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3830b88c4707da898cf266974581f8bf	186 001-3	6.1 Test purposes for Basic Call
3830b88c4707da898cf266974581f8bf	186 001-3	6.1.1 Test purposes for SIP-SIP, Basic call, Successful
3830b88c4707da898cf266974581f8bf	186 001-3	6.1.1.1 Normal call establishment	SS___XX__01 NGN reference to: RFC 3261 [3] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/Successful. Selection criteria: Test purpose: Ensure that call establishment between UE A and UE B is handled correctly when reliable provisional responses and the precondition framework are not used. Ensure that the handling and mapping of the SDP parameters of the INVITE message is performed correctly. The call is released by the called user. Ensure that in the active call state the voice/data transfer on the media channels is performed correctly (e.g. testing QoS parameters). SIP Parameter values: INVITE: Require header without 100rel and precondition option tags sdp: PIXIT (Value should be taken from tables 1 and 2) 180 Ringing: Require header without 100rel Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE SS___XX__02 NGN reference to: RFC 3261 [3], RFC 3312 [17] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/Successful. Selection criteria: Test purpose: Ensure that call establishment between UE A and UE B is handled correctly when reliable provisional responses and the precondition framework are used. Ensure that the messages for the resource negotiation and reservation are delivered correctly. The call is released by the called user. Ensure that in the active call state the voice/data transfer on the media channels is performed correctly (e.g. testing QoS parameters). SIP Parameter values: INVITE: Supported header with 100rel and precondition option tags sdp: PIXIT (Value should be taken from tables 1 and 2) a=curr and a=des lines present 183 Session Progress: Require header with 100rel sdp: a=curr and a=des lines present UPDATE1 sdp: a=curr and a=des lines present ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 13 Comments: SIP UA A SUT SIP UA B INVITE INVITE Start resource negotiation/reservation 183 Session Progress 183 Session Progress PRACK PRACK 200 OK PRACK 200 OK PRACK UPDATE UPDATE1 200 OK UPDATE1 200 OK UPDATE End resource negotiation/reservation 180 Ringing 180 Ringing PRACK PRACK 200 OK PRACK 200 OK PRACK 200 OK INVITE 200 OK INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE SS___XX__03 NGN reference to: RFC 3261 [3] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/Successful. Selection criteria: Test purpose: Ensure that call establishment between UE A and UE B is handled correctly when reliable provisional responses and the precondition framework are not used. Ensure that the handling and mapping of the SDP parameters of the INVITE message is performed correctly. The call is released by the calling user. Ensure that in the active call state the voice/data transfer on the media channels is performed correctly (e.g. testing QoS parameters). SIP Parameter values: INVITE: Require header without 100rel and precondition option tags sdp: PIXIT (Value should be taken from tables 1 and 2) 180 Ringing: Require header without 100rel Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 14 SS___XX__04 NGN reference to: RFC 3261 [3], RFC 3312 [17] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/Successful. Selection criteria: Test purpose: Ensure that call establishment between UE A and UE B is handled correctly when reliable provisional responses and the precondition framework are used. Ensure that the messages for the resource negotiation and reservation are delivered correctly. The call is released by the calling user. Ensure that in the active call state the voice/data transfer on the media channels is performed correctly (e.g. testing QoS parameters). SIP Parameter values: INVITE: Supported header with 100rel and precondition option tags sdp: PIXIT (Value should be taken from tables 1 and 2) a=curr and a=des lines present 183 Session Progress: Require header with 100rel sdp: a=curr and a=des lines present UPDATE1 sdp: a=curr and a=des lines present Comments: SIP UA A SUT SIP UA B INVITE INVITE Start resource negotiation/reservation 183 Session Progress 183 Session Progress PRACK PRACK 200 OK PRACK 200 OK PRACK UPDATE UPDATE1 200 OK UPDATE1 200 OK UPDATE End resource negotiation/reservation 180 Ringing 180 Ringing PRACK PRACK 200 OK PRACK 200 OK PRACK 200 OK INVITE 200 OK INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE Table 1: Values for the test purpose SS___XX__01 to SS___XX__04 m= line b= line a= line VA <media> <transport> <fmt-list> <modifier>:<bandwidth-value> rtpmap:<dynamic-PT> <encoding name>/<clock rate>[/encoding parameters> See note VA_01 Audio RTP/AVP 0 N/A or up to 64 kbit/s N/A 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 VA_04 Audio RTP/AVP Dynamic PT N/A or up to 64 kbit/s rtpmap:<dynamic-PT> PCMA/8000 VA_05 Image Udptl t38 N/A or up to 64 kbit/s Based on T.38 VA_06 Image Tcptl t38 N/A or up to 64 kbit/s Based on T.38 NOTE: <bandwidth value> for <modifier> of AS is evaluated to be B kbit/s. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 15 Table 2: Values for test purposes SS___XX__01 and SS___XX__04 VARIABLE PT Encoding media type clock rate channels 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
3830b88c4707da898cf266974581f8bf	186 001-3	6.1.1.2 Codec negotiation	SS___CN__01 NGN reference to: RFC 3261 [3] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/Codec negotiation. Selection criteria: Test purpose: Ensure that the SUT, when the calling user decides during a session which was set-up without using the precondition mechanism to change the characteristics of the media session by sending a re-INVITE request, transports the re-INVITE request and the related 200 OK and ACK messages correctly. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: re-INVITE: sdp: PIXIT (Value should be taken from tables 1 and 2). Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK re-INVITE re-INVITE 200 OK re-INVITE 200 OK re-INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 16 SS___CN__02 NGN reference to: RFC 3261 [3], RFC 3312 [17] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/Codec negotiation. Selection criteria: Test purpose: Ensure that the IUT, when the calling user decides during a session which was set-up with using the precondition mechanism to change the characteristics of the media session by sending a re-INVITE request, transports the re-INVITE request and the related 200 OK and ACK messages correctly. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: re-INVITE: sdp: PIXIT (Value should be taken from tables 1 and 2). Comments: SIP UA A SUT SIP UA B INVITE INVITE Start resource negotiation/reservation 183 Session Progress SDP 183 Session Progress SDP PRACK PRACK 200 OK PRACK 200 OK PRACK UPDATE UPDATE 200 OK UPDATE 200 OK UPDATE End resource negotiation/reservation 180 Ringing 180 Ringing PRACK PRACK 200 OK PRACK 200 OK PRACK 200 OK INVITE 200 OK INVITE ACK ACK re-INVITE re-INVITE 200 OK re-INVITE 200 OK re-INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE NOTE: Re-Invite may need precondition, too (but is out of scope of this test case). SS___CN__03 NGN reference to: RFC 3261 [3] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/Codec negotiation. Selection criteria: Test purpose: Ensure that the SUT, when the called user decides during a session which was set-up without using the precondition mechanism to change the characteristics of the media session by sending a re-INVITE, transports the re-INVITE request and the related 200 OK and ACK messages correctly. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: re-INVITE: sdp: PIXIT (Value should be taken from tables 1 and 2). Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK re-INVITE re-INVITE 200 OK re-INVITE 200 OK re-INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 17 SS___CN__04 NGN reference to: RFC 3261 [3] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/Codec negotiation. Selection criteria: Test purpose: Ensure that the IUT, when the called user decides during a session which was set-up with using the precondition mechanism to change the characteristics of the media session by sending a re-INVITE, transports the re-INVITE request and the related 200 OK and ACK messages correctly. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: re-INVITE: sdp: PIXIT (Value should be taken from tables 1 and 2). Comments: SIP UA A SUT SIP UA B INVITE INVITE Start resource negotiation/reservation 183 Session Progress SDP 183 Session Progress SDP PRACK PRACK 200 OK PRACK 200 OK PRACK UPDATE UPDATE 200 OK UPDATE 200 OK UPDATE End resource negotiation/reservation 180 Ringing 180 Ringing PRACK PRACK 200 OK PRACK 200 OK PRACK 200 OK INVITE 200 OK INVITE ACK ACK re-INVITE re-INVITE re-INVITE re-INVITE 200 OK 200 OK ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE NOTE: Re-Invite may need precondition, too (but is out of scope of this test case). SS___CN__05 NGN reference to: RFC 3261 [3] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/Codec negotiation Selection criteria: Test purpose: Ensure that the SUT can correctly transport an SDP answer related to the SDP offer in the INVITE request in the 180 Ringing message, which is sent reliably. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: INVITE: sdp: PIXIT (Value should be taken from tables 1 and 2) Supported header with 100rel option tag 180 Ringing: sdp: PIXIT (Value should be taken from tables 1 and 2) Require header with 100rel option tag Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing with SDP answer 180 Ringing with SDP answer PRACK PRACK 200 OK PRACK 200 OK PRACK 200 OK INVITE 200 OK INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 18 SS___CN__06 NGN reference to: RFC 3261 [3] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/Codec negotiation. Selection criteria: Test purpose: Ensure that the SUT can correctly transport an SDP answer related to the SDP offer in the INVITE request in the 183 Session Progress message, which is sent reliably. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: INVITE: sdp: PIXIT (Value should be taken from tables 1 and 2) Supported header with 100rel option tag 183 Session Progress: sdp: PIXIT (Value should be taken from tables 1 and 2) Require header with 100rel option tag Comments: SIP UA A SUT SIP UA B INVITE INVITE 183 Session Progress with SDP answer 183 Session Progress with SDP answer PRACK PRACK 200 OK PRACK 200 OK PRACK 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE SS___CN__07 NGN reference to: RFC 3261 [3] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/Codec negotiation. Selection criteria: Test purpose: Ensure that the SUT can correctly transport an SDP answer related to the SDP offer in the INVITE request in the 200 OK message. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: INVITE: sdp: PIXIT (Value should be taken from tables 1 and 2) 200 OK: sdp: PIXIT (Value should be taken from tables 1 and 2) Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE with SDP answer 200 OK INVITE with SDP answer ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 19
3830b88c4707da898cf266974581f8bf	186 001-3	6.1.1.3 UPDATE method	SS___UP__01 NGN reference to: RFC 3261 [3], RFC 3311 [18] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/update. Selection criteria: Test purpose: Ensure that the SUT, when the calling user decides during a session which was set-up without using the precondition mechanism to change the characteristics of the media session by sending an UPDATE request, transports the UPDATE request and the related 200 OK and ACK messages correctly. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: UPDATE: sdp: PIXIT (Value should be taken from tables 1 and 2). Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK UPDATE UPDATE 200 OK UPDATE 200 OK UPDATE Check media BYE BYE 200 OK BYE 200 OK BYE SS___UP__02 NGN reference to: RFC 3261 [3], RFC 3312 [17], RFC 3311 [18] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/update. Selection criteria: Test purpose: Ensure that the IUT, when the calling user decides during a session which was set-up with using the precondition mechanism to change the characteristics of the media session by sending an UPDATE request, transports the UPDATE request and the related 200 OK message correctly. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: UPDATE: sdp: PIXIT (Value should be taken from tables 1 and 2). Comments: SIP UA A SUT SIP UA B INVITE INVITE Start resource negotiation/reservation 183 Session Progress SDP 183 Session Progress SDP PRACK PRACK 200 OK PRACK 200 OK PRACK UPDATE UPDATE 200 OK UPDATE 200 OK UPDATE End resource negotiation/reservation 180 Ringing 180 Ringing PRACK PRACK 200 OK PRACK 200 OK PRACK 200 OK INVITE 200 OK INVITE ACK ACK UPDATE UPDATE 200 OK UPDATE 200 OK UPDATE Check media BYE BYE 200 OK BYE 200 OK BYE NOTE: UPDATE after session establishment may need precondition, too (but is out of scope of this test case). ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 20 SS___UP__03 NGN reference to: RFC 3261 [3], RFC 3311 [18] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/update. Selection criteria: Test purpose: Ensure that the SUT, when the called user decides during a session which was set-up without using the precondition mechanism to change the characteristics of the media session by sending an UPDATE request, transports the UPDATE request and the related and ACK message correctly. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: UPDATE: sdp: PIXIT (Value should be taken from tables 1 and 2). Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK UPDATE UPDATE 200 OK UPDATE 200 OK UPDATE Check media BYE BYE 200 OK BYE 200 OK BYE SS___UP__04 NGN reference to: RFC 3261 [3], RFC 3312 [17], RFC 3311 [18] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/update. Selection criteria: Test purpose: Ensure that the IUT, when the called user decides during a session which was set-up with using the precondition mechanism to change the characteristics of the media session by sending an UPDATE request, transports the UPDATE request and the related 200 OK message correctly. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: UPDATE: sdp: PIXIT (Value should be taken from tables 1 and 2). Comments: SIP UA A SUT SIP UA B INVITE INVITE Start resource negotiation/reservation 183 Session Progress SDP 183 Session Progress SDP PRACK PRACK 200 OK PRACK 200 OK PRACK UPDATE UPDATE 200 OK UPDATE 200 OK UPDATE End resource negotiation/reservation 180 Ringing 180 Ringing PRACK PRACK 200 OK PRACK 200 OK PRACK 200 OK INVITE 200 OK INVITE ACK ACK UPDATE UPDATE 200 OK UPDATE 200 OK UPDATE Check media BYE BYE 200 OK BYE 200 OK BYE NOTE: UPDATE after session establishment may need precondition, too (but is out of scope of this test case). ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 21 SS___UP__05 NGN reference to: RFC 3261 [3], RFC 3312 [17], RFC 3311 [18] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/update. Selection criteria: Test purpose: Ensure that the IUT, after an SDP offer in an INVITE request from the calling user has been answered in a reliably sent 180 Ringing message by the called user, when the calling user decides before the end of session establishment to change the characteristics of the media session by sending an UPDATE request, transports the UPDATE request and the related 200 OK message correctly. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: INVITE: Allow including UPDATE Supported header include 100rel sdp offer1 180 Ringing: Allow including UPDATE Require header include 100rel sdp answer1 UPDATE: sdp offer2 200 OK UPDATE: sdp answer2 Comments: SIP UA A SUT SIP UA B INVITE (sdp offer1) INVITE (sdp offer1) 180 Ringing (sdp answer1) 180 Ringing (sdp answer1) PRACK PRACK 200 OK PRACK 200 OK PRACK UPDATE (sdp offer2) UPDATE (sdp offer2) 200 OK UPDATE (sdp answer2) 200 OK UPDATE (sdp answer2) 200 OK INVITE 200 OK INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 22 SS___UP__06 NGN reference to: RFC 3261 [3], RFC 3312 [17], RFC 3311 [18] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/update. Selection criteria: Test purpose: Ensure that the IUT, after an SDP offer in an INVITE request from the calling user has been answered in a reliably sent 180 Ringing message by the called user, when the called user decides before the end of session establishment to change the characteristics of the media session by sending an UPDATE request, transports the UPDATE request and the related 200 OK message correctly. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: INVITE: Allow including UPDATE Supported header include 100rel sdp offer1 180 Ringing: Allow including UPDATE Require header include 100rel sdp answer1 UPDATE: sdp offer2 200 OK UPDATE: sdp answer2 Comments: SIP UA A SUT SIP UA B INVITE (sdp offer1) INVITE (sdp offer1) 180 Ringing (sdp answer1) 180 Ringing (sdp answer1) PRACK PRACK 200 OK PRACK 200 OK PRACK UPDATE (sdp offer2) UPDATE (sdp offer2) 200 OK UPDATE (sdp answer2) 200 OK UPDATE (sdp answer2) 200 OK INVITE 200 OK INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 23 SS___UP__07 NGN reference to: RFC 3261 [3], RFC 3312 [17], RFC 3311 [18] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/update. Selection criteria: Test purpose: Ensure that the IUT, after an INVITE request without SDP offer has been sent by the calling user and an SDP offer from the called user reliably sent in a 180 Ringing message has been answered by the calling user in the PRACK message, when the calling user decides before the end of session establishment to change the characteristics of the media session by sending an UPDATE request, transports the UPDATE request and the related 200 OK message correctly. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: INVITE: Allow including UPDATE Supported header include 100rel sdp not present1 180 Ringing: Allow including UPDATE Require header include 100rel sdp offer1 PRACK: sdp answer1 UPDATE: sdp offer2 200 OK UPDATE: sdp answer2 Comments: SIP UA A SUT SIP UA B INVITE (no sdp) INVITE (no sdp) 180 Ringing (sdp offer1) 180 Ringing (sdp offer1) PRACK(sdp answer1) PRACK(sdp answer1) 200 OK PRACK 200 OK PRACK UPDATE (sdp offer2) UPDATE (sdp offer2) 200 OK UPDATE (sdp answer2) 200 OK UPDATE (sdp answer2) 200 OK INVITE 200 OK INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 24 SS___UP__08 NGN reference to: RFC 3261 [3], RFC 3312 [17], RFC 3311 [18] TS 124 229 [10], clauses 5.1.3, 5.1.4 TSS reference: SIP-SIP/Basic_call/update. Selection criteria: Test purpose: Ensure that the IUT, after an INVITE request without SDP offer has been sent by the calling user and an SDP offer from the called user reliably sent in a 180 Ringing message has been answered by the calling user in the PRACK message, when the called user decides before the end of session establishment to change the characteristics of the media session by sending an UPDATE request, transports the UPDATE request and the related 200 OK message correctly. Ensure that the voice/data transfer on the media channels with the re-negotiated media is performed correctly (e.g. testing QoS parameters). SIP Parameter values: INVITE: Allow including UPDATE Supported header:100rel sdp not present1 180 Ringing: Allow including UPDATE Require header include 100rel sdp offer1 PRACK: sdp answer1 UPDATE: sdp offer2 200 OK UPDATE: sdp answer2 Comments: SIP UA A SUT SIP UA B INVITE (no sdp) INVITE (no sdp) 180 Ringing (sdp offer1) 180 Ringing (sdp offer1) PRACK(sdp answer1) PRACK(sdp answer1) 200 OK PRACK 200 OK PRACK UPDATE (sdp offer2) UPDATE (sdp offer2) 200 OK UPDATE (sdp answer2) 200 OK UPDATE (sdp answer2) 200 OK INVITE 200 OK INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE
3830b88c4707da898cf266974581f8bf	186 001-3	6.1.2 Test purposes for SIP-SIP, Basic call, Unsuccessful	SS___XX_U01 NGN reference to: RFC 3261 [3] TS 124 229 [10], clause 5.2.6.3 TSS reference: SIP-SIP/Basic_call/Unsuccessful. Selection criteria: Test purpose: Ensure that the SUT on receipt of a 503 Service Unavailable message from the called user, sends a 500 Server Internal Error or a a 503 Service Unavailable message to the calling user. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B INVITE INVITE 500 Server Internal Error or 503 Service Unavailable 503 Service Unavailable ACK ACK ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 25 SS___XX_U02 NGN reference to: RFC 3261 [3] TS 124 229 [10], clause 5.2.6.3 TSS reference: SIP-SIP/Basic_call/Unsuccessful. Selection criteria: Test purpose: Ensure that the SUT delivers a 486 Busy Here message from the called to the calling user. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B INVITE INVITE 486 Busy Here 486 Busy Here ACK ACK SS___XX_U03 NGN reference to: RFC 3261 [3] TSS reference: SIP-SIP/Basic_call/Unsuccessful. Selection criteria: Test purpose: Ensure that when there is no answer from the called user (there is no response to the INVITE messages), the SUT initiates call clearing to the calling user with a 480 Temporarily Unavailable or 408 Request Timeout message. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B INVITE INVITE 100 Trying INVITE INVITE INVITE INVITE INVITE INVITE INVITE 480 Temporarily Unavailable or 408 Request Timeout ACK NOTE: No 100 Trying response by UA-B. SS___XX_U04 NGN reference to: RFC 3261 [3] TS 124 229 [10], clause 5.2.6.3 TSS reference: SIP-SIP/Basic_call/Unsuccessful. Selection criteria: Test purpose: Ensure that the SUT delivers a 480 Temporarily Unavailable message from the alerting called user to the calling user (do not disturb service). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 480 Temporary unavaible 480 Temporary unavaible ACK ACK ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 26 SS___XX_U05 NGN reference to: RFC 3261 [3] TS 124 229 [10] TSS reference: SIP-SIP/Basic_call/Unsuccessful. Selection criteria: Test purpose: Ensure that when the calling user clears the call with a CANCEL message before receiving an answer to the previously sent INVITE request from the called user, the SUT delivers the CANCEL message to the called user. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B INVITE INVITE 100 Trying CANCEL CANCEL 200 OK CANCEL 200 OK CANCEL 487 Request Terminated 487 Request Terminated ACK ACK NOTE: No 100 Trying response by UA-B. SS___XX_U06 NGN reference to: RFC 3261 [3] TS 124 229 [10] TSS reference: SIP-SIP/Basic_call/Unsuccessful. Selection criteria: Test purpose: Ensure that the IUT, when the calling user decides during a session to change the characteristics of the media session by sending a re-INVITE request and the Re-INVITE is rejected by the called user with a 488 Not Acceptable Here, delivers the 488 Not Acceptable Here to the calling user. Ensure that the voice/data transfer on the media channels with the original media is still performed correctly (e.g. testing QoS parameters). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP SUP SIP INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Re-INVITE Re-INVITE offer 488 Not Acceptable Here 488 Not Acceptable Here Communication BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 27 SS___XX_U07 NGN reference to: RFC 3261 [3] TS 124 229 [10] TSS reference: SIP-SIP/Basic_call/Unsuccessful. Selection criteria: Test purpose: Ensure that the IUT, when the called user decides during a session to change the characteristics of the media session by sending a re-INVITE request and the Re-INVITE is rejected by the calling user with a 488 Not Acceptable Here, delivers the 488 Not Acceptable Here to the called user. Ensure that the voice/data transfer on the media channels with the original media is still performed correctly (e.g. testing QoS parameters). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) No 100 rel; Case b) Supported: 100 rel; Case c) Supported: 100 rel and precondition. Comments: SIP SUT SIP INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Re-INVITE Re-INVITE 488 Not Acceptable Here 488 Not Acceptable Here Communication BYE BYE 200 OK BYE 200 OK BYE SS___XX_U08 NGN reference to: RFC 3261 [3] TS 124 229 [10] TSS reference: SIP-SIP/Basic_call/Unsuccessful. Selection criteria: Test purpose: Ensure that when there is no answer from the called user ("no user responding"), the SUT initiates call clearing to the called user with a CANCEL request and to the calling user with a 408 Request Timeout, a 480 Temporarily Unavailable or a 487 Request Terminated response. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP SUT SIP INVITE INVITE 180 Ringing 180 Ringing Timeout timer C 408/480/487 CANCEL ACK 200 OK CANCEL 487 Request Terminated ACK ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 28
3830b88c4707da898cf266974581f8bf	186 001-3	6.2 Test purposes for SIP-SIP, Supplementary services
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.1 Test purposes for OIP	SS___XXSS_OIP01 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIP. Selection criteria: The originating user subscribes to OIR "temporary mode" default "not restricted". The terminating user subscribes to OIP. Test purpose: Ensure that, when no P-Preferred-Identity header field is provided by the originating UE in the INVITE request, the terminating user receives a P-Asserted-Identity based on the default public user identity associated with the originating UE. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B INVITE INVITE SS___XXSS_OIP02 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIP. Selection criteria: The originating user subscribes to OIR "temporary mode" default "not restricted". The terminating user subscribes to OIP service. Test purpose: Ensure that, when the Privacy header field is set to "none" and no P-Preferred-Identity header field 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. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "none" Comments: SIP UA A SUT SIP UA B INVITE INVITE SS___XXSS_OIP03 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIP. Selection criteria: The originating user subscribes to OIR "temporary mode" default "restricted". The terminating user subscribes to OIP service. Test purpose: Ensure that, when the Privacy header field is set to "none" and no P-Preferred-Identity header field 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. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "none" Comments: SIP UA A SUT SIP UA B INVITE INVITE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 29 SS___XXSS_OIP04 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIP. Selection criteria: The originating user subscribes to OIR "temporary mode" default "not restricted". The terminating user subscribes to OIP service. Test purpose: Ensure that, when no Privacy header field is inserted and a P-Preferred-Identity header field is provided by the originating UE, but the identity information in the P-Preferred- Identity 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. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE INVITE SS___XXSS_OIP05 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIP. Selection criteria: The originating user subscribes to OIR "temporary mode" default "not restricted". The terminating user subscribes to OIP service. Test purpose: Ensure that, when the Privacy header field is set to "none" and a P-Preferred-Identity header field is provided by the originating UE, but the identity information in the P- Preferred-Identity 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. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "none" Comments: SIP UA A SUT SIP UA B INVITE INVITE SS___XXSS_OIP06 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIP. Selection criteria: The originating user subscribes to OIR "temporary mode" default "restricted". The terminating user subscribes to OIP service. Test purpose: Ensure that, when the Privacy header field is set to "none" and a P-Preferred-Identity header field is provided by the originating UE, but the identity information in the P- Preferred-Identity 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. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "none" Comments: SIP UA A SUT SIP UA B INVITE INVITE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 30 SS___XXSS_OIP07 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIP. Selection criteria: The originating user subscribes to OIR "temporary mode" default "not restricted". The terminating user subscribes to OIP service. Test purpose: Ensure that, when no Privacy header field is inserted and a P-Preferred-Identity header field is provided by the originating UE (the identity information in the P-Preferred-Identity must be present in the set of registered public identities of the originating UE and it shall be different from the default public user identity), the terminating UE receives a P-Asserted-Identity based on the information provided by the originating UE. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE INVITE SS___XXSS_OIP08 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIP. Selection criteria: The originating user subscribes to OIR "temporary mode" default "not restricted". The terminating user subscribes to OIP service. Test purpose: Ensure that, when the Privacy header field is set to "none" and a P-Preferred-Identity header field is provided by the originating UE (the identity information in the P-Preferred- Identity must be present in the set of registered public identities of the originating UE and it shall be different from the default public user identity), the terminating UE receives a P-Asserted-Identity based on the information provided by the originating UE. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "none" Comments: SIP UA A SUT SIP UA B INVITE INVITE SS___XXSS_OIP09 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIP. Selection criteria: The originating user subscribes to OIR "temporary mode" default "restricted". The terminating user subscribes to OIP service. Test purpose: Ensure that, when the Privacy header field is set to "none" and a P-Preferred-Identity header field is provided by the originating UE (the identity information in the P-Preferred- Identity must be present in the set of registered public identities of the originating UE and it shall be different from the default public user identity), the terminating UE receives a P-Asserted-Identity based on the information provided by the originating UE. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "none" Comments: SIP UA A SUT SIP UA B INVITE INVITE SS___XXSS_OIP10 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIP. Selection criteria: The terminating user is not subscribed to OIP service. Test purpose: Ensure that, for any INVITE request, the terminating user receives no P-Asserted- Identity header field and no Privacy header field. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE INVITE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 31
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.2 Test purposes for OIR	SS___XXSS_OIR01 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.1.2, 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIR. Selection criteria: The originating user subscribes to OIR "temporary mode" default "restricted". Also, the restricted type is set to "restrict the asserted identity" (see table 1, TS 124 407 [14], clause 4.3.1.2). The terminating user subscribes to OIP service. Test purpose: Ensure that, when no Privacy header field is inserted by the originating UE in the INVITE request, the terminating UE receives an INVITE message where the From header field is set to "anonymous", the Privacy header field is set to "id" and no P-Asserted-Identity header is received. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE INVITE SS___XXSS_OIR02 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.1.2, 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIR. Selection criteria: The originating user subscribes to OIR "temporary mode" default "restricted". Also, the restricted type is set to "restrict all private information appearing in headers" (see table 1, TS 124 407 [14], clause 4.3.1.2). The terminating user subscribes to OIP service. Test purpose: Ensure that, when no Privacy header field is inserted by the originating UE in the INVITE request, the terminating UE receives an INVITE message where the From header field is set to "anonymous", the Privacy header field is set to "header" and no P-Asserted-Identity header is received. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE INVITE SS___XXSS_OIR03 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIR. Selection criteria: The originating user subscribes to OIR "temporary mode" default "restricted". The terminating user subscribes to OIP service. Test purpose: Ensure that, when the Privacy header field is set to "id" by the originating UE in the INVITE request, the terminating UE receives an INVITE message where the From header field is set to "anonymous", the Privacy header field is set to "id" or "header" and no P-Asserted- Identity header is received. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "id" Comments: SIP UA A SUT SIP UA B INVITE INVITE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 32 SS___XXSS_OIR04 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIR. Selection criteria: The originating user subscribes to OIR "temporary mode" default "restricted". The terminating user subscribes to OIP service. Test purpose: Ensure that, when the Privacy header field is set to "header" by the originating UE in the INVITE request, the terminating UE receives an INVITE message where the From header field is set to "anonymous", the Privacy header field is set to "id" or "header" and no P-Asserted- Identity header is received. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "header" Comments: SIP UA A SUT SIP UA B INVITE INVITE SS___XXSS_OIR05 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIR. Selection criteria: The originating user subscribes to OIR "temporary mode" default "not restricted". The terminating user subscribes to OIP service. Test purpose: Ensure that, when the Privacy header field is set to "id" and the From header field is set to "anonymous" by the originating UE in the INVITE request, the terminating UE receives an INVITE message where the From header field is set to "anonymous", the Privacy header field is set to "id" and no P-Asserted-Identity header is received. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "id" From header field is set to: From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag= xxxxxxx Comments: SIP UA A SUT SIP UA B INVITE INVITE SS___XXSS_OIR06 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIR. Selection criteria: The originating user subscribes to OIR "temporary mode" default "not restricted". The terminating user subscribes to OIP service. Test purpose: Ensure that, when the Privacy header field is set to "header" and the From header field is set to "anonymous" by the originating UE in the INVITE request, the terminating UE receives an INVITE message where the From header field is set to "anonymous", the Privacy header field is set to "header" and no P-Asserted-Identity header is received. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "header" From header field is set to: From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag= xxxxxxx Comments: SIP UA A SUT SIP UA B INVITE INVITE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 33 SS___XXSS_OIR07 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIR. Selection criteria: The originating user subscribes to OIR permanent mode. Also, the restricted type is set to "restrict the asserted identity" (see table 1, TS 124 407 [14], clause 4.3.1.2). The terminating user subscribes to OIP service. Test purpose: Ensure that, when no Privacy header field is inserted by the originating UE in the INVITE request, the terminating UE receives an INVITE message where the From header field is set to "anonymous", the Privacy header field is set to "id" and no P-Asserted-Identity header is received. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE INVITE SS___XXSS_OIR08 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIR. Selection criteria: The originating user subscribes to OIR permanent mode. Also, the restricted type is set to "restrict all private information appearing in headers" (see table 1, TS 124 407 [14], clause 4.3.1.2). The terminating user subscribes to OIP service. Test purpose: Ensure that, when no Privacy header field is inserted by the originating UE in the INVITE request, the terminating UE receives an INVITE message where the From header field is set to "anonymous", the Privacy header field is set to "header" and no P-Asserted-Identity header is received. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE INVITE SS___XXSS_OIR09 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIR. Selection criteria: The originating user subscribes to OIR permanent mode. The terminating user subscribes to OIP service. Test purpose: Ensure that, when the Privacy header field is set to "id" by the originating UE in the INVITE request, the terminating UE receives an INVITE message where the From header field is set to "anonymous", the Privacy header field is set to "id" or "header" and no P-Asserted- Identity header is received. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "id" Comments: SIP UA A SUT SIP UA B INVITE INVITE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 34 SS___XXSS_OIR10 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIR. Selection criteria: The originating user subscribes to OIR permanent mode. The terminating user subscribes to OIP service. Test purpose: Ensure that, when the Privacy header field is set to "header" by the originating UE in the INVITE request, the terminating UE receives an INVITE message where the From header field is set to "anonymous", the Privacy header field is set to "id" or "header" and no P-Asserted- Identity header is received. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "header" Comments: SIP UA A SUT SIP UA B INVITE INVITE SS___XXSS_OIR11 OIP/OIR reference to: TS 124 407 [14], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/OIP. Selection criteria: The originating user subscribes to OIR "permanent mode". The terminating user subscribes to OIP service. Test purpose: Ensure that, when the Privacy header field is set to "none" by the originating UE in the INVITE request, the terminating UE receives an INVITE message where the From header field is set to "anonymous", the Privacy header field is set to "id" or "header" and no P-Asserted- Identity header is received. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "none" Comments: SIP UA A SUT SIP UA B INVITE INVITE
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.3 Test purposes for TIP	SS___XXSS_TIP01 TIP/TIR reference to: TS 124 508 [24], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/TIP. Selection criteria: The originating user subscribes to TIP service. Test purpose: Ensure that, when the option tag "from-change" in the Supported header field is provided by the originating UE in the INVITE request: the originating UE receives, in the 2xx SIP response, a P-Asserted-Identity header field with a valid public user identity of the terminating UE. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE (P-Asserted-Identity) 200 OK INVITE ACK ACK ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 35 SS___XXSS_TIP02 TIP/TIR reference to: TS 124 508 [24], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/TIP. Selection criteria: The originating user subscribes to TIP service. The terminating user subscribes to TIR "temporary mode" default "not restricted". Test purpose: Ensure that, when the option tag "from-change" in the Supported header field is provided by the originating UE in the INVITE request: the originating UE receives, in the 2xx SIP response, a P-Asserted-Identity header field with a valid public user identity of the terminating UE. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE (P-Asserted-Identity) 200 OK INVITE ACK ACK SS___XXSS_TIP03 TIP/TIR reference to: TS 124 508 [24], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/TIP. Selection criteria: The originating user subscribes to TIP service. The terminating user subscribes TIR "temporary mode" default "not restricted". Test purpose: 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 Privacy header field is set to "none" by the terminating UE in the 2xx SIP response: the originating UE receives, in the 2xx SIP response, a P-Asserted-Identity header field with a valid public user identity of the terminating UE. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "none" Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE (P-Asserted-Identity) 200 OK INVITE ACK ACK SS___XXSS_TIP04 TIP/TIR reference to: TS 124 508 [24], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/TIP. Selection criteria: The originating user subscribes to TIP service. The terminating user subscribes to TIR "temporary mode" default "restricted". Test purpose: 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 Privacy header field is set to "none" by the terminating UE in the 2xx SIP response: the originating UE receives, in the 2xx SIP response, a P-Asserted-Identity header field with a valid public user identity of the terminating UE. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "none" Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE (P-Asserted-Identity) 200 OK INVITE ACK ACK ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 36
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.4 Test purposes for TIR	SS___XXSS_TIR01 TIP/TIR reference to: TS 124 508 [24], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/TIR. Selection criteria: The originating user subscribes to TIP service. The terminating user subscribes to TIR "temporary mode" default "not restricted". Test purpose: 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 Privacy header field is set to "id" by the terminating UE in any non-100 SIP response (e.g. 180, 183, 200): 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. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "id" Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK SS___XXSS_TIR02 TIP/TIR reference to: TS 124 508 [24], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/TIR. Selection criteria: The originating user subscribes to TIP service. The terminating user subscribes to TIR "temporary mode" default "restricted". Test purpose: Ensure that, when the option tag "from-change" in the Supported header field is provided by the originating UE in the INVITE request and no Privacy header field is inserted by the terminating UE in any non-100 SIP response (e.g. 180, 183, 200): 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. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK SS___XXSS_TIR03 TIP/TIR reference to: TS 124 508 [24], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/TIR. Selection criteria: The originating user subscribes to TIP service. The terminating user subscribes to TIR "temporary mode" default "restricted". Test purpose: 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 Privacy header field is set to "id" by the terminating UE in any non-100 SIP response (e.g. 180, 183, 200): 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. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "id" Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 37 SS___XXSS_TIR04 TIP/TIR reference to: TS 124 508 [24], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/TIR. Selection criteria: The originating user subscribes to TIP service. The terminating user subscribes to TIR "permanent mode". Test purpose: Ensure that, when the option tag "from-change" in the Supported header field is provided by the originating UE in the INVITE request and no Privacy header field is inserted by the terminating UE in any non-100 SIP response (e.g. 180, 183, 200): 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. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK SS___XXSS_TIR05 TIP/TIR reference to: TS 124 508 [24], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/TIR. Selection criteria: The originating user subscribes to TIP service. The terminating user subscribes to TIR "permanent mode". Test purpose: 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 Privacy header field is set to "id" by the terminating UE in any non-100 SIP response (e.g. 180, 183, 200): 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. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "id" Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK SS___XXSS_TIR06 TIP/TIR reference to: TS 124 508 [24], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/TIP. Selection criteria: The originating user subscribes to TIP service. Additionally, the originating user has the "override category". The terminating user subscribes TIR "permanent mode". Test purpose: Ensure that, when the option tag "from-change" in the Supported header field is provided by the originating UE in the INVITE request and no Privacy header field is inserted by the terminating UE in any non-100 SIP response (e.g. 180, 183, 200): The originating UE does not receive a Privacy set to "id" in any non-100 SIP response (e.g. 180, 183, 200) and receives, in the 2xx SIP response, a P-Asserted-Identity header field with a valid public user identity of the terminating UE. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE (P-Asserted-Identity) 200 OK INVITE ACK ACK ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 38 SS___XXSS_TIR07 TIP/TIR reference to: TS 124 508 [24], clauses 4.3.2, 4.5.2.1, 4.5.2.4, 4.5.2.12 TSS reference: SIP-SIP/SupplementaryServices/TIP. Selection criteria: The originating user subscribes to TIP service. The user subscribes to TIR "permanent mode". Test purpose: 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 Privacy header field is set to "none" by the terminating UE in any non-100 SIP response (e.g. 180, 183, 200): 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. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Privacy header field is set to "none" Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.5 Test purposes for Hold
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.5.1 Communication Hold with support for UPDATE	SS__XXSSCH01 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithUPDATE. Selection criteria: Session hold. UPDATE method is used. Test purpose: Ensure that, when the originating UE (user A) sends an UPDATE request containing a SDP with the attribute "a=" sendonly to put the session on hold: • The terminating UE (user B) receives an UPDATE containing a SDP with the attribute "a=" sendonly. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "sendrecv". SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK UPDATE (sendonly) UPDATE (sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 39 SS__XXSSCH02 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithUPDATE. Selection criteria: Session hold. UPDATE method is used. Test purpose: Ensure that, when the originating UE (user A) sends an UPDATE request containing a SDP with the attribute "a=" inactive to change the media stream status to inactive: • The terminating UE (user B) receives an UPDATE containing a SDP with the attribute "a=" inactive. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" inactive. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" inactive. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user B (terminating UE). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK condition: The session was previously put on hold from user B UPDATE (sendonly) UPDATE(sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) UPDATE (inactive) UPDATE (inactive) 200 OK UPDATE (inactive) 200 OK UPDATE (inactive) BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 40 SS__XXSSCH03 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithUPDATE. Selection criteria: Session hold. UPDATE method is used. Test purpose: Ensure that, when the originating UE (user A) sends an UPDATE request containing a SDP with the attribute "a=" sendrecv to resume the session: • The terminating UE (user B) receives an UPDATE containing a SDP with the attribute "a=" sendrecv. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. Then the originating UE (user A) hang up the session. NOTE: The sendrecv SDP attribute can be omitted, since sendrecv attribute is the default. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user A (originating UE). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK UPDATE(sendonly) UPDATE(sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) UPDATE (sendrecv) UPDATE (sendrecv) 200 OK UPDATE (sendrecv) 200 OK UPDATE (sendrecv) BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 41 SS__XXSSCH04 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithUPDATE. Selection criteria: Session hold. UPDATE method is used. Test purpose: Ensure that, when the originating UE (user A) sends an UPDATE request containing a SDP with the attribute "a=" recvonly to resume the media stream status to recvonly: • The terminating UE (user B) receives an UPDATE containing a SDP with the attribute "a=" recvonly. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "inactive" from user A (originating UE). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK UPDATE(sendonly) UPDATE(sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) UPDATE (inactive) UPDATE (inactive) 200 OK UPDATE (inactive) 200 OK UPDATE (inactive) UPDATE (recvonly) UPDATE (recvonly) 200 OK UPDATE (sendonly) 200 OK UPDATE (sendonly) BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 42 SS__XXSSCH05 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithUPDATE. Selection criteria: Session hold. UPDATE method is used. Test purpose: Ensure that, when the terminating UE (user B) sends an UPDATE request containing a SDP with the attribute "a=" sendonly to put the session on hold: • The originating UE (user A) receives an UPDATE containing a SDP with the attribute "a=" sendonly. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. • The terminating UE (user B) receives a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "sendrecv". SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK UPDATE (sendonly) UPDATE (sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 43 SS__XXSSCH06 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithUPDATE. Selection criteria: Session hold. UPDATE method is used. Test purpose: Ensure that, when the terminating UE (user B) sends an UPDATE request containing a SDP with the attribute "a=" inactive to change the media stream status to inactive: • The originating UE (user A) receives an UPDATE containing a SDP with the attribute "a=" inactive. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" inactive. • The terminating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" inactive. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user A (originating UE). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK UPDATE (sendonly) UPDATE (sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) UPDATE (inactive) UPDATE (inactive) 200 OK UPDATE (inactive) 200 OK UPDATE (inactive) BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 44 SS__XXSSCH07 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithUPDATE. Selection criteria: Session hold. UPDATE method is used. Test purpose: Ensure that, when the terminating UE (user B) sends an UPDATE request containing a SDP with the attribute "a=" sendrecv to resume the session: • The originating UE (user A) receives an UPDATE containing a SDP with the attribute "a=" sendrecv. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. • The terminating UE (user B) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. Then the originating UE (user A) hang up the session. NOTE: The sendrecv SDP attribute can be omitted, since sendrecv attribute is the default. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user B (terminating UE). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK UPDATE (sendonly) UPDATE(sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) UPDATE (sendrecv) UPDATE (sendrecv) 200 OK UPDATE (sendrecv) 200 OK UPDATE (sendrecv) BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 45 SS__XXSSCH08 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithUPDATE. Selection criteria: Session hold. UPDATE method is used. Test purpose: Ensure that, when the terminating UE (user B) sends an UPDATE request containing a SDP with the attribute "a=" recvonly to resume the media stream status to recvonly: • The originating UE (user A) receives an UPDATE containing a SDP with the attribute "a=" recvonly. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. • The terminating UE (user B) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "inactive" from user B (terminating UE). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK UPDATE (sendonly) UPDATE (sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) UPDATE (inactive) UPDATE (inactive) 200 OK UPDATE (inactive) 200 OK UPDATE (inactive) UPDATE(recvonly) UPDATE(recvonly) 200 OK UPDATE (sendonly) 200 OK UPDATE (sendonly) BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 46
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.5.2 Communication Hold without support for UPDATE	SS__XXSSCH09 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithoutUPDATE. Selection criteria: Session hold. INVITE method is used. Test purpose: Ensure that, when the originating UE (user A) sends an INVITE request containing a SDP with the attribute "a=" sendonly to put the session on hold: • The terminating UE (user B) receives an INVITE containing a SDP with the attribute "a=" sendonly. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "sendrecv". SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK INVITE (sendonly) INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE(recvonly) ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 47 SS__XXSSCH 10 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithoutUPDATE. Selection criteria: Session hold. INVITE method is used. Test purpose: Ensure that, when the originating UE (user A) sends an INVITE request containing a SDP with the attribute "a=" inactive to change the media stream status to inactive: • The terminating UE (user B) receives an INVITE containing a SDP with the attribute "a=" inactive. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" inactive. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" inactive. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user B (terminating UE). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK INVITE (sendonly) INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE (recvonly) ACK ACK INVITE (inactive) INVITE (inactive) 200 OK INVITE (inactive) 200 OK INVITE (inactive) ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 48 SS__XXSSCH 11 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithoutUPDATE. Selection criteria: Session hold. INVITE method is used. Test purpose: Ensure that, when the originating UE (user A) sends an INVITE request containing a SDP with the attribute "a=" sendrecv to resume the session: • The terminating UE (user B) receives an INVITE containing a SDP with the attribute "a=" sendrecv. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. Then the originating UE (user A) hang up the session. NOTE: The sendrecv SDP attribute can be omitted, since sendrecv attribute is the default. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user A (originating UE). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK INVITE (sendonly) INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE(recvonly) ACK ACK INVITE (sendrecv) INVITE (sendrecv) 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 49 SS__XXSSCH 12 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithoutUPDATE. Selection criteria: Session hold. INVITE method is used. Test purpose: Ensure that, when the originating UE (user A) sends an INVITE request containing a SDP with the attribute "a=" recvonly to resume the media stream status to recvonly: • The terminating UE (user B) receives an INVITE containing a SDP with the attribute "a=" recvonly. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "inactive" from user A (originating UE). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK INVITE(sendonly) INVITE(sendonly) 200 OK INVITE (recvonly) 200 OK INVITE(recvonly) ACK ACK INVITE (inactive) INVITE (inactive) 200 OK INVITE (inactive) 200 OK INVITE (inactive) ACK ACK INVITE (recvonly) INVITE (recvonly) 200 OK INVITE (sendonly) 200 OK INVITE (sendonly) ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 50 SS__XXSSCH 13 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithoutUPDATE. Selection criteria: Session hold. INVITE method is used. Test purpose: Ensure that, when the terminating UE (user B) sends an INVITE request containing a SDP with the attribute "a=" sendonly to put the session on hold: • The originating UE (user A) receives an INVITE containing a SDP with the attribute "a=" sendonly. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. • The terminating UE (user B) receives a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "sendrecv". SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK INVITE (sendonly) INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE (recvonly) ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 51 SS__XXSSCH 14 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithoutUPDATE. Selection criteria: Session hold. INVITE method is used. Test purpose: Ensure that, when the terminating UE (user B) sends an INVITE request containing a SDP with the attribute "a=" inactive to change the media stream status to inactive: • The originating UE (user A) receives an INVITE containing a SDP with the attribute "a=" inactive. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" inactive. • The terminating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" inactive. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user A (originating UE). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK INVITE (sendonly) INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE (recvonly) ACK ACK INVITE(inactive) INVITE (inactive) 200 OK INVITE (inactive) 200 OK INVITE (inactive) ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 52 SS__XXSSCH 15 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithoutUPDATE. Selection criteria: Session hold. INVITE method is used. Test purpose: Ensure that, when the terminating UE (user B) sends an INVITE request containing a SDP with the attribute "a=" sendrecv to resume the session: • The originating UE (user A) receives an INVITE containing a SDP with the attribute "a=" sendrecv. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. • The terminating UE (user B) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. Then the originating UE (user A) hang up the session. NOTE: The sendrecv SDP attribute can be omitted, since sendrecv attribute is the default. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user B (terminating UE). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK INVITE (sendonly) INVITE(sendonly) 200 OK INVITE (recvonly) 200 OK INVITE(recvonly) ACK ACK INVITE (sendrecv) INVITE (sendrecv) 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 53 SS__XXSSCH 16 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithoutAnnounc/WithoutUPDATE. Selection criteria: Session hold. INVITE method is used. Test purpose: Ensure that, when the terminating UE (user B) sends an INVITE request containing a SDP with the attribute "a=" recvonly to resume the media stream status to recvonly: • The originating UE (user A) receives an INVITE containing a SDP with the attribute "a=" recvonly. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. • The terminating UE (user B) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "inactive" from user B (terminating UE). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK INVITE(sendonly) INVITE(sendonly) 200 OK INVITE (recvonly) 200 OK INVITE(recvonly) ACK ACK INVITE (inactive) INVITE (inactive) 200 OK INVITE (inactive) 200 OK INVITE(inactive) ACK ACK INVITE (recvonly) INVITE (recvonly) 200 OK INVITE (sendonly) 200 OK INVITE (sendonly) ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 54
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.5.3 Communication with announcements
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.5.3.1 Communication Hold with support for UPDATE	SS__XXSSCH17 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithUPDATE. Selection criteria: The remote user is put on hold, an announcement starts to the held user. The UPDATE method is used. Test purpose: Ensure that, when the originating UE (user A) sends an UPDATE request containing a SDP with the attribute "a=" sendonly to put the session on hold: • The terminating UE (user B) receives an UPDATE containing a SDP with the attribute "a=" sendonly. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. • An announcement is played to the terminating UE (user B). Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "sendrecv". SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation UPDATE (sendonly) UPDATE (sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) Announcement to UE B BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 55 SS__XXSSCH18 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithUPDATE. Selection criteria: The announcement is stopped after the held user puts the media stream on hold. The UPDATE method is used. Test purpose: Ensure that, when the originating UE (user A) sends an UPDATE request containing a SDP with the attribute "a=" inactive to change the media stream status to inactive: • The terminating UE (user B) receives an UPDATE containing a SDP with the attribute "a=" inactive. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" inactive. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" inactive. • The announcement to the originating UE (user A) is stopped. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user B (terminating UE). • An announcement is played to the originating UE (user A). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation UPDATE (sendonly) UPDATE (sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) Announcement to UE A UPDATE (inactive) UPDATE (inactive) 200 OK UPDATE (inactive) 200 OK UPDATE (inactive) Media stream is stopped BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 56 SS__XXSSCH19 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithUPDATE. Selection criteria: The announcement is stopped after retrieve. Test purpose: Ensure that, when the originating UE (user A) sends an UPDATE request containing a SDP with the attribute "a=" sendrecv to resume the session: • The terminating UE (user B) receives an UPDATE containing a SDP with the attribute "a=" sendrecv. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. • The announcement to the terminating UE (user B) is stopped. Then the originating UE (user A) hang up the session. NOTE: The sendrecv SDP attribute can be omitted, since sendrecv attribute is the default. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user A (originating UE). • An announcement is played to the terminating UE (user B). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation UPDATE (sendonly) UPDATE (sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) Announcement to UE B UPDATE (sendrecv) UPDATE (sendrecv) 200 OK UPDATE (sendrecv) 200 OK UPDATE (sendrecv) Conversation BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 57 SS__XXSSCH20 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithUPDATE. Selection criteria: Announcement is started to user B when user B retrieves the connection. Test purpose: Ensure that, when the originating UE (user A) sends an UPDATE request containing a SDP with the attribute "a=" recvonly to resume the media stream status to recvonly: • The terminating UE (user B) receives an UPDATE containing a SDP with the attribute "a=" recvonly. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. • An announcement is played to the originating UE (user A). Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "inactive" from user A (originating UE). • The announcement to the originating UE (user A) is stopped. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation UPDATE (sendonly) UPDATE (sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) Announcement to UE A UPDATE (inactive) UPDATE (inactive) 200 OK UPDATE (inactive) 200 OK UPDATE (inactive) Media stream is stopped UPDATE (recvonly) UPDATE (recvonly) 200 OK UPDATE (sendonly) 200 OK UPDATE (sendonly) Announcement to UE A BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 58 SS__XXSSCH21 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithUPDATE. Selection criteria: The remote user is put on hold, an announcement starts to the held user. The UPDATE method is used. Test purpose: Ensure that, when the terminating UE (user B) sends an UPDATE request containing a SDP with the attribute "a=" sendonly to put the session on hold: • The originating UE (user A) receives an UPDATE containing a SDP with the attribute "a=" sendonly. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. • The terminating UE (user B) receives a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. • An announcement is played to the terminating UE (user B). Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "sendrecv". SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation UPDATE (sendonly) UPDATE (sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) Announcement to UE A BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 59 SS__XXSSCH22 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithUPDATE. Selection criteria: The announcement is stopped after the held user puts the media stream on hold. The UPDATE method is used. Test purpose: Ensure that, when the terminating UE (user B) sends an UPDATE request containing a SDP with the attribute "a=" inactive to change the media stream status to inactive: • The originating UE (user A) receives an UPDATE containing a SDP with the attribute "a=" inactive. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" inactive. • The terminating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" inactive. • The announcement to the terminating UE (user B) is stopped. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user A (originating UE). • An announcement is played to the terminating UE (user B). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation UPDATE (sendonly) UPDATE(sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) Announcement to UE B UPDATE (inactive) UPDATE (inactive) 200 OK UPDATE (inactive) 200 OK UPDATE (inactive) Media stream is stopped BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 60 SS__XXSSCH23 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithUPDATE. Selection criteria: Announcement is stopped after retrieve. Test purpose: Ensure that, when the terminating UE (user B) sends an UPDATE request containing a SDP with the attribute "a=" sendrecv to resume the session: • The originating UE (user A) receives an UPDATE containing a SDP with the attribute "a=" sendrecv. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. • The terminating UE (user B) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. • The announcement to the originating UE (user A) is stopped. Then the originating UE (user A) hang up the session. NOTE: The sendrecv SDP attribute can be omitted, since sendrecv attribute is the default. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user B (terminating UE). • An announcement is played to the originating UE (user A). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation UPDATE (sendonly) UPDATE (sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) Announcement to UE A UPDATE (sendrecv) UPDATE (sendrecv) 200 OK UPDATE (sendrecv) 200 OK UPDATE (sendrecv) Conversation BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 61 SS__XXSSCH24 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithUPDATE. Selection criteria: Announcement is started to user B when user B retrieves the connection. Test purpose: Ensure that, when the terminating UE (user B) sends an UPDATE request containing a SDP with the attribute "a=" recvonly to resume the media stream status to recvonly: • The originating UE (user A) receives an UPDATE containing a SDP with the attribute "a=" recvonly. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. • The terminating UE (user B) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. • An announcement is played to the terminating UE (user B). Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "inactive" from user B (terminating UE). • The announcement to the terminating UE (user B) is stopped. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation UPDATE (sendonly) UPDATE (sendonly) 200 OK UPDATE (recvonly) 200 OK UPDATE (recvonly) Announcement to UE B UPDATE (inactive) UPDATE (inactive) 200 OK UPDATE (inactive) 200 OK UPDATE (inactive) Media stream is stopped UPDATE (recvonly) UPDATE (recvonly) 200 OK UPDATE (sendonly) 200 OK UPDATE (sendonly) Announcement to UE B BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 62
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.5.3.2 Communication Hold without support for UPDATE	SS__XXSSCH25 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithoutUPDATE. Selection criteria: The remote user is put on hold, an announcement starts to the held user. The INVITE method is used. Test purpose: Ensure that, when the originating UE (user A) sends an INVITE request containing a SDP with the attribute "a=" sendonly to put the session on hold: • The terminating UE (user B) receives an INVITE containing a SDP with the attribute "a=" sendonly. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. • An announcement is played to the terminating UE (user B). Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "sendrecv". SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation INVITE (sendonly) INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE (recvonly) ACK ACK Announcement to UE B BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 63 SS__XXSSCH26 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithoutUPDATE. Selection criteria: The announcement is stopped after the held user puts the media stream on hold. The INVITE method is used. Test purpose: Ensure that, when the originating UE (user A) sends an INVITE request containing a SDP with the attribute "a=" inactive to change the media stream status to inactive: • The terminating UE (user B) receives an INVITE containing a SDP with the attribute "a=" inactive. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" inactive. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" inactive. • The announcement to the originating UE (user A) is stopped. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user B (terminating UE). • An announcement is played to the originating UE (user A). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation INVITE (sendonly) INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE (recvonly) ACK ACK Announcement to UE A INVITE (inactive) INVITE (inactive) 200 OK INVITE (inactive) 200 OK INVITE (inactive) ACK ACK Media stream is stopped BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 64 SS__XXSSCH27 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithoutUPDATE. Selection criteria: Announcement is stopped after retrieve. The INVITE method is used. Test purpose: Ensure that, when the originating UE (user A) sends an INVITE request containing a SDP with the attribute "a=" sendrecv to resume the session: • The terminating UE (user B) receives an INVITE containing a SDP with the attribute "a=" sendrecv. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. • The announcement to the terminating UE (user B) is stopped. Then the originating UE (user A) hang up the session. NOTE: The sendrecv SDP attribute can be omitted, since sendrecv attribute is the default. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user A (originating UE). • An announcement is played to the terminating UE (user B). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation INVITE (sendonly) INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE (recvonly) ACK ACK Announcement to UE B INVITE (sendrecv) INVITE (sendrecv) 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 65 SS__XXSSCH28 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithoutUPDATE. Selection criteria: Announcement is started to user B when user B retrieves the connection. The INVITE method is used. Test purpose: Ensure that, when the originating UE (user A) sends an INVITE request containing a SDP with the attribute "a=" recvonly to resume the media stream status to recvonly: • The terminating UE (user B) receives an INVITE containing a SDP with the attribute "a=" recvonly. • The terminating UE (user B) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. • The originating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. • An announcement is played to the originating UE (user A). Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "inactive" from user A (originating UE). • The announcement to the originating UE (user A) is stopped. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation INVITE(sendonly) INVITE(sendonly) 200 OK INVITE (recvonly) 200 OK INVITE(recvonly) ACK ACK Announcement to UE A INVITE (inactive) INVITE(inactive) 200 OK INVITE (inactive) 200 OK INVITE(inactive) ACK ACK Media stream is stopped INVITE (recvonly) INVITE (recvonly) 200 OK INVITE (sendonly) 200 OK INVITE (sendonly) ACK ACK Announcement to UE A BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 66 SS__XXSSCH29 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithoutUPDATE. Selection criteria: The remote user is put on hold, an announcement starts to the held user. The INVITE method is used. Test purpose: Ensure that, when the terminating UE (user B) sends an INVITE request containing a SDP with the attribute "a=" sendonly to put the session on hold: • The originating UE (user A) receives an INVITE containing a SDP with the attribute "a=" sendonly. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. • The terminating UE (user B) receives a 200 OK SIP response containing a SDP with the attribute "a=" recvonly. • An announcement is played to the terminating UE (user B). Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "sendrecv". SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation INVITE (sendonly) INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE (recvonly) ACK ACK Announcement to UE A BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 67 SS__XXSSCH30 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithoutUPDATE. Selection criteria: The announcement is stopped after the held user puts the media stream on hold. The INVITE method is used. Test purpose: Ensure that, when the terminating UE (user B) sends an INVITE request containing a SDP with the attribute "a=" inactive to change the media stream status to inactive: • The originating UE (user A) receives an INVITE containing a SDP with the attribute "a=" inactive. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" inactive. • The terminating UE (user A) receives a 200 OK SIP response containing a SDP with the attribute "a=" inactive. • The announcement to the terminating UE (user B) is stopped. Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user A (originating UE). • An announcement is played to the terminating UE (user B). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation INVITE (sendonly) INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE (recvonly) ACK ACK Announcement to UE B INVITE (inactive) INVITE (inactive) 200 OK INVITE (inactive) 200 OK INVITE (inactive) ACK ACK Media stream is stopped BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 68 SS__XXSSCH31 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithoutUPDATE. Selection criteria: Announcement is stopped after retrieve. The INVITE method id used. Test purpose: Ensure that, when the terminating UE (user B) sends an INVITE request containing a SDP with the attribute "a=" sendrecv to resume the session: • The originating UE (user A) receives an INVITE containing a SDP with the attribute "a=" sendrecv. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. • The terminating UE (user B) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendrecv. • The announcement to the originating UE (user A) is stopped. Then the originating UE (user A) hang up the session. NOTE: The sendrecv SDP attribute can be omitted, since sendrecv attribute is the default. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The session was previously put on hold from user B (terminating UE). • An announcement is played to the originating UE (user A). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE (sendrecv) 180 Ringing 180 Ringing 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation INVITE (sendonly) INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE (recvonly) ACK ACK Announcement to UE A INVITE (sendrecv) INVITE (sendrecv) 200 OK INVITE (sendrecv) 200 OK INVITE (sendrecv) ACK ACK Conversation BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 69 SS__XXSSCH32 HOLD reference to: TS 124 410 [15], clauses 4.5.2.1, 4.5.2.4, 4.5.2.9 TSS reference: ServedUser/WithAnnounc/WithoutUPDATE. Selection criteria: Announcement is started to user B when user B retrieves the connection. The INVITE method id used. Test purpose: Ensure that, when the terminating UE (user B) sends an INVITE request containing a SDP with the attribute "a=" recvonly to resume the media stream status to recvonly: • The originating UE (user A) receives an INVITE containing a SDP with the attribute "a=" recvonly. • The originating UE (user A) sends a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. • The terminating UE (user B) receives a 200 OK SIP response containing a SDP with the attribute "a=" sendonly. • An announcement is played to the terminating UE (user B). Then the originating UE (user A) hang up the session. Precondition: • A session was established between user A (originating UE) and user B (terminating UE) according to the "basic Call" procedures. • The media stream was previously set to "inactive" from user B (terminating UE). • The announcement to the terminating UE (user B) is stopped. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A SUT SIP UA B INVITE (sendrecv) INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Conversation INVITE(sendonly) INVITE(sendonly) 200 OK INVITE (recvonly) 200 OK INVITE(recvonly) ACK ACK Announcement to UE B INVITE (inactive) INVITE(inactive) 200 OK INVITE (inactive) 200 OK INVITE(inactive) ACK ACK Media stream is stopped INVITE (recvonly) INVITE (recvonly) 200 OK INVITE (sendonly) 200 OK INVITE (sendonly) ACK ACK Announcement to UE B BYE BYE 200 OK BYE 200 OK BYE
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.6 Test purposes for Communication Diversion	The configuration lines in this clause contain only the subscription options to the communication diversion service that are relevant for the test purpose. Subscription options not mentioned can take any value. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 70
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.6.1 CFU	SSS__XXSSCFU01 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.5 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFU. Configuration: The user B has subscribed to CFU, CDIVN is not activated Subscription options: Served user receives indication that a communication has been forwarded = Yes Selection criteria: CFU supported. Test purpose: Ensure that when user A calls user B, the call is forwarded to user C. Ensure that in the active call state the voice transfer on the media channels is performed correctly (e.g. testing QoS parameters). Ensure that User B receives a MESSAGE request indicating the call diversion. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE Communication diversion is performed 181 Call Is Being Forwarded (optional) INVITE MESSAGE 200 OK MESSAGE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Communication BYE BYE 200 OK BYE 200 OK BYE SSS__XXSSCFU02 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.5.1 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFU. Configuration: The user B has subscribed to CFU and CDIVN. Selection criteria: CFU and CDIVN supported. Test purpose: Ensure that when user A calls user B, the call is forwarded to user C. Ensure that in the active call state the voice transfer on the media channels is performed correctly (e.g. testing QoS parameters). Ensure that User B, having activated the CDIVN service, receives a NOTIFY request indicating the call diversion. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C Start Activation CDIVN SUBSCRIBE 200 OK SUBSCRIBE NOTIFY 200 OK NOTIFY End Activation CDIVN INVITE Communication diversion is performed 181 Call Is Being Forwarded (optional) INVITE NOTIFY 200 OK NOTIFY 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Communication BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 71 SSS__XXSSCFU03 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFU. Configuration: The user B has subscribed to CFU Subscription options: Originating user receives notification that his communication has been diverted = No Selection criteria: CFU supported. Test purpose: Ensure that when user A calls user B, the call is forwarded to user C. Ensure that User A does not receive a 181 Call Is Being Forwarded message. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE SSS__XXSSCFU04 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFU. Configuration: The user B has subscribed to CFU and has not activated TIR 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 = No Served user allows the presentation of his/her URI to originating user in diversion notification = No Selection criteria: CFU supported. Test purpose: Ensure that when user A calls user B, the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message containing a Privacy header with value "id" and not containing a P-Asserted-Identity indicating the URI of user B and not containing a History-Info header (with CDIV related cause value) indicating the URI of user B or user A. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C INVITE INVITE 181 Call Is Being Forwarded 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 72 SSS__XXSSCFU05 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFU. Configuration: The user B has subscribed to CFU and has not activated TIR 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 Served user allows the presentation of his/her URI to originating user in diversion notification = Yes Selection criteria: CFU supported. Test purpose: Ensure that when user A calls user B, the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message containing a P-Asserted-Identity indicating the URI of user B and containing a History-Info header • including a first entry with the hi-targeted-to-URI of user B, index = 1, cause param = 302 and • including a second entry with the hi-targeted-to-URI of user C, index = 1.1 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C INVITE INVITE 181 Call Is Being Forwarded 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE SSS__XXSSCFU06 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.2.2 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFU. Configuration: The user B has subscribed to CFU and has not activated OIR Subscription options: Served user allows the presentation of his/her URI to diverted-to user = Yes Selection criteria: CFU supported. Test purpose: Ensure that when user A calls user B, the call is forwarded to user C. Ensure that User C receives an INVITE message containing a History-Info header including an entry (with CDIV related cause value) with the hi-targeted-to-URI of user B. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C INVITE INVITE 181 Call Is Being Forwarded (optional) 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 73
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.6.2 CFB
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.6.2.1 NDUB	SSS__XXSSCFB01 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.5 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFB. Configuration: The user B has subscribed to CFB, CDIVN is not activated Subscription options: Served user receives indication that a communication has been forwarded = Yes The user B has not subscribed to CW Selection criteria: CFB supported, NDUB status can be achieved for user B. Test purpose: Ensure that when user A calls user B which is network determined user busy (NDUB), the call is forwarded to user C. Ensure that in the active call state the voice transfer on the media channels is performed correctly (e.g. testing QoS parameters). Ensure that User B receives a MESSAGE request indicating the call diversion. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C UA B enters NDUB condition (e.g. by establishing a communication) INVITE Communication diversion is performed 181 Call Is Being Forwarded (optional) INVITE MESSAGE 200 OK MESSAGE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Communication BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 74 SSS__XXSSCFB02 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.5.1 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFB. Configuration: The user B has subscribed to CFB and CDIVN The user B has not subscribed to CW Selection criteria: CFB and CDIVN supported, NDUB status can be achieved for user B. Test purpose: Ensure that when user A calls user B which is network determined user busy (NDUB), the call is forwarded to user C. Ensure that in the active call state the voice transfer on the media channels is performed correctly (e.g. testing QoS parameters). Ensure that User B, having activated the CDIVN service, receives a NOTIFY request indicating the call diversion. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C Activation CDIVN SUBSCRIBE 200 OK SUBSCRIBE NOTIFY 200 OK NOTIFY UA B enters NDUB condition (e.g. by establishing a communication) INVITE Communication diversion is performed 181 Call Is Being Forwarded (optional) INVITE NOTIFY 200 OK NOTIFY 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Communication BYE BYE 200 OK BYE 200 OK BYE SSS__XXSSCFB03 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFB. Configuration: The user B has subscribed to CFB Subscription options: Originating user receives notification that his communication has been diverted = No The user B has not subscribed to CW Selection criteria: CFB supported, NDUB status can be achieved for user B. Test purpose: Ensure that when user A calls user B which is network determined user busy (NDUB), the call is forwarded to user C. Ensure that User A does not receive a 181 Call Is Being Forwarded message. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C UA B enters NDUB condition (e.g. by establishing a communication) INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 75 SSS__XXSSCFB04 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFB. Configuration: The user B has subscribed to CFB and has not activated TIR 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 = No Served user allows the presentation of his/her URI to originating user in diversion notification = No The user B has not subscribed to CW Selection criteria: CFB supported, NDUB status can be achieved for user B. Test purpose: Ensure that when user A calls user B which is network determined user busy (NDUB), the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message containing a Privacy header with value "id" and not containing a P-Asserted-Identity indicating the URI of user B and not containing a History-Info header (with CDIV related cause value) indicating the URI of user B or user A. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C UA B enters NDUB condition (e.g. by establishing a communication) INVITE INVITE 181 Call Is Being Forwarded 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 76 SSS__XXSSCFB05 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFB. Configuration: The user B has subscribed to CFB and has not activated TIR 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 Served user allows the presentation of his/her URI to originating user in diversion notification = Yes The user B has not subscribed to CW Selection criteria: CFB supported, NDUB status can be achieved for user B. Test purpose: Ensure that when user A calls user B which is network determined user busy (NDUB), the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message containing a P-Asserted-Identity indicating the URI of user B and containing a History-Info header • including a first entry with the hi-targeted-to-URI of user B, index = 1, cause param = 486 and • including a second entry with the hi-targeted-to-URI of user C, index = 1.1. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C UA B enters NDUB condition (e.g. by establishing a communication) INVITE INVITE 181 Call Is Being Forwarded 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE SSS__XXSSCFB06 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.2.2 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFB. Configuration: The user B has subscribed to CFB and has not activated OIR Subscription options: Served user allows the presentation of his/her URI to diverted-to user = Yes The user B has not subscribed to CW Selection criteria: CFB supported, NDUB status can be achieved for user B. Test purpose: Ensure that when user A calls user B which is network determined user busy (NDUB), the call is forwarded to user C. Ensure that User C receives an INVITE message containing a History-Info header including an entry (with CDIV related cause value) with the hi-targeted-to-URI of user B. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C UA B enters NDUB condition (e.g. by establishing a communication) INVITE INVITE 181 Call Is Being Forwarded (optional) 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 77
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.6.2.2 UDUB	SSS__XXSSCFB07 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.5 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFB. Configuration: The user B has subscribed to CFB, CDIVN is not activated Subscription options: Served user receives indication that a communication has been forwarded = Yes Selection criteria: CFB supported. Test purpose: Ensure that when user A calls user B which is user determined user busy (UDUB), the call is forwarded to user C. Ensure that in the active call state the voice transfer on the media channels is performed correctly (e.g. testing QoS parameters). Ensure that User B receives a MESSAGE request indicating the call diversion. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 486 Busy Here ACK Communication diversion is performed 181 Call Is Being Forwarded (optional) INVITE MESSAGE 200 OK MESSAGE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Communication BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 78 SSS__XXSSCFB08 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.5.1 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFB. Configuration: The user B has subscribed to CFB and CDIVN Selection criteria: CFB and CDIVN supported. Test purpose: Ensure that when user A calls user B which is user determined user busy (UDUB), the call is forwarded to user C. Ensure that in the active call state the voice transfer on the media channels is performed correctly (e.g. testing QoS parameters). Ensure that User B, having activated the CDIVN service, receives a NOTIFY request indicating the call diversion. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C Start Activation CDIVN SUBSCRIBE 200 OK SUBSCRIBE NOTIFY 200 OK NOTIFY End Activation CDIVN INVITE INVITE 486 Busy Here ACK Communication diversion is performed 181 Call Is Being Forwarded (optional) INVITE NOTIFY 200 OK NOTIFY 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Communication BYE BYE 200 OK BYE 200 OK BYE SSS__XXSSCFB09 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFB. Configuration: The user B has subscribed to CFB Subscription options: Originating user receives notification that his communication has been diverted = No Selection criteria: CFB supported. Test purpose: Ensure that when user A calls user B which is user determined user busy (UDUB), the call is forwarded to user C. Ensure that User A does not receive a 181 Call Is Being Forwarded message. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 486 Busy Here ACK Communication diversion is performed INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 79 SSS__XXSSCFB10 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFB. Configuration: The user B has subscribed to CFB and has not activated TIR 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 = No Served user allows the presentation of his/her URI to originating user in diversion notification = No Selection criteria: CFB supported. Test purpose: Ensure that when user A calls user B which is user determined user busy (UDUB), the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message containing a Privacy header with value "id" and not containing a P-Asserted-Identity indicating the URI of user B and not containing a History-Info header (with CDIV related cause value) indicating the URI of user B or user A. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 486 Busy Here ACK Communication diversion is performed 181 Call Is Being Forwarded INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 80 SSS__XXSSCFB11 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFB. Configuration: The user B has subscribed to CFB and has not activated TIR 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 Served user allows the presentation of his/her URI to originating user in diversion notification = Yes Selection criteria: CFB supported. Test purpose: Ensure that when user A calls user B which is user determined user busy (UDUB), the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message containing a P-Asserted-Identity indicating the URI of user B and containing a History-Info header • including a first entry with the hi-targeted-to-URI of user B, index = 1, cause param = 486 and • including a second entry with the hi-targeted-to-URI of user C, index = 1.1 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 486 Busy Here ACK Communication diversion is performed 181 Call Is Being Forwarded INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE SSS__XXSSCFB12 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.2.2 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFB. Configuration: The user B has subscribed to CFB and has not activated OIR Subscription options: Served user allows the presentation of his/her URI to diverted-to user = Yes Selection criteria: CFB supported. Test purpose: Ensure that when user A calls user B which is user determined user busy (UDUB), the call is forwarded to user C. Ensure that User C receives an INVITE message containing a History-Info header (with CDIV related cause value) including an entry with the hi-targeted-to-URI of user B. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 486 Busy Here ACK Communication diversion is performed 181 Call Is Being Forwarded INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 81
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.6.3 CFNR	SSS__XXSSCFNR01 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.5 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNR. Configuration: The user B has subscribed to CFNR, CDIVN is not activated Subscription options: Served user receives indication that a communication has been forwarded = Yes Selection criteria: CFNR supported. Test purpose: Ensure that when user A calls user B which does not answer, the call is forwarded to user C. Ensure that in the active call state the voice transfer on the media channels is performed correctly (e.g. testing QoS parameters). Ensure that User B receives a MESSAGE request indicating the call diversion. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 180 Ringing 180 Ringing No reply timer expires - Communication diversion is performed 181 Call Is Being Forwarded (optional) CANCEL (Note) INVITE 200 OK CANCEL 487 Request Terminated ACK MESSAGE 200 OK MESSAGE 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Communication BYE BYE 200 OK BYE 200 OK BYE NOTE: The communication to user B may be retained until the 180 Ringing from user C has been received. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 82 SSS__XXSSCFNR02 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.5.1 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNR. Configuration: The user B has subscribed to CFNR and CDIVN Selection criteria: CFNR and CDIVN supported. Test purpose: Ensure that when user A calls user B which does not answer, the call is forwarded to user C. Ensure that in the active call state the voice transfer on the media channels is performed correctly (e.g. testing QoS parameters). Ensure that User B, having activated the CDIVN service, receives a NOTIFY request indicating the call diversion. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C Start Activation CDIVN SUBSCRIBE 200 OK SUBSCRIBE NOTIFY 200 OK NOTIFY End Activation CDIVN INVITE INVITE 180 Ringing 180 Ringing No reply timer expires - Communication diversion is performed 181 Call Is Being Forwarded (optional) CANCEL (Note) INVITE 200 OK CANCEL 487 Request Terminated ACK NOTIFY 200 OK NOTIFY 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Communication BYE BYE 200 OK BYE 200 OK BYE NOTE: The communication to user B may be retained until the 180 Ringing from user C has been received. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 83 SSS__XXSSCFNR03 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNR. Configuration: The user B has subscribed to CFNR Subscription options: Originating user receives notification that his communication has been diverted = No Selection criteria: CFNR supported. Test purpose: Ensure that when user A calls user B which does not answer, the call is forwarded to user C. Ensure that User A does not receive a 181 Call Is Being Forwarded message. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 180 Ringing 180 Ringing No reply timer expires - Communication diversion is performed CANCEL (Note) INVITE 200 OK CANCEL 487 Request Terminated ACK 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE NOTE: The communication to user B may be retained until the 180 Ringing from user C has been received. SSS__XXSSCFNR04 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNR. Configuration: The user B has subscribed to CFNR and has not activated TIR 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 = No Served user allows the presentation of his/her URI to originating user in diversion notification = No Selection criteria: CFNR supported. Test purpose: Ensure that when user A calls user B which does not answer, the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message containing a Privacy header with value "id" and not containing a P-Asserted-Identity indicating the URI of user B and not containing a History-Info header indicating the URI of user B or user A. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 180 Ringing 180 Ringing No reply timer expires - Communication diversion is performed 181 Call Is Being Forwarded CANCEL (Note) INVITE 200 OK CANCEL 487 Request Terminated ACK 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE NOTE: The communication to user B may be retained until the 180 Ringing from user C has been received. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 84 SSS__XXSSCFNR05 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNR. Configuration: The user B has subscribed to CFNR and has not activated TIR 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 Served user allows the presentation of his/her URI to originating user in diversion notification = Yes Selection criteria: CFNR supported. Test purpose: Ensure that when user A calls user B which does not answer, the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message containing a P-Asserted-Identity indicating the URI of user B and containing a History-Info header • including a first entry with the hi-targeted-to-URI of user B, index = 1, cause param = 408 and • including a second entry with the hi-targeted-to-URI of user C, index = 1.1 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 180 Ringing 180 Ringing No reply timer expires - Communication diversion is performed 181 Call Is Being Forwarded CANCEL (Note) INVITE 200 OK CANCEL 487 Request Terminated ACK 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE NOTE: The communication to user B may be retained until the 180 Ringing from user C has been received. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 85 SSS__XXSSCFNR06 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.2.2 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNR. Configuration: The user B has subscribed to CFNR and has not activated OIR Subscription options: Served user allows the presentation of his/her URI to diverted-to user = Yes Selection criteria: CFNR supported. Test purpose: Ensure that when user A calls user B which does not answer, the call is forwarded to user C. Ensure that User C receives an INVITE message containing a History-Info header (with CDIV related cause value) including an entry with the hi-targeted-to-URI of user B. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 180 Ringing 180 Ringing No reply timer expires - Communication diversion is performed 181 Call Is Being Forwarded CANCEL (Note) INVITE 200 OK CANCEL 487 Request Terminated ACK 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE NOTE: The communication to user B may be retained until the 180 Ringing from user C has been received. SSS__XXSSCFNR07 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.3 3) TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNR. Configuration: The user B has subscribed to CFNR Subscription options: Served user communication retention on invocation of diversion = Retain communication to the served user until alerting begins at the diverted-to user Selection criteria: CFNR supported. Test purpose: Ensure that when user A calls user B which has not answered before the expiry of the No reply timer, and when the communication has been forwarded to user C and when user B answers the communication before user C starts alerting, the communication is established between user A and user B and the communication is cancelled towards user C. Ensure that in the active call state the voice transfer on the media channels is performed correctly (e.g. testing QoS parameters). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 180 Ringing 180 Ringing No reply timer expires - Communication diversion is performed 181 Call Is Being Forwarded (optional) INVITE 200 OK INVITE 200 OK INVITE ACK ACK CANCEL 200 OK CANCEL Communication BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 86
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.6.4 CFNRc	SSS__XXSSCFNRc01 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.5.1 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNRc. Configuration: The user B has subscribed to CFNRc and CDIVN Selection criteria: CFNRC and CDIVN supported. Test purpose: Ensure that when user A calls user B which is unreachable, the call is forwarded to user C. Ensure that in the active call state the voice transfer on the media channels is performed correctly (e.g. testing QoS parameters). Ensure that User B, having activated the CDIVN service, receives a NOTIFY request indicating the call diversion. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C Activation CDIVN SUBSCRIBE 200 OK SUBSCRIBE NOTIFY 200 OK NOTIFY User B becomes "Not reachable" (indication not specified) INVITE Communication diversion is performed 181 Call Is Being Forwarded (optional) INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Communication BYE BYE 200 OK BYE 200 OK BYE User B becomes "Reachable" NOTIFY 200 OK NOTIFY SSS__XXSSCFNRc02 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNRc. Configuration: The user B has subscribed to CFNRc Subscription options: Originating user receives notification that his communication has been diverted = No Selection criteria: CFNRC supported. Test purpose: Ensure that when user A calls user B which is unreachable, the call is forwarded to user C. Ensure that User A does not receive a 181 Call Is Being Forwarded message. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 87 SSS__XXSSCFNRc03 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNRc. Configuration: The user B has subscribed to CFNRc and has not activated TIR 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 = No Served user allows the presentation of his/her URI to originating user in diversion notification = No Selection criteria: CFNRC supported. Test purpose: Ensure that when user A calls user B which is unreachable, the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message containing a Privacy header with value "id" and not containing a P-Asserted-Identity indicating the URI of user B and not containing a History-Info header (with CDIV related cause value) indicating the URI of user B or user A. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C INVITE INVITE 181 Call Is Being Forwarded 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE SSS__XXSSCFNRc04 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNRc. Configuration: The user B has subscribed to CFNRc and has not activated TIR 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 Served user allows the presentation of his/her URI to originating user in diversion notification = Yes Selection criteria: CFNRC supported. Test purpose: Ensure that when user A calls user B which is unreachable, the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message containing a P-Asserted-Identity indicating the URI of user B and containing a History-Info header • including a first entry with the hi-targeted-to-URI of user B, index = 1, cause param = 503 and • including a second entry with the hi-targeted-to-URI of user C, index = 1.1 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C INVITE INVITE 181 Call Is Being Forwarded 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 88 SSS__XXSSCFNRc05 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.2.2 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNRc. Configuration: The user B has subscribed to CFNRc and has not activated OIR Subscription options: Served user allows the presentation of his/her URI to diverted-to user = Yes Selection criteria: CFNRC supported. Test purpose: Ensure that when user A calls user B which is unreachable, the call is forwarded to user C. Ensure that User C receives an INVITE message containing a History-Info header including an entry (with CDIV related cause value) with the hi-targeted-to-URI of user B. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C INVITE INVITE 181 Call Is Being Forwarded (optional) 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.6.5 CFNL	SSS__XXSSCFNL01 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.5.1 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNL. Configuration: The user B has subscribed to CFNL and CDIVN Selection criteria: CFNL and CDIVN supported. Test purpose: Ensure that when user A calls user B which is not logged in, the call is forwarded to user C. Ensure that in the active call state the voice transfer on the media channels is performed correctly (e.g. testing QoS parameters). Ensure that User B, having activated the CDIVN service, receives a NOTIFY request indicating the call diversion. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C Activation CDIVN SUBSCRIBE 200 OK SUBSCRIBE NOTIFY 200 OK NOTIFY Log off User B INVITE Communication diversion is performed 181 Call Is Being Forwarded (optional) INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Communication BYE BYE 200 OK BYE 200 OK BYE Log in User B NOTIFY 200 OK NOTIFY ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 89 SSS__XXSSCFNL02 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNL. Configuration: The user B has subscribed to CFNL Subscription options: Originating user receives notification that his communication has been diverted = No Selection criteria: CFNL supported. Test purpose: Ensure that when user A calls user B which is not logged in, the call is forwarded to user C. Ensure that User A does not receive a 181 Call Is Being Forwarded message. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C INVITE INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE SSS__XXSSCFNL03 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNL. Configuration: The user B has subscribed to CFNL and has not activated TIR 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 = No Served user allows the presentation of his/her URI to originating user in diversion notification = No Selection criteria: CFNL supported. Test purpose: Ensure that when user A calls user B which is not logged in, the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message containing a Privacy header with value "id" and not containing a P-Asserted-Identity indicating the URI of user B and not containing a History-Info header (with CDIV related cause value) indicating the URI of user B or user A. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C INVITE INVITE 181 Call Is Being Forwarded 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 90 SSS__XXSSCFNL04 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNL. Configuration: The user B has subscribed to CFNL and has not activated TIR 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 Served user allows the presentation of his/her URI to originating user in diversion notification = Yes Selection criteria: CFNL supported. Test purpose: Ensure that when user A calls user B which is not logged in, the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message containing a P-Asserted-Identity indicating the URI of user B and containing a History-Info header • including a first entry with the hi-targeted-to-URI of user B, index = 1, cause param = 404 and • including a second entry with the hi-targeted-to-URI of user C, index = 1.1 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C INVITE INVITE 181 Call Is Being Forwarded 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE SSS__XXSSCFNL05 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.2.2 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFNL. Configuration: The user B has subscribed to CFNL and has not activated OIR Subscription options: Served user allows the presentation of his/her URI to diverted-to user = Yes Selection criteria: CFNL supported. Test purpose: Ensure that when user A calls user B which is not logged in, the call is forwarded to user C. Ensure that User C receives an INVITE message containing a History-Info header (with CDIV related cause value) including an entry with the hi-targeted-to-URI of user B. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA C INVITE INVITE 181 Call Is Being Forwarded (optional) 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 91
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.6.6 CD
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.6.6.1 CD Immediate	SSS__XXSSCD01 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.5 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFB. Configuration: The user B has subscribed to CD, CDIVN is not activated Subscription options: Served user receives indication that a communication has been forwarded = Yes Selection criteria: CD supported. Test purpose: Ensure that when user A calls user B which deflects the communication towards user C immediately (i.e. before alerting starts), the call is forwarded to user C. Ensure that in the active call state the voice transfer on the media channels is performed correctly (e.g. testing QoS parameters). Ensure that User B receives a MESSAGE request indicating the call diversion. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 302 Moved Temporarily ACK Communication diversion is performed 181 Call Is Being Forwarded (optional) INVITE MESSAGE 200 OK MESSAGE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Communication BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 92 SSS__XXSSCD02 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.5.1 TSS reference: SIP-SIP-SIP/Supplementary_Services/CFB. Configuration: The user B has subscribed to CD and CDIVN Selection criteria: CD and CDIVN supported. Test purpose: Ensure that when user A calls user B which deflects the communication towards user C immediately (i.e. before alerting starts), the call is forwarded to user C. Ensure that in the active call state the voice transfer on the media channels is performed correctly (e.g. testing QoS parameters). Ensure that User B, having activated the CDIVN service, receives a NOTIFY request indicating the call diversion. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C Start Activation CDIVN SUBSCRIBE 200 OK SUBSCRIBE NOTIFY 200 OK NOTIFY End Activation CDIVN INVITE INVITE 302 Moved Temporarily ACK Communication diversion is performed 181 Call Is Being Forwarded (optional) INVITE NOTIFY 200 OK NOTIFY 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Communication BYE BYE 200 OK BYE 200 OK BYE SSS__XXSSCD03 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CD. Configuration: The user B has subscribed to CD Subscription options: Originating user receives notification that his communication has been diverted = No Selection criteria: CD supported. Test purpose: Ensure that when user A calls user B which deflects the communication towards user C immediately (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. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 302 Moved Temporarily ACK Communication deflection is performed INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 93 SSS__XXSSCD04 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CD. Configuration: The user B has subscribed to CD and has not activated TIR 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 = No Served user allows the presentation of his/her URI to originating user in diversion notification = No Selection criteria: CD supported. Test purpose: Ensure that when user A calls user B which deflects the communication towards user C immediately (i.e. before alerting starts), the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message containing a Privacy header with value "id" and not containing a P-Asserted-Identity indicating the URI of user B and not containing a History-Info header indicating the URI of user B or user A. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 302 Moved Temporarily ACK Communication deflection is performed 181 Call Is Being Forwarded INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 94 SSS__XXSSCD05 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CD. Configuration: The user B has subscribed to CD and has not activated TIR 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 Served user allows the presentation of his/her URI to originating user in diversion notification = Yes Selection criteria: CD supported. Test purpose: Ensure that when user A calls user B which deflects the communication towards user C immediately (i.e. before alerting starts), the call is forwarded to user C. Ensure that User A receives a 181 Call Is Being Forwarded message containing a P-Asserted-Identity indicating the URI of user B and containing a History-Info header • including a first entry with the hi-targeted-to-URI of user B, index = 1, cause param = 480 and • including a second entry with the hi-targeted-to-URI of user C, index = 1.1 NOTE: "index of these new H-I entries may be different if other entries have been added to H-I header." SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 302 Moved Temporarily ACK Communication deflection is performed 181 Call Is Being Forwarded INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE SSS__XXSSCD06 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.2.2 TSS reference: SIP-SIP-SIP/Supplementary_Services/CD. Configuration: The user B has subscribed to CD and has not activated OIR Subscription options: Served user allows the presentation of his/her URI to diverted-to user = Yes Selection criteria: CD supported. Test purpose: Ensure that when user A calls user B which deflects the communication towards user C immediately (i.e. before alerting starts), the call is forwarded to user C. Ensure that User C receives an INVITE message containing a History-Info header (with CDIV related cause value) including an entry with the hi-targeted-to-URI of user B. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 302 Moved Temporarily ACK Communication deflection is performed 181 Call Is Being Forwarded INVITE 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 95
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.6.6.2 CD during alerting	SSS__XXSSCD07 CDIV reference to: TS 124 504 [12], clause 4.5.2.6.4 TSS reference: SIP-SIP-SIP/Supplementary_Services/CD. Configuration: The user B has subscribed to CD Subscription options: Originating user receives notification that his communication has been diverted = No Selection criteria: CD supported. Test purpose: Ensure that when user A calls user B which deflects the communication towards user C during alerting, the call is forwarded to user C. Ensure that User A does not receive a 181 Call Is Being Forwarded message. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT Comments: SIP UA A SUT SIP UA B SIP UA C INVITE INVITE 180 Ringing 180 Ringing 302 Moved Temporarily ACK Communication deflection is performed INVITE 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 96
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.7 Test purposes for CONF
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.7.1 Conference creation	SSS__XXSSCONF_C RE_001 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.3 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_CRE. Configuration: CONF Selection criteria: Conference creation by Three-way session creation. REFER request to the user, Conference event package is subscribed. Test purpose: Creation of the conference Ensure that, when User A sends an INVITE request with request URI set to a valid conference factory URI: • User A receives a 200 OK SIP response from the conference focus containing "isfocus" feature parameter in Contact header. User A shall store the content of the receive Contact header as the conference URI. • User A sends an ACK SIP request. • User A sends a SUBSCRIBE request with request URI set to the conference URI (previously stored) and the Event header set to "conference". • User A receives a 200 OK SIP response to the SUBSCRIBE request. • User A receives a NOTIFY request (on the same dialog of the SUBSCRIBE previously sent) with the Event header set to "conference". • User A sends a 200 OK SIP response to the NOTIFY request. Inviting users to the conference For each active SIP session, User A sends a REFER request to the remote user (User B or User C) with request URI set to the URI of the address of the remote user and Refer-To header set to the conference URI previously stored (the parameter "method" set to INVITE in the Refer-To header can be included or omitted): • Remote user receives a REFER request containing the Refer-To header set to the conference URI. • Remote user sends a 202 Accepted SIP response to the REFER request. • User A receives a 202 Accepted SIP response to the REFER request. • Remote user sends an INVITE request with request URI set to conference URI to the conference focus. • Remote user sends a NOTIFY request to the User A (on the same dialog of the REFER previously received) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user receives a 200 OK SIP response to the NOTIFY request. • Remote user receives a 200 OK SIP response to the INVITE request from the conference focus. • Remote user sends an ACK to the conference focus. • Remote user sends a NOTIFY request to the User A (on the same dialog of the REFER previously received) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user receives a 200 OK SIP response to the NOTIFY request. • User A sends a BYE request to the remote user in order to release the active SIP session between the user A and the remote user. • Remote user receives a BYE request from user A. • Remote user sends a 200 OK SIP response to the BYE request. • User A receives a 200 OK SIP response to the BYE request. • User A receives a NOTIFY from the conference focus (on the same dialog of the SUBSCRIBE previously sent). • User A sends a 200 OK SIP response to the NOTIFY request. NOTE: Additionally, User A may include the Referred-By header to the REFER and set it to his SIP URI. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 97 Precondition: • User A was participating in two SIP sessions (one with User B and the other with User C). • The SIP session between User A and User B was previously put on HOLD by User A. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: INVITE: Request URI contains the conference factory URI 200 OK: "isfocus" feature parameter indicated in Contact header field conference URI contains in the Contact header field SUBSCRIBE: Request URI contains the conference URI, Event header contains "conference" REFER: Refer-to header contains the conference URI NOTIFY : Event header contains conference; Subscription-State header contains active, application/conference-info+xml contains connected, dialled-in NOTIFY 1: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying NOTIFY 2: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK NOTIFY 3: Event header contains conference; Subscription-State header contains active, application/conference-info+xml contains connected, dialled-in NOTIFY 4: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying NOTIFY 5: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK NOTIFY 6: Event header contains conference; Subscription-State header contains active, application/conference-info+xml contains connected, dialled-in Comments: SIP UA A Focus SIP UA B SIP UA C Establishment of session #1 INVITE INVITE 180 Ringing 180 Ringing 200 OK (INVITE) 200 OK (INVITE) ACK ACK INVITE (sendonly) INVITE (sendonly) 200 OK (recvonly) 200 OK (recvonly) ACK ACK Establishment of session #2 INVITE INVITE 180 Ringing 180 Ringing 200 OK (INVITE) 200 OK (INVITE) ACK ACK Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK SUBSCRIBE SUBSCRIBE 200 OK (SUBSCRIBE) 200 OK (SUBSCRIBE) NOTIFY NOTIFY 200 OK (NOTIFY) 200 OK (NOTIFY) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 98 Inviting UA B to the conference REFER REFER 202 Accepted 202 Accepted INVITE INVITE NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK 200 OK ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) BYE BYE 200 OK (BYE) 200 OK (BYE) NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) Inviting UA C to the conference REFER REFER 202 Accepted 202 Accepted INVITE INVITE NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4) 200 OK 200 OK ACK ACK NOTIFY 5 NOTIFY 5 200 OK (NOTIFY 5) 200 OK (NOTIFY 5) BYE BYE 200 OK (BYE) 200 OK (BYE) NOTIFY 6 NOTIFY 6 200 OK (NOTIFY 6) 200 OK (NOTIFY 6) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 99 SSS__XXSSCONF _CRE_002 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.3 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_CRE. Configuration: CONF Selection criteria: Conference creation by Three-way session creation. REFER request to the user, Conference event package not subscribed. Test purpose: Creation of the conference Ensure that, when User A sends an INVITE request with request URI set to a valid conference factory URI: • User A receives a 200 OK SIP response from the conference focus containing "isfocus" feature parameter in Contact header. User A shall store the content of the receive Contact header as the conference URI. • User A sends an ACK SIP request. Inviting users to the conference For each active SIP session, User A sends a REFER request to the remote user (User B or User C) with request URI set to the URI of the address of the remote user and Refer-To header set to the conference URI previously stored (the parameter "method" set to INVITE in the Refer-To header can be included or omitted): • Remote user receives a REFER request containing the Refer-To header set to the conference URI. • Remote user sends a 202 Accepted SIP response to the REFER request. • User A receives a 202 Accepted SIP response to the REFER request. • Remote user sends an INVITE request with request URI set to conference URI to the conference focus. • Remote user sends a NOTIFY request to the User A (on the same dialog of the REFER previously received) with the Event header set to "refer" and the Content- Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user receives a 200 OK SIP response to the NOTIFY request. • Remote user receives a 200 OK SIP response to the INVITE request from the conference focus. • Remote user sends an ACK to the conference focus. • Remote user sends a NOTIFY request to the User A (on the same dialog of the REFER previously received) with the Event header set to "refer" and the Content- Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user receives a 200 OK SIP response to the NOTIFY request. • User A sends a BYE request to the remote user in order to release the active SIP session between the user A and the remote user. • Remote user receives a BYE request from user A. • Remote user sends a 200 OK SIP response to the BYE request. • User A receives a 200 OK SIP response to the BYE request. NOTE: Additionally, User A may include the Referred-By header to the REFER and set it to his SIP URI. Precondition: • User A was participating in two SIP sessions (one with User B and the other with User C). • The SIP session between User A and User B was previously put on HOLD by User A. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 100 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: INVITE: Request URI contains the conference factory URI. 200 OK: "isfocus" feature parameter indicated in Contact header field conference URI contains in the Contact header field. REFER: Refer-to header contains the conference URI. NOTIFY 1: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 2: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. NOTIFY 3: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 4: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. Comments: SIP UA A Focus SIP UA B SIP UA C Establishment of session #1 INVITE INVITE 180 Ringing 180 Ringing 200 OK (INVITE) 200 OK (INVITE) ACK ACK INVITE (sendonly) INVITE (sendonly) 200 OK (recvonly) 200 OK (recvonly) ACK ACK Establishment of session #2 INVITE INVITE 180 Ringing 180 Ringing 200 OK (INVITE) 200 OK (INVITE) ACK ACK Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK Inviting UA B to the conference REFER REFER 202 Accepted 202 Accepted INVITE INVITE NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK 200 OK (INVITE) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) BYE BYE 200 OK (BYE) 200 OK (BYE) Inviting UA C to the conference REFER REFER 202 Accepted 202 Accepted INVITE INVITE NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) 200 OK 200 OK ACK ACK NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4) BYE BYE 200 OK (BYE) 200 OK (BYE) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 101 SSS__XXSSCONF_ CRE_003 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.3 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_CRE. Configuration: CONF Selection criteria: Conference creation by Three-way session creation. REFER request to the conference focus, Conference event package subscribed. Test purpose: Creation of the conference Ensure that, when User A sends an INVITE request with request URI set to a valid conference factory URI: • User A receives a 200 OK SIP response from the conference focus containing "isfocus" feature parameter in Contact header. User A shall store the content of the receive Contact header as the conference URI. • User A sends an ACK SIP request. • User A sends a SUBSCRIBE request with request URI set to the conference URI (previously stored) and the Event header set to "conference". • User A receives a 200 OK SIP response to the SUBSCRIBE request. • User A receives a NOTIFY request (on the same dialog of the SUBSCRIBE previously sent) with the Event header set to "conference". • User A sends a 200 OK SIP response to the NOTIFY request. Inviting users to the conference For each active SIP session, User A sends a REFER request to the conference focus with request URI set to the conference URI previously stored and Refer-To header set to the SIP URI of the remote user (the parameter "method" set to INVITE in the Refer-To header can be included or omitted): • User A receives a 202 Accepted SIP response to the REFER request. • Remote user receives an INVITE request from the conference focus to be invited to the conference. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user sends a 200 OK SIP response to the INVITE request from the conference focus. • Remote user receives an ACK from the conference focus. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A sends a 200 OK SIP response to the NOTIFY request. • User A sends a BYE request to the remote user in order to release the active SIP session between the user A and the remote user. • Remote user receives a BYE request from user A. • Remote user sends a 200 OK SIP response to the BYE request. • User A receives a 200 OK SIP response to the BYE request. • User A receives a NOTIFY from the conference focus (on the same dialog of the SUBSCRIBE previously sent). • User A sends a 200 OK SIP response to the NOTIFY request. NOTE: Additionally, User A may include the Referred-By header to the REFER and set it to his SIP URI. Precondition: • User A was participating in two SIP sessions (one with User B and the other with User C). • The SIP session between User A and User B was previously put on HOLD by User A. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 102 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: INVITE: Request URI contains the conference factory URI. 200 OK: "isfocus" feature parameter indicated in Contact header field conference URI contains in the Contact header field. SUBSCRIBE: Request URI contains the conference URI. REFER 1: Refer-to header contains the SIP URI of the UA B. NOTIFY: Event header contains conference; Subscription-State header contains active, application/conference-info+xml contains connected, dialled-in NOTIFY 1: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 2: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. NOTIFY 3: Event header contains conference; Subscription-State header contains active, application/conference-info+xml contains connected, dialled-out. REFER 2: Refer-to header contains the URI of the UA C. NOTIFY 4: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 5: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag contains SIP/2.0 200 OK. NOTIFY 6: Event contains conference; Subscription-State contains active application/conference-info+xml contains connected, dialled-out. Comments: SIP UA A Focus SIP UA B SIP UA C Establishment of session #1 INVITE INVITE 180 Ringing 180 Ringing 200 OK (INVITE) 200 OK (INVITE) ACK ACK INVITE (sendonly) INVITE(sendonly) 200 OK (recvonly) 200 OK (recvonly) ACK ACK Establishment of session #2 INVITE INVITE 180 Ringing 180 Ringing 200 OK (INVITE) 200 OK (INVITE) ACK ACK Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK SUBSCRIBE SUBSCRIBE 200 OK (SUBSCRIBE) 200 OK (SUBSCRIBE) NOTIFY NOTIFY 200 OK (NOTIFY) 200 OK (NOTIFY) Inviting UA B to the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE INVITE NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK (INVITE) 200 OK (INVITE) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) BYE BYE 200 OK (BYE) 200 OK (BYE) NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 103 Inviting UA C to the conference REFER 2 REFER 2 202 Accepted 202 Accepted INVITE INVITE NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4) 200 OK (INVITE) 200 OK (INVITE) ACK ACK NOTIFY 5 NOTIFY 5 200 OK (NOTIFY 5) 200 OK (NOTIFY 5) BYE BYE 200 OK (BYE) 200 OK (BYE) NOTIFY 6 NOTIFY 6 200 OK (NOTIFY 6) 200 OK (NOTIFY 6) SSS__XXSSCONF_ CRE_004 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.3 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_CRE. Configuration: CONF Selection criteria: Conference creation by Three-way session creation. REFER request to the focus, Conference event package not subscribed. Test purpose: Creation of the conference Ensure that, when User A sends an INVITE request with request URI set to a valid conference factory URI: • User A receives a 200 OK SIP response from the conference focus containing "isfocus" feature parameter in Contact header. User A shall store the content of the receive Contact header as the conference URI. • User A sends an ACK SIP request. Inviting users to the conference For each active SIP session, User A sends a REFER request to the conference focus with request URI set to the conference URI previously stored and Refer-To header set to the SIP URI of the remote user (the parameter "method" set to INVITE in the Refer-To header can be included or omitted): • User A receives a 202 Accepted SIP response to the REFER request. • Remote user receives an INVITE request from the conference focus to be invited to the conference. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user sends a 200 OK SIP response to the INVITE request from the conference focus. • Remote user receives an ACK from the conference focus. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A sends a 200 OK SIP response to the NOTIFY request. • User A sends a BYE request to the remote user in order to release the active SIP session between the user A and the remote user. • Remote user receives a BYE request from user A. • Remote user sends a 200 OK SIP response to the BYE request. • User A receives a 200 OK SIP response to the BYE request. NOTE: Additionally, User A may include the Referred-By header to the REFER and set it to his SIP URI. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 104 Precondition: • User A was participating in two SIP sessions (one with User B and the other with User C). • The SIP session between User A and User B was previously put on HOLD by User A. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: INVITE: Request URI contains the conference factory URI. 200 OK: "isfocus" feature parameter indicated in Contact header field conference URI contains in the Contact header field. REFER 1: Refer-to header contains the SIP URI of the UA B. NOTIFY 1: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 2: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. REFER 2: Refer-to header contains the URI of the UA C. NOTIFY 3: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 4: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. Comments: SIP UA A Focus SIP UA B SIP UA C Establishment of session #1 INVITE INVITE 180 Ringing 180 Ringing 200 OK (INVITE) 200 OK (INVITE) ACK ACK INVITE (sendonly) INVITE (sendonly) 200 OK (recvonly) 200 OK (recvonly) ACK ACK Establishment of session #2 INVITE INVITE 180 Ringing 180 Ringing 200 OK (INVITE) 200 OK (INVITE) ACK ACK Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK Inviting UA B to the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE INVITE NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK (INVITE) 200 OK (INVITE) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) BYE BYE 200 OK (BYE) 200 OK (BYE) Inviting UA C to the conference REFER 2 REFER 2 202 Accepted 202 Accepted INVITE INVITE NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) 200 OK (INVITE) 200 OK (INVITE) ACK ACK NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4) BYE BYE 200 OK (BYE) 200 OK (BYE) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 105 SSS__XXSSCONF_ CRE_005 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.3 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_CRE. Configuration: CONF Selection criteria: Conference creation by Three-way session creation. REFER request to the focus, Replaces method is used, Conference event package subscribed. Test purpose: Creation of the conference Ensure that, when User A sends an INVITE request with request URI set to a valid conference factory URI: • User A receives a 200 OK SIP response from the conference focus containing "isfocus" feature parameter in Contact header. User A shall store the content of the receive Contact header as the conference URI. • User A sends an ACK SIP request. • User A sends a SUBSCRIBE request with request URI set to the conference URI (previously stored) and the Event header set to "conference". • User A receives a 200 OK SIP response to the SUBSCRIBE request. • User A receives a NOTIFY request (on the same dialog of the SUBSCRIBE previously sent) with the Event header set to "conference". • User A sends a 200 OK SIP response to the NOTIFY request. Inviting users to the conference For each active SIP session, User A sends a REFER request to the conference focus with request URI set to the conference URI previously stored and Refer-To header set to the SIP URI of the remote user. Also, into the Refer-to header the replaces method is used in order to terminate the active SIP session between the user A and the remote user: • User A receives a 202 Accepted SIP response to the REFER request. • Remote user receives an INVITE request from the conference focus to be invited to the conference. The INVITE contains the Replaces header with SIP dialog data ("Call-ID", "From" tag, "To" tag) to be replaced. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user sends a 200 OK SIP response to the INVITE request from the conference focus. • Remote user receives an ACK from the conference focus. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user sends a BYE request to the User A in order to release the active SIP session between the user A and the remote user. • User A receives a BYE request from remote user. • User A sends a 200 OK SIP response to the BYE request. • Remote user receives a 200 OK SIP response to the BYE request. • User A receives a NOTIFY from the conference focus (on the same dialog of the SUBSCRIBE previously sent). • User A sends a 200 OK SIP response to the NOTIFY request. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 106 Precondition: • User A was participating in two SIP sessions (one with User B and the other with User C). • The SIP session between User A and User B was previously put on HOLD by User A. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: INVITE: Request URI contains the conference factory URI. 200 OK: "isfocus" feature parameter indicated in Contact header field conference URI contains in the Contact header field. NOTIFY: Event header contains conference; Subscription-State header contains active, application/conference-info+xml contains connected, dialled-in REFER 1: Refer-to header contains the SIP URI of the UA B. Refer-To: <sip: URI-B?Replaces=call-id1%3Bto-tagsession1%3Bfrom- tagSession1; method=INVITE>. NOTIFY 1: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 2: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. NOTIFY 3: Event header contains conference; Subscription-State header contains active, application/conference-info+xml contains connected, dialled-out. BYE 1: Call-ID: call-id1/ To: ….; tag=session1/ From: …;tag=Session1. REFER 2: Refer-to header contains the SIP URI of the UA Cand Replaces header for session 2. Refer-To: <sip: URI-C?Replaces=call-id2%3Bto-tagsession2%3Bfrom- tagSession2; method=INVITE>. NOTIFY 4: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 5: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag contains SIP/2.0 200 OK. NOTIFY 6: Event contains conference; Subscription-State contains active application/conference-info+xml contains connected, dialled-out. BYE 2: Call-ID: call-id2/ To: ….; tag=session2/ From: …;tag=Session2. Comments: SIP UA A Focus SIP UA B SIP UA C Establishment of session #1 INVITE INVITE 180 Ringing 180 Ringing 200 OK (INVITE) 200 OK (INVITE) ACK ACK INVITE (sendonly) INVITE (sendonly) 200 OK (recvonly) 200 OK (recvonly) ACK ACK Establishment of session #2 INVITE INVITE 180 Ringing 180 Ringing 200 OK (INVITE) 200 OK (INVITE) ACK ACK Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK SUBSCRIBE SUBSCRIBE 200 OK (SUBSCRIBE) 200 OK (SUBSCRIBE) NOTIFY NOTIFY 200 OK (NOTIFY) 200 OK (NOTIFY) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 107 Inviting UA B to the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 4 INVITE 4 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK (INVITE 4) 200 OK (INVITE 4) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) BYE 1 BYE 1 200 OK (BYE 1) 200 OK (BYE 1) NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) Inviting UA C to the conference REFER 2 REFER 2 202 Accepted 202 Accepted INVITE 5 INVITE 5 NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4) 200 OK (INVITE5) 200 OK(INVITE 5) ACK ACK NOTIFY 5 NOTIFY 5 200 OK (NOTIFY 5) 200 OK (NOTIFY 5) BYE 2 BYE 2 200 OK (BYE 2) 200 OK (BYE 2) NOTIFY 6 NOTIFY 6 200 OK (NOTIFY 6) 200 OK (NOTIFY 6) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 108 SSS__XXSSCONF_ CRE_06 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.3 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_CRE. Configuration: CONF Selection criteria: Conference creation by Three-way session creation. REFER request to the focus, Replaces method is used, Conference event package not subscribed. Test purpose: Creation of the conference Ensure that, when User A sends an INVITE request with request URI set to a valid conference factory URI: • User A receives a 200 OK SIP response from the conference focus containing "isfocus" feature parameter in Contact header. User A shall store the content of the receive Contact header as the conference URI. • User A sends an ACK SIP request. Inviting users to the conference For each active SIP session, User A sends a REFER request to the conference focus with request URI set to the conference URI previously stored and Refer-To header set to the SIP URI of the remote user. Also, into the Refer-to header the replaces method is used in order to terminate the active SIP session between the user A and the remote user: • User A receives a 202 Accepted SIP response to the REFER request. • Remote user receives an INVITE request from the conference focus to be invited to the conference. The INVITE contains the Replaces header with SIP dialog data ("Call-ID", "From" tag, "To" tag) to be replaced. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user sends a 200 OK SIP response to the INVITE request from the conference focus. • Remote user receives an ACK from the conference focus. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user sends a BYE request to the User A in order to release the active SIP session between the user A and the remote user. • User A receives a BYE request from remote user. • User A sends a 200 OK SIP response to the BYE request. • Remote user receives a 200 OK SIP response to the BYE request. Precondition: • User A was participating in two SIP sessions (one with User B and the other with User C). • The SIP session between User A and User B was previously put on HOLD by User A. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 109 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: INVITE: Request URI contains the conference factory URI. 200 OK: "isfocus" feature parameter indicated in Contact header field conference URI contains in the Contact header field. REFER 1: Refer-to header contains the URI of user#2 and Replaces header for session 1. Refer-To: <sip:User#2?Replaces=Call-ID1%3Bto-tagsession1%3Bfrom- tagSession1; method=INVITE>. INVITE 4: Replaces: Call-ID1; to-tag=to-tagSession1; from-tag=from-tagSession1. NOTIFY 1: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 2: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. BYE 1: Call-ID: call-id1/ To: ….; tag=session1/ From: …;tag=Session1. REFER 2: Refer-to header contains the URI of user#3 and Replaces header for session 2. Refer-To: <sip:User#3?Replaces=Call-ID2%3Bto-tag session2%3Bfrom- tag Session2; method=INVITE>. INVITE 5: Replaces: Call-ID2; to-tag=to-tagSession2; from-tag=from-tagSession2. NOTIFY 3: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 4: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. BYE 2: Call-ID: call-id2/ To: ….; tag=session2/ From: …;tag=Session2. Comments: SIP UA A Focus SIP UA B SIP UA C Establishment of session #1 INVITE INVITE 180 Ringing 180 Ringing 200 OK (INVITE) 200 OK (INVITE) ACK ACK INVITE (sendonly) INVITE (sendonly) 200 OK (recvonly) 200 OK (recvonly) ACK ACK Establishment of session #2 INVITE INVITE 180 Ringing 180 Ringing 200 OK (INVITE) 200 OK (INVITE) ACK ACK Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK Inviting UA B to the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 4 INVITE 4 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK (INVITE 4) 200 OK (INVITE 4) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) BYE 1 BYE 1 200 OK (BYE 1) 200 OK (BYE 1) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 110 Inviting UA C to the conference REFER 2 REFER 202 Accepted 202 Accepted INVITE 5 INVITE 5 NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) 200 OK (INVITE 5) 200 OK (INVITE5) ACK ACK NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4) BYE 2 BYE 2 200 OK (BYE 2) 200 OK (BYE 2) SSS__XXSSCONF_ CRE_007 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.3 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_CRE. Configuration: CONF Selection criteria: Conference creation by SIP URI-list. Conference event package subscribed. Test purpose: Ensure that, when User A sends an INVITE request with "resource-list+xml" body (which contains a SIP URI-list of the participants that User A wants to invite to the conference) and request URI set to a valid conference factory URI: • User A receives a 200 OK SIP response from the conference focus containing "isfocus" feature parameter in Contact header. User A shall store the content of the receive Contact header as the conference URI. • User A sends an ACK SIP request. • User A sends a SUBSCRIBE request with request URI set to the conference URI (previously stored) and the Event header set to "conference". • User A receives a 200 OK SIP response to the SUBSCRIBE request. • User A receives a NOTIFY request (on the same dialog of the SUBSCRIBE previously sent) with the Event header set to "conference". • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user (User B/User C) receives an INVITE request from the conference focus to be invited to the conference. • Remote user (User B/User C) sends a 180 Ringing SIP response to the INVITE request from the conference focus. • Remote user (User B/User C) sends a 200 OK SIP response to the INVITE request from the conference focus. • Remote user receives an ACK from the conference focus. • User A receives a NOTIFY from the conference focus (on the same dialog of the SUBSCRIBE previously sent). • User A sends a 200 OK SIP response to the NOTIFY request. Precondition: SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: INVITE: Request URI contains the conference factory URI, Require header contains "recipient-list-invite", Content-Disposition header contains "recipient-list", Content-Type header contains "application/resource- lists+xml" and the resource-lists+xml body contains the SIP URI-list of participants at the conference (according to RFC 5366 [25]). 200 OK: "isfocus" feature parameter indicated in Contact header field conference URI contained in the Contact header field. SUBSCRIBE: Request URI contained the conference URI. NOTIFY: Event header contains conference; Subscription-State header contains active, application/conference-info+xml contains connected, dialled-in INVITE 2: The P-Asserted-Identity contains the conference URI. "isfocus" feature parameter indicated in Contact header field conference URI contained in the Contact header field. Referred-By contains SIP or tel URI of UA A. (This is not mandatory) INVITE 3: The P-Asserted-Identity contains the conference URI. "isfocus" feature parameter indicated in Contact header field conference URI contained in the Contact header field. Referred-By contains SIP or tel URI of UA A. (This is not mandatory). ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 111 Comments: SIP UA A Focus SIP UA B SIP UA C Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK SUBSCRIBE SUBSCRIBE 200 OK (SUBSCRIBE) 200 OK (SUBSCRIBE) NOTIFY NOTIFY 200 OK (NOTIFY) 200 OK (NOTIFY) Inviting UA B to the conference INVITE 2 INVITE 2 180 Ringing 180 Ringing 200 OK (INVITE 2) 200 OK(INVITE2) ACK ACK NOTIFY NOTIFY 200 OK (NOTIFY) 200 OK (NOTIFY) Inviting UA C to the conference INVITE 3 INVITE 3 180 Ringing 180 Ringing 200 OK (INVITE 3) 200 OK (INVITE 3) ACK ACK NOTIFY NOTIFY 200 OK (NOTIFY) 200 OK (NOTIFY) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 112 SSS__XXSSCONF_ CRE_008 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.3 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_CRE. Configuration: CONF Selection criteria: Conference creation by SIP URI-list. Conference event package not subscribed. Test purpose: Ensure that, when User A sends an INVITE request with "resource-list+xml" body (which contains a SIP URI-list of the participants that User A wants to invite to the conference) and request URI set to a valid conference factory URI: • User A receives a 200 OK SIP response from the conference focus containing "isfocus" feature parameter in Contact header. User A shall store the content of the receive Contact header as the conference URI. • User A sends an ACK SIP request. • Remote user (User B/User C) receives an INVITE request from the conference focus to be invited to the conference. • Remote user (User B/User C) sends a 180 Ringing SIP response to the INVITE request from the conference focus. • Remote user (User B/User C) sends a 200 OK SIP response to the INVITE request from the conference focus. • Remote user receives an ACK from the conference focus. Precondition: SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: INVITE: Request URI contains the conference factory URI, Require header contains "recipient-list-invite", Content-Disposition header contains "recipient-list", Content-Type header contains "application/resource- lists+xml" and the resource-lists+xml body contains the SIP URI-list of participants at the conference (according to RFC 5366 [25]). 200 OK: "isfocus" feature parameter indicated in Contact header field conference URI contained in the Contact header field. INVITE 2: The P-Asserted-Identity contains the conference URI. "isfocus" feature parameter indicated in Contact header field conference URI contained in the Contact header field. Referred-By contains SIP or tel URI of UA A. (This is not mandatory) INVITE 3: The P-Asserted-Identity contains the conference URI. "isfocus" feature parameter indicated in Contact header field conference URI contained in the Contact header field. Referred-By contains SIP or tel URI of UA A. (This is not mandatory). Comments: SIP UA A Focus SIP UA B SIP UA C Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK Inviting UA B to the conference INVITE 2 INVITE 2 180 Ringing 180 Ringing 200 OK (INVITE 2) 200 OK (INVITE 2) ACK ACK Inviting UA C to the conference INVITE 3 INVITE 3 180 Ringing 180 Ringing 200 OK (INVITE 3) 200 OK(INVITE 3) ACK ACK ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 113 SSS__XXSSCONF_ CRE_09 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.3 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_CRE. Configuration: CONF Selection criteria: Unsuccessful. Conference creation with a conference factory URI not allocated by the conference focus. Test purpose: Ensure that, when User A sends an INVITE request with request URI set to a not valid conference factory URI: • User A receives a 488 Not Acceptable Here SIP response from the conference focus. • User A sends an ACK SIP request Precondition: SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: INVITE: Request URI contains a conference factory URI not allocated by the conference focus. Comments: SIP UA A Focus INVITE INVITE 488 Not Acceptable Here 488 Not Acceptable Here ACK ACK ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 114
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.7.2 Inviting other users to a conference	SSS__XXSSCONF_I NV_001 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.4, 5.3.1.5 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_INV. Configuration: CONF Selection criteria: Inviting participant by sending REFER to the conference focus. The conference event package is subscribed. Test purpose: Ensure that, when User A sends a REFER request to the conference focus with request URI set to the conference URI previously stored and Refer-To header set to the SIP URI of the remote user (the parameter "method" set to INVITE in the Refer-To header can be included or omitted): • User A receives a 202 Accepted SIP response to the REFER request. • Remote user receives an INVITE request from the conference focus to be invited to the conference. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user sends a 180 Ringing SIP response to the INVITE request from the conference focus. • Remote user sends a 200 OK SIP response to the INVITE request from the conference focus. • Remote user receives an ACK from the conference focus. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user sends a SUBSCRIBE request with request URI set to the conference URI (previously stored) and the Event header set to "conference". • Remote User receives a 200 OK SIP response to the SUBSCRIBE request. • Remote user receives a NOTIFY request (on the same dialog of the SUBSCRIBE previously sent) with the Event header set to "conference". • Remote user sends a 200 OK SIP response to the NOTIFY request. Repeat the above steps twice in order to invite to the conference User B (when remote user is UA B) and User C (when remote user is UA C). When User C has joined the conference: • User B receives a NOTIFY request (on the same dialog of the SUBSCRIBE previously sent) with the Event header set to "conference". • User B sends a 200 OK SIP response to the NOTIFY request. NOTE: Additionally, User A may include the Referred-By header to the REFER and set it to his SIP URI. Precondition: • User A has created a conference by using a conference factory URI. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 115 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: REFER 1: Request URI contains the conference URI (previously stored). Refer-To header contains the SIP URI of UA B. Referred-By contains SIP or tel URI of UA A. (This is not mandatory) INVITE 2: The P-Asserted-Identity contains the conference URI. "isfocus" feature parameter indicated in Contact header field conference URI contained in the Contact header field. Referred-By contains SIP or tel URI of UA A. (This is not mandatory) NOTIFY 1: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body, contains SIP/2.0 100 Trying. NOTIFY 2: Event contains refer; Subscription-State header contains terminated, Content- Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. SUBSCRIBE: Request URI contained the conference URI, Event header contains "conference". NOTIFY 3: Event header contains conference; Subscription-State header contains active, application/conference-info+xml contains connected, dialled-out. REFER 2: Request URI contained the conference URI (previously stored). Refer-To header contains the SIP URI of UA C. Referred-By contains SIP or tel URI of UA A. (This is not mandatory) INVITE 3: The P-Asserted-Identity contains the conference URI. "isfocus" feature parameter indicated in Contact header field conference URI contained in the Contact header field. Referred-By contains SIP or tel URI of UA A. (This is not mandatory) NOTIFY 4: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body, contains SIP/2.0 100 Trying. NOTIFY 5: Event contains refer; Subscription-State header contains terminated, Content- Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. SUBSCRIBE: Request URI contained the conference URI, Event header contains "conference". NOTIFY 6: Event header contains conference; Subscription-State header contains active, application/conference-info+xml contains connected, dialled-out. NOTIFY 7: Event header contains conference; Subscription-State header contains active, application/conference-info+xml contains connected, dialled-out. Comments: SIP UA A Focus SIP UA B SIP UA C Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK Inviting UA B to the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 2 INVITE 2 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 180 Ringing 180 Ringing 200 OK (INVITE 2) 200 OK (INVITE 2) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) SUBSCRIBE SUBSCRIBE 200 OK (SUBSCRIBE) 200 OK(SUBSCRIBE) NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 116 Inviting UA C to the conference REFER 2 REFER 2 202 Accepted 202 Accepted INVITE 3 INVITE 3 NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4) 180 Ringing 180 Ringing 200 OK (INVITE 3) 200 OK (INVITE 3) ACK ACK NOTIFY 5 NOTIFY 5 200 OK (NOTIFY 5) 200 OK (NOTIFY 5) SUBSCRIBE SUBSCRIBE 200 OK (SUBSCRIBE) 200 OK(SUBSCRIBE) NOTIFY 6 NOTIFY 6 200 OK (NOTIFY 6) 200 OK (NOTIFY 6) NOTIFY 7 NOTIFY 7 200 OK (NOTIFY 7) 200 OK (NOTIFY 7) SSS__XXSSCONF_I NV_002 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.4, 5.3.1.5 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_INV. Configuration: CONF Selection criteria: Inviting participant by sending REFER to the conference focus. The conference event package is not subscribed. Test purpose: Ensure that, when User A sends a REFER request to the conference focus with request URI set to the conference URI previously stored and Refer-To header set to the SIP URI of the remote user (the parameter "method" set to INVITE in the Refer-To header can be included or omitted): • User A receives a 202 Accepted SIP response to the REFER request. • Remote user receives an INVITE request from the conference focus to be invited to the conference. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user sends a 180 Ringing SIP response to the INVITE request from the conference focus. • Remote user sends a 200 OK SIP response to the INVITE request from the conference focus. • Remote user receives an ACK from the conference focus. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A sends a 200 OK SIP response to the NOTIFY request. Repeat the above steps twice in order to invite to the conference User B (when remote user is UA B) and User C (when remote user is UA C). NOTE: Additionally, User A may include the Referred-By header to the REFER and set it to his SIP URI. Precondition: • User A has created a conference by using a conference factory URI. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 117 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: REFER 1: Request URI contains the conference URI (previously stored). Refer-To header contains the URI of UA B. Referred-By header contains SIP URI of UA A. (This is not mandatory) INVITE 2: The P-Asserted-Identity contains the conference URI. "isfocus" feature parameter indicated in Contact header field conference URI contained in the Contact header field. Referred-By contains SIP or tel URI of UA A. (This is not mandatory) NOTIFY 1: Event header contains refer; Subscription-State header contains active, Content- Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 2: Event header contains refer; Subscription-State header contains active, Content- Type header contains "message/sipfrag", message/sipfrag contains SIP/2.0 200 OK. REFER 2: Request URI contained the conference URI (previously stored). Refer-To header contains the URI of UA C. Referred-By header contains SIP URI of UA A. (This is not mandatory) INVITE 3: The P-Asserted-Identity contains the conference URI. "isfocus" feature parameter indicated in Contact header field conference URI contained in the Contact header field. Referred-By contains SIP or tel URI of UA A. (This is not mandatory) NOTIFY 3: Event header contains refer; Subscription-State header contains active, Content- Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 4: Event header contains refer; Subscription-State header contains active, Content- Type header contains "message/sipfrag", message/sipfrag contains SIP/2.0 200 OK. Comments: SIP UA A Focus SIP UA B SIP UA C Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK Inviting UA B to the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 2 INVITE 2 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 180 Ringing 180 Ringing 200 OK (INVITE 2) 200 OK (INVITE 2) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) Inviting UA C to the conference REFER 2 REFER 2 202 Accepted 202 Accepted INVITE 3 INVITE 3 NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) 180 Ringing 180 Ringing 200 OK (INVITE 3) 200 OK (INVITE 3) ACK ACK NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 118 SSS__XXSSCONF_ INV_003 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.4, 5.3.1.5 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_INV. Configuration: CONF Selection criteria: Inviting participant by sending REFER to the participant. The conference event package is subscribed. Test purpose: Ensure that, when User A sends a REFER request to the remote user with request URI set to the SIP URI of the remote user and Refer-To header set to the conference URI previously stored (the parameter "method" set to INVITE in the Refer-To header can be included or omitted): • Remote user receives a REFER request containing the Refer-To header set to the conference URI. • Remote user sends a 202 Accepted SIP response to the REFER request. • User A receives a 202 Accepted SIP response to the REFER request. • Remote user sends an INVITE request with request URI set to conference URI to the conference focus. • Remote user sends a NOTIFY request to the User A (on the same dialog of the REFER previously received) with the Event header set to "refer" and the Content- Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user receives a 200 OK SIP response to the NOTIFY request. • Remote user receives a 200 OK SIP response to the INVITE request from the conference focus. • Remote user sends an ACK to the conference focus. • Remote user sends a NOTIFY request to the User A (on the same dialog of the REFER previously received) with the Event header set to "refer" and the Content- Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user sends a SUBSCRIBE request with request URI set to the conference URI (previously stored) and the Event header set to "conference". • Remote User receives a 200 OK SIP response to the SUBSCRIBE request. • Remote user receives a NOTIFY request (on the same dialog of the SUBSCRIBE previously sent) with the Event header set to "conference". • Remote user sends a 200 OK SIP response to the NOTIFY request. Repeat the above steps twice in order to invite to the conference User B (when remote user is UA B) and User C (when remote user is UA C). When User C has joined the conference: • User B receives a NOTIFY request (on the same dialog of the SUBSCRIBE previously sent) with the Event header set to "conference". • User B sends a 200 OK SIP response to the NOTIFY request. NOTE: Additionally, User A may include the Referred-By header to the REFER and set it to his SIP URI. Precondition: • User A has created a conference by using a conference factory URI. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 119 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: REFER 1: Request URI contains the SIP URI of UA B Refer-To header contains the conference URI (previously stored). Referred-By contains SIP or tel URI of UA A. (This is not mandatory) INVITE 2: Request URI contains the conference URI. The P-Asserted-Identity contains the URI of UA B. "isfocus" feature parameter indicated in Contact header field conference URI contained in the Contact header field. Referred-By contains SIP or tel URI of UA A. (This is not mandatory) NOTIFY 1: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body, contains SIP/2.0 100 Trying. NOTIFY 2: Event contains refer; Subscription-State header contains terminated, Content- Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. SUBSCRIBE: Request URI contained the conference URI, Event header contains "conference". NOTIFY 3: Event header contains conference; Subscription-State header contains active, application/conference-info+xml contains connected, dialled-out. REFER 2: Request URI contains the SIP URI of UA C. Refer-To header contains the conference URI (previously stored). Referred-By contains SIP or tel URI of UA A. (This is not mandatory) INVITE 3: Request URI contains the conference URI. The P-Asserted-Identity contains the URI of UA C. "isfocus" feature parameter indicated in Contact header field conference URI contained in the Contact header field. Referred-By contains SIP or tel URI of UA A. (This is not mandatory) NOTIFY 4: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body, contains SIP/2.0 100 Trying. NOTIFY 5: Event contains refer; Subscription-State header contains terminated, Content- Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. NOTIFY 6: Event header contains conference; Subscription-State header contains active, application/conference-info+xml contains connected, dialled-out. NOTIFY 7: Event header contains conference; Subscription-State header contains active, application/conference-info+xml contains connected, dialled-out. Comments: SIP UA A Focus SIP UA B SIP UA C Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK Inviting UA B to the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 2 INVITE 2 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK (INVITE 2) 200 OK (INVITE 2) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) SUBSCRIBE SUBSCRIBE 200 OK (SUBSCRIBE) 200 OK (SUBSCRIBE) NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 120 Inviting UA C to the conference REFER 2 REFER 2 202 Accepted 202 Accepted INVITE 3 INVITE 3 NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4) 200 OK (INVITE 3) 200 OK (INVITE 3) ACK ACK NOTIFY 5 NOTIFY 5 200 OK (NOTIFY 5) 200 OK (NOTIFY 5) SUBSCRIBE SUBSCRIBE 200 OK (SUBSCRIBE) 200 OK (SUBSCRIBE) NOTIFY 6 NOTIFY 6 200 OK (NOTIFY 6) 200 OK (NOTIFY 6) NOTIFY 7 NOTIFY 7 200 OK (NOTIFY 7) 200 OK (NOTIFY 7) SSS__XXSSCONF_ INV_004 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.4, 5.3.1.5 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_INV. Configuration: CONF Selection criteria: Inviting participant by sending REFER to the participant. The conference event package is not subscribed. Test purpose: Ensure that, when User A sends a REFER request to the remote user with request URI set to the SIP URI of the remote user and Refer-To header set to the conference URI previously stored (the parameter "method" set to INVITE in the Refer-To header can be included or omitted): • Remote user receives a REFER request containing the Refer-To header set to the conference URI. • Remote user sends a 202 Accepted SIP response to the REFER request. • User A receives a 202 Accepted SIP response to the REFER request. • Remote user sends an INVITE request with request URI set to conference URI to the conference focus. • Remote user sends a NOTIFY request to the User A (on the same dialog of the REFER previously received) with the Event header set to "refer" and the Content- Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • Remote user receives a 200 OK SIP response to the NOTIFY request. • Remote user receives a 200 OK SIP response to the INVITE request from the conference focus. • Remote user sends an ACK to the conference focus. • Remote user sends a NOTIFY request to the User A (on the same dialog of the REFER previously received) with the Event header set to "refer" and the Content- Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A sends a 200 OK SIP response to the NOTIFY request. Repeat the above steps twice in order to invite to the conference User B (when remote user is UA B) and User C (when remote user is UA C). NOTE: Additionally, User A may include the Referred-By header to the REFER and set it to his SIP URI. Precondition: • User A has created a conference by using a conference factory URI. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 121 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: REFER 1: Request URI contains the SIP URI of UA B Refer-To header contains the conference URI (previously stored). Referred-By contains SIP or tel URI of UA A. (This is not mandatory) INVITE 2: Request URI contains the conference URI. The P-Asserted-Identity contains the URI of UA B. "isfocus" feature parameter indicated in Contact header field conference URI contained in the Contact header field. Referred-By contains SIP or tel URI of UA A. (This is not mandatory) NOTIFY 1: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body, contains SIP/2.0 100 Trying. NOTIFY 2: Event contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. REFER 2: Request URI contains the SIP URI of UA C. Refer-To header contains the conference URI (previously stored). Referred-By contains SIP or tel URI of UA A. (This is not mandatory) INVITE 3: Request URI contains the conference URI. The P-Asserted-Identity contains the URI of UA C. "isfocus" feature parameter indicated in Contact header field conference URI contained in the Contact header field. Referred-By contains SIP or tel URI of UA A. (This is not mandatory) NOTIFY 3: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body, contains SIP/2.0 100 Trying. NOTIFY 4: Event contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. Comments: SIP UA A Focus SIP UA B SIP UA C Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK Inviting UA B to the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 2 INVITE 2 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK (INVITE 2) 200 OK (INVITE 2) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) Inviting UA C to the conference REFER 2 REFER 2 202 Accepted 202 Accepted INVITE 3 INVITE 3 NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) 200 OK (INVITE 3) 200 OK (INVITE 3) ACK ACK NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 122 SSS__XXSSCONF_ INV_005 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.4, 5.3.1.5 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_INV. Configuration: CONF Selection criteria: Unsuccessful. User joining a conference by using a not valid conference URI. Test purpose: Ensure that, when User A sends a REFER request to the User B with request URI set to the SIP URI of the User B and Refer-To header set to the conference URI previously stored (the parameter "method" set to INVITE in the Refer-To header can be included or omitted): • User B receives a REFER request containing the Refer-To header set to the conference URI. • User B sends a 202 Accepted SIP response to the REFER request. • User A receives a 202 Accepted SIP response to the REFER request. • Remote user sends an INVITE request with request URI set to a not valid conference URI to the conference focus. • User B sends a NOTIFY request to the User A (on the same dialog of the REFER previously received) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • User B receives a 200 OK SIP response to the NOTIFY request. • User B receives a 488 Not Acceptable Here SIP response to the INVITE request from the conference focus. • User B sends an ACK to the conference focus. • Remote user sends a NOTIFY request to the User A (on the same dialog of the REFER previously received) with the Event header set to "refer" and the Content- Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 503 Service Unavailable. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains 503 Service Unavailable. • User A sends a 200 OK SIP response to the NOTIFY request. Precondition: • User A has created a conference by using a conference factory URI. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: REFER 1: Request URI contains the SIP URI of UA B Refer-To header contains the conference URI (previously stored). Referred-By contains SIP or tel URI of UA A. (This is not mandatory) INVITE 2: URI contained the conference URI not allocated in the conference focus. The P-Asserted-Identity contains the URI of UA B. Referred-By contains SIP or tel URI of UA A. (This is not mandatory) NOTIFY 1: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body, contains SIP/2.0 100 Trying. NOTIFY 2: Event contains refer; Subscription-State header contains terminated, Content- Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 503 Service Unavailable. Comments: SIP UA A Focus SIP UA B Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK Inviting UA B to the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 2 INVITE 2 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 488 Not Acceptable Here 488 Not Acceptable Here ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 123
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.7.3 Leaving a conference	SSS__XXSSCONF _LEAV_001 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.6 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_LEAV. Configuration: CONF Selection criteria: A participant leaves the conference. The conference event package is subscribed. Test purpose: Ensure that, when User B sends a BYE request (in order to leave the conference) to the conference focus with request URI set to the conference URI (previously stored): • User B sends a 200 OK SIP response to the BYE request. • User B receives a NOTIFY request (on the same dialog of the SUBSCRIBE previously sent) with the Event header set to "conference" and Subscription-State header set to "terminated". • User B sends a 200 OK SIP response to the NOTIFY request. Precondition: • User A has created a conference by using a conference factory URI. • User A has invited User B to the conference. • User B has joined the conference. • User B has subscribed to the conference event package. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: NOTIFY 4: Event header contains conference; Subscription-State header contains terminated, Content-Type header contains "application/conference-info+xml". Comments: SIP UA A Focus SIP UA B Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK Inviting UA B to the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 2 INVITE 2 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK (INVITE 2) 200 OK (INVITE 2) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) SUBSCRIBE SUBSCRIBE 200 OK (SUBSCRIBE) 200 OK (SUBSCRIBE) NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) Conference communication UA B leaves the conference BYE BYE 200 OK (BYE) 200 OK (BYE) NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 124 SSS__XXSSCONF _LEAV_002 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.6 TSS reference: S SIP-SIP-SIP/Supplementary_Services/CONF_LEAV. Configuration: CONF Selection criteria: A participant leaves the conference. The conference event package is not subscribed. Test purpose: Ensure that, when User B sends a BYE request (in order to leave the conference) to the conference focus with request URI set to the conference URI (previously stored): • User B sends a 200 OK SIP response to the BYE request. Precondition: • User A has created a conference by using a conference factory URI. • User A has invited User B to the conference. • User B has joined the conference. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; Comments: SIP UA A Focus SIP UA B Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK SIP UA B joining the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 2 INVITE 2 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK (INVITE 2) 200 OK (INVITE 2) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) Conference communication Participant leaves the conference BYE BYE 200 OK (BYE) 200 OK (BYE) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 125
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.7.4 Removing a conference participant from a conference	SSS__XXSSCONF_ REMOV_001 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.6 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_REMOV. Configuration: CONF Selection criteria: A participant removes another conference participant from the conference. The conference event package is subscribed. Test purpose: Ensure that, when User A sends a REFER request to the conference focus with request URI set to the conference URI (previously stored) and Refer-To header set to the SIP URI of User B (the parameter "method" must be set to BYE): • User A receives a 202 Accepted SIP response to the REFER request. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • User B receives a BYE request from the conference focus to be removed from the conference. • User B sends a 200 OK SIP response to the BYE request. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A sends a 200 OK SIP response to the NOTIFY request. • User A receives a NOTIFY request (on the same dialog of the SUBSCRIBE previously sent) with the Event header set to "conference". • User A sends a 200 OK SIP response to the NOTIFY request. Precondition: • User A has created a conference by using a conference factory URI. • User A has subscribed to the conference event package. • User A has invited User B to the conference. • User B has joined the conference. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: REFER 2: Request URI contains conference URI (previously stored). Refer-To header contains the URI of UA B; method=BYE. Referred-By header contains SIP URI of UA A. (This is not mandatory) NOTIFY 4: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 5: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. NOTIFY 6: Event header contains conference; Subscription-State header contains active, Content-Type header contains "application/conference-info+xml". ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 126 Comments: SIP UA A Focus SIP UA B Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK SUBSCRIBE 200 OK (SUBSCRIBE) NOTIFY NOTIFY 200 OK (NOTIFY) 200 OK (NOTIFY) UA B joining the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 2 INVITE 2 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK (INVITE 2) 200 OK (INVITE 2) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) Conference communication UA A removes UA B from the conference REFER 2 REFER 2 202 Accepted 202 Accepted NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4) BYE BYE 200 OK (BYE) 200 OK (BYE) NOTIFY 5 NOTIFY 5 200 OK (NOTIFY 5) 200 OK (NOTIFY 5) NOTIFY 6 NOTIFY 6 200 OK (NOTIFY 6) 200 OK (NOTIFY 6) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 127 SSS__XXSSCONF_ REMOV_002 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.6 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_REMOV. Configuration: CONF Selection criteria: A participant removes another conference participant from the conference. The conference event package is not subscribed. Test purpose: Ensure that, when User A sends a REFER request to the conference focus with request URI set to the conference URI (previously stored) and Refer-To header set to the SIP URI of User B (the parameter "method" must be set to BYE): • User A receives a 202 Accepted SIP response to the REFER request. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • User B receives a BYE request from the conference focus to be removed from the conference. • User B sends a 200 OK SIP response to the BYE request. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A sends a 200 OK SIP response to the NOTIFY request. Precondition: • User A has created a conference by using a conference factory URI. • User A has invited User B to the conference. • User B has joined the conference. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: REFER 2: Request URI contains conference URI (previously stored). Refer-To header contains the URI of UA B; method=BYE. Referred-By header contains SIP URI of UA A. (This is not mandatory) NOTIFY 3: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 4: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. Comments: SIP UA A Focus SIP UA B Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK UA B joining the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 2 INVITE 2 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK (INVITE 2) 200 OK (INVITE 2) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) Conference communication UA A removes UA B from the conference REFER 2 REFER 2 202 Accepted 202 Accepted NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) BYE 200 OK (BYE) NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 128 SSS__XXSSCONF_ REMOV_003 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.6 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_REMOV. Configuration: CONF Selection criteria: The conference owner releases the entire conference by sending a BYE to the focus. The conference event package is subscribed. Test purpose: Ensure that, when User A sends a BYE request to the conference focus with request URI set to the conference URI (previously stored): • User A receives a 200 OK SIP response to the BYE request. • User B receives a BYE request from the conference focus to be removed from the conference. • User B sends a 200 OK SIP response to the BYE request. • User A receives a NOTIFY request (on the same dialog of the SUBSCRIBE previously sent) with the Event header set to "conference". • User A sends a 200 OK SIP response to the NOTIFY request. Precondition: • User A has created a conference by using a conference factory URI. • User A has subscribed to the conference event package. • User A has invited User B to the conference. • User B has joined the conference. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: BYE: Request URI contains the conference URI (previously stored). NOTIFY 4: Event header contains conference; Subscription-State header contains active, Content-Type header contains "application/conference-info+xml". Comments: SIP UA A Focus SIP UA B Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK SUBSCRIBE SUBSCRIBE 200 OK (SUBSCRIBE) 200 OK (SUBSCRIBE) NOTIFY NOTIFY 200 OK (NOTIFY) 200 OK (NOTIFY) UA B joining the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 2 INVITE 2 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK (INVITE 2) 200 OK (INVITE 2) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) Conference communication UA A releases the entire conference BYE BYE 200 OK (BYE) 200 OK (BYE) focus removes UA B from the conference BYE 200 OK BYE NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 129 SSS__XXSSCONF_ REMOV_004 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.6 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_REMOV. Configuration: CONF Selection criteria: The conference owner releases the entire conference by sending a BYE to the focus. The conference event package is not subscribed. Test purpose: Ensure that, when User A sends a BYE request to the conference focus with request URI set to the conference URI (previously stored): • User A receives a 200 OK SIP response to the BYE request. • User B receives a BYE request from the conference focus to be removed from the conference. • User B sends a 200 OK SIP response to the BYE request. Precondition: • User A has created a conference by using a conference factory URI. • User A has invited User B to the conference. • User B has joined the conference. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: BYE: Request URI contains the conference URI (previously stored). Comments: SIP UA A Focus SIP UA B Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK UA B joining the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 2 INVITE 2 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK (INVITE 2) 200 OK (INVITE 2) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) Conference communication UA A releases the entire conference BYE BYE 200 OK (BYE) 200 OK (BYE) focus removes UA B from the conference BYE 200 OK (BYE) ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 130 SSS__XXSSCONF_ REMOV_005 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.6 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_REMOV. Configuration: CONF Selection criteria: The conference owner releases the entire conference by sending a REFER to the focus. The conference event package is subscribed. Test purpose: Ensure that, when User A sends a REFER request to the conference focus with request URI set to the conference URI (previously stored) and Refer-To header set to the conference URI (the parameter "method" must be set to BYE): • User A receives a 202 Accepted SIP response to the REFER request. • User A receives a BYE request from the conference focus to be removed from the conference. • User B receives a BYE request from the conference focus to be removed from the conference. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • User A sends a 200 OK SIP response to the BYE request. • User B sends a 200 OK SIP response to the BYE request. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A sends a 200 OK SIP response to the NOTIFY request. • User A receives a NOTIFY request (on the same dialog of the SUBSCRIBE previously sent) with the Event header set to "conference". • User A sends a 200 OK SIP response to the NOTIFY request. Precondition: • User A has created a conference by using a conference factory URI. • User A has subscribed to the conference event package. • User A has invited User B to the conference. • User B has joined the conference. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: REFER 2: Request URI contains the conference URI (previously stored). Refer-To header contains the conference URI; method=BYE. Referred-By header contains SIP URI of UA A. (This is not mandatory) NOTIFY 4: Event header contains refer; Subscription-State header contains active, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 5: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. NOTIFY 6: Event header contains conference; Subscription-State header contains terminated, Content-Type header contains "application/conference-info+xml". ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 131 Comments: SIP UA A Focus SIP UA B Conference creation INVITE INVITE 200 OK (INVITE) 200 OK INVITE) ACK ACK SUBSCRIBE SUBSCRIBE 200 OK (SUBSCRIBE) 200 OK (SUBSCRIBE) NOTIFY NOTIFY 200 OK (NOTIFY) 200 OK (NOTIFY) UA B joining the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 2 INVITE 2 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK (INVITE 2) 200 OK (INVITE 2) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) Conference communication UA A releases the entire conference REFER 2 REFER 2 202 Accepted 202 Accepted BYE BYE BYE BYE NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK NOTIFY 4 200 OK (BYE) 200 OK (BYE) 200 OK (BYE) 200 OK (BYE) NOTIFY 5 NOTIFY 5 200 OK (NOTIFY 5) 200 OK (NOTIFY 5) NOTIFY 6 NOTIFY 6 200 OK (NOTIFY 6) 200 OK (NOTIFY 6) SSS__XXSSCONF_ REMOV_006 CONF reference to: TS 124 147 [19], clauses 5.2.1, 5.3.1.6 TSS reference: SIP-SIP-SIP/Supplementary_Services/CONF_REMOV. Configuration: CONF Selection criteria: The conference owner releases the entire conference by sending a REFER to the focus. The conference event package is not subscribed. Test purpose: Ensure that, when User A sends a REFER request to the conference focus with request URI set to the conference URI (previously stored) and Refer-To header set to the conference URI (the parameter "method" must be set to BYE): • User A receives a 202 Accepted SIP response to the REFER request. • User A receives a BYE request from the conference focus to be removed from the conference. • User B receives a BYE request from the conference focus to be removed from the conference. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 100 Trying. • User A sends a 200 OK SIP response to the NOTIFY request. • User A sends a 200 OK SIP response to the BYE request. • User B sends a 200 OK SIP response to the BYE request. • User A receives a NOTIFY (on the same dialog of the REFER previously sent) with the Event header set to "refer" and the Content-Type header set to "message/sipfrag". The message/sipfrag body contains SIP/2.0 200 OK. • User A sends a 200 OK SIP response to the NOTIFY request. Precondition: • User A has created a conference by using a conference factory URI. • User A has invited User B to the conference. • User B has joined the conference. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 132 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; SIP header values: REFER 2: Request URI contains the conference URI (previously stored). Refer-To header contains the conference URI; method=BYE. Referred-By header contains SIP URI of UA A. (This is not mandatory) NOTIFY 3: Event header contains refer; Subscription-State header contains active, Content- Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 100 Trying. NOTIFY 4: Event header contains refer; Subscription-State header contains terminated, Content-Type header contains "message/sipfrag", message/sipfrag body contains SIP/2.0 200 OK. Comments: SIP UA A Focus SIP UA B Conference creation INVITE INVITE 200 OK (INVITE) 200 OK (INVITE) ACK ACK UA B joining the conference REFER 1 REFER 1 202 Accepted 202 Accepted INVITE 2 INVITE 2 NOTIFY 1 NOTIFY 1 200 OK (NOTIFY 1) 200 OK (NOTIFY 1) 200 OK (INVITE 2) 200 OK (INVITE 2) ACK ACK NOTIFY 2 NOTIFY 2 200 OK (NOTIFY 2) 200 OK (NOTIFY 2) Conference communication UA A releases the entire conference REFER 2 REFER 2 202 Accepted 202 Accepted BYE BYE BYE BYE NOTIFY 3 NOTIFY 3 200 OK (NOTIFY 3) 200 OK (NOTIFY 3) 200 OK (BYE) 200 OK (BYE) 200 OK (BYE) 200 OK (BYE) NOTIFY 4 NOTIFY 4 200 OK (NOTIFY 4) 200 OK (NOTIFY 4)
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.8 Test purposes for Call Waiting	SS___XXSSCW01 CW reference to: TS 124 615 [21], clause 4.5.5.2 TSS reference: SIP-SIP-SIP/Supplementary_Services/CW. Configuration: The user B has subscribed to CW Selection criteria: CW and approaching NDUB condition supported, NDUB status can be achieved for user B. Test purpose: Ensure that the SUT, when user A sends an INVITE towards user B which is in the approaching NDUB condition, delivers the INVITE to user B containing a Content-Type header set to application/vnd.3gpp.cw+xml and containing a MIME body including a "call-waiting-indication" element. SIP Parameter values: INVITE1 Dial string parameters options=PIXIT TYPE_SDP= PIXIT INVITE2 Content-Type header application/vnd.3gpp.cw+xml MIME body with "call-waiting-indication" element Comments: SIP UA A SUT SIP UA B UA B enters NDUB condition (e.g. by establishing a communication) INVITE1 INVITE2 ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 133 SS___XXSSCW02 CW reference to: TS 124 615 [21], clause 4.5.5.2 TSS reference: SIP-SIP-SIP/Supplementary_Services/CW. Configuration: The user B has subscribed to CW Selection criteria: CW and approaching NDUB condition supported, NDUB status can be achieved for user B. Test purpose: Ensure that the SUT, having delivered an INVITE indicating Call Waiting to user B which is in the approaching NDUB condition, on receipt of a 415 Unsupported Media Type from user B, sends a 486 Busy Here to user A. SIP Parameter values: INVITE1 Dial string parameters options=PIXIT TYPE_SDP= PIXIT INVITE2 Content-Type header application/vnd.3gpp.cw+xml MIME body with "call-waiting-indication" element Comments: SIP UA A SUT SIP UA B UA B enters NDUB condition (e.g. by establishing a communication) INVITE1 INVITE2 415 Unsupported Media Type 486 Busy Here ACK ACK SS___XXSSCW03 CW reference to: TS 124 615 [21], clause 4.5.5.2 TSS reference: SIP-SIP-SIP/Supplementary_Services/CW. Configuration: The user B has subscribed to CW Selection criteria: CW supported, Notification of calling user of CW status is supported. Test purpose: Ensure that the SUT, having delivered an INVITE from user A to user B, on receipt of a 180 Ringing containing an Alert-Info header set to "urn:alert:service:call-waiting", delivers this 180 Ringing to user A and provides an announcement about the CW condition. SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT 180 Ringing Alert-Info header set to "urn:alert:service:call-waiting" Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing Announcement to UE A ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 134 SS___XXSSCW04 CW reference to: TS 124 615 [21], clause 4.5.5.3 TSS reference: SIP-SIP-SIP/Supplementary_Services/CW. Configuration: The user B has subscribed to CW Selection criteria: CW and approaching NDUB condition supported, NDUB status can be achieved for user B. Test purpose: Ensure that the SUT, having delivered an INVITE from user B, which is in the approaching NDUB condition, to user A containing a Content-Type header set to application/vnd.3gpp.cw+xml, when user A leaves the NDUB condition and accepts the waiting call, handles the call with normal establishment procedures. Ensure that the voice/data transfer on the media channels is performed correctly (e.g. testing QoS parameters). SIP Parameter values: INVITE1 Dial string parameters options=PIXIT TYPE_SDP= PIXIT INVITE2 Content-Type header application/vnd.3gpp.cw+xml MIME body with "call-waiting-indication" element Comments: SIP UA A SUT SIP UA B UA B enters approaching NDUB condition (e.g. by establishing a communication) INVITE1 INVITE2 UA B leaves approaching NDUB condition (e.g. by releasing a communication) 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE SS___XXSSCW05 CW reference to: TS 124 615 [21], clause 4.5.5.3 TSS reference: SIP-SIP-SIP/Supplementary_Services/CW. Configuration: The user B has subscribed to CW Selection criteria: CW supported, Notification of calling user of CW status is supported. Test purpose: Ensure that the SUT, having delivered a 180 Ringing containing an Alert-Info header set to "urn:alert:service:call-waiting" from user B to user A, when user B accepts the call by sending a 200OK, handles the call with normal establishment procedures. Ensure that the voice/data transfer on the media channels is performed correctly (e.g. testing QoS parameters). SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT 180 Ringing Alert-Info header set to "urn:alert:service:call-waiting" Comments: SIP UA A SUT SIP UA B INVITE INVITE 180 Ringing 180 Ringing Announcement to UE A 200 OK INVITE 200 OK INVITE ACK ACK Check media BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 135 6.2.9 Test purposes for Completion of Communications to Busy Subscriber NOTE: The descriptions of invocation and operation of the CCBS service by the communication originating user are not yet fully described in TS 124 642 [22]. Therefore no test purposes have been defined for the current version of this document.
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.10 Test purposes for Completion of Communications by No Reply	NOTE: The descriptions of invocation and operation of the CCNR service by the communication originating user are not yet fully described in TS 124 642 [22]. Therefore no test purposes have been defined for the current version of this document.
3830b88c4707da898cf266974581f8bf	186 001-3	6.2.11 Test purposes for Explicit Communication Transfer	NOTE: In this clause the following conventions apply: user A: transferee, user B: transferor (served user), user C: transfer target. SSS__XXSSECT01 ECT reference to: TS 124 529 [23], clause 4.5.2 TSS reference: SIP-SIP-SIP/Supplementary_Services/ECTD. Configuration: The user B has subscribed to ECT Selection criteria: ECT supported. Test purpose: Blind/Assured transfer, served user B is callee in original communication Ensure that the SUT, when user B has established an original communication with user A and user B requests transfer of the communication towards user C by sending a REFER request to user A: • delivers the REFER request to user A containing the ECT Session Identifier URI and when user A responds with a 202 Accepted • delivers the 202 Accepted and a NOTIFY indicating 100 Trying to user B and when user A has held the original communication and sends a new INVITE to the ECT Session Identifier URI • delivers the INVITE to user C and continues normal call establishment between user A and user C and when user B receives a NOTIFY indicating 200 OK and user B sends a BYE to release the original communication. • delivers the BYE to user A and continues normal call release between user A and user B. Ensure that the voice/data transfer on the media channels of the transferred call (A-C) is performed correctly (e.g. testing QoS parameters). SIP Parameter values: REFER1 Request URI: contact URI of user A from original call Refer-To: public address of user C Referred-By: user B REFER2 Request URI: user A Refer-To: ECT Session Identifier Referred-By: user B NOTIFY1 body: 100 Trying INVITE1 Request URI: ECT Session Identifier INVITE2 Referred-By: user B NOTIFY2 body: 20OK ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 136 Comments: SIP UA A SUT SIP UA B SIP UA C Original communication is established from user A to user B REFER2 REFER1 202 Accepted 202 Accepted NOTIFY1 200 OK NOTIFY NOTE: TS 124 529 [23] 4.5.2.5 does not specify the order of events, holding of the original communication A-B could also take place before answering to the REFER request. Re-INVITE (sendonly) Re-INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE(recvonly) ACK ACK INVITE1 INVITE2 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK NOTIFY2 200 OK NOTIFY BYE BYE 200 OK BYE 200 OK BYE Check media (A-C) BYE BYE 200 OK BYE 200 OK BYE SSS__XXSSECT02 ECT reference to: TS 124 529 [23], clause 4.5.2 TSS reference: SIP-SIP-SIP/Supplementary_Services/ECTD. Configuration: The user B has subscribed to ECT Selection criteria: ECT supported. Test purpose: Blind/Assured transfer, served user B is caller in original communication Ensure that the SUT, when user B has established an original communication with user A and user B requests transfer of the communication towards user C by sending a REFER request to user A, • delivers the REFER request to user A containing the ECT Session Identifier URI and when user A responds with a 202 Accepted • delivers the 202 Accepted and a NOTIFY indicating 100 Trying to user B and when user A has held the original communication and sends a new INVITE to the ECT Session Identifier URI • delivers the INVITE to user C and continues normal call establishment between user A and user C and when user B receives a NOTIFY indicating 200 OK and user B sends a BYE to release the original communication • delivers the BYE to user A and continues normal call release between user A and user B. Ensure that the voice/data transfer on the media channels of the transferred call (A-C) is performed correctly (e.g. testing QoS parameters). ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 137 SIP Parameter values: REFER1 Request URI: contact URI of user A from original call Refer-To: public address of user C Referred-By: user B REFER2 Request URI: user A Refer-To: ECT Session Identifier Referred-By: user B NOTIFY1 body: 100 Trying INVITE1 Request URI: ECT Session Identifier INVITE2 Referred-By: user B NOTIFY2 body: 20OK Comments: SIP UA A SUT SIP UA B SIP UA C Original communication is established from user B to user A REFER2 REFER1 202 Accepted 202 Accepted NOTIFY1 200 OK NOTIFY NOTE: TS 124 529 [23], clause 4.5.2.5 does not specify the order of events, holding of the original communication A-B could also take place before answering to the REFER request. Re-INVITE (sendonly) Re-INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE(recvonly) ACK ACK INVITE1 INVITE2 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK NOTIFY2 200 OK NOTIFY BYE BYE 200 OK BYE 200 OK BYE Check media (A-C) BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 138 SSS__XXSSECT03 ECT reference to: TS 124 529 [23], clause 4.5.2 TSS reference: SIP-SIP-SIP/Supplementary_Services/ECTD. Configuration: The user B has subscribed to ECT Selection criteria: ECT supported. Test purpose: Consultative transfer, served user B is callee in original communication Ensure that the SUT, when user A has established an original communication with user B, user B has established a consultation communication with user C and user B requests transfer of the original communication towards user C by sending a REFER request to user A: • delivers the REFER request to user A containing the ECT Session Identifier URI and the call replacement data and when user A responds with a 202 Accepted • delivers the 202 Accepted and a NOTIFY indicating 100 Trying to user A and when user A has held the original communication and sends a new INVITE to the ECT Session Identifier URI • delivers the INVITE to user C and continues normal call establishment between user A and user C and when user C sends a BYE to release the consultation communication (B-C) • delivers the BYE to user B and continues normal call release between user C and user B and when user B receives a NOTIFY indicating 200 OK and user B sends a BYE to release the original communication (A-B) • delivers the BYE to user A and continues normal call release between user A and user B. Ensure that the voice/data transfer on the media channels of the transferred call (A-C) is performed correctly (e.g. testing QoS parameters). SIP Parameter values: REFER1 Request URI: contact URI of user A from original call Refer-To: public address of user C, using Replaces: from-tag and to-tag of communication B-C Referred-By: user B REFER2 Request URI: user A Refer-To: ECT Session Identifier Referred-By: user B NOTIFY1 body: 100 Trying INVITE1 Request URI: ECT Session Identifier INVITE2 Referred-By: user B NOTIFY2 body: 20OK ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 139 Comments: SIP UA A SUT SIP UA B SIP UA C Original communication is established from user A to user B Consultation communication is established from user B to user C REFER2 REFER1 202 Accepted 202 Accepted NOTIFY1 200 OK NOTIFY NOTE: TS 124 529 [23], clause 4.5.2.5 does not specify the order of events, holding of the original communication A-B could also take place before answering to the REFER request. Re-INVITE (sendonly) Re-INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE(recvonly) ACK ACK INVITE1 INVITE2 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE NOTIFY2 200 OK NOTIFY BYE BYE 200 OK BYE 200 OK BYE Check media (A-C) BYE BYE 200 OK BYE 200 OK BYE SSS__XXSSECT04 ECT reference to: TS 124 529 [23], clause 4.5.2 TSS reference: SIP-SIP-SIP/Supplementary_Services/ECTD. Configuration: The user B has subscribed to ECT Selection criteria: ECT supported. Test purpose: Consultative transfer, served user B is caller in original communication Ensure that the SUT, when user B has established an original communication with user A, user B has established a consultation communication with user C and user B requests transfer of the original communication towards user C by sending a REFER request to user A: • delivers the REFER request to user A containing the ECT Session Identifier URI and the call replacement data and when user A responds with a 202 Accepted • delivers the 202 Accepted and a NOTIFY indicating 100 Trying to user A and when user A has held the original communication and sends a new INVITE to the ECT Session Identifier URI • delivers the INVITE to user C and continues normal call establishment between user A and user C and when user C sends a BYE to release the consultation communication (B-C) • delivers the BYE to user B and continues normal call release between user C and user B and when user B receives a NOTIFY indicating 200 and user B sends a BYE to release the original communication (B-A) • delivers the BYE to user A and continues normal call release between user A and user B. Ensure that the voice/data transfer on the media channels of the transferred call (A-C) is performed correctly (e.g. testing QoS parameters). ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 140 SIP Parameter values: REFER1 Request URI: contact URI of user A from original call Refer-To: public address of user C, using Replaces: from-tag and to-tag of communication B-CReferred-By: user B REFER2 Request URI: user A Refer-To: ECT Session Identifier Referred-By: user B NOTIFY1 body: 100 Trying INVITE1 Request URI: ECT Session Identifier INVITE2 Referred-By: user B NOTIFY2 body: 20OK Comments: SIP UA A SUT SIP UA B SIP UA C Original communication is established from user B to user A Consultation communication is established from user B to user C REFER2 REFER1 202 Accepted 202 Accepted NOTIFY1 200 OK NOTIFY NOTE: TS 124 529 [23] 4.5.2.5 does not specify the order of events, holding of the original communication A-B could also take place before answering to the REFER request. Re-INVITE (sendonly) Re-INVITE (sendonly) 200 OK INVITE (recvonly) 200 OK INVITE(recvonly) ACK ACK INVITE1 INVITE2 180 Ringing 180 Ringing 200 OK INVITE 200 OK INVITE ACK ACK BYE BYE 200 OK BYE 200 OK BYE NOTIFY2 200 OK NOTIFY BYE BYE 200 OK BYE 200 OK BYE Check media (A-C) BYE BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 141 Annex A (informative): Bibliography • IETF RFC 2046: "Multipurpose Internet Mail Extensions (MIME) Part Two: Media Types". • IETF RFC 2806: "URLs for Telephone Calls". • IETF RFC 3262: "Reliability of Provisional Responses in the Session Initiation Protocol (SIP)". • IETF RFC 3264: "An Offer/Answer Model with the Session Description Protocol (SDP)". • IETF RFC 3323: "A Privacy Mechanism for the Session Initiation Protocol (SIP)". • IETF RFC 3325: "Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks". • IETF RFC 3326: "The Reason Header Field for the Session Initiation Protocol". • IETF RFC 3515: "The Session Initiation Protocol (SIP) Refer Method". • IETF RFC 3891: "The Session Initiation Protocol (SIP) Replaces Header". • IETF RFC 3892: "The Session Initiation Protocol (SIP) Referred-By Mechanism". • IETF RFC 4967 (2007): "Dial String Parameter for the Session Initiation Protocol Uniform Resource Identifier". • ETSI ES 283 027 (V2.5.1): "Telecommunications and Internet converged Services and Protocols for Advanced Networking (TISPAN); Endorsement of the SIP-ISUP Interworking between the IP Multimedia (IM) Core Network (CN) subsystem and Circuit Switched (CS) networks [3GPP TS 29.163 (Release 7), modified]". • ETSI TS 183 028 (V2.5.0): "Telecommunications and Internet converged Services and Protocols for Advanced Networking (TISPAN); Common Basic Communication procedures; Protocol specification". • ETSI TS 124 505 (V8.0.0): "Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); TISPAN; PSTN/ISDN simulation services: Conference (CONF); Protocol specification (3GPP TS 24.505 version 8.0.0 Release 8)". ETSI ETSI TS 186 001-3 V2.2.1 (2010-11) 142 History Document history V1.0.0 April 2008 Publication V2.1.1 October 2009 Publication V2.2.1 November 2010 Publication
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	1 Scope	The present document specifies the Abstract Test Suite (ATS) and partial Protocol Implementation eXtra Information for Testing (PIXIT) proforma based on the Test Suite Structure and Test Purposes defined in [1]. The TSS&TP have been developed Network Integration Testing between SIP and ISDN/PSTN network signalling protocols. The ATS is sometimes referred to in the present document as "SIP-ISDN-Interworking ATS". The test notation used in the ATS is TTCN-3 ([3]). 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; • the design principles, assumptions, and used interfaces to the TTCN3 tester (System Simulator); • 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 (IXIT) Proforma of the ATS. Annex B provides the Testing and Test Control Notation (TTCN-3) part of the ATS.
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	2 References	References are either specific (identified by date of publication and/or edition number or version number) or non-specific. • For a specific reference, subsequent revisions do not apply. • Non-specific reference may be made only to a complete document or a part thereof and only in the following cases: - if it is accepted that it will be possible to use all future changes of the referenced document for the purposes of the referring document; - for informative references. 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. ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 6
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	2.1 Normative references	The following referenced documents are indispensable for the application of the present document. For dated references, only the edition cited applies. For non-specific references, the latest edition of the referenced document (including any amendments) applies. [1] ETSI TS 186 001 (Parts 1 and 3): "Telecommunications and Internet Converged Services and Protocols for Advanced Networking (TISPAN); Network Integration Testing between SIP and ISDN/PSTN network signalling protocols". [2] ETSI TS 102 351 (V2.1.1): "Methods for Testing and Specification (MTS); Internet Protocol Testing (IPT); IPv6 Testing: Methodology and Framework". [3] ETSI ES 201 873-1 (V3.1.1): "Methods for Testing and Specification (MTS); The Testing and Test Control Notation version 3; Part 1: TTCN-3 Core Language". [4] ETSI ES 201 873-5 (V3.1.1): "Methods for Testing and Specification (MTS); The Testing and Test Control Notation version 3; Part 5: TTCN-3 Runtime Interface (TRI)". [5] ETSI ES 201 873-6 (V3.1.1): "Methods for Testing and Specification (MTS); The Testing and Test Control Notation version 3; Part 6: TTCN-3 Control Interface (TCI)". [6] ISO/IEC 9646-1 (1994): "Information technology - Open Systems Interconnection - Conformance testing methodology and framework - Part 1: General concepts". [7] ISO/IEC 9646-7 (1995): "Information technology - Open Systems Interconnection - Conformance testing methodology and framework - Part 7: Implementation Conformance Statements". [8] ITU-T Recommendation Q.931 (1998): "ISDN user-network interface layer 3 specification for basic call control". [9] ETSI TS 102 027-3 (V3.1.1): "Methods for Testing and Specification (MTS); Conformance Test Specification for SIP (IETF RFC 3261); Part 3: Abstract Test Suite (ATS) and partial Protocol Implementation eXtra Information for Testing (PIXIT) proforma". [10] IETF RFC 3261 (2002): "SIP: Session Initiation Protocol". [11] ETSI EN 383 001: "Telecommunications and Internet converged Services and Protocols for Advanced Networking (TISPAN); Interworking between Session Initiation Protocol (SIP) and Bearer Independent Call Control (BICC) Protocol or ISDN User Part (ISUP) [ITU-T Recommendation Q.1912.5, modified]". [12] ITU-T Recommendations Q.761 to Q.764 (1999): "Signalling System No.7 - ISDN User Part (ISUP)". [13] ITU-T Recommendation E.164 (2005): "The international public telecommunication numbering plan". [14] IETF RFC 4575: "A Session Initiation Protocol (SIP) Event Package for Conference State".
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	2.2 Informative references	The following referenced documents are not essential to the use of the present document but they assist the user with regard to a particular subject area. For non-specific references, the latest version of the referenced document (including any amendments) applies. Not applicable. ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 7
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	3 Definitions and abbreviations
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	3.1 Definitions	For the purposes of the present document, the terms and definitions given in SIP/ISUP interworking reference specification [11], ISDN layer 3 reference specification [8], ISDN User Part (ISUP) reference specification [12], [6], [7], [3] (TTCN-3) and the following apply: Abstract Test Case (ATC): complete and independent specification of the actions required to achieve a specific test purpose, defined at the level of abstraction of a particular Abstract Test Method, starting in a stable testing state and ending in a stable testing state Abstract Test Method (ATM): description of how an IUT is to be tested, given at an appropriate level of abstraction to make the description independent of any particular realization of a Means of Testing, but with enough detail to enable abstract test cases to be specified for this method Abstract Test Suite (ATS): test suite composed of abstract test cases Implementation Under Test (IUT): implementation of one or more OSI protocols in an adjacent user/provider relationship, being part of a real open system which is to be studied by testing Means of Testing (MOT): combination of equipment and procedures that can perform the derivation, selection, parameterization and execution of test cases, in conformance with a reference standardized ATS, and can produce a conformance log PICS proforma: document, in the form of a questionnaire, which when completed for an implementation or system becomes the PICS PIXIT proforma: document, in the form of a questionnaire, which when completed for the IUT becomes the PIXIT Point of Control and Observation (PCO): point within a testing environment where the occurrence of test events is to be controlled and observed, as defined in an Abstract Test Method pre-test condition: setting or state in the IUT which cannot be achieved by providing stimulus from the test environment Protocol Implementation Conformance Statement (PICS): statement made by the supplier of a protocol claimed to conform to a given specification, stating which capabilities have been implemented Protocol Implementation eXtra Information for Testing (PIXIT): statement made by a supplier or implementor of an IUT (protocol) which contains or references all of the information related to the IUT and its testing environment, which will enable the test laboratory to run an appropriate test suite against the IUT SIP number: number conforming to the numbering and structure specified in ITU-T Recommendation E.164 [13] System Under Test (SUT): real open system in which the IUT resides
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	3.2 Abbreviations	For the purposes of the present document, the following abbreviations apply: ASP Abstract Service Primitive NOTE: Exchanged between entities inside the TS or between the user of the ATS (operator) and the TS. ATC Abstract Test Case ATM Abstract Test Method ATS Abstract Test Suite DSS1 Digital Subscriber System No. 1 ETS Executable Test Suite IETF Internet Engineering Task Force ISDN Integrated Services Digital Network ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 8 IUT Implementation Under Test IWU InterWorking Unit LAN Local Area Network MOT Means Of Testing MTC Main Test Component NGN Next Generation Network PA Plateform Adapter PA Platform Adapter PCO Point of Control and Observation PDU Protocol Data Unit NOTE: Message exchanged between TS and SUT at a signalling interface. PICS Protocol Implementation Conformance Statement PIXIT Protocol Implementation eXtra Information for Testing PTC Parallel Test Component SA SUT Adapter SDP Session Description Protocol SIP Session Initiation Protocol SUT System Under Test TC Test Case TCI TTCN-3 Control Interface TCP Test Coordination Procedures TD Test Description TE Test Equipment TL Test Logging TP Test Purpose TS Test System TSS Test Suite Structure TSS&TP Test Suite Structure and Test Purposes TTCN Tree and Tabular Combined Notation TTCN-3 Testing and Test Control Notation edition 3 UDP Unreliable Datagram Protocol
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	4 Abstract Test Method (ATM)
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	4.1 Network architecture	Figures 1 and 2 show the network architecture for SIP-ISDN InterWorking Units (IWU). Figure 1 shows the network architecture for SIP-ISDN Interworking. SIP ISDN <-----------------------------------------> IWU <------------------------------------------> Figure 1: Interworking between SIP and ISDN
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	4.2 Protocol architecture	Figure 1 shows that there are 2 interfaces of the IWU (representing the SUT in the testing environment described in the present document): a SIP interface and an ISDN interface. ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 9 Figure 2 shows the protocol architecture: SUT IUT IUT SIP Q.931 UDP TCP IP Q.921 (LAN) I.431 Figure 2: Protocol architecture of the SIP-ISDN-Interworking ATS
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	4.3 Test architecture
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	4.3.1 Interconnection of TS and SUT	Figure 3 shows the interconnection of TS and SUT in terms of signalling message flows. SIP SUT ISDN <------------------------> <------------------------> IWU Test System Figure 3: Interconnection of TS and SUT
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	4.3.2 Test System architecture	An abstract architecture for a Test System (TS) implementing a TTCN-3 ATS is displayed in figure 4 and also stated in [4]. ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 10 Test Management (TM) Test Control (TC) Test Logging (TL) TCI TTCN-3 Executable (TE) TTCN-3 Runtime System (T3RTS) Executable Test Suite (ETS) Encoding/Decoding System TRI SUT Adapter (SA) Platform Adapter (PA) Figure 4: Abstract Test System Architecture A TS has two interfaces, the TTCN-3 Control Interface (TCI) and the TTCN-3 Runtime Interface (TRI), which specify the interface between Test Management (TM) and TTCN-3 Executable (TE) entities, and TE, SUT Adapter (SA) and Platform Adapter (PA) entities, respectively. Out of these two interfaces the TRI has been standardized in [4], whereas the specification and implementation of the TCI is in [5]. The part of TS that deals with interpretation and execution of TTCN-3 modules, i.e. the Executable Test Suite (ETS), is shown as part of the TTCN-3 Executable (TE). This ETS corresponds either to the executable code produced by a TTCN-3 compiler or a TTCN-3 interpreter from the TTCN-3 ATS in a TS implementation. The remaining part of the TS, which deals with any aspects that cannot be concluded from information being present in the TTCN-3 ATS alone, can be decomposed into Test Management (TM), SUT Adapter (SA), and Platform Adapter (PA) entities. In general, these entities cover a TS user interface, test execution control, test event logging, communication of test data with the SUT, and timer implementation. The part of SA used for SIP message transfer shall implement the TRI adaptation as well as the SIP transport protocol architecture described in clause 4.2. The Encoding/Decoding System (EDS) entity, as far as applied to SIP messages, with the TE and Test Logging (TL) entity within the TM shall comply with the conventions defined in clause 4.3.2 of [9]. The part of SA used for ISDN message transfer shall implement the TRI adaptation as well as the ISDN transport protocol architecture described in clause 4.2. The Encoding/Decoding System (EDS) entity, as far as applied to ISDN messages, shall comply with the conventions and requirements defined in the following clauses.
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	5 The ATS development process
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	5.1 Requirements and Test Purposes	For each test purpose there is a table defined in clause 6 of [1]. The requirements applicable to this TP are given by a reference to the relevant base specification. There are no explicit formulations of requirements. ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 11
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	5.2 ATS structure
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	5.2.1 Test case grouping	The ATS structure defined in table 1 is based on the structuring of Test Purposes in clause 6 of [1]. The group names in columns 1 to 3 of table 1 are those assigned in the ATS; they are based on the names provided in clause 6 of [1], but use the naming conventions defined for the ATS (see clause 5.3.2.2). Table 1: ATS structure Group Subgroup Sub-Subgroup Group Index ISDN-SIP 1 Basic call 11 Successful - Speech 1101 Codec negotiation 1102 Successful - UPDATE 1103 Successful - DTMF Tests 1104 Successful - UDI 1105 Unsuccessful 1106 Sup. Services 12 CLIP 1201 CLIR 1202 COLP/COLR 1203 CFU 1204 CFB 1205 CFNR 1206 CFNL 1207 CD 1208 HOLD 1209 3PTY 1210 CONF 1211 SIP-ISDN 2 Basic call 21 Successful 3,1 kHz audio 2101 Codec negotiation 2102 DTMF 2103 UDI 2104 Unsuccessful 2105 Sup. Services 22 CLIP/OIP 2201 OIR/CLIR 2202 TIP/COLP 2203 TIR/COLR 2204 CFU 2205 CFB 2206 CFNR 2207 CD 2208 TP 2209 3PTY 2210 HOLD 2211 CONF 2212 CW 2213 ACR 2214 CUG 2215 ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 12 Group Subgroup Sub-Subgroup Group Index SIP-SIP 5 Basic call 51 Successful 5101 Codec negotiation 5102 Update 5103 Unsuccessful 5104 Sup. Services 53 OIP 5201 OIR 5202 TIP 5203 TIR 5204 HOLD 5205 CFU 5206 CFB 5207 CFNR 5208 CFNL 5209 CD 5210 CONF 5211
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	5.2.2 Test case identifiers	The test case names are built up according to the following scheme: <"TC"><Group path index>"_"<TC number> where: a) double quotes (") are used to enclose literal strings; b) <Group path index> is the 4-digit number which uniquely identifies the path of groups/subgroups; c) <TC number> is the identifier from the TSS&TP document. EXAMPLE: TC1101_IS___XX__001: i) the identifier has Group index "1101", i.e. it is in the subgroup having complete path: SIP-ISDN/BasicCall/Successful - Speech/; ii) the identifier is the first test case of this group/subgroup. NOTE: This naming scheme provides a 1-1 correspondence of TP identifiers as defined in [1] and test case names.
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	5.3 ATS specification framework
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	5.3.1 ATS Library	For this interworking ATS there are 2 applicable base protocols: a) SIP protocol ([10]); and b) ISDN protocol (ITU-T Recommendation Q.931 [8] series, plus associated standards for supplementary services etc.). Since e.g. the data structures of these 2 base protocols are independent, and other objects like test cases are common, the TTCN-3 library modules are basically organized as: 1) SIP modules; 2) ISDN modules; 3) Common modules (generated for the present ATS); ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 13 4) LibCommon modules (taken from [2]). Table 2 shows the organization of the ATS as library of modules. Table 2: Library of modules Module Class Module Id Description LibCommon LibCommon_AbstractData Generic data types for a stack and its operations. LibCommon_BasicTypesAndValues Basic type and value definitions (integer and Boolean). LibCommon_DataStrings Bit and Octet string types. LibCommon_Sync Co-ordination/synchronization of test components. LibCommon_TextStrings Basic character and string types with fixed length. LibCommon_Time Time handling functions and moduleparameter. LibCommon_VerdictControl Basic functions for setting of test component verdicts. AtsCommon General_Types Definitions are based on component type definitions from IPv6, SCOP and common synchronization libraries. SipIsdn_PICS Module Parameter declarations associated with PICS. SipIsdn_PIXITS SIP-ISDN common Module Parameter declarations associated with PIXIT. SipIsdn_Testcases Test case functions. SipIsdn_TestConfiguration Functions which implement the configuration of the SUT adapter and mapping of test components for establishing and tearing down different test configurations. SipIsdn_TestExecution Module control: execute test cases depending on selection conditions; repeat parameterized test cases based on the "Variant-tables" defined in the test prose. SipIsdn_TestSystem Common functions, components, ASPs controlling the test system. SipAts SipIsdn_SIP_SDPTypes SIP SDP data types. SipIsdn_SIP_TCFunctions PTC root functions for SIP-to-ISDN test cases. SipIsdn_SIPSIP_TCFunctions PTC root functions for SIP-to-SIP test cases. SipIsdn_SIP_Types SIP data types (messages, header fields) and parallel test component (according to [9]). SipIsdn_SIP_Templates Templates for SIP messages and header fields (according to [10]). SipIsdn_SIP_Steps SIP auxiliary functions. SipIsdn_SIP_XMLTypes SIP data types defined in XML (message body according to RFC 4575 [14]). XSDAUX Generic XML type system for TTCN-3. IsdnAts SipIsdn_ISDN_Types ISDN data types (information element and message types according to ITU-T Recommendation Q.931 [8] and ASP type declarations). SipIsdn_ISDN_Templates Templates for ISDN information elements, messages and ASPs. SipIsdn_ISDN_Steps Test step declarations, including preambles, postambles and default. SipIsdn_ISDN_TCFunctions Test case functions running on the Isdn component. SipIsdnASN1Types ASN.1 definitions for ISDN message parts (ASN.1 and TTCN-3 notation).
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	5.3.2 Use of TTCN-3
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	5.3.2.1 General	TTCN-3 as defined in [3] is used as ATS specification language. A number of requirements have been identified for the development and production of the TTCN-3 specification for the SIP/ISUP Interworking ATS: 1) Top-down design. 2) A uniquely defined testing architecture and test method. 3) Uniform TTCN-3 style and naming conventions. 4) TTCN-3 is human-readability. ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 14 5) TTCN-3 specification is feasible, implementable, compilable and maintainable. 6) Test cases shall be designed in a way to be easily adaptable, upwards compatible with the evolution of the base protocol and protocol interworking of future releases. 7) The test declarations, data structures and data values shall be largely reusable. 8) Modularity and modular working method. 9) Minimizing the requirements of intelligence on the emulators of the lower testers. 10) Giving enough design freedom to the test equipment manufacturers. Fulfilling these requirements should ensure the investment of the test equipment manufacturers and users of the ATS having stable testing means for a relatively long period.
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	5.3.2.2 TTCN-3 naming conventions	Like in other software projects using a programming language, the use of naming conventions supports or increases: a) the readability; b) the detection of semantic errors; c) the shared work of several developers; d) the maintainability. The naming conventions applied to the SIP/ISUP Interworking ATS are based on the following underlying principles: • when constructing meaningful identifiers, the general guidelines specified for naming in clause 9 of [2] should be followed; • for the SIP ATS part, which is based on a subset of [9], with extensions, the naming conventions defined in [9] should be followed; • the names of TTCN-3 objects being associated with standardized data types (e.g. in the base protocols) should reflect the names of these data types as close as possible (of course not conflicting with syntactical requirements or other conventions being explicitly stated); • the subfield names of TTCN-3 objects being associated with standardized data type should also be similar to corresponding element names in the base standards (be recognizable in the local context); • in most other cases, identifiers should be prefixed with a short alphabetic string (specified in table 3) indicating the type of TTCN-3 element it represents; • prefixes should be separated from the body of the identifier with an underscore ("_"); • only test case names, module names, data type names and module parameters should begin with an upper-case letter. All other names (i.e. the part of the identifier following the prefix) should begin with a lower-case letter. Table 3 specifies the naming guidelines for each element of the TTCN-3 language indicating the recommended prefix and capitalization. ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 15 Table 3: TTCN-3 naming conventions Language element Naming convention Prefix Example Notes Module Use upper-case initial letter none SipIsdn_ISDN_Types TSS grouping Use all upper-case letters as specified in clause 5.2.2 none TP_RT_PS_TR Item group within a module Use lower-case initial letter none messageGroup ISDN message type Use prefix PDU_DSS1 and upper-case initial letter and message name abbreviations none PDU_DSS1_Setup ISDN parameter type Use prefix DSS1 and upper- case initial letter and parameter name none DSS1_CallReference SIP message type Use upper-case initial letter none Request, Response note 4 SIP header type Use upper-case initial letter none MaxForwards note 4 Basic common data types (e.g. bit string types of fixed length) Use upper-case initial letter none Take from common module Other Data types Use upper-case initial letter none SetupContents Template None m_ m_IAM_Basic note 1 Message template with wildcard or matching expression None mw_ mw_AnyUserReply note 2 Signature template Use lower-case initial letter s_ s_callSignature Port instance Use lower-case initial letter none signallingPort Test component ref Use lower-case initial letter none userTerminal Constant Use lower-case initial letter c_ c_maxRetransmission External constant Use lower-case initial letter cx_ cx_macId Function Use lower-case initial letter f_ f_authentication() External function Use lower-case initial letter fx_ fx_calculateLength() Altstep (incl. Default) Use lower-case initial letter a_ a_receiveSetup() Test case Use naming as specified in clause 5.2.2 TC_ TC_101_001 Variable (local) Use lower-case initial letter v_ v_macId Variable (defined within a component) Use lower-case initial letters vc_ vc_systemName Timer (local) Use lower-case initial letter t_ t_wait Timer (defined within a component) Use lower-case initial letters tc_ tc_authMin Module parameter Use initial upper case letters PX PX_MAC_ID note 3 Parameterization Use lower-case initial letter p_ p_macId Enumerated Value Use lower-case initial letter e_ e_syncOk NOTE 1: This prefix must be used for all template definitions which do not assign or refer to templates with wildcards or matching expressions, e.g. templates specifying a constant value, parameterized templates without matching expressions, etc. NOTE 2: This prefix must be used in identifiers for templates which either assign a wildcard or matching expression (e.g. ?, *, value list, if present, pattern, etc.) or reference another template which assigns a wildcard or matching expression. NOTE 3: In this case it is acceptable to use underscore as a word delimiter. NOTE 4: This convention has been used in [9] (SIP ATS).
1ac6a1947ee8d396f7041a17f85f4694	186 001-2	5.4 ATS archive	Annex B contains the ATS archive (.zip file expanding to text files with TTCN-3 code). ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 16 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 According to the interworking type of ATS defined in the present document, the PIXIT are divided in SIP-related PIXIT and ISDN-related PIXIT. A.2.1 SIP-related PIXIT The SIP-related PIXIT of table A.1 apply, which have been provided for the particular purposes of this ATS. Each PIXIT item corresponds to a Module Parameter of the ATS. Table A.1: SIP-related PIXIT items Item Module Parameter Description Type Value SDP Parameter 1.1 PX_SIP_SDP_dyn SDP dynamic port. integer 1.2 PX_SIP_SDP_b_modifier SDP bandwidth modifier. charstring 1.3 PX_SIP_SDP_b_bandwidth SDP bandwidth value. integer 1.4 PX_SIP_SDP_encoding SDP media attribute encoding supported by the IUT. charstring 1.5 PX_SIP_SDP_encoding_unavail SDP media attribute encoding unavailable at the IUT. charstring 1.6 PX_SIP_SDP_encoding_unsup SDP media attribute encoding unsupported by the IUT. charstring 1.7 PX_SIP_SDP_transport SDP media T.38 transport (used in TC2101_IS___AU__09) charstring Supported options 1.8 PX_SIP_100rel True if 100rel mechanism is supported in SIP. boolean 1.9 PX_SIP_precondition True if precondition mechanism is supported in SIP. boolean 1.10 PX_SIP_UDP True if UDP Transport is used by the IUT to run campaign. boolean 1.11 PX_SIP_TRANSPORT Used Transport in upper case, i.e. "UDP" or "TCP". charstring Ports and addresses of the IUT 1.12 PX_SIP_IUT_PORT IUT port number to exchange SIP messages. integer 1.13 PX_SIP_IUT_IPADDR IUT IP address to exchange SIP messages. charstring 1.14 PX_SIP_IUT_HOME_DOMAIN IUT domain. charstring ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 17 Item Module Parameter Description Type Value 1.15 PX_SIP_IUT_HOME_DOMAIN_UNKNO WN Unknown IUT domain. charstring 1.16 PX_SIP_IUT_USER User identity at ISDN side. charstring 1.17 PX_SIP_IUT_USER_UNKNOWN Unknown user identity at ISDN side. charstring Ports and addresses of the ETS (tester), first access, used for SIP-ISDN and SIP-SIP testing 1.18 PX_SIP_ETS_PORT Port number used by the ETS to exchange SIP messages. integer 1.19 PX_SIP_ETS_IPADDR IP address used by the ETS to exchange SIP messages. charstring 1.20 PX_ETS_LOCAL_DOMAIN Identity of the tester local domain. charstring 1.21 PX_ETS_LOCAL_USER Identity of the tester local user. charstring 1.22 PX_ETS_LOCAL_USER_DIV Identity of the user with active call diversion service. charstring 1.23 PX_SIP_ETS_LOCAL_USER_FULL Identity of the tester local user (format "+"cc+ndc+sn). charstring 1.24 PX_SIP_ETS_BEARER_PORT Port number used by the ETS to exchange media streams. integer 1.25 PX_SIP_ETS_BEARER_PORT2 Second Port number used by the ETS to exchange media streams. integer 1.26 PX_SIP_ETS_BEARER_IPADDR IP address used by the ETS to exchange media streams. charstring Ports and addresses of the ETS2 (tester), second access, used for SIP-SIP testing only 1.27 PX_SIP_ETS2_PORT Port number used by the ETS2 to exchange SIP messages. integer 1.28 PX_SIP_ETS2_IPADDR IP address used by the ETS2 to exchange SIP messages. charstring 1.29 PX_ETS2_LOCAL_DOMAIN Identity of the tester local domain. charstring 1.30 PX_ETS2_LOCAL_USER Identity of the tester local user. charstring 1.31 PX_ETS2_LOCAL_USER_DIV Identity of the user with active call diversion service. charstring 1.32 PX_SIP_ETS2_LOCAL_USER_FULL Identity of the tester local user (format "+"cc+ndc+sn). charstring 1.33 PX_SIP_ETS2_BEARER_PORT Port number used by the ETS2 to exchange media streams. integer 1.34 PX_SIP_ETS2_BEARER_PORT2 Second Port number used by the ETS2 to exchange media streams. integer 1.35 PX_SIP_ETS2_BEARER_IPADDR IP address used by the ETS2 to exchange media streams. charstring Ports and addresses of the ETS3 (tester), second access, used for SIP-SIP (CONF) testing only 1.36 PX_SIP_ETS3_PORT Port number used by the ETS2 to exchange SIP messages. integer 1.37 PX_SIP_ETS3_IPADDR IP address used by the ETS2 to exchange SIP messages. charstring 1.38 PX_ETS3_LOCAL_DOMAIN Identity of the tester local domain. charstring PX_ETS3_LOCAL_USER Identity of the tester local user. charstring Registration parameters 1.39 PX_SIP_REGISTRATION Does the SIP user have to register itself before executing a test case? boolean 1.40 PX_SIP_REGISTRAR_PORT Registrar port number to exchange SIP messages. integer 1.41 PX_SIP_REGISTRAR_DOMAIN Registrar domain. charstring Release cause 1.42 PX_SIP_BYE_CAUSE Release cause to be used in BYE and in Failure messages. integer RTP stream control and check 1.43 PX_SIP_CheckConversation True, if conversation check is implemented. boolean 1.44 PX_SIP_CheckDTMF True, if DTMF check is implemented. boolean 1.45 PX_SIP_SendAnnouncement True, if Announcement sending is implemented. boolean 1.46 PX_SIP_CheckRinging True, if ringing check is implemented. boolean ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 18 Item Module Parameter Description Type Value Parameters for HTTP authentication 1.47 PX_SIP_REGISTRATION_AUTHENTIC ATION_ENABLED Option controlling if authentication is enabled/disabled for registration messages. boolean 1.48 PX_SIP_RFC2617_QOP Quoted string of one or more tokens indicating the "quality of protection" values supported by the server. The value "auth" indicates authentication; the value "auth-int" indicates authentication with integrity protection. charstring 1.49 PX_SIP_RFC2617_USERNAME The name of user in the specified realm. charstring 1.50 PX_SIP_RFC2617_PASSWD A known shared secret, the password of user of the specified username. charstring 1.51 PX_SIP_RFC2617_URI URI for HTTP authentication. charstring 1.52 PX_SIP_RFC2617_USERNAME_T The name of terminating user in the specified realm. charstring 1.53 PX_SIP_RFC2617_PASSWD_T A known shared secret, the password of terminating user of the specified username. charstring 1.54 PX_SIP_RFC2617_URI_T URI for HTTP authentication (terminating user). charstring 1.55 PX_SIP_RFC2617_USERNAME_T3 The name of 3rd user in the specified realm. charstring 1.56 PX_SIP_RFC2617_PASSWD_T3 A known shared secret, the password of 3rd user of the specified username. charstring 1.57 PX_SIP_RFC2617_URI_T3 URI for HTTP authentication (3rd user). charstring SIP Timers 1.58 PX_SIP_T1 T1 RTT estimate. float 1.59 PX_T2 T2 Maximum retransmit interval for non-INVITE requests and INVITE response. float 1.60 PX_T4 4 Maximum duration a message will remain in the network. float 1.61 PX_SIP_TWAIT TWait default value for waiting an operator action. float 1.62 PX_SIP_TACK TAck default value for waiting an acknowledgement. float 1.63 PX_SIP_TRESP TResp default value for waiting for a response from the IUT. float 1.64 PX_SIP_TNOACT TNoAct default value for waiting no message from the IUT. float 1.65 PX_SIP_TSYN TSYNC default value to synchronize ptc. float 1.66 PX_SIP_TGUARD TGUARD default value for an extra long timer to limit test execution. float 1.67 PX_TRespRetention TRespRetention minimum time that a Proxy will wait before sending a final response. float Test Case variant management 1.68 PX_TC_VA Testcase variant according to table entry in table to test purpose description, if present. integer 1.69 PX_TC_HistoryInfoUsage testcase variant on the use of the HistoryInfo-header field. boolean 1.70 PX_TC_VA_NO180 testcase variant that do not use of 180 response for user-C (SSS-tests). boolean 1.71 PX_TC_VA_NO181 testcase variant that do not expect of 181 response for SUT (SSS-tests). boolean 1.72 PX_VA_IS___XX_U14 Value to choose SIP message from table below IS___XX_U14. 1 = 415 Unsupported Media type 2 = 420 Bad Extension 3 = 421 Extension Required. integer ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 19 Item Module Parameter Description Type Value 1.73 PX_VA_IS___XX_U14_CV_SIP Cause value for SIP message sent in IS___XX_U14. integer 1.74 PX_VA_IS___XX_U14_CV_ISDN Cause value for ISDN RELEASE message received in IS___XX_U14. integer 1.75 PX_VA_IS___XX_U15 Value to choose SIP message from table below IS___XX_U15. 1 = 415 Unsupported Media type 2 = 420 Bad Extension 3 = 421 Extension Required. integer 1.76 PX_VA_SIP_IS___XXSSCOLP01_MSG Value to choose SIP message from table below IS___XXSSCOLP01. 1 = 180 Ringing 2 = 183 Session Progress. integer 1.77 PX_VA_SIP_IS___XXSSCOLP02_MSG Value to choose SIP message from table below IS___XXSSCOLP02. 1 = 180 Ringing 2 = 183 Session Progress. integer 1.78 PX_VA_SIP_IS___XXSSCOLP03_MSG Value to choose SIP message from table below IS___XXSSCOLP03. 1 = 180 Ringing 2 = 183 Session Progress. integer 1.78 PX_VA_SIP_IS___XXSSCOLP04_MSG Value to choose SIP message from table below IS___XXSSCOLP04. 1 = 180 Ringing 2 = 183 Session Progress. integer 1.79 PX_VA_SIP_IS___XXSSCOLP07_PRIV Value to choose SIP Privacy header to be sent from table below IS___XXSSCOLP07. 1 = Id 2 = User 3 = Header. integer 1.80 PX_VA_SIP_IS___XXSSCOLP08_MSG Value to choose SIP message from table below IS___XXSSCOLP08. 1 = 180 Ringing 2 = 183 Session Progress. integer 1.81 PX_VA_SIP_IS___XXSSCOLP09_MSG Value to choose SIP message from table below IS___XXSSCOLP09. 1 = 180 Ringing 2 = 183 Session Progress. integer 1.82 PX_VA_SIP_IS___XXSSCOLP10_MSG Value to choose SIP message from table below IS___XXSSCOLP10. 1 = 180 Ringing 2 = 183 Session Progress. integer 1.83 PX_VA_SIP_IS___XXSSCOLP11_MSG Value to choose SIP message from table below IS___XXSSCOLP11. 1 = 180 Ringing 2 = 183 Session Progress. integer 1.84 PX_VA_SIP_IS___XXSSCOLP14_PRIV Value to choose SIP Privacy header to be sent from table below IS___XXSSCOLP14. 1 = Id 2 = User 3 = Header. integer ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 20 Item Module Parameter Description Type Value 1.85 PX_VA_SIP_IS___XXSSCOLP16_PRIV Value to choose SIP Privacy header to be sent from table below IS___XXSSCOLP16. 1 = Id 2 = User 3 = Header. integer 1.86 PX_VA_SIP_IS___XXSSCOLP17_MSG Value to choose SIP message from table below IS___XXSSCOLP17. 1 = 180 Ringing 2 = 183 Session Progress. integer 1.87 PX_VA_SIP_IS___XXSSCOLP19_MSG Value to choose SIP message from table below IS___XXSSCOLP19. 1 = 180 Ringing 2 = 183 Session Progress. integer 1.88 PX_VA_SIP_SS___XX__01_SDP Value to choose SDP parameters to be sent from table below SS___XX__04 for use in SS___XX__01. integer 1.89 PX_VA_SIP_SS___XX__01_CODEC Value to choose codec to be sent from table below SS___XX__04 for use in SS___XX__01. integer 1.90 PX_VA_SIP_SS___XX__02_SDP Value to choose SDP parameters to be sent from table below SS___XX__04 for use in SS___XX__02. integer 1.91 PX_VA_SIP_SS___XX__02_CODEC Value to choose codec to be sent from table below SS___XX__04 for use in SS___XX__02. integer 1.92 PX_VA_SIP_SS___XX__03_SDP Value to choose SDP parameters to be sent from table below SS___XX__04 for use in SS___XX__03. integer 1.93 PX_VA_SIP_SS___XX__03_CODEC Value to choose codec to be sent from table below SS___XX__04 for use in SS___XX__03. integer 1.94 PX_VA_SIP_SS___XX__04_SDP Value to choose SDP parameters to be sent from table below SS___XX__04 for use in SS___XX__04. integer 1.95 PX_VA_SIP_SS___XX__04_CODEC Value to choose codec to be sent from table below SS___XX__04 for use in SS___XX__04. integer 1.96 PX_SIP_CallClearingMsg Value to choose call clearing message to be sent from table below SI___XX__U10. Valid choices are: 404, 500, 410, 484. integer 1.97 PX_SIP_phonecontext SIP URL should include phone-context (compare SIP parameter value). boolean ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 21 A.2.2 ISDN-related PIXIT Tables A.2 to A.5 list the ISDN-related PIXIT items associated with the ATS. Each PIXIT item corresponds to a Module Parameter of the ATS. Default values are not provided. Table A.2: General SS/SUT-related ISDN PIXIT items Item IModule Parameter Description Type Value 2.1 PX_Isdn_Basic1 Select whether basic or primary rate access applies on the ISDN side for the first ISDN access. True = Basic access False = Primary Rate Access boolean 2.2 PX_Isdn_PtP1 Select whether point-to-point or point- to-multipoint configuration applies on the ISDN side for the first ISDN access. True = point-to-point False = point-to-multipoint boolean 2.3 PX_Isdn_Basic2 Select whether basic or primary rate access applies on the ISDN side for the second ISDN access. True = Basic access False = Primary Rate Access boolean 2.4 PX_Isdn_PtP2 Select whether point-to-point or point- to-multipoint configuration applies on the ISDN side for the second ISDN access. True = point-to-point False = point-to-multipoint boolean 2.5 PX_Isdn_L2Init Select whether the data link has to be released and re-established at the start of each test on the ISDN side. True = Data link reset for each test False = Keep data link established boolean 2.6 PX_Isdn_WaitRestart Select whether the IUT sends RESTART messages after re-establishment of the multiple frame operation. True = Wait for RESTART False = Do not wait for RESTART boolean ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 22 Table A.3: Timer-related ISDN PIXIT items Item IModule Parameter Description Type Value 3.1 PX_Isdn_TAC Time to control the reception of a message. float 3.2 PX_Isdn_TNOAC Time to control that IUT sends nothing. float 3.3 PX_Isdn_TWAIT Time to control that IUT reacts prior to Upper Tester action. float 3.4 PX_Isdn_TSYNC Time to control synchronization. float 3.5 PX_TDelay Time to delay messages before sending. float 3.6 PX_Isdn_WaitRestart_Duration Time to wait for RESTART messages after L2 re-establishment. float 3.7 PX_Isdn_T301 Maximum time for ISDN protocol timer T301, T301 is started on receipt of ALERTING and stopped on receipt of CONNECT. float 3.8 PX_Isdn_T304 Maximum time for ISDN protocol timer T304, T304 is started on sending of SETUP ACKNOWLEDGE and stopped on receipt of INFORMATION. float 3.9 PX_Isdn_T307 Maximum time for ISDN protocol timer T307, T307 is started on sending of SUSPEND ACKNOWLEDGE and stopped on receipt of RESUME. float 3.10 PX_Isdn_T_CFNR Maximum time for ISDN protocol timer T_CFNR, T_CFNR is started on receipt of ALERTING, if CFNR is activated and stopped on receipt of CONNECT. float Table A.4: Operator-check-related ISDN PIXIT items Item IModule Parameter Description Type Value 4.1 PX_Isdn_CheckConversation True if conversation check is implemented and used. Otherwise false (see note 1). boolean 4.2 PX_Isdn_CheckRinging True if ringing check is implemented and used. Otherwise false (see note 2). boolean 4.3 PX_Isdn_CheckDTMF True if DTMF tone check is implemented and used. Otherwise false (see note 3). NOTE 1: If true, test execution will stop at positions where the TP indicates "conversation" until the operator enters the check result. NOTE 2: If true, test execution will stop at positions where the TP indicates "ringing" until the operator enters the check result. NOTE 3: If true, test execution will stop at positions where the TP indicates "DTMF" until the operator enters the check result. ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 23 Table A.5: ISDN PIXIT items associated with message fields Item Module Parameter Description Type Value Called Party Number 5.1.1 PX_Isdn_CDPN_numberingPlan IdentificationDefault Default value of the numbering plan field of the called party number information. bitstring(4) 5.1.2 PX_Isdn_CDPN_TypeOfNumbe r_SIP_Access Default value of the type of number field of the called party number information to call a (first) SIP access. bitstring(3) 5.1.3 PX_Isdn_CDPN_SIP_Access Default value of the digits field of the called party number information to call a (first) SIP access. charstring 5.1.4 PX_Isdn_CDPN_DigitsFirstPorti on Value of the digits field of an incomplete called party number information. The number digits do not allow routing to the SIP side. charstring 5.1.5 PX_Isdn_CDPN_DigitsSecondP ortion Value of the digits field of a called party number information complementing the number given in PX_Isdn_CDPN_DigitsFirstPortion. charstring 5.1.6 PX_Isdn_CDPN_TypeOfNumbe r_1stISDN_Access Default value of the type of number field of the called party number information to call the first ISDN access. bitstring(3) 5.1.7 PX_Isdn_CDPN_1stISDN_Acce ss Default value of the digits field of the called party number information to call the first ISDN access. charstring 5.1.8 PX_Isdn_CDPN_TypeOfNumbe r_2ndISDN_Access Default value of the type of number field of the called party number information to call the second ISDN access. bitstring(3) 5.1.9 PX_Isdn_CDPN_2ndISDN_Acc ess Default value of the digits field of the called party number information to call the second ISDN access. charstring 5.1.10 PX_Isdn_CDPN_TypeOfNumbe r_3rdISDN_Access Default value of the type of number field of the called party number information to call the third ISDN access. bitstring(3) 5.1.11 PX_Isdn_CDPN_3rdISDN_Acce ss Default value of the digits field of the called party number information to call the third ISDN access. charstring 5.1.12 PX_Isdn_CDPN_TypeOfNumbe r_2ndSIP_Access Default value of the type of number field of the called party number information to call a second SIP access. bitstring(3) 5.1.13 PX_Isdn_CDPN_2ndSIP_Acces s Default value of the digits field of the called party number information to call a (second) SIP access. charstring Calling Party Number and Subaddress 5.2.1 PX_Isdn_CGPN_numberingPla nIdentificationDefault Default value of the numbering plan field of the calling party number information. bitstring(4) 5.2.2 PX_Isdn_CGPN_TypeOfNumbe rDefault Default value of the type of number field of the calling party number information. bitstring(3) 5.2.3 PX_Isdn_CGPN_DigitsDefault Default value of the digits field of the calling party number information. charstring 5.2.4 PX_Isdn_CGPS_DigitsDefault Default value of the digits field of the calling party subaddress information. charstring 5.2.5 PX_Isdn_CGPS_TypeDefault Default value of the type of subaddress field of the calling party subaddress information. bitstring(3) ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 24 Item Module Parameter Description Type Value Connected number 5.3.1 PX_Isdn_CODN_numberingPla nIdentificationDefault Default value of the numbering plan field of the connected number information. bitstring(4) 5.3.2 PX_Isdn_CODN_TypeOfNumbe rDefault Default value of the type of number field of the connected number information. bitstring(3) 5.3.3 PX_Isdn_CODN_DigitsDefault Default value of the digits field of the connected number information. charstring Bearer Capability 5.4.1 PX_Isdn_BCAP_TransferCapab ility_tx Default value of the transfer capability of the bearer capability information (to be sent when the TP does not specify a specific value for that field). bitstring(5) 5.4.2 PX_Isdn_BCAP_TransferCapab ility_rx Default value of the transfer capability of the bearer capability information (to be received when the TP does not specify a specific value for that field). bitstring(5) A.2.3 General PIXIT The PIXIT of table A.6 are general timer items that control the synchronization between the SIP and the ISDN test components. Each PIXIT item corresponds to a Module Parameter of the ATS. Table A.6: General PIXIT items Item IModule Parameter Description Type Value 6.10 PX_TSYNC_TIME_LIMIT Default time limit for a sync client to reach a synchronization point. float 6.11 PX_TSHUT_DOWN_TIME_LIMIT Default time limit for a sync client to finish its execution of the shutdown default. float ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 25 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_18600102v010201p0.zip which accompanies the present document. ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 26 Annex A (informative): Change history Date WG Doc. CR Rev CAT Title / Comment Current Version New Version STF306 validation output 1.1.1 1.2.1 ETSI ETSI TS 186 001-2 V1.2.1 (2009-09) 27 History Document history V1.1.1 July 2008 Publication V1.2.1 September 2009 Publication
f72e5b464f10bf0050bab19795493770	186 001-1	1 Scope	The present document specifies the Test Suite Structure and Test Purposes (TSS&TP) for Network Integration Testing (NIT) to verify the overall compatibility of SIP, ISDN and non-ISDN (PSTN) over the national or international ISDN networks. The TSS&TP specification covers the procedures described in Recommendation ITU-T Q.1912.5 [51] or ETSI EN 383 001 [49] or ETSI TS 129 163 [i.20] and ETSI EN 300 899-1 [23]. For SIP and SDP specific terminology, the references are ETSI TS 124 229 [55] and IETF RFC 3261 [28]. Figure 1: SIP-ISDN and SIP-PSTN inter-working testing architecture with SIP-I and ISUP ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 8 Figure 2: SIP-ISDN and SIP-PSTN inter-working testing architecture with SIP II NNI
f72e5b464f10bf0050bab19795493770	186 001-1	2 References
f72e5b464f10bf0050bab19795493770	186 001-1	2.1 Normative references	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. The following referenced documents are necessary for the application of the present document. [1] Void. [2] Recommendation ITU-T Q.1902.2 (2001): "Bearer Independent Call Control protocol (Capability Set 2) and Signalling System No.7 ISDN User Part: General functions of messages and parameters". [3] Void. [4] Void. [5] Void. [6] Void. [7] Void. [8] Void. [9] Void. [10] Void. [11] Void. ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 9 [12] Void. [13] Void. [14] Recommendation ITU-T Q.734.1 (03-1993): "Stage 3 description for multiparty supplementary services using Signalling System No. 7: Conference calling". [15] Recommendation ITU-T Q.734.2 (07-1996): "Stage 3 description for multiparty supplementary services using Signalling System No. 7: Three-party service". [16] Void. [17] Void. [18] Void. [19] Void. [20] Void. [21] Void. [22] Recommendation ITU-T Q.850 (05-1998): "Usage of cause and location in the Digital Subscriber Signalling System No. 1 and the Signalling System No. 7 ISDN User Part". [23] ETSI EN 300 899-1: "Integrated Services Digital Network (ISDN); Signalling System No.7; Interworking between ISDN User Part (ISUP) version 2 and Digital Subscriber Signalling System No. one (DSS1); Part 1: Protocol specification [ITU-T Recommendation Q.699, modified]". [24] Void. [25] IETF RFC 4566 (2006): "SDP: Session Description Protocol". [26] IETF RFC 3966 (2004): "The tel URI for Telephone Numbers". [27] Void. [28] IETF RFC 3261 (2002): "SIP: Session Initiation Protocol". [29] Void. [30] IETF RFC 3264 (2002): "An Offer/Answer Model with Session Description Protocol (SDP)". [31] IETF RFC 3311 (2002): "The Session Initiation Protocol (SIP) UPDATE Method". [32] IETF RFC 3312 (2002): "Integration of Resource Management and Session Initiation Protocol (SIP)". [33] IETF RFC 3323 (2002): "A Privacy Mechanism for the Session Initiation Protocol (SIP)". [34] IETF RFC 3325 (2002): "Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks". [35] Void. [36] Void. [37] Void. [38] Void. [39] Void. [40] Void. [41] Void. ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 10 [42] ETSI ES 283 003: "Telecommunications and Internet converged Services and Protocols for Advanced Networking (TISPAN); IP Multimedia Call Control Protocol based on Session Initiation Protocol (SIP) and Session Description Protocol (SDP) Stage 3 [3GPP TS 24.229 [Release 7], modified]". [43] ETSI TS 124 607 V10.1.0 (03-2014): "Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; Originating Identification Presentation (OIP) and Originating Identification Restriction (OIR) using IP Multimedia (IM) Core Network (CN) subsystem; Protocol specification (3GPP TS 24.607 version 10.1.0 Release 10". [44] ETSI TS 124 608 V10.1.0 (07-2013): "Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; Terminating Identification Presentation (TIP) and Terminating Identification Restriction (TIR) using IP Multimedia (IM) Core Network (CN) subsystem; Protocol specification (3GPP TS 24.608 version 10.1.0 Release 10)". [45] ETSI TS 124 604 V10.10.0 (07-2015): "Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; Communication Diversion (CDIV) using IP Multimedia (IM) Core Network (CN) subsystem; Protocol specification (3GPP TS 24.604 version 10.10.0 Release 10)". [46] ETSI TS 124 605 V10.1.0 (01-2013): "Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; Conference (CONF) using IP Multimedia (IM) Core Network (CN) subsystem; Protocol specification (3GPP TS 24.605 version 10.1.0 Release 10)". [47] Void. [48] Void. [49] ETSI EN 383 001: "Telecommunications and Internet converged Services and Protocols for Advanced Networking (TISPAN); Interworking between Session Initiation Protocol (SIP) and Bearer Independent Call Control (BICC) Protocol or ISDN User Part (ISUP) [ITU-T Recommendation Q.1912.5, modified]". [50] Void. [51] Recommendation ITU-T Q.1912.5 (2004): "Interworking between Session Initiation Protocol (SIP) and Bearer Independent Call Control Protocol or ISDN User Part". [52] Recommendation ITU-T Q.699 (09-1997): "Interworking between ISDN access and non-ISDN access over ISDN User Part of Signalling System No. 7". [53] Recommendation ITU-T Q.931 (05-1998): "ISDN user-network interface layer 3 specification for basic call control". [54] 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 ETSI TS 34.229-1 version 6.3.0 Release 6)". [55] ETSI TS 124 229: "Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; IP multimedia call control protocol based on Session Initiation Protocol (SIP) and Session Description Protocol (SDP); Stage 3 (3GPP ETSI TS 24.229 Release 10)". [56] IETF RFC 2833: "RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals". [57] ETSI TS 183 036: "Telecommunications and Internet converged Services and Protocols for Advanced Networking (TISPAN); ISDN/SIP interworking; Protocol specification". [58] ETSI TS 183 043: "Telecommunications and Internet converged Services and Protocols for Advanced Networking (TISPAN); IMS-based PSTN/ISDN Emulation; Stage 3 specification". ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 11
f72e5b464f10bf0050bab19795493770	186 001-1	2.2 Informative references	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. NOTE: While any hyperlinks included in this clause were valid at the time of publication, ETSI cannot guarantee their long term validity. 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. [i.1] Void. [i.2] 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". [i.3] ETSI EN 300 403-1: "Integrated Services Digital Network (ISDN); Digital Subscriber Signalling System No. one (DSS1) protocol; Signalling network layer for circuit-mode basic call control; Part 1: Protocol specification [Recommendation ITU-T Q.931 (1993), modified]". [i.4] 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". [i.5] ETSI EN 300 207-1: "Integrated Services Digital Network (ISDN); Diversion supplementary services; Digital Subscriber Signalling System No. One (DSS1); Part 1: Protocol specification". [i.6] 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". [i.7] 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". [i.8] 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". [i.9] 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". [i.10] 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". [i.11] 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 ETSI TS 24.147 version 9.1.0 Release 9)". [i.12] 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". [i.13] ETSI ETS 300 648: "Public Switched Telephone Network (PSTN); Calling Line Identification Presentation (CLIP) supplementary service; Service description". [i.14] 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 ETSI TS 186 001-1 V3.1.1 (2015-11) 12 [i.15] 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". [i.16] ETSI TBR 008: "Integrated Services Digital Network (ISDN); Telephony 3,1 kHz teleservice; Attachment requirements for handset terminals". [i.17] Recommendation ITU-T Q.951: "Stage 3 description for number identification supplementary services using DSS 1". [i.18] Recommendation ITU-T Q.939: "Typical DSS 1 service indicator codings for ISDN telecommunications services". [i.19] ETSI TS 183 028: "Telecommunications and Internet converged Services and Protocols for Advanced Networking (TISPAN); Common Basic Communication procedures; Protocol specification". [i.20] 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 ETSI TS 29.163 version 9.1.0 Release 9)". [i.21] ISO/IEC 9646 (1994): "Information technology - Open Systems Interconnection -Conformance testing methodology and framework". [i.22] ETSI TS 133 203: "Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; 3G security; Access security for IP-based services (3GPP TS 33.203 version 10.3.0 Release 10)". [i.23] ETSI TR 133 978: "Universal Mobile Telecommunications System (UMTS); Security aspects of early IP Multimedia Subsystem (IMS) (3GPP TR 33.978 version 7.0.0 Release 7)". [i.24] IETF RFC 2617: "HTTP Authentication: Basic and Digest Access Authentication". [i.25] IETF RFC 3761: "The E.164 to Uniform Resource Identifiers (URI) Dynamic Delegation Discovery System (DDDS) Application (ENUM)". [i.26] Void. [i.27] ETSI TR 123 981: "Universal Mobile Telecommunications System (UMTS); LTE; Interworking aspects and migration scenarios for IPv4-based IP Multimedia Subsystem (IMS) implementations (3GPP TR 23.981 version 10.0.0 Release 10)". [i.28] ETSI TS 186 011-2: "Core Network and Interoperability Testing (INT); IMS NNI Interoperability Test Specifications (3GPP Release 10); Part 2: Test descriptions for IMS NNI Interoperability". [i.29] ETSI TS 129 165: "Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; Inter-IMS Network to Network Interface (NNI) (3GPP TS 29.165 version 10.20.0 Release 10)".
f72e5b464f10bf0050bab19795493770	186 001-1	3 Definitions and abbreviations
f72e5b464f10bf0050bab19795493770	186 001-1	3.1 Definitions	For the purposes of the present document, the following terms and definitions apply. For BICC or ISUP specific terminology, the reference is Recommendation ITU-T Q.1902.2 [2]. For SIP and SDP specific terminology, the reference is ETSI TS 124 229 [55]. Basic Call Control (BCC): signalling protocol associated with the DSS1 - ISDN Basic Call control procedures of Recommendation ITU-T Q.931 [53] (ETSI EN 300 403-1 [i.3]) Incoming Interworking Unit (I-IWU): physical entity, (which can be combined with a BICC ISN or ISUPexchange) that terminates incoming calls using SIP and originates outgoing calls using the BICC or ISUP protocols ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 13 incoming or outgoing: direction of a call (not signalling information) with respect to a reference point incoming SIP or BICC/ISUP (network): network, from which the incoming calls are received, that uses the SIP or BICC/ISUP protocol (without the term "network", it simply refers to the protocol) inopportune: specifies a test purpose covering a signalling procedure where an inopportune message (type of message not expected in the IUT current state) is sent to the IUT Outgoing Interworking Unit (O-IWU): physical entity, (which can be combined with a BICC ISN or ISUP exchange) that terminates incoming calls using BICC or ISUP protocols and originates outgoing calls using the SIP outgoing SIP or BICC/ISUP (network): network, to which the outgoing calls are sent, that uses the SIP or BICC/ISDN protocol NOTE: Without the term "network", it simply refers to the protocol. SIP precondition: indicates the support of the SIP "precondition procedure" as defined in IETF RFC 3312 [32] test purpose: non-formal test description, mainly using text NOTE: TSIs test description can be used as the basis for a formal test specification (e.g. Abstract Test Suite in TTCN). See ISO/IEC 9646 [i.21]. valid: specifies a test purpose covering a signalling procedure where all the messages sent to or received from the IUT are valid (expected in the current status of the IUT) and correctly encoded
f72e5b464f10bf0050bab19795493770	186 001-1	3.1.1 Conventions for representation of SIP/SDP information	1) All letters of SIP method names are capitalized. EXAMPLE 1: INVITE, INFO. 2) SIP header fields are identified by the unabbreviated header field name as defined in the relevant RFC, including capitalization and enclosed hyphens but excluding the following colon. EXAMPLE 2: To, From, Call-ID. 3) Where it is necessary to refer with finer granularity to components of a SIP message, the component concerned is identified by the ABNF rule name used to designate it in the defining RFC (generally 25/IETF RFC 3261 [28]), in plain text without surrounding angle brackets. EXAMPLE 3: Request-URI, the user info portion of a sip: URI. 4) URI types are represented by the lower-case type identifier followed by a colon and the abbreviation "URI" EXAMPLE 4: sip: URI, tel: URI. 5) SIP provisional responses and final responses other than 2XX are represented by the status code followed by the normal reason phrase for that status code, with initial letters capitalized. EXAMPLE 5: 100 Trying, 484 Address Incomplete. 6) Because of potential ambiguity within a call flow about which request a 200 OK final response answers, 200 OK is always followed by the method name of the request. EXAMPLE 6: 200 OK INVITE, 200 OK PRACK. 7) A particular line of an SDP session description is identified by the two initial characters of the line -- that is, the line type character followed by "=" EXAMPLE 7: m=line, a=line. 8) Where it is necessary to refer with finer granularity to components of a session description, the component concerned is identified by its rule name in the ABNF description of the SDP line concerned, delimited with angle brackets. EXAMPLE 8: the <media> and <fmt> components of the m= line. ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 14
f72e5b464f10bf0050bab19795493770	186 001-1	3.2 Abbreviations	For the purposes of the present document, the following abbreviations apply: ACR Anonymous Call Rejection AKA Authentication and Key Agreement ANM Answer Message APRI Address Presentation Restriction Indicator AS Aplication Server ATS Abstract Tests Suite BC Bearer Capability CC Call Control CD Call Diversion CD-ISS Call Diversion ISDN - SIP - SIP CF Call Forwarding CFB Call Forwarding Busy CFNL Communication Forwarding on No Logged-in CFNR Call Forwarding No Reply CFU Call Forwarding Unconditional CLIP Calling Line Identification Presentation CLIR Calling Line Identification Restriction CN Core Network COLP Connected Line Identification Presentation COLR Called Line Identification Restriction COMP Complete message CON Connect message CONF Conference CONN Connect Massage CR Call Reference CS Circuit switched CSCF Call Session Control Function CUG Closed User Group CV Call Variable CV_SIP Call Variable for SIP CW Call Waiting DDI Direct Dialling In DHCP Dynamic Host Configuration Protocol DNS Domain Name System DTMF Dual-tone multi-frequency signaling ENUM Telephone Number Mapping ES European Standard FAC Facility message FCI Forward Call Indicator GSM Global System for Mobile communications GW GateWay HLC High Layer Capability HTTP Hypertext Transfer Protocol I Inopportune IA Incoming Allowed ICB Incoming Call Baring IE Information Element IMS IP Multimedia Subsystem IP Internet Protocol IPSEC Internet Protocol Security IPX Internetwork Packet Exchange IS_UD ISDN SIP - Unrestricted Digital information ISDN Integrated Services Digital Network ISI ISDN- SIP- ISDN ISS ISDN - SIP - SIP ISUP ISDN User Part IUT Implementation Under Test ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 15 IWU Interworking Unit LLC Low Layer Capability LPC Linear Predictive Coding MGCF Media Gateway Control Funktion MGW Media Gateway NAPTR Naming Authority Pointer NDC National Destination Code NDUB Network Determined User Busy NGN New Generation Network NNI Network - Network - Interface NS Name Server OBCI Optional Backward Call Indicator OIP Originating Identification Presentation OIR Originating Identification presentation Restriction PA Progress Indicator PBX Private Branch Exchange PCMA Puls-Code-Modulation- A law PCMU Puls-Code-Modulation- U law PER Packed Encoding Rules PI Progress Indicator PI_VA Progress Indicator Variable PICS Protocol Implementation Conformance Statement PIXIT Protocol Implementation eXtra Information for Testing PRACK PRACK message PROC Proceeding message PSTN Public Switched Telephone Network PT Posture Transport QCELP Q-Code Excitation Linear Prediction QoS Quality of Service REL Release Message RLC Release Complete Message S Syntactically invalid SA Security Association SC Sending Complete SCN Switched Circuit Network SDP Session Description Protocol SII SIP ISDN ISDN SIP Session Initiation Protocol SIP-I Session Initiation Protocol - ISUP (SIP with encapsulated ISUP) SIPS Session Initiation Protocol Security SIS SIP ISDN SIP SN Subscriber Number SUS Suspend message SUT System Under Test TIR Terminating Identification Restriction TP Test Purpose TR Technical Report TSS Test Suite Structure UA User Agent UAC User Agent Client UDI Unrestricted Digital Information UDUB User Determined User Busy URI Uniform Resource Identifier V Valid VA Variable XML Extensible Markup Language ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 16
f72e5b464f10bf0050bab19795493770	186 001-1	4 Test Suite Structure (TSS)
f72e5b464f10bf0050bab19795493770	186 001-1	4.1 Test Suite Structure (TSS)
f72e5b464f10bf0050bab19795493770	186 001-1	4.1.1 ISDN-SIP	C - Plane / U - Plane Basic_Call Successful Voice IS_XX_xx Codec negotiation IS_CN_xx Update Tests IS_XX_UP_xx DTMF IS_DTMF_xx UDI IS_UD_xx C - Plane Unsuccessful IS_XX_Uxx Supplementary Services CLIP IS_XXSSCLIPxx CLIR IS_XXSSCLIRxx COLP/COLR (TIP/TIR) IS_XXSSCOLPxx CFU ISI_XXSSCFUxx ISS_XXSSCFUxx CFB ISI_XXSSCFBxx ISS_XXSSCFBxx CFNR ISI_XXSSCFNRxx ISS_XXSSCFNRx CFNL ISS_XXSSCFNLxx 3PTY ISI_XXSS3PTYxx ISS_XXSS3PTYxx HOLD ISI_XXSSHOLDxx CONF IS_XXSSCONFxx ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 17
f72e5b464f10bf0050bab19795493770	186 001-1	4.1.2 SIP-ISDN	C - Plane / U - Plane Basic_Call Successful 3,1 kHz audio SI_AU_xx Codec negotiation SI_XX_CN_xx DTMF SI_XX_DT_xx UDI SI_UD_xx C - Plane Unsuccessful SI_XX _Uxx Supplementary Services CLIP SI_XXSSOIPxx CLIR SI_XXSSOIRxx COLP/COLR (TIP/TIR) SI_XXSSCOLPxx CFU SIS_XXSSCFUxx SII_XXSSCFUxx CFB SIS_XXSSCFBxx SII_XXSSCFBxx CFNR SIS_XXSSCFNRxx SII_XXSSCFNRxx 3PTY SII_XXSS3PTYXX SIS_XXSS3PTYXX TP SI_XXSSTPxx CUG SI_XXSSCUGxx HOLD SI_XXSSHOLDxx CONF SI_XXSSCONFxx CW SI_XXSSCWxx ACR SI_XXSSACRxx
f72e5b464f10bf0050bab19795493770	186 001-1	4.1.3 PSTN-SIP	C - Plane / U - Plane Basic_Call Successful PS_AU_Xxx C - Plane Unsuccessful PS_AU_Uxx Supplementary Services CLIP PS_XXSSCLIPxx CLIR PS_XXSSCLIRxx CFU PSP_XXSSCFUxx PSS_XXSSCFUxx CFB PSP_XXSSCFBxx PSS_XXSSCFBxx CFNR PSP_XXSSCFNRxx PSS_XXSSCFNRxx CFNL PSP_XXSSCFNLxx ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 18
f72e5b464f10bf0050bab19795493770	186 001-1	4.1.4 SIP-PSTN	C - Plane / U - Plane Basic_Call Successful 3,1 kHz audio SP_AU_xx Unsuccessful SP_XX_Uxx Supplementary Services CLIP SP_XXSSCLIPxx CLIR SP_XXSSCLIRxx CFU SPS_XXSSCFUxx SPP_XXSSCFUxx CFB SP_XXSSCFBxx SPP_XXSSCFBxx CFNR SPS_XXSSCFNRxx SPP_XXSSCFNRxx
f72e5b464f10bf0050bab19795493770	186 001-1	5 Numbering Scheme
f72e5b464f10bf0050bab19795493770	186 001-1	5.1 General description	Pos 1: Network of the A-Subscriber. Pos. 2: Network of the B-Subscriber. Pos. 3: Network of the C-Subscriber. Pos. 4: Network of the D-Subscriber. Pos. 5: Network of the E-Subscriber. The following Network Codes apply: _: No such network used (used e.g. for C-Subscriber in successful A to B Calls) (underscore makes it easier to read the name) P: PSTN I: ISDN S: SIP (Extensions will be added when needed) Pos. 6 and 7: Bearer- or Teleservice involved XX: defined per PIXIT value NOTE: TSIs may be appropriate for Test Purposes (provided the Test Purpose states for which Bearer- and/or Tele Services it should be tested). It is however NOT appropriate for Test Cases since it would be detrimental to Test Automation. SP: Speech AU: 3,1 kHz Audio UD: UDI UT: UDI/TA CN: Codec negotiation ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 19 DT: DTMF UP: UPDATE Method Pos. 8 and 9: _: No Supplementary Services Involved / Successful _U: No Supplementary Services Involved / Unsuccessful SS: Supplementary Services Involved SI: Supplementary Services interaction SN: Nonsymmetrical Supplementary Services Involved ST: Supplementary Services transparent
f72e5b464f10bf0050bab19795493770	186 001-1	5.2 Basic Call	Speech IS_XX_XX 1 2 3 4 5 6 7 8 9 10 11 I S _ _ _ S P _ _ x x
f72e5b464f10bf0050bab19795493770	186 001-1	5.3 Supplementary Services	CLIP IS_XXSSCLIP XX 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 I S _ _ _ X X S S C L I P x X
f72e5b464f10bf0050bab19795493770	186 001-1	6 Test purposes	The registration and application usage procedures in the ETSI TS shall be compliant to IETF RFC 3261 [28] and ETSI TS 124 229 [55]. The validation of the registration procedure is out of scope of the present document and is part of the Preamble used in the test cases. The registration conformance tests based on ETSI TS 124 229 [55] described in ETSI TS 134 229-1 [54]. The preconditions mechanism shall be supported by the UE in case of supporting IMS. The handling of preconditions at the originating or /and terminating UE (MGCF in case if interworking) is described in table 0. ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 20 Table 0 PIXIT Values UE (MGCF) originating case UE (MGCF) terminating case VA precondition" option-tag in the Supported header local resource reservation is required at the terminating UE local resource reservation is not required by the terminating UE and the terminating UE supports the precondition mechanism VA_1 "precondition" option-tag in the Supported header the terminating UE shall make use of the precondition mechanism VA_2.1 "precondition" option-tag in the Supported header and required resources at the originating network are not reserved the terminating UE shall make use of the precondition mechanism VA_2.2 "precondition" option-tag in the Supported header and required resources at the originating network are not reserved the terminating UE shall use the precondition mechanism VA_3.1 "precondition" option-tag in the Supported header and required local resources at the originating network the terminating UE shall make use of the precondition mechanism VA_3.2 "precondition" option-tag in the Supported header and required local resources at the originating network the required local resources at the originating UE and the terminating UE are available, the terminating UE may use the precondition mechanism VA_4.1 INVITE request does not include the "precondition" option-tag in the Supported header the terminating UE shall not make use of the precondition mechanism. VA_4.2 INVITE request does not include the "precondition" option-tag in the Supported header the terminating UE shall not make use of the precondition mechanism. Dial string parameters options To header field- UE originated VA_5.1 sip: dialled digits@homehostportion;user=dialstring VA_5.2 sip: dialled digits@homehostportion;user=phone VA_5.3 sip: dialled digits; phone-context=<"+"CC>@homehostportion;user=phone VA_5.3 sip: dialled digits; phone-context=<"+"CC+NDC>@homehostportion;user=phone Request-URI VA_6.1 E164 Address (format "+"CC+NDC+SN) (e.g. as User info in SIP URI with user= phone, or as tel URI) ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 21
f72e5b464f10bf0050bab19795493770	186 001-1	6.1 Test Prerequisites
f72e5b464f10bf0050bab19795493770	186 001-1	6.1.1 IP Version	These test specifications are based on the use of IPv4 for SIP message transport throughout all IMS nodes as specified in ETSI TR 123 981 [i.27] but do not exclude the use of IPv6 in the case that all involved IMS nodes support this version of the IP protocol.
f72e5b464f10bf0050bab19795493770	186 001-1	6.1.2 Authentication and Security	The current test specification supports as default full IMS ETSI TS 133 203 [i.22] 3GPP security. Non-compliance with full IMS security features defined in ETSI TS 133 203 [i.22] is expected to be a problem mainly at the UE side, because of the potential lack of support of the USIM/ISIM interface (especially in 2G-only devices) and of the potential inability to support IPsec on some UE platforms. For those reasons, fallback to early IMS ETSI TR 133 978 [i.23] and SIP Digest authentication without key agreement and null authentication may be used to achieve satisfactory test results. Tests should however be executed with full IMS security if all required IMS nodes support it.
f72e5b464f10bf0050bab19795493770	186 001-1	6.1.3 Registration and Subscription
f72e5b464f10bf0050bab19795493770	186 001-1	6.1.3.1 SIP Call Flow	This clause describes the registration call flow under the authentication and security scope described in clause 4.2.2 in ETSI TS 186 011-2 [i.28].
f72e5b464f10bf0050bab19795493770	186 001-1	6.1.3.1.1 Early IMS Registration and Subscription Call Flow [i.28]	Early IMS security does not allow SIP requests to be protected using an IPsec Security Association (SA) because it does not perform a key agreement procedure. IPsec security associations are not set up between UE and P-CSCF, as they are in the full IMS security solution. For early IMS security, the expected registration and subscription sequence is: Step Direction Message Comment UE IMS 1 The UE establishes an IP bearer as required by its specific access network (optional). 2 P-CSCF address discovery using DHCP procedures for IPv4 (optional). 3 REGISTER The UE sends initial registration for IMS services. Unprotected 4 200 OK The IMS responds with 200 OK. 5 SUBSCRIBE The UE subscribes to its registration event package. 6 200 OK or 202 Accepted The IMS responds with 200 OK or 202 Accepted. 7 NOTIFY The IMS sends initial NOTIFY for registration event package, containing full registration state information for the registered public user identity in the XML body. 8 200 OK The UE responds with 200 OK. ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 22
f72e5b464f10bf0050bab19795493770	186 001-1	6.1.3.1.2 Full IMS Registration and Subscription Call Flow [i.28]	For full IMS security, the expected registration and subscription sequence is: Step Direction Message Comment UE IMS 1 The UE establishes an IP bearer as required by its specific access network (optional). 2 P-CSCF address discovery using DHCP procedures for IPv4 (optional). 3 REGISTER The UE sends initial registration for IMS services. Unprotected 4 401 Unauthorized The IMS responds with a valid Digest AKA authentication challenge and a list of integrity and encryption algorithms supported by the network as defined in the IMS_AKA procedure of ETSI TS 133 203 [i.22]. 5 Upon receipt of 401 Unauthorized, the UE selects the first integrity and encryption algorithm combination on the list received from the P-CSCF in 401 Unauthorized which is also supported by the UE. If the P-CSCF did not include any confidentiality algorithm in 401 Unauthorized then the UE shall select the NULL encryption algorithm. The UE then proceeds to establish two new pairs of IPSEC Security Associations (SA1 and SA2). 6 REGISTER The UE sends another REGISTER with authentication credentials over IPSEC security association SA1. Protected by SA1 7 200 OK The IMS responds with 200 OK over the same IPSEC security association SA1. 8 SUBSCRIBE The UE subscribes to its registration event package over the IPSEC security association SA2. Protected by SA2 9 200 OK or 202 Accepted The IMS responds with 200 OK or 202 Accepted over the IPSEC security association SA2. 10 NOTIFY The IMS sends initial NOTIFY for registration event package, containing full registration state information for the registered public user identity in the XML body, over the IPSEC security association SA2. 11 200 OK The UE responds with 200 OK over the IPSEC security association SA2.
f72e5b464f10bf0050bab19795493770	186 001-1	6.1.3.1.3 SIP Digest Registration and Subscription Call Flow	For SIP Digest authentication without key agreement and null authentication, the expected registration and subscription sequence is: Step Direction Message Comment UE IMS 1 The UE establishes an IP bearer as required by its specific access network (optional). 2 P-CSCF address discovery using DHCP procedures for IPv4 (optional). 3 REGISTER The UE sends initial registration for IMS services. Unprotected 4 401 Unauthorized The IMS responds with a valid HTTP Digest authentication challenge as defined in IETF RFC 2617 [i.24]. 5 REGISTER The UE sends another REGISTER with authentication credentials. 6 200 OK The IMS responds with 200 OK. 7 SUBSCRIBE The UE subscribes to its registration event package. 8 200 OK or 202 Accepted The IMS responds with 200 OK or 202 Accepted. 9 NOTIFY The IMS sends initial NOTIFY for registration event package, containing full registration state information for the registered public user identity in the XML body. 10 200 OK The UE responds with 200 OK. ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 23
f72e5b464f10bf0050bab19795493770	186 001-1	6.1.4 Supported Options
f72e5b464f10bf0050bab19795493770	186 001-1	6.1.4.1 Security	Support for security agreement is optional in case of Full IMS Reg. It shall only be used in case all IMS nodes support it.
f72e5b464f10bf0050bab19795493770	186 001-1	6.1.4.2 Signalling Compression	"No SigComp" is the default signalling configuration in all test descriptions. Tests may be executed with signalling compression if the required nodes support it.
f72e5b464f10bf0050bab19795493770	186 001-1	6.1.5 Number Resolution	"ENUM (IETF RFC 3761 [i.25]) is a capability that transforms E.164 numbers into domain names and then uses the DNS (Domain Name System) to discover NAPTR records that specify the services available for a specific domain name." The test infrastructure focuses on the use of Infrastructure ENUM to map a telephone number into a SIP URI that could identify a specific point of interconnection (PoI) to that communication provider's network that could enable the originating party to establish communication with the associated terminating party either directly or through an IPX. The Infrastructure ENUM platform has a tiered structure and provides authoritative, service specific information to the quering party. A combination of Tier 0, Tier 1 and Tier 2 registries enables global discovery of ENUM data. When returning the SIP URI of an PoI the ENUM solution acts a hosted T2 ENUM registry for the number range holder. When returning a NS record the ENUM solution acts as either a Tier 0 or Tier 1 registry.
f72e5b464f10bf0050bab19795493770	186 001-1	6.2 ISDN - SIP
f72e5b464f10bf0050bab19795493770	186 001-1	6.2.1 Basic Call
f72e5b464f10bf0050bab19795493770	186 001-1	6.2.1.1 Test purposes for ISDN-SIP Basic call Successful - Speech or 3,1 kHz audio	Successful Speech or 3,1 kHz audio calls IS_XX_01 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1.1 ETSI EN 383 001 [49], clause 7.1.1 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call Test purpose: Ensure that the call establishment using en-bloc sending is performed correctly. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). The call is released from the calling user. At the call establishment the SDP parameters in table 1 can be used. ISDN Parameter values: BC=PIXIT, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 24 Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case b) IMS with 100 rel SETUP INVITE CALL PROC ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) IMS SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 25 IS_XX_02 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3.1 ETSI EN 383 001 [49], clause 7.3.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.5 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call, SIP Profile A or ETSI EN 383 001 [49] Profile B or ETSI TS 129 163 [i.20] Test purpose: Ensure that call establishment using en-bloc sending is performed correctly. Ensure that the ISDN user in the state U3 receives an ALERTING message with the progress indicator information element "call is not end-to-end ISDN (#1)" when the SIP user answers with 180 Ringing message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). The call is released from the called user. ISDN Parameter values: BC=PIXIT, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING PI #1 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING PI #1 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 26 IS_XX_02A ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3.1 ETSI EN 383 001 [49], clause 7.3.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.5 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call, ETSI TS 129 163 [i.20] Test purpose: Ensure that call establishment using en-bloc sending is performed correctly. P-Early-Media header not supported, 183 is not interworked sending complete indication received. Ensure that the ISDN user in the state U3 does not receive a Progress message when the SIP user answers with 183 Session Progress. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). The call is released from the called user. ISDN Parameter values: BC=PIXIT, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC 183 Session Progress ALERTING PI #1 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING PI #1 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 27 IS_XX_02B ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ETSI TS 129 163 [i.20], clause 7.2.3.2.5 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call, ETSI TS 129 163 [i.20]; P-Early-Media header supported and inserted by the network Test purpose: Ensure that call establishment using overlap sending is performed correctly. P-Early-Media header supported. Ensure that the ISDN user in the state U2 receive a Call Proceeding message when the SIP user answers with 183 Session Progress and P-Early-Media header supported and inserted by the network. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). The call is released from the called user. ISDN Parameter values: BC=PIXIT, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC PI #8 183 Session Progress ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 28 IS_XX_02C ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3.1 ETSI EN 383 001 [49], clause 7.3.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.5 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call, ETSI TS 129 163 [i.20]; P-Early-Media header supported and inserted by the network Test purpose: Ensure that call establishment using en-bloc sending is performed correctly. P-Early-Media header supported and inserted by the network. Ensure that the ISDN user in the state U3 receives a Progress message when the SIP user answers with 183 Session Progress. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). The call is released from the called user. ISDN Parameter values: BC=PIXIT, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC PROGRESS PI#8 183 Session Progress ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 29 IS_XX_03 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; ITU-T Q.1912.5 [51] Profile B with PI Test purpose: Ensure that call establishment using en-bloc sending is performed correctly. Ensure that the ISDN user in the state U3 receives an ALERTING message with the progress indicator information element #1 or #2 or both location "Network beyond Interworking point" when the SIP user answers with 180 Ringing message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). NOTE: According to ITU-T Q.699 [52] every message sent to the ISDN access may contain 2 Progress indicator information elements. ISDN Parameter values: BC=PIXIT , no HLC PI_VA (PIXIT) SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING PI 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING PI# VA 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 30 IS_XX_03A ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ETSI TS 129 163 [i.20], clause 7.2.3.2.5 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; ETSI TS 129 163 [i.20] P-Early-Media header not supported Test purpose: Ensure that call establishment using en-bloc sending is performed correctly. Ensure that the ISDN user in the state U3 receives an ALERTING message including a progress indicator I.E. with the descriptions "call is not end-to-end ISDN (#1)" or #2 or both, location "Network beyond Interworking point" when the SIP user answers with 180 Ringing message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). NOTE: According to ITU-T Q.699 [52] every message sent to the ISDN access may contain 2 Progress indicator information elements. ISDN Parameter values: BC=PIXIT , no HLC PI_VA (PIXIT) SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING PI 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING PI# 1 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 31 IS_XX_03B ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; ETSI TS 129 163 [i.20] P-Early-Media header supported Test purpose: Ensure that call establishment using en-bloc sending is performed correctly. Ensure that the ISDN user in the state U3 receives an ALERTING message including a PI#8 when the SIP user answers with 180 Ringing message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). NOTE: According to ITU-T Q.699 [52] every message sent to the ISDN access may contain 2 Progress indicator information elements. ISDN Parameter values: BC=PIXIT , no HLC PI_VA (PIXIT) SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING PI#8 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING PI# 1, PI#8 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 32 IS_XX_04 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3.1 ETSI EN 383 001 [49], clause 7.1.1 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; ITU-T Q.1912.5 [51] Profile B without PI Test purpose: Ensure that call establishment using en-bloc sending is performed correctly. Ensure that the ISDN user in the state U3 receives an ALERTING message without the progress indicator information element "call is not end-to-end ISDN (#1)" when the SIP user answers with 180 Ringing message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=PIXIT, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 33 IS_XX_05 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 ETSI EN 383 001 [49], clause 7.3 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; SIP Profile A or ETSI EN 383 001 [49] Profile B or ETSI TS 129 163 [i.20] Test purpose: Ensure that the call establishment using en-bloc sending is performed correctly. Ensure that the ISDN user in the state U3 receives a CONNECT message including a progress indicator I.E. with the descriptions "call is not end-to-end ISDN (#1)" location "Network beyond Interworking point" when the SIP user answers with a 200 OK message. Ensure that in the active call state (N10) the voice transfer on the media and B- channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=PIXIT, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC CONN PI# 1 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK CONN PI# 2 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 34 IS_XX_06 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; ITU-T Q.1912.5 [51] Profile B with PI Test purpose: Ensure that call establishment using en-bloc sending is performed correctly. Ensure that the ISDN user in the state U3 receives a CONNECT message with the progress indicator information element #1 or #2 or both location "Network beyond Interworking point" when the SIP user answers with 180 Ringing message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). NOTE: According to ITU-T Q.699 [52] every message sent to the ISDN access may contain 2 Progress indicator information elements. ISDN Parameter values: BC=PIXIT , no HLC PI_VA (PIXIT) SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC CONN PI# VA 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) Supported: 100 rel and precondition SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK CONN PI# VA 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 35 IS_XX_07 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3. ETSI EN 383 001 [49], clause 7.3 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; ITU-T Q.1912.5 [51] without PI Test purpose: Ensure that the call establishment using en-bloc sending is performed correctly. Ensure that the ISDN user in the state U3 receives a CONNECT message without a progress indicator I.E. with the descriptions "call is not end-to-end ISDN (#1)" location "Network beyond Interworking point" when the SIP user answers with a 200 OK message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=PIXIT, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 36 IS_SP_08 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.2 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 ETSI EN 383 001 [49], clause 7.3 ETSI TS 129 163 [i.20], clause 7.2.3.5.1 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; SIP Profile A or ETSI EN 383 001 [49] Profile B or ETSI TS 129 163 [i.20] Test purpose: Ensure that call establishment using overlap sending is performed correctly. Ensure that the ISDN user in the state U2 receives an ALERTING message with a progress indicators "call is not end-to-end ISDN (#1)" location Network beyond Interworking point" when the SIP user answers with a 180 Ringing message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP SETUP ACK INFO INVITE ALERTING PI #1 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP SETUP ACK INFO INVITE 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING PI #1 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 37 IS_SP_08A ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.2 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ETSI TS 129 163 [i.20], clause 7.2.3.5.1 TSS reference: SIP-ISDN/Basic_call/Successful/3,1 kHz audio Selection criteria: Basic_call; ETSI TS 129 163 [i.20] overlap receiving supported; In-Dialog Method SIP PBX Test purpose: Ensure that call establishment using overlap sending is performed correctly. Ensure that the ISDN user in the state U2 receives an ALERTING message when the SIP user answers with a 180 Ringing message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP SETUP ACK INFO INVITE cseg1 183 Session Progress cseg1 INFO INFO cseg2 200 OK cseg2 Call proceeding 183 cseg1 ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 38 IS_SP_08B ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.2 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ETSI TS 129 163 [i.20], clause 7.2.3.5.1 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; ETSI TS 129 163 [i.20] overlap receiving supported; multiple INVITE Overlap Dialling Procedures; one dialog open Test purpose: Ensure that call establishment using overlap sending is performed correctly. Ensure that the ISDN user in the state U2 receives an ALERTING message when the SIP user answers with a 180 Ringing message. Ensure that in the active call state (N10) the voice transfer on the media and B- channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE SETUP ACK 484 ACK INFO INVITE 484 ACK Call proceeding 183 Session Progress ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 39 IS_SP_08C ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.2 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ETSI TS 129 163 [i.20], clause 7.2.3.5.1 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; ETSI TS 129 163 [i.20] overlap receiving supported; multiple INVITE Overlap Dialling Procedures; two dialogs open Test purpose: Ensure that call establishment using overlap sending is performed correctly. Ensure that the ISDN user in the state U2 receives an ALERTING message when the SIP user answers with a 180 Ringing message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE csq 1 SETUP ACK INFO INVITE csq 2 484 csq 1 ACK Call proceeding 183 Session csq 2 Progress ALERTING 180 Ringing csq2 CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 40 IS_XX_09 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.2 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 ETSI EN 383 001 [49], clause 7.3 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call;ITU-T Q.1912.5 [51] Profile B with PI Test purpose: Ensure that call establishment using overlap sending is performed correctly. Ensure that the ISDN user in the state U2 receives a ALERTING message with a progress indicators #1 or #2 or both location "Network beyond Interworking point" when the SIP user answers with a 180 Ringing message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). NOTE: According to ITU-T Q.699 [52] every message sent to the ISDN access may contain 2 Progress indicator information elements. ISDN Parameter values: BC=PIXIT, no HLC PI_VA (PIXIT) SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP SETUP ACK INFO INVITE ALERTING PI #VA 180 Ringing INVITE CONN 200 OK INVITE ACK Conversa tion DISC BYE REL 200 OK BYE Case c) SETUP SETUP ACK INFO INVITE 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING PI #VA 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversa tion DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 41 IS_XX_10 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3.1 ETSI EN 383 001 [49], clause 7.1.1 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; ITU-T Q.1912.5 [51] Profile B without PI Test purpose: Ensure that call establishment using overlap sending is performed correctly. Ensure that the ISDN user in the state U2 receives an ALERTING message without the progress indicator information element "call is not end-to-end ISDN (#1)" when the SIP user answers with 180 Ringing message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=PIXIT, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP SETUP ACK INFO INVITE ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP SETUP ACK INFO INVITE 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 42 IS_XX_11 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.2 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 ETSI EN 383 001 [49], clause 7.3 ETSI TS 129 163 [i.20], clause 7.2.3.5.1 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; SIP Profile A or ETSI EN 383 001 [49] Profile B or ETSI TS 129 163 [i.20]; Test purpose: Ensure that call establishment using overlap sending is performed correctly. Ensure that the ISDN user in the state U2 receives a CONNECT message with a progress indicators "call is not end-to-end ISDN (#1)" location "Network beyond Interworking point" when the SIP user answers with a 200 OK message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=PIXIT, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP SETUP ACK INFO INVITE CONN PI #1 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK CONN PI #1 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 43 IS_XX_12 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.2 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 ETSI EN 383 001 [49], clause 7.3 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; ITU-T Q.1912.5 [51] Profile B with PI Test purpose: Ensure that call establishment using overlap sending is performed correctly. Ensure that the ISDN user in the state U2 receives a CONNECT message with a progress indicators set to PI_VA location "Network beyond Interworking point" when the SIP user answers with a 200 OK message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC= PIXIT, no HLC PI_VA (PIXIT) SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP SETUP ACK INFO INVITE CONN PI #VA 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) Supported: 100 rel and precondition SETUP SETUP ACK INFO INVITE 183 Session Progress PRACK 200 OK UPDATE 200 OK CONN PI #VA 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Values for test purposes IS_XX_12 VA PI information element VA_1 # 1 (call is not end-to-end ISDN) VA_2 # 2 (destination address is non-ISDN) VA_3 PI # 1 and PI # 2 ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 44 IS_XX_13 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.2 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 ETSI EN 383 001 [49], clause 7.3 ETSI TS 129 163 [i.20], clause 7.2.3.5.1 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; ITU-T Q.1912.5 [51] Profile B without PI Test purpose: Ensure that call establishment using overlap sending is performed correctly. Ensure that the ISDN user in the state U3 receives an ALERTING message without a progress indicators "call is not end-to-end ISDN (#1)" location Network beyond Interworking point" when the SIP user answers with a 180 Ringing message . Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=PIXIT, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP SETUP ACK INFO INVITE ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP SETUP ACK INFO INVITE 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE ISDN 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 45 IS_XX_14 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.3.3 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7 ETSI EN 383 001 [49], clause 7.7 ETSI TS 129 163 [i.20], clause 7.2.3.2.13 TSS reference: ISDN-SIP/Basic_call/Successful/Voice/ Selection criteria: Basic_call Test purpose: Ensure that the call establishment and the call clearing procedure is performed correctly when the calling user clears after answering with a DISCONNECT message indicating the Cause value # 16 "normal call clearing". The called user shall receive a BYE message. According to ETSI TS 129 163 [i.20] and ETSI EN 383 001 [49] the Reason Header field shall be included with Cause Value #16. Ensure that in the Call Delivered call state U4 the transfer of tone or announcement on the B- channel is performed correctly. ISDN Parameter values: BC=PIXIT, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE ISDN 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 46 IS_XX_15 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.3.3 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7 ETSI EN 383 001 [49], clause 7.7 ETSI TS 129 163 [i.20], clause 7.2.3.2.13 TSS reference: ISDN-SIP/Basic_call/Successful/Voice/ Selection criteria: Basic_call Test purpose: Ensure that the call clearing procedure is performed correctly when the called user clears after answering with a BYE message. The calling user shall receive a DISCONNECT message with the Cause value # 16 "normal call clearing". Ensure that in the Call Delivered call state U4 and disconnect indication state (N12) the transfer of tone or announcement on the B- channel is performed correctly. ISDN Parameter values: BC=PIXIT, no HLC SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) Supported: 100 rel and precondition SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 47 IS_SP_16 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 4.5.16 ETSI EN 300 899-1 [23], clause 2.1.1 ETSI TBR 008 [i.16], clause 5.1.3, ETSI EG 201 018 [i.2], clause 6.3.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 ETSI EN 383 001 [49], clause 7.3 ETSI TS 129 163 [i.20], clause 7.2.3.2.2.2 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; Test purpose: Ensure that call establishment supporting the telephony 3,1 kHz teleservice is performed correctly. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=speech, HLC=telephony SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) Supported: 100 rel and precondition SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 48 Table 1: PIXIT Values for test purposes IS_XX_01 to IS_XX_16 Variable m= line b= line a= line <media> <transport> <fmt-list> <modifier>:<bandwidth-value> rtpmap:<dynamic-PT> <encoding name>/<clock rate> [/encoding parameters> VA_01 audio RTP/AVP 0 (and possibly 8) AS:64 rtpmap:0 PCMU/8000 (and possibly rtpmap:8 PCMA/8000) VA_02 audio RTP/AVP Dynamic PT (and possibly a second Dynamic PT) AS:64 rtpmap:<dynamic-PT> PCMU/8000 (and possibly rtpmap:<dynamic-PT> PCMA/8000) VA_03 audio RTP/AVP 8 AS:64 rtpmap:8 PCMA/8000 VA_04 audio RTP/AVP Dynamic PT AS:64 rtpmap:<dynamic-PT> PCMA/8000 VA_05 audio RTP/AVP 0 and/or 8 AS:64 rtpmap:0 PCMU/8000 and/or rtpmap:8 PCMA/8000 VA_06 audio RTP/AVP 0 (and possibly 8) AS:64 rtpmap:0 PCMU/8000 (and possibly rtpmap:8 PCMA/8000) VA_07 audio RTP/AVP 8 AS:64 rtpmap:8 PCMA/8000 ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 49 IS_AU_17 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 4.5.17 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; Telefax G3 terminals; Telefax G3 terminals with T.38 Test purpose: Support of Telefax G3. Ensure that in the active call state (N10) the Fax transfer on the media and B-channels is performed correctly. ISDN Parameter values: BC=3,1 kHz audio, HLC = Facsimile G2/G3 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line Based on T.38. b = line AS: 64 m = line: udptl; T38 Comments: ISDN SUT SIP Case a) SETUP INVITE ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP SETUP ACK INFO INVITE 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 50 IS_AU_18 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 4.5.17 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1 ETSI EN 383 001 [49], clause 7.1 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; Telefax G3 terminals-inband Test purpose: Support of Telefax G3. Ensure that in the active call state (N10) the Fax transfer on the media and B-channels is performed correctly and the echo cancellers in the GW are not activated. No transcoding in the gateway takes place. ISDN Parameter values: BC=3,1 kHz audio, HLC = Facsimile G2/G3 SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line RTP/AVP b = line 64 kbit/s m = line: 8 Comments: ISDN SUT SIP Case a) SETUP INVITE ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP SETUP ACK INFO INVITE 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 51 IS_AU_19 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 4.5.5 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; Bearer service 3,1 kHz audio Test purpose: Ensure that the ISDN SETUP with the BC parameter value information transfer capability 3,1 kHz audio, voice band data via modem, synchronous/ asynchronous mode is set to MODE is mapped to the SIP INVITE Message. Ensure that in the active call state (N10) the data transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters) and the echo cancellers in the GW are not activated. ISDN Parameter values: BC= 3,1 kHz audio, voice band data via modem, synchronous/ asynchronous mode: MODE user rate: USER_RATE (table 2) SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP Case a) SETUP INVITE ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 52 IS_AU_20 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 4.5.18 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1 ETSI EN 383 001 [49], clause 7.1 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; Bearer service 3,1 kHz audio Test purpose: Ensure that the ISDN SETUP with the BC parameter value information transfer capability 3,1 kHz audio and the LLC ISDN Parameter values: 3,1 kHz audio, voice band data via modem, synchronous/ asynchronous mode is set to MODE, user rate set to USER_RATE is mapped to the SIP INVITE Message. Ensure that in the active call state (N10) the data transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters) and the echo cancellers in the GW are not activated. ISDN Parameter values: BC = 3,1 kHz audio, LLC = 3,1 kHz audio, voice band data via modem, synchronous/ asynchronous mode: MODE user rate: USER_RATE (table 2) SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP Case a) SETUP INVITE ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 53 IS_AU_21 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 4.5.18 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1 ETSI EN 383 001 [49], clause 7.1 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; Bearer service 3,1 kHz audio SIP selection criteria: Audio Test purpose: Ensure that the ISDN SETUP with the BC parameter value information transfer capability 3,1 kHz audio voice band data via modem, synchronous/ asynchronous mode is set to MODE, user rate set to USER_RATE and the LLC ISDN Parameter values: 3,1 kHz audio, voice band data via modem, synchronous/ asynchronous mode is set to MODE, user rate set to USER_RATE is mapped to the SIP INVITE. In the active call state (N10) ensure that the data transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters) and the echo cancellers in the GW are not activated. ISDN Parameter values: BC=LLC=3,1 kHz audio, voice band data via modem, synchronous/ asynchronous mode: MODE user rate: USER_RATE (table 2) SIP Parameter values: Dial string parameters options=PIXIT TYPE_SDP= PIXIT; PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP Case a) SETUP INVITE ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 54 Table 2: Values for test purposes IS_AU_19 to IS_AU_21 VA_01 MODE: synchronous USER_RATE: 1,2 kbit/s VA_02 MODE: synchronous USER_RATE: 2,4 kbit/s VA_03 MODE: synchronous USER_RATE: 3,6 kbit/s VA_04 MODE: synchronous USER_RATE: 4,8 kbit/s VA_05 MODE: synchronous USER_RATE: 7,2 kbit/s VA_06 MODE: synchronous USER_RATE: 8 kbit/s VA_07 MODE: synchronous USER_RATE: 9,6 kbit/s VA_08 MODE: synchronous USER_RATE: 14,4 kbit/s VA_09 MODE: synchronous USER_RATE: 16 kbit/s VA_10 MODE: synchronous USER_RATE: 19,2 kbit/s VA_11 MODE: synchronous USER_RATE: 32 kbit/s VA_12 MODE: synchronous USER_RATE: 48 kbit/s VA_13 MODE: synchronous USER_RATE: 56,0 kbit/s VA_14 MODE: synchronous USER_RATE: 64 kbit/s VA_15 MODE: asynchronous USER_RATE: 1,2 kbit/s VA_16 MODE: asynchronous USER_RATE: 2,4 kbit/s VA_17 MODE: asynchronous USER_RATE: 3,6 kbit/s VA_18 MODE: asynchronous USER_RATE: 4,8 kbit/s VA_19 MODE: asynchronous USER_RATE: 7,2 kbit/s VA_20 MODE: asynchronous USER_RATE: 8 kbit/s VA_21 MODE: asynchronous USER_RATE: 9,6 kbit/s VA_22 MODE: asynchronous USER_RATE: 14,4 kbit/s VA_23 MODE: synchronous USER_RATE: 16 kbit/s VA_24 MODE: asynchronous USER_RATE: 19,2 kbit/s VA_25 MODE: asynchronous USER_RATE: 32 kbit/s VA_26 MODE: asynchronous USER_RATE: 48 kbit/s VA_27 MODE: asynchronous USER_RATE: 56 kbit/s VA_28 MODE: asynchronous USER_RATE: 64 kbit/s ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 55 Table 3: PIXIT Values for the test purpose IS_AU_19 to IS_AU_21 m= line b= line a= line VA <media> <transport> <fmt-list> <modifier>:<bandwidth-value> (see note) rtpmap: <payload type> <encoding name>/ <clock rate> [/<encoding parameters>] VA_01 Audio RTP/AVP 0 N/A or up to 64 kbit/s N/A 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 VA_04 Audio RTP/AVP Dynamic PT N/A or up to 64 kbit/s rtpmap:<dynamic-PT> PCMA/8000 VA_05 Image Udptl t38 N/A or up to 64 kbit/s Based on T.38 VA_06 Image Tcptl t38 N/A or up to 64 kbit/s Based on T.38 NOTE: <bandwidth value> for <modifier> of AS is evaluated to be B kbit/s. IS_XX_22 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 4.5.18 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ETSI TS 129 163 [i.20] clause A.1.4 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; announcement towards a PSTN/ISDN Providing announcements to a user during the establishment of a communication session SIP selection criteria: Audio Test purpose: Ensure that an announcement towards a PSTN/ISDN can be provided. During the establishment an AS in the IP network provides an announcement e.g. "The communication is forwarded" or "The user is not reachable". The announcement should be received after the CALL PROCEEDING message with PI#8. ISDN Parameter values: BC= PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 56 IS_AU_23 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 4.5.17 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1 ETSI EN 383 001 [49], clause 7.1 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic_call; Telefax G3 terminals-inband Test purpose: Support of Telefax G3. Ensure that in the active call state (N10) the Fax transfer on the media and B-channels is performed correctly and the echo cancellers in the GW are not activated. In the active call state the callee sends a re-INVITE with the T38. ISDN Parameter values: BC=3,1 kHz audio, HLC = Facsimile G2/G3 SIP Parameter values: Dial string parameters options=PIXIT a = line RTP/AVP b = line 64 kbit/s m = line: 8 re-INVITE a = line Based on T.38. b = line AS: 64 m = line: udptl; T38 Comments: ISDN SUT SIP Case a) SETUP INVITE ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation INVITE 200 OK INVITE ACK DISC BYE REL 200 OK BYE Case c) Supported: 100 rel and precondition SETUP SETUP ACK INFO INVITE 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK INVITE 200 OK INVITE ACK DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 57
f72e5b464f10bf0050bab19795493770	186 001-1	6.2.1.2 Codec negotiation	IS_CN_01 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1.1 ETSI EN 383 001 [49], clause 7.1.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.1 IETF RFC 3264 [30] TSS reference: ISDN-SIP/Basic_call/Codec negotiation Selection criteria: Basic_call Test purpose: Ensure that the call establishment is performed correctly. The answer related to the SDP offer is contained in the 183 Session Progress message Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP a) SDP pre-condition not requested SETUP INVITE offer1 CALL PROC 183 Session Progress answer 1 ALERTING 180 Ringing CONNECT 200 OK INVITE ACK DISC BYE REL 200 OK BYE b) pre-condition and 100 rel SETUP INVITE offer 1 CALL PROC 183 Session Progress answer 1 PRACK 200 OK PRACK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK PRACK CONNECT 200 OK INVITE DISC ACK REL BYE 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 58 IS_CN_02 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1.1 ETSI EN 383 001 [49], clause 7.1.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.1 IETF RFC 3264 [30] TSS reference: ISDN-SIP/Basic_call/Codec negotiation Selection criteria: Basic_call Test purpose: Ensure that the call establishment is performed correctly. The answer related to the SDP offer is contained in the 180 Ringing message Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP a) SDP pre-condition not requested ISDN 1 UE 1 SETUP INVITE offer1 CALL PROC ALERTING 180 Ringing answer 1 CONNECT 200 OK INVITE ACK DISC BYE REL 200 OK BYE b) pre-condition and 100 rel SETUP INVITE offer1 CALL PROC 183 Session Progress PRACK 200 OK PRACK ALERTING 180 Ringing Answer 1 PRACK 200 OK PRACK CONNECT 200 OK INVITE DISC ACK REL BYE 200 OK BYE 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 59 IS_CN_03 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1.1 ETSI EN 383 001 [49], clause 7.1.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.1 IETF RFC 3264 [30] TSS reference: ISDN-SIP/Basic_call/Codec negotiation Selection criteria: Basic_call Test purpose: Ensure that the call establishment is performed correctly. The answer related to the SDP offer is contained in the 200 OK INVITE message. Ensure that in the active call state (N10) the voice transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP a) SDP pre-condition not requested SETUP INVITE offer1 CALL PROC ALERTING 180 Ringing CONNECT 200 OK INVITE answer 1 ACK DISC BYE REL 200 OK BYE b) pre-condition and 100 rel SETUP INVITE offer 1 CALL PROC 183 Session Progress PRACK 200 OK PRACK ALERTING 180 Ringing PRACK 200 OK PRACK CONNECT 200 OK INVITE answer 1 DISC ACK REL BYE 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 60 IS_CN_04 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1.1 ETSI EN 383 001 [49], clause 7.1.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.1 IETF RFC 3264 [30] TSS reference: ISDN-SIP/Basic_call/Codec negotiation Selection criteria: Basic_call Test purpose: Ensure that the call establishment is performed correctly. Ensure that answer related to the SDP offer is contained in the 183 Session Progress message. A new offer (codec) is sent in the 180 Ringing. Ensure that in the active call state (N10) the voice transfer on the media and B- channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP a) SDP pre-condition not requested SETUP INVITE Offer 1 CALL PROC 183 Session Progress Answer 1 ALERTING 180 Ringing offer 2 CONNECT 200 OK INVITE ACK answer 2 DISC BYE REL 200 OK BYE b) pre-condition and 100 rel Option a) SETUP INVITE Offer 1 CALL PROC 183 Session Progress Answer 1 PRACK 200 OK PRACK ALERTING 180 Ringing offer 2 PRACK 200 OK PRACK CONNECT 200 OK INVITE ACK answer 2 DISC BYE REL 200 OK BYE Option c) SETUP INVITE Offer 1 CALL PROC 183 Session Progress Answer 1 PRACK 200 OK PRACK ALERTING 180 Ringing offer 2 PRACK UPDATE answer 2 200 OK 200 OK PRACK CONNECT 200 OK INVITE ACK DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 61 IS_CN_05 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1.1 ETSI EN 383 001 [49], clause 7.1.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.1 IETF RFC 3264 [30] TSS reference: ISDN-SIP/Basic_call/Codec negotiation Selection criteria: Basic_call; RE-INVITE Test purpose: 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. ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP a) SDP pre-condition not requested SETUP INVITE Offer 1 CALL PROC 183 Session Progress ALERTING 180 Ringing CONNECT 200 OK INVITE ACK RE-INVITE offer 2 200 OK answer 2 ACK DISC BYE REL 200 OK BYE b) ETSI TS 124 229 [55] / ETSI ES 283 003 [42] (pre-condition and 100 rel ) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK PRACK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK PRACK CONNECT 200 OK INVITE ACK RE-INVITE offer 2 200 OK answer 2 ACK DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 62
f72e5b464f10bf0050bab19795493770	186 001-1	6.2.1.3 Test purposes for ISDN-SIP Basic call Successful - UPDATE	IS_XX_UP_01 NGN reference to: ETSI EN 383 001 [49], clause 7.1.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.1 IETF RFC 3311 [31], clause 5.1 TSS reference: ISDN -SIP/Basic_call/Successful Selection criteria: UPDATE procedure for the callee after INVITE transaction Test purpose: Ensure that the callee can send UPDATE after completion of the initial INVITE transaction. ISDN Parameter values: BC=PIXIT, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE Offer 1 CALL PROC 183 Session Progress answer 1 PRACK 200 OK PRACK ALERTING 180 Ringing CONNECT 200 OK INVITE ACK Conversation UPDATE offer 200 OK UPDATE Conversation DISC BYE REL 200 OK BYE IS_XX_UP_02 NGN reference to: ETSI EN 383 001 [49], clause 7.1.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.1 IETF RFC 3311 [31], clause 5.1 TSS reference: ISDN - SIP/Basic_call/Successful Selection criteria: Subsequent UPDATE is rejected if a pending offer (PRACK) is not answered Test purpose: Ensure that a subsequent UPDATE following an UPDATE is rejected (500 Server Internal Error) as long as the pending offer (2) is not answered. ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 63 Comments: ISDN SUT SIP SETUP INVITE with Offer 1 CALL PROC 183 Session Progress answer 1 PRACK 200 OK PRACK UPDATE offer 2 UPDATE offer 3 500 Server Internal Error 200 OK UPDATE answer 2 ALERTING 180 Ringing CONNECT 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE IS_XX_UP_03 NGN reference to: ETSI EN 383 001 [49], clause 7.1.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.1 IETF RFC 3311 [31], clause 5.1 TSS reference: ISDN - SIP/Basic_call/Successful Selection criteria: Received UPDATE is rejected if a pending offer (INVITE) is not answered Test purpose: Ensure that a subsequent UPDATE following an INVITE with SDP offer is rejected (500 Server Internal Error) as long as the first offer is not answered. ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE offer 1 CALL PROC UPDATE offer 2 500 Server Internal Error 183 Session Progress answer 1 PRACK 200 OK PRACK ALERTING 180 Ringing CONNECT 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 64 IS_XX_UP_04 NGN reference to: ETSI EN 383 001 [49], clause 7.1.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.1 IETF RFC 3311 [31], clause 5.1 TSS reference: ISDN - SIP/Basic_call/Successful Selection criteria: Execution of UPDATE procedure, SDP version identifier exists Test purpose: Ensure that an UPDATE procedure is executed (e.g. bandwidth parameter changed), if the SDP version identifier is different. ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE with offer 1 with SDP version identifier CALL PROC 183 Session Progress answer 1 PRACK 200 OK (PRACK) UPDATE with offer 2 different SDP version identifier 200 OK UPDATE answer 2 Session parameter changed ALERTING 180 Ringing CONNECT 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 65 IS_XX_UP_05 NGN reference to: ETSI EN 383 001 [49], clause 7.1.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.1 IETF RFC 3311 [31], clause 5.1 TSS reference: ISDN - SIP/Basic_call/Successful Selection criteria: Execution of UPDATE procedure, SDP version identifier exists Test purpose: Ensure that an UPDATE procedure is executed but bandwidth parameters are not changed, if the SDP version identifier and SDP content have not changed. ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE with offer 1 with SDP version identifier CALL PROC 183 Session Progress answer 1 PRACK 200 OK (PRACK) UPDATE with offer 2 identical SDP as offer 1 200 OK UPDATE answer 2 Session parameter are not changed ALERTING 180 Ringing CONNECT 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 66
f72e5b464f10bf0050bab19795493770	186 001-1	6.2.1.4 Test purposes for ISDN-SIP Basic call Successful - DTMF Tests	IS_XX_DT_01 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1.1 ETSI EN 383 001 [49], clause 7.1.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.1 TSS reference: ISDN-SIP/Basic_call/Successful/DTMF Selection criteria: Basic call; DTMF - Inband Test purpose: Ensure that the call establishment using en-bloc sending is performed correctly. Ensure that in the active call state (N10) the DTMF Digits (events 0 through 15) can be transmitted inband to the called user. ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP Case a) SETUP INVITE with offer 1 CALL PROC ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) Supported: 100 rel and precondition SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Values of codecs for test purposes IS_XX_DT_01 VARIABLE PT Encoding media type clock rate channels VA_01 0 PCMU A 8,000 1 VA_02 3 GSM A 8,000 1 VA_03 8 PCMA A 8,000 1 ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 67 IS_XX_DT_02 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1.1 ETSI EN 383 001 [49], clause 7.1.1 ETSI TS 129 163 [i.20], clause 7.2.3.2.1 TSS reference: ISDN - SIP/Basic_call/Successful/DTMF Selection criteria: Basic call; DTMF-RFC 2833 [56] Test purpose: Ensure that the call establishment using en-bloc sending is performed correctly. Ensure that in the active call state (N10) the DTMF Digits (events 0 through 15) can be transmitted as payload for DTMF Digits (IETF RFC 2833 [56]) to the called user. ISDN Parameter values: BC=speech, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP Case a) SETUP INVITE with offer 1 CALL PROC ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) Supported: 100 rel and precondition SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 68
f72e5b464f10bf0050bab19795493770	186 001-1	6.2.1.5 Test purposes for ISDN-SIP Basic call Successful -UDI	Successful UDI IS_UD_01 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1 ETSI EN 383 001 [49], clause 7.1 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN - SIP/Basic_call/Successful/UDI Selection criteria: Basic call; UDI Test purpose: Ensure that the call establishment using en-bloc sending is performed correctly. Ensure that the mapping of the SETUP parameters and the INVITE message parameters is performed correctly. Ensure that in the active call state (N10) the data transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC= UDI, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line : rtpmap:<dynamic-PT> CLEARMODE/8000 b = line AS: 64 m = RTP/AVP Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 69 IS_UD_02 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 ETSI EN 383 001 [49], clause 7.1 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Successful/UDI Selection criteria: Basic call; UDI; ITU-T Q.1912.5 [51] Profile A Test purpose: Ensure that call establishment using en-bloc sending is performed correctly. Ensure that the ISDN user in the state U3 receives an ALERTING message with the progress indicator information element "call is not end-to-end ISDN (#1)" location "Network beyond Interworking point" when the SIP user answers with 180 Ringing message. Ensure that in the active call state (N10) the data transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC= UDI, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line : rtpmap:<dynamic-PT> CLEARMODE/8000 b = line AS: 64 m = RTP/AVP Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING PI #1 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING PI #1 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 70 IS_UD _03 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic call; UDI; ITU-T Q.1912.5 [51] Profile B with PI Test purpose: Ensure that call establishment using en-bloc sending is performed correctly. Ensure that the ISDN user in the state U3 receives an ALERTING message with the progress indicator information element PI#1 or PI#2 or both location "Network beyond Interworking point" when the SIP user answers with 180 Ringing message. Ensure that in the active call state (N10) the data transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC= UDI, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line : rtpmap:<dynamic-PT> CLEARMODE/8000 b = line AS: 64 m = RTP/AVP Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING PI 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING PI# VA 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 71 IS_UD_04 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 ETSI EN 383 001 [49], clause 7.3 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Successful/UDI Selection criteria: Basic call; UDI; ETSI EN 383 001 [49] or ETSI TS 129 163 [i.20] Test purpose: Ensure that call establishment using en-bloc sending is performed correctly. Ensure that the ISDN user in the state U3 receives an ALERTING message without the progress indicator information element when the SIP user answers with 180 Ringing message. Ensure that in the active call state (N10) the data transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC= UDI, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line : rtpmap:<dynamic-PT> CLEARMODE/8000 b = line AS: 64 m = RTP/AVP Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 72 IS_UD_05 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 TSS reference: ISDN-SIP/Basic_call/Successful/UDI Selection criteria: Basic call; UDI; SIP Profile A or Profile B optional Test purpose: Ensure that call establishment using en-bloc sending is performed correctly. Ensure that the ISDN user in the state U3 receives a CONNECT message with the progress indicator information element "call is not end-to-end ISDN (#1)"location "Network beyond Interworking point" when the SIP user answers with 200 OK message. Ensure that in the active call state (N10) the data transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC= UDI, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line : rtpmap:<dynamic-PT> CLEARMODE/8000 b = line AS: 64 m = RTP/AVP Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC CONN PI# 1 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) Supported: 100 rel and precondition SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK CONN PI# 2 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 73 IS_UD_06 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic call; UDI; ITU-T Q.1912.5 [51] Profile B with PI Test purpose: Ensure that call establishment using en-bloc sending is performed correctly. Ensure that the ISDN user in the state U3 receives an CONNECT message with the progress indicator information element PI#1 or PI#2 or both location "Network beyond Interworking point" when the SIP user answers with 180 Ringing message. Ensure that in the active call state (N10) the data transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC= UDI, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line : rtpmap:<dynamic-PT> CLEARMODE/8000 b = line AS: 64 m = RTP/AVP Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC CONN PI 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) Supported: 100 rel and precondition SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK CONN PI# VA 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 74 IS_UD_07 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.3 ETSI EN 383 001 [49], clause 7.1 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Successful/UDI Selection criteria: Basic call; UDI; ETSI EN 383 001 [49] or ETSI TS 129 163 [i.20] Test purpose: Ensure that call establishment using en-bloc sending is performed correctly. Ensure that the ISDN user in the state U3 receives a CONNECT message without a progress indicator information element when the SIP user answers with 200 OK message. Ensure that in the active call state (N10) the data transfer on the media and B-channels is performed correctly (e.g. testing QoS parameters). ISDN Parameter values: BC= UDI, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line : rtpmap:<dynamic-PT> CLEARMODE/8000 b = line AS: 64 m = RTP/AVP Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 75 IS_UD_08 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.3.3 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7 ETSI EN 383 001 [49], clause 7.7 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Successful/UDI Selection criteria: Basic call; UDI Test purpose: Ensure that the call establishment and the call clearing procedure is performed correctly when the calling user clears after answering with a DISCONNECT message indicating the Cause value # 16 "normal call clearing". The called user shall receive a BYE message. The Reason header shall contain Cause Value #16 in the case of ETSI EN 383 001 [49] and ETSI TS 129 163 [i.20]. ISDN Parameter values: BC= UDI, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line : rtpmap:<dynamic-PT> CLEARMODE/8000 b = line AS: 64 m = RTP/AVP Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing PRACK 200 OK CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 76 IS_UD_09 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.3.3 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7 ETSI EN 383 001 [49], clause 7.7 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Successful/UDI Selection criteria: Basic call; UDI; ETSI EN 383 001 [49] or ETSI TS 129 163 [i.20] and ITU optional Test purpose: Ensure that the call clearing procedure is performed correctly when the called user clears after answering with a BYE message. The calling user shall receive a DISCONNECT message with the Cause value # 16 "normal call clearing". ISDN Parameter values: BC= UDI, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line : rtpmap:<dynamic-PT> CLEARMODE/8000 b = line AS: 64 m = RTP/AVP Comments: ISDN SUT SIP Case a) SETUP INVITE CALL PROC ALERTING 180 Ringing CONN 200 OK INVITE ACK Conversation DISC BYE REL 200 OK BYE Case c) SETUP INVITE CALL PROC 183 Session Progress PRACK 200 OK UPDATE 200 OK ALERTING 180 Ringing CONN 200 OK INVITE PRACK 200 OK ACK Conversation DISC BYE REL 200 OK BYE ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 77
f72e5b464f10bf0050bab19795493770	186 001-1	6.2.1.6 Test purposes for ISDN-SIP Basic call Unsuccessful	Unsuccessful IS_XX_U01 ISDN reference to: ETSI EN 300 403-1 [i.3], clauses 5.2.5.1, G.1.7 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6 ETSI EN 383 001 [49], clause 7.7.6 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Unsuccessful Selection criteria: Basic call; Test purpose: Ensure that, when the called user is busy and responds with a 486 Busy Here message the circuit switched side is initiating call clearing with a DISCONNECT or RELEASE message indicating cause value #17 "user busy". ISDN Parameter values: BC = PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: The originating exchange sends a DISCONNECT message to the calling user with progress indicator #8 thus indicating that in-band information is available. Normal release procedure applies after the in-band information has been connected. The calling user shall receive in the disconnect indication state (N12) the in-band tone/announcement on the B-channel. ISDN SUT SIP SETUP INVITE DISC 486 Busy Here REL ACK RLC ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 78 IS_XX_U02 ISDN reference to: ETSI EN 300 403-1 [i.3], clauses 5.2.5.1, G.1.7 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6 TSS reference: ISDN-SIP/Basic_call/Unsuccessful Selection criteria: Basic call; Test purpose: Ensure that, when the called user is busy and the PROXY responds with a 486 Busy Here (NDUB). The circuit switched side is initiating call clearing with a DISCONNECT or RELEASE message indicating cause value #17 "user busy". ISDN Parameter values: BC = PIXIT SIP Parameter values: PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: The originating exchange sends a DISCONNECT message to the calling user with progress indicator #8 thus indicating that in-band information is available. Normal release procedure applies after the in-band information has been connected. The calling user shall receive in the disconnect indication state (N12) the in-band tone/announcement on the B-channel. ISDN SUT SIP SETUP DISC REL RLC IS_XX_U03 ISDN reference to: ETSI EN 300 403-1 [i.3], clauses 5.2.5.4, G.1.9 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6 ETSI EN 383 001 [49], clause 7.7.6 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Unsuccessful/ Selection criteria: Basic call; Reason Header field is supported Test purpose: Ensure that when there is no answer from the called user (but user alerted), the ISDN network initiate call clearing to the calling user with a DISCONNECT message indicating cause value #19 "no answer from user (user alerted)" and sends to the called user a CANCEL or BYE message indicating cause # 102 "recovery on timer expire" in the Reason Header field. ISDN Parameter values: BC = PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE CALL PROCEEDING . … . ALERTING 180 Ringing DISC#19 CANCEL 200 OK 487 Request terminated ACK ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 79 IS_XX_U04 ISDN reference to: ETSI EN 300 403-1 [i.3], clauses 5.2.5.4, G.1.9 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6 ETSI EN 383 001 [49], clause 7.7.6 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Unsuccessful/ Selection criteria: Basic call; Reason Header field is supported Test purpose: Ensure that when there is no answer from the called user (but user alerted) and if the SIP network initiate call clearing before the SCN release the call, the SIP network shall send to the calling user a 480 Temporarily unavailable message and the SCN network initiate call clearing to the calling user with a DISCONNECT message indicating cause value # 20 Subscriber absent. ISDN Parameter values: BC = PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE CALL PROCEEDING . . . ALERTING 180 Ringing DISC#20 480 temp. Unavailable REL ACK IS_XX_U05 ISDN reference to: ETSI EN 300 403-1 [i.3], clauses 5.2.5.4, G.1.9 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6 ETSI EN 383 001 [49], clause 7.7.6 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Unsuccessful/ Selection criteria: Basic call; Reason Header field is not supported Test purpose: Ensure that when there is no answer from the called user (but user alerted), the ISDN network initiate call clearing to the calling user with a DISCONNECT message indicating cause value #19 "no answer from user (user alerted)" and sends to the called user a CANCEL or BYE. ISDN Parameter values: BC = PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE CALL PROCEEDING … ALERTING 180 Ringing DISC#19 CANCEL 200 OK ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 80 IS_XX_U06 ISDN reference to: ETSI EN 300 403-1 [i.3], clauses 5.2.5.4, G.1.9 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6 ETSI EN 383 001 [49], clause 7.7.6 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Unsuccessful/ Selection criteria: Basic call; Reason Header field is not supported Test purpose: Ensure that when there is no answer from the called user (but user alerted) and if the SIP network initiate call clearing before the SCN release the call, the SIP network shall send to the calling user a 480 Temporarily unavailable message and the SCN network initiate call clearing to the calling user with a DISCONNECT message indicating cause value # 20 Subscriber absent. ISDN Parameter values: BC = PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE CALL PROCEEDING . . . ALERTING 180 Ringing DISC#20 480 temp. Unavailable REL ACK IS_XX_U07 ISDN reference to: ETSI EN 300 403-1 [i.3], clauses 5.1.9, 5.3.2, G.1.10 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6 ETSI EN 383 001 [49], clause 7.7.6 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Unsuccessful/ Selection criteria: Basic call Test purpose: Ensure that when the called side rejects the call and responds with a 603 Decline message containing the Cause information element indicating the cause value #21 "call reject". The circuit switched network initiates call clearing to the calling user with a DISCONNECT or RELEASE message indicating cause value # 21"call reject". ISDN Parameter values: BC = PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE CALL PROCEEDING . DISC#21 603 Decline Unavailable ACK ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 81 IS_XX_U08 ISDN reference to: ETSI EN 300 403-1 [i.3], clauses 5.1.9, 5.3.2, G.1.10 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6 ETSI EN 383 001 [49], clause 7.7.6 ETSI TS 129 163 [i.20], clause 7.2.3.2 TSS reference: ISDN-SIP/Basic_call/Unsuccessful Selection criteria: Basic call; Test purpose: Ensure that the call will be released when the called number is incomplete. The circuit switched network initiates call clearing to the calling user with a DISCONNECT or RELEASE COMPLETE message with a cause value # 28. ISDN Parameter values: BC = PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: In some networks tones or announcement can be generated in the destination exchange (or intermediate exchange) during call establishment. The originating exchange sends a DISCONNECT message to the calling user with progress indicator #8 thus indicating that in-band information is available. Normal release procedure applies after the in-band information has been connected. The calling user shall receive in the disconnect indication state (N12) the in-band tone/announcement. ISDN SUT SIP-S-CSCF SETUP INVITE CALL PROCEEDING DISC#28 484 Address Incomplete ACK IS_XX_U09 ISDN reference to: ETSI EN 300 403-1 [i.3], clause G.1.13 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6 ETSI EN 383 001 [49], clause 7.7.6 ETSI TS 129 163 [i.20], clause 7.2.3.2.12 TSS reference: ISDN-SIP/Basic_call/Unsuccessful Selection criteria: Basic call; Test purpose: Ensure that when the called party is not registered. The circuit switched network initiates call clearing to the calling user with a DISCONNECT or RELEASE COMPLETE message with a cause: # 20 "subscriber absent" ISDN Parameter values: BC = PIXIT SIP Parameter values: PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SETUP CALL PROCEEDING DISC#20 ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 82 IS_XX_U10 ISDN reference to: ETSI EN 300 403-1 [i.3], clauses 5.2.2, G.5.7 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6 ETSI EN 383 001 [49], clause 7.7.6 ETSI TS 129 163 [i.20], clause 7.2.3.2.12 TSS reference: ISDN-SIP/Basic_call/Unsuccessful Selection criteria: Basic call Test purpose: Ensure that when the called user is not compatible and responds with a 503 Service Unavailable, the circuit switched network initiates call clearing to the calling user with a DISCONNECT with Cause value # 127. ISDN Parameter values: BC = PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE CALL PROCEEDING . DISC#127 503 Service Unavailable ACK IS_XX_U11 ISDN reference to: ETSI EN 300 403-1 [i.3], clause G.1.6 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6 ETSI EN 383 001 [49], clause 7.7.6 ETSI TS 129 163 [i.20], clause 7.2.3.2.12 TSS reference: ISDN-SIP/Basic_call/Unsuccessful Selection criteria: Basic call; Test purpose: Ensure that when the calling user clears with cause value #16 "normal call clearing" before answer from called user, the network initiates call clearing to the called user with a CANCEL or BYE message. ISDN Parameter values: BC=PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE CALL PROCEEDING . ALERTING 180 Ringing DISC#16 CANCEL REL 200 OK CANCEL RLC 487 Request terminated ACK ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 83 IS_XX_U012 ISDN reference to: ETSI EN 300 403-1 [i.3], clause G.1.6 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6 ETSI EN 383 001 [49], clause 7.7.6 ETSI TS 129 163 [i.20], clause 7.2.3.2.12 TSS reference: ISDN-SIP/Basic_call/Unsuccessful Selection criteria: Basic call; Test purpose: Ensure that the call will be released when the number is changed, the circuit switched network initiates call clearing to the calling user with a DISCONNECT with Cause value # 22 number changed. ISDN Parameter values: BC=PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SETUP CALL PROCEEDING DISC#22 REL IS_XX_U13 ISDN reference to: ETSI EN 300 403-1 [i.3], clause G.1.6 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6, ETSI EN 383 001 [49], clause 7.7.6 ETSI TS 129 163 [i.20], clause 7.2.3.2.12 IETF RFC 3261 [28] IETF RFC 4566 [25] TSS reference: ISDN-SIP/Basic_call/Unsuccessful Selection criteria: Basic call Test purpose: Ensure that when there is no answer from the called user (there is no response from INVITE messages), the network initiate call clearing to the calling user with a DISCONNECT message indicating cause value # 20 "Subscriber absent". ISDN Parameter values: BC=PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE CALL PROCEEDING INVITE INVITE INVITE INVITE INVITE DISC#20 INVITE REL RLC ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 84 IS_XX_U14 ISDN reference to: ETSI EN 300 403-1 [i.3], clause G.1.6 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6, ETSI EN 383 001 [49], clause 7.7.6 ETSI TS 129 163 [i.20], clause 7.2.3.2.12 IETF RFC 3261 [28] IETF RFC 4566 [25] TSS reference: ISDN-SIP/Basic_call/Unsuccessful Selection criteria: Basic call; Reason Header field is supported Test purpose: Ensure that the SUT if the SIP Failure response is interworked on receipt of a Failure message 4XX defined as SIP_Failure_VA. Sends a DISC or RELEASE message. The Cause Value in the header field set to CV_SIP is mapped to the ISDNCause Value field in the ISDNREL message with the Cause value set to CV_ ISDN. ISDN Parameter values: BC=PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE CALL PROCEEDING DISC#CV_ISDN SIP_Failure_VA REL ACK RLC Values for test purposes IS_XX_U14 ←REL (Cause Value) CV_ ISDN ←4XX/5XX/6XX SIP message SIP_Failure_VA VA_01 CV_ ISDN 415 Unsupported Media type CV_SIP (PIXIT) VA_02 CV_ ISDN 420 Bad Extension CV_SIP (PIXIT) VA_03 CV_ ISDN 421 Extension required CV_SIP (PIXIT) CV_SIP = CV_ISDN ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 85 IS_XX_U15 ISDN reference to: ETSI EN 300 403-1 [i.3], clause G.1.6 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7.6, ETSI EN 383 001 [49], clause 7.7.6 ETSI TS 129 163 [i.20], clause 7.2.3.2.12 IETF RFC 3261 [28] IETF RFC 4566 [25] TSS reference: ISDN-SIP/Basic_call/Unsuccessful Selection criteria: Basic call; Reason Header field is not supported Test purpose: Ensure that the SUT if the SIP Failure response is interworked on receipt of a Failure message 4XX defined as SIP_Failure_VA sends a DISC with Cause Value 127. ISDN Parameter values: BC=PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SETUP INVITE CALL PROCEEDING DISC#127 SIP_Failure_VA REL ACK RLC Values for test purposes IS_XX_U15 ←REL (Cause Value) ←4XX/5XX/6XX SIP message SIP_Failure_VA VA_01 127 415 Unsupported Media type VA_02 127 420 Bad Extension VA_03 127 421 Extension required IS_UD_U16 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.3.3 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7 ETSI EN 383 001 [49], clause 7.7 ETSI TS 129 163 [i.20], clause 7.2.3.2.12 TSS reference: ISDN-SIP/Basic_call/Unsuccessful Selection criteria: Basic call; SIP Network does not support UDI Test purpose: Ensure that when the SIP Network is not supporting UDI, the network initiate call clearing to the calling user with a DISCONNECT message indicating cause value # 65 "Bearer capability not implemented" ISDN Parameter values: BC= UDI, no HLC SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SETUP INVITE CALL PROCEEDING ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 86 DISC#65 REL RLC IS_AU_U17 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.3.3 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7 ETSI EN 383 001 [49], clause 7.7 ETSI TS 129 163 [i.20], clause 7.2.3.2.12 TSS reference: ISDN-SIP/Basic_call/Unsuccessful Selection criteria: Basic call; SIP Network does not support Teleservice FAX G3 Test purpose: Ensure that when the SIP Network is not supporting the Teleservice Fax G3, the network initiate call clearing to the calling user with a DISCONNECT message indicating cause value # 79 "Service or option not implemented" ISDN Parameter values: BC=3,1 kHz audio, HLC= Facsimile G2/G3 SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SETUP INVITE CALL PROCEEDING DISC#79 REL RLC IS_XX_U18 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.3.3 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.7 ETSI EN 383 001 [49], clause 7.7 ETSI TS 129 163 [i.20], clause 7.2.3.2.12 TSS reference: ISDN-SIP /Basic_call/Unsuccessful Selection criteria: Basic call Test purpose: During the session, the called user decide 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. On the ISDN side the user is still in the active state. ISDN Parameter values: BC=PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 87 IS_XX_U19 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1.1 ETSI EN 383 001 [49], clause 7.1.1 ETSI TS 129 163 [i.20], clause 7.2.3.2 IETF RFC 3264 [30], clause 6 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic call Test purpose: Ensure that answer related to the SDP offer is contained in the 180 Ringing message. The media stream is rejected (port number is set to zero). Ensure that the call is rejected by sending a CANCEL or BYE. ISDN Parameter values: BC=PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE CALL PROCEEDING . ALERTING 180 Ringing answer 1 Case a) DISC CANCEL REL 200 OK CANCEL RLC 487 Request terminated ACK Case b) DISC BYE REL 200 OK RLC 487 Request terminated ACK ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 88 IS_XX_U20 ISDN reference to: ETSI EN 300 403-1 [i.3], clause 5.1.5.1 ETSI EN 300 899-1 [23], clause 2.1.1 NGN reference to: ITU-T Q.1912.5 [51], clause 7.1.1 IETF RFC 3264 [30], clause 6 TSS reference: ISDN-SIP/Basic_call/Successful/Voice Selection criteria: Basic call Test purpose: Ensure that answer related to the SDP offer is contained in the 200 OK INVITE message. The media stream is rejected (port number is set to zero). Ensure that the call is rejected by sending a BYE. ISDN Parameter values: BC=PIXIT SIP Parameter values: Dial string parameters options=PIXIT PIXIT for supported header: Case a) no 100 rel Case b) Supported: 100 rel Case c) Supported: 100 rel and precondition a = line (PIXIT) b = line (PIXIT) m = line (PIXIT) Comments: ISDN SUT SIP SETUP INVITE offer 1 CALL PROCEEDING . 200 OK INVITE answer 1 ACK DISC BYE REL 200 OK RLC 487 Request terminated ACK ETSI ETSI TS 186 001-1 V3.1.1 (2015-11) 89
f72e5b464f10bf0050bab19795493770	186 001-1	6.2.2 Test purposes for ISDN-SIP Supplementary services

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