ETSI TS 186 001-1 V1.0.0 (2008-05)

Telecommunications and Internet Converged Services and Protocols for Advanced Networking (TISPAN); Network Integration Testing between SIP and ISDN/PSTN network signalling protocols; Part 1: Test Suite Structure and Test Purposes (TSS&TP) for SIP-ISDN

DTS/TISPAN-06012-1-NGN

General Information

• Status: Published
• Publication Date: 26-May-2008
• Technical Committee: TISPAN 6 - TISPAN Testing
• Current Stage: 12 - Completion
• Due Date: 06-Jun-2008
• Completion Date: 27-May-2008

Standards Content (Sample)

Technical Specification
Telecommunications and Internet Converged Services and
Protocols for Advanced Networking (TISPAN);
Network Integration Testing between SIP and ISDN/PSTN
network signalling protocols
Part 1: Test Suite Structure and
Test Purposes (TSS&TP) for SIP-ISDN

2 ETSI TS 186 001-1 V1.0.0 (2008-05)

Reference
DTS/TISPAN-06012-1-NGN
Keywords
SIP, ISDN, IMS, NIT
ETSI
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ETSI
3 ETSI TS 186 001-1 V1.0.0 (2008-05)
Contents
Intellectual Property Rights.5
Foreword.5
1 Scope.6
2 References.6
2.1 Normative references.7
2.2 Informative references.9
3 Definitions and abbreviations.10
3.1 Definitions.10
3.1.1 Conventions for representation of SIP/SDP information.10
3.2 Abbreviations.11
4 Test Suite Structure (TSS).12
4.1 Test Suite Structure (TSS).12
4.1.1 ISDN-SIP.12
4.1.2 SIP-ISDN.13
4.1.3 PSTN-SIP.13
4.1.4 SIP-PSTN.14
5 Numbering Scheme.14
5.1 General description.14
5.2 Basic Call.15
5.3 Supplementary Services .15
6 Test purposes.15
6.1 ISDN - SIP .17
6.1.1 Basic Call.17
6.1.1.1 Test purposes for ISDN-SIP Basic call Successful - Speech or 3,1 kHz audio.17
6.1.1.2 Codec negotiation.42
6.1.1.3 Test purposes for ISDN-SIP Basic call Successful - UPDATE .47
6.1.1.4 Test purposes for ISDN-SIP Basic call Successful - DTMF Tests .52
6.1.1.5 Test purposes for ISDN-SIP Basic call Successful -UDI.53
6.1.1.6 Test purposes for ISDN-SIP Basic call Unsuccessful.62
6.1.2 Test purposes for ISDN-SIP Supplementary services.79
6.1.2.1 CLIP/OIP.79
6.1.2.2 CLIR/OIR.84
6.1.2.3 COLP/COLR (TIP/TIR).88
6.1.2.4 CFU.126
6.1.2.4.1 CFU - ISI.126
6.1.2.4.2 CFU - ISS.133
6.1.2.5 CFB.140
6.1.2.5.1 CFB - ISI .140
6.1.2.5.1 CFB - ISS .152
6.1.2.6 CFNR.164
6.1.2.6.1 CFNR - ISI .164
6.1.2.6.2 CFNR - ISS .185
6.1.2.7 CFNL.205
6.1.2.8 CD.216
6.1.2.8.1 Call Deflection-ISI.216
6.1.2.8.2 CD-ISS.228
6.1.2.9 3PTY.238
6.1.2.10 HOLD.245
6.1.2.11 CONF (Outgoing Call).250
6.2 Test purposes for SIP-ISDN.255
6.2.1 Basic Call.255
6.2.1.1 Test purposes for SIP-ISDN, Basic call, Successful 3,1 kHz audio.255
6.2.1.2 Codec negotiation.271
ETSI
4 ETSI TS 186 001-1 V1.0.0 (2008-05)
6.2.1.3 Test purposes for SIP-ISDN, Basic call, DTMF .272
6.2.1.4 Test purposes for SIP-ISDN, Basic call, UDI .274
6.2.1.5 Test purposes for SIP-ISDN, Basic call, Unsuccessful.278
6.2.2 Test purposes for SIP - ISDN Supplementary services .295
6.2.2.1 OIP/CLIP.295
6.2.2.2 OIR/CLIR .325
6.2.2.3 TIP/COLP.340
6.2.2.4 TIR/COLR.342
6.2.2.5 CFU.345
6.2.2.5.1 CFU - SIS.345
6.2.2.5.2 CFU - SII.357
6.2.2.6 CFB.369
6.2.2.6.1 CFB - SIS .369
6.2.2.6.2 CFB - SII .385
6.2.2.7 CFNR.404
6.2.2.7.1 CFNR - SIS .404
6.2.2.7.2 CFNR - SII .422
6.2.2.8 Call Deflection.440
6.2.2.8.1 CD - SIS.440
6.2.2.8.2 CD - SII.445
6.2.2.9 Three Party service 3PTY .450
6.2.2.10 TP.454
6.2.2.11 CUG.457
6.2.2.12 Hold.460
6.2.2.13 CONF.466
6.2.2.14 Call waiting (CW).474
6.2.2.15 Anonymous Call Rejection (ACR).474
6.3 Test purposes for PSTN - SIP .475
6.3.1 Basic call.475
6.3.1.1 Basic call Successful.475
6.3.1.2 Test purposes for PSTN - SIP Basic call Unsuccessful.478
6.3.2 Test purposes for PSTN - SIP Supplementary services .481
6.3.2.1 CLIP.481
6.3.2.2 CLIR.482
6.3.2.3 CFU.483
6.3.2.4 CFB.490
6.3.2.5 CFNR.495
6.3.2.6 CFNL.501
6.4 Test purposes for SIP-PSTN .504
6.4.1 Test purposes for SIP-PSTN, Basic call .504
6.4.1.1 Test purposes for SIP-PSTN, Basic call, Successful.504
6.4.1.2 Test purposes for SIP-PSTN, Basic call, Unsuccessful.506
6.4.2 Test purposes for SIP - PSTN Supplementary services .512
6.4.2.1 OIP/ CLIP.512
6.4.2.2 OIR/CLIR .550
6.4.2.3 CFU.564
6.4.2.4 CFB.567
6.4.2.5 CFNR.570
History .575

ETSI
5 ETSI TS 186 001-1 V1.0.0 (2008-05)
Intellectual Property Rights
IPRs essential or potentially essential to the present document may have been declared to ETSI. The information
pertaining to these essential IPRs, if any, is publicly available for ETSI members and non-members, and can be found
in ETSI SR 000 314: "Intellectual Property Rights (IPRs); Essential, or potentially Essential, IPRs notified to ETSI in
respect of ETSI standards", which is available from the ETSI Secretariat. 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 updates on the ETSI Web
server) wSIch are, or may be, or may become, essential to the present document.
Foreword
This Technical Specification (TS) has been produced by ETSI Technical Committee Telecommunications and Internet
converged Services and Protocols for Advanced Networking (TISPAN).
The present document is part 1 of a multi-part standard covering Network Integration Testing between SIP and
ISDN/PSTN network signalling protocols:
Part 1: "Test Suite Structure and Test Purposes (TSS&TP) for SIP-ISDN";
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&TP) for SIP-SIP".
ETSI
6 ETSI TS 186 001-1 V1.0.0 (2008-05)
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 ITU-T Recommendation Q.1912.5 [51] or
EN 383 001 [49] or ES 283 027 [48] and EN 300 899-1 [23]. For SIP and SDP specific terminology, reference shall be
made to ES 283 003 [42] respectively RFC 3261 [28].
SUT
SIP SIP ISUP mapping mapping ISDN ISDN
terminal terminal
SIP->ISUP ISUP->ISDN
Q.1912.5 EN 300 899-1
or (ITU-T Q.699)
EN 383 001
or
ES 283 027
Figure 1: SIP-ISDN inter-working testing architecture
SUT
SIP SIP ISUP mapping mapping PSTN PSTN
terminal terminal
SIP->ISUP ISUP->PSTN
EN 383 001 EN 300 899-1
or (ITU-T Q.699)
Q.1912.5
or
ES 283 027
Figure 2: SIP-PSTN inter-working testing architecture
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.
For online referenced documents, information sufficient to identify and locate the source shall be provided. Preferably,
the primary source of the referenced document should be cited, in order to ensure traceability. Furthermore, the
reference should, as far as possible, remain valid for the expected life of the document. The reference shall include the
method of access to the referenced document and the full network address, with the same punctuation and use of upper
case and lower case letters.
ETSI
7 ETSI TS 186 001-1 V1.0.0 (2008-05)
NOTE: While any hyperlinks included in this clause were valid at the time of publication ETSI cannot guarantee
their long term validity.
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] ITU-T Recommendations Q.761 to Q.764 (2000): "Specifications of Signalling System No.7
ISDN User Part (ISDN)".
[2] ITU-T Recommendations Q.1902.1 to Q.1902.4 (2001): "Specifications of the Bearer Independent
Call Control Protocol (BICC)".
[3] ITU-T Recommendation Q.731.7 (06/97): "Stage 3 description for number identification
supplementary services using Signalling System No. 7: Malicious call identification (MCID)".
[4] ITU-T Recommendation Q.732.2 (12/99): "Stage 3 description for call offering supplementary
services using Signalling System No. 7: Call diversion services: Call Forwarding Busy (CFB)".
[5] ITU-T Recommendation Q.732.3 (12/99): "Stage 3 description for call offering supplementary
services using Signalling System No. 7: Call Forwarding No Reply (CFNR)".
[6] ITU-T Recommendation Q.732.4 (12/99): "Stage 3 description for call offering supplementary
services using Signalling System No. 7: Call Forwarding Unconditional (CFU)".
[7] ITU-T Recommendation Q.732.5 (12/99): "Stage 3 description for call offering supplementary
services using Signalling System No. 7: Call Deflection (CALL DEFLECTION)".
[8] ITU-T Recommendation Q.732.7 (07/96): "Stage 3 description for call offering supplementary
services using Signalling System No. 7: Explicit Call Transfer".
[9] ITU-T Recommendation Q.733.1 (02/92): "Stage 3 description for call completion supplementary
services using Signalling System No. 7: Call waiting (CW)".
[10] ITU-T Recommendation Q.733.2 (03/93): "Stage 3 description for call completion supplementary
services using Signalling System No. 7: Call hold (HOLD)".
[11] ITU-T Recommendation Q.733.3 (06/97): "Stage 3 description for call completion supplementary
services using Signalling System No. 7: Completion of calls to busy subscriber (CCBS)".
[12] ITU-T Recommendation Q.733.4 (03/93): "Stage 3 description for call completion supplementary
services using Signalling System No. 7: Terminal portability (TP)".
[13] ITU-T Recommendation Q.733.5 (12/99): "Stage 3 description for call completion supplementary
services using Signalling System No. 7: Completion of calls on no reply".
[14] ITU-T Recommendation Q.734.1 (03/93): "Stage 3 description for multiparty supplementary
services using Signalling System No. 7: Conference calling".
[15] ITU-T Recommendation Q.734.2 (07/96): "Stage 3 description for multiparty supplementary
services using Signalling System No. 7: Three-party service".
[16] ITU-T Recommendation Q.735.1 (03/93): "Stage 3 description for community of interest
supplementary services using Signalling System No. 7: Closed user group (CUG)".
[17] ITU-T Recommendation Q.735.3 (03/93): "Stage 3 description for community of interest
supplementary services using Signalling System No. 7: Multi-level precedence and preemption".
[18] ITU-T Recommendation Q.735.6 (07/96): "Stage 3 description for community of interest
supplementary services using Signalling System No. 7: Global Virtual Network Service (GVNS)".
[19] ITU-T Recommendation Q.736.1 (10/95): "Stage 3 description for charging supplementary
services using Signalling System No. 7: International Telecommunication Charge Card (ITCC)".
ETSI
8 ETSI TS 186 001-1 V1.0.0 (2008-05)
[20] ITU-T Recommendation Q.736.3 (10/95): "Stage 3 description for charging supplementary
services using Signalling System No. 7: Reverse charging (REV)".
[21] ITU-T Recommendation Q.737.1 (06/97): "Stage 3 description for additional information transfer
supplementary services using Signalling System No. 7: User-to-user signalling (UUS)".
[22] ITU-T Recommendation Q.850 (05/98): "Usage of cause and location in the Digital Subscriber
Signalling System No. 1 and the Signalling System No. 7 ISDN User Part".
[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] IETF RFC 2046 (1996): "Multipurpose Internet Mail Extensions (MIME) Part Two: Media
Types".
[25] IETF RFC 2327 (1998): "SDP: Session Description Protocol".
[26] IETF RFC 2806 (2000): "URLs for Telephone Calls".
[27] IETF RFC 3204 (2001): "MIME media types for ISDNand QSIG Objects".
[28] IETF RFC 3261 (2002): "SIP: Session Initiation Protocol".
[29] IETF RFC 3262 (2002): "Reliability of Provisional Responses in the Session Initiation Protocol
(SIP)".
[30] IETF RFC 3264 (2002): "An Offer/Answer Model with the Session Description Protocol (SDP)".
[31] IETF RFC 3311 (2002): "The Session Initiation Protocol 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] IETF RFC 3326 (2002): "The Reason Header Field for the Session Initiation Protocol".
[36] ISO/IEC 9646-1 (1994): "Conformance testing methodology and framework - Part 1: General
Concepts".
[37] ISO/IEC 9646-2 (1994): "Conformance testing methodology and framework - Part 2: Abstract
Test Suite Specification".
[38] ISO/IEC 9646-3 (1992): "Conformance testing methodology and framework - Part 3: The Tree
and Tabular Combined Notation".
[39] ISO/IEC 9646-3/DAM 1 (1992): "Conformance testing methodology and framework - Part 3: The
Tree and Tabular Combined Notation; Amendment 1: TTCN extensions".
[40] ISO/IEC 9646-5 (1994): "Conformance testing methodology and framework - Part 5:
Requirements on test laboratories and clients for the conformance assessment process".
[41] ISO/IEC 9646-7 (1994): "Conformance testing methodology and framework - Part 7:
Implementation Conformance Statement".
[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 183 007: "Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN); PSTN/ISDN simulation services; Originating Identification
Presentation (OIP) and Originating Identification Restriction (OIR); Protocol specification".
ETSI
9 ETSI TS 186 001-1 V1.0.0 (2008-05)
[44] ETSI TS 183 008: "Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN); PSTN/ISDN simulation services Terminating Identification
Presentation (TIP) and Terminating Identification Restriction (TIR); Protocol specification".
[45] ETSI TS 183 004: "Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN); PSTN/ISDN simulation services: Communication Diversion
(CDIV); Protocol specification".
[46] ETSI TS 183 005: "Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN); PSTN/ISDN simulation services: Conference (CONF); Protocol
specification".
[47] ETSI TS 183 010: "Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN); NGN Signalling Control Protocol; Communication HOLD
(HOLD) PSTN/ISDN simulation services; Protocol specification".
[48] ETSI ES 283 027: "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]".
[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] IETF RFC 4967 (2007): "Dial String Parameter for the Session Initiation Protocol Uniform
Recource Identifier".
[51] ITU-T Recommendation Q.1912.5 (2004): "Interworking between Session Initiation Protocol
(SIP) and Bearer Independent Call Control Protocol or ISDN User Part".
[52] ITU-T Recommendation Q.699 (09/97): "Interworking between ISDN access and non-ISDN
access over ISDN User Part of Signalling System No. 7".
[53] ITU-T Recommendation Q.931 (05/98): "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 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);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 7.9.0 Release 7)".
[56] IETF RFC 2833: " RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals ".
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
10 ETSI TS 186 001-1 V1.0.0 (2008-05)
3 Definitions and abbreviations
3.1 Definitions
For the purposes of the present document, the following terms and definitions apply:
For BICC or ISUP specific terminology, reference shall be made to ITU-T Recommendation Q.1902.2 [2]. For SIP and
SDP specific terminology, reference shall be made to RFC 3261 [28] and RFC 2327 [25] respectively. Definitions for
additional terminology used in this interworking Recommendation are as follows:
Adjacent SIP Node (ASN): SIP node (e.g. SIP Proxy or Back-to-Back User Agent or the SIP side of an IWU) that has
established a direct trust relation (association) with Incoming or Outgoing IWU entities
NOTE: The SIP Proxy and Back-to-Back User Agent are defined in accordance with RFC 3261 [28].
Basic Call Control (BCC): signalling protocol associated with the DSS1 - ISDN Basic Call control procedures of ITU-
T recommendation Q.931 [53] (EN 300 403-1)
incoming or outgoing: direction of a call (not signalling information) with respect to a reference point
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
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/ISDNprotocol
NOTE: Without the term "network", it simply refers to the protocol.
SIP precondition: indicates the support of the SIP "precondition procedure" as defined in RFC 3312 [32]
syntactically invalid: specifies a test purpose covering a signalling procedure where a valid (expected in the current
status of the IUT) but not correctly encoded (unknown or incorrect parameter values) message is sent to the IUT, wSIch
shall react correctly and eventually reject the message
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 9646.
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
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.
ETSI
11 ETSI TS 186 001-1 V1.0.0 (2008-05)
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 [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 and components of the m= line.
3.2 Abbreviations
For the purposes of the present document, the following abbreviations apply:
GW GateWay
I Inopportune
IUT Implementation Under Test
PER Packed Encoding Rules
PICS Protocol Implementation Conformance Statement
PIXIT Protocol Implementation eXtra Information for Testing
S Syntactically invalid
TP Test Purpose
TSS Test Suite Structure
V Valid
ETSI
12 ETSI TS 186 001-1 V1.0.0 (2008-05)
4 Test Suite Structure (TSS)
4.1 Test Suite Structure (TSS)
4.1.1 ISDN-SIP
C - Plane / U - Plane
Basic_Call Successful Voice IS_XX_xx
Codec IS_CN_xx
negotiation
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 IS_XXSSCOLPxx
(TIP/TIR)
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
13 ETSI TS 186 001-1 V1.0.0 (2008-05)
4.1.2 SIP-ISDN
C - Plane / U - Plane
Basic_Call Successful 3,1 kHz audio SI_AU_xx
Codec SI_XX_CN_xx
negotiation
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 SI_XXSSCOLPxx
(TIP/TIR)
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
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
14 ETSI TS 186 001-1 V1.0.0 (2008-05)
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
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 (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 wSIch 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
ETSI
15 ETSI TS 186 001-1 V1.0.0 (2008-05)
CN: Codec negotiation
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
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 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 [28] and ETSI ES
283 003 [42] (modified 3GPP TS 24.229). 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 3GPP TS 24.229 [55] described in 3GPP TS 34.229-1 [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
the following table.
ETSI
16 ETSI TS 186 001-1 V1.0.0 (2008-05)
PIXIT Values
UE (MGCF) originating case UE (MGCF) terminating case
VA precondition" option-tag in the Supported header local resource local resource
reservation is reservation is not
required at the required by the
terminating UE 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 the terminating UE
required resources at the originating network are not
shall make use of
reserved the precondition
mechanism
VA_2.2 "precondition" option-tag in the Supported header and the terminating UE
required resources at the originating network are not shall use the
reserved precondition
mechanism
VA_3.1 "precondition" option-tag in the Supported header and the terminating UE
required local resources at the originating network shall make use of
the precondition
mechanism
VA_3.2 "precondition" option-tag in the Supported header and the required local
required local resources at the originating network 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" the terminating UE
option-tag in the Supported header shall not make use
of the precondition
mechanism.
VA_4.2 INVITE request does not include the "precondition" the terminating UE
option-tag in the Supported header 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: dialed digits; phone-context=<"+"CC>@homehostportion;user=phone
VA_5.3 sip: dialed 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
17 ETSI TS 186 001-1 V1.0.0 (2008-05)
6.1 ISDN - SIP
6.1.1 Basic Call
6.1.1.1 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: NGN reference to:
EN 300 403-1, clause 5.1.5.1 Q.1912.5 clause 7.1.1
EN 300 899-1 clause 2.1.1
EN 383 001 clause 7.1.1
ES 283 027 clause 7.2.3.2
TSS reference: ISDN-SIP/Basic_call/Successful/Voice
Selection criteria: Basic_call
Ensure that the call establishment using en-bloc sending is performed correctly.
Test purpose:
Ensure that in the active call state (N10) the voice transfer on
...