A.5.6 Signalling flows with call termination delivery attempt failure to the IM CN subsystem

24.2063GPPStage 3TSVoice call continuity between Circuit Switched (CS) and IP Multimedia Subsystem (IMS)

Figure A.5.6-1 shows the termination of a call that is capable of being subject to VCC, where the call delivery attempt to the IM CN subsystem fails. In this example and following information available at the VCC application, the call termination is re-attempted on the CS domain, 2nd domain available.

In this example, when the VCC Application receives the indication that delivery of the terminating call through the selected IM CN subsystem cannot be completed, the VCC application performs an implementation option to attempt delivery of the terminating call through the CS domain. Use of this implementation option depends on operator policies.

Figure A.5.6-1: Call termination delivery attempt failure to the IM CN subsystem

The details of the signalling flows are as follows:

Steps 1 to 8 are identical to the example in subclause A.5.2.

9. User unreachable

The delivery of the call fails due to an error detected in the termination procedure. This could be due to, for example, destination busy (error code 486), destination service denied (error code 403), destination currently out of coverage (error code 480), or some other error. In this example the intermediate IM CN subsystem returns SIP 480 (Temporarily unavailable) response to the DTF.

10. SIP 480 (Temporarily Unavailable) response (intermediate IM CN subsystem entities to DTF) – see example in table A.5.6-10

The intermediate IM CN subsystem returns SIP 480 (Temporarily Unavailable) response to the DTF.

Table A.5.6-10: SIP 480 (Temporarily Unavailable) response (intermediate IM CN subsystem entities to DTF)

SIP/2.0 480 Temporarily unavailable

Via: SIP/2.0/UDP sip:dtf2.home2.net; branch=z9jG4bK332b23.3, SIP/2.0/UDP scscf2.home2.net;branch=z9hG4bK332b23.1, pcscf2.home2.net;branch=z9hG4bK431h23.1, SIP/2.0/UDP

From:

To:

Call-ID:

Cseq:

Retry-After: 3600

Content-Length: 0

11. Interaction between the DTF and DSF

Information is exchanged between the DTF and DSF. The nature of this signalling is outside the scope of this version of the specification.

12. Domain Selection (re-attempt)

In this example the VCC user has been simultaneously registered in IM CN subsystem and in the CS domain (CS attached). This information is known to the DSF, and after the first attempt and failure to deliver the terminated call in the IM CN subsystem, the DSF re-attempts on the second domain available to the terminated user i.e. the CS domain.

NOTE 1: The choice for the DSF to re-attempt the call termination on the second domain available following a failure on the first domain is subject to operator policy.

The DSF determines the CSRN. The CSRN will allow the call to be routed further into the CS domain. In this example the CSRN = +1-241-555-4444

NOTE 2: it is an implementation option as how the DSF determines the CSRN. The DSF can collaborate with the CSAF for determination of the CSRN.

13. Interaction between the DTF and DSF

Information is exchanged between the DTF and DSF. The nature of this signalling is outside the scope of this version of the specification.

14. SIP INVITE request (DTF to intermediate IM CN subsystem entities) – see example in table A.5.6-12

The DTF sends SIP INVITE request with the CSRN as the Request-URI to the intermediate IM CN subsystem entities.

The DTF modifies the message received in step 3 in accordance with routeing B2BUA functionality, e.g. mapping of From, To, Cseq and Call-ID headers from one side of the B2BUA to the other.

The DTF inserts the original SDP from the originating SIP INVITE request.

The History-Info header can be included to contain the public user identity of the terminating user which was previously included in the Request-URI.

Table A.5.6-14: SIP INVITE request (DTF to intermediate IM CN subsystem entities)

INVITE tel:+1-241-555-4444 SIP/2.0

Via: SIP/2.0/UDP sip:dtf2.home2.net; branch=z9jG4bK332b23.3, SIP/2.0/UDP scscf2.home2.net;branch=z9hG4bK332b23.1, SIP/2.0/UDP icscf2_s.home2.net;branch=z9hG4bK871y12.1, SIP/2.0/UDP scscf1.home1.net;branch=z9hG4bK332b23.1, SIP/2.0/UDP pcscf1.home1.net;branch=z9hG4bK431h23.1, SIP/2.0/UDP [5555::aaa:bbb:ccc:ddd]:1357;comp=sigcomp;branch=z9hG4bKnashds7

Max-Forwards: 65

Route: <sip:scscf2.home2.net;lr>; dia-id=6574839201

Record-Route: <sip:dtf2.home2.net;lr>, <sip:scscf2.home2.net;lr>, <sip:scscf1.home1.net;lr>, <sip:pcscf1.home1.net;lr>

P-Asserted-Identity: "John Doe" <tel:+1-212-555-1111>

P-Access-Network-Info:

P-Charging-Vector: icid-value="AyretyU0dm+6O2IrT5tAFrbHLso=023551024"; orig-ioi="type 3ashome1.net"

Privacy: none

Supported: histinfo

From:

To: <tel:+1-241-555-4444>

Call-ID:

Cseq:

Contact:

History-Info: <sip:user2_public1@home2.net>; index=1, <tel:+ 1-241-555-4444>; index=1.1

Allow:

Content-Type:

Content-Length: (…)

v=

o=

s=

c=

t=

m=

b=

a=

a=

a=

a=

a=

a=

a=

a=

The termination of the call can proceed toward the CS domain as described in the subclause A.5.3 starting from the step 10.