Ims call flow

IMS/MMD Call Flow Examples X.S0013-009-0 v1.0
IMS/MMD Call Flow Examples
CONTENTS
1 Introduction 1
2 Glossary and Definitions 2
2.1 Acronyms.......................................................................................................................................2
2.2 Definitions.....................................................................................................................................2
3 References 3
3.1 Normative References....................................................................................................................3
3.2 Informative References..................................................................................................................3
4 Methodology 4
4.1 Functional Entities covered by call flows......................................................................................4
4.2 Identities.........................................................................................................................................4
4.2.1 User and Network Identities...........................................................................................4
4.3 Notation for call flows...................................................................................................................5
5 Registration Procedures 5
6 Signaling flows for session establishment 6
6.1 General assumptions......................................................................................................................6
6.2 Scenarios........................................................................................................................................6
6.2.1 Scenario 1.......................................................................................................................7
6.2.1.1 Assumptions..................................................................................................7
6.2.1.2 Call Flow.......................................................................................................7
6.2.2 Scenario 2.....................................................................................................................12
6.2.2.1 Assumptions................................................................................................12
6.2.2.2 Call Flow.....................................................................................................12
6.2.3 Scenario 3.....................................................................................................................13
6.2.3.1 Assumptions................................................................................................13
6.2.3.2 Call flow......................................................................................................13
6.2.4 Scenario 4.....................................................................................................................20
6.2.4.1 Assumptions................................................................................................20
6.2.4.2 Call flow......................................................................................................20
Appendix A (Informative): VoIP example with QoS reservation, activation, and updating at RAN 28
A.1 QoS Configuration for VoIP in advance 28
A.2 Resource activation on originating side 30
A.3 Resource activation on terminating side 31
A.4 Updating of resource reservation 32
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X.S0013-009-0 v1.0 IMS/MMD Call Flow Examples
A.4.1 Resource updating on originating side 32
A.4.1 Resource updating on terminating side 34
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LIST OF FIGURES
Figure 1 Originating UE resource ready, terminating UE resource ready 7
Figure 2 Originating UE resource ready, terminating UE resource not ready 12
Figure 3 Originating UE not ready, Terminating UE ready 14
Figure 4 Originating UE not ready, Terminating UE not ready 21
Figure 5 QoS configuration for VoIP 29
Figure 6 QoS Activation on Originating UE 30
Figure 7 QoS Activation on Terminating UE 31
Figure 8 Resource Updating on Originating UE 33
Figure 9 Resource Updating on Terminating UE 34
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Revision History
Revision Changes Date
v1.0 Initial Publication December, 2007
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1 Introduction
The document provides examples of signaling flows for the IP multimedia call control based on
SIP and SDP. The signaling flows specified in this document are only for informational purposes.
If there is ambiguity between this specification and [1], the text specified in [1] shall be followed.
The call flows describe the behavior of the mobile stations under various conditions for setting up
real time services like VoIP and PSVT.
In this document, several key words are used to signify the requirements. The key words “shall”,
“shall not”, “should”, “should not” and “may” are to be interpreted as described in [6] and the TIA
Engineering Style Manual.
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2 Glossary and Definitions
2.1 Acronyms
AAA Authentication, Authorization, and Accounting
AN Access Network
AT Access Terminal
EMPA Enhanced Multi-Flow Packet Application
HRPD High Rate Packet Data
HSS Home Subscriber Server
IMS IP Multimedia Subsystem
IP Internet Protocol
IP-CAN IP-Connectivity Access Network
MMD Multi Media Domain
MPA Multi-Flow Packet Application
PDSN Packet Data Serving Node
PFO PPP Free Operation
PPP Point to Point Protocol
PSVT Packet Switched Video Telephony
QoS Quality of Service
RAN Radio Access Network
RSVP Resource ReSerVation Protocol
SBBC Service Based Bearer Control
SDP Session Description Protocol
SIP Session Initiation Protocol
TFT Traffic Flow Template
UDP User Datagram Protocol
UE User Equipment
VoIP Voice Over IP
2.2 Definitions
Caller The person placing a call
Called party The recipient or destination of a call
Alert The audible notification given to the Called party of an incoming call.
Ring Back The audible notification given to a Caller to indicate that the Called
party has been located and is being alerted
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3 References
The following documents contain provisions, which, through reference in this text, constitute
provisions of this document.
 References are either specific (identified by date of publication, edition number, version
number, etc.) or non-specific.
 For a specific reference, subsequent revisions do not apply.
 For a non-specific reference, the latest version applies. In the case of a reference to a
3GPP2 document, a non-specific reference implicitly refers to the latest version of that
document in the same Release as the present document.
3.1 Normative References
[1] 3GPP2 X.S0013-004-A v1.0: “IP Multimedia Call Control Protocol Based on SIP and SDP
Stage 3”.
[2] 3GPP2 X.S0013-002-A v1.0: “IP Multimedia (IM) Subsystem – Stage 2”.
[3] IETF RFC 2429, “ RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video
(H.263+)”.
[4] IETF RFC 3558, “RTP Payload Format for Enhanced Variable Rate Codecs (EVRC) and
Selectable Mode Vocoders (SMV)”.
[5] IETF RFC 3262, “Reliability of provisional Responses in the Session Initiation Protocol
(SIP)”
[6] IETF RFC 2119, “Key words for use in RFCs to Indicate Requirement Levels”, March
1997.
[7] 3GPP2 X.S0013-005-A v1.0: “All-IP Core Network Multimedia Domain IP Multimedia
Subsystem Cx Interface Signaling flows and Message Contents” – Annex A.
3.2 Informative References
[8] 3GPP2 X.S0013-012-0 v1.0: “All-IP Core Network Multimedia Domain; Service Based
Bearer Control – Stage 2”.
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4 Methodology
4.1 Functional Entities covered by call flows
The flows show the signaling exchanges between the following functional entities:
 User Equipment (UE)
 Proxy-CSCF (P-CSCF)
 Interrogating-CSCF (I-CSCF)
 Serving-CSCF (S-CSCF)
The call flows mainly show the interactions between the UE-1 and UE-2 for call origination and
termination. The procedures and the message exchanged between these elements are as described in [1].
4.2 Identities
4.2.1 User and Network Identities
The public identity of UE-1 is sip:UE-1@domain1.net. The public identity of UE-2 is sip:UE-
2@domain2.net
The following are network entities associated with UE-1:
 Home Domain of UE-1: domain1.net
 P-CSCF serving UE-1: p-cscf1.domain1.net
 I-CSCF serving UE-1: i-cscf1.domain1.net
 S-CSCF serving UE-1: s-cscf1.domain1.net
The following are the network entities associated with UE-2:
 Home Domain of UE-2: domain2.net
 P-CSCF serving UE-2: p-cscf2.domain2.net
 I-CSCF serving UE-2: i-cscf2.domain2.net
 S-CSCF serving UE-2: s-cscf2.domain2.net
In the example session establishment flows, both UE-1 and UE-2 are assumed to be in the home
network. So, the P-CSCF1 and P-CSCF2 are in UE-1’s and UE-2’s respective home networks.
However, according to [2], the P-CSCF can be either in the home or visited network. The session
establishment call flows described in this document are not affected whether the P-CSCF is in the
visited or home network.
For brevity, I-CSCF and S-CSCF are shown together in the session establishment call flows.
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4.3 Notation for call flows
Offer/Answer exchange process is also shown in the call flows. For brevity, the following notation is used to
represent offer and answer:
• “O” and “A” in the call flows represent “Offer” and “Answer”.
• “On” represents the nth
SDP offer. For example, if there are two offer/answer exchanges during the call
setup process, then the first offer will be noted as “O1” and the second as “O2”.
• “An” represents the nth
SDP Answer. Answer “An” corresponds to Offer “On”.
5 Registration Procedures
The registration procedures for UE-1 and UE-2 are as specified in [1] and [7]. UE-1 registers with S-
CSCF1 through P-CSCF1 and UE-2 registers with S-CSCF2 through P-CSCF2.
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6 Signaling flows for session establishment
In all of the call flows provided in this document, registration procedures are assumed to have already been
completed.
6.1 General assumptions
All the call flows shown in this document assume the following:
• The originating UE and the terminating UE both support precondition and reliable provisional responses
100 (rel).
• The originating UE will only include “Supported: precondition” in the SIP INVITE it sends out to its peer.
o If the originating UE wishes to both send and receive media with its peer, and the resources are
reserved for the stream, the originating UE marks the stream as sendrecv using the “a=sendrecv”
attribute. (Note: for sendonly streams, the originating UE marks the stream as “a=sendonly” and
for recvonly streams, the originating UE marks the stream as “a=recvonly”).
o If the originating UE wishes to communicate with its peer, and the resources are not reserved for
the stream, the originating UE marks the particular stream as inactive using the “a=inactive”
attribute.
• If the “Supported” header in the incoming INVITE includes “precondition” and the terminating UE decides
to use precondition, the terminating UE includes “Require: precondition” in all reliable responses that carry
SDP (i.e., offer or answer).
• The terminating UE sends 180 (Ringing) response reliably. Note that sending the 180 (Ringing) response
reliably does not increase the call setup time experienced by the caller.
• If the 180 (Ringing) response contains an answer (or offer), the called party is alerted only after receiving a
PRACK request for the 180 (Ringing) response. This enhances the caller/called party user experience. For
example, if the called party picks up the phone before the 180 (Ringing) response reaches the caller, the
caller will only be able to hear the called party but will not be able to respond,
• Both UEs have the required resources ready before the terminating UE can alert the called party of an
incoming call.
• For brevity, 100 (Trying) messages are not shown in the figures.
• The SBBC interactions [8] are not shown in the call flows.
6.2 Scenarios
The following scenarios are considered for the session establishment process:
• Scenario 1: Originating UE has resources ready before sending INVITE, terminating UE has resources
ready before sending the first provisional response;
• Scenario 2: Originating UE has resources ready before sending INVITE, terminating UE does not have
resources ready before sending the first provisional response;
• Scenario 3: Originating UE does not have resources ready before sending INVITE, terminating UE has
resources ready before sending the first provisional response;
• Scenario 4: Originating UE does not have resources ready before sending INVITE, terminating UE does
not have resources ready before sending the first provisional response.
The call flows for the above four scenarios are depicted in the subsequent subsections.
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6.2.1 Scenario 1
This section covers the scenario where the originating UE’s resources are ready before sending INVITE, and the
terminating UE’s resources are ready before sending the first provisional response.
6.2.1.1 Assumptions
6.2.1.2 Call Flow
The following section applies to the case where UE-1 has its resource ready before sending the
INVITE request to UE-2. The terminating UE, UE-2, also has its resource ready before answering the
INVITE request with the first provisional response.
Note: This flow assumes UE-2 has sufficient resources available prior to receiving the INVITE.
Figure 1 Originating UE resource ready, terminating UE resource ready
1. INVITE (UE-1 to P-CSCF1)
UE-1 determines the set of codecs or media streams that it wishes to support for the session. It builds an SDP offer
containing characteristics of each codec, and assigns local port numbers for each possible media flow. Multiple
media flows may be offered, and for each media flow (m= line in SDP), there may be multiple codec choices
offered.
For this example, it is assumed that UE-1 is willing to establish a multimedia session comprising a video stream and
an audio stream. The video stream supports one H.263 codec as specified in [3]. The audio stream supports both
EVRC and SMV codecs as specified in [4].
In addition, UE-1 indicates that precondition is supported for this session. In the SDP offer, it indicates that resource
is already available at the local end point.
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Table 6.2.1.2-1 INVITE (UE-1 to P-CSCF1)
INVITE sip:UE-2@domain2.net SIP/2.0
From: <sip:UE-1@domain1.net>;tag=169f498-0-13c4-78e-2044f20e-78e
To: <sip:UE-2@domain2.net>
Call-ID: 16a1b80-0-13c4-78e-6a540802-78e@10.20.1.100
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.20.1.100:1357;comp=sigcomp;branch=z9hG4bK-78f-1d86b6-24d29348
Max-Forwards: 70
Route: <sip:pcscf1.domain1.net:7531;lr;comp=sigcomp>, <sip:scscf1.domain1.net;lr>
P-Preferred-Identity: "User-1" <sip:UE-1@domain1.net>
P-Access-Network-Info: 3GPP2-1X-HRPD; ci-3gpp2=1234123412341234123412341234123411
Privacy: none
Require: sec-agree
Proxy-Require: sec-agree
Security-Verify: ipsec-3gpp; q=0.1; alg=hmac-sha-1-96;spi-c=98765432; spi-s=76543210;
port-c=13579; port-s=23456
Contact: <sip:UE-1@10.20.1.100:1357;comp=sigcomp>
Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER, MESSAGE
Supported: 100rel, precondition
Content-Type: application/sdp
Content-Length: xxx
v=0
o=- 3323527065117000 3323527065117000 IN IP4 10.20.1.100
s=-
c=IN IP4 10.20.1.100
t=0 0
m=audio 49500 RTP/AVP 97 99
b=AS:25.4
a=curr:qos local sendrecv
a=curr:qos remote none
a=des:qos optional local sendrecv
a=des:qos optional remote sendrecv
a=rtpmap:97 EVRC/8000
a=ptime:20
a=rtpmap:99 SMV/8000
m=video 49600 RTP/AVP 34
b=AS:75
a=rtpmap:34 H263/90000
a=cif:1
2. INVITE (P-CSCF1 to I/S-CSCF1)
The P-CSCF1 adds itself to the Record-Route header and Via header. As the request is forwarded to an interface that
is not compressed, the P-CSCF1 SIP URI does not contain the "comp=sigcomp" parameter. The P-CSCF1 removes
the Security-Verify header and associated "sec-agree" option-tags prior to forwarding the request. As the Require
and Proxy-Require headers are empty, P-CSCF1 removes those headers completely.
The INVITE request is forwarded to the I/S-CSCF1.
3. INVITE (I/S-CSCF1 to I/S-CSCF2)
S-CSCF1 performs an analysis of the destination address, and determines the network operator to whom the
destination subscriber belongs. Since the originating operator does not desire to keep their internal configuration
hidden, S-CSCF1 forwards the INVITE request directly to I-CSCF2 in the destination network.
The I-CSCF2 sends a query to the HSS to find out the S-CSCF2 of the called user. The HSS responds with the
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address of the current S-CSCF2 for the terminating subscriber. I-CSCF2 forwards the INVITE request to the S-
CSCF2 that will handle the session termination. S-CSCF2 validates the service profile of this subscriber and
evaluates the initial filter criteria.
4-5. INVITE (I/S-CSCF2 to UE-2)
S-CSCF2 forwards the INVITE request to UE-2 via P-CSCF2.
6. 180 Ringing (UE-2 to P-CSCF2)
UE-2 has accepted both video and audio streams, and EVRC is the chosen codec for the audio stream. Since
resources are already available for UE-2, it sends a 180 (Ringing) response reliably with the SDP answer indicating
that resources are reserved at both endpoints.
7-10. 180 Ringing (P-CSCF2 to UE-1)
P-CSCF2 forwards the 180 (Ringing) response to UE-1 via I/S-CSCF2, I/S-CSCF1, and P-CSCF1. Table 6.2.1.2-2
shows the 180 (Ringing) response in detail.
Table 6.2.1.2-2 180 Ringing (P-CSCF1 to UE-1)
SIP/2.0 180 Ringing
To: <sip:UE-2@domain2.net>;tag=169f7f8-0-13c4-9f6-33835972-9f6
Call-ID: 16a1b80-0-13c4-78e-6a540802-78e@10.20.1.100
CSeq: 1 INVITE
Record-Route:<sip:pcscf2.domain2.net;lr>,<sip:scscf2.domain2.net;lr>,<sip:scscf1.domain1.net;lr>,
<sip:pcscf1.domain1.net:7531;lr;comp=sigcomp>
P-Asserted-Identity: "User 2" <sip:UE-2@domain2.net>, <tel:+1-858-335-7341>
Privacy: none
Require: 100rel, precondition
RSeq: 1000
Content-Length: xxx
v=0
o=- 33235270718000 33235270718000 IN IP4 10.30.1.24
s=-
c=IN IP4 10.30.1.24
t=0 0
m=audio 49700 RTP/AVP 97
a=curr:qos remote sendrecv
a=ptime:20
a=rtpmap:34 H263/90000
a=cif:1
11. PRACK (UE-1 to P-CSCF1)
UE-1 acknowledges the 180 (Ringing) response from UE-2 with a PRACK request as specified in [5]. (Table
6.2.1.2-3)
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If UE-1 determines to make any change in media flows, it includes a new SDP offer in the PRACK request sent to
UE-2. Otherwise, no SDP offer is included in the PRACK request.
Table 6.2.1.2-3 PRACK (UE-1 to P-CSCF1)
PRACK sip:UE-2@10.30.1.24:8805;comp=sigcomp SIP/2.0
To: <sip:UE-2@domain2.net>; tag=169f7f8-0-13c4-9f6-33835972-9f6
Call-ID: 16a1b80-0-13c4-78e-6a540802-78e@10.20.1.100
P-Access-Network-Info: 3GPP2-1X-HRPD;ci-3gpp2=1234123412341234123412341234123411
CSeq: 2 PRACK
Max-Forwards: 70
Route:
<sip:pcscf1.domain1.net:7531;lr;comp=sigcomp>,<sip:scscf1.domain1.net;lr>,<sip:scscf2.d
omain2.net;lr>,
<sip:pcscf2.domain2.net;lr>
RAck: 1000 1 INVITE
Content-Length: 0
12. PRACK (P-CSCF1 to I/S-CSCF1)
The P-CSCF forwards the PRACK request to S-CSCF. As the Proxy-Require header is empty, the P-CSCF removes
this header completely.
13-14. PRACK (I/S-CSCF1 to P-CSCF2)
The I/S-CSCF1 forwards the PRACK request to P-CSCF2 via I/S-CSCF2.
15. PRACK (P-CSCF2 to UE-2)
UE-2 starts alerting the user after it receives the PRACK request for the 180 (Ringing) response. UE-2 may also
alert the user before receiving the PRACK request, but there may be media clipping if the user answers the call
before the 180 (Ringing) response reaches UE-1.
16. 200 OK (UE-2 to P-CSCF2)
The 200 OK response is generated by UE-2 to acknowledge the reception of the PRACK request.
17-20. 200 OK (P-CSCF2 to UE-1)
The P-CSCF2 forwards the 200 OK response to UE-1 via I/S-CSCF2, I/S-CSCF1, and P-CSCF1. Table 6.2.1.2-4
shows the 200 OK message going from P-CSCF1 to UE-1.
Table 6.2.1.2-4 200 OK (P-CSCF1 to UE-1)
SIP/2.0 200 OK
Call-ID: 16a1b80-0-13c4-78e-6a540802-78e@10.20.1.100
CSeq: 2 PRACK
Content-Length: 0
21. 200 OK (UE-2 to P-CSCF2)
When the user at UE-2 answers the call, UE-2 generates a 200 OK response towards UE-1 to answer the INVITE
request.
22-25. 200 OK (P-CSCF2 to UE-1)
The P-CSCF2 forwards the 200 OK response to UE-1 via I/S-CSCF2, I/SCSCF1, and P-CSCF1. Table 6.2.1.2-5
shows the 200 OK message going from P-CSCF1 to UE-1.
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Table 6.2.1.2-5 200 OK (P-CSCF1 to UE-1)
SIP/2.0 200 OK
Call-ID: 16a1b80-0-13c4-78e-6a540802-78e@10.20.1.100
CSeq: 1 INVITE
Content-Length: 0
26. ACK (UE-1 to P-CSCF1)
UE responds to the 200 OK with an ACK request sent to P-CSCF1. (Table 6.2.1.2-6).
Table 6.2.1.2-6 ACK (UE-1 to P-CSCF1)
27-30. ACK (P-CSCF1 to UE-2)
The P-CSCF1 forwards the ACK response to UE-2 via I/S-CSCF1, I/S-CSCF2, and P-CSCF2.
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ACK sip:UE-2@10.30.1.24:8805;comp=sigcomp SIP/2.0
Call-ID: 16a1b80-0-13c4-78e-6a540802-78e@10.20.1.100
CSeq: 1 ACK
Via: SIP/2.0/UDP 10.20.1.100:5060;comp=sigcomp;branch=z9hG4bK-792-1d9290-49ff0c31
Max-Forwards: 70
Route: <sip:pcscf1.domain1.net:7531;lr;comp=sigcomp>,<sip:scscf1.domain1.net;lr>,
<sip:scscf2.domain2.net;lr>,
<sip:pcscf2.domain2.net;lr>
Content-Length: 0
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6.2.2 Scenario 2
This section covers the scenario where the originating UE’s resources are ready before sending INVITE, and the
terminating UE’s resources are not ready before sending the first provisional response
6.2.2.1 Assumptions
6.2.2.2 Call Flow
This scenario assumes that the terminating UE, UE-2, does not have the resource ready before sending the first
provisional response. UE-2 may send a 183 (Session Progress) response unreliably. At the same time, UE-2 also
starts the resource reservation process. Once the resource is ready at UE-2 side, the UE-2 sends 180 (Ringing)
response reliably to UE-1, including an SDP answer to indicate that the resource is ready. The SDP offer/answer
exchanges are the same as those in Scenario 1.
Figure 2 Originating UE resource ready, terminating UE resource not ready
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6.2.3 Scenario 3
This section covers the scenario where the originating UE’s resources are not ready before sending INVITE, and
terminating UE’s resources are ready before sending the first provisional response.
6.2.3.1 Assumptions
The call flow in this section assumes that the originating UE, UE-1, has resources ready before sending the PRACK
request to the first provisional response. In case UE-1’s resources are not ready before sending the PRACK request
for the first provisional response (183), then UE-1 needs to send an UPDATE request to UE-2 to indicate that
resources are ready, after the resource reservation is completed.
6.2.3.2 Call flow
The following section applies to the case where UE-1 has no resource reserved before sending the initial INVITE
request to UE-2.
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Figure 3 Originating UE not ready, Terminating UE ready
1-5. INVITE (UE-1 to P-CSCF1)
UE-1 determines the set of codecs or media streams that it wishes to support for the session. It builds a SDP
offered.
For this example, it is assumed that UE-1 desires to establish a multimedia session comprising a video stream and an
audio stream. The video stream supports one H.263 codec. The audio stream supports both EVRC and SMV codec.
UE-1 does not indicate that precondition is required for this session, but indicates that it is supported. (This approach
optimizes compatibility and performance when interworking with 3rd
party (non-MMD) terminals). In the SDP body,
UE-1 indicates the current resource status and that the desired resource status is optional. UE-1 also sets every media
stream to inactive mode by using the ‘a=inactive’ SDP attribute in the SDP offer. Detecting the QoS precondition
content in the SDP, UE-2 indicates resource reservation, but the session can continue regardless of whether or not
this reservation is possible.
UE-1 does not indicate that reliable provisional responses are required, but indicates that they are supported. This
gives UE-2 the ability to reliably send only those responses that are most appropriate.
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Table 6.2.3.2-7 INVITE (UE-1 to P-CSCF1)
Via: SIP/2.0/UDP 100.200.1.1:1357;comp=sigcomp;branch=z9hG4bK4d29348
From: <sip:UE-1@domain1.net>;tag= a48s
Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@domain1.net
CSeq: 1 INVITE
Max-Forwards: 70
Privacy: none
Require: sec-agree
Security-Verify: ipsec-3gpp; q=0.1; alg=hmac-sha-1-96; spi-c=9876543; spi-s=3456789;
port-c=8642; port-s=7531
Content-Length: xxx
v=0
o=- 2987935614 2987935614 IN IP4 100.200.1.1
s=-
c=IN IP4 100.200.1.1
t=0 0
b=AS:25.4
a=inactive
a=curr:qos local none
a=ptime:20
b=AS:75
a=inactive
a=rtpmap:34 H263/90000
a=cif:1
6-10. 183 (P-CSCF1 to UE-1)
UE-2 accepts both video and audio streams, and chooses EVRC as the codec for the audio stream. An SDP answer is
included to assist UE-1 in completing resource reservation as early as possible. The SDP answer includes
precondition status.
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3
4
5
6
7
8
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Table 6.2.3.2-8 183 Session Progress (P-CSCF1 to UE-1)
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 100.200.1.1:1357;comp=sigcomp;branch= z9hG4bK4d29348
From: <sip:UE-1@domain1.net>;tag=a48s
To: <sip:UE-2@domain2.net>;tag=a9f6
CSeq: 1 INVITE
Record-Route:<sip:pcscf2.domain2.net;lr>,<sip:scscf2.domain2.net;lr>,
<sip:scscf1.domain1.net;lr>,<sip:pcscf1.domain1.net:7531;lr;comp=sigcomp>
Privacy: none
RSeq: 1000
Content-Length: xxx
v=0
o=- 35270718123 35270718123 IN IP4 100.300.1.2
s=-
c=IN IP4 100.300.1.2
t=0 0
b=AS:25.4
a=inactive
a=conf:qos remote sendrecv
a=ptime:20
b=AS:75
a=inactive
a=rtpmap:34 H263/90000
a=cif:1
11-15. PRACK (UE-1 to P-CSCF1)
UE-1 acknowledges the 183 (Session Progress) response from UE-2 with a PRACK request. Resource reservation at
UE-1 is assumed to have been completed at some point prior to sending the PRACK request. The local resource
status is included in the SDP offer.
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3
4
5
6
7
8
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Table 6.2.3.2-3 – PRACK (UE-1 to P-CSCF1)
CSeq: 2 PRACK
Max-Forwards: 70
<sip:scscf2.domain2.net;lr>, <sip:pcscf2.domain2.net;lr>
Require: precondition
RAck: 1000 1 INVITE
Content-Length: xxx
v=0
o=- 2987935614 2987935615 IN IP4 100.200.1.1
s=-
c=IN IP4 100.200.1.1
t=0 0
b=AS:25.4
a=sendrecv
a=ptime:20
b=AS:75
a=sendrecv
a=rtpmap:34 H263/90000
a=cif:1
16-20. 200 OK (P-CSCF1 to UE-1)
The 200 OK response is generated by UE-2 to acknowledge the reception of the PRACK request. UE-2 has
completed resource reservation for both video and audio streams. The response will include an SDP answer to
reflect successful resource reservation.
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7
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Table 6.2.3.2-4 - 200 OK (P-CSCF1 to UE-1)
SIP/2.0 200 OK
CSeq: 2 PRACK
Content-Length: xxx
v=0
o=- 35270718123 35270718124 IN IP4 100.300.1.2
s=-
c=IN IP4 100.300.1.2
t=0 0
b=AS:25.4
a=sendrecv
a=ptime:20
b=AS:75
a=sendrecv
a=rtpmap:34 H263/90000
a=cif:1
UE-2 generates a 180 (Ringing) response to UE-1 and sends it reliably. UE-2 also alerts the called user of the
incoming call.
Table 6.2.3.2-5 - 180 Ringing (P-CSCF1 to UE-1)
SIP/2.0 180 Ringing
CSeq: 1 INVITE
Privacy: none
Require: 100rel
RSeq: 1001
Content-Length: 0
UE-1 acknowledges the 180 (Ringing) response from UE-2 with a PRACK request.
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3
4
5
6
7
8
9
10
11
2

Table 6.2.3.2-6 - PRACK (UE-1 to P-CSCF1)
CSeq: 3 PRACK
Max-Forwards: 70
RAck: 1001 1 INVITE
Content-Length: 0
31-35. 200 OK (P-CSCF1 to UE-1)
SIP/2.0 200 OK
CSeq: 3 PRACK
Content-Length: 0
36-40. 200 OK (P-CSCF1 to UE-1)
request.
SIP/2.0 200 OK
CSeq: 1 INVITE
Record-
Route:<sip:pcscf2.domain2.net;lr>,<sip:scscf2.domain2.net;lr>,<sip:scscf1.domain1.net;
lr>, <sip:pcscf1.domain1.net:7531;lr;comp=sigcomp>
Content-Length: 0
41-45. ACK (UE-1 to P-CSCF1)
UE-1 sends an ACK request in response to the 200 OK response from UE-2.
Table 6.2.3.2-9 - ACK (UE-1 to P-CSCF1)
19
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
1
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5
6
7
8
9
10
11
12
13
14
15
2

6.2.4 Scenario 4
This section covers the scenario where the originating UE’s resources are not ready before sending INVITE,
and the terminating UE’s resources are not ready before sending the first provisional response.
6.2.4.1 Assumptions
This scenario assumes that the originating UE, UE-1, completes resource reservation before, or shortly
after, the arrival of the 183 (Session Progress) response from UE-2, and that the terminating UE completes
local resource reservation prior to sending 180 (Ringing) message.
In case the originating UE, UE-1, has not completed its resource reservation before sending the PRACK
request to the first provisional response (183), the UE-1 needs to send an UPDATE request to indicate
resource ready once it completes the resource reservation. Otherwise , if the terminating UE, UE-2, cannot
finish resource reservation before the offer/answer wherein the UE-1 indicates resources are ready, UE-2
can send a 180 (Ringing) response as soon as its resources are reserved without going through further
offer/answer exchanges.
Note: The UE-1/UE-2 may start reservation anytime after it sends/receives the initial INVITE.
6.2.4.2 Call flow
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Figure 4 Originating UE not ready, Terminating UE not ready
1-5. INVITE (UE-1 to P-CSCF1)
UE-1 determines the set of codecs or media streams that it wishes to support for the session. It builds a SDP
offered.
For this example, it is assumed that UE-1 desires to establish a multimedia session comprising a video stream and an
audio stream. The video stream supports one H.263 codec. The audio stream supports both EVRC and SMV codecs.
UE-1 does not indicate that precondition is required for this session, but that it is supported. (This approach
optimizes compatibility and performance when interworking with 3rd
party (non-MMD) terminals). In the SDP body,
UE-1 indicates the current resource status and that the desired resource status is optional. UE-1 also sets every media
stream to inactive mode by using the ‘a=inactive’ SDP attribute in the SDP offer. Detecting the QoS precondition
content in the SDP, UE-2 indicates resource reservation, but the session can continue regardless of whether or not
this reservation is possible.
UE-1 does not indicate that reliable provisional responses are required, but indicates that they are supported. This
gives UE-2 the ability to reliably send only those responses that are most appropriate.
UE-1 may perform local resource reservation at any time it feels it has enough information to begin this process.
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This is assumed to be after sending the INVITE request, but could be sooner.
Table 6.2.4.2-1 - INVITE (UE-1 to P-CSCF1)
From: <sip:UE-1@domain1.net>;tag= a48s
CSeq: 1 INVITE
Max-Forwards: 70
Privacy: none
Require: sec-agree
Security-Verify: ipsec-3gpp; q=0.1; alg=hmac-sha-1-96; spi-c=9876543; spi-s=3456789;
port-c=8642; port-s=7531
Content-Length: xxx
v=0
o=- 2987935614 2987935614 IN IP4 100.200.1.1
s=-
c=IN IP4 100.200.1.1
t=0 0
b=AS:25.4
a=inactive
a=ptime:20
b=AS:75
a=inactive
a=rtpmap:34 H263/90000
a=cif:1
6-10. 183 Progress (P-CSCF1 to UE-1)
UE-2 accepts both video and audio streams, and chooses EVRC as the codec for the audio stream. An SDP answer is
included to assist UE-1 in completing resource reservation as early as possible. The SDP answer includes
precondition status.
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Table 6.2.4.2-2 - 183 Session Progress (P-CSCF1 to UE-1)
SIP/2.0 183 Session Progress
CSeq: 1 INVITE
Privacy: none
RSeq: 1000
Content-Length: xxx
v=0
o=- 35270718123 35270718123 IN IP4 100.300.1.2
s=-
c=IN IP4 100.300.1.2
t=0 0
b=AS:25.4
a=inactive
a=ptime:20
b=AS:75
a=inactive
a=rtpmap:34 H263/90000
a=cif:1
UE-1 acknowledges the 183 (Session Progress) response from UE-2 with a PRACK request. Resource reservation at
UE-1 is assumed to have completed at some point prior to sending the PRACK request. The local resource status is
included in the SDP offer.
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7
8
9
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CSeq: 2 PRACK
Max-Forwards: 70
RAck: 1000 1 INVITE
Content-Length: xxx
v=0
o=- 2987935614 2987935615 IN IP4 100.200.1.1
s=-
c=IN IP4 100.200.1.1
t=0 0
b=AS:25.4
a=sendrecv
a=ptime:20
b=AS:75
a=sendrecv
a=rtpmap:34 H263/90000
a=cif:1
16-20. 200 OK (P-CSCF1 to UE-1)
The 200 OK response is generated by UE-2 to acknowledge the reception of the PRACK request. UE-2 has
completed resource reservation for both video and audio streams. The response will include an SDP answer to
reflect successful resource reservation.
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SIP/2.0 200 OK
CSeq: 2 PRACK
Content-Length: xxx
v=0
o=- 35270718123 35270718124 IN IP4 100.300.1.2
s=-
c=IN IP4 100.300.1.2
t=0 0
b=AS:25.4
a=sendrecv
a=ptime:20
b=AS:75
a=sendrecv
a=rtpmap:34 H263/90000
a=cif:1
UE-2 generates a 180 (Ringing) response to UE-1 and sends it reliably.
Table 6.2.4.2-5 - 180 Ringing (P-CSCF1 to UE-1)
SIP/2.0 180 Ringing
CSeq: 1 INVITE
Privacy: none
Require: 100rel
RSeq: 1001
Content-Length: 0
UE-1 acknowledges the 180 (Ringing) response from UE-2 with a PRACK request.
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6
7
8
9
10
11
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CSeq: 3 PRACK
Max-Forwards: 70
RAck: 1001 1 INVITE
Content-Length: 0
31-35. 200 OK (P-CSCF1 to UE-1)
SIP/2.0 200 OK
CSeq: 3 PRACK
Content-Length: 0
36-40. 200 OK (P-CSCF1 to UE-1)
request.
SIP/2.0 200 OK
CSeq: 1 INVITE
Content-Length: 0
41-45. ACK (UE-1 to P-CSCF1)
UE-1 sends an ACK request in response to the 200 OK response from UE-2.
Table 6.2.4.2-9 - ACK (UE-1 to P-CSCF1)
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27
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
1
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2

Appendix A (Informative): VoIP example with QoS
reservation, activation, and updating at RAN
In this section, VoIP application is used as an example to show how the QoS reservation, activation, and updating
process at RAN and IP-CAN works together with the call flows described in this document. The call flow described
in Scenario 1 is used. The following call flows are examples where an HRPD Rev-A (or later) access network is
used.
A.1 QoS Configuration for VoIP in advance
In order to reduce the call setup delay, VoIP application on the AT sets up two independent QoS flows in advance
(e.g. right after the AT powers up):
1) SIP flow for SIP signaling messages;
2) RTP flow for VoIP RTP traffic.
The steps for configuration of the QoS are the same for these two flows, except for the actual activation part of the
SIP flow. The SIP flow reservation always remains in the “on” state while the VoIP RTP flow needs to be explicitly
turned on when actual VoIP call is being set up.
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Figure 5 QoS configuration for VoIP
Step 1: The UE powers up for the first time.
Step 2: UE performs session negotiation. The AN provides the list of QoS User Profiles that the AN can handle as
part of the parameter negotiation for MPA/EMPA. This step also includes the terminal authentication. UE gets the
RANSupportedQoSProfiles as part of the session negotiation if AN plans to send one to the UE.
Step 3: VoIP Application is started.
Step 4: The PPP/PFO negotiation with the PDSN is initiated. The user is authenticated with the AAA and the user’s
allowed QoS User Profiles is updated in the AN and PDSN.
Step 5: The UE starts SIP flow setup.
Step 6: The UE negotiates with the AN to configure the radio protocols requires to carry the SIP flow and once the
PDSN is also successfully configured requests the AN to turn the reservation on.
Step 7: AN performs A11 signaling to establish the required A10 connections.
Step 8: The TFT filters and the channel treatment information are configured at the PDSN using RSVP messaging.
Step 9: The SIP flow is activated.
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Step 10: AN informs the PDSN that the QoS flow is activated.
Step 11: The UE starts RTP flow setup. This may happen in parallel to step 5~10.
Step 12: The UE negotiates with the AN to configure the radio protocols required to carry the RTP flow.
Step 13: AN performs A11 signaling to establish the required A10 connections.
Step 14: The TFT filters and the channel treatment information are configured at the PDSN. The RTP flow is now
successfully configured and reservation is left in the off state.
Step 15: Once step 10 and step 14 are completed, The UE performs SIP registration.
A.2 Resource activation on originating side
The call flow in this section show how the pre-reserved resources are turned on before the UE-1 makes a VoIP call.
Figure 6 QoS Activation on Originating UE
Step 1. After the mobile user initiates the VoIP call, UE-1 performs connection setup for both SIP and RTP flows as
described in Section A.1 if the flows are not set up in advance.
Step 2. UE-1 sends ReservationOnRequest to turn on the QoS for the RTP flow.
Step 3. AN-1 sends ReservationAccept to UE-1 indicating that the QoS for the RTP flow is changed to “on” state.
Step 4~5. AN-1 informs the PDSN that the QoS flow for RTP is now activated.
The rest of the call flow follows the procedures described in Scenario 1 where the resources at UE-1 are ready.
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A.3 Resource activation on terminating side
This section shows how the terminating side, UE-2, quickly turns on its pre-configured QoS flow for RTP after it
receives the INVITE and before it sends 180 (Ringing) to indicate resource ready.
Figure 7 QoS Activation on Terminating UE
Step 1. UE-1 activates its QoS upon VoIP call initiation.
Step 2. UE-1 sends INVITE to UE-2 and the INVITE request is eventually forwarded to AN-2.
Step 3. AN-2 optionally performs connection setup (e.g., traffic channel setup etc) with UE-2.
Step 4. AN-2 sends the INVITE to UE-2.
Step 5. UE-2 sends ReservationOnRequest to turn on the QoS for the RTP flow.
Step 6. AN-2 sends ReservationAccept to UE-2 indicating that the QoS for the RTP flow is changed to “on” state.
Step 7~8. AN-2 informs the PDSN that the QoS flow for RTP is now activated.
The rest of the call flow follows the procedures described in Scenario 1 where the resources at UE-2 are ready
before the first provisional response is sent.
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A.4 Updating of resource reservation
In some cases, the UE may need to adjust the pre-configured QoS based on the actual codec negotiated. In this
section, VoIP application with multiple codecs in the initial offer is used as an example to show how QoS resources
can be updated without affecting the call setup delay.
For example, in the initial INVITE sent from UE-1 to UE-2, two different codecs may be offered, one is supported
by RS1 with FlowProfileID 256 (e.g. EVRC), while the other may be supported by RS2 with FlowProfileID 257
(e.g. AMR). Typically, the VoIP calls will be RS1 with RS2 calls occurring occasionally. In order to expedite the call
setup delay using the approach described in Sections A.1 through A.3, FlowProfileID 256 is always granted initially
to the UE for the pre-configured RTP QoS flow. The reservation is turned to the open state with FlowProfileID 256
when the UE initiates VoIP calls. If both ends actually agrees on RS2 Vocoder, the QoS flow may be switched to
FlowProfileID=257.
A.4.1 Resource updating on originating side
Call flow in this section shows how UE-1 can update its QoS to support RS2 even though QoS flow for RS1 is pre-
configured.
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Figure 8 Resource Updating on Originating UE
Step 1. UE-1 performs connection setup if no flows have been pre-configured.
Step 2~4. UE-1 sends INVITE to UE-2 and also turns on the QoS flow for RTP stream with FlowProfileID 256.
Step 5~6. UE-2 completes resource reservation and returns 180 (Ringing) with the answer included.
Step 7. UE-1 learns that RS2 codec is chosen for the session instead of the RS1 codec. It changes reservation label if
different labels are used for the two reservations.
Step 8~9. UE-1 requests the AN-1 to update the QoS flow profile to 257.
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Step 10~11. AN-1 indicates to UE-1 that the flow profile has been updated to 257.
Step 12~13. AN-1 informs the PDSN that the updated flow has been activated.
Step 14. UE-1 and AN-1 updates RLP/RTCMAC parameters to match the updated flow if necessary.
The rest of the SIP messages are exchanged following the same procedure described in Scenario 1.
A.4.1 Resource updating on terminating side
Call flow in this section shows how UE-2 can update its QoS to support RS2 even though QoS flow for RS1 is pre-
configured.
Figure 9 Resource Updating on Terminating UE
Step 1~2. UE-1 completes resource reservation and sends INVITE to UE-2 with QoS ready indication in the offer.
The INVITE arrives at AN-2 first.
Step 3. AN-2 performs connection setup if no flows have been pre-configured.
Step 4. AN-2 forwards the INVITE to UE-2.
Step 5~6. UE-2 determines that RS2 codec will be used instead of the RS1 codec. It requests AN-2 to update the
QoS flow for RTP to profile 257 and at the same time requests AN-2 to turn on the reservation.
Step 7~8. AN-2 indicates to UE-2 that the flow profile has been updated to 257 and the flow is turned on.
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Step 9. UE-2 and AN-2 updates RLP/RTCMAC parameters to match the updated flow if necessary.
Step 10~11. AN-2 informs the PDSN that the updated flow has been activated.
The rest of the SIP messages are exchanged following the same procedure described in Scenario 1.
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Ims call flow

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