A.4 Signalling flows for call origination for service continuity
24.2373GPPIP Multimedia (IM) Core Network (CN) subsystem IP Multimedia Subsystem (IMS) service continuityRelease 17Stage 3TS
A.4.1 Session origination for CS calls
An example flow for session origination for CS calls can be found in 3GPP TS 24.292 [4].
A.4.2 Session origination with PS to CS SRVCC enhancements
The signalling flow shown in figure A.4.2-1 gives an example of originating session set up when ATCF anchors the media of the session. This flow assumes that ATCF was invoked during registration.
Figure A.4.2-1 Signalling flows for service continuity using PS to CS SRVCC enhancements
1. SIP INVITE request (UE to P-CSCF) – see example in table A.4.2-1
Table A.4.2-1: SIP INVITE request (UE to P-CSCF)
INVITE tel:+1-212-555-2222 SIP/2.0
Via: SIP/2.0/UDP [5555::aaa:bbb:ccc:ddd]:1357;comp=sigcomp;branch=z9hG4bKnashds7
Max-Forwards: 70
Route: <sip:pcscf1.visited2.net:7531;lr;comp=sigcomp>, <sip:orig@scscf1.home1.net;lr>
P-Preferred-Identity: "John Doe" <sip:user1_public1@home1.net>
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
P-Access-Network-Info: 3GPP-UTRAN-TDD; utran-cell-id-3gpp=234151D0FCE11
Privacy: none
From: <sip:user1_public1@home1.net>;tag=171828
To: <tel:+1-212-555-2222>
Call-ID: cb03a0s09a2sdfglkj490333
Cseq: 127 INVITE
Require: sec-agree
Supported: precondition, 100rel, gruu
Proxy-Require: sec-agree
Security-Verify: ipsec-3gpp; q=0.1; alg=hmac-sha-1-96; spi-c=98765432; spi-s=87654321; port-c=8642; port-s=7531
Contact: <sip:user1_public1@home1.net;gr=urn:uuid:f81d4fae-7dec-11d0-a765-00a0c91e6bf6;comp=sigcomp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER, MESSAGE
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=audio 3456 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 none remote sendrecv
a=rtpmap:97 AMR
a=fmtp:97 mode-set=0,2,5,7; maxframes=2
a=rtpmap:96 telephone-event
2. SIP INVITE request (P-CSCF to ATCF) – see example in table A.4.2-2
Since a Feature-Caps header field with the g.3gpp.atcf feature-capability indicator was included in SIP 2xx response to the SIP REGISTER request which created the binding of the contact address using which the SIP INVITE request is sent, the P-CSCF routes the SIP INVITE request to the ATCF.
Table A.4.2-2: SIP INVITE request (P-CSCF to ATCF)
INVITE tel:+1-212-555-2222 SIP/2.0
Record-Route: <sip:pcscf1.visited1.net;lr>
Via: SIP/2.0/UDP pcscf1.visited2.net:5060;branch=z9hG4bKnas56565, SIP/2.0/UDP [5555::aaa:bbb:ccc:ddd]:1357;comp=sigcomp;branch=z9hG4bKnashds7
Max-Forwards: 69
Route: <sip:orig@atcf.visited2.net;lr>, <sip:orig@scscf1.home1.net;lr>
P-Asserted-Identity: "John Doe" <sip:user1_public1@home1.net>
P-Preferred-Service:
P-Access-Network-Info:
Privacy:
From:
To:
Call-ID:
Cseq:
Require:
Supported:
Proxy-Require:
Contact:
Accept-Contact
Allow:
Content-Type:
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=audio 3456 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 none remote sendrecv
a=rtpmap:97 AMR
a=fmtp:97 mode-set=0,2,5,7; maxframes=2
a=rtpmap:96 telephone-event
Route: ATCF URI for originating requests (as configured in P-CSCF) followed by the remaining Route header fields determined by P-CSCF.
3. ATGW resource reservation
The ATCF decides to anchor the media of the session and reserves the resources in the ATGW.
4-9. SIP INVITE request (ATCF towards remote UE) – see example in table A.4.2-4
The ATCF modifies the SDP offer without changing the dialog identifier and forwards the SIP INVITE request. The ATCF replaces the IP address, ports, … with values provided by ATGW.
Table A.4.2-4: SIP INVITE request (ATCF towards remote UE)
INVITE tel:+1-212-555-2222 SIP/2.0
Record-Route: <sip:pcscf1.visited1.net;lr>, <sip:atcf.visited.net;lr>
Via: SIP/2.0/UDP atcf.visited2.net:5060;branch=z9hG4bKnas55889, SIP/2.0/UDP pcscf1.visited2.net:5060;branch=z9hG4bKnas56565, SIP/2.0/UDP [5555::aaa:bbb:ccc:ddd]:1357;comp=sigcomp;branch=z9hG4bKnashds7
Max-Forwards: 68
Route: <sip:orig@scscf1.home1.net;lr>
P-Asserted-Identity:
P-Preferred-Service:
P-Access-Network-Info:
Privacy:
From:
To:
Call-ID:
Cseq:
Require:
Supported:
Proxy-Require:
Contact:
Accept-Contact
Allow:
Content-Type:
Content-Length:
v=0
o=- 22 333 IN IP6 8888::111:222:333:444
s=-
c=IN IP6 8888::111:222:333:444
t=0 0
m=audio 8899 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 none remote sendrecv
a=rtpmap:97 AMR
a=fmtp:97 mode-set=0,2,5,7; maxframes=2
a=rtpmap:96 telephone-event
SDP offer: the IP address and ports are updated to contain the values provided by ATGW .
10-12. SIP 183 (Session Progress) response (remote UE towards SCC AS)
The remote UE responds with SIP 183 (Session progress) response.
13.-15. SIP 183 (Session Progress) response (SCC AS towards ATCF) – see example in table A.4.2-13
The SCC AS forwards the SIP 183 (Session progress) response.
Table A.4.2-13: SIP 183 (Session Progress) response (SCC AS towards ATCF)
SIP/2.0 183 Session Progress
Feature-Caps: *;+g.3gpp.srvcc
Record-Route: <sip:pcscf1.visited1.net;lr>, <sip:atcf.visited.net;lr>, <sip:scscf.home1.net;lr>, <sip:icscf.home1.net;lr>, <sip:sccas.home1.net;lr>
Via: SIP/2.0/UDP sccas.home1.net:5060;branch=z9hG4bKnas522, SIP/2.0/UDP scscf.home1.net:5060;branch=z9hG4bKnas889, SIP/2.0/UDP icscf.home1.net:5060;branch=z9hG4bKnas225, SIP/2.0/UDP atcf.visited2.net:5060;branch=z9hG4bKnas55889, SIP/2.0/UDP pcscf1.visited2.net:5060;branch=z9hG4bKnas56565, SIP/2.0/UDP [5555::aaa:bbb:ccc:ddd]:1357;comp=sigcomp;branch=z9hG4bKnashds7
Max-Forwards: 60
P-Asserted-Identity: <tel:+1-212-555-2222>
Privacy:
From:
To: <tel:+1-212-555-2222>; tag=aaa
Call-ID:
Cseq:
Require:
Supported:
Contact: <sip:user2_public1@home2.net;gr=urn:uuid:2ad8950e-48a5-4a74-8d99-ad76cc7fc74>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Allow:
Content-Type:
Content-Length:
v=0
o=- 462346 5654 IN IP6 1234::55:66:77:88
s=-
c=IN IP6 1234::55:66:77:88
t=0 0
m=audio 4456 RTP/AVP 97 96
b=AS:25.4
a=curr:qos local none
a=curr:qos remote sendrecv
a=des:qos mandatory local sendrecv
a=des:qos mandatory remote sendrecv
a=rtpmap:97 AMR
a=fmtp:97 mode-set=0,2,5,7; maxframes=2
a=rtpmap:96 telephone-event
Feature-Caps: The header field contains
– g.3gpp.srvcc indicating that the session has been anchored in the SCC AS.
Supported: The SCC AS adds the "tdialog" and the "replaces" option tags in the Supported header field header, if not already included. In this example the "tdialog" and the "replaces" option tags were already included.
16. ATGW resource configuration
The ATCF configures the resources of ATGW.
17. SIP 183 (Session Progress) response (ATCF towards UE) – see example in table A.4.2-17
The ATCF replaces the IP address, ports, … in SDP answer with values provided by ATGW.
Table A.4.2-17: SIP 183 (Session Progress) response (ATCF towards UE)
SIP/2.0 183 Session Progress
Feature-Caps: *;+g.3gpp.srvcc
Record-Route: <sip:pcscf1.visited1.net;lr>, <sip:atcf.visited.net;lr>, <sip:scscf.home1.net;lr>, <sip:icscf.home1.net;lr>, <sip:sccas.home1.net;lr>
Via: SIP/2.0/UDP pcscf1.visited2.net:5060;branch=z9hG4bKnas56565, SIP/2.0/UDP [5555::aaa:bbb:ccc:ddd]:1357;comp=sigcomp;branch=z9hG4bKnashds7
Max-Forwards: 60
P-Asserted-Identity: <tel:+1-212-555-2222>
Privacy:
From:
To:
Call-ID:
Cseq:
Require:
Supported:
Contact: <sip:user2_public1@home2.net;gr=urn:uuid:2ad8950e-48a5-4a74-8d99-ad76cc7fc74>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Allow:
Content-Type:
Content-Length:
v=0
o=- 44 555 IN IP6 8888::111:222:333:444
s=-
c=IN IP6 8888::111:222:333:444
t=0 0
m=audio 11234 RTP/AVP 97 96
b=AS:25.4
a=curr:qos local none
a=curr:qos remote sendrecv
a=des:qos mandatory local sendrecv
a=des:qos mandatory remote sendrecv
a=rtpmap:97 AMR
a=fmtp:97 mode-set=0,2,5,7; maxframes=2
a=rtpmap:96 telephone-event
SDP answer: the IP address and ports are updated to contain the values provided by ATGW.
A.4.3 Call origination prior to CS to PS SRVCC
The signalling flow shown in figure A.4.3-1 gives an example of originating session set up. In this flow, the ATCF decides to anchor the media of the session in ATGW.
NOTE: For clarity, the SIP 100 (Trying) responses are not shown in the signalling flow.
Figure A.4.3-1 Signalling flows for service continuity using CS to PS SRVCC.
1. SC UE A registers in IMS
The SC UE A registers in IMS.
2. The SC UE A performs a CS attach and the MSC server registers in IMS.
3. CC SETUP message (SC UE A to MSC server).
The SC UE sends a CC SETUP message according to 3GPP TS 24.008 [8].
4. SIP INVITE request (MSC server to ATCF) – see example in table A.4.3-4
The MSC server enhanced for ICS sends the SIP INVITE request towards the ATCF.
Table A.4.3-4: SIP INVITE request (MSC server to ATCF)
INVITE tel:+1-237-555-2222 SIP/2.0
Via: SIP/2.0/UDP [5555::aaa:bbb:ccc:ddd]:1357;branch=z9hG4bKnashds7
Max-Forwards: 70
Route: <sip:atcf2.visited2.net:7531;lr><sip:orig@scscf1.home1.net;lr>
P-Asserted-Identity: tel:+1-212-555-1111
P-Charging-Vector: icid-value="1234bc9876e";icid-generated-at"5555::aaa:bbb:ccc:ddd";orig-ioi=visited2.net
Privacy: none
From: <tel:+1-237-555-1111>;tag=171828
To: < tel:+1-237-555-2222 >
P-Access-Network-Info: 3GPP-UTRAN-TDD; utran-cell-id-3gpp=234151D0FCE11
P-Visited-Network-ID: "Visited Network Number 1 for MSC Server"
Call-ID: cb03a0s09a2sdfglkj490333
Cseq: 127 INVITE
Supported: precondition, 100rel
Accept: application/vnd.3gpp.access-transfer-events+xml;et="2"
Recv-Info: g.3gpp.access-transfer-events
Contact: <sip:user1_public1@visited2.net>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";+g.3gpp.ti="F0CA"
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER, MESSAGE
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=audio 3456 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 none remote sendrecv
a=rtpmap:97 AMR
a=fmtp:97 mode-set=0,2,5,7; maxframes=2
a=rtpmap:96 telephone-event
–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>initiator</direction>
</Setup-info>
</srvcc-ext>
–boundary1–
Route: The ATCF management URI received from SCC AS in the SIP MESSAGE request containing CS to PS SRVCC information during the registration of the user is added at the top of the URIs received in the Service-Route header field of the SIP 200 (OK) response to REGISTER.
application/vnd.3gpp.srvcc-ext+xml: Contains the direction of call and the C-MSISDN of the UE.
Accept: Indicate that the MSC server is able to receive the application/vnd.3gpp.access-transfer-events+xml with the event type 2.
Recv-Info: Indicate the support for g.3gpp.access-transfer-events package.
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 74 (decimal) and the transaction identifier flag as sent by the MSC server in CS signalling of the originating CS call are shown.
5. 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.
6-9. SIP INVITE request (ATCF to remote UE B) – see example in table A.4.3-6
Table A.4.3-6: SIP INVITE request (ATCF to remote UE B)
INVITE tel:+1-237-555-2222 SIP/2.0
Record-Route: <sip:atcf2.visited2.net;lr>
Via: SIP/2.0/UDP atcf.visited2.net:5060;branch=z9hG4bKnas55889, SIP/2.0/UDP [5555::aaa:bbb:ccc:ddd]:1357;;branch=z9hG4bKnashds7
Max-Forwards:
Route: <sip:orig@scscf1.home1.net;lr>
P-Asserted-Identity:
P-Access-Network-Info:
P-Charging-Vector:
Privacy:
From:
To:
Call-ID:
Cseq:
Require:
Supported:
Proxy-Require:
Contact:
Accept-Contact
Allow:
Content-Type:
Content-Length:
v=0
o=- 22 333 IN IP6 8888::111:222:333:444
s=-
c=IN IP6 8888::111:222:333:444
t=0 0
m=audio 8899 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 none remote sendrecv
a=rtpmap:97 AMR
a=fmtp:97 mode-set=0,2,5,7; maxframes=2
a=rtpmap:96 telephone-event
SDP offer: The IP address and ports are updated to contain the values provided by ATGW.
10-11. SIP 183 (Session Progress) response (remote UE B to SCC AS)
The remote UE B responds with SIP 183 (Session progress) response.
12-13. SIP 183 (Session Progress) response (SCC AS to ATCF) – see example in table A.4.3-13
Table A.4.3-13: SIP 183 (Session Progress) response (SCC AS towards ATCF)
SIP/2.0 183 Session Progress
Record-Route: <sip:atcf2.visited2.net;lr>,<sip:icscf1.home1.net;lr>,<sip:scscf1.home1.net;lr>, <sip:sccas1.home1.net;lr>;
Via: SIP/2.0/UDP scscf1.home1.net:5060;branch=z9hG4bKnas889, SIP/2.0/UDP atcf.visited2.net:5060;branch=z9hG4bKnas55889, SIP/2.0/UDP [5555::aaa:bbb:ccc:ddd]:1357;branch=z9hG4bKnashds7
Max-Forwards:
P-Asserted-Identity: <tel:+1-237-555-2222>
P-Charging-Vector: icid-value="1234bc9876e@5555::aaa:bbb:ccc:ddd";orig-ioi= visited2.net
Privacy:
From:
To: <tel:+1-237-555-2222>; tag=aaa
Call-ID:
Cseq:
Require:
Supported:
Contact: <sip:user2_public1@home2.net;gr=urn:uuid:2ad8950e-48a5-4a74-8d99-ad76cc7fc74>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Allow:
Content-Type:
Content-Length:
Feature-Caps: *;+g.3gpp.srvcc
v=0
o=- 462346 5654 IN IP6 1234::55:66:77:88
s=-
c=IN IP6 1234::55:66:77:88
t=0 0
m=audio 4456 RTP/AVP 97 96
b=AS:25.4
a=curr:qos local none
a=curr:qos remote sendrecv
a=des:qos mandatory local sendrecv
a=des:qos mandatory remote sendrecv
a=rtpmap:97 AMR
a=fmtp:97 mode-set=0,2,5,7; maxframes=2
a=rtpmap:96 telephone-event
14. Configure ATGW resources.
The ATCF configures the resources of ATGW.
15. SIP 183 (Session Progress) response (ATCF to MSC server) – see example in table A.4.3-15
Table A.4.3-15: SIP 183 (Session Progress) response (ATCF to MSC server)
SIP/2.0 183 Session Progress
Record-Route: <sip:atcf2.visited2.net;lr>, <sip:scscf1.home1.net;lr>, <sip:icscf1.home1.net;lr>, <sip:sccas1.home1.net;lr>
Via: SIP/2.0/UDP [5555::aaa:bbb:ccc:ddd]:1357;
Max-Forwards:
P-Asserted-Identity:
P-Charging-Vector:
Privacy:
From:
To:
Call-ID:
Cseq:
Require:
Supported:
Recv-Info: g.3gpp.access-transfer-events;et="1,3,4"
Contact: <sip:user2_public1@home2.net;gr=urn:uuid:2ad8950e-48a5-4a74-8d99-ad76cc7fc74>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Allow:
Content-Type:
Content-Length:
v=0
o=- 44 555 IN IP6 8888::111:222:333:444
s=-
c=IN IP6 8888::111:222:333:444
t=0 0
m=audio 11234 RTP/AVP 97 96
b=AS:25.4
a=curr:qos local none
a=curr:qos remote sendrecv
a=des:qos mandatory local sendrecv
a=des:qos mandatory remote sendrecv
a=rtpmap:97 AMR
a=fmtp:97 mode-set=0,2,5,7; maxframes=2
a=rtpmap:96 telephone-event
SDP answer: the IP address and ports are updated to contain the values provided by ATGW.
Recv-Info: Indicates the support for the info package g.3gpp.access-transfer-events and is able to receive the event types 1, 3 and 4.
16. CC CALL CONFIRM message (MSC server to SC UE A)
Regular call setup continues.