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Voice and Internet multimedia in UMTS networks

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UMTS (Universal Mobile Telecommunications System), UMTS 3GPP International Organization for Standardization is the development of the global 3G standard. Its main access networks, including CDMA and packet-based core network and a series of technical standards and interface protocols.

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									     Voice and Internet multimedia in UMTS networks


     M C Bale




     Voice telephony is the predominant service on today’s cellular mobile networks, in terms of number of customers, revenues
     and network usage. However, it is difficult to predict how long this will be the case given the rising demand for new Internet
     multimedia services. It is therefore essential that 3rd generation (3G) mobile networks support a voice telephony service, but
     also that these networks are also capable of providing Internet multimedia services using the same technology.

     This paper provides an overview of how voice telephony is provided in the initial phase of the universal mobile
     telecommunications system (UMTS). It then describes how this is expected to evolve in later phases — so that voice
     telephony becomes one of a large number of multimedia services provided from a common Internet protocol-based mobile
     network.



     1.   Introduction
                                                                       information services with which it will be integrated. This
     T     he main driver behind 2nd generation digital mobile
           networks, such as the global system for mobile
     communications (GSM) [1], was the need to provide a voice
                                                                       requires a more radical approach to the provision of voice
                                                                       services, one that is more aligned with the Internet and the
     telephony service to mobile users. This has been achieved         protocols standardised by the Internet Engineering Task
     with incredible success. Moreover, GSM has established the        Force (IETF) [5]. This challenge is being addressed by
     starting point from which future mobile networks must             3GPP in the production of the Release 4 and 5 standards,
     evolve and an important benchmark for voice services that         and by the IETF in the production of the protocols needed to
     the 3rd generation of mobile networks must exceed in terms        realise mobile Internet multimedia.
     of functionality and quality.
                                                                           This paper initially provides an overview of how a voice
          The Universal Mobile Telecommunications System               telephony service is supported by a UMTS network
     (UMTS), the 3rd generation network and systems                    conforming to the 3GPP Release 1999 standards. It then
     standardised by the 3rd Generation Partnership Programme          describes the proposed solution currently being standardised
     (3GPP) [2], aims to provide voice services that will meet the     by 3GPP for Internet multimedia services (including voice)
     needs of mobile users. This is being done in collaboration        known as the Release 5 standards. This solution is
     with the International Telecommunications Union (ITU)             illustrated with message sequence flows to show the
     ‘International Mobile Telecommunications — 2000’ project          dynamic aspects of the solution and the application of the
     [3].                                                              various protocols. It is assumed that the reader already has
                                                                       an awareness of GSM and general packet radio service
         In the initial phase of UMTS, defined by the 3GPP             (GPRS) networks.
     Release 1999 standards, the voice telephony service is
     essentially an evolution of the GSM voice service that                 Work to address the challenges of providing voice and
     benefits from the 3rd generation technologies adopted for         multimedia services in a mobile and wireless Internet
     the UMTS Terrestrial Radio Access Network (UTRAN) [4].            environment is progressing rapidly within 3GPP as well as
                                                                       the other bodies producing standards for this area (such as
         However, the customer’s needs for mobile voice                the IETF). However, the reader should be aware that there is
     telephony must also be considered in the light of the             still much work to be done, especially at a detailed level. At
     growing demand for mobile Internet multimedia services. In        the time of writing, this paper reflects current views, which
     particular, voice will be a feature of many of these              may differ from the actual standards when they are
     multimedia services, e.g. videoconferencing, mobile               completed. To aid understanding of some of the issues,
     commerce (mCommerce), games and multimedia mail. To               potential solutions are described, but it should be recognised
     enable such services, it is important that the voice service is   that these are only illustrative and may not be endorsed as
48   as much part of the mobile Internet as the data and               standards in the future.


     BT Technol J Vol 19 No 1 January 2001
                                                           VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

2.   Voice in the 3GPP Release 1999 network                            Figure 1 shows the overall network for the support of
                                                                    voice services in the 3GPP Release 1999 standards, and is
R    elease 1999 is the first phase of the 3GPP standards for
     UMTS. This is a completed set of standards that
defines a UMTS network able to provide users with voice
                                                                    more fully described in Lobley [6].

                                                                        To achieve compatibility with GSM, the Release 1999
and data services fully compatible with those of GSM and
                                                                    network effectively adopts the GSM core network and
GPRS. The standards allow users to migrate on to the
                                                                    service architecture. This has a significant benefit to the
UMTS and to roam seamlessly between UMTS and GSM/
                                                                    network operator since it enables a cost-effective and low-
GPRS networks without any loss of capability. It also has
                                                                    risk evolutionary approach to be taken for the deployment
the benefit to the network operator of being able to target
                                                                    of UMTS. However, in the UTRAN, changes in both the
the introduction of UMTS to specific geographical areas,
                                                                    architecture and the radio and transport technologies
while relying on existing GSM and GPRS networks to
                                                                    employed result in differences from GSM, but also enable
provide coverage in other areas.
                                                                    some improvements to the way in which voice services are
    Specifically, current GSM networks support voice and            provided. The main areas where the UTRAN affects voice
low-speed data services that are circuit-switched, so called        services are described below.
because the voice or data is carried between users in bearer
circuits that are switched into place across the network for a      •     Improved quality of service in the radio access
time period, under the control of signalling from the users.
In contrast, the GPRS network supports packet-switched                    The use of wideband code-division multiple access
data services. For the purposes of this paper, only the voice             (WCDMA) and the various modes of operation in the
services in Release 1999 are described, but the descriptions              radio access can improve the quality of the voice
also apply to low-speed circuit-switched data services.                   service in terms of availability and reliability [4].


                                                                  application
                                                                 and service                               HLR
                                                                 environment                                                 A



                                          BSS
                                                               GSM A
                                                              interface


                        GSM radio access
                           network

                                                                                  VLR
                                                                                                                             B

                                                                                 MSC                  GMSC



                                          RNC                                            circuit-switched domain
                                                           UMTS lu-CS                       (TDM or ATM AAL2
                                                            interface                         core network)



                                                                                   EIR

                                          RNC

                                                   UMTS lu-PS       C
                      UMTS terrestrial radio        interface
                     access network (UTRAN)

                           signalling       A   mobility management signalling to other networks

                           speech paths     B   speech circuits and call signalling to other networks (e.g. PSTN)

                           packet data      C   packet data and signalling to the packet-switched domain (i.e. GPRS)

                                           Fig 1 3GPP Release 1999 voice network overview.                                                 49


                                                                                                   BT Technol J Vol 19 No 1 January 2001
     VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

     •    Well-defined interface between a UTRAN and a core           although these may not meet all the quality-of-service
          network                                                     requirements of a Release 1999 network.

          The interface between the radio network controllers             In Release 1999, the user’s speech is digitally sampled
          (RNCs) and the core network, the Iu interface, is more      by the mobile user equipment, and then coded for
          clearly defined and open, such that a UTRAN from one        transmission. The default speech coding, which must be
          vendor will interoperate with a core network from           supported by all mobile user equipment (terminals) and the
          another vendor. The Iu interface itself is separated into   UTRAN, is adaptive multi-rate (AMR). The AMR coder
          the Iu-CS interface between the RNC and the core            supports eight source rates ranging from 4.75 kbit/s− 1 to
          network circuit-switched domain, and the Iu-PS              12.2 kbit/s− 1, and is rate-controlled which enables it to
          interface between the RNC and the core network              rapidly switch between these at any point in the call. The
          packet-switched domain (not shown in Fig 1). The            AMR coder encodes and decodes the digitally sampled
          separation of the core network domains and the Iu           speech to make optimum use of the battery power and
          interface allows the deployment and evolution of voice      bandwidth available, particularly on the radio link between
          services independently of packet data services in           the mobile equipment and the radio base stations (node B).
          Release 1999.                                               The bit rates are selected depending on the quality of speech
                                                                      required and the quality of the transport provided by the
     •    Use of ATM as the transport technology
                                                                      network, and primarily that of the radio link. The AMR
          ATM is used as the transport technology between the         coder also supports a low-rate background noise encoding
          radio base-stations and RNCs, between RNCs, and             mode to reduce transmission during silence, further
          between the RNCs and the core network (the Iu               reducing bandwidth and battery usage in the user
          interface). Both circuit-switched and packet-switched       equipment. In addition to AMR, other speech coding may
          services are carried in ATM cells, using appropriate        be optionally selected, such as enhanced full rate (EFR) or
          adaptation layer protocols. In the case of the voice        full rate (FR), as also specified for GSM. Within the core
          bearer circuits this is ATM adaptation layer 2 (AAL2),      network, the ITU-T Recommendation G.711 speech coding
          and for the signalling is ATM adaptation layer 5            at 64 kbit/s− 1 or 56 kbit/s− 1 is generally used as in the public
          (AAL5). ATM provides a number of benefits in the            switched telephony network (PSTN) and GSM core
          access network, such as the ability to transport packet-    networks. Transcoding from AMR (or other speech coding)
          and circuit-switched services with low delay, high          to G.711 is performed in the MSC.
          bandwidth and manageable quality of service.
          Conversion of ATM to the circuit-switched time                  If the user’s equipment at both ends of a voice call use
          division multiplexed (TDM) technology, if used to           the same coding, then transcoding to G.711 (or other
          switch the voice paths in the core network, can be          codings) is not necessary. There are two procedures that can
          performed by the mobile switching centre (MSC) or by        be adopted to remove or reduce transcoding, namely:
          a gateway function between the RNC and the MSC.
                                                                      •    tandem-free operation of transcoders — where inband
     •    Speech transcoders located in the core network                   signalling between the transcoders determines the
                                                                           transcoders in use and allows the transcoders to drop
          Speech transcoding is performed in the MSC in                    out of the speech circuit if both terminals are using the
          Release 1999, rather than at the base-station sub-               same speech coding,
          system of the GSM radio access network. The
          relocation of this function into the core network allows    •    transcoder-free operation — where the mobile
          operators to provide lower cost access transmission              terminals negotiate the speech coding during call set-
          networks, and eases the introduction of transcoder and           up, and transcoders are only inserted into the speech
          tandem-free operation.                                           path if end-to-end compatibility cannot be achieved.
                                                                          Although considered for Release 1999, it is not until
         A significant benefit of retaining the GSM core is that      Releases 4 and 5 that standards will be available for
     the MSCs can interface to both the UTRAN and existing            tandem-free operation and transcoder-free operation.
     GSM radio access networks, and more easily support user
     roaming and in-call handover from the UMTS to GSM                    As with GSM, signalling from the user to the network
     networks. Within the core network, the only notable change       broadly falls into two categories — call-related signalling
     from GSM is that voice services may be supported either on       for establishing, maintaining and terminating voice calls,
     circuit-switched TDM (as in GSM) or via ATM transport.           and non-call-related signalling for mobility management
     Again, AAL2 is recommended for providing the voice               (e.g. for location registration, roaming and in-call
     bearer circuits and switching if ATM is used. Other              handover). The signalling protocols and procedures are
     transport protocols such as ATM adaptation layer 1 or            generally the same as for GSM, although new lower layer
50   voice-over-IP solutions could in theory be used instead —        protocols provide adaptation to the underlying ATM


     BT Technol J Vol 19 No 1 January 2001
                                                           VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

transport in the UTRAN. Within the core network and for            sessions and the interconnection to other networks, such as
interconnect to other networks, the ITU-T recommendations          the PSTN and other UMTS networks. The IM domain also
for Signalling System 7 (SS7) are used, again with lower-          relies on a managed core IP network that is enabled to
level adaptation layer protocols in the case where ATM             provide the quality of service needed for voice and
transport is used.                                                 multimedia services.

    Supplementary services, such as call diversion and                 The main reasons for the introduction of the IM domain
caller identity, are provided from the MSCs, which also            are to enable new services and to reduce cost. The IM
provide tones and announcements to the user. More                  architecture uses IP and the other protocols standardised by
advanced voice services can be provided from the                   the Internet engineering task force (IETF) as interfaces to
application and service environment [7]. A user profile,           component ‘building blocks’ of the Release 5 network.
containing information on the individual’s subscribed
services, is provisioned into the home location register               These protocols provide a very adaptable suite of
(HLR) for that user. This is then copied into the                  technologies for building packet-based networks and
corresponding visitor location register (VLR) in the MSC           services, and the growth in the use of these protocols and
responsible for controlling users’ calls, so that their services   associated networking equipment over the last decade has
can be provided as they change location. For billing               resulted in considerable cost reductions. However, while the
purposes, call detail records, for example containing              IETF protocols can be adopted to provide many of the
information on call duration and destination, are generated        functions of the IM domain, each UMTS service has
by the MSCs and sent to the operator’s billing engine.             specific requirements that impact on the overall design of
Information may also be collected from the HLR for billing         the network and the detailed information carried within the
purposes. The MSC also communicates with an equipment              protocols. Therefore, to determine the IM network and
identification register (EIR), for example to validate             protocol design, the services to be supported must be
whether the mobile terminal is a stolen one.                       understood.

    With the Release 1999 standards completed, it is                  Examples of the services that will be supported in
anticipated that UMTS Release 1999 voice networks will be          Release 5 by the IM domain are:
operational by 2002, with operators already beginning to           •   voice telephony,
deploy network and UTRAN equipment in order to meet
this date. However, it is not until the second phase of UMTS       •   real-time interactive games,
standards that support for other real-time multimedia              •   videotelephony,
services is defined.
                                                                   •   instant messaging,
3.   Voice and multimedia in the 3GPP Release 5 network
                                                                   •   emergency calls,

T    he following phases of UMTS evolution specify how
     voice and multimedia can be supported by an Internet
Protocol (IP) transport service. Currently, two phases are
                                                                   •   multimedia conferencing.

                                                                        These services tend to share a number of characteristics
defined:                                                           — they are generally a conversational session between two
•    Release 4, which includes the migration of the Release        or more parties requiring some degree of real-time
     1999 circuit-switched domain core network and                 interactivity. The real-time aspects of the service can be
     services to an IP transport, and is described further in      described in terms of the quality of service of the transport
     section 5 of this paper,                                      (such as transmission delay or packet jitter) and of the
                                                                   session (or call) control, such as time to establish the
•    Release 5, which takes a more radical approach to the         session.
     introduction of conversational and interactive
     multimedia services on to an end-to-end IP transport              To meet the interactive needs of these services, the
     provided by an enhanced general packet radio service          GPRS network provides quality-of-service levels — for
     in the packet-switched domain.                                example by operating at low levels of network utilisation or
                                                                   by employing mechanisms such as Diffserv (see RFCs 2474
   These releases were previously known singly as Release          and 2475 [5]). Additionally, IP version 6 (IPv6 — see RFC
2000.                                                              2460 [5]) has been recommended as the transport protocol
                                                                   to be used for the IM domain, since this has a number of
    Release 5 specifies voice and multimedia services that         features that are beneficial to UMTS networks (such as a
make use of GPRS for the transport of speech and                   large address space, support for packet prioritisation, and
signalling, rather than the circuit-switched domain trans-         easier manageability).
port. A new core network domain, the Internet multimedia
(IM) core network subsystem, or IM domain for short, is               The IM domain has four important roles in meeting the
introduced for the control of voice and multimedia calls and       requirements of services:                                         51


                                                                                             BT Technol J Vol 19 No 1 January 2001
     VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

     •    it enables users and applications to control the sessions                 •     it generates call detail records (CDRs), for example
          and calls between multiple parties, for example to                              containing information on time, duration, volume of
          establish, maintain, modify and terminate calls1,                               data sent/received, and the call participants — the
     •    it controls and supports network resources (such as
                                                                                          CDRs, together with records from the GPRS network
                                                                                          on the data volumes transmitted and received are used
          media gateways and GPRS gateway support nodes
                                                                                          for charging purposes.
          (GGSNs), multimedia resource functions (MRF) and
          the core IP network) to provide the functionality,                            An overview of the IM domain and its relationship with
          security and quality required for the call,                               the GPRS packet-switched domain is shown in Fig 2. The
     •    it provides for registration of users on the ‘home’ and                   purpose of each of the functional entities is more fully
          ‘roamed to’ networks, so that users may access their                      described in Lobley [6].
          services from any UMTS network,                                               The IM domain architecture complements the voice
     1
       Strictly speaking, sessions and calls are different (see RFC 2543 [5] for    over IP (VoIP) protocols and architectures developed by the
     a definition of each). However, for the purposes of this paper the term        IETF [5], ETSI Tiphon [8] and ITU-T Study Group 16 [3],
     ‘call’ is used to refer to simple cases where calls and sessions can be
     considered the same, for example, in the case of a point-to-point voice        although these were primarily developed for fixed IP
     telephony call.                                                                networks. Supporting VoIP in a mobile and wireless




                                                                                application                                               A
                                                                                                                             signalling
                                                                               and service              HSS                  gateway
                                                                               environment




                                                                                                                                          B
                                                                                                   CSCF



                                                                                                                                          C
                                                                                                                             signalling
                                                                                                              MGCF
                                                                     EIR                                                     gateway
                                    RNC



                                                                                         DHCP
                                                                                        and DNS                                           D
                                                                                         servers
                                                                                                                              media
                                                                                                              MRF            gateway
                                    RNC
                                                                                   GGSN
                                                                SGSN                                                                      E

                 UMTS terrestrial radio                                     packet-switched               Internet multimedia domain
                access network (UTRAN)                                      domain (GPRS)                     (IPv6 core network)
                                                      UMTS lu-PS
                                                       interface

                             signalling          A     mobility management signalling to other networks

                             speech paths         B    call related and mobility management signalling to other Release 5 networks

                                                 C     call related signalling to other circuit-switched and VoIP networks

                                                 D     circuit-switched speech circuits to other networks (e.g. PSTN and GSM)

                                                 E     speech paths to other Release 5 and other VoIP networks

52                                                           Fig 2   3GPP Release 5 network overview.


     BT Technol J Vol 19 No 1 January 2001
                                                           VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

environment raises a number of additional requirements,                    user A                IM domain               user B
which are being addressed by the 3GPP group. In particular,              equipment               call control          equipment
they include:                                                                      INVITE
                                                                          1                              INVITE
•     the ability of the network to hand over a call (signalling                              trying
      and speech paths) from one radio base-station (node B)                                                        ringing
                                                                                                                                  2
      to another, without perceivable loss of speech quality,                                 ringing
                                                                                                                       OK
      for example as a user moves between radio base-                                                                             3
                                                                                                 OK
      stations,
•     the ability of the network to cope with the additional              4
                                                                                   ACK
                                                                                                          ACK
      delays imposed on the speech path due to radio access
      and use of AMR coding, without perceivable loss of                           both way speech path established
      speech quality,
•     the ability of the network to allow users to roam to                5
                                                                                   BYE
                                                                                                          BYE
      another operator’s network (a visited network), and
      still receive service,                                                                                           OK
                                                                                                OK
•     the ability of the network to control the voice service of
      a roaming user from either the user’s home network or
      the visited network.                                                    Fig 3      Simple establishment of a speech path.


    The first two points are addressed by the mechanisms               name representing the called user (either similar to an
used to transport IP packets carrying speech and IP packets            Internet e-mail address or a telephone number), a
carrying signalling over the GPRS network and IM domain                description of the call (e.g. codec to be used) and the
core IP network. The subsequent points are addressed by the            address of the endpoint of the speech path on user A’s
registration, discovery and call control procedures of the IM          equipment (e.g. a telephone). A call control entity in
domain.                                                                the IM domain receives this invitation, and confirms
                                                                       back to user A’s telephone that it is trying to contact
3.1     Overview of VoIP in 3GPP Release 5                             user B’s telephone. The call control entity then
                                                                       performs a database look-up to translate user B’s name
    In common with fixed network VoIP, digitised speech                to an address to which it can route the invitation. On
from each user is carried in IP packets between one user’s             resolving the address, the IM domain call control
terminal equipment and another by an IP network. The path              routes the invitation on to user B’s telephone.
that these packets take through the network is referred to as
the speech path. Unlike a circuit-switched environment, the
                                                                   •   Alerting (2)
packets may individually take different routes through the             On receiving the invitation, user B’s telephone alerts
IP core network to a common exit point of the IP core                  user B of the incoming call, and informs user A via the
network, rather than be forced along a specific circuit.               IM domain that the called telephone is ringing.
However, in reality, it is likely that the packets will follow
the same route through the network if the network is not           •   Answer (3)
congested.                                                             When user B answers, the telephone accepts the call by
                                                                       sending an OK back to user A’s telephone via the IM
    To establish a speech path, and synchronise the users
                                                                       domain. This message contains the address on user B’s
and their equipment, call control functionality is
                                                                       telephone on which the speech path should terminate,
programmed into the user’s equipment and network. These
                                                                       as well as the agreed call description.
call control functions communicate using signalling
messages. For example, the call control enables passing of         •   Acknowledge (4)
the endpoint addresses for the speech paths on the user’s
                                                                       User A’s telephone acknowledges acceptance of the
equipment and the negotiation of the network and user
                                                                       call, and the speech path is established — both
equipment resources needed for the call, such as codecs and
                                                                       telephones now know each other’s address and are able
the quality of service required. Figure 3 shows a simple call
                                                                       to send speech packets to each other.
establishment to create a VoIP speech path, which is
described below.                                                   •   Clear (5)

•     Invite (1)                                                       When the users have finished talking to each other, the
                                                                       call is cleared, for example by user A’s telephone
      The calling user (A) initiates the call by inviting the          sending a BYE to user B’s telephone via the IM
      called user (B) to the call. This invitation contains a          domain. Both telephones then free up any resources                  53


                                                                                                   BT Technol J Vol 19 No 1 January 2001
     VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

           allocated for the speech path, and user B’s telephone                                 session initiation session description
           confirms that the speech path has cleared by sending an                                protocol (SIP)      protocol (SDP)
           OK back to user A’s telephone via the IM domain.

     3.2     Signalling                                                           TCP or UDP
                                                                                                          TCP or UDP payload
                                                                                    header
         The signalling protocol for registration and call control
     in the IM domain is based on the session initiation protocol
     (SIP) (see RFC 2543 [5]). In simple terms, the control of the    IP header                          IP payload
     call relates to inviting and synchronising the various
     participants in the call. It also enables the participants to
                                                                                    Fig 4   Transport of SIP and SDP in IP.
     describe and share information about the characteristics of
     the terminating equipment and the speech path between the            To send and receive SIP messages over the GPRS
     users. This information is known as the session description,     network, the user’s equipment must establish a bi-
     and could include, for example, the coder used for the           directional packet data session with the IM domain for the
     speech and the bandwidth needed for the speech paths. SIP        signalling path. This is known as a packet data protocol
     essentially provides the invitation and synchronisation of       (PDP) context activation and is a common GPRS procedure
     the participants, and it uses the session description protocol   for establishing an IP data path between the user’s terminal
     (SDP) (see RFC 2327 [5]) to describe the session.                and the network. The signalling path is a separate PDP
                                                                      context to the speech path, and must be done before any SIP
         Both of these protocols are standardised by the IETF,        messages can be sent (e.g. for registration). The GPRS
     and are simple to use and program, text-based, and can be        quality-of-service class for signalling is interactive,
     readily adapted to support a wide range of multimedia            although the detailed parameters define the specific
     applications.                                                    transport quality requirements (such as high priority, but
                                                                      lower sensitivity to delay and jitter). The establishment of
         In the SIP protocol, users are addressed by a SIP            the PDP context for signalling assigns an IP address to the
     uniform resource locator (URL), which has the form               mobile terminal and allocates bandwidth and the required
     user_name@network_domain_name, where the user_name               quality of service over the UTRAN and GPRS network for
     and network_domain_name are textual names (similar to an         the signalling. The assigned IP address, together with the
     Internet e-mail address). PSTN telephone numbers may be          SIP port number is used to address the SIP client in the
     textualised so that they can conform to this format, and thus    mobile terminal. This IP address can also be used
     allow users to be addressed from the PSTN (and vice versa).      subsequently as the IP address for the speech paths.
                                                                          When the mobile terminal is to be switched off or roams
         The SIP URLs provide a flexible means of addressing,         to another network, the PDP context for the signalling is
     but can also be easily included in Web pages as hyperlinks,      deactivated.
     that when activated initiate a SIP session to that user.
                                                                           One of the benefits of using GPRS to carry the
                                                                      signalling path is that the GPRS controls the handover of
         The functional entity in the IM domain that performs the
                                                                      the signalling path as a user moves between the radio cells.
     call control is known as the call state control function
                                                                      It is therefore not essential that the IM domain be aware of
     (CSCF). These have been classified into different types [6],
                                                                      the geographical location of a user. However, some voice
     and provide the functions of a stateful SIP proxy server, as
                                                                      services may require location information (for example, to
     defined in RFC 2543. Correspondingly, the user’s
                                                                      restrict the user to certain cellular areas or for emergency
     equipment provides the functions of a SIP user agent, as
                                                                      services). In these cases, the IM domain will need to obtain
     defined in RFC 2543. SIP messages are usually transported
                                                                      location information from the GPRS network or home
     by the transmission control protocol (TCP) (see RFC 761
                                                                      subscriber server (HSS).
     [5]) or user datagram protocol (UDP) (see RFC 768 [5]).
     However, SIP is transport independent and other protocols,           It should be recognised that SIP only supports the call
     such as the stream control transmission protocol (SCTP)          control procedures for the establishment of the speech path.
     which runs over UDP, can be used to provide a higher level       The allocation of the actual bandwidth and quality-of-
     of quality than TCP.                                             service needed to provide the IP transport for the speech
                                                                      packets over the UTRAN and GPRS network is requested
         SIP itself includes reliability mechanisms that can be       by the user’s equipment as additional PDP contexts using
     used if running over an unreliable transport, but these can be   the GPRS protocols. Similarly, the control of quality-of-
     omitted if a reliable protocol transport such as TCP or SCTP     service mechanisms in the core IP network is independent
     is used. These protocols are carried over IP over the GPRS       of the call control procedures, and instead relies on other
54   network and IM domain IP core network (see Fig 4)                solutions (such as Diffserv).


     BT Technol J Vol 19 No 1 January 2001
                                                          VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

    Within the IM domain, additional protocols are used                                                 AMR coded speech samples
between network elements in order to provide the full voice                                               (e.g. 20ms of speech)
service. These include a mobility management protocol
between the CSCFs and the HSS (this could be MAP [2] or
LDAP (see RFC 2251 [5])), and media gateway control                                            RTP
                                                                                                                RTP payload
protocol, such as the H.248/Megaco protocol (see RFC                                          header
2885 [5]), jointly produced by the ITU-T and IETF.

3.3    Transport of speech packets                                                  UDP
                                                                                                         UDP payload
                                                                                   header

    So that speech may be sent and received in IP packets,
the user’s actual speech is sampled by the user’s equipment
and coded for transmission (e.g. using the AMR coder).                IP header                      IP payload
Once a certain number of samples have been taken, usually
between 10 ms and 40 ms, the coded samples are packetised                          Fig 5   Transport of speech in IP.
and sent to the network. The time taken to packetise the               To provide a higher quality of service than ‘best-effort’,
speech samples adds considerable delay to the speech path         the GPRS network specifies a conversational class of
and can necessitate echo cancellation devices within the          service that prioritises speech packets for low delay and low
terminal equipment or network. However, it is inefficient to      jitter. Similarly, mechanisms such as Diffserv or the
simply send smaller packets of speech samples, since this         resource reservation protocol (RSVP) (see RFC 2205 [5])
increases the bandwidth needed and requires the IP routers        may be used within the IM domain core IP network to
to route more speech packets. In addition to the end-to-end       provide a high-quality service.
transmission delay, delay due to packet jitter is also
encountered at the termination of the speech path where the           The AMR coder is the default codec that all Release 5
packets have to be buffered so that the digitised speech can      terminals must support, although other codecs may also be
be synchronised before it is played out.                          supported. As transcoder-free operation is supported by the
                                                                  call control signalling, AMR coding of the speech can be
    The speech packets are transported between users’
                                                                  used end-to-end between the users’ items of equipment,
equipment in UDP/IP packets by the GPRS network and IM
                                                                  without the need to transcode to another standard. However,
                                                                  with AMR coding of speech at 12.2 kbit/s− 1, a 20 ms
domain core IP network. A framing protocol is required for
the speech samples, e.g. to synchronise samples and control
                                                                  sample of speech results in an RTP speech payload that is
the sampling rate. The IETF protocols for framing voice and
                                                                  roughly half the size of the combined IP, UDP and RTP
multimedia are the real-time transport protocol (RTP) (see
                                                                  packet headers. This makes for a very inefficient use of
RFC 1889 [5]) and RTP control protocol (RTCP), which are
                                                                  bandwidth, especially in the costly radio access. Increasing
carried in UDP/IP packets. This is shown in Fig 5.
                                                                  the sample size reduces the problem, but increases the end-
    Currently, as RTP and RTCP do not support rate-               to-end delay of the speech packets, as well as increasing the
controlled codecs such as AMR, another framing protocol           likelihood of packet loss on the radio interface.
such as the Iu user plane protocol (IuUP) [2]], that is used to
frame the speech on the Iu interface in Release 1999, could           One solution to this is to perform header compression
be used instead of RTP and RTCP. However, for the                 between the user’s equipment and the UTRAN, where the
purposes of this paper, RTP and RTCP are used to illustrate       bandwidth is most expensive. This could theoretically bring
the framing and transport of speech and multimedia in the         the overhead of the IP, UDP and RTP headers down to less
IM domain.                                                        than 10% (not including the overhead of the lower layer
                                                                  GPRS and UTRAN protocols). Another possible solution is
    For the user’s equipment to be able to send and receive       to transport the speech from the user’s equipment through
speech packets to and from the IM domain, it must activate        the UTRAN using AAL2, and to packetise the speech into
a bi-directional PDP context between itself and the IM            IP payloads at the node B or RNC. However, both of these
domain. This allocates bandwidth and the required quality         solutions require that speech be transported differently to
of service over the UTRAN and GPRS network for the                other real-time and best-effort services that can be sent in
transport of speech packets. The entry point to the IM            uncompressed packets all the way to the user’s equipment.
domain will contain firewalls for security and prevention of      For example, media such as video have transport-quality
denial of service attacks, and these may also be controlled       requirements that are similar to voice, but the higher
dynamically by the call control, on a call-by-call basis, to      bandwidth nature means that the payload-to-header ratio is
prevent speech packets being sent or received before the          much greater, and hence less wasteful of bandwidth.
call is established. Deactivating the appropriate GPRS PDP
context disconnects the speech path between the user’s                As with the signalling path, a benefit of using GPRS to
equipment and the IM domain.                                      carry the speech paths is that the GPRS controls the                 55


                                                                                               BT Technol J Vol 19 No 1 January 2001
     VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

     handover of the speech paths as a user moves between the               user’s equipment is switched on. Once registered, users can
     radio cells. However, this does require that the GPRS                  make and receive IM domain calls until they deregister.
     handover procedures be enhanced to ensure that the quality             Before the registration procedure can take place, the user
     of service required for voice is met throughout the                    equipment has to connect to the network and discover an
     handover.                                                              entry point into the IM domain. This entry point is the proxy
                                                                            call state control function (P-CSCF), and it provides a
     3.4    Roaming, registration and discovery                             simple, generic call control function as well as potentially
                                                                            providing a SIP firewall to ensure security of the IM
         One of the main benefits of current GSM networks is                domain.
     the ability for the user to make and receive calls while
     travelling abroad. To provide such a benefit, the user must                The P-CSCF always resides in the network to which the
     have the capability to be able to connect to a network that is         UE is connected, and therefore the procedure for discovery
     controlled by an operator other than that to which they are            of the P-CSCF is the same, irrespective of whether the user
     subscribed. This benefit is also an essential feature of the           is roaming or not. Additionally, the P-CSCF could provide
     Release 5 standards, although additional procedures are                access to services that are not user specific but that are
     required to provide a roaming capability for the IM domain.            specific to the ‘roamed to’ network, such as emergency
     The reasoning behind this is the fact that the user’s voice            calls.
     service can be controlled by one of two methods — home or
     visited.
                                                                                The procedure for discovery relies on GPRS signalling
         In Lobley [6] it is shown that the call is controlled by an        with the use of the IETF dynamic host configuration
     entity known as a serving call state control function (S-              protocol (DHCP) (see RFC 2131 [5]) and domain name
     CSCF). The S-CSCF can be located either in the network                 system (DNS) (see RFC 1035 [5]) protocols. The idea of the
     owned by the operator to which the user is subscribed,                 procedure is for any UE to be able to attach to a GPRS
     known as home control, or alternatively in the network                 network, and be provided with an IP address of the P-CSCF.
     owned by another operator if the user has roamed to that               All SIP-based signalling from the UE then goes via the P-
     network, known as visited control. This is the main                    CSCF which is responsible for routeing the messages on to
     difference when roaming in an IM domain compared to                    the S-CSCF.
     today’s GSM network and Release 1999, where the visited
     network always controls a roaming user’s voice service.                   Figure 6 shows the sequence of events in the ‘discovery’
                                                                            procedure.
        In order that the user can make and receive calls, the
     user equipment (UE) has to be registered with an S-CSCF.                  The sequence of events that make up the discovery
     The registration procedure happens immediately after the               procedure is described below.

                 user                                                                            DHCP                       DNS
                                             SGSN                     GGSN
               equipment                                                                         server                    server

                      activate PDP context
                      activation                create PDP context
                1                               activation

                                                    create PDP context
                         activate PDP context                response
                                    response

                        DHCP DISCOVER
                2

                                                                                   DHCP OFFER

                        DHCP REQUEST
                3

                                                                                      DHCP ACK

                          QUERY
                4
                                                                                                          QUERY response


56                                                      Fig 6   Discovery message sequence.


     BT Technol J Vol 19 No 1 January 2001
                                                            VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

•   PDP context activation (1)                                               to the server(s). Each server checks the returned IP
                                                                             address. If it does not match, the server considers it as
    The UE activates a PDP context to the GPRS network,                      an implicit decline. However, the selected DHCP
    which will be used for the discovery procedures, and                     server sends a DHC PACK to the UE.
    later for the IM domain registration and call control
    procedures using SIP. To achieve this, the UE sends an             •     DNS query (4)
    activate PDP context activation request to the SGSN.
                                                                             The UE sends a DNS QUERY to the DNS server for
    Upon receipt of the request, the SGSN sends a create
                                                                             resolution of the predefined name for P-CSCFs to an IP
    PDP context activation request to the GGSN. If the
                                                                             address. The DNS server replies to the UE with a
    GGSN is able to establish a PDP context (e.g. after
                                                                             QUERY response containing the IP address of an
    checking that the UE has the necessary permission), it
                                                                             appropriate P-CSCF.
    creates a PDP context response to the SGSN, which in
    turn replies to the UE with an activate PDP context                    On disconnection of the UE, such as just before the
    response. This is a standard GPRS procedure, although              device is turned off, the IP address can be released back to
    the details, such as the GPRS address point name used              the DHCP server and the signalling PDP context can be
    and the nature of the PDP address returned, may be                 deactivated.
    specific to the discovery procedure.
                                                                           Now that the UE has knowledge of the proxy CSCF
•   DHCP discovery (2)                                                 address, the registration procedure can take place in order
                                                                       that an S-CSCF can be selected. Unlike the discovery
    The UE broadcasts a DHCP DISCOVER message to
                                                                       procedure, the registration procedure differs depending on
    the network. Upon receiving this message the DHCP
                                                                       whether the S-CSCF is to be located in the home network,
    Server can respond with a DCHP OFFER message or it
                                                                       or the visited network. However, the home network, the
    may not respond at all. If the DHCP server decides to
                                                                       network to which the user is subscribed, always carries out
    respond it broadcasts the DHCP OFFER message with
                                                                       the decision on whether home control or visited control is
    a specified available IP address. Note: at this stage
                                                                       used. Figure 7 shows the functional entities involved in
    there is no agreement of an assignment between the
                                                                       registration for visited network control, and Fig 8 shows the
    DHCP server and the UE. The UE may receive more
                                                                       message sequence required. The message sequences for
    than one DHCP OFFER response (if more than one
                                                                       home network control are the same except the visited I-
    DHCP server responds) and therefore will have to
                                                                       CSCF is not required since the S-CSCF is in the home
    choose one.
                                                                       network. If a user is connected to the home network rather
•   DHCP request (3)                                                   than a visited network, the visited I-CSCF is not required
                                                                       and the S-CSCF and P-CSCF will be in the home network.
    Using the IP address received within the DHCP
    OFFER response, the UE broadcasts a DHCP                             After the UE has obtained a signalling path through the
    REQUEST message containing the chosen IP address                   GPRS network, it can perform the IM registration.



                                                                                                          HSS

                                               visited                                                          home
                                             IM domain                                                        IM domain

                                                serving              interrogating                    interrogating
                                                 CSCF                    CSCF                             CSCF

                                                                                                        home core
                                                                                                        IP network
                                                 proxy
                                                 CSCF


                    user           GPRS                              visited core
                  equipment       network                            IP network

                                                  registration signalling (SIP)
                                                  mobility management signalling
                                                  discovery signalling (GPRS, DHCP, DNS)
                                     Fig 7    Functional entities for visited network registration.                                               57


                                                                                                          BT Technol J Vol 19 No 1 January 2001
     VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS


           user                                              visited                     visited                    home
                                  P-CSCF                                                                                                         HSS
         equipment                                          S-CSCF                      I-CSCF                     I-CSCF


               REGISTER
         1                             REGISTER
                                                                                                                         Cx-Query
                                                                                                                  2

                                                                                                                          Cx-Select-Pull
                                                                                                                  3

                                                                                                        REGISTER
                                                                                                                           4
                                                                             REGISTER
                                                                                               5
                                                                  Cx-Put
                                                            6

                                                                  Cx-Pull
                                                            7

                                                                   200 OK
                                                            8                                200 OK

                                                                                                            200 OK
                            200 OK
                                             9


                                                 Fig 8   Message sequences for visited network registration.


     Signalling based on SIP is used to perform the registration                       home network or the visited network (for example,
     between the UE and the CSCFs. The protocol between the                            based on the user’s service profile). The HSS then
     CSCFs and the HSS is as yet undefined, but is represented                         issues a response indicating the serving network
     in this paper by information flows prefixed by the letters Cx                     selection back to the home I-CSCF.
     (since this is the Cx reference point in the architecture). IM
     domain registration for visited network control requires the                •     Cx-Select-Pull (3)
     following steps.
                                                                                       At this stage, it is assumed that the authentication of
     •    Register (1)                                                                 the user has been completed (although it may have
                                                                                       been determined at an earlier point in the message
          The UE sends a REGISTER message to the P-CSCF.                               sequence). The home I-CSCF then sends a Cx-Select-
          This message contains the subscriber identity and the                        Pull to the HSS to request the information related to the
          domain name of the home network. Upon receipt of the                         S-CSCF capabilities required by the user. The HSS
          REGISTER, it examines the home domain name to                                responds with the necessary information on the
          discover the entry point to the home network. This                           required S-CSCF capabilities to the home I-CSCF.
          entry point is an interrogating CSCF (I-CSCF), which
          provides policing of the SIP interface to other networks               •     Home I-CSCF forwards message (4)
          and interrogation of the home subscriber server. The P-
          CSCF forwards the REGISTER message on to the I-                              The home I-CSCF determines the address of an I-
          CSCF in the home network, adding the name of the P-                          CSCF in the visited network from the visited network
          CSCF, a visited network contact point name, and the                          contact point name, and forwards the REGISTER
          visited network capabilities. A name-address                                 message on to the visited I-CSCF2.
          resolution mechanism is utilised in order to determine
          the address of the home network from the home                          •     Visited I-CSCF forwards message (5)
          domain name.

     •    Cx-Query(2)                                                                  The visited I-CSCF, using its role of S-CSCF selection,
                                                                                       determines the name and address of an appropriate S-
          When the I-CSCF receives the REGISTER message, it                            CSCF based on the required S-CSCF capabilities, and
          queries the HSS by sending a Cx-Query containing the                         forwards the REGISTER message on to it.
          parameters of the REGISTER message. The HSS
                                                                                 2
          checks whether the user is already registered, and if                    This step is not required if home network control is selected. Instead, the
                                                                                 functions performed by the visited I-CSCF are performed by the home
58        not, selects whether the serving CSCF is to be in the                  I-CSCF.


     BT Technol J Vol 19 No 1 January 2001
                                                            VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

•   S-CSCF contacts HSS using Cx-Put (6)                               multimedia calls with the IM Domain, irrespective of the
                                                                       location.
    On receiving the REGISTER, the S-CSCF associates
    the subscriber and the S-CSCF name in the HSS using                    A deregistration procedure is invoked by either the UE
    the Cx-Put, which is acknowledged by the HSS.                      or the network in order to remove the registration of the user
                                                                       on a S-CSCF, for example if a user roams to a different
•   S-CSCF retrieves profile using Cx-Pull (7)                         visited network or the user disconnects from the network.
    The S-CSCF then uses the Cx-Pull request/response to
                                                                       3.5      Control of voice (and multimedia) calls
    retrieve the subscriber’s profile for the user from the
    HSS, which it then stores locally. The S-CSCF also
                                                                           Once the user is registered with an S-CSCF, voice and
    stores the name of the P-CSCF.
                                                                       multimedia calls may be made to other users. The S-CSCF
•   Serving contact name determination (8)                             provides the main point of control of the call and any
                                                                       supplementary or advanced service features for that user.
    The S-CSCF then determines whether the serving                     SIP signalling between the user equipment and the S-CSCF
    contact name should be that of the S-CSCF or the                   is routed via a P-CSCF, which provides a (secure) entry
    visited I-CSCF. The S-CSCF then returns a 200 OK                   point to the IM domain and a point of flexibility for routeing
    message with this information to the visited I-CSCF.               SIP messages to home or visited network S-CSCFs.
    The visited I-CSCF forwards the 200 OK to the home
    I-CSCF, and then releases all knowledge of the                         Each user will be registered with an S-CSCF, so that a
    registration information for that user. Similarly, the             simple voice call between two users will usually require two
    home I-CSCF forwards the 200 OK to the P-CSCF,                     S-CSCFs to communicate (i.e. one for each user).
    and then releases all knowledge of the registration                Additionally, an I-CSCF is required in order to interrogate
    information for that user.                                         the HSS to find the S-CSCF on which the called user is
                                                                       registered. Figure 9 shows the main functional entities
•   Registration completion                                            involved in the control of voice calls between two mobile
                                                                       users on a Release 5 network. For simplicity, this scenario
    On receiving the 200 OK message, the P-CSCF stores                 assumes that both users are connected to, and registered on,
    the serving network contact name, before sending the               their home network (i.e. they are not roaming). However,
    200 OK to the UE and completing the registration                   the sequence of events is similar for roaming users, with
    procedure.                                                         home or visited control.

    The user is now registered with an S-CSCF in the                      An IM domain call comprises the following five distinct
visited network and is able to make and receive voice and              phases.




                                                                        HSS
                                                                        (B)
                                     IM domain                                                 IM domain
                                        (A)                                                       (B)

                                       serving                     interrogating            serving
                                      CSCF (A)                       CSCF (B)              CSCF (B)



                                       proxy                                                 proxy
                                      CSCF (A)                                              CSCF (B)


        user A           GPRS           core IP                      core IP                              GPRS
      equipment                                                                                                              user B
                      network (A)     network (A)                  network (B)                          network (B)
                                                                                                                           equipment

                                              speech path (IP media stream)
                                              call control signalling (SIP)
                                              location management signalling

                                    Fig 9   Functional entities in a Release 5 mobile-to-mobile call.
                                                                                                                                                 59


                                                                                                         BT Technol J Vol 19 No 1 January 2001
     VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

     •    Call invitation                                              •   PDP content activation
          The calling user invites the called user to participate in       The UE IP address for the IP speech paths may not be
          a call. This is supported by the SIP INVITE method               known until the GPRS PDP context for the speech path
          and the 100 trying provisional response.                         has been activated. In this case, without a resource
                                                                           reservation phase, the PDP context would have to be
     •    Resource reservation                                             activated prior to the INVITE, before the session
                                                                           description is sent.
          The resources (such as the GPRS and UTRAN
          bandwidth) are reserved so that early tones and              •   Tone/announcement provision
          announcements can be played, and that transport for
                                                                           Without early reservation of the speech path, it is not
          the speech path is available when the called user
                                                                           possible for the network or end equipment to provide
          answers. This is currently not supported by SIP,
                                                                           tones or announcements in the speech path back to user
          although the use of the 183 session progress
                                                                           A prior to the call being answered (such as ring tone or
          provisional response and 200 OK final response with a
                                                                           busy tone).
          new SIP method, COMET, has been proposed in the
          IETF and 3GPP.                                                  Figure 10 illustrates the SIP signalling flows for a
                                                                       simple mobile-to-mobile call with a resource reservation
     •    Call offering                                                phase, based on the scenario in Fig 9. It assumes that the
          The called user is alerted to the incoming call. Support     underlying GPRS and IP core network provides the
          for informing the calling user of this event is provided     necessary quality of service for the speech paths.
          by the SIP 180 ringing provisional response.                     The message sequences in Fig 10 are described below.
     •    Call connection                                              •   Invite (1)
          The called user answers, the speech path is connected            User A initiates the call by sending an INVITE
          and charging begins. This is supported in SIP by the             message to the P-CSCF, which contains the names
          200 OK final response and the ACK method.                        (SIP URLs) of the calling and called users. The session
                                                                           description part of the message includes the IP address
     •    Call termination                                                 of user A’s UE and a description of the speech path
          The call and speech path is cleared by one of the users.         (e.g. AMR coded speech using RTP, with the UDP port
          This is supported in SIP by the BYE method and the               number). This description may include options, such as
          200 OK final response.                                           a range of codecs that could be used. Additionally, the
                                                                           session description indicates that the reservation of the
         The addition of the resource reservation phase of the             speech path IP transport and quality of service is a
     call to the SIP protocol is necessary in the mobile                   mandatory pre-condition to ringing.
     environment for a number of reasons.                                  The P-CSCF confirms receipt of the INVITE by
                                                                           replying with a 100 trying message, and forwards the
     •    Path establishment prior to ringing
                                                                           INVITE on to the S-SCSF, adding the name of user
          The establishment of the PDP contexts for transport of           A’s S-CSCF to the message. This allows tracing of the
          the speech path should occur prior to the called user’s          signalling route back through the network.
          telephone ringing. While this may not need to be the             The S-CSCF confirms receipt of the INVITE by
          case for all multimedia services, there is a user                replying with a 100 trying message, and then invokes
          expectation that when a ringing telephone is answered,           any necessary service features for user A (for example,
          a speech path will be in place. Given the scarcity of            outgoing call-barring). The S-CSCF then determines a
          bandwidth on the radio interface, if reservation is not          SIP entry point for user B’s home network from the
          performed early, then in some cases a ringing                    SIP URL for user B, for example by performing a DNS
          telephone could be answered only for the users to find           query. The SIP entry point to user B’s home network
          that there is no speech path available.                          will usually be an I-CSCF. User A’s SCSF then sends
     •    Quality of service                                               the INVITE on to user B’s I-CSCF, adding the name of
                                                                           user A’s S-CSCF to the message.
          The service may require that the quality of service of
          the speech channels be established end-to-end, using a
                                                                       •   Cx-Query (2)
          protocol such as the IETF RSVP, or that the speech               On receiving the INVITE, the I-CSCF interrogates the
          paths need to be secure. If so, the procedures to reserve        HSS with a Cx-Query to determine the address of user
          the appropriate quality-of-service level or to implement         B’s S-CSCF. It also confirms receipt of the INVITE
          the security should occur prior to the call ringing and          from user A’s S-CSCF by replying with a 100 trying
60        alerting user B.                                                 message. Once the HSS responds with the address of


     BT Technol J Vol 19 No 1 January 2001
                                                                                               VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

                               user A             proxy             serving            interrogating                           serving             proxy              user B
                             equipment                             CSCF (A)              CSCF (B)            HSS (B)          CSCF (B)            CSCF (B)
                                                 CSCF (A)                                                                                                           equipment
                                    INVITE                                                        2
                              1                        INVITE
                                                                            INVITE
                                                                                                Cx-Query
    call invitation




                                          100 trying         100 trying
                                                                                  100 trying
                                                                                                INVITE
                                                                                                                                      INVITE
                                                                                                                                                        INVITE
                                                                                                                       100 trying                                             3
                                                                                                                                          100 trying
                                                                                                                                                           183 session
                                                                                                                                          183 session
                                                                                                                                                              progress
                                                                               183 session                             183 session           progress
                                       183 session       183 session
                                                                                  progress                                progress
                                          progress          progress
    resource reservation




                              4

                                  activate PDP                                                                                                                 activate PDP
                                     context                                                                                                                      context
                                    COMET              COMET                COMET               COMET
                                                                                                                                      COMET             COMET
                                                                                                                                                                 200 OK
                                                                                                                           200 OK              200 OK                         5
                                                                200 OK               200 OK
                                                                                                                                                              180 ringing
                                                                                                                        180 ringing       180 ringing
                                                                                 180 ringing                                                                                  6
offering




                                                            180 ringing
  call




                                        180 ringing
                                                                                                                                                                 200 OK
                                                                                                                           200 OK              200 OK                          7
                                                                200 OK               200 OK
                                            200 OK
     call connection




                                    ACK                ACK
                             8                                              ACK                 ACK
                                                                                                                                      ACK               ACK


                                                                               speech transmission (both way RTP media)

                                    BYE
                              9                        BYE                  BYE                 BYE
                                                                                                                                      BYE               BYE
    call termination




                                                                                                                                                                              10

                                 deactivate PDP                                                                                                               deactivate PDP
                                    context                                                                                                                      context
                                                                                                                                                                 200 OK
                                                                                                                           200 OK              200 OK
                                                                200 OK               200 OK
                                            200 OK


                                                               Fig 10     Signalling message sequences for a simple mobile-mobile call.

                           the S-CSCF on which user B is registered, the I-CSCF                              with a 183 session progress message, indicating in the
                           forwards the INVITE on to that S-CSCF, adding its                                 session description that it accepts the pre-condition,
                           name to the message.                                                              and requesting confirmation that user A’s UE has itself
                                                                                                             met the pre-condition. This message traverses the
                           User B’s S-CSCF receives the INVITE and invokes                                   signalling path via the CSCFs back to user A’s UE. In
                           any necessary service features for user B, before                                 the meantime, user B’s UE activates a GPRS PDP
                           forwarding the INVITE on to user B’s P-CSCF adding                                context for the user-plane speech path through to an IM
                           its name to the message. The S-CSCF confirms receipt                              domain IP entry point (e.g. a firewall that protects the
                           of the INVITE by replying to the I-CSCF with a 100                                IM domain IP core network).
                           trying message. The P-CSCF receives the INVITE and
                           forwards it on to user B’s UE. The P-CSCF confirms                            •   PDP context activation (4)
                           receipt of the INVITE by replying to the S-CSCF with
                                                                                                             User A’s UE receives the 183 session progress, and
                           a 100 trying message.
                                                                                                             activates a GPRS PDP context for the speech path
•                          Invite acceptance (3)                                                             through to the IM domain IP entry point. As required,
                                                                                                             the UE confirms that the speech path is reserved by
                           User B’s equipment accepts the call invitation, but                               sending a COMET message back to user B’s UE along
                           does not alert user B at this stage. Instead, it responds                         the signalling path, which also contains the address                  61


                                                                                                                                        BT Technol J Vol 19 No 1 January 2001
     VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

          details of the speech path on user A’s UE and can also     if necessary. The circuit-switched networks generally use
          confirm the agreed session description.                    the ITU-T SS7 integrated services user part (ISUP) to
     •    Acknowledgement of reserved speech paths (5)               control calls. Signalling gateways (SGW) in the IM domain
                                                                     map between the message transfer part levels of SS7 and the
          On receiving the COMET, user B’s UE now knows              SIP transport protocol (e.g. TCP/IP) used in the IM domain.
          that the necessary IP transport and quality of service
          for the speech paths has been reserved at both ends,           It is not possible to simply map ISUP signalling
          and the address to use for the speech path. It             messages into SIP messages, since the service context of the
          acknowledges the COMET with a 200 OK, which can            messages must be known. A media gateway control
          contain the the address details of the speech path on      function (MGCF) is used to perform the mapping of the IM
          user B’s equipment.                                        domain voice service (and SIP signalling) to the voice
     •    User B alert (6)                                           service of the other network (e.g. PSTN voice service and
                                                                     ISUP signalling). The MGCF communicates with the S-
          User B’s equipment alerts user B, for example by           CSCF or I-CSCF using SIP. The MGCF also controls the
          ringing. It indicates this back to user B’s S-CSCF using   MGW, for example using the H.248/Megaco protocol,
          a 180 ringing message, which is sent back via the          jointly developed by the IETF and ITU-T.
          signalling path to user A’s UE. User A’s UE will then
          provide an indication of this back to user A, such as a
                                                                         Interworking with other VoIP networks that are not
          locally generated ringing tone.
                                                                     compatible with 3GPP Release 5, such as those based on
     •    User B answer (7)                                          ITU-T Recommendation H.323, also requires signalling and
          User B answers the call. User B’s UE sends a 200 OK        media gateways in order to map any differences in lower
          message via the signalling path to user A’s UE. If not     layer protocols and police the IM domain. An MGCF is also
          already sent, this message will contain the address        needed to ensure appropriate mapping of the voice service
          details of the speech path on user B’s equipment.          between the networks.

     •    User A acknowledgement (8)                                     Figure 11 shows the functional entities involved in
          User A’s UE acknowledges the establishment of the          interworking 3GPP Release 5 voice calls with PSTN or
          call by sending an ACK, which traverses the signalling     GSM networks.
          path back to user B’s UE. The UEs are now able to
          send IP speech packets to each other. It is likely that       Figure 12 illustrates the message sequences for a simple
          the P-CSCFs will have some control over the IM             mobile originated voice call that terminates in the PSTN
          domain speech-path entry points (firewalls), and not       network. The message sequences are described below.
          permit the speech packets through until this stage, or
          on receipt of the prior 200 OK (the choice may be          •   Invite (1)
          service dependent). This control could also be the point
                                                                         The UE of the mobile calling party, initiates the call to
          at which the call charging commences.
                                                                         the PSTN user by sending an INVITE to their P-CSCF,
     •    Call release (9)                                               which contains an appropriate session description. The
          To release the call, user A’s UE sends a BYE message           PSTN user is identified as such by the SIP URL, which
          to user B’s UE via the signalling path, and deactivates        contains the PSTN telephone number encoded into the
          its PDP context for the speech path. At this point, the        SIP URL format. Additionally, the session description
          P-CSCF may close the IM domain speech-path entry               indicates that the reservation of the speech path IP
          point to further traffic and cease charging.                   transport and quality of service is a mandatory pre-
                                                                         condition to ringing.
     •    Deactivation (10)
                                                                         The P-CSCF confirms receipt of the INVITE by
          User B’s UE responds by deactivating its PDP context
                                                                         replying with a 100 trying message, and forwards the
          for the speech path and acknowledging the BYE with a
                                                                         INVITE on to the S-SCSF, adding the name of user
          200 OK. This traverses the signalling path back to user
                                                                         A’s S-CSCF to the message. This allows tracing of the
          A’s UE, releasing each of the CSCFs from the call.
                                                                         signalling route back through the network.
     4.   Interworking 3GPP Release 5 with other networks
                                                                         The S-CSCF confirms receipt of the INVITE by

     I  nterworking with circuit-switched networks, such as the
        PSTN, GSM and 3GPP Release 1999 networks, require
     interworking at both the speech path level and the signalling
                                                                         replying with a 100 trying message, and then invokes
                                                                         any necessary service features for user A (for example,
                                                                         outgoing call-barring). The S-CSCF then determines
     level. Media gateways (MGWs) are included in the IM                 that the call is destined for the PSTN, and routes the
     domain to interface terminate the IP transported speech             INVITE to an appropriate MGCF, adding the name of
62   paths and convert this to circuit-switched TDM, transcoding         user A’s S-CSCF to the message.


     BT Technol J Vol 19 No 1 January 2001
                                                                     VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

                                                                                                    IM domain

                                                     serving                 media gateway              signalling
                                                    CSCF (A)                   controller               gateway

                                                                                                                             PSTN
                                                                                                                              or
                                                      proxy                                                                  GSM
                                                     CSCF (A)
                                                                                                          media
                                                                                                         gateway
               user                 GPRS                                     core IP
             equipment             network                                   network


                           speech path (IP media stream)                                   call control signalling (SIP)

                           speech path (circuit-switched TDM)                              call control signalling (ISUP)
                                                                                           media G/W control signalling (H.248/Megaco)

                                         Fig 11   Functional entities in a Release 5 mobile-to-PSTN or GSM call.


•    MGW configuration (2)                                                             so that the backward speech path from the PSTN to the
                                                                                       UE is switched through so that the mobile user can hear
     The MGCF initially responds to the S-CSCF with a
                                                                                       tones and announcements from the PSTN. It may also
     100 trying. It then configures the MGW for the speech
                                                                                       select the chosen codec if the codecs have been
     path3 (for example using the H.248/Megaco protocol),
                                                                                       negotiated. The MGCF then acknowledges the
     by seizing an already created circuit-switched trunk
                                                                                       COMET with a 200 OK, which can contain the address
     termination on the PSTN side of the MGW, and adding
                                                                                       details of the IP termination on the MGW.
     a new IP speech-path (e.g. RTP) termination to the IM
     domain side of the MGW. This is done by the ‘add’                                 The MGCF now initiates the call establishment to the
     command, which additionally creates a new context in                              PSTN by sending an initial address message (IAM) to
     the MGW, and associates the IP termination and PSTN                               the signalling gateway (SGW). The SGW relays the
     termination. The PSTN termination is configured for                               IAM from the IP-based transport protocol (for example
     both-way speech. The MGW returns a description of                                 SCTP/UDP/IP) to the SS7 message transfer part, and
     the ports to the MGCF in response.                                                on to the PSTN entry point (for example a PSTN
                                                                                       gateway trunk exchange).
     The MGCF, knowing the description of the IP speech-
     path port and its capabilities, sends a 183 session                        •      PSTN call acceptance (5)
     progress message back to the UE, via the signalling
     path. This indicates that the precondition can be met by                          The PSTN accepts the call with an address complete
     the MGW, and that confirmation that the UE can meet                               message (ACM), which is sent back to the MGCF via
     the precondition is required.                                                     the SGW. So that the mobile user may now hear any
                                                                                       in-band tones and announcements from the PSTN, the
•    PDP context activation (3)                                                        MGCF sends a 183 session progress message back to
     The UE receives the 183 session progress, and                                     the UE. This contains a session description indicating
     activates a GPRS PDP context for the speech path                                  that one-way IP speech packets may be received and
     through to the IM domain entry point. As required, the                            the address of the RTP termination on the MGW, if not
     UE confirms that the speech path is reserved by                                   already sent. This message follows the signalling path,
     sending a COMET message back to MGCF along the                                    and may cause the P-CSCF to control the IM domain
     signalling path, which also contains the address details                          IP speech-path entry point (e.g. firewall) from the
     of the speech path on the UE.                                                     GPRS network to allow the media to be played to the
                                                                                       UE.
•    MGW 1-way connection (4)
                                                                                •      PSTN alerting (6)
     On receiving the COMET, the MGCF now knows both
     that the necessary IP transport and quality of service                            The PSTN sends a call progress message (CPG) to the
     for the speech paths has been reserved at both ends,                              SGW, indicating that the called user’s telephone is
     and the address to use for the speech path. It then                               ringing. This is accompanied by in-band ring tone in
     modifies the IP speech-path termination on the MGW                                the speech path back to user A. The SGW relays this
3                                                                                      message back to the MGCF. The MGCF sends a 180
 It is assumed that the MGW has already established a control relationship
with the MGCF and the terminations on the TDM circuit-switched side                    ringing message to the S-CSCF, which is forwarded
have already been provisioned and configured.                                          via the signalling path to the UE.                               63


                                                                                                                BT Technol J Vol 19 No 1 January 2001
     VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

                                   mobile             proxy              serving         media G/W              media          signalling          PSTN
                                  terminal            CSCF                CSCF            controller            G/W               G/W            entry point
                                         INVITE
                                  1                        INVITE
                                                                              INVITE
           call invitation


                                              100 trying         100 trying
                                                                                    100 trying     2
                                                                                                 Add
                                                              183 session          183 session
                                             183 session
                                                                 progress             progress
                                                progress
                                  3

                                       activate PDP
                                          context
           resource reservation




                                         COMET
                                                           COMET                                   4
                                                                              COMET




                                                                                                                                                               call invitation
                                                                                                 modify

                                                                    200 OK             200 OK
                                                  200 OK
                                                                                                 IAM
                                                                                                                                     IAM


                                                              183 session          183 session                                                  ACM




                                                                                                                                                               reservation
                                            183 session                               progress                                ACM                        5




                                                                                                                                                                resource
                                                                 progress
                                               progress

                                                                                                                                                 CPG
                                                                                   180 ringing                                CPG                       6
                                             180 ringing       180 ringing
           call offering




                                                                                                                                                               call offering
                                                in-band ringing tone (one-way IP media)                            in-band ringing tone

                                                                                                                                                ANM
                                                                                                                              ANM                        7
                                                                                           8
                                                                                                 modify
                                                                                       200 OK




                                                                                                                                                               call connection
                                                                    200 OK
           call connection




                                                  200 OK

                                         ACK
                                  9                        ACK
                                                                              ACK

                                             speech transmission (both way IP media)              speech transmission (both way circuit-switched TDM)

                                         BYE
                                  10                       BYE
                                                                              BYE
           call termination




                                                                                                                                                               call termination
                                                                                                  11
                                                                                                 REL
                                                                                       200 OK                                        REL
                                                  200 OK            200 OK
                                                                                                 subtract

                                                                                                                                                 RLC
                                                                                                                              RLC                       12


                                                              Fig 12   Signalling message sequences for a simple mobile-PSTN call.

     •    PSTN answer (7)                                                                                   message back to the UE, with a session description
          When the call is answered, the PSTN sends an answer                                               indication that two-way media may be sent and
          message (ANM) to the SGW, which relays it back to                                                 received. This message follows the signalling path, and
          the MGCF.                                                                                         may cause the P-CSCF to control the IM domain IP
                                                                                                            speech-path entry point to allow both-way media. This
     •    MGW 2-way connection (8)                                                                          is the point at which call charging commences.
          At this point, the MGCF issues another modify                                            •        Acknowledgement of call establishment (9)
          command to change the IP speech-path termination in
          the MGW to allow both-way speech paths to be                                                      The UE receives the 200 OK. The call is now
64        switched through. The MGCF then sends a 200 OK                                                    established and two-way speech can take place


     BT Technol J Vol 19 No 1 January 2001
                                                       VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS

     between the mobile and PSTN user. The UE                  MSC call server by an appropriate protocol such as the
     acknowledges this by sending an ACK message back          H.248/Megaco protocol.
     to the MGCF via the signalling path.

•    PDP context deactivation (10)                                 The signalling from the GMSC call server to other
                                                               networks (including Release 5 CSCFs) is via a signalling
     When the mobile user clears the call, the UE sends a      gateway. The speech paths are interconnected to other
     BYE to the P-CSCF and deactivates the PDP context         circuit-switched and other VoIP networks via a media
     for the speech path. The P-CSCF forwards the BYE on       gateway. The operation of these gateways is similar to the
     to the MGCF via the S-CSCF.                               PSTN interconnect case for the Release 5 voice service.
•    Call release (11)
                                                                   Additionally, interconnect of the speech paths to
     The MGCF releases the call into the PSTN with a           Release 5 networks may not require a media gateway if the
     release message (REL) and confirms the BYE by             speech paths are compatible, although additional security
     sending a 200 OK message back to the UE via the           measures (such as firewalls) will be required in the case of
     CSCFs on the signalling path, which each release the      interconnect to other operators.
     call in turn.
                                                                  The MSC call server supports the Release 1999 call
     The MGCF then clears the speech path in the MGW by        control, service features and mobility management of an
     issuing a subtract command to delete the terminations     MSC, while the GMSC call server performs the call control
     from the call context, and the call context itself. The   and HSS interrogation of a Release 1999 GMSC — both,
     MGW optionally responds by sending an audit report        however, using media gateways to perform the circuit-
     for the call to the MGCF that contains information such   switching functions, with IP providing the core transport
     as the number of packets sent/received and the packet     network.
     loss.

•    Release confirmation (12)                                    Using this design, the Release 4 networks are capable of
                                                               supporting the Release 1999 voice service with minimal
     The release message sent to the PSTN is confirmed         enhancement to the network and little, if any, impact on the
     back to the MGCF (via the SGW) by a release               end user.
     complete message (RLC), which completes the release
     procedure.                                                6.   Conclusions

5.   Voice and multimedia in the 3GPP Release 4
     network                                                   V     oice telephony is an essential service for many mobile
                                                                     network users, and one that must be supported by 3rd
                                                               generation networks. This paper has shown how the initial

T  he 3GPP Release 4 standards provide a means for oper-
   ators to migrate the Release 1999 circuit-switched
domain to an IP-based core network infrastructure.
                                                               3GPP UMTS standards have taken an evolutionary approach
                                                               to providing a voice service compatible with GSM to max-
                                                               imise the benefits of the new radio access technologies. It has
                                                               then described how a more innovative approach to providing
    An overview of the Release 4 network is shown in Fig       voice and multimedia integration with the Internet protocols
13. The MSC call server concept is more fully described in     is being developed for the Release 4 and 5 standards.
Lobley [6].
                                                                   As a founder member of the 3G.IP group [9], BT has
    In this case, the Release 4 circuit-switched domain
                                                               played an influential role in moving the mobile network
connects to the UTRAN via the Iu-CS interface, which
                                                               standards towards an Internet solution for voice and
supports the same speech transport and signalling protocols
                                                               multimedia.
as in Release 1999. GSM radio networks can also be
connected via the GSM A interface. The Iu-CS interface
(and similarly the GSM A interface) is terminated at the          BT continues to make an active contribution to 3G.IP
circuit-switched domain entry point by a media gateway.        and the 3GPP standards needed to realise a mobile voice
This relays the signalling path from the ATM transport on      and multimedia mobile solution on a global scale.
to IP transport (such as TCP or SCTP) and on to the MSC
call server. Speech paths in the Iu-CS interface are relayed   Acknowledgements
into the IP core network from ATM AAL2 transport on to
UDP/IP transport. For the speech circuits, the media
gateway operates in a similar way to the one used for PSTN     T    he author would like to thank his colleagues in the
                                                                    BTexaCT 3G Networks unit who provided information
                                                               for, and valuable discussion on, this paper.
interconnect (in Fig 2 and Fig 11), and is controlled by the                                                                      65


                                                                                          BT Technol J Vol 19 No 1 January 2001
     VOICE AND INTERNET MULTIMEDIA IN UMTS NETWORKS


                             GSM radio
                           access network


                                                                                    application                                                            A
                                                                EIR                                           HSS                    signalling
                                                                                   and service
                                                                                                                                     gateway
                                                                                   environment



                                          BSS

                                                                                               VLR            GMSC
                                                                                                               call                                        B
                                                                                                              server                 signalling
                                                                                    MSC call server
                                                                                                                                     gateway




                                                            GSM A
                                                           interface
                                          RNC
                                                                                                                                                           C
                                                                               media                                                  media
                                                                              gateway                                                gateway


                                                                                                                                                           D
                                                            UMTS lu-CS
                                          RNC
                                                             interface


                       UMTS terrestrial radio                                                     circuit-switched domain
                      access network (UTRAN)                                                         (IPv6 core network)


                                        signalling         A      mobility management signalling to other networks

                                        speech paths        B     call-related signalling to other networks

                                                           C      circuit-switched speech circuits to other networks (e.g. PSTN and GSM)

                                                           D      speech paths to Release 5 and other VoIP networks


                                                            Fig 13     3GPP Release 4 network overview.



     References                                                                      8   European Telecommunications Standards Institute project — Tiphon
                                                                                         http://www.etsi.org/tiphon

     1   Mehrotra A: ‘GSM System Engineering’, Artech House (1997).                  9   3G.IP — http://www.3gip.org

     2   3GPP — http://www.3gpp.org
                                                                                                                    Mel Bale is a senior technical consultant on
                                                                                                                    3rd generation networks in BTexaCT. He
     3   International Telecommunication Union — Telecommunication
                                                                                                                    joined BT in 1987 after graduating from the
         Standardization Sector — http://www.itu.int
                                                                                                                    University of East Anglia, initially leading a
                                                                                                                    number of Unix software developments for
     4   Harris J W: ‘The future of radio access in 3G’, BT Technol J, 19, No 1,                                    network test systems. In 1993, he moved into
         pp 106—113 (January 2001).                                                                                 the field of intelligent networks, where he
                                                                                                                    managed a team of voice network designers
     5   Internet Engineering Task Force — http://www.ietf.org                                                      and had responsibility for defining service
                                                                                                                    architectures for future networks, including
                                                                                                                    the Parlay API.
     6   Lobley N C: ‘GSM to UMTS : Architecture evolution to support multi-
         media’, BT Technol J, 19, No 1, pp 38—47 (January 2001).                                                He is currently leading teams designing fixed
                                                                                                                 and mobile VoIP networks and researching
     7   Cookson M D: ‘3G service control’, BT Technol J, 19, No 1, pp 67—           future IP mobile network technologies. He is a Chartered Engineer and a
66       79 (January 2001).                                                          Member of the IEE.



     BT Technol J Vol 19 No 1 January 2001

								
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