12.13 MT MTSI speech call with 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
12.13.1 Definition
Test to verify that the UE correctly performs IMS mobile terminated speech call setup when using IMS Multimedia Telephony. 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].
12.13.2 Conformance requirement
[TS 24.229, clause 5.1.4.1]
If an initial INVITE request is received the terminating UE shall check whether the terminating UE requires local resource reservation.
NOTE 1: The terminating UE can decide if local resource reservation is required based on e.g. application requirements, current access network capabilities, local configuration, etc.
If local resource reservation is required at the terminating UE and the terminating UE supports the precondition mechanism, and:
a) the received INVITE request includes the "precondition" option-tag in the Supported header or Require header, the terminating UE shall make use of the precondition mechanism and shall indicate a Require header with the "precondition" option-tag in any response or subsequent request it sends towards to the originating UE; or
…
If local resource reservation is not required by the terminating UE and the terminating UE supports the precondition mechanism and:
a) the received INVITE request includes the "precondition" option-tag in the Supported header and:
– the required resources at the originating UE are not reserved, the terminating UE shall use the precondition mechanism; or
[TS 24.229, clause 6.1.1]
During session establishment procedure, and during session modification procedures, SIP messages shall only contain SDP payload if that is intended to modify the session description, or when the SDP payload is included in the message because of SIP rules described in RFC 3261.
[TS 24.229, clause 6.1.3]
If the terminating UE had previously set one or more media streams to inactive mode and the QoS resources for those media streams are now ready, it shall set the media streams to active mode by applying the procedures described in RFC 4566 with respect to setting the direction of media streams.
…
Upon sending a SDP answer to an SDP offer, with the SDP answer including one or more media streams for which the originating side did indicate its local preconditions as not met, if the precondition mechanism is supported by the terminating UE, the terminating UE shall indicate its local preconditions and request the confirmation for the result of the resource reservation at the originating end point.
[TS 26.114, clause 5.2.1]
MTSI terminals offering speech communication shall support:
– AMR speech codec (3GPP TS 26.071, 3GPP TS 26.090, 3GPP TS 26.073 and 3GPP TS 26.104) including all 8 modes and source controlled rate operation 3GPP TS 26.093. The terminal shall be capable of operating with any subset of these 8 codec modes.
[TS 26.114, clause 6.2.2.1]
An MTSI client offering a speech media session for narrow-band speech and/or wide-band speech should offer SDP according to the examples in clauses A.1 to A.3.
An MTSI client shall at least offer AVP for speech media streams. An MTSI client should also offer AVPF for speech media streams. RTP profile negotiation shall be done as described in clause 6.2.1a.
[TS 26.114, clause 6.2.5]
The SDP shall include bandwidth information for each media stream and also for the session in total. The bandwidth information for each media stream and for the session is defined by the Application Specific (AS) bandwidth modifier as defined in RFC 4566.
[TS 26.114, clause 7.3.1]
The bandwidth for RTCP traffic shall be described using the "RS" and "RR" SDP bandwidth modifiers at media level, as specified by RFC 3556.
Reference(s)
3GPP TS 24.229 [10] clauses 5.1.4.1, 6.1.1, 6.1.3, TS 26.114 [66] clause 5.2.1, 6.2.2.1, 6.2.5 and 7.3.1.
12.13.3 Test purpose
1) To verify that, when initiating MT MTSI speech call and SS needs to reserve resources, 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 SIP header and parameter contents.
3) To verify that within SIP signalling the UE performs the correct exchange of SDP contents.
4) To verify that the UE is able to release the call.
12.13.4 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 |
|||
|
1-15 |
Steps 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. |
||
NOTE: The default messages contents in annex A are used with condition “IMS security“ or “GIBA” when applicable
Specific Message Content
None.
12.13.5 Test requirements
The UE shall send requests and responses as described in clause 12.13.4