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The concept of the IMS (IP Multimedia Subsystem) was originally driven by 3GPP to
provide an additional subsystem for GSM/UMTS networks that would enable the
delivery of IP-based multimedia services.
Since its conception within 3GPP it was also adopted by 3GPP2, who are responsible
for standards for mobile networks based on CDMA and IS-41 technology.
The IMS concept also closely parallels work being done by various bodies focused on
service delivery in fixed networks, such as the ITU (International Telecommunication
Union) and ETSI (European Telecommunications Standards Institute).
The IMS standards for fixed and mobile networks are now aligned, and beyond the
original intention of providing rich multimedia services, IMS is now seen as a
mechanism for supporting convergence, particularly FMC (Fixed-Mobile
Convergence).

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The reference architecture for the initial Release 5 IMS included all the functions
necessary to interface to the 3GPP PS and CS domains and to control services within
these domains, the IMS domain and the shared network elements such as the
Location Registers.
The IMS consists of:

CSCFs (Call Session Control Functions)

MRF (Multimedia Resource Function)

MRFC (Media Resource Function Controller)
MRFP (Media Resource Function Processor)

BGCF (Breakout Gateway Control Function)

SLF (Subscription Locator Function)

MGCF (Media Gateway Control Function)

IMS- MGW (IP Multimedia Subsystem Media Gateway Function)

ASs (Application Servers)

SGWs (Signalling Gateways)

In common with GSM and UMTS standardisation, the IMS is based on series of
defined interfaces or reference points between these functions.

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The IMS is made up from a number of component parts connected together using an
IP backbone. This will be a separate backbone to the one connecting the GGSNs and
SSGNs in the packet switching core of the GPRS network (Gn). Mobile users wishing
to connect to the IMS must first establish a GPRS connection, and a PDP context
must be activated. It is likely that the user will be authenticated during this process.
Data transfer between users of the IMS is arranged into sessions. The CSCF (Call
Session Control Function) is responsible for session control and is the control point for
the following functions:
User authentication
Call routing establishment
QoS over the IP network
Controlling the generation of CDRs (Call Detail Records) for accounting (billing)
All call/session control signalling in the IMS is performed using SIP.
Three types of CSCF are defined:
P-CSCF (Proxy Call Session Control Function)
S-CSCF (Serving Call Session Control Function)
I-CSCF (Interrogating Call Session Control Function)
There is likely to be more than one each of these functions in the networks for loadsharing purposes.

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Signalling from the user equipment will first arrive at the P-CSCF. The P-CSCF will
forward the call signalling to the S-CSCF, which is the home networks point of control
for the call.
The P-CSCF is located in the visited network for roaming subscribers.
CDRs will be generated for mobile originating calls at this point also.

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The S-CSCF handles the call/session control and accounting control for any activity
from the mobile subscriber. It is always located in the subscribers home network.
All signalling is routed via this device for billing purposes. For example, a mobile
subscriber from the UK roaming in America who calls a device in Australia will have
the call set-up signalling routed through the S-CSCF. However, the traffic will take a
different and perhaps a more optimal path between the networks.

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Any signalling arriving from outside the operators network will arrive at the I-CSCF.
This signalling might include:
A SIP call set-up request destined to a subscriber of the operators network
A SIP call set-up request destined to a roaming subscriber within the network
operators network
A registration request
The I-CSCF is responsible for assigning an S-CSCF to the subscriber for registration
requests. The I-CSCF may make the choice on the basis of traffic load or user identity.

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The service for which a user makes a request for example video streaming, music
download or MMS is resident on the AS (Application Server).
The PoC (Push-to-talk over Cellular) service is also supported by an application
server.

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Calls made to any CS domain will be routed to a BGCF by the S-CSCF, and the BGCF
will select an appropriate MGCF (Media Gateway Control Function).

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The IMS is based on an all-IP architecture, but it may at some point have to connect to
legacy networks such as the PSTN. To achieve this interconnection the packetised
media (voice) in the IP network will need to be converted into a legacy format, perhaps
TDM (Time Division Multiplexing). This conversion is performed by an MGW (Media
Gateway).
The MGW(s) within the infrastructure will be controlled and managed by an MGCF
(Media Gateway Control Function).
The relationship between the MGCF(s) and the MGW(s) may be one to many,
allowing for a flexible and scalable architecture.

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The MRF (Media Resource Function) is the only element that does not have a
mandatory presence in the IMS architecture. It is used for providing conferencing and
announcement services.
The MRFC (MRF Controller) and the MRFP (MRF Processor) are responsible for the
following:
Mixing media for voice/video
Conferencing (bridging)
Providing multimedia announcements
Processing media streams, e.g. audio transcoding
The functional split between the two elements is much the same as the split between
the MGW and the MGCF. It is controlled using the H.248/MEGACO protocol. The
MRFC receives call control signalling via SIP protocol messages.

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The HSS (Home Subscriber Server) represents the evolution of the HLR in legacy
mobile networks. It is the centralised database for network subscriptions and is
involved in registration and subscriber profile management.
The HSS will interact with other network nodes during location/registration procedures
and will also be involved in policy control, where that control is based on the
subscriber profile. Transactions between the HSS and other network nodes will
increasingly be based on the Diameter protocol.

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Like many areas of 3GPP standardisation, the IMS has continued to evolve to support
the changes seen in network and service architectures. By Release 8 this meant that
the IMS was able to support service delivery over a range of access and network
types, and specifically 3GPP LTE (Long Term Evolution).
As LTE is an all-IP architecture it was logical for 3GPP to propose that service control
and delivery within this architecture should be based on the IMS concept.

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Amongst other refinements 3GPP Release 8 created an IMS reference architecture

which could support generic access networks or IP-CANs (IP Connectivity Access
Networks), including LTE and SAE (System Architecture Evolution).
The 3GPP Release 8 architecture is largely identical to that of Release 5, but with a
few additional functional elements:

TrGW (Transition Gateway): Together with IBCF, the TrGW acts as a border
control function between operator networks. The TrGW is responsible for
handling the transport of traffic (e.g. RTP), while the IBCF handles SIP
signalling

IBCF (Interconnection Border Control Function): The IBCF supports functions

associated with roaming between operators of different networks.

MRB (Media Resource Broker): The MRB acts a broker between a range of
MRFs and various applications that may need access to these resources. The
MRB is responsible for the assignment, and subsequent release, of multimedia
resources.

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A number of network operators have either launched or are trialling IMS networks and
services. The services include PoC, multimedia conferencing, video sharing, FMC,
and gaming.
The benefits for a network operator introducing IMS include the ability to introduce
innovative services within very tight timescales (compared to traditional service
delivery platforms) and to embrace a potential pool of millions of developers (the IMS
being based on Internet standards).
As the IMS centralises control of services it is much easier to share data between
applications. For example, presence information may underpin many services and by
making it available on a server within the IMS those services can draw on this data.
The IMS is designed to be highly scalable, so network operators can enter new
markets very cost effectively. Combined with the migration to a single network
architecture to support both voice and non-voice services, the IMS offers operators a
cost-effective network migration plan.

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The IMS provides a single centralised point for service control, and importantly this
allows services to interact with one another. This aspect provides an important tool for
service enrichment; for example, a voice call could be enhanced with an
accompanying video stream.
Many innovative services are being targeted at IMS-based networks, but it is likely that
initial launches will only offer limited service sets which will expand over time. The
services launched may be aimed in some cases at the enterprise market, for example
multimedia conferencing, whereas gaming services would be aimed at the consumer
market.
It is likely that IMS services will be capable of supporting a high degree of user
personalisation, probably by means of web-based interfaces through which the user
can interact with the servers in the network.
With an expansion of access technology alternatives, such as 3G, 4G, Wi-Fi and fixed
line, IMS offers the possibility of seamless service access irrespective of customer
location and access means. This would really represent a convergence from the
service perspective.
However, there are a number of critical factors for user acceptance of IMS-based
services, including security, QoS and ease of use. These have got to be perceived as
robust by the end user if the new services are to be readily accepted.
Migration of users and services from legacy platforms to the IMS will affect both
operators and users; this migration needs to be as seamless as possible to minimise
transitional impact on customers.

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One of the main objectives of the LTE architecture is an overall simplification of the
network with a reduction in the number of nodes required in the radio access and core
network components. The evolution of the network is designed to optimise
performance and improve cost efficiency. Also, interoperability with the existing 3.5G
infrastructure is important, particularly mobility and handover between the networks.
The EPS (Evolved Packet System) is divided into the radio access network and the
core network.

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17

The E-UTRAN (Evolved Universal Terrestrial Radio Access Network) contains a single
element known as the eNB (evolved Node B). The eNB supports all the user plane
and control plane protocols to enable communication with the UE. It also supports
radio resource management, admission control, scheduling, uplink QoS enforcement,
cell broadcast, encryption and compression/decompression of user data.
Each eNB is connected to the core network on the S1 interface. The S1 interface
allows the eNB to communicate with the MME (Mobility Management Entity) via the
S1-MME interface, and the SGW (Serving Gateway) via the S1-U interface. The
interfaces support a many-to-many relationship between the eNB and the SGW and
MME.
The eNBs are networked together using the X2 interface. The X2 interface is based on
the same set of protocols as the S1 and is primarily in place to allow user plane
tunnelling of packets during handover to minimise packet loss.

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The Evolved Packet Core has two principal functions: high-speed packet handling and
mobility management, these functions are carried out by the SGW and MME. This
separation of function allows each to be implemented on a platform optimised for data
handling or message processing. This results in optimised performance, allowing
independent scaling of each component and efficient topological optimisation of
platforms to ensure consistent service, i.e. reduced latencies and maximised
throughput.
The SGW acts as a router, routing and forwarding packets of user data, it is able to
provide transport level packet marking, and the marking process may be used for QoS
management by other network elements. It also has some accounting functions for
UL/DL services.
The MME is the primary signalling node in the EPC. NAS (Non-Access Stratum)
signalling is terminated at this point, as well as signalling related to bearer
establishment and authentication of the UEs through interaction with the HSS (Home
Subscriber Server). It is also the decision point for SGW selection, and MME, SGW
selection during handover where EPC node change is necessary.

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S1 provides access to evolved RAN radio resources for the transport of user plane
and control plane traffic. The S1 reference point enables MME and UPE separation
and also deployments of a combined MME and UPE solution.
S2a/b provides the user plane with related control and mobility support between a
trusted/non-trusted non-3GPP IP access and the SAE Anchor.
S3 enables user and bearer information exchange for inter 3GPP access system
mobility in idle and/or active state. It is based on Gn reference point defined between
SGSNs.
S4 provides the user plane with related control and mobility support between the
GPRS core and the 3GPP Anchor. It is based on Gn reference point as defined
between SGSN and GGSN.
S5a provides the user plane with related control and mobility support between the
MME/UPE and the 3GPP anchor. It is for further study whether a standardised S5a
exists or whether the MME/UPE and 3GPP anchor are combined into one entity.
S5b provides the user plane with related control and mobility support between the
3GPP anchor and the SAE anchor. It is for further study whether a standardised S5b
exists or whether 3GPP anchor and SAE anchor are combined into one entity.
S6 enables transfer of subscription and authentication data for authenticating/
authorising user access to the evolved system (AAA interface).

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SGi is the reference point between the Inter-AS Anchor and the PDN. The PDN may
be an operator external public or private PDN, or an intra-operator PDN, e.g. for
provision of IMS services. This reference point corresponds to Gi and Wi
functionalities and supports any 3GPP and non-3GPP access systems.
The interfaces between the SGSN in 2G/3G core network and the EPC (Evolved
Packet Core), and between the SAE MME/UPE and the 2G/3G Core Network, are
based on GTP (GPRS Tunnelling Protocol). The interfaces will be based on GTP.

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The SGW (Serving Gateway) terminates the interface towards the E-UTRAN and is
the main packet routing and forwarding node in the EPC.
It has the following functions:

Provides transport-level packet marking in the uplink and downlink by setting the
DSCP (DiffServ Code Point), depending upon the QCI (QoS Class Identifier) of
the associated EPS bearer, which may be used for QoS management by other
network elements

Provides accounting functions based on the user and QCI, inter-operator

charging and uplink and downlink charging per UE, PDN, and QCI for roaming
within home routed traffic

Acts as a local anchoring point for inter-eNB handover and assists in the
reordering function by sending one or more end marker packets to the source
eNB immediately after switching the path

Acts as a 3GPP anchoring point for inter-RAT (Radio Access Technology)

handovers by providing the termination point for the S4 interface and relaying
traffic between 2G/3G systems and the PDN-GW (Packet Data Network
Gateway)

Provides idle mode functions such as packet buffering and initiation of network
triggered service request.

The S-GW is also one of the lawful interception points in the network.

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The PDN-GW (Packet Data Network Gateway is the link between the mobile device
and the services that reside in an external packet network such as IMS. The PDN-GW
provides an entry and exit point for UE connectivity with external data networks and
terminates the SGi interface towards the PDN.
It has the following functions:
Acts as a mobility anchor between 3GPP and non-3GPP technologies such as
3GPP2 CDMA2000 and WiMAX
Has responsibility for the allocation of user IP addresses
Provides support for charging, packet filtering and lawful interception

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The MME (Mobility Management Entity) is the primary signalling node in the EPC.
NAS signalling is terminated at this point. It is responsible for signalling related to
bearer establishment and authentication of UEs through interaction with the HSS. It is
also the decision point for SGW selection, and MME/SGW selection during handover
where EPC node change is necessary.
The MME handles roaming functions such as allocation of temporary identities,
admission control, and communication with the home HSS on the S6a interface.
A comprehensive list of MME functions is shown in the diagram.

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The HSS has the functionality of the 2G/3G HLR and AuC. It is the main database
containing subscription-related data for any given user, and it will use that data to
support the network entities providing services to that user. For instance, it may
provide support to call control servers in respect of routing/roaming procedures by
resolving issues relating to authentication, authorisation, naming/address resolution,
and so on.
The user-related information held in the HSS includes:
User identification, numbering and addressing information
User security information, including network access control information for
authentication and authorisation
Inter-system location information

User profile information

The HSS will also generate user security information for mutual authentication,
communication integrity checking, and ciphering.
As well as EPC functionality, the HSS may also be responsible for supporting call
control and session management entities of the CS domain, PS domain and the IMS.

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The end-to-end service is an application-level relationship established between the

application present on the UE and a peer application resident on a server or peer
device located inside or outside the current network. This service is dependent on the
underlying connections or bearers to ensure the required QoS.
The EPS bearer represents the connection between the UE and the EPC and is
established through NAS signalling. The QoS experienced by the end-to-end service is
determined by the EPS bearer. When the EPS bearer is established between the UE
and the EPC, the QoS parameters and packet forwarding treatment will be negotiated.
The external bearer is established by the gateway device and the interconnected
network.
The radio bearer, S1 bearer and S5/S8 bearers are all established to carry the data,
and the entitles managing the bearers will maintain a one-to-one mapping in order to
bind the radio to the S1 bearer, etc. All of these bearers are established using their
respective AS (Access Stratum) signalling.

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An EPS bearer is composed of the following elements:

The S5/S8 bearer, which transports packets between the SGW and PDN-GW
The S1 bearer, which transports packets between the SGW and the eNB
The radio bearer, which is implemented by a RLC (Radio Link Control)
connection between the eNB and the UE
The EPS system provides IP connectivity between a UE and an external PDN. This is
referred to as a PDN connectivity service. The PDN connectivity service supports the
transport of traffic flow aggregates consisting of one or more SDFs (Service Data
Flows).

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An important aspect of any all-IP network is a mechanism to guarantee differentiation

of packet flows based on QoS requirements. Applications such as video streaming,
HTTP, or video telephony have special QoS needs, and should receive differentiated
services over the network. With EPS, QoS flows called EPS bearers are established
between the UE and the PDN-GW.
Each EPS bearer is associated with a QoS profile, and is composed of a radio bearer
and a mobility tunnel. Thus, each QoS IP flow (e.g. VoIP) will be associated with a
different EPS bearer, and the network can prioritise packets accordingly.
The QoS procedure for packets arriving from the Internet is similar to that of HSPA.
When receiving an IP packet, the PDN-GW performs packet classification based on
parameters such as rules received from the PCRF, and sends it through the proper
mobility tunnel. Based on the mobility tunnel, the eNB can map packets to the
appropriate radio QoS bearer.
Within the EPC the Policy Charging Rule Function (PCRF) is responsible for
controlling the QoS policy; elsewhere in the architecture there will be a Policy
Charging Enforcement Function (PCEF), which will implement the policy rules for each
traffic stream.
The concept of the QCI (QoS Class Identifier) and the associated bit rates specify the
QoS information for service data flows and bearers on the Gx reference point.

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The EPS bearer QoS profile includes the parameters shown in the diagram.
The QCI is a label used as a reference to access-node-specific parameters that
control bearer level packet forwarding treatment and that have been preconfigured by
the operator owning the access node.
On the radio interface and S1, each PDU is indirectly associated with one QCI via the
bearer identifier carried in the PDU header. This also applies to the S5 and S8
interfaces.
Parameters include:
Scheduling weights
Link layer protocol configuration

Admission thresholds
Queue management

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The primary purpose of the ARP is to decide if a bearer establishment or modification

request can be either accepted or rejected due to resource limitations. In the case of
GBR bearers this is typically due to radio capacity.
The ARP can be used to decide which bearers should be dropped due to resource
limitations, for example by the eNB during handover. Once a bearer has been
established, the ARP has no impact on bearer-level packet forwarding as this is
provided by the QCI, GBR, MBR and AMBR parameters. The ARP is not included in
the EPS QoS profile sent to the UE.
It is possible to have bearers with different values of ARP. For example, an operator
could map voice to a bearer with a high ARP value whilst allocating video to a lowervalue ARP. If there is any congestion, such as at cell edge, the video bearer could be
dropped without affecting the voice bearer.
The ARP can also be used to free up capacity by allowing the eNB to drop low ARP
bearers.

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GBR (Guaranteed Bit Rate) defines the bit rate that can be expected to be provided by
a GBR bearer. This is only relevant to bearers where a guaranteed QoS is required,
for example voice services.
MBR (Maximum Bit Rate) limits the bit rate that can be expected to be provided by a
GBR bearer, and excess traffic may be discarded. The MBR parameters help to set a
limit on the data rate expected for the service. If the bit rate exceeds this figure then
the network can reduce the effective rate by applying traffic-shaping functions.
AMBR (Aggregate Maximum Bit Rate) applies to non-guaranteed bit rate bearers only.
Its purpose is to limit the overall bit rate of all bearers associated with this limit for a
given PDN. This means that when this limit is exceeded, the network may apply
traffic-conditioning algorithms for both uplink and downlink transmission, as for the
MBR limit which is defined at the EPS bearer level.

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The Bearer Type parameter indicates whether or not resources associated with the
bearer need to be permanently allocated during the whole bearer lifetime, reflecting
the difference between GBR and non-GBR bearers
A default bearer is provided during start up; this bearer is used to carry IP traffic that is
not associated with a dedicated bearer. This bearer is always non-GBR, which means
that the resources for the IP flows are not guaranteed at the eNB and have no
admission control. Multiple non-GBR bearers belonging to the same UE share an
AMBR and can suffer packet loss during congested periods.
A dedicated bearer is used to carry traffic that has been identified as requiring a
specific packet forwarding treatment. The dedicated bearer can be either GBR or nonGBR with a GBR bearer having a GBR and MBR.

L2PDB (L2 Packet Delay Budget) describes the maximum time that a link layer SDU
(Service Data Unit) can spend transiting through RLC (Radio Link Control) and MAC
(Medium Access Control) layers between the access node and the UE. In principle,
this attribute is used to derive waiting queues and MAC HARQ (Hybrid Automatic
Repeat Request) operating parameters. For GBR bearers, L2PDB indicates a
maximum limit for packet transmission time. For non-GBR bearers, this parameter
does not have the same strict meaning.
L2PLR (L2 Packet Loss Rate) describes the maximum ratio of L2 packets that have
not been successfully delivered. As for L2PDB, this parameter is intended for RLC and
MAC HARQ configuration.

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SIP (Session Initiation Protocol) is an Internet protocol which is used to establish,

maintain and clear down multimedia sessions. SIP was developed by the IETF
(Internet Engineering Taskforce) and has many similarities to both HTTP (Hypertext
Transfer Protocol), used for web transactions, and SMTP (Simple Mail Transfer
Protocol), used for email.
SIP was originally intended as a protocol for supporting Internet multimedia and
conferencing applications and was originally published in the RFC (Request for
Comments) 2543 in 1999.
There was growing interest in the use of SIP as a signalling protocol not just to set up
multimedia sessions, but also simple services such as telephony. The standard
continued to evolve with the release of RFC 2543bis and, in 2002, the release of RFC
3261, which obsoleted RFC 2543. This version of SIP currently has the status of a
proposed standard.
SIP is a significant protocol and underpins many VoIP (Voice over IP) services, and
has been selected by 3GPP (Third Generation Partnership Project) as the control
protocol within the IMS (IP Multimedia Subsystem).
It is important to note that SIP is a control or signalling protocol which can be used to
discover users and establish sessions between them. Session format is not dealt with
directly by SIP, but by a related protocol, SDP (Session Description Protocol). SDP is
used to negotiate media formats for sessions and the relevant SDP data is conveyed
within SIP messages. Nor is SIP used to transfer information (the media streams)
between users: again, a related protocol such as RTP (Real-time Transfer Protocol) is
used for this task.

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SIP is an application-level protocol that can operate either over the reliable service
provided by TCP (Transmission Control Protocol) or the unreliable service provided by
UDP (User Datagram Protocol). In turn, TCP or UDP will operate over IP. SIP will
work with both IPv4 and IPv6.

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SIP should be viewed as a signalling protocol for session establishment. Specifically, it

supports five elements of the overall communication requirements:
Location to determine the location of the endpoints for a session
User availability to establish the status of a user and their willingness to
participate in a session
User capabilities to establish the media format for the session and the ability
of the user devices to support particular formats
Session set-up the signalling required between the calling party and called
party (or parties) for session establishment
Session control procedures for transferring and terminating sessions, invoking
additional services and changing the session characteristics

Although this list incorporates all the functions required to establish and manage
sessions between parties to offer a complete service to a user, other protocols need to
be used in conjunction with SIP. Examples are:
SDP (Session Description Protocol) for defining session media formats
RTP (Real-Time Protocol) for conveying real-time media such as voice
RTSP (Real-Time Streaming Protocol) for controlling streamed media

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A basic SIP call only needs to involve two terminals; there does not have to be any
infrastructure in terms of servers between these devices. This may well be the case for
calls on a small private network where all the terminals are aware of the address of
other terminals. In other cases, one or more servers may be required to manage
functions such as locating parties for the call or interacting with service control logic.
A SIP session starts with a message known as an INVITE (in SIP terms this is known
as a method). This is forwarded by a SIP server towards the destination, in this case
the second terminal.
An exchange of a RINGING response (which is optional) and an OK response then
follows. From these responses the originating terminal can obtain the address of the
terminating device and therefore does not need to send signalling via the server
(unless the server stipulates that it should remain in the call).
In this example the ACK (acknowledge) is sent directly from the originating to the
terminating terminal. After this media flows can be established between the terminals,
although this is not done using SIP.
When the session is complete either terminal may end it by sending a BYE, which
results in an OK response.
The minimum sequence required to establish a SIP session is INVITE, OK and ACK,
although in practice many call scenarios use more messages than this because of the
complexities of media negotiation and to provide progress information.

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While SIP UA (User Agents) can connect directly without using proxy servers, a typical
call or session is more likely to use one or more proxies to achieve the end-to-end
routing between the UAs. The illustration shows a call between two UAs which transits
through two proxy servers.
The lines between the entities are SIP messages, which are either requests or
responses. The responses are numbered with each series, (100, 200, etc.),
representing a grouping of functions. Requests will invoke a function on a server
(including the UAS [User Agent Server]), which is referred to in SIP as a method.
Methods include INVITE, BYE, CANCEL and REFER.

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SIP can establish, manage and clear down multimedia sessions, but does not contain
the functionality to describe the media formats for a session. This description is
performed by another protocol, SDP (Session Description Protocol). SDP is described
in RFC 2327, and like SIP was originally intended for Internet multimedia multicasting.
The descriptive capabilities of SDP include:

Session name and purpose

Time(s) the session is active

The media comprising the session (media type, transport protocol, media
format)

The information to receive those media (addresses, ports, formats and so on)

Other elements specifically used for multicast

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The overall architecture for policy control in LTE and other networks, consists of a
PDP (Policy Decision Point) and a PEP (Policy Enforcement Point).
The PDP makes policy decisions, although these may be arrived at after
communication with other network nodes such as the HSS or an AS. These policy
decisions are communicated to the PEP.
The PEP is responsible for enforcing the policy rules on a session and flow basis.

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The PCRF (Policy and Charging Rules Function) includes the policy control decision
and flow-based charging control functionalities and provides network control regarding
the SDF detection, gating, QoS and flow-based charging towards the PCEF. The
PCRF may activate, modify and deactivate a PCC rule at any time, over the Gx
reference point. However, the modification procedure is applicable to dynamic PCC
rules only.
The PCEF (Policy and Charging Enforcement Function) is located in the PDN-GW and
provides SDF detection, user plane traffic handling, triggering control plane session
management (where the IP-CAN permits), QoS handling, SDF measurement and
online and offline charging interactions.
The PCEF enforces policy control as directed by the PCRF in two ways: QoS
enforcement, and gate enforcement, whereby the PCEF enables an SDF that is
subject to policy control to pass through the PCEF if the corresponding gate is open.
If an SDF (defined by an active PCC rule) is subject to charging control, the PCEF will
allow the SDF to pass through the PCEF as long as the PCC rule is active and the
OCS has authorised credit for the charging key. The PCEF may let a service data flow
pass through the PCEF during the course of the credit reauthorisation procedure.

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A PCC rule holds the information required to enable the user plane detection of the
policy control and proper charging for a service data flow.
Policy and charging control rule operations consist of activation, modification and
deactivation of PCC rules.
There are two different types of PCC rules defined by the operator: dynamic rules, and
predefined rules.
The activation of a dynamic PCC rule provides the PCC rule information to the PCEF
and is provisioned by the PCRF via the Gx reference point.
Predefined PCC rules are directly provisioned into the PCEF and only referenced by
the PCRF. Activation of a predefined PCC rule provides an identifier of the relevant
PCC rule to the PCEF via the Gx reference point. If it is not known in the PCRF it may
be done by the PCEF based on operator policy; however, the PCEF may only activate
it if there are no UE-provided traffic mapping information related to the IP-CAN bearer.

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A PCC rule identifies the service data flow and specifies the parameters for charging
control which may depend on the subscription data.
The following charging models are supported:
Volume
Time
Volume and time
Event
No charging
The charging model may change as a result of events identified by the OCS. For
example, as a result of having used the SDF for a certain amount of time and/or
volume, the user gets to use some services for free.
When a user is identified as roaming, a different rate or model may be applied to the
one used when the user is in the home network. It is also possible to apply different
rates and charging models dependent on a users location, beyond the granularity of
roaming.
It is possible to apply a separate rate to a specific service, for example to allow the
user to download a certain volume of data associated with a particular service for free
whilst any additional downloads are charged at the required rate.
It is also possible to change the rate based on the time of day.

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The flexible charging scheme within the EPS is based on the collection of events and
information. This is stored in a formatted record called the CDR (Call Details Record),
the format of which is determined by the type of service, and includes:
PS domain access
IMS service
Any type of service hosted by an AS, such as MMS
MBMS (Multimedia Broadcast and Multicast Service)
The CDF (Charging Data Function) gathers information from the nodes via the Rf
interface and creates the CDR. The Rf interface is based on the Diameter protocol and
makes use of extensions specific to the charging process.

The CGF (Charging Gateway Function) is the gateway between the core network
nodes and the billing domain and is responsible for the collection of the CDR via the
Ga interface using FTP (File Transfer Protocol).
During a session, information is generated by the relevant node and is used by the
CDF or CGF to build the CDR. Any redundant information such as data traffic volumes
or session start and stop timestamp is used to check the consistency between the
views reported by the network elements.

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Diameter is an AAA (Authentication, Authorisation and Accounting) protocol that was

defined to overcome some of the problems or limitations inherent in the RADIUS
protocol.
RADIUS is widely deployed for applications such as Internet access through an ISP
(Internet Service Provider), where typical access might be dial-up and the user is
expected to enter (or have stored) a usernamepassword pair for authentication.
Diameter was designed to overcome some of the issues that surround RADIUS and is
specifically for supporting access to IP-based networks.
The features included in diameter which overcome the limitations of RADIUS include:
Operation over reliable connections (TCP or SCTP)

Increased length of attribute values

Increased number of pending messages
Server fail-over capability
Replay protection
End-to-end capability

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As a minimum requirement, a Diameter implementation will include a client and a

server. The Diameter client is typically at the edge of a network and controlling access,
for example a NAS (Network Access Server), mobility agent or CSCF (in an IMS).
Servers are the devices which handle the AAA requests for a particular domain
(realm).
However, other entities may be present, described below.
Relay agents (or relays) agents route Diameter messages based on the value of
routing-related AVPs. They do not analyse any of the other AVPs in a message nor do
they originate messages, but they may change the value of a routing AVP depending
on circumstances.
Proxy agents, like relays, can route Diameter messages. However, it can also
implement policies and therefore may be used to control resources and provide
admission control. Proxy devices therefore provide a service to devices such as NASs.
Redirect agents return special responses to requests, which allow the requesting client
to resend its request to a redirected end-point. These agents can be used in scenarios
where a centralised routing function is implemented.
Translation agents support translation between two AAA protocols, such as RADIUS
and Diameter, and are therefore important elements in the transition towards Diameter
implementation.

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