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本文(ETSI TS 186 001-3-2010 IMS Network Testing (INT) Network Integration Testing Part 3 Test Suite Structure and Test Purposes (TSS&TP) for SIP-SIP (V2 2 1)《IMS网络测试(INT) 网络综合测试 第3部分 SI_1.pdf)为本站会员(postpastor181)主动上传,麦多课文库仅提供信息存储空间,仅对用户上传内容的表现方式做保护处理,对上载内容本身不做任何修改或编辑。 若此文所含内容侵犯了您的版权或隐私,请立即通知麦多课文库(发送邮件至master@mydoc123.com或直接QQ联系客服),我们立即给予删除!

ETSI TS 186 001-3-2010 IMS Network Testing (INT) Network Integration Testing Part 3 Test Suite Structure and Test Purposes (TSS&TP) for SIP-SIP (V2 2 1)《IMS网络测试(INT) 网络综合测试 第3部分 SI_1.pdf

1、 ETSI TS 186 001-3 V2.2.1 (2010-11)Technical Specification IMS Network Testing (INT);Network Integration Testing;Part 3: Test Suite Structure andTest Purposes (TSS Essential, or potentially Essential, IPRs notified to ETSI in respect of ETSI standards“, which is available from the ETSI Secretariat.

2、Latest updates are available on the ETSI Web server (http:/webapp.etsi.org/IPR/home.asp). Pursuant to the ETSI IPR Policy, no investigation, including IPR searches, has been carried out by ETSI. No guarantee can be given as to the existence of other IPRs not referenced in ETSI SR 000 314 (or the upd

3、ates on the ETSI Web server) which are, or may be, or may become, essential to the present document. Foreword This Technical Specification (TS) has been produced by ETSI Technical Committee IMS Network Testing (INT). The present document is part 3 of a multi-part deliverable covering Network Integra

4、tion Testing, as identified below: Part 1: “Test Suite Structure and Test Purposes (TSS Part 2: “Abstract Test Suite (ATS) and partial Protocol Implementation eXtra Information for Testing (PIXIT) proforma specification“; Part 3: “Test Suite Structure and Test Purposes (TSS Part 4: “Abstract Test Su

5、ite (ATS) and partial Protocol Implementation eXtra Information for Testing (PIXIT) proforma specification“. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11)61 Scope The present document specifies the Test Suite Structure and Test Purposes (TSS IP Multimedia Call Control Protocol based on Session Initiation

6、Protocol (SIP) and Session Description Protocol (SDP) Stage 3 3GPP TS 24.229 Release 7, modified“. 2 ETSI TS 124 503 (V8.5.0): “Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; TISPAN; IP Multimedia Call Control Protocol based on Session

7、Initiation Protocol (SIP) and Session Description Protocol (SDP) Stage 3 3GPP TS 24.229 (Release 7), modified (3GPP TS 24.503 version 8.5.0 Release 8)“. 3 IETF RFC 3261 (2002): “SIP: Session Initiation Protocol“. 4 ISO/IEC 9646-1 (1994): “Information technology - Open Systems Interconnection - Confo

8、rmance testing methodology and framework - Part 1: General concepts“. 5 ISO/IEC 9646-2 (1994): “Information technology - Open Systems Interconnection - Conformance testing methodology and framework - Part 2: Abstract Test Suite specification“. 6 ISO/IEC 9646-3 (1998): “Information technology - Open

9、Systems Interconnection - Conformance testing methodology and framework - Part 3: The Tree and Tabular Combined Notation (TTCN)“. 7 Void. 8 ISO/IEC 9646-5 (1994): “Information technology - Open Systems Interconnection - Conformance testing methodology and framework - Part 5: Requirements on test lab

10、oratories and clients for the conformance assessment process“. 9 ISO/IEC 9646-7 (1995): “Information technology - Open Systems Interconnection - Conformance testing methodology and framework - Part 7: Implementation Conformance Statements“. 10 ETSI TS 124 229 (V7.15.0): “Digital cellular telecommuni

11、cations system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; Internet Protocol (IP) multimedia call control protocol based on Session Initiation Protocol (SIP) and Session Description Protocol (SDP); Stage 3 (3GPP TS 24.229 version 7.15.0 Release 7)“. 11 Void. ETSI ETSI TS 186

12、001-3 V2.2.1 (2010-11)712 ETSI TS 124 504 (V8.5.0): “Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; TISPAN; PSTN/ISDN simulation services: Communication Diversion (CDIV); Protocol specification (3GPP TS 24.504 version 8.5.0 Release 8)“.

13、 13 Void. 14 ETSI TS 124 407 (V7.0.0): “Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); TISPAN; PSTN/ISDN simulation services; Originating Identification Presentation (OIP) and Originating Identification Restriction (OIR); Protocol specificat

14、ion (3GPP TS 24.407 version 7.0.0 Release 7)“. 15 ETSI TS 124 410 (V7.0.0): “Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); TISPAN; NGN Signalling Control Protocol; Communication HOLD (HOLD) PSTN/ISDN simulation services; Protocol specificat

15、ion (3GPP TS 24.410 version 7.0.0 Release 7)“. 16 IETF RFC 4566 (2006): “SDP: Session Description Protocol“. 17 IETF RFC 3312 (2002): “Integration of Resource Management and Session Initiation Protocol (SIP)“. 18 IETF RFC 3311 (2002): “The Session Initiation Protocol (SIP) UPDATE Method“. 19 ETSI TS

16、 124 147 (V8.2.0): “Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; Conferencing using the IP Multimedia (IM) Core Network (CN) subsystem; Stage 3 (3GPP TS 24.147 version 8.2.0 Release 8)“. 20 Void. 21 ETSI TS 124 615 (V8.2.0): “Digital

17、cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; Communication Waiting (CW) using IP Multimedia (IM) Core Network (CN) subsystem; Protocol Specfication (3GPP TS 24.615 version 8.2.0 Release 8)“. 22 ETSI TS 124 642 (V8.2.0): “Digital cellular tele

18、communications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; Completion of Communications to Busy Subscriber (CCBS) and Completion of Communications by No Reply (CCNR) using IP Multimedia (IM) Core Network (CN) subsystem; Protocol Specification (3GPP TS 24.642 version 8.

19、2.0 Release 8)“. 23 ETSI TS 124 529 (V8.1.0): “Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); LTE; TISPAN; PSTN/ISDN simulation services: Explicit Communication Transfer (ECT); Protocol specification (3GPP TS 24.529 version 8.1.0 Release 8)“

20、. 24 ETSI TS 124 508 (V8.1.0): “Digital cellular telecommunications system (Phase 2+); Universal Mobile Telecommunications System (UMTS); PSTN/ISDN simulation services Terminating Identification Presentation (TIP) and Terminating Identification Restriction (TIR); Protocol specification (3GPP TS 24.5

21、08 version 8.1.0 Release 8)“. 25 IETF RFC 5366: “Conference Establishment Using Request-Contained Lists in the Session Initiation Protocol (SIP)“. 2.2 Informative references The following referenced documents are not necessary for the application of the present document but they assist the user with

22、 regard to a particular subject area. Not applicable. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11)83 Definitions and abbreviations 3.1 Definitions For the purposes of the present document, the following terms and definitions apply: For SIP and SDP specific terminology, reference shall be made to RFC 3261

23、 3 and RFC 4566 16 respectively. SIP precondition: Indicates the support of the SIP “precondition procedure“ as defined in RFC 3312 17. 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 Tes

24、t Suite in TTCN). See ISO 9646 (all parts) 4 to 9. The test purposes have been defined from the users viewpoint and the abbreviation “UE“ is used in the description. However, the detailed comments section uses the abbreviation “UA“ for test system instances of the users. 3.2 Conventions for represen

25、tation of SIP/SDP information 1) All letters of SIP method names are capitalised. 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. EXAMP

26、LE 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/RFC 3261 3), in plain text without surrounding angle brackets. EXAMPLE

27、3: Request-URI, the userinfo 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 followe

28、d 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.

29、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

30、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 and components of the m= line. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11)93.3 Abbreviations For the purposes of the present docu

31、ment, the following abbreviations apply: ABNF Augmented Backus-Naur Form ATS Abstract Test Suite CCBS Completion of Communications to Busy Subscriber CCNR Completion of Communications by No Reply CD Communication Deflection CDIV Communication DIVersionCDIVN Communication DIVersion Notification CFB C

32、ommunication Forwarding Busy CFNL Communication Forwarding on Not Logged-in CFNR Communication Forwarding No Replay CFNRc Communication Forwarding on subscriber Not Reachable CFU Communication Forwarding Unconditional CONF CONFerence CW Call WaitingECT Explicit Communication Transfer HOLD communicat

33、ion HOLD IUT Implementation Under Test NDUB Network Determined User Busy OIP Originating Identification Presentation OIR Originating Identification Restriction PIXIT Protocol Implementation eXtra Information for Testing SDP Session Description Protocol SIP Session Initiation Protocol SUT System Unde

34、r Test TIP Terminating Identification Presentation TIR Terminating Identification Restriction TP Test Purpose TSI Test System Interface TSS Test Suite Structure TTCN Test and Test Control Notation UA User AgentUDUB User Determined User Busy UE User Equipment ETSI ETSI TS 186 001-3 V2.2.1 (2010-11)10

35、4 Test Suite Structure (TSS) 4.1 SIP-SIP C - Plane / U - Plane Basic_Call Successful Normal call establishment SS_XX_xx Codec negotiation SS_CN_xx UPDATE SS_XX_UP_xx Unsuccessful SS_XX_Uxx Supplementary_Services OIP SS_XXSS_OIPxx OIR SS_XXSS_OIRxx TIP SS_XXSS_TIPxx TIR SS_XXSS_TIRxx HOLD SS_XXSS_CHx

36、x CDIV CFU SS_XXSS_CFUxx CFB SS_XXSS_CFBxx CFNR SS_XXSS_CFNRxx CFNRc SS_XXSS_CFNRcxx CFNL SS_XXSS_CFNLxx CD SS_XXSS_CDxx CONF CONF_CRE SS_XXSS_CONF_CRExx CONF_IN SS_XXSS_CONF_INVxx CONF_LEAV SS_XXSS_CONF_LEAVxx CONF_REMOV SS_XXSS_CONF_REMOVxx CW SS_XXSS_CWxx CCBS SS_XXSS_CCBSxx CCNR SS_XXSS_CCNRxx E

37、CT SS_XXSS_ECT 5 Numbering Scheme 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 (us

38、ed e.g. for C-Subscriber in successful A to B Calls) (underscore makes it easier to read the name) P: PSTN I: ISDN ETSI ETSI TS 186 001-3 V2.2.1 (2010-11)11S: SIP (Extensions will be added when needed) Pos. 6 and 7: Bearer- or Teleservice involved XX: Defined per PIXIT value NOTE: TSIs may be approp

39、riate 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 DT: DTMF UP: UPD

40、ATE Method Pos. 8 and 9: _: No Supplementary Services Involved / Successful _U: No Supplementary Services Involved / Unsuccessful SS: Supplementary Services Involved 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 5.3 Supplementary Services CLIP IS_XXSSCLIP XX 1 2 3 4 5

41、6 7 8 9 10 11 12 13 14 15 I S _ _ _ X X S S C L I P x X 6 Test purposes The registration and application usage procedures in the ATS shall be compliant to RFC 3261 3 and ES 283 003 1 (modified TS 124 229 10 and TS 124 503 2). The validation of the registration procedure is out of scope of the presen

42、t document and will be part of the preambles used in the abstract test cases. ETSI ETSI TS 186 001-3 V2.2.1 (2010-11)126.1 Test purposes for Basic Call 6.1.1 Test purposes for SIP-SIP, Basic call, Successful 6.1.1.1 Normal call establishment SS_XX_01 NGN reference to: RFC 3261 3 TS 124 229 10, claus

43、es 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 p

44、arameters 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 pr

45、econdition 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 barb4right barb4right INVITE 180 Ringing barb4left barb4left 180 Ringing 200 OK INVITE barb4left barb4left 200 OK INVITE ACK barb4right barb

46、4right ACK Check media BYE barb4left barb4left BYE 200 OK BYE barb4right barb4right 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

47、 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

48、 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 wit

49、h 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)13Comments: SIP UA A SUT SIP UA B INVITE barb4right barb4right INVITE Start resource negotiation/reservation 183 Session Progress barb4left barb4left 183 Session Progress PRACK barb4right barb4right PRACK 200 OK PRACK barb4left barb4left 200 OK PRACK UPDATE barb4right barb4right UPDATE1 200 OK UPDATE1 barb4left b

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