ETSI TS 186 001-3 V1.0.0 (2008-04)

Telecommunications and Internet Converged Services and Protocols for Advanced Networking (TISPAN); Network Integration Testing between SIP and ISDN/PSTN network signalling protocols; Part 3: Test Suite Structure and Test Purposes (TSS&TP) for SIP-SIP

DTS/TISPAN-06012-3-NGN-1

General Information

• Status: Published
• Publication Date: 01-Apr-2008
• Technical Committee: TISPAN 6 - TISPAN Testing
• Current Stage: 12 - Completion
• Due Date: 15-May-2008
• Completion Date: 02-Apr-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 3: Test Suite Structure and Test Purposes (TSS&TP)
for SIP-SIP
2 ETSI TS 186 001-3 V1.0.0 (2008-04)

Reference
DTS/TISPAN-06012-3-NGN-1
Keywords
ISDN, SIP, IP, TSS&TP
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ETSI
3 ETSI TS 186 001-3 V1.0.0 (2008-04)

Contents
Intellectual Property Rights.4
Foreword.4
1 Scope.5
2 References.5
2.1 Normative references.5
2.2 Informative references.6
3 Definitions and abbreviations.7
3.1 Definitions.7
3.2 Conventions for representation of SIP/SDP information .7
3.3 Abbreviations.8
4 Test Suite Structure (TSS).9
4.1 SIP-SIP.9
5 Numbering Scheme.9
5.1 General description.9
5.2 Basic Call.10
5.3 Supplementary Services .10
6 Test purposes.10
6.1 Void.12
6.2 Void.12
6.3 Void.12
6.4 Void.12
6.5 Test purposes for SIP-SIP .12
6.5.1 Test purposes for SIP-SIP, Basic call, Successful .12
6.5.1.1 Codec negotiation.20
6.5.1.2 UPDATE method.28
6.5.1.3 Test purposes for SIP-SIP, Basic call, Unsuccessful .31
6.5.1.3.1 Unsuccessful.31
6.5.2 Test purposes for SIP-SIP, Supplementary services .56
6.5.2.1 Test purposes for OIP.56
6.5.2.2 Test purposes for OIR .62
6.5.2.3 Test purposes for TIP.72
6.5.2.4 Test purposes for TIR.74
6.5.2.5 Hold.79
6.5.2.5.1 Communication Hold with support for UPDATE .79
6.5.2.5.2 Communication Hold without support for UPDATE .87
6.5.2.5.3 Communication with announcements.99
6.5.2.6 CFU.111
6.5.2.7 CFB.126
6.5.2.8 CFNR.154
6.5.2.9 CFNL.188
6.5.2.10 CALL DEFLECTION.194
6.5.2.11 CONF.204
6.5.2.11.1 Conference creation.204
6.5.2.11.2 Joining a conference .219
6.5.2.11.3 Inviting other users to a conference.221
6.5.2.11.4 Leaving a conference.229
6.5.2.11.5 Removing a conference participant from a conference .233
Annex A (informative): Bibliography.241
History .242

ETSI
4 ETSI TS 186 001-3 V1.0.0 (2008-04)

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) 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 Telecommunications and Internet
converged Services and Protocols for Advanced Networking (TISPAN).
The present document is part 3 of a multi-part deliverable covering Network Integration Testing between SIP and
ISDN/PSTN network signalling protocols, as identified below:
Part 1: "Test Suite Structure and Test Purposes (TSS&TP) specification for SIP-ISDN";
Part 2: "Test Suite Structure and Test Purposes (TSS&TP) ATS and PIXIT";
Part 3: "Test Suite Structure and Test Purposes (TSS&TP) for SIP-SIP".
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5 ETSI TS 186 001-3 V1.0.0 (2008-04)

1 Scope
This document specifies the Test Suite Structure and Test Purposes (TSS&TP) for Network Integration Testing (NIT) to
verify the overall compatibility of SIP networks. For SIP and SDP specific terminology, reference shall be made to
ES 283 003 [1] and RFC 3261 [2] respectively".
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.
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] 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]".
[2] IETF RFC 3261 (2002): "SIP: Session Initiation Protocol".
[3] Void.
[4] Void.
[5] ISO/IEC 9646-1 (1994): "Conformance testing methodology and framework - Part 1: General
Concepts".
[6] ISO/IEC 9646-2 (1994): "Conformance testing methodology and framework - Part 2: Abstract
Test Suite Specification".
[7] ISO/IEC 9646-3 (1992): "Conformance testing methodology and framework - Part 3: The Tree
and Tabular Combined Notation".
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6 ETSI TS 186 001-3 V1.0.0 (2008-04)

[8] ISO/IEC 9646-3/DAM 1 (1992): "Conformance testing methodology and framework - Part 3: The
Tree and Tabular Combined Notation; Amendment 1: TTCN extensions".
[9] ISO/IEC 9646-5 (1994): "Conformance testing methodology and framework - Part 5:
Requirements on test laboratories and clients for the conformance assessment process".
[10] ISO/IEC 9646-7 (1994): "Conformance testing methodology and framework - Part 7:
Implementation Conformance Statement etworking (TISPAN); PSTN/ISDN simulation services:
Conference (CONF); Protocol specification".
[11] 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)".
[12] 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)".
[13] Void.
[14] 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 [ITU-T Recommendation Q.931 (1993), modified]".
[15] 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]".
[16] 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]".
[17] ETSI TS 183 004: "Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN); PSTN/ISDN simulation services: Communication Diversion
(CDIV); Protocol specification".
[18] ETSI TS 183 028: "Telecommunications and Internet converged Services and Protocols for
Advanced Networking (TISPAN); Common Basic Communication procedures; Protocol
specification".
[19] ETSI TS 183 007: "Originating Identification Presentation (OIP) and Originating Identification
Restriction (OIR) PSTN/ISDN Simulation Services".
[20] ETSI TS 183 005: "Telecommunications and Internet converged Services and Protocols for
Advanced IETF RFC 3204 (2001), MIME media types for ISDNand QSIG Objects".
[21] 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".
2.2 Informative references
[22] IETF RFC 2327 (1998): "SDP: Session Description Protocol".
[23] IETF RFC 3312 (2002): "Integration of Resource Management and Session Initiation Protocol
(SIP)".
[24] IETF RFC 3311 (2002): "The Session Initiation Protocol UPDATE Method".
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7 ETSI TS 186 001-3 V1.0.0 (2008-04)

3 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 [2] and RFC 2327 [22] respectively.
SIP precondition: Indicates the support of the SIP "precondition procedure" as defined in RFC 3312 [23].
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
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 (all parts) [5] to [10].
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
3.2 Conventions for representation 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.
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/RFC 3261 [2]), in
plain text without surrounding angle brackets.
EXAMPLE 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 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.
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8 ETSI TS 186 001-3 V1.0.0 (2008-04)

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.
3.3 Abbreviations
For the purposes of the present document, the following abbreviations apply:
GW GateWay
I Inopportune
IUT Implementation Under Test
MCU Multipoint Control Unit
MGCF Media Gateway Control Function
MSI Manufacturer Specific Information
PDU Protocol Data Unit
PER Packed Encoding Rules
PICS Protocol Implementation Conformance Statement
PIXIT Protocol Implementation eXtra Information for Testing
PSA PSase A : Call setup signalling procedures
PSE PSase E : Call termination signalling procedures
RAS Registration, Admission and Status
RCF Register Confirm
REG REGistration
RRJ Register Reject
RRQ Register Request
S Syntactically invalid
STA STAtus
TE Terminal
TP Test Purpose
TSS Test Suite Structure
UCF Unregistration ConFirm
UE User Equipment
URJ Unregistration ReJect
URQ Unregistration ReQuest
V Valid
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9 ETSI TS 186 001-3 V1.0.0 (2008-04)

4 Test Suite Structure (TSS)
4.1 SIP-SIP
C – Plane / U – Plane
Basic_Call Successful
SS___XX___xx
Codec SS___CN___xx
negotiation
UPDATE SS___XX__UP__xx
Unsuccessful SS___XX__Uxx
Supplementary
Services OIP/OIR SS___XXSS_OIPxx
SS___XXSS_OIRxx
TIP/TIR SS___XXSS_TIPxx
HOLD SS___XXSS_CHxx
CFU SS___XXSS_CFUxx
CFB SS___XXSS_CFBxx
CFNR SS___XXSS_CFNRxx
CFNL SS___XXSS_CHFNLxx
CONF SS___XXSS_CONFxx
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
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10 ETSI TS 186 001-3 V1.0.0 (2008-04)

AU: 3,1 kHz Audio
UD: UDI
UT: UDI/TA
CN: Codec negotiation
DT: DTMF
UP: UPDATE Method
Pos. 8&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 [2] and ES 283 003 [1]
(modified TS 124 229 [11]). 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 TS 124 229 [11] are contained in TS 134 229-1 [12]
The preconditions mechanism shall be supported by the UE in case of supporting IMS.
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11 ETSI TS 186 001-3 V1.0.0 (2008-04)

The handling of preconditions at the originating or /and terminating UE (MGCF in case if interworking) is described in
the following table.
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 headerd and the terminating UE
required resources at the originating netwotk are not shall make use of
reserved the precondition
mechanism
VA_2.2 "precondition" option-tag in the Supported headerd 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)

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12 ETSI TS 186 001-3 V1.0.0 (2008-04)

6.1 Void
6.2 Void
6.3 Void
6.4 Void
6.5 Test purposes for SIP-SIP
6.5.1 Test purposes for SIP-SIP, Basic call, Successful
SS___XX__01 NGN reference to:
RFC 3261 [2]
TS 124 229 [11] / ES 283 003 [1]
TSS reference: SIP-SIP/Basic_call/Successful.
Selection criteria:
Test purpose: Ensure that call establishment and the correct handling and mapping of the SDP
parameters of the INVITE message defined as : TYPE_SDP is performed correctly.
Ensure that in the active call state the voice/data transfer on the media channels is
performed correctly (e.g. testing QoS parameters). In case when the parameter in the
SDP rtpmap: is used the codecs in table 1 applies. The call is released
from the called user.
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: A) SDP pre-condition not requested
SIP SUT SIP
INVITE  INVITE
100 Trying
180 Ringing  180 Ringing
200 OK INVITE 200 OK INVITE
ACK  ACK
BYE  BYE
200 OK BYE  200 OK BYE
C) pre-condition and 100 rel
INVITE  INVITE
100 Trying
183 Session Progress SDP
Authorize QoS resource at P-SCCF
183 Session Progress SDP
PRACK  PRACK
Resource Reservation at UE1
200 OK PRACK  200 OK PRACK
UPDATE  UPDATE
200 OK UPDATE 200 OK UPDATE
180 Ringing  180 Ringing
PRACK  PRACK
200 OK PRACK 200 OK PRACK
200 OK INVITE  200 OK INVITE
ACK  ACK
BYE BYE
200 OK BYE  200 OK BYE
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13 ETSI TS 186 001-3 V1.0.0 (2008-04)

SS___XX__02 NGN reference to:
RFC 3261 [2]
TS 124 229 [11] / ES 283 003 [1]
TSS reference: SIP-SIP/Basic_call/Successful.
Selection criteria:
Test purpose: Ensure that call establishment and the correct handling and mapping of the SDP
parameters of the INVITE message defined as : TYPE_SDP is performed correctly.
Ensure that in the active call state the voice/data transfer on the media channels is
performed correctly (e.g. testing QoS parameters). In case when the parameter in the
SDP rtpmap: is used the codecs in table 1 applies. The call is released
from the calling user.
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:
A) SDP pre-condition not requested
SIP SUT SIP
INVITE  INVITE
100 Trying
180 Ringing  180 Ringing
200 OK INVITE 200 OK INVITE
ACK  ACK
BYE  BYE
200 OK BYE 200 OK BYE
C) pre-condition and 100 rel
INVITE  INVITE
100 Trying
183 Session Progress SDP
Authorize QoS resource at P-SCCF
183 Session Progress SDP
PRACK  PRACK
Resource Reservation at UE1
200 OK PRACK  200 OK PRACK
UPDATE  UPDATE
200 OK UPDATE 200 OK UPDATE
180 Ringing  180 Ringing
PRACK  PRACK
200 OK PRACK 200 OK PRACK
200 OK INVITE  200 OK INVITE
ACK  ACK
BYE BYE
200 OK BYE  200 OK BYE
SS___XX__03 NGN reference to:
RFC 3261 [2]
TS 124 229 [11] / ES 283 003 [1]
TSS reference: SIP-SIP/Basic_call/Successful.
Selection criteria:
Test purpose: Ensure that call establishment and the correct if the called 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).

In case when the parameter in the SDP rtpmap: is used the codecs in
table 2 applies.
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
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14 ETSI TS 186 001-3 V1.0.0 (2008-04)

SS___XX__03 NGN reference to:
RFC 3261 [2]
TS 124 229 [11] / ES 283 003 [1]
Comments:
A) SDP pre-condition not requested
SIP SUT SIP
INVITE   INVITE
100 Trying
180 Ringing 180 Ringing
200 OK INVITE  200 OK INVITE
ACK  ACK
BYE BYE
200 OK BYE  200 OK BYE
SS___XX__04 NGN reference to:
RFC 3261 [2]
TS 124 229 [11] / ES 283 003 [1]
TSS reference: SIP-SIP/Basic_call/Successful.
Selection criteria:
Test purpose: Ensure that call establishment and the correctly if the called user answers with with a
200 OK message.
In case when the parameter in the SDP rtpmap: is used the codecs in
table 2 applies.
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:
A) SDP pre-condition not requested
SIP SUT SIP
INVITE INVITE
100 Trying
200 OK INVITE  200 OK INVITE

ACK ACK
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 : rtpmap:
/ rate>[/encoding parameters>
NOTE: for
of AS is
evaluated to be B kbit/s.
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:
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:
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

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15 ETSI TS 186 001-3 V1.0.0 (2008-04)

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
SS___XX__05 NGN reference to:
RFC 3261 [2]
TS 124 229 [11] / ES 283 003 [1]
TS 183 028 [18]
TSS reference: SIP-SIP/Basic_call/Successful
Selection criteria:
Test purpose: Ensure that an AS can send an announcement to the calling user during the
establishment of a communication.

The AS serving the calling party apply for example when a communication is going to be
diverted and the AS serving the diverting user inform the calling party that the
communication is going to be diverted.
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
16 ETSI TS 186 001-3 V1.0.0 (2008-04)

Comments:
S-CSCF MRFC/AS MRFP
1. INVITE
2. 100 Trying
3. Evaluation of initial
Filter Criteria
4. INVITE
5. 100 Trying
6. Service logic decides to
start announcement
7. Interaction to reserv resources for
announcement
8. 183 Session
9. 183 Session
Progress
Progress
10. PRACK
11. PRACK
12. 200 OK
13. 200 OK
14. Interaction to start announcemt
15. Announcement
16. Interaction to indicate that the
announcement is sent
17. Interaction to release resources
reserved for the announcement
18. INVITE
19. 100 Trying
20. INVITE
21. 100 Trying
22. 200 OK
23. 200 OK
24. 200 OK
25. 200 OK
26. ACK
27. ACK
28. ACK
29. ACK
ETSI
17 ETSI TS 186 001-3 V1.0.0 (2008-04)

SS___XX__06 NGN reference to:
RFC 3261 [2]
TS 124 229 [11] / ES 283 003 [1]
TS 183 028 [18]
TSS reference: SIP-SIP/Basic_call/Successful
Selection criteria: Including Alert-Info header field in the 180 (Ringing) response.
Test purpose: Ensure that according RFC 3261 [2] an Alert-Info header field, to indicate a source of
media to play an alternative ring tone by an originating endpoint can be used.
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
18 ETSI TS 186 001-3 V1.0.0 (2008-04)

Comments:
S-CSCF AS
1. INVITE
2 . 1 00 T r ying
3. Evaluation of initial
FilterC riteria
4. INVITE
5. 10 0 T ryin g
6. INVITE
7. 10 0 T ryin g
8. INVIT E
9. 10 0 T r yin g
10. 180 Ringing
11. 180 Ringing
12. 180 Ringing
13. 180 Ringing
Alert-Info:http://
Alert-Info:http://
url.wav
url.wav
14. Retrieving and
playing http://
url.wav
15. 200 OK
16. 200 OK
17. 200 OK
18. 200 OK
19. Stops playing
retrievedh ttp://url.wav
20 . AC K
21 . ACK
22 . ACK
23 . AC K
ETSI
2. re-INVITE
6 200 OK
.
12 A CK
.
13. ACK
19 ETSI TS 186 001-3 V1.0.0 (2008-04)

SS___XX__07 NGN reference to:
RFC 3261 [2]
TS 124 229 [11] / ES 283 003 [1]
TS 183 028 [18]
TSS reference: SIP-SIP/Basic_call/Successful.
Selection criteria: Providing announcements to a user during an established communication.
Test purpose: Two users are communicating with (at least) one AS in the signalling path.
Ensure that a user can place a communication session on hold and sends a re-INVITE
request towards the remote user involved in the communication. The Call-Info header
field in a re-INVITE request to indicate media.
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:
AS
S-CSCF
¨1. re-INVITE
3. Service logic decides to include
a reference to an wav file.
4. re-INVITE
(Call-Info: http://url.wav)
5. re-INVITE
(Call-Info: http://url.wav)
7. 200 OK
8. 200 OK
9. 200 OK
10. ACK
11. ACK
14. Retrieves and plays the http://
url.wav
ETSI
20 ETSI TS 186 001-3 V1.0.0 (2008-04)

6.5.1.1 Codec negotiation
SS___CN__01 NGN reference to:
RFC 3261 [2]
TS 124 229 [11] / ES 283 003 [1]
RFC 3311 [24]
TSS reference: SIP-SIP/Basic_call/Codec negotiation.
Selection criteria:
Test purpose: During the session, the calling user decides to change the characteristics of the media
session. This is accomplished by sending a re-INVITE containing a new media
description. This re-INVITE references the existing dialog so that the other party
knowsthat 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.
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: A) SDP pre-condition not requested
SIP SUT SIP
INVITE  INVITE
100 Trying
180 Ringing  180 Ringing
200 OK INVITE 200 OK INVITE
ACK  ACK
re-INVITE  re-INVITE
200 OK   200 OK
BYE  BYE
200 OK BYE 200 OK BYE
C) pre-condition and 100 rel
INVITE  INVITE
100 Trying
183 Session Progress
SDP
Authorize QoS resource at P-SCCF

183 Session Progress
SDP
PRACK  PRACK
Resource Reservation at UE1
200 OK PRACK  200 OK PRACK
UPDATE  UPDATE
200 OK UPDATE 200 OK UPDATE
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   200 OK
BYE  BYE
200 OK BYE 200 OK BYE
ETSI
21 ETSI TS 186 001-3 V1.0.0 (2008-04)

SS___CN__02 NGN reference to:
RFC 3261 [2]
TS 124 229 [11] / ES 283 003 [1]
RFC 3311 [24]
TSS reference: SIP-SIP/Basic_call/Codec negotiation
Selection criteria:
During the session, the called user decides to change the characteristics of the media
Test purpose:
session. This is accomplished by sending a re-INVITE containing a new media
description. This re- INVITE references the existing dialog so that the other party knows
that it is to modify an existing session instead of establishing a new session. The other
party sends a 200 (OK) to accept the change.The requestor responds to the 200 (OK)
with an ACK. In case when the parameter in the SDP rtpmap: is used the
codecs in table 2 applies.
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: A) SDP pre-condition not requested

INVITE  INVITE
100 Trying
180 Ringing  180 Ringing
200 OK INVITE 200 OK INVITE
ACK  ACK
re-INVITE   re-INVITE
200 OK  200 OK
BYE  BYE
200 OK BYE 200 OK BYE
C) pre-condition and 100 rel
INVITE  INVITE
100 Trying
183 Session Progress SDP
Authorize QoS resource at P-SCCF
183 Session Progress SDP
PRACK  PRACK
Resource Reservation at UE1
200 OK PRACK  200 OK PRACK
UPDATE UPDATE
200 OK UPDATE  200 OK UPDATE
180 Ringing  180 Ringing
PRACK PRACK
200 OK PRACK  200 OK PRACK
200 OK INVITE  200 OK INVITE

ACK ACK
BYE  BYE
re-INVITE   re-INVITE
200 OK  200 OK
200 OK BYE  200 OK BYE
ETSI
22 ETSI TS 186 001-3 V1.0.0 (2008-04)

SS___CN__03 NGN reference to:
RFC 3261 [2]
TS 124 229 [11] / ES 283 003 [1]
RFC 3311 [24]
TSS reference: SIP-SIP/Basic_call/Codec negotiation
Selection criteria:
Test purpose: Ensure that the call establishment is performed correctly.
• The initial INVITE contains a SDP with the offer 1.
• 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).
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: SDP pre-condition not requested
SIP SUT SIP
INVITE offer 1  INVITE offer1
100 Trying
180 Ringing Ringing answer 1 180 Ringing answer 1
200 OK INVITE  200 OK INVITE
ACK  ACK
BYE BYE
A) SDP pre-condition not requested
INVITE   INVITE offer1
100 Trying
180 Ringing answer 1 180 Ringing answer 1
200 OK INVITE  200 OK INVITE
ACK  ACK
BYE BYE
200 OK BYE  200 OK BYE
C) pre-condition and 100 rel
INVITE offer1  INVITE offer1

100 Trying
183 Session Progress SDP
Authorize QoS resource at P-SCCF

183 Session Progress SDP
PRACK  PRACK
Resource Reservation at UE1
200 OK PRACK 200 OK PRACK
180 Ringing answer 1  180 Ringing answer 1
PRACK  PRACK
200 OK PRACK 200 OK PRACK
200 OK INVITE  200 OK INVITE

ACK ACK
BYE  BYE
200 OK BYE  200 OK BYE
ETSI
23 ETSI TS 186 001-3 V1.0.0 (2008-04)

SS___CN__04 NGN reference to:
RFC 3261 [2]
TS 124 229 [11] / ES 283 003 [1]
RFC 3311 [24]
TSS reference: SIP-SIP/Basic_call/Codec negotiation.
Selection criteria:
Test purpose: Ensure that the call establishment performed correctly.
• The initial INVITE contains a SDP with the offer 1.
• 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).
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: A) SDP pre-condition not requested
SIP SUT SIP
INVITE offer 1  INVITE offer 1
100 Trying
183 Session Progress answer 1  183 Session Progress answer 1
180 Ringing  180 Ringing
200 OK INVITE  200 OK INVITE

ACK ACK
BYE  BYE
200 OK BYE  200 OK BYE
C) pre-condition and 100 rel
INVITE offer 1 INVITE offer 1
100 Trying
183 Session Progress answer 1
Authorize QoS resource at P-SCCF
183 Session Progress SDP
answer 1
PRACK PRACK
Resource Reservation at UE1
200 OK PRACK  200 OK PRACK
180 Ringing 180 Ringing
PRACK  PRACK
200 OK PRACK  200 OK PRACK
200 OK INVITE 200 OK INVITE
ACK  ACK
BYE  BYE
200 OK BYE 200 OK BYE
SS___CN__05 NGN reference to:
RFC 3261 [2]
TS 124 229 [11] / ES 283 003 [1]
RFC 3311 [24]
TSS reference: SIP-SIP/Basic_call/Codec negotiation.
Selection criteria:
Test purpose: Ensure that mobile originated while in home network, establishing a session with another
user. After the multimedia session is established, it is possible for either endpoint to
change the set of media flows or codec for a media flow.
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
24 ETSI TS 186 001-3 V1.0.0 (2008-04)

SS___CN__05 NGN reference to:
RFC 3261 [2]
TS 124 229 [11] / ES 283 003 [1]
RFC 3311 [24]
Comments:
UE#1 P-CSCF#1 S-CSCF#1 S-CSCF#2 P-CSCF#2 UE#2
1.
...