12.13b MT MTSI speech call Successful without preconditions at both originating UE and terminating UE

34.229-13GPPInternet Protocol (IP) multimedia call control protocol based on Session Initiation Protocol (SIP) and Session Description Protocol (SDP)Part 1: Protocol conformance specificationRelease 16TSUser Equipment (UE) conformance specification

12.13b.1 Definition

Test to verify that the UE correctly performs IMS mobile terminated speech call setup when using IMS Multimedia Telephony without preconditions. This process is described in 3GPP TS 24.229 [10], clauses 5.1.3 and 6.1, TS 24.173 [65] and TS 26.114 [66].

12.13b.2 Conformance requirement

same as 12.13 except

[TS 24.229, Rel-8, clause 5.1.3.1]:

The "integration of resource management and SIP" extension is hereafter in this subclause referred to as "the precondition mechanism" and is defined in RFC 3312 [30] as updated by RFC 4032 [64].

The preconditions mechanism should be supported by the originating UE.

[TS 24.229, Rel-8, clause 5.1.4.1]:

The preconditions mechanism should be supported by the terminating UE.

If local resource reservation is not required by the terminating UE and the terminating UE supports the precondition mechanism and:

…

b) the received INVITE request does not include the "precondition" option-tag in the Supported header field or Require header field, the terminating UE shall not make use of the precondition mechanism;

Reference(s)

3GPP TS 24.229 [10] clauses 5.1.3.1 and 5.1.4.1.

12.13b.3 Test purpose

1) To verify that, when initiating MT MTSI speech call the UE performs correct exchange of SIP protocol signalling messages for setting up the session.

2) To verify that within SIP signalling the UE performs the correct exchange of SDP contents for negotiating media without preconditions.

12.13b.4 Method of test

Initial conditions

UE contains either ISIM and USIM applications or only USIM application on UICC. UE has discovered P-CSCF and registered to IMS services, by executing the generic test procedure in Annex C.2 or C.2a (GIBA only) up to the last step.

SS is configured with the shared secret key of IMS AKA algorithm, related to the IMS private user identity (IMPI) configured on the UICC card equipp5ed into the UE. SS has performed AKAv1-MD5 authentication with the UE and accepted the registration (IMS security).

Test procedure applicable for a UE with E-UTRA support (TS 34.229-2 [5] A.18/1)

1-26) UE executes the procedures described in TS 36.508 [94] table 4.5A.7.3-1 steps 1 to26.

Expected sequence

NOTE: Only the IMS procedure relevant to the test purpose is described below.

Step

Direction

Message

Comment

UE

SS

1-6

Steps 1-6 defined in annex C.11

MTSI MT speech call. Referred from 36.508 [94] table 4.5A.7.3-1 for a UE with E-UTRA support.

7

Step 9 defined in annex C.11

The UE responds to INVITE with 180 Ringing

8-9

Step 12-13 defined in annex C.11

SS responds to INVITE with a 200 OK final response and SS acknowledges the receipt of 200 OK for INVITE.

10-11

Step 14-15 defined in annex C.11

The SS sends BYE to release the call and UE sends 200 OK for the BYE request and ends the call.

NOTE: The default messages contents in annex A are used with condition “IMS security” or “GIBA” when applicable

Specific Message Content

INVITE (Step 1)

Use the default message “INVITE for MT Call” in annex A.2.9 with the following exceptions:

Message-body

The following SDP types and values.

Session description:

  • v=0
  • o=- 1111111111 1111111111 IN (addrtype) (unicast-address for SS)
  • s=-
  • c=IN (addrtype) (connection-address for SS)
  • b=AS:37

Time description:

  • t=0 0

Media description:

  • m=audio (transport port) RTP/AVP 97 98 99 100
  • b=AS:37
  • b=RS:0
  • b=RR:2000

Attributes for media:

  • a=rtpmap:97 AMR-WB/16000/1
  • a=fmtp:97 mode-change-capability=2; max-red=220
  • a=rtpmap: 98 telephone-event/16000
  • a=fmtp: 98 0-15
  • a=rtpmap:99 AMR/8000/1
  • a=fmtp:99 mode-change-capability=2; max-red=220
  • a=rtpmap: 100 telephone-event/8000
  • a=fmtp: 100 0-15
  • a=ptime:20
  • a=maxptime:240

183 Session Progress (Step 4)

Use the default message "183 Session Progress" in annex A.2.3 with the following exceptions:

Header/param

Value/remark

Status-Line

Reason-Phrase

Not checked

Message-body

The following SDP types and values shall be present.

Session description:

  • v=0
  • o=(user-name) (sess-id) (sess-version) IN (addrtype) (unicast-address for UE)
  • s=(session name)
  • c=IN (addrtype) (connection-address for UE) [Note 1]
  • b=AS: (bandwidth-value)

Time description:

  • t=0 0

Media description:

  • m=audio (transport port) RTP/AVP (fmt) [Note 3]
  • c=IN (addrtype) (connection-address for UE) [Note 1]
  • b=AS: (bandwidth-value)
  • b=RS: (bandwidth-value)
  • b=RR: (bandwidth-value)

Attributes for media:

  • a=rtpmap:(payload type) AMR-WB/16000 [Note 2]
  • a=fmtp:(format) [Note 2, 3]

Note 1: At least one "c=" field shall be present.

Note 2: The values for fmt, payload type and format are not checked

Note 3: Parameters for the AMR codec are not checked

180 Ringing (Step 7)

Use the default message “180 Ringing for INVITE” in annex A.2.6 applying condition A2 (in addition to any other applicable conditions).

BYE (Step 10)

Use the default message “BYE” in annex A.2.8 applying condition A3 or A4 as appropriate.

200 OK for BYE (Step 11)

Use the default message “200 OK for other requests than REGISTER or SUBSCRIBE” in annex A.3.1 applying condition A5 and A11 (in addition to any other applicable conditions).