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IMS Originating to IMS Terminating Call (Caller and Called are IMS Subscribers)

Calling UE IMS Network Called UE


EventStudio System Designer 4.0
Caller User Visited IMS 1 Home IMS 1 Home IMS 2 Called User
Equipment Equipment
15-Dec-07 08:49 (Page 1)
Caller Orig P-CSCF Orig S-CSCF Term I-CSCF Term S-CSCF Term P-CSCF Called
This sequence diagram describes the call setup of a call from one IMS subscriber to another IMS subscriber. The calling subscriber is roaming in another IMS supporting network. The
called subscriber is in the home IMS network.

The call flow focuses on the IMS routing of SIP dialog. The major steps in the call flow are:

(1) IMS Routing of Initial SIP INVITE.

(2) IMS Routing of First Response to the SIP Invite.

(3) PDP Context Activation and Audio/Video Path Setup.


This sequence diagram was generated with EventStudio System Designer 4.0 (http://www.EventHelix.com/EventStudio). Copyright © 2007 EventHelix.com Inc. All Rights Reserved.
IMS Routing of Initial SIP INVITE
Initiate Call The user initiates a call to called@hims2.net.
called@hims2.net

Prepare a list of supported voice The calling includes all supported codecs. This
and video codecs information is included as the first SDP offer
in the initial invite.
INVITE The SIP phone sends the invite to
INVITE called@hims2.net SIP/2.0,
called@hims2.net. The message contains
P-Preferred-Identity: Route entries for the terminal and the S-CSCF
<caller@hims1.net>,
Via: <Calling UE IP> :Port,
address that was extracted from the
Route: <P-CSCF address>, Service-Route header in the registration "200
Route: <S-CSCF address>,
Contact: <Calling UE IP> :Port,
OK" message. Security ports setup for IPSec
SDP: <Caller Supported Codec List> SA establishment are used. "To" and "From"
headers are also included in the message.
These headers do not play a role in call
processing.
Make sure that INVITE was received over the The INVITE was sent using the registration
IPSec Security Association time SA so the P-CSCF accepts the request.
Make sure that the preferred public identity is P-CSCF verifies that the preferred public
currently registered identity specified in the INVITE is currently
registered.
The S-CSCF address for the user was obtained
at the time of registration (Service-Route
header in the "200 OK" response to the
REGISTER message.)
Use DNS to translate from scscf1.hims1.net to The originating P-CSCF queries the DNS to
'Orig S-CSCF' IP address obtain the IP address of the S-CSCF in the
called subscriber's home network. The
S-CSCF address for the user was obtained at
the time of registration (Service-Route header
in the "200 OK" response to the REGISTER
message).
IMS Originating to IMS Terminating Call (Caller and Called are IMS Subscribers)
Calling UE IMS Network Called UE
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Caller Orig P-CSCF Orig S-CSCF Term I-CSCF Term S-CSCF Term P-CSCF Called
INVITE The P-CSCF replaces the preferred identity
INVITE called@hims2.net SIP/2.0,
header with the asserted identity header and
P-Asserted-Identity: forwards the message to the S-CSCF in the
<caller@hims1.net>,
Via: <Orig P-CSCF> <Calling-UE>,
home network. It adds a Record-Route header
Record-Route: <Orig P-CSCF>, with its own address.
Route: <Orig S-CSCF>,
Contact: <Calling UE IP> :Port,
SDP: <Caller Supported Codec List>

100 Trying The P-CSCF just acknowledges the INVITE to


the UE. The "100 Trying" message indicates
that the call setup is in progress.
Use DNS to translate from hims2.net to 'Term The originating S-CSCF queries the DNS to
I-CSCF' IP address obtain the IP address of the I-CSCF in the
called subscriber's home network.
INVITE The S-CSCF removes the Route header and
INVITE called@hims2.net SIP/2.0,
routes the INVITE to the I-CSCF IP address
P-Asserted-Identity: obtained from the DNS query. Note that the
<caller@hims1.net>,
<tel:+13015556666>,
S-CSCF has added the telephone URL to the
Via: <Orig S-CSCF> <Orig P-CSCF> P-Asserted-Identity. The Via and
<Calling-UE>,
Record-Route: <Orig S-CSCF> <Orig
Record-Route headers are also updated with
P-CSCF>, self address.
Contact: <Calling UE IP> :Port,
SDP: <Caller Supported Codec List>

Query the SLF to locate HSS The I-CSCF queries the Subscription Location
Function (SLF) to identify the HSS that needs
to be queried.
100 Trying The S-CSCF acknowledges the INVITE that
was received from P-CSCF.
Query HSS to identify the S-CSCF for this SIP Query the HSS to obtain the S-CSCF for the
Dialog user.
INVITE As a part of the message processing, a route
INVITE called@hims2.net SIP/2.0,
entry is added for the Term S-CSCF. A new Via
P-Asserted-Identity: header is added to record that the message
<caller@hims1.net>,
<tel:+13015556666>,
traversed this I-CSCF. The message is
Via: <Term I-CSCF> <Orig S-CSCF> forwarded to the first route header (in this
<Orig P-CSCF> <Calling-UE>,
Route: <Term S-CSCF>,
case, the "Term S-CSCF").
Record-Route: <Orig S-CSCF> <Orig
P-CSCF>,
Contact: <Calling UE IP> :Port,
SDP: <Caller Supported Codec List>

Map from public request URI to the registered Map from the public URI to the called
address subscriber's registered IP address and port
number.
INVITE The public URI in the SIP INVITE is replaced
INVITE CALLED-IP SIP/2.0,
with the called subscriber's registered IP
P-Asserted-Identity: address and port number. The message is
<caller@hims1.net>,
<tel:+13015556666>,
routed to the P-CSCF IP address that was
Via: <Term S-CSCF> <Term I-CSCF> recorded at the time of registration. The Via
<Orig S-CSCF> <Orig P-CSCF>
<Calling-UE>,
and Record-Route headers are updated.
Route: <Term P-CSCF>,
Record-Route: <Term S-CSCF> <Orig
S-CSCF> <Orig P-CSCF>,
Contact: <Calling UE IP> :Port,
IMS Originating to IMS Terminating Call (Caller and Called are IMS Subscribers)
Calling UE IMS Network Called UE
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Caller Orig P-CSCF Orig S-CSCF Term I-CSCF Term S-CSCF Term P-CSCF Called
Obtain a media authorization token from the The terminating P-CSCF requests the Policy
PDF Decision Function (PDF) to generate a media
authorization token. The token will be included
in the INVITE sent to the terminating UE.
INVITE The P-CSCF updates the Via and
INVITE CALLED-IP SIP/2.0,
Route-Record headers and forwards the
P-Asserted-Identity: request to the Called UE. Note that the secure
<caller@hims1.net>,
<tel:+13015556666>,
port is included in the Via address
Via: <Term P-CSCF>;port <Term specification. The message also includes the
S-CSCF> <Term I-CSCF> <Orig
S-CSCF> <Orig P-CSCF>
media authorization token. This token will have
<Calling-UE>, to be passed to the GGSN in the PDP context
Route: <Term P-CSCF>;port,
Record-Route: <Term S-CSCF> <Orig
activation request.
S-CSCF> <Orig P-CSCF>,
Contact: <Called UE IP> :Port,
SDP: <Caller Supported Codec
List>,
P-Media-Authorization

100 Trying

100 Trying

100 Trying

Prepare a list of Codecs common The Caller examines the SDP list of available
between the Caller and the Called codec. It prunes the list by excluding codecs
subscriber that are not supported by the called
subscriber. This list will be included in the 183
message sent to the caller.
IMS Routing of First Response to the SIP Invite
183 Session Progress The UE replies indicating that the session is in
Via: <Term P-CSCF>;port <Term
progress. The contact address is set its own
S-CSCF> <Term I-CSCF> <Orig IP address. The Via and the Record-Route
S-CSCF> <Orig P-CSCF>
<Calling-UE>,
headers are copied from the received INVITE.
Record-Route: <Term S-CSCF>;port
<Orig S-CSCF> <Orig P-CSCF>,
Contact: <Calling UE IP> :Port,
SDP: <Codecs supported by Caller
and Called>

183 Session Progress The P-CSCF removes its own Via header entry
Via: <Term S-CSCF> <Term I-CSCF>
and addresses the message to the top via
<Orig S-CSCF> <Orig P-CSCF> header (Term S-CSCF in this case). The
<Calling-UE>,
Record-Route: <Term S-CSCF> <Orig
P-CSCF also removes the secure port from the
S-CSCF> <Orig P-CSCF>, Record-Route.
Contact: <Calling UE IP> :Port,
SDP: <Codecs supported by Caller
and Called>

183 Session Progress 183 message just retraces the path of the
Via: <Term I-CSCF> <Orig S-CSCF>
original INVITE. Each note removes its down
<Orig P-CSCF> <Calling-UE>, entry from the via header and forwards the
Record-Route: <Term S-CSCF> <Orig
S-CSCF> <Orig P-CSCF>,
message to the Via entry at the top. The
Contact: <Calling UE IP> :Port, Record-Route header is not touched.
SDP: <Codecs supported by Caller
and Called>
IMS Originating to IMS Terminating Call (Caller and Called are IMS Subscribers)
Calling UE IMS Network Called UE
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Caller Orig P-CSCF Orig S-CSCF Term I-CSCF Term S-CSCF Term P-CSCF Called
183 Session Progress
Via: <Orig S-CSCF> <Orig P-CSCF>
<Calling-UE>,
Record-Route: <Term S-CSCF> <Orig
S-CSCF> <Orig P-CSCF>,
SDP: <Codecs supported by Caller
and Called>

183 Session Progress


Via: <Orig P-CSCF> <Calling-UE>,
Record-Route: <Term S-CSCF> <Orig
S-CSCF> <Orig P-CSCF>,
SDP: <Codecs supported by Caller
and Called>

Obtain a media authorization token from the The originating P-CSCF requests the Policy
PDF Decision Function (PDF) to generate a media
authorization token. The token will be included
in the "183 Session Progress" sent to the
originating UE.
183 Session Progress Just like other nodes, the Orig P-CSCF
Via: <Calling-UE>,
removes its own entry from the Via header.
Record-Route: <Term S-CSCF>;port The P-CSCF also updates the Record-Route
<Orig S-CSCF> <Orig P-CSCF>,
SDP: <Codecs supported by Caller
header to include the protected port number in
and Called>, its entry. This forces the terminal to send all
P-Media-Authorization responses using the protected IPSec SA. The
message also includes the media authorization
token. This token will have to be passed to the
GGSN in the PDP context activation request.
PDP Context Activation and Audio/Video Path Setup
Select one Codec from the common The Caller examines the received common
codec list codec list and selects the codec to activate.
PRACK PRACK PRACK PRACK PRACK The Caller now sends a PRACK to inform the
SDP: <Selected Codec>, SDP: <Selected Codec>, SDP: <Selected Codec>, <Local-QOS: none> SDP: <Selected Codec>, SDP: <Selected Codec>,
called subscriber about the selected Codec.
<Local-QOS: none> <Local-QOS: none> <Local-QOS: none> <Local-QOS: none> The message also indicates that currently the
resources needed for meeting the quality of
service requiements of the session are not
available.
begin Now that the codec to be used has been
Caller PDP Context Activation selected, the PDP context activation is initiated
for allocating resources for meeting the
Quality of Service (QoS) requirements for the
codec.
200 OK 200 OK 200 OK 200 OK 200 OK The called subscriber acknowledges the
SDP: <Selected Codec>, SDP: <Selected Codec>, SDP: <Selected Codec>, <Local-QOS: none> SDP: <Selected Codec>, SDP: <Selected Codec>,
PRACK. The message also indicates that
<Local-QOS: none> <Local-QOS: none> <Local-QOS: none> <Local-QOS: none> quality of service for the session is not met for
the called subscriber.
begin The final codec at the called side is decided.
Called PDP Context Activation So initiate the PDP context activation to
allocate resources for meeting the QoS of the
terminating leg of the call.
IMS Originating to IMS Terminating Call (Caller and Called are IMS Subscribers)
Calling UE IMS Network Called UE
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Caller Orig P-CSCF Orig S-CSCF Term I-CSCF Term S-CSCF Term P-CSCF Called
end The caller PDP context activation has been
Caller PDP Context Activation completed.

UPDATE UPDATE UPDATE UPDATE UPDATE Since the caller PDP context has been
SDP: <Local-QOS: SDP: <Local-QOS: SDP: <Local-QOS: sendrecv> SDP: <Local-QOS: SDP: <Local-QOS:
activated, notify the called end that the caller
sendrecv> sendrecv> sendrecv> sendrecv> can now meet the quality of service in the
send and receive direction.
200 OK 200 OK 200 OK 200 OK 200 OK The caller replies back to the called user. Note
SDP: <Local-QOS: none> SDP: <Local-QOS: none> SDP: <Local-QOS: none> SDP: <Local-QOS: none> SDP: <Local-QOS: none>
that the Local QoS is still set to none as the
called PDP context activation has not been
completed.
end The called PDP context activation has been
Called PDP Context Activation completed. At this point, the caller and the
called PDP contexts are both active. The QoS
for the call can now be met.
Ringing Now all the resources for the call are in place.
Ring the called subscriber to notify the user
about the incoming call.
180 Ringing 180 Ringing 180 Ringing 180 Ringing 180 Ringing 180 Ringing Inform the caller that the called subscriber is
being rung. This serves as an implicit
indication to the caller that the QoS at the
called side has also been met.
PRACK PRACK PRACK PRACK PRACK The caller acknowledges the ringing message.

200 OK 200 OK 200 OK 200 OK 200 OK The called subscriber acknowledges the
PRACK.
Answer The called subscriber answers the call.

200 OK 200 OK 200 OK 200 OK 200 OK 200 OK Notify the caller that that the call has been
answered.
ACK ACK ACK ACK ACK The caller acknowledges the "200 OK"
message. The call is now ready to enter
conversation mode.
Conversation on a direct RTP/RTCP connection between the caller and called subscriber SIP phones.