user C User A invokes ECT to transfer the session to user C INVITE1 (user B) 200 OK INVITE ACK INVITE2 (user C) 200 OK INVITE ACK Apply post test routine Comments Check: Is a reINVITE is sent from network A to user B update the session parameter in the SDP. Check: Is a reINVITE is sent from network A to user C update the session parameter in the SDP. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 199 Test case number SS_ect_005 Test case group SIP-SIP/Service/ECT Reference 5.4.3.2/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 60 Test purpose SIP-I support. Call Transfer invoked in active state, call was previous on HOLD. BICC/ISUP - SIP-I interworking applies in the originating network User A and C are located in network A and user B is located in network B. Ensure that an User A can successfully invoke the ECT supplementary service and transfer the call with User B to User C in active state. Configuration User A is subscribed to the Explicit Call Transfer supplementary service SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/sdp a=sendrecv --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required FAC Generic Notification Call transfer active Call transfer number --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session is established between user A and user B and set on hold User A invokes ECT to transfer the session to user C INFO (LOP request) 200 OK INFO INFO (LOP response) 200 OK INFO CASE A INVITE (sendrecv; FAC) 200 OK INVITE ACK CASE B INFO (FAC) 200 OK INFO INVITE (sendrecv) 200 OK INVITE ACK Apply post test routine Comments A session from User A to User B is already established User A sets the User B on hold User A invokes the ECT service Check: Is (optional) an INFO request is sent from Network A to Network B and an ISUP LOP message is present the Loop prevention indicator set to 'request'? Check: Is (optional) an INFO request is sent from Network A to Network B and an ISUP LOP message is present the Loop prevention indicator set to 'response'? Check: Is (CASE A) an INVITE request sent and an ISUP FAC message is present containing a Generic notification indicator is set to 'Call transfer active' and in addition the media stream is set to 'sendrecv'? Check: Is (CASE B) an INFO request sent and an ISUP FAC message is present containing a Generic notification indicator is set to 'Call transfer active'? In addition is an INVITE request sent and the media stream is set to 'sendrecv' to resume the held session? NOTE: The content of the FAC in the INVITE request is Equal to the content of the FAC in the INFO request. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 200 Test case number SS_ect_006 Test case group SIP-SIP/Service/ECT Reference 5.4.3.2/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 60 Test purpose SIP-I support. Call Transfer invoked in alerting state, call was previous on HOLD. BICC/ISUP - SIP-I interworking applies in the originating network User A and C are located in network A and user B is located in network B. Ensure that an User A can successfully invoke the ECT supplementary service and transfer the call with User B to User C in alerting state. Configuration User A is subscribed to the Explicit Call Transfer supplementary service SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/sdp a=sendrecv --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required FAC Generic Notification Call transfer alerting Call transfer number --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session is established between user A and user B and set on hold User A invokes ECT to transfer the session to user C INFO (LOP request) 200 OK INFO INFO (LOP response) 200 OK INFO CASE A INVITE (sendrecv; FAC) 200 OK INVITE ACK CASE B INFO (FAC) 200 OK INFO INVITE (sendrecv) 200 OK INVITE ACK Apply post test routine Comments A session from User A to User B is already established User A sets the User B on hold A session from User A to User C is already established User A invokes the ECT service Check: Is (optional) an INFO request is sent from Network A to Network B and an ISUP LOP message is present the Loop prevention indicator set to 'request'? Check: Is (optional) an INFO request is sent from Network A to Network B and an ISUP LOP message is present the Loop prevention indicator set to 'response'? Check: Is (CASE A) an INVITE request sent and an ISUP FAC message is present containing a Generic notification indicator is set to 'Call transfer alerting' and in addition the media stream is set to 'sendrecv'? Check: Is (CASE B) an INFO request sent and an ISUP FAC message is present containing a Generic notification indicator is set to 'Call transfer alerting'? In addition is an INVITE request sent and the media stream is set to 'sendrecv' to resume the held session? NOTE: The content of the FAC in the INVITE request is Equal to the content of the FAC in the INFO request. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 201 Test case number SS_ect_007 Test case group SIP-SIP/Service/ECT Reference 5.4.3.2/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 60 Test purpose SIP-I support. Call Transfer invoked in active state. BICC/ISUP - SIP-I interworking applies in the originating network Users A and B are located in network A and User C is located in network B. Ensure that an User A can successfully invoke the ECT supplementary service and transfer the call with User B to User C in active state. Configuration User A is subscribed to the Explicit Call Transfer supplementary service SIP Parameter INFO Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required FAC Generic Notification Call transfer active Call transfer number Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session is established between user A and user C User A invokes ECT to transfer the session to user C INFO (LOP request) 200 OK INFO INFO (LOP response) 200 OK INFO INFO (FAC) 200 OK INFO Apply post test routine Comments A session from User A to User B is already established User A sets the User B on hold A session from User A to User C is already established User A invokes the ECT service Check: Is (optional) an INFO request is sent from Network A to Network B and an ISUP LOP message is present the Loop prevention indicator set to 'request'? Check: Is (optional) an INFO request is sent from Network A to Network B and an ISUP LOP message is present the Loop prevention indicator set to 'response'? Check: Is (CASE B) an INFO request sent and an ISUP FAC message is present containing a Generic notification indicator is set to 'Call transfer active'? NOTE: The content of the FAC in the INVITE request is Equal to the content of the FAC in the INFO request. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 202 Test case number SS_ect_008 Test case group SIP-SIP/Service/ECT Reference 5.4.3.2/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 60 Test purpose SIP-I support. Call Transfer invoked in alerting state. BICC/ISUP - SIP-I interworking applies in the originating network User A and B are located in network A and user C is located in network B. Ensure that an User A can successfully invoke the ECT supplementary service and transfer the call with User B to User C in alerting state. Configuration User A is subscribed to the Explicit Call Transfer supplementary service SIP Parameter INFO Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required CPG Generic Notification Call transfer alerting Call transfer number Message flow SIP (Network A) Interconnection Interface SIP (Network B) A session in the early dialogue is established between user A and user C User A invokes ECT to transfer the session to user C INFO (LOP request) 200 OK INFO INFO (LOP response) 200 OK INFO INFO (CPG) 200 OK INFO Apply post test routine Comments A session from User A to User B is already established User A sets the User B on hold A session from User A to User C is already established User A invokes the ECT service Check: Is (optional) an INFO request is sent from Network A to Network B and an ISUP LOP message is present the Loop prevention indicator set to 'request'? Check: Is (optional) an INFO request is sent from Network A to Network B and an ISUP LOP message is present the Loop prevention indicator set to 'response'? Check: Is (CASE B) an INFO request sent and an ISUP CPG message is present containing a Generic notification indicator is set to 'Call transfer alerting'? NOTE: The content of the FAC in the INVITE request is Equal to the content of the FAC in the INFO request. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 203
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7f110eeff44cca5c59b5fc967e08482a
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101 585
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7.1.5.12 Malicious Communication Identification (MCID)
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Test case number SS_mcid_001 Test case group SIP-SIP/Service/MCID Reference 4.5.2.5/ [18] SELECTION EXPRESSION SE 38 Test purpose Network B sends a MCID request, no response. User A is located in network A, user B is located in network B and subscribed to the Malicious Communication Identification service. When user A call user B and no originating identification is present in the INVITE request, the network B sends an INFO request to network A requesting the originating identity. After timeout of timer TO-ID the network B sends the 180 Ringing response. Configuration User B is subscribed to the MCID service SIP Parameter INFO: <…:mcid……………....> <…:request> <…:McidRequestIndicator>01</…:McidRequestIndicator> <…:HoldingIndicator >…</…:HoldingIndicator> </…:request> </…:mcid> Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE INFO 200 OK INFO Timeout TO-ID 180 Ringing Apply post test routine Comments Check: Is an INFO request sent to network A? Check: Is the McidRequestIndicator element set to ‚01'? Check: is a 200 OK INFO response sent to network B? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 204 Test case number SS_mcid_002 Test case group SIP-SIP/Service/MCID Reference 4.5.2.5/ [18] SELECTION EXPRESSION SE 38 AND SE 47 Test purpose Network B sends a MCID request, MCID response. PSTN user A is located in network A, user B is located in network B and subscribed to the Malicious Communication Identification service. When user A call user B and no originating identification is present in the INVITE request, the network B sends an INFO request to network B requesting the originating identity. After receipt of an INFO request from network A the network B sends the 180 Ringing response. Configuration User B subscribed to the MCID service User A is a ISDN or POTS user in the PSTN of network A SIP Parameter INFO: <…:mcid ……………….....> <…:request> <…:McidRequestIndicator>01</…:McidRequestIndicator> <…:HoldingIndicator >…</…:HoldingIndicator> </…:request> </…:mcid> INFO: <…:mcid…………….......> <…:response> <…:McidResponseIndicator>01</…:McidResponseIndicator> <…:HoldingProvidedIndicator>…</…:HoldingProvidedIndicator> <…:OrigPartyIdentity>any URI</…:OrigPartyIdentity> <…:OrigPartyPresentationRestriction> true/false </…:OrigPartyPresentationRestriction> </…:response> </…:mcid> Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE INFO 200 OK INFO INFO 200 OK INFO 180 Ringing Apply post test routine Comments Check: Is an INFO request sent to network A? Check: Is the McidRequestIndicator element set to ‚01'? Check: Is a 200 OK INFO response sent to network B? Check: Is an INFO request sent to network B? Check: Is the McidResponseIndicator element set to ‚01'? Check: Is the OrigPartyIdentity element present in the response element? Check: Is a 200 OK INFO response sent to network A? A INFO request containing a mcid response element sent by the MGCF in network A is optional. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 205 Test case number SS_mcid_003 Test case group SIP-SIP/Service/MCID Reference 5.4.3.2/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 61 Test purpose SIP-I support. Network B sends a MCID request, no response. User A is located in network A, user B is located in the PSTN/PLMN part of network B and subscribed to the Malicious Call Identification service. When user A call user B and no originating identification is present in the INVITE request, the network B sends a 183 Session Progress to network A and an ISUP/BICC IDR message is present the MCID request indicator is set to 'MCID requested' requesting the originating identity. After timeout of timer (ISUP) T39 the network B sends the 180 Ringing response. Configuration User B is subscribed to the MCID service SIP Parameter INFO: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IDR MCID request indicators MCID request indicator MCID requested Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 183 Session Progress(IDR) Timeout TO-ID 180 Ringing Apply post test routine Comments Check: Is an INFO request sent to network A? Check: Is a ISUP/BICC IDR message is present and the MCID request indicator is set to 'MCID requested'? Check: Is a 200 OK INFO response sent to network B? NOTE: Based on network policies the MCID request indicator can be set to'MCID not requested'. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 206 Test case number SS_mcid_004 Test case group SIP-SIP/Service/MCID Reference 5.4.3.2/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 61 Test purpose SIP-I support. Network B sends a MCID request, MCID response. PSTN user A is located in network A, user B is located in the PSTN/PLMN part of network B and SIP-I - ISUP/BICC interworking applies and User B is subscribed to the Malicious Call Identification service. When user A call user B and no originating identification is present in the INVITE request, the network B sends a INFO request to network A requesting the originating identity. After receipt of an INFO request from network A the network B sends the 180 Ringing response. Configuration User B subscribed to the MCID service User A is a ISDN or POTS user in the PSTN of network A SIP Parameter INFO: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IDR MCID request indicators MCID request indicator MCID requested INFO: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IRS MCID response indicators MCID response indicator MCID included Calling party number Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE INFO(IDR) 200 OK INFO INFO(IRS) 200 OK INFO 180 Ringing Apply post test routine Comments Check: Is an INFO request sent to network A and a ISUP/BICC IDR is present and the MCID request indicator is set to 'MCID requested'? Check: Is a 200 OK INFO response sent to network B? Check: Is an INFO request sent to network B and a ISUP/BICC IRS is present and the MCID response indicator is set to 'MCID included'? Check: Is the Calling party number present in the attached ISUP/BICC IRS? Check: Is a 200 OK INFO response sent to network A? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 207
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7f110eeff44cca5c59b5fc967e08482a
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101 585
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7.1.5.13 Message Waiting Indication (MWI)
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Test case number SS_mwi_001 Test case group SIP-SIP/Service/MWI Reference 4.7.2/ [16] SELECTION EXPRESSION [Network A] SE 39 AND [Network B] SE 39 Test purpose Initial subscription of a Voicemail box. The Voicemail owner is in network A, his Voicemail box is located in network B. Ensure that a Voicemail owner is able to activate his Voicemail box. Configuration Voicemail in network B Voicemail owner in network A SIP Parameter SUBCRIBE Event: message-summary Expires: [any value] Accept: application/simple-message-summary NOTIFY Subscription-State: active;expires=[any value] Event: message-summary Message flow SIP (Network A) Interconnection Interface SIP (Network B) SUBCRIBE 200 OK SUBSCRIBE NOTIFY 200 OK NOTIFY 200 OK BYE NOTIFY 200 OK NOTIFY Apply post test routine Comments Check: Is it possible for a user in network A to subscribe to a Voicemail box in network B? Check: Is the Event header in the SUBCRIBE set to 'message-summary'? Check: Is the Accept header in the SUBCRIBE set to 'application/simple- message-summary'? Check: Is the Event header in the NOTIFY is set to 'message-summary'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 208 Test case number SS_mwi_002 Test case group SIP-SIP/Service/MWI Reference 4.7.2/ [16] SELECTION EXPRESSION [Network A] SE 39 AND [Network B] SE 39 Test purpose A new entry in the Voicemail box is indicated to the owner. The Voicemail owner is in network A, his Voicemail box is located in network B. Ensure when a user calls user A and the call is not answered, the call is forwarded to the Voicemail box of user A in network B. Ensure that the user A is notified by message waiting indication that there is a new message present in his voicemail account. Configuration Voicemail in network B Voicemail owner in network A SIP Parameter NOTIFY Subscription-State: active;expires=[any value] Event: message-summary Content-Type: application/simple-message-summary Messages-Waiting: yes Message-Account: sip:userA@networkA (optional) Voice-Message: [any new value]/[any old value] (optional) Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 200 OK INVITE ACK BYE 200 OK BYE NOTIFY 200 OK NOTIFY Apply post test routine Comments Check: Is the Event header in the NOTIFY set to 'message-summary'? Check: Is the Content-Type header in the NOTIFY set to 'application/simple- message-summary'? Check: Contains the MIME body the header 'Messages-Waiting' set to 'yes'? Check: Contains the MIME body the optional header 'Message-Account'? Check: Contains the MIME body the optional header 'Voice-Message'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 209 7.1.5.14 Completion of Communications to Busy Subscriber (CCBS), Completion of Communications by No Reply (CCNR) Test case number SS_cc_001 Test case group SIP-SIP/Service/CC Reference 4.5.4.3/ [14] SELECTION EXPRESSION [Network A] SE 40 AND [Network B] SE 40 Test purpose Indicating of CCBS possible. User A is located in network A and user B is located in network B. Ensure when user A calls user B and user B is busy, the network B sends an indication that CCBS is possible in the 486 Busy Here final response. Configuration SIP Parameter 486: Call-Info: <sip:UE-B>;purpose=call-completion;m=BS Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 486 Busy Here ACK Comments Check: The 486 final response contains the Call-Info header. Check: The Call-Info header contains the URI of user B as the monitor point in network B. Check: The Call-Info header contains the purpose parameter set to 'call-completion' and the m parameter set to 'BS'. Repeat this test in reverse direction. Test case number SS_cc_002 Test case group SIP-SIP/Service/CC Reference 4.5.4.3/ [14] SELECTION EXPRESSION [Network A] SE 41 AND [Network B] SE 41 Test purpose Indicating of CCNR possible. User A is located in network A and user B is located in network B. Ensure when user A calls user B and user B is free, the network B sends an indication that CCNR is possible in the 180 Ringing provisional response. Configuration SIP Parameter 180: Call-Info: <sip:UE-B>;purpose=call-completion;m=NR Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing Apply post test routine Comments Check: The 180 provisional response contains the Call-Info header. Check: The Call-Info header contains the URI of user B as the monitor point in network B. Check: The Call-Info header contains the purpose parameter set to 'call-completion' and the m parameter set to 'NR'. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 210 Test case number SS_cc_003 Test case group SIP-SIP/Service/CC Reference 4.5.4.2/ [14] SELECTION EXPRESSION ([Network A] SE 40 OR [Network A] SE 41) AND ([Network B] SE 40 OR [Network B] SE 41) Test purpose Invocation of CCBS or CCNR. User A is located in network A and user B is located in network B. • Ensure when user A call user B and user B is busy, the indication that CCBS is possible is sent to the network A. when user A invokes CCBS, a SUBSCRIBE request is sent to the network B, the Event header is set to 'call-completion' and the m parameter in the Request line is set to 'BS'. • Ensure when user A call user B and user B is free, the indication that CCNR is possible is sent to the network A. when user A invokes CCNR, a SUBSCRIBE request is sent to the network B, the Event header is set to 'call-completion' and the m parameter in the Request line is set to 'NR'. Ensure that the network B sends a NOTIFY request to network A to confirm that the request is in the Call completion queue at the terminating Application Server. Configuration SIP Parameter SUBSRIBE sip:B-AS;m=BS or m=NR From:<UE-A> To:<UE-B> Contact:<A-AS> Event:call-completion NOTIFY sip:A-AS Event:call-completion Content-Type: application/call-completion state: queued service-retention Message flow SIP (Network A) Interconnection Interface SIP (Network B) An indication whether CCBS or CCNR is possible is sent by network B SUBSCRIBE 202 Accepted NOTIFY 200 OK NOTIFY Apply post test routine Comments Check: Is a SUBCRIBE request is sent to network B? Check: Is the m parameter in the Request URI is set to 'BS' in case of CCBS request or set to 'NR' in case of CCNR? Check: Is a NOTIFY request is sent to network A and the Event header is set to 'call-completion' and the state header in the message body is set to 'queued''. Repeat this test in reverse direction. NOTE: The service-retention header in the NOTIFY body is a network option. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 211 Test case number SS_cc_004 Test case group SIP-SIP/Service/CC Reference 4.5.4.3/ [14] SELECTION EXPRESSION ([Network A] SE 40 OR [Network A] SE 41) AND ([Network B] SE 40 OR [Network B] SE 41) Test purpose Invocation of CCBS or CCNR unsuccessful; short term denial User A is located in network A and user B is located in network B. Ensure that user A invokes a CCBS or CCNR request to network B and the network B is currently unable to process the request (e.g. the B-queue is full), a 480 Temporarily Unavailable final response is sent. Configuration SIP Parameter SUBSRIBE sip:B-AS;m=BS or m=NR From:<UE-A> To:<UE-B> Contact:<A-AS> Event:call-completion Message flow SIP (Network A) Interconnection Interface SIP (Network B) An indication whether CCBS or CCNR is possible is sent by network B SUBSCRIBE 480 (Temporarily Unavailable) Comments Check: Is a SUBCRIBE request is sent to network B? Check: Is the m parameter in the Request URI is set to 'BS' in case of CCBS request or set to 'NR' in case of CCNR? Check: Is a 480 Temporarily Unavailable sent from network B indicates the CCBS or CCNR request is unsuccessful e.g. CC queue is full? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 212 Test case number SS_cc_005 Test case group SIP-SIP/Service/CC Reference 4.5.4.3/ [14] SELECTION EXPRESSION ([Network A] SE 40 OR [Network A] SE 41) AND ([Network B] SE 40 OR [Network B] SE 41) Test purpose Successful CC operation User A is located in network A and user B is located in network B. User A has successfully invoked a CCBS or CCNR request. • Ensure when the user B becomes available for CC recall, the CC recall procedure is started. The network B sends a NOTIFY request to network A and a state header is present in the message body set to 'ready'. • Ensure that the recall from user A to user B is successful. • Ensure that a CC revocation notification is dent to network A to indicate the subscription is terminated; the reason header is set to 'noresource'. Configuration SIP Parameter NOTIFY sip:O-AS Event:call-completion Content-Type: application/call-completion state: ready NOTIFY sip:O-AS Event:call-completion Subscription-State: terminated; reason=noresource Message flow SIP (Network A) Interconnection Interface SIP (Network B) A CCBS or CCNR request was already successful NOTIFY 200 OK NOTIFY INVITE 180 Ringing NOTIFY 200 OK NOTIFY 200 OK INVITE ACK Apply post test routine Comments Check: Is a NOTIFY request is sent to network A and the Event header is set to 'call-completion' and the state header in the message body is set to 'ready'? Check: Is the recall from user A to user B is successful? Check: Is the CC revocation is performed after the 180 Ringing or the 200 OK INVITE was sent to user A? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 213 Test case number SS_cc_006 Test case group SIP-SIP/Service/CC Reference 4.5.4.31/ [14] SELECTION EXPRESSION ([Network A] SE 40 OR [Network A] SE 41) AND ([Network B] SE 40 OR [Network B] SE 41) Test purpose No CC call as result. User A is located in network A and user B is located in network B. User A has successfully invoked a CCBS or CCNR request. Ensure when no recall result is performed while CC-T9 is running (user A does not calls to user B) the network B sends a NOTIFY request to network A with an indication that the subscription is terminated, the reason header is set to 'rejected'. Configuration SIP Parameter NOTIFY sip:O-AS Event:call-completion Content-Type: application/call-completion state: ready NOTIFY sip:O-AS Event:call-completion Subscription-State: terminated; reason=rejected Message flow SIP (Network A) Interconnection Interface SIP (Network B) A CCBS or CCNR request was already successful User B is available for recall NOTIFY 200 OK NOTIFY CC-T9 expires NOTIFY 200 OK NOTIFY Comments Check: Is a NOTIFY request is sent to network A and the Event header is set to 'call-completion' and the state header in the message body is set to 'ready'? User A does not perform the recall Check: Is the CC revocation is performed after timer CC-T9 expires? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 214 Test case number SS_cc_007 Test case group SIP-SIP/Service/CC Reference 4.5.4.2/ [14] SELECTION EXPRESSION ([Network A] SE 40 OR [Network A] SE 41) AND ([Network B] SE 40 OR [Network B] SE 41) Test purpose User A is unavailable while CC recall is performed. User A is located in network A and user B is located in network B. User A has successfully invoked a CCBS or CCNR request. User B is available for CC-recall and network B sends a CC-recall notification to network A. • Ensure that network A sends PUBLISH request to suspend the recall procedure. • Ensure that network A sends PUBLISH request to resume the recall procedure if user A is available to complete the recall procedure. • Ensure the network B sends a NOTIFY request to indicate the CC-recall procedure. Configuration SIP Parameter NOTIFY sip:O-AS Event:call-completion Content-Type: application/call-completion state: ready PUBLISH sip B-AS To: SIP 2 Event: presence Content-Type: application/pidf+xml <?xml version="1.0" encoding="UTF-8"?> <presence <status> <basic>closed</basic> PUBLISH sip B-AS To: SIP 2 Event: presence Content-Type: application/pidf+xml <?xml version="1.0" encoding="UTF-8"?> <presence <status> <basic>open</basic> Message flow SIP (Network A) Interconnection Interface SIP (Network B) A CCBS or CCNR request was already successful User B is available for recall NOTIFY 200 OK NOTIFY User A is busy PUBLISH 200 OK PUBLISH User A is no longer busy PUBLISH 200 OK PUBLISH User B is available for recall NOTIFY 200 OK NOTIFY Apply post test routine Comments ETSI ETSI TS 101 585 V1.2.1 (2014-04) 215 Test case number SS_cc_008 Test case group SIP-SIP/Service/CC Reference 6.11.2/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 Test purpose SIP-I support: Indicating of CCBS possible. BICC/ISUP - SIP-I interworking applies in the terminating network and User A is located in network A and user B is located in network B. Ensure when user A calls user B and user B is busy, the network B sends a 486 Busy Here final response and an encapsulated ISUP REL is present, the Cause value indicator is set to #17 or #34 and the CCBS possible indicator is set to 'CCBS possible'. Configuration SIP Parameter 486: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL Cause value #17 or #34 Diagnostics CCBS possible Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 486 Busy Here (REL) ACK Comments Check: The 486 final response contains an encapsulated BICC/ISUP REL, the Cause value set to 17 or 34 and the Diagnostics set to 'CCBS possible'. Repeat this test in reverse direction. Test case number SS_cc_009 Test case group SIP-SIP/Service/CC Reference 6.5/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 Test purpose SIP-I support: Indicating of CCNR possible. BICC/ISUP - SIP-I interworking applies in the terminating network User A is located in network A and user B is located in network B. Ensure when user A calls user B and user B is free, the network B sends a 180 Ringing provisional response and an encapsulated ACM is present containing a CCNR possible indicator set to 'CCNR possible'. Configuration SIP Parameter 180: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM CCNR possible indicator CCNR possible Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing (ACM) Apply post test routine Comments Check: The 180 provisional response contains an encapsulated ACM. Check: The CCNR possible indicator in the ACM is set to 'CCNR possible'. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 216
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101 585
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7.1.6 Other PSTN services (SIP-I interworking)
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7f110eeff44cca5c59b5fc967e08482a
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101 585
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7.1.6.1 User-to-User Signalling (UUS)
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Test case number SS_uus_001 Test case group SIP-SIP/SIP-I/UUS Reference 7.1/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 1 implicit in initial INVITE request. BICC/ISUP - SIP-I interworking applies in the originating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 1 implicit request calls user B and User-to-user Information parameter is present in the encapsulated IAM of the initial INVTE request. Configuration User A is subscribed to the User-to-User service 1 implicit request SIP Parameter INVITE: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM User-to-user Information User Information Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (IAM) Apply post test routine Comments Check: Is an ISUP/BICC IAM encapsulated in the initial INVITE request? Check: Is a User-to-user Information parameter present in the encapsulated ISUP/BICC IAM? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 217 Test case number SS_uus_002 Test case group SIP-SIP/SIP-I/UUS Reference 7.1, 6.5/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47) AND ([Network B] SE 17 AND SE 47) AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 1 implicit response in 180. BICC/ISUP - SIP-I interworking applies in the originating and terminating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 1 implicit request calls user B subscribed to User-to-User service 1 an User-to-user Information parameter is present in the encapsulated ACM of the 180 response. Configuration User A is subscribed to the User-to-User service 1 implicit request SIP Parameter INVITE: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM User-to-user Information User Information 180 Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM User-to-user Information User Information Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (IAM) 180 Ringing (ACM) Apply post test routine Comments Check: Is an ISUP/BICC IAM encapsulated in the initial INVITE request? Check: Is a User-to-user Information parameter present in the encapsulated ISUP/BICC IAM? Check: Is an ISUP/BICC ACM encapsulated in the 180 response? Check: Is a User-to-user Information parameter present in the encapsulated ISUP/BICC ACM? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 218 Test case number SS_uus_003 Test case group SIP-SIP/SIP-I/UUS Reference 7.1/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 1 explicit in initial INVITE request. BICC/ISUP - SIP-I interworking applies in the originating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 1 explicit request calls user B an User-to-user Indicator parameter is present set to 'Request service 1', 'not essential' or 'essential' in the encapsulated IAM of the initial INVTE request. Configuration User A is subscribed to the User-to-User service 1 explicit request SIP Parameter INVITE: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM User-to-user Indicator Request service 1 not essential or essential User-to-user Information User Information Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (IAM) Apply post test routine Comments Check: Is an ISUP/BICC IAM encapsulated in the initial INVITE request? Check: Is a User-to-user Indicator parameter present in the encapsulated ISUP/BICC IAM? Check: Is the Request service 1 set to 'not essential' or 'essential'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 219 Test case number SS_uus_004 Test case group SIP-SIP/SIP-I/UUS Reference 7.1, 6.5/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47) AND ([Network B] SE 17 AND SE 47) AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 1 explicit response in 180. BICC/ISUP - SIP-I interworking applies in the originating and terminating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 1 explicit request calls user B subscribed to User-to-User service 1 an User-to-user Indicator parameter is present set to 'Response', 'service 1 provided' in the encapsulated ACM of the 180 response. Configuration User A is subscribed to the User-to-User service 1 explicit request SIP Parameter INVITE: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM User-to-user Indicator Request service 1 essential or not essential 180 Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM User-to-user Indicator Response service 1 provided Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (IAM) 180 Ringing (ACM) Apply post test routine Comments Check: Is an ISUP/BICC IAM encapsulated in the initial INVITE request? Check: Is a User-to-user Information parameter present in the encapsulated ISUP/BICC IAM? Check: Is an ISUP/BICC ACM encapsulated in the 180 response? Check: Is an User-to-user Indicator parameter present set to 'Response', 'service 1 provided' in the encapsulated ISUP/BICC ACM? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 220 Test case number SS_uus_005 Test case group SIP-SIP/SIP-I/UUS Reference 7.1, 6.5/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47) AND ([Network B] SE 17 AND SE 47) AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 1 not essential explicit rejected in 180. BICC/ISUP - SIP-I interworking applies in the originating and terminating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 1 explicit request calls user B not subscribed to User-to-User service 1 the call is rejected by the network an User-to-user Indicator parameter is present set to 'Response', 'service 1 not provided' in the encapsulated ACM of the 180 response. Configuration User A is subscribed to the User-to-User service 1 explicit request User B is not subscribed to the User-to-User service 1 SIP Parameter INVITE: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM User-to-user Indicator Request service 1 not essential 180 Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM User-to-user Indicator Response service 1 not provided Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (IAM) 180 Ringing (ACM) Apply post test routine Comments Check: Is an ISUP/BICC IAM encapsulated in the initial INVITE request? Check: Is a User-to-user Information parameter present in the encapsulated ISUP/BICC IAM? Check: Is an ISUP/BICC ACM encapsulated in the 180 response? Check: Is an User-to-user Indicator parameter present set to 'Response', 'service 1 not provided' in the encapsulated ISUP/BICC ACM? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 221 Test case number SS_uus_006 Test case group SIP-SIP/SIP-I/UUS Reference 6.11.2, 7.1/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47) AND ([Network B] SE 17 AND SE 47) AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 1 essential explicit rejection. BICC/ISUP - SIP-I interworking applies in the originating and terminating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 1 explicit request calls user B subscribed to User-to-User service 1 essential is rejected by the network or by the user. A 500 Server Internal Error is sent and an encapsulated ISUP/BICC REL is present, the Cause value is set to #29 or #69. Configuration User A is subscribed to the User-to-User service 1 explicit request SIP Parameter INVITE: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM User-to-user Indicator Request service 1 essential 500 Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL Cause value #29 or #69 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (IAM) 500 Server Internal Error (REL) ACK Apply post test routine Comments Check: Is an ISUP/BICC IAM encapsulated in the initial INVITE request? Check: Is a User-to-user Indicator parameter present in the encapsulated ISUP/BICC IAM set to 'Request', 'service 1', 'essential'? Check: Is an ISUP/BICC REL encapsulated in the 500 response? Check: Is the Cause value set to #29 or #69 in the encapsulated REL? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 222 Test case number SS_uus_007 Test case group SIP-SIP/SIP-I/UUS Reference 7.1/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 2 in initial INVITE request. BICC/ISUP - SIP-I interworking applies in the originating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 2 calls user B an User-to-user Indicator parameter is present set to 'Request service 2', 'not essential' or 'essential' in the encapsulated IAM of the initial INVTE request. Configuration User A is subscribed to the User-to-User service 2 SIP Parameter INVITE: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM User-to-user Indicator Request service 2 not essential or 'essential' Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (IAM) Apply post test routine Comments Check: Is an ISUP/BICC IAM encapsulated in the initial INVITE request containing a User-to-user Indicator parameter, and the indicator Request service 2 is set to the value 'not essential' or 'essential'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 223 Test case number SS_uus_008 Test case group SIP-SIP/SIP-I/UUS Reference 5.4.3.2, 6.5, 7.1/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47) AND ([Network B] SE 17 AND SE 47) AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 2 in initial INVITE request successful. BICC/ISUP - SIP-I interworking applies in the originating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 2 calls user B an User-to-user Indicator parameter is present set to 'Request service 2', 'not essential' or 'essential' in the encapsulated IAM of the initial INVTE request. The User-to-User service is successful. Configuration User A is subscribed to the User-to-User service 2 User B is subscribed to the User-to-User service 2 SIP Parameter INVITE: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM User-to-user Indicator Request service 2 not essential or 'essential' 180 Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM User-to-user Indicator Response service 2 provided INFO Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required USR User-to-user Information User Information 183 Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required USR User-to-user Information User Information Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (IAM) 180 Ringing (ACM) INFO (USR) 200 OK INFO 183 Session Progress (USR) Apply post test routine Comments Check: Is an ISUP/BICC IAM encapsulated in the initial INVITE request containing a User-to-user Indicator parameter, and the indicator Request service 2 is set to the value 'not essential' or 'essential'? Check: Is an ISUP/BICC ACM encapsulated in the 180 and the User-to-user Indicator parameter is set to 'Response', 'service 2 provided'? Check: Is an ISUP/BICC USR encapsulated in the INFO message sent from network A to network B containing an User-to-user Information parameter? Check: Is an ISUP/BICC USR encapsulated in the 183 response sent from network B to network A containing an User-to-user Information parameter? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 224 Test case number SS_uus_009 Test case group SIP-SIP/SIP-I/UUS Reference 7.1, 6.5/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47) AND ([Network B] SE 17 AND SE 47) AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 2 not essential rejected in 180 response. BICC/ISUP - SIP-I interworking applies in the originating and terminating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 2 not essential calls user B not subscribed to User-to-User service 2 the call is rejected by the network an User-to-user Indicator parameter is present set to 'Response', 'service 2 not provided' in the encapsulated ACM of the 180 response. Configuration User A is subscribed to the User-to-User service 2 User B is not subscribed to the User-to-User service 2 SIP Parameter INVITE: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM User-to-user Indicator Request service 2 not essential 180 Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ACM User-to-user Indicator Response service 2 not provided Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (IAM) 180 Ringing (ACM) Apply post test routine Comments Check: Is an ISUP/BICC IAM encapsulated in the initial INVITE request? Check: Is a User-to-user Information parameter present in the encapsulated ISUP/BICC IAM set to 'Request', 'service 2' 'not essential'? Check: Is an ISUP/BICC ACM encapsulated in the 180 response? Check: Is an User-to-user Indicator parameter present set to 'Response', 'service 2 not provided' in the encapsulated ISUP/BICC ACM? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 225 Test case number SS_uus_010 Test case group SIP-SIP/SIP-I/UUS Reference 6.11.2, 7.1/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47) AND ([Network B] SE 17 AND SE 47) AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 2 essential rejection. BICC/ISUP - SIP-I interworking applies in the originating and terminating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 2 essential calls user B not subscribed to User-to-User service 2 the call is rejected by the network. A 500 Server Internal Error is sent and an encapsulated ISUP/BICC REL is present, the Cause value is set to #29 or #69. Configuration User A is subscribed to the User-to-User service 2 SIP Parameter INVITE: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM User-to-user Indicator Request service 2 essential 500 Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL Cause value #29 or #69 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (IAM) 500 Server Internal Error (REL) ACK Apply post test routine Comments Check: Is an ISUP/BICC IAM encapsulated in the initial INVITE request? Check: Is a User-to-user Indicator parameter present in the encapsulated ISUP/BICC IAM set to 'Request', 'service 1', 'essential'? Check: Is an ISUP/BICC REL encapsulated in the 500 response? Check: Is the Cause value set to #29 or #69 in the encapsulated REL? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 226 Test case number SS_uus_011 Test case group SIP-SIP/SIP-I/UUS Reference 7.1/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 3 in initial INVITE request. BICC/ISUP - SIP-I interworking applies in the originating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 3 calls user B an User-to-user Indicator parameter is present set to 'Request service 3', 'not essential' or 'essential' in the encapsulated IAM of the initial INVTE request. Configuration User A is subscribed to the User-to-User service 3 SIP Parameter INVITE: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM User-to-user Indicator Request service 3 not essential or 'essential' Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (IAM) Apply post test routine Comments Check: Is an ISUP/BICC IAM encapsulated in the initial INVITE request containing a User-to-user Indicator parameter, and the indicator Request service 3 is set to the value 'not essential' or 'essential'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 227 Test case number SS_uus_012 Test case group SIP-SIP/SIP-I/UUS Reference 5.4.3.2, 6.5, 7.1/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47) AND ([Network B] SE 17 AND SE 47) AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 3 in initial INVITE request successful. BICC/ISUP - SIP-I interworking applies in the originating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 3 calls user B an User-to-user Indicator parameter is present set to 'Request service 3', 'not essential' or 'essential' in the encapsulated IAM of the initial INVTE request. The User-to-User service is successful. Configuration User A is subscribed to the User-to-User service 3 User B is subscribed to the User-to-User service 3 SIP Parameter INVITE: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM User-to-user Indicator Request service 3 not essential or 'essential' 200 OK Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM User-to-user Indicator Response service 3 provided INFO Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required USR User-to-user Information User Information Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (IAM) 180 Ringing (ACM) 200 OK INVITE (ANM) ACK INFO (USR) 200 OK INFO INFO (USR) 200 OK INFO Apply post test routine Comments Check: Is an ISUP/BICC IAM encapsulated in the initial INVITE request containing a User-to-user Indicator parameter, and the indicator Request service 3 is set to the value 'not essential' or 'essential'? Check: Is an ISUP/BICC ANM encapsulated in the 200 OK INVITE and the User-to-user Indicator parameter is set to 'Response', 'service 3 provided'? Check: Is an ISUP/BICC USR encapsulated in the INFO message sent from network A to network B containing an User-to-user Information parameter? Check: Is an ISUP/BICC USR encapsulated in the INFO message sent from network B to network A containing an User-to-user Information parameter? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 228 Test case number SS_uus_013 Test case group SIP-SIP/SIP-I/UUS Reference 7.1, 6.5/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47) AND ([Network B] SE 17 AND SE 47) AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 3 not essential rejected in 200 OK response. BICC/ISUP - SIP-I interworking applies in the originating and terminating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 3 not essential calls user B not subscribed to User-to-User service 3 the call is rejected by the network an User-to-user Indicator parameter is present set to 'Response', 'service 3 not provided' in the encapsulated ANM of the 200 OK final response. Configuration User A is subscribed to the User-to-User service 3 User B is not subscribed to the User-to-User service 3 SIP Parameter INVITE: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM User-to-user Indicator Request service 3 not essential 200 OK Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM User-to-user Indicator Response service 3 not provided Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (IAM) 180 Ringing (ACM) 200 OK INVITE (ANM) ACK Apply post test routine Comments Check: Is an ISUP/BICC IAM encapsulated in the initial INVITE request? Check: Is a User-to-user Information parameter present in the encapsulated ISUP/BICC IAM set to 'Request', 'service 3' 'not essential'? Check: Is an ISUP/BICC ANM encapsulated in the 200 OK response? Check: Is an User-to-user Indicator parameter present set to 'Response', 'service 3 not provided' in the encapsulated ISUP/BICC ANM? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 229 Test case number SS_uus_014 Test case group SIP-SIP/SIP-I/UUS Reference 6.11.2, 7.1/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47) AND ([Network B] SE 17 AND SE 47) AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 3 essential rejection. BICC/ISUP - SIP-I interworking applies in the originating and terminating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 3 essential calls user B not subscribed to User-to-User service 3 the call is rejected by the network. A 500 Server Internal Error is sent and an encapsulated ISUP/BICC REL is present, the Cause value is set to #29 or #69. Configuration User A is subscribed to the User-to-User service 3 SIP Parameter INVITE: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM User-to-user Indicator Request service 3 essential 500 Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL Cause value #29 or #69 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE (IAM) 500 Server Internal Error (REL) ACK Apply post test routine Comments Check: Is an ISUP/BICC IAM encapsulated in the initial INVITE request? Check: Is a User-to-user Indicator parameter present in the encapsulated ISUP/BICC IAM set to 'Request', 'service 1', 'essential'? Check: Is an ISUP/BICC REL encapsulated in the 500 response? Check: Is the Cause value set to #29 or #69 in the encapsulated REL? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 230 Test case number SS_uus_015 Test case group SIP-SIP/SIP-I/UUS Reference 5.4.3.2, 6.5, 7.1/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47) AND ([Network B] SE 17 AND SE 47) AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 3 during a session is established successful. BICC/ISUP - SIP-I interworking applies in the originating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 3 user A is able to request the User-to-User service 3 while the session is established. The User-to- User service is successful. Configuration User A is subscribed to the User-to-User service 3 User B is subscribed to the User-to-User service 3 SIP Parameter INFO: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required FAR Facility indicator user-to-user service User-to-user Indicator Request service 3 not essential INFO: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required FAA Facility indicator user-to-user service User-to-user Indicator Response service 3 provided INFO Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required USR User-to-user Information User Information Message flow SIP (Network A) Interconnection Interface SIP (Network B) A session is already established INFO (FAR) 200 OK INFO INFO (FAA) 200 OK INFO INFO (USR) 200 OK INFO INFO (USR) 200 OK INFO Apply post test routine ETSI ETSI TS 101 585 V1.2.1 (2014-04) 231 Comments A session is already established Check: Is an ISUP/BICC FAR encapsulated in the INFO request sent from Network A to Network B and the a User-to-user Indicator parameter is set to Is the Request service 3 'not essential'? Check: Is an ISUP/BICC FAA encapsulated in the INFO request sent from Network B to Network A and the User-to-user Indicator parameter is set to 'Response', 'service 3 provided'? Check: Is an ISUP/BICC USR encapsulated in the INFO message sent from network A to network B containing an User-to-user Information parameter? Check: Is an ISUP/BICC USR encapsulated in the INFO message sent from network B to network A containing an User-to-user Information parameter? Repeat this test in reverse direction. Test case number SS_uus_016 Test case group SIP-SIP/SIP-I/UUS Reference 5.4.3.2, 6.5, 7.1/ [24] SELECTION EXPRESSION ([Network A] SE 17 AND SE 47) AND ([Network B] SE 17 AND SE 47) AND SE 63 Test purpose SIP-I support: Indicating of User-to-User service 3 during a session is established unsuccessful. BICC/ISUP - SIP-I interworking applies in the originating network User A is located in network A and user B is located in network B. Ensure when user A subscribed to the User-to-User service 3 user A is able to request the User-to-User service 3 while the session is established. The service request is rejected by Network B. Configuration User A is subscribed to the User-to-User service 3 User B is not subscribed to the User-to-User service 3 SIP Parameter INFO: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required FAR Facility indicator user-to-user service User-to-user Indicator Request service 3 not essential INFO: Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required FRJ Facility indicator user-to-user service User-to-user Indicator Response service 3 not provided Message flow SIP (Network A) Interconnection Interface SIP (Network B) A session is already established INFO (FAR) 200 OK INFO INFO (FRJ) 200 OK INFO Apply post test routine Comments A session is already established Check: Is an ISUP/BICC FAR encapsulated in the INFO request sent from Network A to Network B and the a User-to-user Indicator parameter is set to Is the Request service 3 'not essential'? Check: Is an ISUP/BICC FAA encapsulated in the INFO request sent from Network B to Network A and the User-to-user Indicator parameter is set to 'Response', 'service 3 not provided'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 232
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101 585
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7.1.6.2 Subaddressing (SUB)
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Test case number SS_sub_001 Test case group SIP-SIP/SIP-I/SUB Reference 7.1/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 62 Test purpose SIP-I support: Calling party subaddress can be correctly transferred in the Access Transport parameters. BICC/ISUP - SIP-I interworking applies in the originating network User A is located in network A and user B is located in network B. Ensure that an ISUP/BICC ATP parameter present in the encapsulated IAM of the INVITE request and contains a Calling party subaddress. Configuration User A is subscribed to the SUB supplementary service SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Access transport Calling party subaddress --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(IAM) Comments Establish a call from User A subscribed to the SUB supplementary service to user B Check: Is an ISUP/BICC IAM present in the initial INVITE request? Check: Is an ISUP/BICC ATP parameter present in the encapsulated IAM containing a Calling party subaddress? Repeat this test in reverse direction. Test case number SS_sub_002 Test case group SIP-SIP/SIP-I/SUB Reference 7.1/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 62 Test purpose SIP-I support. Called party subaddress can be correctly transferred in the Access Transport parameters. BICC/ISUP - SIP-I interworking applies in the originating network User A is located in network A and user B is located in network B. Ensure that an ISUP/BICC ATP parameter present in the encapsulated IAM of the INVITE request and contains a Called party subaddress. Configuration User A is subscribed to the SUB supplementary service SIP Parameter INVITE Content-Type: multipart/mixed;boundary=[any boundary name] --[any boundary name] Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required IAM Access transport Called party subaddress --[any boundary name]-- Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(IAM) Apply post test routine Comments Check: Is the BICC/ISUP ANM encapsulated in the 200 OK INVITE final response? Check: Is an ISUP/BICC ATP parameter present in the encapsulated ANM containing a Called party subaddress? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 233 Test case number SS_sub_003 Test case group SIP-SIP/SIP-I/SUB Reference 6.7/ [24] SELECTION EXPRESSION [Network B] SE 17 AND SE 47 AND SE 62 Test purpose SIP-I support. Connected party subaddress can be correctly transferred in the Access Transport parameters. BICC/ISUP - SIP-I interworking applies in the terminating network User A is located in network A and user B is located in network B. Ensure that an ISUP/BICC ATP parameter present in the encapsulated ANM of the 200 OK INVITE final response and a Connected party subaddress is contained. Configuration User B is subscribed to the SUB supplementary service SIP Parameter 200 OK INVITE Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required ANM Access transport Connected party subaddress Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE(IAM) 180 Ringing(ACM) 200 OK INVITE(ANM) ACK Apply post test routine Comments Check: Is the BICC/ISUP ANM encapsulated in the 200 OK INVITE final response? Check: Is an ISUP/BICC ATP parameter present in the encapsulated ANM containing a Called party subaddress? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 234
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101 585
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7.1.6.3 Terminal Portability (TP)
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Test case number SS_tp_001 Test case group SIP-SIP/SIP-I/TP Reference 5.4.3.2/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 64 Test purpose SIP-I support. SUS and RES messages transferred in an INFO request. BICC/ISUP - SIP-I interworking applies in the originating network User A is located in network A and user B is located in network B. A session is already established. Ensure that an INFO request is sent from Network A to Network B and an ISUP SUS message is encapsulated containing a Suspend/resume indicator set to ISDN subscriber initiated. Ensure that an INFO request is sent from Network A to Network B and an ISUP RES message is encapsulated containing a Suspend/resume indicator set to ISDN subscriber initiated. Configuration User A is subscribed to the Terminal Portability supplementary service SIP Parameter INFO Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required SUS Suspend/resume indicator ISDN subscriber initiated INFO Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required RES Suspend/resume indicator ISDN subscriber initiated Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists INFO(SUS) 200 OK INFO INFO(RES) 200 OK INFO Apply post test routine Comments A session is already established Check: Is an ISUP SUS message encapsulated in the INFO request and the Suspend/resume indicator set to 'ISDN subscriber initiated'? Check: Is an ISUP RES message encapsulated in the INFO request and the Suspend/resume indicator set to 'ISDN subscriber initiated'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 235 Test case number SS_tp_002 Test case group SIP-SIP/SIP-I/TP Reference 5.4.3.2, 6.11.2, 6.11.2/ [24] SELECTION EXPRESSION [Network A] SE 17 AND SE 47 AND SE 64 Test purpose SIP-I support. SUS message transferred in an INFO request call released. BICC/ISUP - SIP-I interworking applies in the originating network User A is located in network A and user B is located in network B. A session is already established. Ensure that an INFO request is sent from Network A to Network B and an ISUP SUS message is encapsulated containing a Suspend/resume indicator set to ISDN subscriber initiated. Ensure that an BYE request is sent from Network A to Network B and an ISUP REL message is encapsulated containing a Cause value set to #102. Configuration User A is subscribed to the Terminal Portability supplementary service SIP Parameter INFO Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required SUS Suspend/resume indicator ISDN subscriber initiated BYE Content-Type: application/isup;version=itu-t92 Content-Disposition: signal;handling=required REL Location public network serving remote user Cause value 102 Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists INFO(SUS) 200 OK INFO BYE(REL) 200 OK BYE Comments A session is already established Check: Is an ISUP SUS message encapsulated in the INFO request and the Suspend/resume indicator set to ISDN 'subscriber initiated'? Check: Is an ISUP REL message encapsulated in the BYE request and the Cause value set to #102? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 236
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101 585
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7.2 Number Portability
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Test case number SS_NP_001 Test case group SIP-SIP/NubP Reference 5.3, 5.4/ [2] SELECTION EXPRESSION [Network A] SE 13 Test purpose Request line in the INVITE contains the number portability indication. User A attempts to call user B ported to network B. Ensure that the userinfo in the INVITE contains a destination number in the global number format, an 'rn' parameter containing the Number Portability Routing Number in a global number format with hex digits and optional the 'npdi' parameter. Configuration SIP Parameter INVITE: Request line sip: + <CC> <NDC> <SN>[;npdi][; rn=(Number portability routing number)] @<hostname>;user = phone SIP/2.0 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Check: Is the URI in the userinfo of the Request line in a global number format? Check: Is the URI rn parameter containing the Number Portability Routing Number in a global number format? Check: Is optional the URI parameter 'npdi' present? Check: Is the user parameter set to 'phone'? Repeat this test in reverse direction. Test case number SS_NP_002 Test case group SIP-SIP/NubP Reference 5.3, 5.4/ [2] SELECTION EXPRESSION NOT [Network A] SE 13 Test purpose Request line in the INVITE without npdi parameter. The Network A does not have a Number Portability database. User A attempts to call user B ported to network B. Ensure that the userinfo in the INVITE contains a destination number in a global number format and a npdi URI parameter is not present. Configuration SIP Parameter INVITE: Request line sip: + <CC> <NDC> <SN>@<hostname>;user = phone SIP/2.0 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Check: Is the URI in the userinfo of the Request line in a global number format without npdi parameter and number portability routing number? Check: Is the user parameter set to 'phone'? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 237
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101 585
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7.3 Accounting
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Test case number SS_acc_001 Test case group SIP-SIP/ACCOUNTING Reference SELECTION EXPRESSION Test purpose Comparison of Charging Data Records > 1 s. Accounting of a confirmed session with a duration > 1 s. Verify the duration of the active session stored in the CDR of both networks compared with the duration in the monitored message flow at the Interconnection Interface. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK Communication BYE 200 OK BYE Comments 1. Setup a call from network A to network B. 2. Verify is the session confirmed. 3. Terminate the session after 5 s. 4. Determine the duration of the session from the trace of the call monitor. 5. Compare the following information elements indicated in the CDR´s of both networks: • calling party number • called party number • timestamp • callduration • callsetuptime (optional) 6. Check the duration indicated in the CDR against the duration in the call trace. 7. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 238 Test case number SS_acc_002 Test case group SIP-SIP/ACCOUNTING Reference SELECTION EXPRESSION Test purpose Comparison of Charging Data Records < 1 s Accounting of a confirmed session with a duration of < 1 min. Verify the duration of the active session stored in the CDR of both networks compared with the duration in the monitored message flow at the Interconnection Interface. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK Communication BYE 200 OK BYE Comments 1. Setup a call from network A to network B. 2. Verify is the session confirmed. 3. Terminate the session after 5 s. 4. Determine the duration of the session from the trace of the call monitor. 5. Compare the following information elements indicated in the CDR´s of both networks: • calling party number • called party number • timestamp • callduration • callsetuptime (optional) 6. Check the duration indicated in the CDR against the duration in the call trace. 7. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 239 Test case number SS_acc_003 Test case group SIP-SIP/ACCOUNTING Reference SELECTION EXPRESSION Test purpose Comparison of Charging Data Records > 15 min. Accounting of a confirmed session with a duration of > 15 min. Verify the duration of the active session stored in the CDR of both networks compared with the duration in the monitored message flow at the Interconnection Interface. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK Communication BYE 200 OK BYE Comments 1. Setup a call from network A to network B. 2. Verify is the session confirmed. 3. Terminate the session after 15 min. 4. Determine the duration of the session from the trace of the call monitor. 5. Compare the following information elements indicated in the CDR´s of both networks: • calling party number • called party number • timestamp • callduration • callsetuptime (optional) 6. Check the duration indicated in the CDR against the duration in the call trace. 7. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 240 Test case number SS_acc_004 Test case group SIP-SIP/ACCOUNTING Reference SELECTION EXPRESSION Test purpose Comparison of Charging Data Records 25 min. Accounting of a confirmed session with a duration of 25 min. Verify the duration of the active session stored in the CDR of both networks compared with the duration in the monitored message flow at the Interconnection Interface. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK Communication BYE 200 OK BYE Comments 1. Setup a call from network A to network B. 2. Verify is the session confirmed. 3. Terminate the session after 25 min. 4. Determine the duration of the session from the trace of the call monitor. 5. Compare the following information elements indicated in the CDR´s of both networks: • calling party number • called party number • timestamp • callduration • callsetuptime (optional) 6. Check the duration indicated in the CDR against the duration in the call trace. 7. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 241 Test case number SS_acc_005 Test case group SIP-SIP/ACCOUNTING Reference SELECTION EXPRESSION Test purpose Comparison of Charging Data Records more than 30 min. Accounting of a confirmed session with a duration of > 30 min. Verify the duration of the active session stored in the CDR of both networks compared with the duration in the monitored message flow at the Interconnection Interface. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK Communication BYE 200 OK BYE Comments 1. Setup a call from network A to network B. 2. Verify is the session confirmed. 3. Terminate the session after 35 min. 4. Determine the duration of the session from the trace of the call monitor. 5. Compare the following information elements indicated in the CDR´s of both networks: • calling party number • called party number • timestamp • callduration • callsetuptime (optional) 6. Check the duration indicated in the CDR against the duration in the call trace. 7. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 242 Test case number SS_acc_006 Test case group SIP-SIP/ACCOUNTING Reference SELECTION EXPRESSION Test purpose Comparison of Charging Data Records more than 60 min. Accounting of a confirmed session with a duration between 60 min and 120 min. Verify the duration of the active session stored in the CDR of both networks compared with the duration in the monitored message flow at the Interconnection Interface. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK Communication BYE 200 OK BYE Comments 1. Setup a call from network A to network B. 2. Verify is the session confirmed. 3. Terminate the session at the earliest 61 min and at the latest 119 min. 4. Determine the duration of the session from the trace of the call monitor. 5. Compare the following information elements indicated in the CDR´s of both networks: • calling party number • called party number • timestamp • callduration • callsetuptime (optional) 6. Check the duration indicated in the CDR against the duration in the call trace. 7. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 243 Test case number SS_acc_007 Test case group SIP-SIP/ACCOUNTING Reference SELECTION EXPRESSION Test purpose Comparison of Charging Data Records more than 24 hours. Accounting of a confirmed session with duration > 24 h with change of date. Verify the duration of the active session stored in the CDR of both networks compared with the duration in the monitored message flow at the Interconnection Interface. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK Communication BYE 200 OK BYE Comments 1. Setup a call from network A to network B. 2. Verify is the session confirmed. 3. Terminate the session after 24 hours. 4. Determine the duration of the session from the trace of the call monitor. 5. Compare the following information elements indicated in the CDR´s of both networks: • calling party number • called party number • timestamp • callduration • callsetuptime (optional) 6. Check the duration indicated in the CDR against the duration in the call trace. 7. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 244 Test case number SS_acc_008 Test case group SIP-SIP/ACCOUNTING Reference SELECTION EXPRESSION Test purpose Comparison of Charging Data Records less than 1 s. Accounting of a confirmed session with duration <1 s. Verify the duration of the active session stored in the CDR of both networks compared with the duration in the monitored message flow at the Interconnection Interface. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing 200 OK INVITE ACK Communication BYE 200 OK BYE Comments 1. Setup a call from network A to network B. 2. Verify is the session confirmed. 3. Terminate the session after 0,9 s. 4. Determine the duration of the session from the trace of the call monitor. 5. Compare the following information elements indicated in the CDR´s of both networks: • calling party number • called party number • timestamp • callduration • callsetuptime (optional) 6. Check the duration indicated in the CDR against the duration in the call trace. 7. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 245 Test case number SS_acc_009 Test case group SIP-SIP/ACCOUNTING Reference SELECTION EXPRESSION Test purpose Comparison of Charging Data Records session not confirmed. Accounting of an unsuccessful session in the early dialogue. Verify the duration of the call attempt stored in the CDR of both networks compared with the duration in the monitored message flow at the Interconnection Interface if applicable. Configuration SIP Parameter Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 180 Ringing BYE/CANCEL 200 OK BYE/CANCEL 487 Request Terminated ACK Comments 1. Setup a call from network A to network B. 2. Verify is an early dialogue established. 3. Terminate the early dialogue after 20 s. 4. Determine the duration of the session from the trace of the call monitor. 5. Compare the following information elements indicated in the CDR´s of both networks: • calling party number • called party number • timestamp • callduration • callsetuptime (optional) 6. Check the duration indicated in the CDR against the duration in the call trace. 7. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 246
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7.4 Carrier Selection
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Test case number SS_csel_001 Test case group SIP-SIP/CS Reference 5.7.1.10/ [2] SELECTION EXPRESSION [Network A] SE14 AND [Network B] SE15 Test purpose User selects an operator 'call-by-call'. User A and user B are located in network A. Ensure that user A is able to call user B and user A is able to select network B as a selected carrier 'call-by-call'. Configuration User in network A is not presubscribed SIP Parameter INVITE: Request line sip: + <CC> <NDC> <SN>[;cic=(carrier ID)]@<hostname> user=phone SIP/2.0 INVITE: Request line sip: + <CC> <NDC> <SN>;npdi [;rn=<Number portability routing number>]@<hostname>; user=phone SIP/2.0 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 1 INVITE 2 Apply post test routine Comments Check: Is the optional 'cic' tel uri parameter present in the Request URI in the INVITE sent from network A to network B identifying the selected carrier? Check: Is the 'npdi' parameter present in the Request URI of the INVITE request sent from network B to network A? Check: Is optional the 'rn' parameter present in the Request URI of the INVITE request sent from network B to network A? NOTE 1: The 'cic' parameter may be absent according national regulations or national agreements. NOTE 2: It is possible that further information is available in the Request line regarding the end user charging in case of Carrier selection. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 247 Test case number SS_csel_002 Test case group SIP-SIP/CS Reference 5.7.1.10/ [2] SELECTION EXPRESSION [Network A] SE14 AND [Network B] SE15 Test purpose User is presubscribed to operator B. User A and user B are located in network A. Ensure that user A is able to call user B and user A is preselected to network B as a selected carrier. Configuration User in network A is presubscribed to network B SIP Parameter INVITE: Request line sip: + <CC> <NDC> <SN>[;cic=(carrier ID)]@<hostname> user=phone SIP/2.0 INVITE: Request line sip: + <CC> <NDC> <SN>;npdi [;rn=<Number portability routing number>]@<hostname>; user=phone SIP/2.0 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 1 INVITE 2 Apply post test routine Comments Check: Is the optional 'cic' tel uri parameter present in the Request URI in the INVITE sent from network A to network B identifying the selected carrier? Check: Is the 'npdi' parameter present in the Request URI of the INVITE request sent from network B to network A? Check: Is optional the 'rn' parameter present in the Request URI of the INVITE request sent from network B to network A? NOTE 1: The 'cic' parameter may be absent according national regulations or national agreements. NOTE 2: It is possible that further information is available in the Request line regarding the end user charging in case of Carrier selection. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 248 Test case number SS_csel_003 Test case group SIP-SIP/CS Reference 5.7.1.10/ [2] SELECTION EXPRESSION [Network A] SE14 AND [Network B] SE15 Test purpose User is presubscribed to an operator unequal to B, and overrides the preselection with call-by-call via operator B. User A and user B are located in network A. User A is preselected to a network unequal to network B. Ensure that user A is able to call user B and user A is able to select network B as a selected carrier 'call-by-call'. The preselected carrier is ignored. Configuration User in network A is not presubscribed to network B SIP Parameter INVITE: Request line sip: + <CC> <NDC> <SN>[;cic=(carrier ID)]@<hostname> user=phone SIP/2.0 INVITE: Request line sip: + <CC> <NDC> <SN>;npdi [;rn=<Number portability routing number>]@<hostname>; user=phone SIP/2.0 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 1 INVITE 2 Apply post test routine Comments Check: Is the optional 'cic' tel uri parameter present in the Request URI in the INVITE sent from network A to network B identifying the selected carrier? Check: Is the 'npdi' parameter present in the Request URI of the INVITE request sent from network B to network A? Check: Is optional the 'rn' parameter present in the Request URI of the INVITE request sent from network B to network A? NOTE 1: The 'cic' parameter may be absent according national regulations or national agreements. NOTE 2: It is possible that further information is available in the Request line regarding the end user charging in case of Carrier selection. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 249 Test case number SS_csel_004 Test case group SIP-SIP/CS Reference 5.7.1.10/ [2] SELECTION EXPRESSION [Network A] SE14 AND [Network B] SE15 Test purpose User is presubscribed to operator B, and overrides the preselection with call-by-call via operator B. User A and user B are located in network A. User A is preselected to network B. Ensure that user A is able to call user B and user A is able to select network B as a selected carrier 'call-by-call'. The preselected carrier is ignored. Configuration User in network A is presubscribed to network B SIP Parameter INVITE: Request line sip: + <CC> <NDC> <SN>[;cic=(carrier ID)]@<hostname> user=phone SIP/2.0 INVITE: Request line sip: + <CC> <NDC> <SN>;npdi [;rn=<Number portability routing number>]@<hostname>; user=phone SIP/2.0 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 1 INVITE 2 Apply post test routine Comments Check: Is the optional 'cic' tel uri parameter present in the Request URI in the INVITE sent from network A to network B identifying the selected carrier? Check: Is the 'npdi' parameter present in the Request URI of the INVITE request sent from network B to network A? Check: Is optional the 'rn' parameter present in the Request URI of the INVITE request sent from network B to network A? NOTE 1: The 'cic' parameter may be absent according national regulations or national agreements. NOTE 2: It is possible that further information is available in the Request line regarding the end user charging in case of Carrier selection. Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 250 Test case number SS_csel_005 Test case group SIP-SIP/CS Reference SELECTION EXPRESSION [Network A] SE14 AND [Network B] SE15 AND [Network A] SE34 Test purpose User is preselected to operator B. Transit of CUG information -OA. An originating user in a CUG Outgoing Access not allowed preselected to Network B and calls to a user in the same CUG. The session establishment is successful. Configuration User in network A is presubscribed to network B Users in network A are in the same CUG SIP Parameter INVITE: Request line sip: + <CC> <NDC> <SN>@ <hostname> user=phone SIP/2.0 Content-Type: application/vnd.etsi.cug+xml Content-Disposition: ….;handling= required ……. <…:cug> …… <..: cugCommunicationIndicator>11</…: cugCommunicationIndicator> <…:cug> INVITE: Request line sip: + <CC> <NDC> <SN@<hostname>;user=phone SIP/2.0 Content-Type: application/vnd.etsi.cug+xml Content-Disposition: ….;handling= required ……. <…:cug> …… <..: cugCommunicationIndicator>11</…: cugCommunicationIndicator> <…:cug> Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE 1 INVITE 2 Apply post test routine Comments Check: Is the 'npdi' parameter present in the userinfo of the INVITE request sent from network B to network A? Check: Is optional the 'rn' parameter present in the userinfo of the INVITE request sent from network B to network A? Check: Contains the XML body in the INVITE a 'cugCommunicationIndicator' element set to '11' as a 'cug' child element? Check: Is the session setup not rejected? ETSI ETSI TS 101 585 V1.2.1 (2014-04) 251
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7.5 Emergency call
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Test case number SS_ecall_001 Test case group SIP-SIP/EmC Reference 5.2.10, 5.7.1.14/ [2] SELECTION EXPRESSION Test purpose Request line in the INVITE. User A attempts to call a PSAP located in network B. Ensure that the Request line in the INVITE contains the emergency number and a 'rn' parameter containing the PSAP routing number. In addition a location information may be present: • Geolocation header and corresponding PIDF-LO Element • User-to-User header • National solution to convey location information to make location information available for the PSAP. Configuration SIP Parameter INVITE: Request line sip+ <(emergency number)>[; rn =+<(PSAP routing number)] @hostname>;user = phone SIP/2.0 Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE Apply post test routine Comments Check: Is the URI in the userinfo of the Request line in a global number format containing the PSAP routing number? Check: Optional: Is the URI 'rn' parameter containing the PSAP Routing Number? Check: Is the user parameter set to 'phone'? Check: Is the location information present in the initial INVITE request. Geolocation header PIDF-LO Element XML 'geopriv' sub element Or User-to-User header Or National solution Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 252
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7.6 SIP Support of Charging
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Test case number SS_sipc_001 Test case group SIP-SIP/ SIP_charging Reference B.2.3/ [19] SELECTION EXPRESSION SE 16 Test purpose Successful session from user A to user B via network B one single tariff. User A is located in network A and network B is responsible for charging (CDP) in case of carrier selection or service. Ensure that the network B sends a tariff information with one single tariff covered in a XML MIME body in a reliable provisional or successful final response. Configuration SIP Parameter INVITE: Supported: 100rel 18x or 200 OK Require: 100rel ContentType: application/vnd.etsi.sci+xml Content-Disposition: render; handling=optional messageType crgt chargingControlIndicators chargingTariff tariffCurrency currentTariffCurrency communicationChargeSequenceCurrency currencyFactorScale currencyFactor currencyScale tariffDuration subTariffControl tariffControlIndicators originationIdentification currency (optional) Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE CASE A 18x(crgt) PRACK 200 OK PRACK CASE B 200 OK INVITE(crgt) Apply post test routine Comments Check: Is the supported header in the initial INVITE set to '100rel' Check: Is the Require header in the response containing the tariff information set to '100rel'? Check: Is the messageType 'crgt' present in a 1xx provisional or a 200 OK INVITE final response? Check: Is the tariffCurrency element set to 'currentTariffCurrency'? Check: Represents the currencyFactorScale in the communicationChargeSequenceCurrency element the applicable tariff? Check: Is the tariffDuration element set to '0'? Check: Is the optional element 'currency' set to 'EUR' if present? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 253 Test case number SS_sipc_002 Test case group SIP-SIP/ SIP_charging Reference B.2.3/ [19] SELECTION EXPRESSION SE 16 Test purpose Successful session from user A to user B via network B several tariffs in one sequence. User A is located in network A and network B is responsible for charging (CDP) in case of carrier selection or service. Ensure that the network B sends a tariff information with several tariffs in a sequence covered in a XML MIME body in a reliable provisional or successful final response. Configuration SIP Parameter INVITE: Supported: 100rel 18x or 200 OK Require: 100rel ContentType: application/vnd.etsi.sci+xml Content-Disposition: render; handling=optional messageType crgt chargingControlIndicators chargingTariff tariffCurrency currentTariffCurrency communicationChargeSequenceCurrency currencyFactorScale currencyFactor currencyScale tariffDuration subTariffControl communicationChargeSequenceCurrency currencyFactorScale currencyFactor currencyScale tariffDuration subTariffControl tariffControlIndicators originationIdentification currency (optional) Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE CASE A 18x(crgt) PRACK 200 OK PRACK CASE B 200 OK INVITE(crgt) Apply post test routine Comments Check: Is the Supported header in the initial INVITE set to '100rel'? Check: Is the Require header in the response containing the tariff information set to '100rel'? Check: Is the messageType 'crgt' present in a 1xx provisional or a 200 OK INVITE final response? Check: Is the tariffCurrency element set to 'currentTariffCurrency'? Check: Are there more than one communicationCharge SequenceCurrency elements present in the currentTariffCurrency element? Check: Represents the currencyFactorScale in the communicationCharge SequenceCurrency elements the applicable tariffs? Check: Is the tariffDuration element in the last applicable tariff set to '0'? Check: Is the optional element 'currency' set to 'EUR' if present? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 254 Test case number SS_sipc_003 Test case group SIP-SIP/ SIP_charging Reference B.2.3/ [19] SELECTION EXPRESSION SE 16 Test purpose Successful session from user A to user B via network B with call attempt charge. User A is located in network A and network B is responsible for charging (CDP) in case of carrier selection or service. Ensure that the network B sends a tariff information with a call attempt charge covered in a XML MIME body in a reliable provisional or successful final response. Configuration SIP Parameter INVITE: Supported: 100rel 18x or 200 OK Require: 100rel ContentType: application/vnd.etsi.sci+xml Content-Disposition: render; handling=optional messageType crgt chargingControlIndicators chargingTariff tariffCurrency currentTariffCurrency communicationChargeSequenceCurrency currencyFactorScale currencyFactor currencyScale tariffDuration subTariffControl tariffControlIndicators callAttemptChargeCurrency currencyFactor currencyScale originationIdentification currency (optional) Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE CASE A 18x(crgt) PRACK 200 OK PRACK CASE B 200 OK INVITE(crgt) Apply post test routine Comments Check: Is the supported header in the initial INVITE set to '100rel'? Check: Is the Require header in the response containing the tariff information set to '100rel'? Check: Is the messageType a 'crgt' present in a 1xx provisional or a 200 OK INVITE final response? Check: Is the tariffCurrency element set to 'callAttemptChargeCurrency'? Check: Represents the currencyFactorScale in the callAttemptChargeCurrency element the applicable tariff? Check: Is the optional element 'currency' set to 'EUR' if present? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 255 Test case number SS_sipc_004 Test case group SIP-SIP/ SIP_charging Reference B.2.3/ [19] SELECTION EXPRESSION SE 16 Test purpose Successful session from user A to user B via network B with call setup charge. User A is located in network A and network B is responsible for charging (CDP) in case of carrier selection or service. Ensure that the network B sends a tariff information with a call setup charge covered in a XML MIME body in a reliable provisional or successful final response. Configuration SIP Parameter INVITE: Supported: 100rel 18x or 200 OK Require: 100rel ContentType: application/vnd.etsi.sci+xml Content-Disposition: render; handling=optional messageType crgt chargingControlIndicators chargingTariff tariffCurrency currentTariffCurrency communicationChargeSequenceCurrency currencyFactorScale currencyFactor currencyScale tariffDuration subTariffControl tariffControlIndicators callSetupChargeCurrency currencyFactor currencyScale originationIdentification currency (optional) Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE CASE A 18x(crgt) PRACK 200 OK PRACK CASE B 200 OK INVITE(crgt) Apply post test routine Comments Check: Is the supported header in the initial INVITE set to '100rel'? Check: Is the Require header in the response containing the tariff information set to '100rel'? Check: Is the messageType a 'crgt' present in a 1xx provisional or a 200 OK INVITE final response? Check: Is the tariffCurrency element set to 'callSetupChargeCurrency'? Check: Represents the currencyFactorScale in the callSetupChargeCurrency element the applicable tariff? Check: Is the optional element 'currency' set to 'EUR' if present? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 256 Test case number SS_sipc_005 Test case group SIP-SIP/ SIP_charging Reference B.2.3/ [19] SELECTION EXPRESSION SE 16 Test purpose Successful session from user A to user B via network B with a next tariff. User A is located in network A and network B is responsible for charging (CDP) in case of carrier selection or service. Ensure that the network B sends a tariff information with a next tariff and tariff switch over time covered in a XML MIME body in a reliable provisional or successful final response. Configuration SIP Parameter INVITE: Supported: 100rel 18x or 200 OK Require: 100rel ContentType: application/vnd.etsi.sci+xml Content-Disposition: render; handling=optional messageType crgt chargingControlIndicators chargingTariff tariffCurrency currentTariffCurrency communicationChargeSequenceCurrency currencyFactorScale currencyFactor currencyScale tariffDuration subTariffControl tariffControlIndicators tariffSwitchCurrency nextTariffCurrency communicationChargeSequenceCurrency currencyFactorScale currencyFactor currencyScale tariffDuration subTariffControl tariffControlIndicators tariffSwitchOverTime originationIdentification currency (optional) Message flow SIP (Network A) Interconnection Interface SIP (Network B) INVITE CASE A 18x(crgt) PRACK 200 OK PRACK CASE B 200 OK INVITE(crgt) Apply post test routine Comments Check: Is the supported header in the initial INVITE set to '100rel'? Check: Is the Require header in the response containing the tariff information set to '100rel'? Check: Is the messageType 'crgt' present in a 1xx provisional or a 200 OK INVITE final response? Check: Is the tariffSwitchCurrency element set to 'nextTariffCurrency'? Check: Represents the currencyFactorScale in the communicationChargeSequenceCurrency element the next tariff? Check: Is the time to change the tariff indicated in the tariffSwitchOverTime element? Check: Is the optional element 'currency' set to 'EUR' if present? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 257 Test case number SS_sipc_006 Test case group SIP-SIP/ SIP_charging Reference B.2.3/ [19] SELECTION EXPRESSION SE 16 Test purpose Successful change of a current tariff and next tariff during an active session. User A is located in network A and network B is responsible for charging (CDP) in case of carrier selection or service. Ensure that the network B sends a new tariff information with several current tariffs and several next tariffs covered in a XML MIME body in an INFO request. Configuration SIP Parameter INFO ContentType: application/vnd.etsi.sci+xml messageType crgt chargingControlIndicators chargingTariff tariffCurrency currentTariffCurrency communicationChargeSequenceCurrency currencyFactorScale currencyFactor currencyScale tariffDuration subTariffControl communicationChargeSequenceCurrency currencyFactorScale currencyFactor currencyScale tariffDuration subTariffControl tariffControlIndicators tariffSwitchCurrency nextTariffCurrency communicationChargeSequenceCurrency currencyFactorScale currencyFactor currencyScale tariffDuration subTariffControl communicationChargeSequenceCurrency currencyFactorScale currencyFactor currencyScale tariffDuration subTariffControl tariffControlIndicators tariffSwitchOverTime originationIdentification currency (optional) Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists INFO 200 OK INFO Apply post test routine Comments Check: Is the messageType 'crgt' present in the INFO request? Check: Is the tariffCurrency element set to 'currentTariffCurrency'? Check: Represents the currencyFactorScale in the communicationChargeSequenceCurrency elements the current tariffs? Check: Is the tariffSwitchCurrency element set to 'nextTariffCurrency'? Check: Represents the currencyFactorScale in the communicationChargeSequenceCurrency elements the next tariffs? Repeat this test in reverse direction. ETSI ETSI TS 101 585 V1.2.1 (2014-04) 258 Test case number SS_sipc_007 Test case group SIP-SIP/SIP_charging Reference B.2.3/ [19] SELECTION EXPRESSION SE 16 Test purpose Successful additional charge during an active session. User A is located in network A and network B is responsible for charging (CDP) in case of carrier selection or service. Ensure that the network B sends a new tariff information with additional charge covered in a XML MIME body in an INFO request. Configuration SIP Parameter INFO ContentType: application/vnd.etsi.sci+xml messageType aocrg chargingControlIndicators addOnCharge addOnChargeCurrency currencyFactor currencyScale originationIdentification currency (optional) Message flow SIP (Network A) Interconnection Interface SIP (Network B) A confirmed session already exists INFO 200 OK INFO Apply post test routine Comments Check: Is the messageType 'aocrg' present in the INFO request? Check: Is the addOnCharge element set to 'addOnChargeCurrency'? Check: Represents the currencyFactorScale the add on tariff? Repeat this test in reverse direction
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7.7 Quality of Service
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7.7.1 Reference Configurations
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7.7.1.1 Backbone Configuration
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Figure 7.7-1 shows the backbone configuration. IP Transit SBC ADM DL CL SBC SBC SBC ADM DL CL Figure 7.7-1: Backbone
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7.7.1.2 PSTN/ISDN classic access Configuration
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Figure 7.7-2 shows the PSTN/ISDN classic access configuration. Figure 7.7-2: Reference configuration for PSTN/ISDN with classical access ETSI ETSI TS 101 585 V1.2.1 (2014-04) 259
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7.7.1.3 NGN PSTN/ISDN access Configuration
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Figure 7.7-3 shows the NGN PSTN/ISDN classic access configuration. Figure 7.7-3: Reference configuration for NGN with PSTN/ISDN access
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7.7.1.4 Access DSL Configuration
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Figure 7.7-4 shows the xDSL access configuration. Figure 7.7-4: Reference configuration for DSL access
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7.7.1.5 Delay Values
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The requirements for the backbone delay, Network parameters: End-to-End Delay, Talker Echo Loudness Rating, R Value Delay with regional propagation delay (1 400 km/11 ms) are contained in clause 4 of TR 102 775 [i.3]
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7.7.2 Test purposes for Quality of Service test
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Test case number SS_qos_001 Test case group SIP-SIP/QoS Transmission Type: Voice Preconditions user segment A: Reset Jitter Buffer 1 and Jitter Buffer 2 (e.g. by establishing a new call) Apply signal "single-talk" to Interface A and determine Delay DJB1 Apply signal "single-talk" to Interface B and determine Delay DJB2 Preconditions user segment B: Reset Jitter Buffer 1 and Jitter Buffer 2 (e.g. by establishing a new call) Apply signal single-talk to Interface A and determine Delay DJB1 Apply signal single-talk to Interface B and determine Delay DJB2 Requirement DJB1 = DJB2 Delay jitter for Voice Test objective Delay Voice test with loopback Measurement procedure After establishing a voice call from the user segment A to user segment B, determine the round trip delay in the sending and receiving direction. Based on the measured delays in the user segment A and user segment B determine the transit segment delay. Loop in user segment B Dtr seg A-B = (Dsum seg A-B- DJB1seg B- DJB2segA)/2 Loop in user segment A Dtr seg B-A = (Dsum seg B-A - DJB1seg B- DJB2segA)/2 Calling station The amplitude of the tone is -16 dBm0 Called station The amplitude of the tone is -16 dBm0 Delay loop 1 000 ms ETSI ETSI TS 101 585 V1.2.1 (2014-04) 260 Test case number SS_qos_002 Test case group SIP-SIP/QoS Transmission Type: Voice Preconditions user segment A: Reset Jitter Buffer 1 and Jitter Buffer 2 (e.g. by establishing a new call) Apply signal "single-talk" to Interface A and determine Delay DJB1 and DJB2 Preconditions user segment B: Reset Jitter Buffer 1 and Jitter Buffer 2 (e.g. by establishing a new call) Apply signal "single-talk" to Interface A and determine Delay DJB1 and DJB2 Requirement DJB1 = DJB2 Delay jitter for Voice Test objective Delay Voice test with synchronous tests system Measurement procedure After establishing a voice call from the user segment A to user segment B, determine the delay of the end-to-end in the sending and receiving direction. Based on the measured delays in the user segment A and user segment B determine the transit segment delay. Dtr-seg A-B = Dsum-seg A-B - DJB1seg B Dtr-seg B-A= Dsum-seg B-A- DJB2segA Calling station The amplitude of the tone is -16 dBm0 Called station The amplitude of the tone is -16 dBm0 ETSI ETSI TS 101 585 V1.2.1 (2014-04) 261 Annex A (informative): Bibliography • IETF RFC 3966 (2004): "The tel URI for Telephone Numbers". • IETF RFC 3311 (2002): "The Session Initiation Protocol (SIP) UPDATE Method". • IETF RFC 3323 (2002): "A Privacy Mechanism for the Session Initiation Protocol (SIP)". • IETF RFC 3325 (2002): "Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks". • ETSI TS 129 163: "Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; Interworking between the IP Multimedia (IM) Core Network (CN) subsystem and Circuit Switched (CS) networks (3GPP TS 29.163 Release 10)". • IETF RFC 2833: "RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals". • ETSI TS 134 229-1: "Universal Mobile Telecommunications System (UMTS); Internet Protocol (IP) multimedia call control protocol based on Session Initiation Protocol (SIP) and Session Description Protocol (SDP); Part 1: Protocol conformance specification (3GPP TS 34.229-1 version 6.3.0 Release 6)". • ETSI EG 201 018: "Integrated Services Digital Network (ISDN); Application of the Bearer Capability (BC), High Layer Compatibility (HLC) and Low Layer Compatibility (LLC) information elements by terminals supporting ISDN services". • ETSI EN 300 093-1: "Integrated Services Digital Network (ISDN); Calling Line Identification Restriction (CLIR) supplementary service; Digital Subscriber Signalling System No. one (DSS1) protocol; Part 1: Protocol specification". • ETSI EN 300 207-1: "Integrated Services Digital Network (ISDN); Diversion supplementary services; Digital Subscriber Signalling System No. One (DSS1); Part 1: Protocol specification". • ETSI EN 300 188-1: "Integrated Services Digital Network (ISDN); Three-Party (3PTY) supplementary service; Digital Subscriber Signalling System No. one (DSS1) protocol; Part 1: Protocol specification". • ETSI EN 300 141-1: "Integrated Services Digital Network (ISDN); Call Hold (HOLD) supplementary service; Digital Subscriber Signalling System No. one (DSS1) protocol; Part 1: Protocol specification". • ETSI EN 300 185-1: "Integrated Services Digital Network (ISDN); Conference call, add-on (CONF) supplementary service; Digital Subscriber Signalling System No. one (DSS1) protocol; Part 1: Protocol specification". • ETSI EN 300 196-1: "Integrated Services Digital Network (ISDN); Generic functional protocol for the support of supplementary services; Digital Subscriber Signalling System No. one (DSS1) protocol; Part 1: Protocol specification". • ETSI EN 300 138-1: "Integrated Services Digital Network (ISDN); Closed User Group (CUG) supplementary service; Digital Subscriber Signalling System No. one (DSS1) protocol; Part 1: Protocol specification". • ETSI TS 124 147: "Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; Conferencing using the IP Multimedia (IM) Core Network (CN) subsystem; Stage 3 (3GPP TS 24.147 version 9.1.0 Release 9)". • ETSI EN 300 001: "Attachments to the Public Switched Telephone Network (PSTN); General technical requirements for equipment connected to an analogue subscriber interface in the PSTN". • ETSI ETS 300 648: "Public Switched Telephone Network (PSTN); Calling Line Identification Presentation (CLIP) supplementary service; Service description". ETSI ETSI TS 101 585 V1.2.1 (2014-04) 262 • ETSI EN 300 092-1: "Integrated Services Digital Network (ISDN); Calling Line Identification Presentation (CLIP) supplementary service; Digital Subscriber Signalling System No. one (DSS1) protocol; Part 1: Protocol specification". • ETSI EN 300 659: "Access and Terminals (AT); Analogue access to the Public Switched Telephone Network (PSTN); Subscriber line protocol over the local loop for display (and related) services". • ETSI TBR 008: "Integrated Services Digital Network (ISDN); Telephony 3,1 kHz teleservice; Attachment requirements for handset terminals". • Recommendation ITU-T Q.951: "Stage 3 description for number identification supplementary services using DSS 1". • Recommendation ITU-T Q.939: "Typical DSS 1 service indicator codings for ISDN telecommunications services". • Recommendation ITU-T Q.850 (05/98): "Usage of cause and location in the Digital Subscriber Signalling System No. 1 and the Signalling System No. 7 ISDN User Part". • ETSI EG 201 299-1: "Integrated Services Digital Network (ISDN); Network Integration Testing (NIT); ISDN/PSTN end-to-end testing; Part 1: Test Suite Structure and Test Purposes (TSS&TP) specification". ETSI ETSI TS 101 585 V1.2.1 (2014-04) 263 History Document history V1.1.1 August 2012 Publication V1.1.2 September 2012 Publication V1.2.1 April 2014 Publication
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1 Scope
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The present document specifies the Abstract Test Suite (ATS) and partial Protocol Implementation eXtra Information for Testing (PIXIT) proforma for the test specifications for Diameter protocol on the Rx interface as specified in TS 129 214 [1] in compliance with the relevant requirements and in accordance with the relevant guidance given in ISO/IEC 9646-7 [4] and ETS 300 406 [5]. The test notation used in the ATS is TTCN-3 (see ES 201 873-1 [6]). The following test specification and design considerations can be found in the body of the present document: • the overall test suite structure; • the testing architecture; • the test methods and port definitions; • the test configurations; • TTCN styles and conventions; • the partial PIXIT proforma; • the modules containing the TTCN-3 ATS. Annex A provides the Partial Implementation Extra Information for Testing (PIXIT) Proforma of the ATS. Annex B provides the Testing and Test Control Notation (TTCN-3) part of the ATS.
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2 References
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References are either specific (identified by date of publication and/or edition number or version number) or non-specific. For specific references, only the cited version applies. For non-specific references, the latest version of the reference document (including any amendments) applies. Referenced documents which are not found to be publicly available in the expected location might be found at http://docbox.etsi.org/Reference. NOTE: While any hyperlinks included in this clause were valid at the time of publication, ETSI cannot guarantee their long term validity.
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2.1 Normative references
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The following referenced documents are necessary for the application of the present document. [1] ETSI TS 129 214: "Universal Mobile Telecommunications System (UMTS); LTE; Policy and charging control over Rx reference point (3GPP TS 29.214 version 10.5.0 Release 10)". [2] ETSI TS 101 580-2: "IMS Network Testing (INT); Diameter Conformance testing for Rx interface; Part 2: Test Suite Structure (TSS) and Test Purposes (TP)". [3] ISO/IEC 9646-1: "Information technology - Open Systems Interconnection - Conformance testing methodology and framework - Part 1: General concepts". [4] ISO/IEC 9646-7: "Information technology - Open Systems Interconnection - Conformance testing methodology and framework - Part 7: Implementation Conformance Statements". [5] ETSI ETS 300 406: "Methods for testing and Specification (MTS); Protocol and profile conformance testing specifications; Standardization methodology". ETSI ETSI TS 101 580-3 V1.1.1 (2012-04) 6 [6] ETSI ES 201 873-1: "Methods for Testing and Specification (MTS); The Testing and Test Control Notation version 3; Part 1: TTCN-3 Core Language".
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2.2 Informative references
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The following referenced documents are not necessary for the application of the present document but they assist the user with regard to a particular subject area. Not applicable.
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3 Definitions and abbreviations
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3.1 Definitions
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For the purposes of the present document, the terms and definitions given in ISO/IEC 9646-7 [4] and TS 129 214 [1] apply.
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3.2 Abbreviations
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For the purposes of the present document, the abbreviations given in ISO/IEC 9646-1 [3], ISO/IEC 9646-7 [4], TS 129 214 [1] and the following apply: LLP Lower Layer Primitives SDP Session Description Protocol
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4 ATS conventions
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Test purposes of the present document address the Diameter protocol on the Rx interface.
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4.1 Test Architecture
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The test architecture defined in figures 1 and 2 apply. The communication covered by the test purposes of TS 101 580-2 [2] focuses on the Rx interface. For some tests the Gm interface is needed to trigger events on the Rx interface.
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4.1.1 Test configuration
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4.1.1.1 Configurations using Rx and Gm interface
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The Gm interface is located between UE and the SUT. The Rx interface is located between PCRF and the SUT. ETSI ETSI TS 101 580-3 V1.1.1 (2012-04) 7 Figure 1: Test architecture with IMS as SUT
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4.1.1.2 Configurations using Rx interface only
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The Rx interface is located between AF and the SUT. Figure 2: Test architecture with PCRF as SUT
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4.1.2 Interconnection of TS and SUT
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4.1.2.1 AF Role
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Figure 3 shows the interconnection of TS and SUT in terms of signalling message flows. Diameter component exists from two ports which are connected to Test System. Diameter messages are transferred over DIAM port. Lower Layer Primitives are transferred over LLP port. For execution of tests the Test Adapter shall be developed. There are two possibilities to communicate over Test Adapter: • ATS povide only Diameter messages; or • ATS provide Diameter messages and LL primitives. ETSI ETSI TS 101 580-3 V1.1.1 (2012-04) 8 Figure 3: Interconnection for AF role
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4.1.2.2 PCRF Role
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Figure 4 shows the interconnection of TS and SUT in terms of signalling message flows. Diameter component exists from two ports which are connected to Test System. Diameter messages are transferred over DIAM port. Lower Layer Primitives are transferred over LLPP port. For execution of tests the Test Adapter shall be developed. There are two possibilities to communicate over Test Adapter: • ATS povide only Diameter messages; or • ATS provide Diameter messages and LL primitives. port:LLPCR TS UE1 port:SIP port:DIAMP port:UE port:UE UE2 port:SIP PCRF port:PCRF SUT AF(IMS) map(p_imsComponent_ue1:SIPP, system: UE1); map(p_imsComponent_ue2:SIPP, system: UE2); map(p_imsComponent_pcrf:DIAMP, system: PCRF); map(p_imsComponent_pcrf:LLPP, system: LLPCRF); port:LLPP ETSI ETSI TS 101 580-3 V1.1.1 (2012-04) 9 Figure 4: Interconnection for PCRF role
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4.2 ATS structure
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4.2.1 Test case grouping
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The ATS structure is based on the Test Purposes for the Diameter protocol on the Rx interface as defined in TS 101 580-2 [2].
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4.2.2 Test case identifiers
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The test case names are built up according to the following scheme: "<TC>"_"<Group index>"_"<TC number>" NOTE: This naming scheme provides a 1-1 correspondence of TP identifiers as defined in TS 101 580-2 [2] and test case names. The TP identifier of TC_xxx_01 is TP_xxx_01. The test cases have been divided according to the functionalities into several groups. - TC_xxx_xx xxx port:LLIMS TS port:DIAMP AF(IMS) port:IMS SUT PCRF map(p_imsComponent_ims:DIAMP, system: IMS); map(p_imsComponent_ims:LLPP, system: LLIMS); port:LLPP ETSI ETSI TS 101 580-3 V1.1.1 (2012-04) 10 Annex A (normative): Partial PIXIT proforma Notwithstanding the provisions of the copyright clause related to the text of the present document, ETSI grants that users of the present document may freely reproduce the PIXIT proforma in this annex so that it can be used for its intended purposes and may further publish the completed PIXIT proforma. A.1 Introduction This partial PIXIT proforma contained in the present document is provided for completion, when the related Abstract Test Suite is to be used against the Implementation Under Test (IUT). The completed partial PIXIT will normally be used in conjunction with the completed PICS, as it adds precision to the information provided by the PICS. A.2 PIXIT items Each PIXIT item corresponds to a Module Parameter of the ATS. A.2.1 Diameter related PIXIT items Table A.1: Diameter related PIXIT items Id Identifier Type Description Configuration 1 PX_DIAM_LLP_ENABLED Boolean True, if Lower Layer Primitives and Port are enabled False, if Lower Layer Primitives and Port are disabled 2 PX_IPv6 Boolean True, if IPv6 addresses are used False, if IPv4 addresses are used IP addresses and port numbers 3 PX_DIAMETER_RX_ETS_IPADDR Charstring IP address of the test system 4 PX_DIAMETER_RX_SUT_IPADDR Charstring IP address of the system under test 5 PX_DIAMETER_RX_ETS_PORT Integer Port number of the test system 6 PX_DIAMETER_RX_ETS_PORT Integer Port number of the system under test 7 PX_UE_framedIpAddress Octetstring IPv4 address of the User Equipment having initiated the session that causes the Diameter messages exchange between AF and PCRF 8 PX_UE_framedIp6Address Octetstring IPv6 address of the User Equipment having initiated the session that causes the Diameter messages exchange between AF and PCRF Field Values 9 PX_SessionID UTF8String The Session-Id identifying a specific session 10 PX_OriginHost Charstring The Origin-Host identifying the endpoint that originates the Diameter messages 11 PX_OriginRealm Charstring The Origin-Realm identifying the Realm of the originator of any Diameter messages 12 PX_DestinationHost Charstring The Destination-Host identifying the endpoint to which the Diameter messages are destined 13 PX_DestinationRealm Charstring The Destination -Realm identifying the Realm of the destination of any Diameter messages 14 PX_ANCA_ipv4 IPv4Addr The Access-Network-Charging-Address in IPv4 format 15 PX_ANCA_ipv6 IPv6Addr The Access-Network-Charging-Address in IPv6 format ETSI ETSI TS 101 580-3 V1.1.1 (2012-04) 11 A.2.2 IMS related PIXIT items Table A.2 contains PIXIT items related to the communication between UE1 and UE via the AF that will cause the Diameter exchange between AF and PCRF. The UE1 and UE2 are simulated by the test system, the system under test is the AF. For testing the PCRF the values in table A.2 will not be used. Table A.2: IMS related PIXIT items Id Identifier Type Description P-CSCF IP parameters 1 PX_IMS_SUT_PCSCF1_PORT Integer SUT – P-CSCF port number to exchange SIP messages - connection point for the UE1. 2 PX_IMS_SUT_PCSCF1_IPADDR Charstring SUT – P-CSCF IP address to exchange SIP messages - connection point for the UE1. 4 PX_IMS_SUT_PCSCF1_HOME_D OMAIN Charstring SUT – P-CSCF domain - connection point for UE1. 5 PX_IMS_SUT_PCSCF2_PORT Integer SUT – P-CSCF port number to exchange SIP messages - connection point for the UE2. 6 PX_IMS_SUT_PCSCF2_IPADDR Charstring SUT – P-CSCF IP address to exchange SIP messages - connection point for the UE2. 7 PX_IMS_SUT_PCSCF2_HOME_D OMAIN Charstring SUT – P-CSCF domain - connection point for UE2. UE1 parameters 8 PX_IMS_TS_UE1_PORT Integer Port number used by UE1 to exchange SIP messages. 9 PX_IMS_TS_UE1_IPADDR Charstring IP address used by UE1 to exchange SIP messages. 10 PX_IMS_SUT_UE1_BEARER_IPA DDR Charstring IP address used by the test system to exchange media streams for the UE1. 11 PX_IMS_SUT_UE1_HOME_DOMAI N Charstring Identity of the UE1 local domain. 12 PX_IMS_SUT_UE1_PUBLIC_USER Charstring Identity of the UE1 local user. 13 PX_IMS_SUT_UE1_QOP Charstring Quoted string of one or more tokens indicating the "quality of protection" values for UE1. 14 PX_IMS_SUT_UE1_PRIVAT_USER NAME Charstring Private user name for UE1. 15 PX_IMS_SUT_UE1_PRIVAT_PASS WD Charstring Private password for UE1. 16 PX_IMS_SUT_UE1_REGISTRAR Charstring SUT- REGISTRAR domain of UE1. UE2 parameters 17 PX_IMS_TS_UE2_PORT Integer Port number used by UE2 to exchange SIP messages. 18 PX_IMS_TS_UE2_IPADDR Charstring IP address used by UE2 to exchange SIP messages. 19 PX_IMS_SUT_UE2_BEARER_IPA DDR Charstring IP address used by the test system to exchange media streams for UE2. 20 PX_IMS_SUT_UE2_HOME_DOMAI N Charstring Identity of the UE2 local domain. 21 PX_IMS_SUT_UE2_PUBLIC_USER Charstring Identity of the UE2 local user. 22 PX_IMS_SUT_UE2_QOP Charstring Quoted string of one or more tokens indicating the "quality of protection" values for UE2. 23 PX_IMS_SUT_UE2_PRIVAT_USER NAME Charstring Private user name for the UE2. 24 PX_IMS_SUT_UE2_PRIVAT_PASS WD Charstring Private password for the UE2. 25 PX_IMS_SUT_UE2_REGISTRAR Charstring SUT- REGISTRAR domain of UE2. SDP parameters 26 PX_SIP_SDP_dyn Charstring SDP dynamic port. 27 PX_SIP_SDP_encoding Charstring SDP media attribute encoding. 28 PX_SIP_REGISTER_AUTHENTICA TION_ENABLED Boolean True, if authentication for REGISTER messages is enabled. False, if authentication for REGISTER messages is disabled. ETSI ETSI TS 101 580-3 V1.1.1 (2012-04) 12 Annex B (informative): TTCN-3 library modules B.1 Electronic annex, zip file with TTCN-3 code The TTCN-3 library modules are contained in archive ts_10158003v010101p0.zip which accompanies the present document. This ATS has been produced using the Testing and Test Control Notation (TTCN) according to ES 201 873-1 [6]. ETSI ETSI TS 101 580-3 V1.1.1 (2012-04) 13 History Document history V1.1.1 April 2012 Publication
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1 Scope
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The present document provides the Test Suite Structure (TSS) and Test Purposes (TP) for the test specifications for the Diameter protocol on the Rx interface as specified in TS 129 214 [1] in compliance with the relevant requirements and in accordance with the relevant guidance given in ISO/IEC 9646-7 [4] and ETS 300 406 [5].
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2 References
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References are either specific (identified by date of publication and/or edition number or version number) or non-specific. For specific references, only the cited version applies. For non-specific references, the latest version of the reference document (including any amendments) applies. Referenced documents which are not found to be publicly available in the expected location might be found at http://docbox.etsi.org/Reference. NOTE: While any hyperlinks included in this clause were valid at the time of publication, ETSI cannot guarantee their long term validity.
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2.1 Normative references
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The following referenced documents are necessary for the application of the present document. [1] ETSI TS 129 214: "Universal Mobile Telecommunications System (UMTS); LTE; Policy and charging control over Rx reference point (3GPP TS 29.214 version 10.5.0 Release 10)". [2] ETSI TS 101 580-1: "IMS Network Testing (INT); Diameter Conformance testing for Rx interface; Part 1: Protocol Implementation Conformance Statement (PICS)". [3] ISO/IEC 9646-1: "Information technology - Open Systems Interconnection - Conformance testing methodology and framework - Part 1: General concepts". [4] ISO/IEC 9646-7: "Information technology - Open Systems Interconnection - Conformance testing methodology and framework - Part 7: Implementation Conformance Statements". [5] ETSI ETS 300 406: "Methods for testing and Specification (MTS); Protocol and profile conformance testing specifications; Standardization methodology". [6] IETF RFC 3588: "Diameter Base Protocol". [7] ETSI TS 129 213: "Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; Policy and charging control signalling flows and Quality of Service (QoS) parameter mapping (3GPP TS 29.213 version 10.3.0 Release 10)".
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2.2 Informative references
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The following referenced documents are not necessary for the application of the present document but they assist the user with regard to a particular subject area. Not applicable. ETSI ETSI TS 101 580-2 V1.1.1 (2012-04) 6
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3 Definitions and abbreviations
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3.1 Definitions
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For the purposes of the present document, the terms and definitions given in TS 129 214 [1] and the following apply: Abstract Test Method (ATM): Refer to ISO/IEC 9646-1 [3]. Abstract Test Suite (ATS): Refer to ISO/IEC 9646-1 [3]. Implementation Under Test (IUT): Refer to ISO/IEC 9646-1 [3]. Test Purpose (TP): Refer to ISO/IEC 9646-1 [3].
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3.2 Abbreviations
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For the purposes of the present document, the abbreviations given in TS 129 214 [1] and the following apply: TP Test Purpose TSS Test Suite Structure
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4 Test Suite Structure (TSS)
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4.1 TP naming convention
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TPs are numbered, starting at 001, within each group. Groups are organized according to the TSS. Table 1: TP identifier naming convention scheme Identifier: <TP>_<iut>_<scope>_<nn> <tp> = Test Purpose: fixed to "TP" <iut> = type of IUT: PCRF or AF <scope> = group IPS Initial Provisioning Session MSI Modification of Session Information GRP Gate Related Procedure ST Session Termination SN Subscription Notification TPE Traffic Plane Events <nn> = sequential number (01-99)
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4.2 Test strategy
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As the base standard TS 129 214 [1] contains no explicit requirements for testing, the TPs were generated as a result of an analysis of the base standard and the PICS specification TS 101 580-1 [2].
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4.3 TP structure
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Each TP has been written in a manner which is consistent with all other TPs. The intention of this is to make the TPs more readable and checkable. A particular structure has been used which is illustrated in table 2. This table should be read in conjunction with any TP, i.e. please use a TP as an example to facilitate the full comprehension of table 2. ETSI ETSI TS 101 580-2 V1.1.1 (2012-04) 7 Table 2: Structure of a single TP TP part Text Example Header <Identifier> see table 1 <clause number in base TS 129 214 [1]> clause 4.4.1 <PICS reference> A.2/3 Start point Ensure that the IUT in the <state> see RFC 3588 [6] clause 5.6 Open state and/or further actions before stimulus if the action is sending/receiving see below for message structure having sent an AA-Request Stimulus <trigger>, see below for message structure on receipt of a Capabilities-Exchange- Request (see note 2) or <goal> to require PCC supervision, etc. Reaction <action>. sends, saves, does, etc. if the action is sending see below for message structure <next action>, etc. Message structure <message type> Capabilities-Exchange-Answer, etc. (see note 2) a) containing a(n) <avp name> AVP b) indicating <coding of the field> and back to a) or b) (see note 3) Vendor-Id, etc. NOTE 1: Text in italics will not appear in TPs and text between <> is filled in for each TP and may differ from one TP to the next. NOTE 2: All messages shall be considered as "valid and compatible" unless otherwise specified in the test purpose. This includes the presence of all mandatory AVPs as specified in RFC 3588 [6] and in TS 129 214 [1], clause 5.6. NOTE 3: An AVP can be embedded into another AVP. This is expressed by indentations, e.g. if Message1 contains AVP1 and AVP2 where AVP1 has AVP3 embedded this will be expressed like this: sends/receives Message 1 containing AVP1 containing AVP3 indicating ... containing AVP2 indicating ...
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4.4 Test Purposes
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All PICS items referred to in this clause are as specified in TS 101 580-1 [2] unless indicated otherwise by another numbered reference. PICS items are only meant for test selection, therefore only PICS items with status optional or conditional are explicitly mentioned. Call flow information for described test purposes is specified in TS 129 213 [7].
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4.4.1 AF Role
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Test Selection: IUT takes the role of the AF; PICS A.2/1
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4.4.1.1 Initial Provisioning of Session Information for AF Role
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NOTE: In this clause it is assumed that two user equipments (UE) have registered to the IMS network via the AF (acting as P-CSCF) and that one UE has sent a SIP INVITE request addressed to a second UE towards the AF with the intention to establish a SIP session between the two UEs. ETSI ETSI TS 101 580-2 V1.1.1 (2012-04) 8 TP_AF_IPS_01 Standards Reference: 4.4.1 ¶ 1 PICS item: Test purpose: Ensure that the IUT in the Open state to indicate that a new AF session has been established and media information is available and requires PCC supervision, sends an AA-Request containing a Framed-IP-Address AVP indicating the full IP address of the UE containing a Media-Component-Description AVP containing the Flow-Status AVP. Comments: In the case of IPv6 the Framed-IP-Address AVP is replaced by the Framed-IPv6-Prefix AVP. TP_AF_IPS_02 Standards Reference: 4.4.1 (last two ¶ in clause) PICS item: Test purpose: Ensure that the IUT in the Open state having sent an AA-Request on receipt of an AA-Answer, containing a Result-Code AVP indicating DIAMETER_SUCCESS containing an Access-Network-Charging-Address AVP, does not reject the AA-Answer. Comments: TP_AF_IPS_03 Standards Reference: 4.4.1 ¶ 10 PICS item: A.3/2 Test purpose: Ensure that the IUT in the Open state to indicate that a new AF session has been established and media information is available and requires PCC supervision, for sponsored connectivity, sends an AA-Request containing a Sponsored-Connectivity-Data AVP containing a Sponsor-Identity AVP containing an Application-Service-Provider-Identity AVP. Comments:
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4.4.1.2 Modification of Session Information for AF Role
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NOTE: In this clause it is assumed that two user equipments (UE) have registered to the IMS network and have established a SIP session between each other via the AF (acting as P-CSCF) and the corresponding AF session has successfully been established with the exchange of AA-Request and AA-Answer messages. TP_AF_MSI_01 Standards Reference: 4.4.2 ¶ 1 PICS item: A.3/3 Test purpose: Ensure that the IUT in the Open state with an AF session successfully established, to indicate modification of the session information, sends an AA-Request containing a Media-Component-Description AVP. Comments: TP_AF_MSI_02 Standards Reference: 4.4.2 (last ¶ in clause) PICS item: A.3/3 Test purpose: Ensure that the IUT in the Open state with an AF session successfully established and having requested modification of the session information within an AA-Request, on receipt of an AA-Answer, containing a Result-Code AVP indicating DIAMETER_SUCCESS containing an Access-Network-Charging-Address AVP, does not reject the AA-Answer. Comments: ETSI ETSI TS 101 580-2 V1.1.1 (2012-04) 9 TP_AF_MSI_03 Standards Reference: 4.4.2 ¶ 5 PICS item: A.3/2, A.3/3 Test purpose: Ensure that the IUT in the Open state with an AF session successfully established, to indicate modification of the session information, for sponsored connectivity, sends an AA-Request containing a Sponsored-Connectivity-Data AVP containing a Sponsor_Identity AVP containing an Application-Service-Provider-Identity AVP. Comments:
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101 580-2
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4.4.1.3 Gate Related Procedures for AF Role
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TP_AF_GRP_01 Standards Reference: 4.4.3 ¶ 6 PICS item: A.3.4 Test purpose: Ensure that the IUT in the Open state having sent an AA-Request and having received an AA-Answer, on receipt of an RA-Request containing a Specific-Action AVP indicating INDICATION_OF_FAILD_RESOURCES_ALLOCATION, sends an RA-Answer. Comments:
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101 580-2
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4.4.1.4 Session Termination for AF Role
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TP_AF_ST_01 Standards Reference: 4.4.4 ¶ 1 PICS item: Test purpose: Ensure that the IUT in the Open state having sent an AA-Request and having received an AA-Answer containing a Result-Code AVPindicating DIAMETER_SUCCESS, on termination of the AF session, sends an ST-Request. Comments:
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101 580-2
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4.4.1.5 Subscription to Notification of Signaling Path Status for AF Role
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TP_AF_SN_01 Standards Reference: 4.4.5 ¶ 1 PICS item: A.3/6 Test purpose: Ensure that the IUT in the Open state to open an Rx diameter session and to subscribe to notifications of the status of the AF signalling transmission path, sends an AA-Request containing a Framed-IP-Address AVP indicating the full IP address of the UE containing a Specific-Action AVP indicating INDICATION_OF_RELEASE_OF_BEARER or INDICATION_OF_LOSS_OF_BEARER containing a Media-Component-Description AVP containing one Media-Sub-Component AVP containing a Flow-Usage AVP indicating AF_SIGNALLING containing a Media-Component-Number AVP indicating ‘0’. Comments: In the case of IPv6 the Framed-IP-Address AVP is replaced by the Framed-IPv6-Prefix AVP. IMS UE Actions: Registration of UE. ETSI ETSI TS 101 580-2 V1.1.1 (2012-04) 10 TP_AF_SN_02 Standards Reference: 4.4.5 ¶ 2 PICS item: A.3/6, NOT A.3/7 Test purpose: Ensure that the IUT in the Open state to open an Rx diameter session and to subscribe to notifications of the status of the AF signalling transmission path without the provision of AF signalling flow information, sends an AA-Request containing a Framed-IP-Address AVP indicating the full IP address of the UE containing a Specific-Action AVP indicating INDICATION_OF_RELEASE_OF_BEARER or INDICATION_OF_LOSS_OF_BEARER containing a Media-Component-Description AVP containing one Media-Sub-Component AVP containing a Flow-Number AVP indicating ‘0’ and not containing any other AVPs containing a Media-Component-Number AVP indicating ‘0’ and not containing any other AVPs. Comments: In the case of IPv6 the Framed-IP-Address AVP is replaced by the Framed-IPv6-Prefix AVP. IMS UE Actions: Registration of UE. TP_AF_SN_03 Standards Reference: 4.4.5 ¶ 5 PICS item: A.3/6 Test purpose: Ensure that the IUT in the Open state having established an Rx diameter session with subscription to notifications of the status of the AF signalling transmission path, to cancel this subscription, sends an ST-Request. Comments: IMS UE Actions: Registration and Deregistration of UE. TP_AF_SN_04 Standards Reference: 4.4.5 ¶ 5 PICS item: A.3/6, NOT A.3/7 Test purpose: Ensure that the IUT in the Open state having established an Rx diameter session with subscription to notifications of the status of the AF signalling transmission path without the provision of AF signalling flow information, to cancel this subscription, sends an ST-Request. Comments: IMS UE Actions: Registration and Deregistration of UE. TP_AF_SN_05 Standards Reference: 4.4.5a ¶ 2 PICS item: A.3/6, A.3/7 Test purpose: Ensure that the IUT in the Open state to open an Rx diameter session and to subscribe to notifications of the status of the AF signalling transmission path with the provision of AF signalling flow information, sends an AA-Request containing a Framed-IP-Address AVP indicating the full IP address of the UE containing a Media-Component-Description AVP containing one or more Media-Sub-Component AVP, each containing a Flow-Number AVP containing one or two Flow-Description AVP containing Flow-Usage AVP indicating AF_SIGNALLING containing Flow-Status AVP indicating ENABLED containing AF-Signalling-Protocol AVP indicating the signalling protocol between UE and AF containing Media-Component-Number AVP indicating ‘0’. Comments: In the case of IPv6 the Framed-IP-Address AVP is replaced by the Framed-IPv6-Prefix AVP. sd IMS UE Actions: Registration of UE. ETSI ETSI TS 101 580-2 V1.1.1 (2012-04) 11 TP_AF_SN_06 Standards Reference: 4.4.5a ¶ 5 PICS item: A.3/6, A.3/7 Test purpose: Ensure that the IUT in the Open state having established an Rx diameter session with subscription to notifications of the status of the AF signalling transmission path with the provision of AF signalling flow information, to cancel this subscription, sends an ST-Request. Comments: IMS UE Actions: Registration and Deregistration of UE.
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101 580-2
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4.4.1.6 Traffic Plane Events for AF Role
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TP_AF_TPE_01 Standards Reference: 4.4.6.1 ¶ 2 PICS item: A.3/8 Test purpose: Ensure that when the IUT in the Open state having established an Rx diameter session, on receipt of an AS-Request, sends an AS-Answer and an ST-Request. Comments: The ST-Request is sent to indicate the termination of the session.
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101 580-2
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4.4.2 PCRF Role
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Test Selection: IUT takes the role of the PCRF; PICS A.2/2
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101 580-2
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4.4.2.1 Initial Provisioning of Session Information for PCRF Role
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TP_PCRF_IPS_01 Standards Reference: 4.4.1 (last two ¶ in clause) PICS item: Test purpose: Ensure that the IUT in the Open state, on receipt of an AA-Request containing a Framed-IP-Address AVP containing a Media-Component-Description AVP containing the Flow-Status AVP, sends an AA-Answer containing a Result-Code AVP indicating DIAMETER_SUCCESS containing a Access-Network-Charging-Address AVP. Comments: In the case of IPv6 the Framed-IP-Address AVP is replaced by the Framed-IPv6-Prefix AVP. TP_PCRF_IPS_02 Standards Reference: 4.4.1 ¶ 1, 4, 5, 6, 7, 8, 10, 12 PICS item: Test purpose: Ensure that the IUT in the Open state, on receipt of an AA-Request containing a Framed-IP-Address AVP containing a Media-Component-Description AVP containing the Flow-Status AVP containing an AF-Application-Identifier AVP containing an AF-Charging-Identifier AVP containing a Service-URN AVP containing a MPS-Identifier AVP containing a Service-Info-Status AVP containing a Sponsored-Connectivity-Data AVP containing an Application-Service-Provider-Identity AVP containing a Sponsor-Identity AVP containing a Granted-Service-Unit AVP containing a Specific-Action AVP indicating USAGE_REPORT, sends an AA-Answer. Comments: In the case of IPv6 the Framed-IP-Address AVP is replaced by the Framed-IPv6-Prefix AVP. ETSI ETSI TS 101 580-2 V1.1.1 (2012-04) 12 TP_PCRF_IPS_03 Standards Reference: 4.4.1 first item of dashed list PICS item: NOT A.4/4 Test purpose: Ensure that the IUT in the Open state, on receipt of an AA-Request containing a Sponsored-Connectivity-Data AVP containing a Sponsor-Identity AVP containing an Application-Service-Provider-Identity AVP, sends an AA-Answer containing an Experimental-Result AVP containing an Experimental-Result-Code AVP indicating REQUEST_SERVICE_NOT_AUTHORISED. Comments:
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101 580-2
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4.4.2.2 Modification of Session Information for PCRF Role
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TP_PCRF_MSI_01 Standards Reference: 4.4.3 (last ¶ in clause) PICS item: Test purpose: Ensure that the IUT in the Open state having established an AF session, on receipt of an AA-Request modifying the session information containing a Framed-IP-Address AVP containing a Media-Component-Description AVP containing the Flow-Status AVP, sends an AA-Answer containing a Result-Code AVP indicating DIAMETER_SUCCESS containing a Access-Network-Charging-Address AVP. Comments: In the case of IPv6 the Framed-IP-Address AVP is replaced by the Framed-IPv6-Prefix AVP. TP_PCRF_MSI_02 Standards Reference: 4.4.3 ¶ 1, 2, 4, 5, 10 PICS item: Test purpose: E Ensure that the IUT in the Open state having established an AF session, on receipt of an AA-Request modifying the session information containing a Framed-IP-Address AVP containing a Media-Component-Description AVP containing a MPS-Identifier AVP containing a Service-Info-Status AVP indicating FINAL_SERVICE_INFORMATION containing a Sponsored-Connectivity-Data containing an Application-Service-Provider-Identity AVP containing a Sponsor-Identity AVP containing a Granted-Service-Unit AVP, sends an AA-Answer. Comments: In the case of IPv6 the Framed-IP-Address AVP is replaced by the Framed-IPv6-Prefix AVP.
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101 580-2
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4.4.2.3 Gate Related Procedures for PCRF Role
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TP_PCRF_GRP_01 Standards Reference: 4.4.3 ¶ 6 PICS item: Test purpose: Ensure that the IUT in the Open state having received an AA-Request and having sent an AA-Answer, to indicate a resource allocation failure during the modification of PCC/QoS rules, sends an RA-Request containing a Specific-Action AVP indicating INDICATION_OF_FAILD_RESOURCES_ALLOCATION. Comments: It may be impossible to trigger the condition for sending of the RA-Request. ETSI ETSI TS 101 580-2 V1.1.1 (2012-04) 13
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101 580-2
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4.4.2.4 Session Termination for PCRF Role
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TP_PCRF_ST_01 Standards Reference: 4.4.4 ¶ 2 PICS item: Test purpose: Ensure that the IUT in the Open state having established an AF session, on receipt of an ST-Request, sends an ST-Answer. Comments: TP_PCRF_ST_02 Standards Reference: 4.4.4 ¶ 4 PICS item: A.4/4 Test purpose: Ensure that the IUT in the Open state having established an AF session for sponsored connectivity, on receipt of an ST-Request, sends an ST-Answer containing a Sponsored-Connectivity-Data AVP containing an Used-Service-Unit AVP. Comments:
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101 580-2
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4.4.2.5 Subscription to Notification of Signaling Path Status for PCRF Role
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TP_PCRF_SN_01 Standards Reference: 4.4.5 ¶ 3 PICS item: Test purpose: Ensure that the IUT in the Open state, on receipt of an AA-Request subscribing to notifications of the status of the AF containing a Framed-IP-Address AVP containing a Specific-Action AVP indicating INDICATION_OF_RELEASE_OF_BEARER containing a Specific-Action AVP indicating INDICATION_OF_LOSS_OF_BEARER containing a Media-Component-Description AVP containing a Media-Sub-Component AVP containing a Flow-Usage AVP indicating AF_SIGNALLING containing a Flow-Number AVP indicating ‘0’ containing a Media-Component-Number AVP indicating ‘0’, sends a AA-Answer. Comments: In the case of IPv6 the Framed-IP-Address AVP is replaced by the Framed-IPv6-Prefix AVP. TP_PCRF_SN_02 Standards Reference: 4.4.5 ¶ 3 PICS item: Test purpose: Ensure that the IUT in the Open state, on receipt of an AA-Request subscribing to notifications of the status of the AF without the provision of AF signalling flow information, containing a Framed-IP-Address AVP indicating the full IP address of the UE containing a Specific-Action AVP indicating INDICATION_OF_RELEASE_OF_BEARER containing a Specific-Action AVP indicating INDICATION_OF_LOSS_OF_BEARER containing a Media-Component-Description AVP containing a Media-Sub-Component AVP containing a Flow-Number AVP indicating ‘0’ containing a Media-Component-Number AVP indicating ‘0’, sends an AA-Answer. Comments: In the case of IPv6 the Framed-IP-Address AVP is replaced by the Framed-IPv6-Prefix AVP. ETSI ETSI TS 101 580-2 V1.1.1 (2012-04) 14 TP_PCRF_SN_03 Standards Reference: 4.4.5 ¶ 5 PICS item: Test purpose: Ensure that the IUT in the Open state having established an Rx diameter session with subscription to notifications of the status of the AF signalling transmission path without the provision of AF signalling flow information, on receipt of an ST-Request, sends an ST-Answer. Comments: TP_PCRF_SN_04 Standards Reference: 4.4.5a ¶ 3 PICS item: Test purpose: Ensure that the IUT in the Open state, on receipt of an AA-Request subscribing to notifications of the status of the AF with the provision of AF signalling flow information, containing a Framed-IP-Address AVP indicating the full IP address of the UE containing a Media-Component-Description AVP containing a Media-Sub-Component AVP, containing a Flow-Number AVP containing two Flow-Description AVPs containing Flow-Usage AVP indicating AF_SIGNALLING containing Flow-Status AVP indicating ENABLED containing AF-Signalling-Protocol AVP indicating the signalling protocol between UE and AF containing Media-Component-Number AVP indicating ‘0’, sends an AA-Answer. Comments: In the case of IPv6 the Framed-IP-Address AVP is replaced by the Framed-IPv6-Prefix AVP. TP_PCRF_SN_05 Standards Reference: 4.4.5a ¶ 5 PICS item: Test purpose: Ensure that the IUT in the Open state having established an Rx diameter session with subscription to notifications of the status of the AF signalling transmission path with the provision of AF signalling flow information, on receipt of an ST-Request, sends an ST-Answer. Comments:
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101 580-2
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4.4.2.6 Traffic Plane Events for PCRF Role
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TP_PCRF_TPE_01 Standards Reference: 4.4.6.1 ¶ 1 PICS item: Test purpose: Ensure that when the IUT in the Open state having established an Rx diameter session, to indicate termination of an IP-CAN session, sends an AS- Request. Comments: It may be impossible to trigger the condition for sending of the AS-Request. TP_PCRF_TPE_01 Standards Reference: 4.4.6.1 ¶ 2 PICS item: Test purpose: Ensure that when the IUT in the Open state having established an Rx diameter session and having indicated termination of an IP-CAN session by sending an AS-Request, on receipt of an AS-Answer followed by an ST-Request, sends an ST-Answer. Comments: It may be impossible to trigger the condition for sending of the AS-Request. ETSI ETSI TS 101 580-2 V1.1.1 (2012-04) 15 History Document history V1.1.1 April 2012 Publication
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101 580-1
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1 Scope
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The present document provides the Protocol Implementation Conformance Statement (PICS) proforma for the test specifications for the Diameter protocol on the Rx interface as specified in TS 129 214 [1] in compliance with the relevant requirements and in accordance with the relevant guidance given in ISO/IEC 9646-7 [3] and ETS 300 406 [4]. The supplier of a protocol implementation which is claimed to conform to TS 129 214 [1] is required to complete a copy of the PICS proforma provided in annex A of the present document and is required to provide the information necessary to identify both the supplier and the implementation.
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2 References
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References are either specific (identified by date of publication and/or edition number or version number) or non-specific. For specific references, only the cited version applies. For non-specific references, the latest version of the reference document (including any amendments) applies. Referenced documents which are not found to be publicly available in the expected location might be found at http://docbox.etsi.org/Reference. NOTE: While any hyperlinks included in this clause were valid at the time of publication, ETSI cannot guarantee their long term validity.
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2.1 Normative references
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The following referenced documents are necessary for the application of the present document. [1] ETSI TS 129 214: "Universal Mobile Telecommunications System (UMTS); LTE; Policy and charging control over Rx reference point (3GPP TS 29.214 version 10.5.0 Release 10)". [2] ISO/IEC 9646-1: "Information technology - Open Systems Interconnection - Conformance testing methodology and framework - Part 1: General concepts". [3] ISO/IEC 9646-7: "Information technology - Open Systems Interconnection - Conformance testing methodology and framework - Part 7: Implementation Conformance Statements". [4] ETSI ETS 300 406: "Methods for testing and Specification (MTS); Protocol and profile conformance testing specifications; Standardization methodology". [5] ETSI TS 129 212: "Universal Mobile Telecommunications System (UMTS); LTE; Policy and charging control over Gx/Sd reference point (3GPP TS 29.212 version 10.5.0 Release 10)".
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2.2 Informative references
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The following referenced documents are not necessary for the application of the present document but they assist the user with regard to a particular subject area. Not applicable. ETSI ETSI TS 101 580-1 V1.1.1 (2012-04) 6
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3 Definitions and abbreviations
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3.1 Definitions
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For the purposes of the present document, the terms and definitions given in TS 129 214 [1] and the following apply: PICS proforma: document, in the form of a questionnaire, designed by the protocol specifier or conformance test suite specifier, which, when completed for an OSI implementation or system, becomes the PICS NOTE: See ISO/IEC 9646-1 [2]. Protocol Implementation Conformance Statement (PICS): statement made by the supplier of an Open Systems Interconnection (OSI) implementation or system, stating which capabilities have been implemented for a given OSI protocol NOTE: See ISO/IEC 9646-1 [2]. static conformance review: review of the extent to which the static conformance requirements are met by the IUT, accomplished by comparing the PICS with the static conformance requirements expressed in the relevant standard(s) NOTE: See ISO/IEC 9646-1 [2].
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3.2 Abbreviations
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For the purposes of the present document, the abbreviations given in TS 129 214 [1] and the following apply: PICS Protocol Implementation Conformance Statement
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101 580-1
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4 Conformance
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A PICS proforma which conforms to this PICS proforma specification shall be technically equivalent to annex A, and shall preserve the numbering and ordering of the items in annex A. A PICS which conforms to this PICS proforma specification shall: a) describe an implementation which claims to conform to TS 129 214 [1]; b) be a conforming ICS proforma which has been completed in accordance with the instructions for completion given in clause A.1; c) include the information necessary to uniquely identify both the supplier and the implementation. ETSI ETSI TS 101 580-1 V1.1.1 (2012-04) 7 Annex A (normative): PICS proforma Notwithstanding the provisions of the copyright clause related to the text of the present document, ETSI grants that users of the present document may freely reproduce the PICS proforma in this annex so that it can be used for its intended purposes and may further publish the completed PICS proforma. A.1 Guidance for completing the ICS proforma A.1.1 Purposes and structure The purpose of this PICS proforma is to provide a mechanism whereby a supplier of an implementation of the requirements defined in relevant specifications may provide information about the implementation in a standardised manner. The PICS proforma is subdivided into clauses for the following categories of information: - instructions for completing the PICS proforma; - identification of the implementation; - identification of the protocol; - PICS proforma tables (for example: Major capabilities, etc). A.1.2 Abbreviations and conventions This annex does not reflect dynamic conformance requirements but static ones. In particular, a condition for support of a PDU parameter does not reflect requirements about the syntax of the PDU (i.e. the presence of a parameter) but the capability of the implementation to support the parameter. In the sending direction, the support of a parameter means that the implementation is able to send this parameter (but it does not mean that the implementation always sends it). In the receiving direction, it means that the implementation supports the whole semantic of the parameter that is described in the main part of the present document. As a consequence, PDU parameter tables in this annex are not the same as the tables describing the syntax of a PDU in the reference specification. The PICS proforma contained in this annex is comprised of information in tabular form in accordance with the guidelines presented in ISO/IEC 9646-7 [3]. Item column The item column contains a number which identifies the item in the table. Item description column The item description column describes in free text each respective item (e.g. parameters, timers, etc.). It implicitly means "is <item description> supported by the implementation?". Reference column The reference column gives reference to the relevant sections in core specifications. ETSI ETSI TS 101 580-1 V1.1.1 (2012-04) 8 Status column The various status used in this annex are in accordance with the rules in table A.1. Table A.1: Key to status codes Status code Status name Meaning m mandatory The capability shall be supported. It is a static view of the fact that the conformance requirements related to the capability in the reference specification are mandatory requirements. This does not mean that a given behaviour shall always be observed (this would be a dynamic view), but that it shall be observed when the implementation is placed in conditions where the conformance requirements from the reference specification compel it to do so. For instance, if the support for a parameter in a sent PDU is mandatory, it does not mean that it shall always be present, but that it shall be present according to the description of the behaviour in the reference specification (dynamic conformance requirement). o optional The capability may or may not be supported. It is an implementation choice. n/a not applicable It is impossible to use the capability. No answer in the support column is required. c.<integer> conditional The requirement on the capability ("m", "o", "n/a") depends on the support of other optional or conditional items. <integer> is the identifier of the conditional expression. o.<integer> qualified optional For mutually exclusive or selectable options from a set. <integer> is the identifier of the group of options, and the logic of selection of the options. Mnemonic column The Mnemonic column contains mnemonic identifiers for each item. Support column The support column shall be filled in by the supplier of the implementation. The following common notations, defined in ISO/IEC 9646-7 [3], are used for the support column: Y or y supported by the implementation N or n not supported by the implementation N/A, n/a or - no answer required (allowed only if the status is N/A, directly or after evaluation of a conditional status) References to items For each possible item answer (answer in the support column) within the PICS proforma there exists a unique reference, used, for example, in the conditional expressions. It is defined as the table identifier, followed by a solidus character "/", followed by the item number in the table. EXAMPLE: A.5/4 is the reference to the answer of item 4 in table A.5. A.1.3 Instructions for completing the PICS proforma The supplier of the implementation may complete the PICS proforma in each of the spaces provided. More detailed instructions are given at the beginning of the different clauses of the PICS proforma. A.2 Identification of the Network Equipment Identification of the Network Equipment should be filled in so as to provide as much detail as possible regarding version numbers and configuration options. The product supplier information and client information should both be filled in if they are different. ETSI ETSI TS 101 580-1 V1.1.1 (2012-04) 9 A person who can answer queries regarding information supplied in the PICS should be named as the contact person. A.2.1 Date of the statement ......................................................................................................................................................................................... A.2.2 Network Equipment Under Test identification Name: ......................................................................................................................................................................................... ......................................................................................................................................................................................... Hardware configuration: ......................................................................................................................................................................................... ......................................................................................................................................................................................... ......................................................................................................................................................................................... Software configuration: ......................................................................................................................................................................................... ......................................................................................................................................................................................... ......................................................................................................................................................................................... A.2.3 Product supplier Name: ......................................................................................................................................................................................... Address: ......................................................................................................................................................................................... ......................................................................................................................................................................................... ......................................................................................................................................................................................... Telephone number: ......................................................................................................................................................................................... Facsimile number: ......................................................................................................................................................................................... E-mail address: ......................................................................................................................................................................................... Additional information: ......................................................................................................................................................................................... ......................................................................................................................................................................................... ......................................................................................................................................................................................... ETSI ETSI TS 101 580-1 V1.1.1 (2012-04) 10 A.2.4 Client Name: ......................................................................................................................................................................................... Address: ......................................................................................................................................................................................... ......................................................................................................................................................................................... ......................................................................................................................................................................................... Telephone number: ......................................................................................................................................................................................... Facsimile number: ......................................................................................................................................................................................... E-mail address: ......................................................................................................................................................................................... Additional information: ......................................................................................................................................................................................... ......................................................................................................................................................................................... ......................................................................................................................................................................................... A.2.5 PICS contact person Name: ......................................................................................................................................................................................... Telephone number: ......................................................................................................................................................................................... Facsimile number: ......................................................................................................................................................................................... E-mail address: ......................................................................................................................................................................................... Additional information: ......................................................................................................................................................................................... ......................................................................................................................................................................................... A.3 Identification of the protocol This PICS proforma applies to the following specification: • TS 129 214 [1]. ETSI ETSI TS 101 580-1 V1.1.1 (2012-04) 11 A.4 Global statement of conformance The implementation described in this PICS meets all the mandatory requirements of the referenced standard? [ ] Yes [ ] No NOTE: Answering "No" to this question indicates non-conformance to the protocol specification. Non-supported mandatory capabilities are to be identified in the PICS, with an explanation of why the implementation is non-conforming. Explanations may be entered in the comments field at the bottom of each table or on attached pages. In the tabulations which follow, all references are to TS 129 214 [1] unless another numbered reference is explicitly indicated. A.5 PICS proforma tables A.5.1 Roles Table A.2: Roles Item Roles Reference clause Status Support 1 AF 4.3.1 o.1 2 PCRF 4.3.2 o.1 o.1: At least one of these roles shall be supported. A.5.2 System Capabilities for AF The tables provided in this clause need only to be completed for AF implementations, where item A.2/1 above is supported. Table A.3: System Capabilities for AF Item Does the IUT support ... Reference clause Status Support 1 Procedures for the initial provision of service information? 4.4.1 m 2 The sponsored data connectivity feature? 4.4.1, TS 129 212 [5], 4.5.20 o 3 Procedures for the modification of the session information? 4.4.2 o 4 Gate related procedures? 4.4.3 o 5 Procedures for the termination of AF sessions? 4.4.4 m 6 Procedures for the subscription to the notification of the status of the AF Signalling transmission path? 4.4.5 o 7 Procedures for the provision of information about AF Signalling IP flows? 4.4.5a o 8 Procedures for the handling of traffic plane events? 4.6 m ETSI ETSI TS 101 580-1 V1.1.1 (2012-04) 12 A.5.3 System Capabilities for PCRF The tables provided in this clause need only to be completed for PCRF implementations, where item A.2/2 above is supported. Table A.4: System Capabilities for PCRF Item Does the IUT support ... Reference clause Status Support 1 Checking of the service information provided by the AF? 4.3.2 o 2 Usage of the subscription information as basis for the policy and charging control decisions? 4.3.2 o 3 Procedures for the initial provision of service information? 4.4.1 m 4 The sponsored data connectivity feature? 4.4.1, TS 129 212 [5], 4.5.20 o 5 Procedures for the modification of the session information? 4.4.2 m 6 Gate related procedures? 4.4.3 m 7 Procedures for the termination of AF sessions? 4.4.4 m 8 Procedures for the subscription to the notification of the status of the AF Signalling transmission path? 4.4.5 m 9 Procedures for the provision of information about AF Signalling IP flows? 4.4.5a m 10 Procedures for the handling of traffic plane events? 4.6 m ETSI ETSI TS 101 580-1 V1.1.1 (2012-04) 13 History Document history V1.1.1 April 2012 Publication
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2ff50194a724b72091ab07d6d4848112
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101 416
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1 Scope
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The present document specifies the stage 2 description of the SoLSA service, which gives the network operator the basis to offer subscribers or group of subscribers different services, different tariffs and different access rights depending on the geographical location of the subscriber, according to GSM 02.43.
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2ff50194a724b72091ab07d6d4848112
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101 416
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2 References
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The following documents contain provisions which, through reference in this text, constitute provisions of the present document. • References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. • For a specific reference, subsequent revisions do not apply. • For a non-specific reference, the latest version applies. • A non-specific reference to an ETS shall also be taken to refer to later versions published as an EN with the same number. • For this Release 1998 document, references to GSM documents are for Release 1998 versions (version 7.x.y).
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2ff50194a724b72091ab07d6d4848112
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101 416
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2.1 Normative references
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[1] GSM 02.43: "Digital cellular telecommunications system (Phase 2+); Support of Localised Service Area (SoLSA), Stage 1". [2] GSM 03.03: " Digital cellular telecommunications system (Phase 2+); Numbering, Addressing and Identification". [3] GSM 03.22: "Digital cellular telecommunications system (Phase 2+); Functions Related to Mobile Station (MS) in Idle Mode". [4] GSM 04.08: "Digital cellular telecommunications system (Phase 2+); Mobile Radio Interface – Layer 3 Specification". [5] GSM 04.60: "Digital cellular telecommunications system (Phase 2+); Mobile Station (MS) – Base Station System (BSS) interface; Radio Link Control/ Medium Access Control (RLC/MAC) protocol (GPRS)". [6] GSM 05.02: " Digital cellular telecommunications system (Phase 2+); Multiplexing and Multiple Access on the Radio Path". [7] GSM 05.08: "Digital cellular telecommunications system (Phase 2+); Radio subsystem link control". [8] GSM 08.08: "Digital cellular telecommunications system (Phase 2+);Mobile Switching Centre – Base Station System (MSC – BSS) interface Layer 3 specification". [9] GSM 08.18: "Digital cellular telecommunications system (Phase 2+); General Packet Radio Service (GPRS); Base Station System (BSS) – Serving GPRS Support Node (SGSN); BSS GPRS Protocol (BSSGP)". [10] GSM 08.58: "Digital cellular telecommunications system (Phase 2+); BSC-BTS Interface – Layer 3 Specification". [11] GSM 11.11: "Digital cellular telecommunications system (Phase 2+); Specification of the Subscriber Identity Module – Mobile Equipment (SIM-ME) Interface". ETSI ETSI TS 101 416 V7.2.0 (2000-01) 7 (GSM 03.73 version 7.2.0 Release 1998) [12] GSM 11.14: "Digital cellular telecommunications system (Phase 2+); Specification of the SIM Application Toolkit for the Subscriber Identity Module –Mobile Equipment".
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2ff50194a724b72091ab07d6d4848112
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101 416
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2.2 Informative references
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[13] GSM 10.43: "Digital cellular telecommunications system; SoLSA, Work item status".
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2ff50194a724b72091ab07d6d4848112
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101 416
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3 Definitions and abbreviations
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2ff50194a724b72091ab07d6d4848112
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101 416
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3.1 Definitions
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For the purposes of the present document, the following terms and definitions apply: Allowed localised service area: A Localised Service Area where the subscriber has allowed service. Current localised service area: The Localised Service Area of the serving cell to which the mobile station has subscription. In the case of overlapping LSAs, the LSA with the highest priority shall be selected as current LSA. If there are more than one overlapping LSA with the highest priority any of those LSAs may be selected as current LSA. Escape PLMN: A specific PLMN code that may be broadcast for non SoLSA compatible mobile stations that do not understand the exclusive access indicator. Localised Service Area: A Localised Service Area consists of a cell or a number of cells. The cells constituting a LSA may not necessarily provide continuous coverage. Network operator: Entity that provides the network operating elements and resources for the execution of the LSA service. Service provider: Entity that offers the LSA services for subscription. The network operator may be the service provider. Service subscriber: Mobile subscriber, which subscribes to the LSA service. In principle, if a network provides LSA service, all users are able to subscribe to this service.
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2ff50194a724b72091ab07d6d4848112
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101 416
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3.2 Abbreviations
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For the purposes of the present document, the following abbreviations apply: LSA Localised Service Area LSA ID Localised Service Area Identity CSE Camel Service Environment
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2ff50194a724b72091ab07d6d4848112
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101 416
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4 Main concepts
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2ff50194a724b72091ab07d6d4848112
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101 416
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4.1 Localised Service Area definition
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The network operator can define a Localised Service Area, LSA, consisting of a cell or a number of cells. It is possible for the network operator to set certain characteristics/attributes per LSA. Some LSA related attributes may be managed as part of cell management, e.g. exclusive access. The LSA is identified by a LSA ID. It shall be possible for the service subscriber to define a name on each of her allowed LSAs. ETSI ETSI TS 101 416 V7.2.0 (2000-01) 8 (GSM 03.73 version 7.2.0 Release 1998)
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