A.5 Signalling flows for call termination for service continuity

24.2373GPPIP Multimedia (IM) Core Network (CN) subsystem IP Multimedia Subsystem (IMS) service continuityRelease 17Stage 3TS

A.5.1 Session termination using CS media

An example flow for session termination using CS calls can be found in 3GPP TS 24.292 [4].

A.5.2 Call termination prior to CS to PS SRVCC

The signalling flow shown in figure A.5.2-1 gives an example of a terminating session set up. In this flow, the ATCF anchors the media of the session in ATGW.

NOTE: For clarity, the SIP 100 (Trying) responses are not shown in the signalling flow.

Figure A.5.2-1 Signalling flows for service continuity using CS to PS SRVCC.

1. SC UE A registers in IMS.

2. The SC UE A performs a CS attach and the MSC server registers in IMS.

3-4. SIP INVITE request (Remote UE B to SCC AS)

The remote UE sends a SIP INVITE request towards the user at SC UE A.

5-6. SIP INVITE request (SCC AS to MSC server) – see example in table A.5.2-5

The SCC AS forwards the SIP INVITE request towards the MSC server.

Table A.5.2-5: SIP INVITE request (SCC AS to MSC server)

INVITE tel:+1-237-555-1111 SIP/2.0

Via: SIP/2.0/UDP sccas1.home1.net;branch=z9hG4bK871y12.1

Max-Forwards: 70

Route: <sip:scscf1.home1.net;lr>,<sip:msc2.visited2.net;lr>

Record-Route: <sip:sccas1.home1.net;lr>

P-Asserted-Identity: "John Doe" <sip:user1_public2@visited2.net>, <tel:+1-237-555-2222>

P-Asserted-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel

Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"

Privacy: none

From: <tel:+1-237-555-2222>;tag=171828

To: tel:+1-237-555-1111

P-Charging-Vector: icid-value="1234bc9876e";icid-generated-at=5555::aaa:bbb:ccc:eeee";orig-ioi=visited2.net

Call-ID: cb03a0s09a2sdfglkj490333

Cseq: 127 INVITE

Supported: 100rel, precondition, gruu, 199

Accept: applicatiom/sdp,application/3gpp-ims+xml

Contact: <sip:user2_public1@visited2.net;gr=urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel">

Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER, MESSAGE

Feature-Caps: *;+g.3gpp.srvcc

Content-Type: application/sdp

Content-Length: (…)

v=0

o=- 2987933615 2987933615 IN IP6 5555::aaa:bbb:ccc:ddd

s=-

c=IN IP6 5555::aaa:bbb:ccc:ddd

t=0 0

m=video 3400 RTP/AVP 98 99

b=AS:75

a=curr:qos local none

a=curr:qos remote none

a=des:qos mandatory local sendrecv

a=des:qos none remote sendrecv

a=inactive

a=rtpmap:98 H263

a=fmtp:98 profile-level-id=0

a=rtpmap:99 MP4V-ES

m=audio 3456 RTP/AVP 97 0 96

b=AS:25.4

a=curr:qos local none

a=curr:qos remote none

a=des:qos mandatory local sendrecv

a=des:qos none remote sendrecv

a=rtpmap:97 AMR

a=fmtp:97 mode-set=0,2,5,7; mode-change-period=2

a=rtpmap:96 telephone-event

a=maxptime:20

Feature-Caps: The header field contains

– g.3gpp.srvcc indicating that the session has been anchored in the SCC AS.

7. INVITE request (MSC server to ATCF) – see example in table A.5.2-7

Table A.5.2-7: SIP INVITE request (MSC server to ATCF)

INVITE tel:+1-237-555-1111 SIP/2.0

Via: SIP/2.0/UDP msc2.visited2.net;branch=z9hG4bK879l11.1,SIP/2.0/UDP scscf1.home1.net;branch=z9hG4bK871z34.1,SIP/2.0/UDP sccas1_s.home1.net;branch=z9hG4bK871y12.1

Max-Forwards: 70

Route: <sip:atcf2.visited2.net;lr>,<sip:msc2.visited2.net;lr>

Record-Route: <sip:atcf2.visited2.net;lr>,<sip:msc2.visited2.net;lr>,<sip:scscf1.home1.net;lr>,<sip:sccas1.home1.net;lr>

P-Asserted-Identity: "John Doe" <sip:user1_public2@visited2.net>, <tel:+1-237-555-2222>

P-Asserted-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel

Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"

Privacy: none

From: <sip:user2_public1@visited2.net>;tag=171828

To: tel:+1-237-555-1111

Call-ID: cb03a0s09a2sdfglkj490333

Cseq: 127 INVITE

Supported: 100rel, precondition, gruu, 199

Accept: applicatiom/sdp,application/3gpp-ims+xml

Contact: <sip:user2_public1@visited2.net;gr=urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel">;g.3gpp.rsrvcc

Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER, MESSAGE

P-Charging-Vector: icid-value="1234bc9876e";icid-generated-at=5555::aaa:bbb:ccc:eeee";orig-ioi=visited2.net

Feature-Caps: *;+g.3gpp.srvcc

Content-Type: multipart/mixed;boundary="boundary1"

Content-Length: (…)

–boundary1

Content-Type: application/sdp

v=0

o=- 2987933615 2987933615 IN IP6 5555::aaa:bbb:ccc:ddd

s=-

c=IN IP6 5555::aaa:bbb:ccc:ddd

t=0 0

m=video 3400 RTP/AVP 98 99

b=AS:75

a=curr:qos local none

a=curr:qos remote none

a=des:qos mandatory local sendrecv

a=des:qos none remote sendrecv

a=inactive

a=rtpmap:98 H263

a=fmtp:98 profile-level-id=0

a=rtpmap:99 MP4V-ES

m=audio 3456 RTP/AVP 97 0 96

b=AS:25.4

a=curr:qos local none

a=curr:qos remote none

a=des:qos mandatory local sendrecv

a=des:qos none remote sendrecv

a=rtpmap:97 AMR

a=fmtp:97 mode-set=0,2,5,7; mode-change-period=2

a=rtpmap:96 telephone-event

a=maxptime:20

–boundary1

Content-Type: application/vnd.3gpp.srvcc-ext+xml

<?xml version="1.0"?>

<srvcc-ext>

<Setup-info>

<C-MSISDN>tel:+1-212-555-1111</C-MSISDN>

<direction>receiver</direction>

</Setup-info>

</srvcc-ext>

–boundary1–

Route: ATCF management URI received from SCC AS in the SIP MESSAGE request containing CS to PS SRVCC information during the registration of the user followed by MSC server URI.

application/vnd.3gpp.srvcc-ext+xml: Contains the direction of call and the C-MSISDN of the UE.

8. ATCF decides whether to anchor the media in the ATGW. In this flow, the ATCF decides to anchor the media in the ATGW and reserves the resources in the ATGW.

9. SIP INVITE request (ATCF to MSC server) – see example in table A.5.2-9

The ATCF forwards the SIP INVITE request to MSC server according to the received Route header field.

Table A.5.2-9: SIP INVITE request (ATCF to MSC server)

INVITE tel:+1-237-555-1111 SIP/2.0

Via: SIP/2.0/UDP atcf2.visited2.net;branch=z9hG4bK871x99ja,SIP/2.0/UDP msc2.visited2.net;branch=z9hG4bK879l11.1,SIP/2.0/UDP scscf1.home1.net;branch=z9hG4bK871z34.1,SIP/2.0/UDP sccas1.home1.net;branch=z9hG4bK871y12.1

Max-Forwards: 68

Route:<sip:msc2.visited2.net;lr>

Record-Route:<sip:atcf2.visited2.net:7531;lr>,<sip:scscf1.visited2.net;lr>,<sccas1.home1.net;lr>

P-Asserted-Identity:

P-Charging-Vector: icid-value="1234bc9876e";icid-generated-at=5555::aaa:bbb:ccc:eeee";orig-ioi=visited2.net

P-Asserted-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel

Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"

Privacy:

From:

To:tel:

Call-ID: cb03a0s09a2sdfglkj490333

Cseq: 127 INVITE

Supported:

Accept: applicatiom/sdp,application/3gpp-ims+xml,application/3gpp.access-transfer-events+xml;et="1,3,4"

Recv-Info: g.3gpp.access-transfer-events

Contact: sip:<+12375551111 @msc2.visited2.net;user=phone>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel">

Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER, MESSAGE

Feature-Caps: *;+g.3gpp.srvcc

Content-Type: application/sdp

Content-Length: (…)

v=0

o=- 2987933615 2987939999 IN IP6 8888::111:222:333:444

s=-

c=IN IP6 8888::111:222:333:444

t=0 0

m=video 3400 RTP/AVP 98 99

b=AS:75

a=curr:qos local none

a=curr:qos remote none

a=des:qos mandatory local sendrecv

a=des:qos none remote sendrecv

a=inactive

a=rtpmap:98 H263

a=fmtp:98 profile-level-id=0

a=rtpmap:99 MP4V-ES

m=audio 3456 RTP/AVP 97 0 96

b=AS:25.4

a=curr:qos local none

a=curr:qos remote none

a=des:qos mandatory local sendrecv

a=des:qos none remote sendrecv

a=rtpmap:97 AMR

a=fmtp:97 mode-set=0,2,5,7; mode-change-period=2

a=rtpmap:96 telephone-event

a=maxptime:20

SDP offer: the IP address and ports are updated to contain the values provided by ATGW.

Accept: Indicates that the ATCF is able to receive the application/3gpp.access-transfer-events+xml with event types 1, 3 and 4.

Recv-Info: Indicates that the ATCF support receiving the g.3gpp.access-transfer-events info package.

10. CC SETUP message (MSC server to UE A)

The MSC server sends a CC SETUP message. The mapping of the INVITE request to the CC SETUP message is described by 3GPP TS 29.292 [18].

11. CC CALL CONFIRM message (UE A to MSC server)

The UE sends a CC CALL CONFIRM message in accordance to 3GPP TS 24.008 [8].

12. SIP 183 (Session Progress) response (MSC server to ATCF) – see example in table A.5.2-12

The MSC server sends a SIP 183 (Session Progress) response. The CC CONFIRMED message is mapped to the SIP 183 (Session Progress) response as described in 3GPP TS 29.292 [18].

Table A.5.2-12: SIP 183 (Session Progress) response (MSC server to ATCF)

SIP/2.0 183 Session Progress

Via:

Route: sip:atcf2.visited2.net:7531;lr;>; <sip:scscf1.visited2.net;lr>>

Record-Route: <sip:scscf1.visited2.net;lr>,<sip:atcf2.visited2.net:7531;lr;>

P-Asserted-Identity: tel:<+12375551111>

P-Called-Party-ID:tel:<+12375551111>

P-Charging-Vector: icid-value="1234bc9876e";icid-generated-at=5555::aaa:bbb:ccc:eeee";orig-ioi=visited2.net

From:

To: <tel:+1-237-555-2222>;tag=314159

Call-ID:

Cseq:

Require: 100rel

Contact: <sip:user1_public1@visited2.net;gr=urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";+g.3gpp.ti="70D8"

Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER

Rseq: 9021

Content-Type: application/sdp

Content-Length: (…)

Recv-Info: g.3gpp.access-transfer-events;et="2"

v=0

o=- 2987933615 2987939999 IN IP6 8888::111:222:333:666

s=-

c=IN IP6 8888::111:222:333:666

t=0 0

m=video 0 RTP/AVP 98 99

m=audio 6544 RTP/AVP 97 96

b=AS:25.4

a=curr:qos local sendrecv

a=curr:qos remote none

a=des:qos mandatory local sendrecv

a=des: qos mandatory remote sendrecv

a=conf:qos remote sendrecv

a=rtpmap:97 AMR

a=fmtp:97 mode-set=0,2,5,7; mode-change-period=2

a=rtpmap:96 telephone-event

a=maxptime:20

Recv-Info: Indicates that the ATCF support receiving the g.3gpp.access-transfer-events info package with event type 2.

Contact: g.3gpp.ti media feature tag with value containing the transaction identifier specified in figure 11.9 and table 11.3 of 3GPP TS 24.007 [75] encoded by hexadecimal digit. In this example, the transaction identifier 88 (decimal) and the transaction identifier flag as sent by the MSC server in CS signalling of the terminating CS call are shown.

13. Configure ATGW resources

The ATCF configures resources in the ATGW.

14. SIP 183 (Session Progress) response (ATCF to MSC server) – see example in table A.5.2-14

Table A.5.2-14: SIP 183 (Session Progress) response (ATCF to MSC server)

SIP/2.0 183 Session Progress

Via:

Route: sip:atcf2.visited2.net:7531;lr;>; <sip:scscf1.home1.net;lr>;g.3gpp.srvcc me

Record-Route: <sip:scscf1.home1.net;lr>,<sip:atcf2.visited2.net:7531;lr;>

P-Asserted-Identity: <tel:+1-237-555-1111>

P-Called-Party-ID: <tel:+1-237-555-1111>

P-Charging-Vector: icid-value="1234bc9876e";icid-generated-at=5555::aaa:bbb:ccc:eeee";orig-ioi=visited2.net

From:

To: <tel:+1-237-555-2222>;tag=314159

Call-ID:

Cseq:

Require: 100rel

Contact:

Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER

Rseq: 9021

Content-Type: application/sdp

Content-Length: (…)

v=0

o=- 2987933615 2987332299 IN IP6 8888::111:222:333:446

s=-

c=IN IP6 8888::111:222:333:446

t=0 0

m=video 0 RTP/AVP 98 99

m=audio 53261 RTP/AVP 97 96

b=AS:25.4

a=curr:qos local sendrecv

a=curr:qos remote none

a=des:qos mandatory local sendrecv

a=des: qos mandatory remote sendrecv

a=conf:qos remote sendrecv

a=rtpmap:97 AMR

a=fmtp:97 mode-set=0,2,5,7; mode-change-period=2

a=rtpmap:96 telephone-event

a=maxptime:20

SDP answer: the IP address and ports are updated to contain the values provided by ATGW.

15-16. SIP 183 (Session Progress) response (MSC server to SCC AS) – see example in table A.5.2-15

Table A.5.2-15: SIP 183 (Session Progress) response (ATCF to SCC AS)

SIP/2.0 183 Session Progress

Via:

Route: sip:atcf2.visited2.net:7531;lr;>; <sip:scscf1.home1.net;lr>;g.3gpp.srvcc me

Record-Route: <sip:scscf1.home1.net;lr>,<sip:atcf2.visited2.net:7531;lr;>

P-Asserted-Identity: <tel:+1-237-555-1111>

P-Called-Party-ID:tel: <tel:+1-237-555-1111>

P-Charging-Vector: P-Charging-Vector: icid-value="1234bc9876e";icid-generated-at=5555::aaa:bbb:ccc:eeee";orig-ioi=visited2.net

From:

To: <tel:+1-237-555-2222>;tag=314159

Call-ID:

Cseq:

Require: 100rel

Contact:

Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER

Rseq: 9021

Content-Type: application/sdp

Content-Length: (…)

v=0

o=- 2987933615 2987332299 IN IP6 8888::111:222:333:446

s=-

c=IN IP6 8888::111:222:333:446

t=0 0

m=video 0 RTP/AVP 98 99

m=audio 53261 RTP/AVP 97 96

b=AS:25.4

a=curr:qos local sendrecv

a=curr:qos remote none

a=des:qos mandatory local sendrecv

a=des: qos mandatory remote sendrecv

a=conf:qos remote sendrecv

a=rtpmap:97 AMR

a=fmtp:97 mode-set=0,2,5,7; mode-change-period=2

a=rtpmap:96 telephone-event

a=maxptime:20

SDP answer: the IP address and ports are the values provided by ATGW.

16-17. SIP 183 (Session Progress) response (SCC AS to remote UE)

The SCC AS sends the SIP 183 (Session Progress) response towards the remote UE.

Regular call setup continues.