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3GPP IMS
Cellular VoIP with SIP
Overview
• The cellular industry sees the benefits of
VoIP
• Standards have been locked down and
vendors are preparing products
• Within a few years we will see large
deployments of cell phones using IP and
SIP for voice and multimedia services
Why is this important?
• Current GSM / UMTS system has 70% market share
with 475 Million subscribers
• Scope of the change is tremendous. Consider the
impact of changing the entire PSTN system to VoIP in a
few years. IMS similar in scale.
• Enables seamless integration with wired VoIP and all
other Web / IP based services and Cell Phones.
• Enables (and drives) many new services such as video
phone, streaming media and presence based services
• May drive widespread adoption of IPv6
Why is this interesting?
• IMS represents the largest planed deployment of
VoIP / SIP (that I am aware of)
• IMS addresses the details. The standards define
all aspects of a large, real system.
• IMS integrates with VoIP and other services on
the wired internet. Anything you do with SIP in
the future will likely involve some interaction with
terminals in the IMS networks.
• IMS enables 3rd
party solutions.
What will I discuss?
• What is IMS?
• IMS Implementation / Components
• Key Use Cases – What happens
– Data Link Establishment
– Registration
– Call Establishment
IMS – Internet Multimedia Core
Network Subsystem
• IMS is a set of standards defined by 3gpp to
replace GSM signaling with VoIP / SIP.
• More information on 3gpp, and all of the IMS
standards can be found at www.3gpp.org.
22.228, 23.228 and 24.228 (and others)
describe IMS.
• IMS standards are very comprehensive and
designed to address all aspects of the system to
promote worldwide interoperability.
What Is IMS?
IMS Network Components
Use Cases to be Considered
• Data Link Establishment
• Registration
• Calling
Call Session Control Function
IMS Components
• Central part of network implementation is
set of “Call Session Control Function”
Components
• Proxy (P-CSCF)
• Serving (S-CSCF)
• Interrogating (I-CSCF)
Other IMS Components
• Media Gateway (MGW)
• Media Gateway Control Function (MGCF)
• Application Server (AS)
• Home Subscriber Server (HSS)
• Others Not Mentioned
Proxy Call Session Control
Function (P-CSCF)
• First Point of contact between Mobile and the IMS in home
network
• P-CSCF is a proxy; Accepts requests then services them
internally or forwards request on
• P-CSCF IP address is discovered by UEs as part of or
immediately after data link establishment
• REGISTER Message from UE forwarded to I-CSCF Other SIP
messages from UE forwarded to S-CSCF
• Generation of CDRs (Billing Records)
• Maintain a Security Association between itself and each UE
• Should perform SIP message compression/decompression.
• Authorization of bearer resources and QoS management.
Serving Call Session Control
Function (S-CSCF)
• S-CSCF is a proxy; Accepts requests then services them
internally or forwards request on (possibly after translation)
• Routes requests to other proxies for servicing.
– Local P-CSCF for call to mobile in network
– Remote P-CSCF (I-CSCF) for mobile in remote network
– MGCF for access to PSTN
– MRFC for special function such as multi-party calls
• E.164 address translation to SIP routable URL using ENUM DNS
translation
• Registration - accepts registration requests and makes its
information available through the location server (eg. HSS).
• Prevent access by unauthorized users
• Billing records
• Interacts with HSS to store / retrieve user information
• Interacts with AS (Application Server)
Interrogating Call Session Control
Function (I-CSCF)
• Optional
• Hides details of network operator’s network
• Used by visiting UE
• Used to interface out of home area UE to
home network
• Route a SIP message received from another
network towards the S-CSCF
• Address of S-CSCF obtained from HSS by I-
CSCF
Use Cases
• Data Link Establishment
• Registration
• Calling
Data Link Establishment
• Mobiles must establish a data link for the exchange of IP packets
(IMS exists above the IP layer)
• Existing GSM support for high speed data links will remain in IMS
• GSM Packet Switched Data Links
– Are secure
– Can be established when roaming
– Can provide the address of the P-CSCF as part of activate confirm
– Allow mobile to specify and control the QoS
– Should assign the mobile a globally routable IP address (IPv6) for VoIP
media streaming
• Other data links types may also be supported
• WLAN
– Could be activated when in coverage as cheaper bearer
– Likely requires a VPN to the IMS network for security??
Registration Use Cases
• Registering New User
– In Network / Roaming
– Authentication
• Re-Registering
• Deregistering
– Network Initiated
• HSS
• Serving S-CSCF
– Client Initiated
UE RAN GPRS/DHCP P-CSCF
(pcscf1)
DNS I-CSCF
(icscf1_1)
S-CSCF
(scscf1)
HSS
Visited Network (visited1.net) Home Network (home1.net)
1. GPRS Attach procedure
PDP Context Establishment
and
P-CSCF Discovery
2. REGISTER
3. DNS: DNS-Q
4. REGISTER
6. REGISTER
19. Cx: S-CSCF
registration
notification
20. 200 OK
21. 200 OK
22.200 OK
5. Cx: User registration status query
7. Cx:
Authentication
9. 401Unauthorized
10. 401 Unauthorized
11. 401 Unauthorized
13. REGISTER
14. DNS: DNS-Q
15. REGISTER
17 REGISTER
16. Cx: User registration status query
8. Autentication
Vector Selection
12. Generation
of Response and
session keys
18.
Authentication
New UE Registration
• Figure 6.2-1 from
24.228
• UE Roaming
• Shows Authentication
of the private user
identity.
• No I-CSCF
(Configuration Hiding)
Node Before Registration During Registration After Registration
UE
- in local network
Credentials
Home Domain
Proxy Name/Address
Same Same
Proxy-CSCF
- in Home or Visited
network
Routing Function Initial Network Entry
point
UE Address
Public and Private User
IDs
Final Network Entry point
UE Address
Public and Private User
IDs
Interrogating-CSCF
- in Home network
HSS or SLF Address Serving-CSCF
address/name
P-CSCF Network ID
Home Network contact
Information
No State Information
HSS User Service Profile P-CSCF Network ID Serving-CSCF
address/name
Serving-CSCF
(Home)
No state information HSS Address/name
User profile (limited – as
per network
scenario)
Proxy address/name
P-CSCF Network ID
Public/Private User ID
UE IP Address
May have session state
Information
Same as during
registration
Registration – Information Storage
• Table 5.1 from
23.228
• Information
Storage before,
during and after
the registration
process
Call Use Cases
• Mobile Originated Call (Placed)
• Mobile Terminated Call (Received)
• Mobile in Home Network
• Mobile out of Home Network (Roaming)
• Mobile to Mobile Call
• Mobile to PSTN Call
• Mobile to Legacy GSM Call?
• Mobile to Wired VoIP Terminal
Mobile Originated Call Use Case
• Figure 7.2.3.1-1: 24.228
• Mobile is in network
• Shows interaction of UE,
Proxy CSCF and Serving
CSCF
• Peer could be PSTN,
another mobile (In
network or roaming) a
multi-conferencing server
or a wired VoIP client.
Mobile
Originated Call
Use Case
(1) INVITE (UE to P-CSCF)
• UE#1 determines the complete set of codecs that it is capable of supporting for this session.
• It builds a SDP containing bandwidth requirements and characteristics of each, 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.
• UE sends the INVITE request, containing an initial SDP, to the P-CSCF determined via the CSCF
discovery mechanism.
• Headers
– Request-URI: Contains the international E.164 number from the user as <tel:E.164_number
– Via: contains the IP address or FQDN of the originating UE.
– Route: contains the P-CSCF address learnt during P-CSCF discovery, including the port number
negotiated during the security agreement, plus the elements from the Service-Route header from
registration.
– P-Preferred-Identity: The user provides a hint about the identity to be used for this session.
– P-Access-Network-Info: the UE provides the access-type and access-info, related to the serving
access network.
– Cseq: A random starting number.
– Contact: is a SIP URI that contains the IP address or FQDN of the originating UE. It also contains the
port number where the UE wants to receive protected messages.
– Security-Verify: Contains the security agreement as represented by the received Security-
Server header.
– SDP The SDP contains a set of codecs supported by UE#1 and desired by the user at UE#1 for
this session.
Table 7.2.3.1-1 INVITE Message
Table 7.2.3.1-1: INVITE (UE to P-CSCF)
1. INVITE tel:+1-212-555-2222 SIP/2.0
2. Via: SIP/2.0/UDP
[5555::aaa:bbb:ccc:ddd]:1357;comp=sigcomp;branch=z9hG4b
Knashds7
3. Max-Forwards: 70
4. Route: <sip:pcscf1.home1.net:7531;lr;comp=sigcomp>,
<sip:scscf1.home1.net;lr>
5. P-Preferred-Identity: "John Doe" <tel:+1-212-555-1111>
6. P-Access-Network-Info: 3GPP-UTRAN-TDD; utran-cell-id-
3gpp=234151D0FCE11
7. Privacy: none
8. From: <sip:user1_public1@home1.net>;tag=171828
9. To: <tel:+1-212-555-2222>
10. Call-ID: cb03a0s09a2sdfglkj490333
11. Cseq: 127 INVITE
12. Require: precondition, sec-agree
13. Proxy-Require: sec-agree
14. Supported: 100rel
15. Contact: <sip:[5555::aaa:bbb:ccc:ddd]:1357;comp=sigcomp>
16. Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE,
REFER, MESSAGE
17. Security-Verify: ipsec-3gpp; q=0.1; alg=hmac-sha-1-96; spi-
c=98765432; spi-s=87654321; port-c=8642; port-s=7531
SDP
1. Content-Type: application/sdp
2. Content-Length: (…)
3. v=0
4. o=- 2987933615 2987933615 IN IP6 5555::aaa:bbb:ccc:ddd
5. s=-
6. c=IN IP6 5555::aaa:bbb:ccc:ddd
7. t=0 0
8. m=video 3400 RTP/AVP 98 99
9. b=AS:75
10. a=curr:qos local none
11. a=curr:qos remote none
12. a=des:qos mandatory local sendrecv
13. a=des:qos none remote sendrecv
14. a=rtpmap:98 H263
15. a=rtpmap:99 MP4V-ES
16. a=fmtp:98 profile-level-id=0
17. m=audio 3456 RTP/AVP 97 96
18. b=AS:25.4
19. a=curr:qos local none
20. a=curr:qos remote none
21. a=des:qos mandatory local sendrecv
22. a=des:qos none remote sendrecv
23. a=rtpmap:97 AMR
24. a=fmtp:97 mode-set=0,2,5,7; maxframes=2
1. a=rtpmap:96 telephone-event
(3) INVITE (P-CSCF to S-CSCF)
• The P-CSCF adds itself to the Record-Route
header and Via header.
• Headers
– P-Asserted-Identity: P-CSCF inserts the TEL URI in
the P-Asserted-Identity header field and it also
removes P-Preferred-Identity header field.
– P-Access-Network-Info: This header contains
information from the UE.
– P-Charging-Vector: The P-CSCF inserts this header
and populates the icid parameters with a unique
globally value
(5) Evaluation of initial filter
criterias
• S-CSCF validates the service profile of
this subscriber, and evaluates the initial
filter criterias.
(6) INVITE (MO#2 to S-S)
• Headers
– Request-URI: ENUM translated to a
globally routable SIP-URL before applying it
as Request-URI of the outgoing INVITE
request.
– P-Charging-Vector: The S-CSCF adds the
identifier of its own network to the originating
Inter Operator Identifier (IOI) parameter of this
header.
(8) 183 Session Progress
(S-S to MO#2)
• The media stream capabilities of the
destination are returned along the
signaling path, in a 183
183 Session Progress
Table 7.2.3.1-8:
The media stream capabilities of the destination are returned
• SIP/2.0 183 Session Progress
• Via: 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=z9hG4b
Knashds7
• Record-Route: <sip:pcscf2.home2.net;lr>,
<sip:scscf2.home2.net;lr>, <sip:scscf1.home1.net;lr>,
<sip:pcscf1.home1.net;lr>
• P-Asserted-Identity: "John Smith" <tel:+1-212-555-2222>
• P-Charging-Vector: icid-
value="AyretyU0dm+6O2IrT5tAFrbHLso=023551024"; orig-
ioi=home1.net; term-ioi=home2.net
• Privacy: none
• From:
• To: <tel:+1-212-555-2222>;tag=314159
• Call-ID:
• CSeq:
• Require: 100rel
• Contact: <sip:[5555::eee:fff:aaa:bbb]:8805;comp=sigcomp>
• Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE,
REFER, MESSAGE
• RSeq: 9021
• Content-Type: application/sdp
• Content-Length: (…)
• v=0
• o=- 2987933623 2987933623 IN IP6 5555::eee:fff:aaa:bbb
• s=-
• c=IN IP6 5555::eee:fff:aaa:bbb
• t=0 0
• m=video 10001 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 mandatory remote sendrecv
• a=conf:qos remote sendrecv
• a=rtpmap:98 H263
• a=rtpmap:99 MP4V-ES
• a=fmtp:98 profile-level-id=0
• m=audio 6544 RTP/AVP 97 96
• b=AS:25.4
• a=curr:qos local none
• 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; maxframes=2
• a=rtpmap:96 telephone-event
(11) 183 Session Progress
(P-CSCF to UE)
• Headers
– P-Media-Authorization: a P-CSCF
generated authorization token
– Record-Route: The P-CSCF rewrites the
Record-Route header to add the port number
negotiated during the security agreement and
the comp=sigcomp parameter to its own SIP
URI.
(12) PRACK (UE to P-CSCF)
• UE#1 determines which media flows
should be used for this session, and which
codecs should be used for each of those
media flows. If there was any change in
media flows, or if there was more than one
choice of codec for a media flow, then
UE#1 include a new SDP offer in the
PRACK request sent to UE#2).
• PRACK sip:[5555::eee:fff:aaa:bbb]:8805;comp=sigcomp SIP/2.0
• Via: SIP/2.0/UDP
[5555::aaa:bbb:ccc:ddd]:1357;comp=sigcomp;branch=z9hG4bK
nashds7
• Max-Forwards: 70
• P-Access-Network-Info: 3GPP-UTRAN-TDD; utran-cell-id-
3gpp=234151D0FCE11
• Route: <sip:pcscf1.home1.net:7531;lr;comp=sigcomp>,
<sip:scscf1.home1.net;lr>, <sip:scscf2.home2.net;lr>,
<sip:pcscf2.home2.net;lr>
• From: <sip:user1_public1@home1.net>;tag=171828
• To: <tel:+1-212-555-2222>;tag=314159
• Call-ID: cb03a0s09a2sdfglkj490333
• Cseq: 128 PRACK
• Require: precondition, sec-agree
• 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
• RAck: 9021 127 INVITE
• Content-Type: application/sdp
• Content-Length: (…)
• v=0
• o=- 2987933615 2987933616 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
• b=AS:75
• a=curr:qos local none
• a=curr:qos remote none
• a=des:qos mandatory local sendrecv
• a=des:qos mandatory remote sendrecv
• a=rtpmap:98 H263
• a=fmtp:98 profile-level-id=0
• m=audio 3456 RTP/AVP 97 96
• b=AS:25.4
• a=curr:qos local none
• a=curr:qos remote none
• 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
(13) Resource Reservation
• After determining the final media streams
in step #11, UE initiates the reservation
procedures for the resources needed for
this session.
(19) UPDATE (UE to P-CSCF)
• When the resource reservation is
completed, UE sends the UPDATE
request to the terminating endpoint, via
the signalling path established by the
INVITE request. The request is sent first to
P-CSCF.
(38) ACK (UE to P-CSCF)
• UE starts the media flow for this session,
and responds to the 200 OK (39) with an
ACK request sent to P-CSCF.
• PSTN Call
Origination
• Figure 7.2.4.1-
1, 24.228
•
MGW MGCF
Home Network
2. H.248 interaction
to create connection
4. 100 Trying
5. 183 Session Progress
8. 200 OK
9. H.248 interaction
to modify connection
to reserve resources
10. Resource
Reservation
12. UPDATE
14. 180 Ringing
15. PRACK
17.
ACM
1. IAM
3. INVITE
7. PRACK
11.
COT
19.
ANM
20. H.248 interaction to
modify connection to
start media flow
21. ACK
CS Networks
6. Bearer related negotiation(if any)
13. 200 OK
16. 200 OK
18. 200 OK
IMS Standards Define
• Phone Operation
• Network Operation
– Billing
– PSTN Interface
• Communication between Phone and
Network
• Communication between Networks
IMS Services
• What existing services will be offered?
• How will the existing services be better?
• What new services will be offered?
• Can the old voice services still be offered?
IMS vs Wired VoIP
• How is this the same as wired VoIP with
SIP?
• How is this different from Wired VoIP with
SIP?
References
• [5] 3GPP TS 24.228 V5.4.0 (2003-03),
“Signaling Flows for the IP Multimedia Call
Control Based on SIP and SDP;Stage 3
(Release 5).”

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3GPP IMS

  • 2. Overview • The cellular industry sees the benefits of VoIP • Standards have been locked down and vendors are preparing products • Within a few years we will see large deployments of cell phones using IP and SIP for voice and multimedia services
  • 3. Why is this important? • Current GSM / UMTS system has 70% market share with 475 Million subscribers • Scope of the change is tremendous. Consider the impact of changing the entire PSTN system to VoIP in a few years. IMS similar in scale. • Enables seamless integration with wired VoIP and all other Web / IP based services and Cell Phones. • Enables (and drives) many new services such as video phone, streaming media and presence based services • May drive widespread adoption of IPv6
  • 4. Why is this interesting? • IMS represents the largest planed deployment of VoIP / SIP (that I am aware of) • IMS addresses the details. The standards define all aspects of a large, real system. • IMS integrates with VoIP and other services on the wired internet. Anything you do with SIP in the future will likely involve some interaction with terminals in the IMS networks. • IMS enables 3rd party solutions.
  • 5. What will I discuss? • What is IMS? • IMS Implementation / Components • Key Use Cases – What happens – Data Link Establishment – Registration – Call Establishment
  • 6. IMS – Internet Multimedia Core Network Subsystem • IMS is a set of standards defined by 3gpp to replace GSM signaling with VoIP / SIP. • More information on 3gpp, and all of the IMS standards can be found at www.3gpp.org. 22.228, 23.228 and 24.228 (and others) describe IMS. • IMS standards are very comprehensive and designed to address all aspects of the system to promote worldwide interoperability.
  • 9. Use Cases to be Considered • Data Link Establishment • Registration • Calling
  • 10. Call Session Control Function IMS Components • Central part of network implementation is set of “Call Session Control Function” Components • Proxy (P-CSCF) • Serving (S-CSCF) • Interrogating (I-CSCF)
  • 11. Other IMS Components • Media Gateway (MGW) • Media Gateway Control Function (MGCF) • Application Server (AS) • Home Subscriber Server (HSS) • Others Not Mentioned
  • 12. Proxy Call Session Control Function (P-CSCF) • First Point of contact between Mobile and the IMS in home network • P-CSCF is a proxy; Accepts requests then services them internally or forwards request on • P-CSCF IP address is discovered by UEs as part of or immediately after data link establishment • REGISTER Message from UE forwarded to I-CSCF Other SIP messages from UE forwarded to S-CSCF • Generation of CDRs (Billing Records) • Maintain a Security Association between itself and each UE • Should perform SIP message compression/decompression. • Authorization of bearer resources and QoS management.
  • 13. Serving Call Session Control Function (S-CSCF) • S-CSCF is a proxy; Accepts requests then services them internally or forwards request on (possibly after translation) • Routes requests to other proxies for servicing. – Local P-CSCF for call to mobile in network – Remote P-CSCF (I-CSCF) for mobile in remote network – MGCF for access to PSTN – MRFC for special function such as multi-party calls • E.164 address translation to SIP routable URL using ENUM DNS translation • Registration - accepts registration requests and makes its information available through the location server (eg. HSS). • Prevent access by unauthorized users • Billing records • Interacts with HSS to store / retrieve user information • Interacts with AS (Application Server)
  • 14. Interrogating Call Session Control Function (I-CSCF) • Optional • Hides details of network operator’s network • Used by visiting UE • Used to interface out of home area UE to home network • Route a SIP message received from another network towards the S-CSCF • Address of S-CSCF obtained from HSS by I- CSCF
  • 15. Use Cases • Data Link Establishment • Registration • Calling
  • 16. Data Link Establishment • Mobiles must establish a data link for the exchange of IP packets (IMS exists above the IP layer) • Existing GSM support for high speed data links will remain in IMS • GSM Packet Switched Data Links – Are secure – Can be established when roaming – Can provide the address of the P-CSCF as part of activate confirm – Allow mobile to specify and control the QoS – Should assign the mobile a globally routable IP address (IPv6) for VoIP media streaming • Other data links types may also be supported • WLAN – Could be activated when in coverage as cheaper bearer – Likely requires a VPN to the IMS network for security??
  • 17. Registration Use Cases • Registering New User – In Network / Roaming – Authentication • Re-Registering • Deregistering – Network Initiated • HSS • Serving S-CSCF – Client Initiated
  • 18. UE RAN GPRS/DHCP P-CSCF (pcscf1) DNS I-CSCF (icscf1_1) S-CSCF (scscf1) HSS Visited Network (visited1.net) Home Network (home1.net) 1. GPRS Attach procedure PDP Context Establishment and P-CSCF Discovery 2. REGISTER 3. DNS: DNS-Q 4. REGISTER 6. REGISTER 19. Cx: S-CSCF registration notification 20. 200 OK 21. 200 OK 22.200 OK 5. Cx: User registration status query 7. Cx: Authentication 9. 401Unauthorized 10. 401 Unauthorized 11. 401 Unauthorized 13. REGISTER 14. DNS: DNS-Q 15. REGISTER 17 REGISTER 16. Cx: User registration status query 8. Autentication Vector Selection 12. Generation of Response and session keys 18. Authentication New UE Registration • Figure 6.2-1 from 24.228 • UE Roaming • Shows Authentication of the private user identity. • No I-CSCF (Configuration Hiding)
  • 19. Node Before Registration During Registration After Registration UE - in local network Credentials Home Domain Proxy Name/Address Same Same Proxy-CSCF - in Home or Visited network Routing Function Initial Network Entry point UE Address Public and Private User IDs Final Network Entry point UE Address Public and Private User IDs Interrogating-CSCF - in Home network HSS or SLF Address Serving-CSCF address/name P-CSCF Network ID Home Network contact Information No State Information HSS User Service Profile P-CSCF Network ID Serving-CSCF address/name Serving-CSCF (Home) No state information HSS Address/name User profile (limited – as per network scenario) Proxy address/name P-CSCF Network ID Public/Private User ID UE IP Address May have session state Information Same as during registration Registration – Information Storage • Table 5.1 from 23.228 • Information Storage before, during and after the registration process
  • 20. Call Use Cases • Mobile Originated Call (Placed) • Mobile Terminated Call (Received) • Mobile in Home Network • Mobile out of Home Network (Roaming) • Mobile to Mobile Call • Mobile to PSTN Call • Mobile to Legacy GSM Call? • Mobile to Wired VoIP Terminal
  • 22. • Figure 7.2.3.1-1: 24.228 • Mobile is in network • Shows interaction of UE, Proxy CSCF and Serving CSCF • Peer could be PSTN, another mobile (In network or roaming) a multi-conferencing server or a wired VoIP client. Mobile Originated Call Use Case
  • 23. (1) INVITE (UE to P-CSCF) • UE#1 determines the complete set of codecs that it is capable of supporting for this session. • It builds a SDP containing bandwidth requirements and characteristics of each, 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. • UE sends the INVITE request, containing an initial SDP, to the P-CSCF determined via the CSCF discovery mechanism. • Headers – Request-URI: Contains the international E.164 number from the user as <tel:E.164_number – Via: contains the IP address or FQDN of the originating UE. – Route: contains the P-CSCF address learnt during P-CSCF discovery, including the port number negotiated during the security agreement, plus the elements from the Service-Route header from registration. – P-Preferred-Identity: The user provides a hint about the identity to be used for this session. – P-Access-Network-Info: the UE provides the access-type and access-info, related to the serving access network. – Cseq: A random starting number. – Contact: is a SIP URI that contains the IP address or FQDN of the originating UE. It also contains the port number where the UE wants to receive protected messages. – Security-Verify: Contains the security agreement as represented by the received Security- Server header. – SDP The SDP contains a set of codecs supported by UE#1 and desired by the user at UE#1 for this session.
  • 24. Table 7.2.3.1-1 INVITE Message Table 7.2.3.1-1: INVITE (UE to P-CSCF) 1. INVITE tel:+1-212-555-2222 SIP/2.0 2. Via: SIP/2.0/UDP [5555::aaa:bbb:ccc:ddd]:1357;comp=sigcomp;branch=z9hG4b Knashds7 3. Max-Forwards: 70 4. Route: <sip:pcscf1.home1.net:7531;lr;comp=sigcomp>, <sip:scscf1.home1.net;lr> 5. P-Preferred-Identity: "John Doe" <tel:+1-212-555-1111> 6. P-Access-Network-Info: 3GPP-UTRAN-TDD; utran-cell-id- 3gpp=234151D0FCE11 7. Privacy: none 8. From: <sip:user1_public1@home1.net>;tag=171828 9. To: <tel:+1-212-555-2222> 10. Call-ID: cb03a0s09a2sdfglkj490333 11. Cseq: 127 INVITE 12. Require: precondition, sec-agree 13. Proxy-Require: sec-agree 14. Supported: 100rel 15. Contact: <sip:[5555::aaa:bbb:ccc:ddd]:1357;comp=sigcomp> 16. Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER, MESSAGE 17. Security-Verify: ipsec-3gpp; q=0.1; alg=hmac-sha-1-96; spi- c=98765432; spi-s=87654321; port-c=8642; port-s=7531 SDP 1. Content-Type: application/sdp 2. Content-Length: (…) 3. v=0 4. o=- 2987933615 2987933615 IN IP6 5555::aaa:bbb:ccc:ddd 5. s=- 6. c=IN IP6 5555::aaa:bbb:ccc:ddd 7. t=0 0 8. m=video 3400 RTP/AVP 98 99 9. b=AS:75 10. a=curr:qos local none 11. a=curr:qos remote none 12. a=des:qos mandatory local sendrecv 13. a=des:qos none remote sendrecv 14. a=rtpmap:98 H263 15. a=rtpmap:99 MP4V-ES 16. a=fmtp:98 profile-level-id=0 17. m=audio 3456 RTP/AVP 97 96 18. b=AS:25.4 19. a=curr:qos local none 20. a=curr:qos remote none 21. a=des:qos mandatory local sendrecv 22. a=des:qos none remote sendrecv 23. a=rtpmap:97 AMR 24. a=fmtp:97 mode-set=0,2,5,7; maxframes=2 1. a=rtpmap:96 telephone-event
  • 25. (3) INVITE (P-CSCF to S-CSCF) • The P-CSCF adds itself to the Record-Route header and Via header. • Headers – P-Asserted-Identity: P-CSCF inserts the TEL URI in the P-Asserted-Identity header field and it also removes P-Preferred-Identity header field. – P-Access-Network-Info: This header contains information from the UE. – P-Charging-Vector: The P-CSCF inserts this header and populates the icid parameters with a unique globally value
  • 26. (5) Evaluation of initial filter criterias • S-CSCF validates the service profile of this subscriber, and evaluates the initial filter criterias.
  • 27. (6) INVITE (MO#2 to S-S) • Headers – Request-URI: ENUM translated to a globally routable SIP-URL before applying it as Request-URI of the outgoing INVITE request. – P-Charging-Vector: The S-CSCF adds the identifier of its own network to the originating Inter Operator Identifier (IOI) parameter of this header.
  • 28. (8) 183 Session Progress (S-S to MO#2) • The media stream capabilities of the destination are returned along the signaling path, in a 183
  • 29. 183 Session Progress Table 7.2.3.1-8: The media stream capabilities of the destination are returned • SIP/2.0 183 Session Progress • Via: 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=z9hG4b Knashds7 • Record-Route: <sip:pcscf2.home2.net;lr>, <sip:scscf2.home2.net;lr>, <sip:scscf1.home1.net;lr>, <sip:pcscf1.home1.net;lr> • P-Asserted-Identity: "John Smith" <tel:+1-212-555-2222> • P-Charging-Vector: icid- value="AyretyU0dm+6O2IrT5tAFrbHLso=023551024"; orig- ioi=home1.net; term-ioi=home2.net • Privacy: none • From: • To: <tel:+1-212-555-2222>;tag=314159 • Call-ID: • CSeq: • Require: 100rel • Contact: <sip:[5555::eee:fff:aaa:bbb]:8805;comp=sigcomp> • Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER, MESSAGE • RSeq: 9021 • Content-Type: application/sdp • Content-Length: (…) • v=0 • o=- 2987933623 2987933623 IN IP6 5555::eee:fff:aaa:bbb • s=- • c=IN IP6 5555::eee:fff:aaa:bbb • t=0 0 • m=video 10001 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 mandatory remote sendrecv • a=conf:qos remote sendrecv • a=rtpmap:98 H263 • a=rtpmap:99 MP4V-ES • a=fmtp:98 profile-level-id=0 • m=audio 6544 RTP/AVP 97 96 • b=AS:25.4 • a=curr:qos local none • 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; maxframes=2 • a=rtpmap:96 telephone-event
  • 30. (11) 183 Session Progress (P-CSCF to UE) • Headers – P-Media-Authorization: a P-CSCF generated authorization token – Record-Route: The P-CSCF rewrites the Record-Route header to add the port number negotiated during the security agreement and the comp=sigcomp parameter to its own SIP URI.
  • 31. (12) PRACK (UE to P-CSCF) • UE#1 determines which media flows should be used for this session, and which codecs should be used for each of those media flows. If there was any change in media flows, or if there was more than one choice of codec for a media flow, then UE#1 include a new SDP offer in the PRACK request sent to UE#2).
  • 32. • PRACK sip:[5555::eee:fff:aaa:bbb]:8805;comp=sigcomp SIP/2.0 • Via: SIP/2.0/UDP [5555::aaa:bbb:ccc:ddd]:1357;comp=sigcomp;branch=z9hG4bK nashds7 • Max-Forwards: 70 • P-Access-Network-Info: 3GPP-UTRAN-TDD; utran-cell-id- 3gpp=234151D0FCE11 • Route: <sip:pcscf1.home1.net:7531;lr;comp=sigcomp>, <sip:scscf1.home1.net;lr>, <sip:scscf2.home2.net;lr>, <sip:pcscf2.home2.net;lr> • From: <sip:user1_public1@home1.net>;tag=171828 • To: <tel:+1-212-555-2222>;tag=314159 • Call-ID: cb03a0s09a2sdfglkj490333 • Cseq: 128 PRACK • Require: precondition, sec-agree • 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 • RAck: 9021 127 INVITE • Content-Type: application/sdp • Content-Length: (…) • v=0 • o=- 2987933615 2987933616 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 • b=AS:75 • a=curr:qos local none • a=curr:qos remote none • a=des:qos mandatory local sendrecv • a=des:qos mandatory remote sendrecv • a=rtpmap:98 H263 • a=fmtp:98 profile-level-id=0 • m=audio 3456 RTP/AVP 97 96 • b=AS:25.4 • a=curr:qos local none • a=curr:qos remote none • 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
  • 33. (13) Resource Reservation • After determining the final media streams in step #11, UE initiates the reservation procedures for the resources needed for this session.
  • 34. (19) UPDATE (UE to P-CSCF) • When the resource reservation is completed, UE sends the UPDATE request to the terminating endpoint, via the signalling path established by the INVITE request. The request is sent first to P-CSCF.
  • 35. (38) ACK (UE to P-CSCF) • UE starts the media flow for this session, and responds to the 200 OK (39) with an ACK request sent to P-CSCF.
  • 36.
  • 37. • PSTN Call Origination • Figure 7.2.4.1- 1, 24.228 • MGW MGCF Home Network 2. H.248 interaction to create connection 4. 100 Trying 5. 183 Session Progress 8. 200 OK 9. H.248 interaction to modify connection to reserve resources 10. Resource Reservation 12. UPDATE 14. 180 Ringing 15. PRACK 17. ACM 1. IAM 3. INVITE 7. PRACK 11. COT 19. ANM 20. H.248 interaction to modify connection to start media flow 21. ACK CS Networks 6. Bearer related negotiation(if any) 13. 200 OK 16. 200 OK 18. 200 OK
  • 38.
  • 39. IMS Standards Define • Phone Operation • Network Operation – Billing – PSTN Interface • Communication between Phone and Network • Communication between Networks
  • 40. IMS Services • What existing services will be offered? • How will the existing services be better? • What new services will be offered? • Can the old voice services still be offered?
  • 41.
  • 42. IMS vs Wired VoIP • How is this the same as wired VoIP with SIP? • How is this different from Wired VoIP with SIP?
  • 43. References • [5] 3GPP TS 24.228 V5.4.0 (2003-03), “Signaling Flows for the IP Multimedia Call Control Based on SIP and SDP;Stage 3 (Release 5).”