umts services

2UMTS Services and ApplicationsJouni Salonen, Antti Toskala and Harri Holma

2.1 Introduction

The best known new feature of UMTS is higher user bit rates: on circuit-switched con-nections 384 kbps, and on packet-switched connections up to 2 Mbps, can be reached.Higher bit rates naturally facilitate some new services, such as video telephony and quickdownloading of data. If there is to be a killer application, it is most likely to be quickaccess to information and its filtering appropriate to the location of a user: see Figure 2.1.Often the requested information is on the Internet, which calls for effective handling ofTCP/UDP/IP traffic in the UMTS network. At the start of the UMTS era almost all trafficwill be voice, but later the share of data will increase. It is, however, difficult to predictthe pace at which the share of data will start to dominate the overall traffic volume. Atthe same time that transition from voice to data occurs, traffic will move from circuit-switched connections to packet-switched connections. At the start of UMTS service notall of the Quality of Service (QoS) functions will be implemented, and therefore delay-critical applications such as speech and video telephony will be carried on circuit-switchedbearers. Later, it will be possible to support delay-critical services as packet data withQoS functions.

Compared to GSM and other existing mobile networks, UMTS provides a new andimportant feature, namely it allows negotiation of the properties of a radio bearer.Attributes that define the characteristics of the transfer may include throughput, transferdelay and data error rate. To be a successful system, UMTS has to support a wide rangeof applications that possess different quality of service (QoS) requirements. At present itis not possible to predict the nature and usage of many of these applications. Therefore itis neither possible nor sensible to optimise UMTS to only one set of applications. UMTSbearers have to be generic by nature, to allow good support for existing applicationsand to facilitate the evolution of new applications. Since most of the telecommunicationsapplications today are Internet or N-ISDN applications, it is natural that these applicationsand services dictate primarily the procedures for bearer handling.

WCDMA for UMTS, edited by Harri Holma and Antti Toskala 2002 John Wiley & Sons, Ltd

12 WCDMA for UMTS

Figure 2.1. One of the concept phones for UMTS

2.2 UMTS Bearer Service

UMTS allows a user/application to negotiate bearer characteristics that are most appro-priate for carrying information. It is also possible to change bearer properties via a bearerrenegotiation procedure in the course of an active connection. Bearer negotiation is initi-ated by an application, while renegotiation may be initiated either by the application or bythe network (e.g. in handover situations). An application-initiated negotiation is basicallysimilar to a negotiation that occurs in the bearer establishment phase: the applicationrequests a bearer depending on its needs, and the network checks the available resourcesand the user’s subscription and then responds. The user either accepts or rejects the offer.The properties of a bearer affect directly the price of the service.

The bearer class, bearer parameters and parameter values are directly related to anapplication as well as to the networks that lie between the sender and the receiver. Theset of parameters should be selected so that negotiation and renegotiation proceduresare simple and unambiguous. In addition, parameters should allow easy policing andmonitoring. The format and semantics will take into account the existing reservationprotocols such as RSVP and those used in GPRS. Furthermore, the QoS concept shouldbe flexible and versatile enough to allow bearer negotiation in the future with as yetunknown applications.

The layered architecture of a UMTS bearer service is depicted in Figure 2.2; eachbearer service on a specific layer offers its individual services using those provided by

UMTS Services and Applications 13

TE UTRANMT TECN

GatewayCN IuEDGENODE

End-to-End Service

TE/MT LocalBearer Service UMTS Bearer Service

External BearerService

Radio Access BearerService

CN BearerService

Iu BearerService

BackboneNetwork Service

Radio BearerService

UTRAFDD/TDDService

Physical BearerService

UMTS

Figure 2.2. Architecture of a UMTS bearer service

the layers below. As can be seen from the figure, the UMTS bearer service plays a majorrole in end-to-end service provisioning [1].

2.3 UMTS QoS Classes

In general, applications and services can be divided into different groups, depending onhow they are considered. Like new packet-switched protocols, UMTS attempts to fulfilQoS requests from the application or the user. In UMTS four traffic classes have beenidentified:

— conversational,— streaming,— interactive, and— background classes.

The main distinguishing factor between these classes is how delay-sensitive the trafficis: the conversational class is meant for very delay-sensitive traffic, while the backgroundclass is the most delay-insensitive. The UMTS QoS classes are summarised in Table 2.1.

The conversational and streaming classes are typically transmitted as real-time connec-tions over the WCDMA Release’99 air interface, while the interactive and backgroundclasses are transmitted as non-real-time packet data using packet scheduling. The packetscheduling in WCDMA is described in detail in Chapter 10.

14 WCDMA for UMTS

Table 2.1. UMTS QoS classes

Traffic class Conversationalclass

Streaming class Interactive class Background

Fundamentalcharacteristics

Preserve timerelation (variation)betweeninformationentities of thestream

Conversationalpattern (stringentand low delay)

Preserve timerelation (variation)betweeninformationentities of thestream

Request responsepattern

Preserve dataintegrity

Destination is notexpecting the datawithin a certaintime

Preserve dataintegrity

Example oftheapplication

Voice,videotelephony,video games

Streamingmultimedia

Web browsing,network games

Backgrounddownload ofemails

2.3.1 Conversational Class

The best-known application of this class is speech service over circuit-switched bearers.With Internet and multimedia, a number of new applications will require this type, forexample voice over IP and video telephony. Real-time conversation is always performedbetween peers (or groups) of live (human) end-users. This is the only type of the fourwhere the required characteristics are strictly imposed by human perception.

Real-time conversation is characterised by the fact that the end-to-end delay is low andthe traffic is symmetric or nearly symmetric. The maximum end-to-end delay is given bythe human perception of video and audio conversation: subjective evaluations have shownthat the end-to-end delay has to be less than 400 ms. Therefore the limit for acceptabledelay is strict, as failure to provide sufficiently low delay will result in unacceptablequality.

2.3.1.1 AMR Speech ServiceThe speech codec in UMTS will employ the Adaptive Multi-rate (AMR) technique. Themulti-rate speech coder is a single integrated speech codec with eight source rates: 12.2(GSM-EFR), 10.2, 7.95, 7.40 (IS-641), 6.70 (PDC-EFR), 5.90, 5.15 and 4.75 kbps. TheAMR bit rates can be controlled by the radio access network. To facilitate interoperabilitywith existing cellular networks, some of the modes are the same as in existing cellularnetworks. The 12.2 kbps AMR speech codec is equal to the GSM EFR codec, 7.4 kbps isequal to the US-TDMA speech codec, and 6.7 kbps is equal to the Japanese PDC codec.The AMR speech coder is capable of switching its bit rate every 20 ms speech frameupon command. For AMR mode switching in-band signalling is used.

The AMR coder operates on speech frames of 20 ms corresponding to 160 samples atthe sampling frequency of 8000 samples per second. The coding scheme for the multi-ratecoding modes is the so-called Algebraic Code Excited Linear Prediction Coder (ACELP).The multi-rate ACELP coder is referred to as MR-ACELP. Every 160 speech samples,the speech signal is analysed to extract the parameters of the CELP model (LP filtercoefficients, adaptive and fixed codebooks’ indices and gains). The speech parameter bits


delivered by the speech encoder are rearranged according to their subjective importancebefore they are sent to the network. The rearranged bits are further sorted based on theirsensitivity to errors and are divided into three classes of importance: A, B and C. ClassA is the most sensitive, and the strongest channel coding is used for class A bits in theair interface.

During a normal telephone conversation, the participants alternate so that, on the aver-age, each direction of transmission is occupied about 50% of the time. The AMR hasthree basic functions to utilise effectively discontinuous activity:

• Voice Activity Detector (VAD) on the TX side.

• Evaluation of the background acoustic noise on the TX side, in order to transmitcharacteristic parameters to the RX side.

• The transmission of comfort noise information to the RX side is achieved by meansof a Silence Descriptor (SID) frame, which is sent at regular intervals.

• Generation of comfort noise on the RX side during periods when no normal speechframes are received.

DTX has some obvious positive implications: in the user terminal battery life willbe prolonged or a smaller battery could be used for a given operational duration. Fromthe network point of view, the average required bit rate is reduced, leading to a lowerinterference level and hence increased capacity.

The AMR specification also contains error concealment. The purpose of frame sub-stitution is to conceal the effect of lost AMR speech frames. The purpose of muting theoutput in the case of several lost frames is to indicate the breakdown of the channel tothe user and to avoid generating possibly annoying sounds as a result of the frame sub-stitution procedure [2] [3]. The AMR speech codec can tolerate about a 1% frame errorrate (FER) of class A bits without any deterioration of speech quality. For class B and Cbits a higher FER is allowed. The corresponding bit error rate (BER) of class A bits willbe about 10−4.

The bit rate of the AMR speech connection can be controlled by the radio accessnetwork depending on the air interface loading and the quality of the speech connections.During high loading, such as during busy hours, it is possible to use lower AMR bit ratesto offer higher capacity while providing slightly lower speech quality. Also, if the mobileis running out of the cell coverage area and using its maximum transmission power,a lower AMR bit rate can be used to extend the cell coverage area. The capacity andcoverage of the AMR speech codec is discussed in Chapter 12. With the AMR speechcodec it is possible to achieve a trade-off between the network’s capacity, coverage andspeech quality according to the operator’s requirements.

Release 5 contains enhancement of the AMR technology, the Adaptive Multi-RateWideband (AMR-WB) speech codec. The term wideband comes from the sampling rate,which has been increased from 8 kHz to 16 kHz, thus resulting to the 14 bit sampleswith 16000 samples/s sampling rate. This allows covering twice the audio bandwidthcompared to the classical telephone voice bandwidth of 4 kHz. The end result is a clearimprovement in voice and audio quality. The data rates range from 23.85 kbps down

16 WCDMA for UMTS

to 6.6 kbps. Further details of the AMR-WB can be found from [4]. In case of packetswitched streaming AMR-WB is part of Release 4 already.

2.3.1.2 Video TelephonyVideo telephony has similar delay requirements to speech service. Due to the nature ofvideo compression, the BER requirement is more stringent than that of speech. UMTShas specified that ITU-T Rec. H.324M should be used for video telephony in circuit-switched connections and Session Initiation Protocol (SIP) for supporting IP multimediaapplications, including video telephony.

Multimedia Architecture for Circuit Switched ConnectionsOriginally Rec. H.324 was intended for multimedia communication over a fixed telephonenetwork, i.e. PSTN. It is specified that for PSTN connections, a synchronous V.34 modemis used. Later on, when wireless networks evolved, mobile extensions were added to thespecification to make the system more robust against transmission errors. The overallpicture of the H.324 system is shown in Figure 2.3 [5].

H.324 consists of the following mandatory elements: H.223 for multiplexing and H.245for control. Elements that are optional but are typically employed are H.263 video codec,G.723.1 speech codec, and V.8bis. Later, MPEG-4 video and AMR were added as optionalcodecs into the system. The recommendation defines the seven phases of a call: set-up,speech only, modem training, initialisation, message, end, and clearing. Level 0 of H.223multiplexing is exactly the same as that of H.324, thus providing backward compatibility

Video I/Oequipment

Audio I/Oequipment

User dataapplication T.120

etc.

System controlUser

interface

Video codecH.263 or MPEG-4

Simple Profile

Speech codecG.723.1or AMR

Data protocolsV.14, LAPM, etc.

System control

H.245 control

SRP/LAPMprocedures

H.223Multiplexing/

demulti-plexing

Level 0Level 1Level 2

Scope of Rec. H.324 M

Modem V.34V.8/V.8bisfor PSTN

Correspondinginterface for

wirelessnetwork

PSTN,Wireless circuit

switchednetwork (GSM,WCDMA, ...)

ModemcontrolV.25ter

Receive pathdelay

Figure 2.3. Scope of ITU Rec. H.324


with older H.324 terminals. With a standardised negotiation procedure the terminal canadapt to the prevailing radio link conditions by selecting the appropriate error resiliencylevel.

V.8bis contains procedures for the identification and selection of common modes ofoperation between data circuit-terminating equipment (DCE) and between data termi-nal equipment (DTE) over general switched telephone network and leased point-to-pointtelephone types. The basic features of V.8bis are as follows:

• It allows a desired communication mode to be selected by either the calling or theanswering station.

• It allows terminals to automatically identify common operating modes (applications).

• It enables automatic selection between multiple terminals that share a common tele-phone circuit.

• It provides user-friendly switching from normal voice telephony to a modem-basedcommunication mode.

The capabilities exchange feature of V.8bis permits a list of communication modes, as wellas software applications, to be exchanged between terminals. Each terminal is thereforeable to establish the modes of operation it shares with the remote station. A capabilityexchange between stations thus ensures, a priori, that a selected communication mode ispossible. Attempts to establish incompatible modes of operation are thus avoided, whichspeeds up the application level connection.

As with the mode selection procedure, a capabilities exchange may be performed eitherat call set-up, automatically under the control of either the calling or the answering station,or during the course of telephony. In the latter case, on completion of the informationexchange, the communication link may be configured either to return to voice telephonymode or to adopt immediately one of the common modes of communication.

V.8bis has been designed so that, when a capabilities exchange takes place in telephonymode, and the capabilities exchanged are limited to standard features, the interruption invoice communications is short (less than approximately 2 seconds) and as unobtrusive aspossible.

In order to guarantee seamless data services between UMTS and PSTN, the call controlmechanism of UMTS should take the V.8bis messages into account. V.8bis messagesshould be interpreted and converted into UMTS messages and vice versa.

One of the recent developments of H.324 is an operating mode that makes it possibleto use an H.324 terminal over ISDN links. This mode of operation is defined in Annex Dof the H.324 recommendation and is also referred to as H.324/I. H.324/I terminals use theI.400 series ISDN user-network interface in place of the V.34 modem. The output of theH.223 multiplex is applied directly to each bit of the digital channel, in the order definedby H.223. Operating modes are defined bit rates ranging from 56 kbps to 1920 kbps, sothat H.324/I allows the use of several 56 or 64 kbps links at the same time.

H.324/I provides direct interoperability with H.320 terminals, H.324 terminals on theGSTN (using GSTN modems), H.324 terminals operating on ISDN through user substitu-tion of I.400 series ISDN interfaces for V.34 modems, and voice telephones (both GSTNand ISDN). H.324/I terminals support H.324/Annex F (= V.140) which is for establishing

18 WCDMA for UMTS

communication between two multiprotocol audio-visual terminals using digital channelsat a multiple of 64 or 56 kbps [6].

Multimedia Architecture for Packet Switched ConnectionsIn the beginning of 3GPP standardization there were two competing standard proposalsfor IP multimedia: ITU-T H.323 and IETF’s SIP. After thorough evaluation, SIP wasselected and forms now basis for IP multimedia signalling in UMTS. SIP is part of IETFMultimedia Architecture (see Figure 2.4), which covers several areas:

• SIP (Session Initiation Protocol): Signalling protocol to be used instead ofH.323/H.245.

• SAP (Session Announcement Protocol): Multicast announcement protocol (advertisesInternet A/V sessions such as pop concerts, lectures, etc.). Current MBone is basedon this.

• SDP (Session Description Protocol): Text-based syntax to describe sessions (replacesASN.1/BER in H.323).

• RTSP (Real Time Streaming Protocol): Protocol for controlling remote servers (e.g.VOD servers to play a file).

RTP is used for media encapsulation and RTCP for control information delivery and lipsync purposes. Multiparty application sharing is also possible: at least shared workspace

Video I/Oequipment

Audio I/Oequipment

User dataapplication T.120

etc.

System controlUser interface

Video codecH.261, H.263,

H.263+, MobiVideo

Audio codecG.711, G.722,

G.723.1,G.728, G.729, EFR

Security (optional)

SIP

SAP

RTCP (optional)

RSVP (optional)

SDP

RTP-capsulation,receive path delay

RTCP senderreports (optional)

RTP-capsulation,receive path delay

RTCP senderreports (optional)

TCP/IP driver Packetnetwork

Scope of IETFMultimedia terminal

Figure 2.4. IETF Multimedia architecture


and a network text editor have been developed. Both approaches are based on reliablemulticast.

SIP provides the necessary protocol mechanisms so that end systems and proxy serverscan provide services:

— call forwarding, call-forwarding no answer;

— call-forwarding busy;

— call-forwarding unconditional;

— other address translation services;

— callee and calling number delivery, where numbers can be any (preferably a unique)naming scheme;

— personal mobility, i.e., the ability to reach a called party under a single, location-independent address even when the user changes terminals;

— terminal-type negotiation and selection.

Callers can be given a choice how to reach the party, e.g., via Internet telephony, mobilephone, an answering service, etc.:

— terminal capability negotiation;— caller and callee authentication;— blind and supervised call transfer;— invitations to multicast conferences.

Extensions of SIP are available to allow third-party signalling, for example for click-to-dial services, fully meshed conferences and connections to multipoint control units(MCUs), as well as mixed modes and the transition between them. SIP is addressing-neutral, with addresses expressed as URLs of various types such as SIP, H.323 ortelephone (E.164). SIP is independent of the packet layer and requires only an unreliabledatagram service, as it provides its own reliability mechanism [7] [8] [9].

Figure 2.5 shows one of the concept phones for video telephony.

2.3.2 Streaming Class

Multimedia streaming is a technique for transferring data such that it can be processedas a steady and continuous stream. Streaming technologies are becoming increasinglyimportant with the growth of the Internet because most users do not have fast enoughaccess to download large multimedia files quickly. With streaming, the client browser orplug-in can start displaying the data before the entire file has been transmitted.

For streaming to work, the client side receiving the data must be able to collect the dataand send it as a steady stream to the application that is processing the data and convertingit to sound or pictures. Streaming applications are very asymmetric and therefore typicallywithstand more delay than more symmetric conversational services. This also means thatthey tolerate more jitter in transmission. Jitter can be easily smoothed out by buffering.

Internet video products and the accompanying media industry as a whole is clearlydivided into two different target areas: (1) Web broadcast and (2) video streaming on-demand. Web broadcast providers usually target very large audiences that connect to a

20 WCDMA for UMTS

Figure 2.5. 3G concept phone for video telephony

highly performance-optimised media server (or choose from a multitude of servers) viathe actual Internet, which at present is very slow. The on-demand services are moreoften used by big corporations that wish to store video clips or lectures to a serverconnected to a higher bandwidth local intranet—these on-demand lectures are seldomused simultaneously by more than hundreds of people.

Both application types use basically similar core video compression technology, but thecoding bandwidths, level of tuning within network protocol use, and robustness of servertechnology needed for broadcast servers differ from the technology used in on-demand,smaller-scale systems. This has led to a situation where the few major companies devel-oping and marketing video streaming products have specialised their end-user products tomeet the needs of these two target groups. Basically, they have optimised their core prod-ucts differently: those directed to the ‘28.8 kbps market’ for bandwidth variation-sensitivestreaming over the Internet and those for the 100–7300 kbps intranet market.

At the receiver the streaming data or video clip is played by a suitable independentmedia player application or a browser plug-in. Plug-ins can be downloaded from the Web,usually free of charge, or may be readily bundled to a browser. This depends largely onthe browser and its version in use—new browsers tend to have integrated plug-ins forthe most popular streaming video players.

In conclusion, a client player implementation in a mobile system seems to lead to anapplication-level module that could handle video streams independently (with independentconnection and playback activation) or in parallel with the browser application when theservice is activated from the browser. The module would interface directly to the socket


interface of applied packet network protocol layers, here most likely UDP/IP or TCP/IP[10] [11].

2.3.3 Interactive Class

When the end-user, either a machine or a human, is on line requesting data from remoteequipment (e.g. a server), this scheme applies. Examples of human interaction with theremote equipment are Web browsing, database retrieval, and server access. Examplesof machine interaction with remote equipment are polling for measurement records andautomatic database enquiries (tele-machines).

Interactive traffic is the other classical data communication scheme that is broadlycharacterised by the request response pattern of the end-user. At the message destinationthere is an entity expecting the message (response) within a certain time. Round-trip delaytime is therefore one of the key attributes. Another characteristic is that the content ofthe packets must be transparently transferred (with low bit error rate).

2.3.3.1 Computer GamesPlaying a computer game interactively across the network is one example of applicationsthat can be seen to be part of the interactive class. However, depending on the natureof a game, i.e. how intensive data transfer is, it may rather belong to the conversationalclass due to high requirements for the maximum allowed end-to-end delay. Games usuallydeploy Java 2 Micro Edition (J2ME) technology.

2.3.4 Background Class

Data traffic of applications such as e-mail delivery, SMS, downloading of databases andreception of measurement records can be delivered background since such applicationsdo not require immediate action. The delay may be seconds, tens of seconds or evenminutes. Background traffic is one of the classical data communication schemes that isbroadly characterised by the fact that the destination is not expecting the data within acertain time. It is thus more or less insensitive to delivery time. Another characteristic isthat the content of the packets does not need to be transparently transferred. Data to betransmitted has to be received error free.

Multimedia Messaging Service (MMS) is an extension of successful Short MessagingService (SMS). As the name suggests, MMS contains not only text, but several elementssuch as text, voice, animated GIF images, JPG images, MIDI ringing tones and applica-tions. Since MMS uses MIME encapsulation, any element that has its own MIME type canbe sent as MMS. In addition to multimedia elements, MMS contains presentation infor-mation that conforms to Synchronised Multimedia Integration Language (SMIL). WhenSMIL is used for the presentation of multimedia messages on mobile terminals, the sizeof the window will be limited by the resolution and appearance of the terminal display.The layout of a multimedia message represents the content as created by the originatorand thus it may well be possible that the original layout does not fit into the display ofthe receiving terminal. Therefore the receiving terminal must be capable of replacing thelayout section with a terminal specific one where the size and the position of the text andimage regions are appropriately defined.

22 WCDMA for UMTS

MMS is jointly standardized by 3GPP and WAP Forum. 3GPP is responsible for generalrequirements, architecture and functionality [14] [15] whereas WAP Forum focuses ontransport protocol [16] [17]. MMS transport is done using WAP transport and any bearercapable of WAP can be used. Thus, MMS is bearer independent, e.g. MMS is not limitedto only GSM or WCDMA. WAP Wireless Session Protocol WSP is used for messagetransport from phone to MMSC and from MMSC to phone. In addition WAP push featuresare used to deliver the message from server to recipient. Shortly, sending MMS from anMMS client to an MMS client consists of the following phases:

• Client Sending MMS to MMS Server

• MMS Server Sending Notification to recipient Client (WAP push message using SMSas bearer)

• Client Fetching MMS from MMS Server

• MMS Server Sending Delivery Report to Client

If the recipient client is switched off, the line is busy, the terminal is out of thecoverage area, or for some other reason is not able to retrieve the message soon afterthe notification, then the MMS is stored in MMSC and can be fetched later. When acertain time has elapsed (configurable by the operator), MMS is removed from MMSCand moved to more permanent storage area that can be accessed via Web or WAP. Usercan then later delete, forward, save, or retrieve un-fetched messages [18] [19].

It is easy to predict that once terminals have built-in cameras and large colour displays(see Figure 2.6), MMS will soon take off MMS will soon take off and provide hugepossibilities for both operators and service providers.

Figure 2.6. Nokia 7650 is one example of Multimedia Messaging terminals


2.4 Service Capabilities with Different Terminal Classes

In WCDMA the same principle as with GSM with Terminal class mark is not used.WCDMA terminals shall tell the network, upon connection set up, larger set of parame-ters indicating the radio access capabilities of the particular terminal. These capabilitiesdetermine e.g. what is the maximum user data rate supported in particular radio configu-ration, given independently for the uplink and downlink directions. To provide guidanceon which capabilities should be applied together, reference terminal radio access capa-bility combinations have been specified in 3GPP standardisation, see [20]. The followingreference combinations have been defined for Release’99:

• 32 kbps class: This is intended to provide basic speech service, including AMR speechas well as some limited data rate capabilities up to 32 kbps

• 64 kbps class: This is intended to provide speech and data service, with also simulta-neous data and AMR speech capability

• 128 kbps class: This class has the air interface capability to provide for example videotelephony or then various other data services

• 384 kbps class is being further enhanced from 128 kbps and has for example multicodecapability which points toward support of advanced packet data methods provided inWCDMA

• 768 kbps class has been defined as an intermediate step between 384 kbps and 2 Mbpsclass

• 2 Mbps class: This is the state of the art class and has been defined for downlinkdirection only

These classes are defined so that a higher class has all the capabilities covered by a lowerclass. It should be noted that terminals may deviate from these classes when giving theirparameters to the network, thus 2 Mbps is possible for the uplink also though not coveredby any of the classes directly.

3GPP specifications include performance requirements for the bit rates up to 384 kbps,for more details see Section 12.5. Therefore, it is expected that terminals up to 384 kbpswill be available in the initial deployment phase.

High Speed Downlink Packet Access, HSDPA, further enhances the WCDMA bitrate capabilities. HSDPA terminal capabilities are defined in 3GPP Release 5 and extendbeyond 10 Mbps. HSDPA is covered in detail in Chapter 11.

2.5 Location Service in WCDMA

2.5.1 Location Services

Location-based services and applications are expected to become one of the new dimen-sions in UMTS. A location-based service is provided either by a teleoperator or by athird party service provider that utilises available information on the terminal location.The service is either push (e.g. automatic distribution of local information) or pull type

24 WCDMA for UMTS

Figure 2.7. 3G concept phone showing location-based service

(e.g. localisation of emergency calls). Other possible location-based services are discountcalls in a certain area, broadcasting of a service over a limited number of sites (broad-casting video on demand), and retrieval and display of location-based information, suchas the location of the nearest gas stations, hotels, restaurants, and so on. Figure 2.7 showsan example. Depending on the service, the data may be retrieved interactively or asbackground. For instance, before travelling to an unknown city abroad one may requestnight-time download of certain points of interest from the city. The downloaded informa-tion typically contains a map and other data to be displayed on top of the map. By clickingthe icon on the map, one gets information from the point. Information to be downloadedbackground or interactively can be limited by certain criteria and personal interest.

The location information can be input by the user or detected by the network ormobile station. The network architecture of the location services is discussed in Chapter 5.Release-99 of UMTS specifies the following positioning methods:

— the cell coverage-based positioning method,— Observed Time Difference Of Arrival—Idle Period DownLink (OTDOA-IPDL),— network-assisted GPS methods.

These methods are complementary rather than competing, and are suited for differentpurposes. These approaches are introduced in the following sections.

2.5.2 Cell Coverage Based Location Calculation

Cell coverage based location method is a network based approach, i.e., it does not requireany new functionalities in the mobile. The radio network has the location information with


Forced location update

Mobile in idle or URA_PCH stateMobile location is known with an

accuracy of location/registration area

Mobile in cell_DCH or cell_FACH or cell_PCH state

Mobile location is known with anaccuracy of one cell

Round triptime estimate

Location area

Figure 2.8. Location calculation with cell coverage combined with round trip time

a cell level accuracy when the mobile has been allocated a dedicated channel or whenthe mobile is in cell FACH or cell PCH states. These states are introduced in Chapter 7.If the mobile is in idle state, its location with cell accuracy can be obtained by forcingthe mobile to cell FACH state with a location update as illustrated in Figure 2.8.

The accuracy of the cell coverage based method depends heavily on the cell size. Thetypical cell ranges in the urban area are below 1 km and in the dense urban a few hundredmeters providing fairly accurate location information.

The accuracy of the cell coverage based approach can be improved by using the roundtrip time measurement that can be obtained from the base station. That information isavailable in cell DCH state and it gives the distance between the base and the mobilestation.

2.5.3 Observed Time Difference of Arrival, OTDOA

The OTDOA method is based on the mobile measurements of the relative arrival times ofthe pilot signals from different base stations. At least three base stations must be receivedby the mobile for the location calculation as shown in Figure 2.9. A measurement fromtwo base stations defines a hyperbola. With two measurement pairs, i.e. with three basestations, the location can be calculated.

In order to facilitate the OTDOA location measurements and to avoid near-far prob-lems, WCDMA standard includes idle periods in downlink, IPDL. During those idleperiods the mobile is able to receive the pilot signal of the neighbour cells even if thebest pilot signal on the same frequency is very strong. Typical frequency of the idle peri-ods is 1 slot every 100 ms, i.e. 0.7% of the time. The IPDL-OTDOA measurements areshown in Figure 2.10.

26 WCDMA for UMTS

A pair of base stations defines a hyperbola. Two pairs (= three base stations) gives the location.

Figure 2.9. Location calculation with three base stations

IPDL = idle periods indownlink signal

Arrivaltime t1

Arrivaltime t2 Location measurement

unit in base stations

Pilot signal Arrivaltime t3

Figure 2.10. IPDL (Idle Period Downlink)—OTDOA (Observed Time Difference of Arrival)

The network needs to know the relative transmission times of the pilot signals fromdifferent base stations to calculate the mobile location. That relative timing informationcan be obtained by

1. OTDOA measurements by the location measurement unit at the base station. Thebase station measures the relative timing of the adjacent cells. The measurement issimilar to the OTDOA measurements by the mobile.

2. GPS receiver at the base station.

The accuracy of the OTDOA measurements can be in the order of tens of meters invery good conditions when several base stations in line-of-sight can be received by themobile. In practise, such ideal measurement conditions are not typically available incellular networks. The accuracy depends on the following factors


— Number of base stations that the mobile can be receive. Minimum three is required.If more base stations can be received the accuracy is improved.

— Relative locations of the base stations. If the base stations are located in differentdirections from the mobile, the accuracy is improved.

— Line-of-sight. If there is a line-of-sight between the mobile and the base station, theaccuracy is improved.

The requirement of receiving at least three base stations is challenging in the cellularnetworks. The target of the network planning is to create clear dominance areas of thecells and to avoid unnecessary overlapping of the cells. That approach maximizes thecapacity. The clear dominance areas and limited cell overlapping reduces the probabilityof accurate OTDOA measurements as it is difficult to receive at least three pilot signals.Figure 2.11 shows the probability of mobile receiving several pilot signals in realisticnetwork scenarios. The probability of receiving at least three pilots is 74% in Figure 2.11.IPDL allows to receive the strongest pilot and the second strongest with 100% probability,but it is challenging to receive at least three pilots with very high probability. The requiredpilot Ec/I0 was −18 dB in these simulations and fully loaded network was assumed. Theresults show that IPDL greatly improves the performance of OTDOA: without IPDL theprobability of receiving at least three pilots would be only 31%. The results also showthat it is difficult to obtain very high probability of OTDOA measurements. The accuracycan be improved by combining OTDOA with cell coverage based location method.

2.5.4 Assisted GPS

Most accurate location measurements can be obtained with integrated GPS receiver inthe mobile. The network can provide additional information, like visible GPS satellites,

0%

10%

20%

30%

40%

50%

60%

Per

cen

tag

e o

f A

rea

1 2 3 4 5 6 7

Number of Pilots

Without IPDL

With IPDL

OTDOAmeasurements

can be done with≥3 pilots

Figure 2.11. Probability of receiving several pilot signals [21]

28 WCDMA for UMTS

GPS signal

Assistancedata Reference GPS

receiver in theradio network

GPS receiver in the mobile

GPSsatellite

GPSsatellite

GPSsatellite

Figure 2.12. Assisted GPS

reference time and Doppler, to assist the mobile GPS measurements. The assistance dataimproves the GPS receiver sensitivity for indoor measurements, makes the acquisitiontimes faster and reduces the GPS power consumption. The principle of assisted GPS isshown in Figure 2.12.

A reference GPS receiver in every base station provides most accurate assistance dataand most accurate GPS measurements by the mobile. The assisted GPS measurements canachieve accuracy of 10 meters outdoors and a few tens of meters indoors. That accuracymeets also the FCC requirements in USA. If the most stringent measurement probabilitiesand accuracies are not required, the reference GPS receiver is not needed in every basestations, but only a few reference GPS receivers are needed in the radio network. Itis also possible to let the mobile GPS make the measurements without any additionalassistance data.

2.6 Concluding Remarks

In this chapter we have briefly looked at UMTS from the perspective of services andapplications. The list is by no means complete, but hopefully it helps readers to understandthe variety of different services and gives some flavour of what we will see just a few yearsfrom now. UMTS provides high bit rates for both circuit-switched and packet-switchedconnections, effective bearer handling, multicall, and many other new features to make itpossible to create new applications in a cost-efficient manner.

References[1] 3GPP, Technical Specification Group Services and System Aspects, QoS Concept (3G TR

23.907 version 1.3.0), 1999.[2] 3GPP, Mandatory Speech Codec Speech Processing Functions, AMR Speech Codec: General

Description (3G TS 26.071 version 3.0.1), 1999.[3] 3GPP, Mandatory Speech Codec Speech Processing Functions, AMR Speech Codec: Frame

Structure General Description (3G TS 26.101 version 1.4.0), 1999.[4] 3GPP, Technical Specification Group Services and System Aspects, Speech Coded speech

processing functions, AMR Wideband Speech Codec, General Description, (3G TS 26.171version 5.0.0), 2001.


[5] 3GPP, Technical Specification Group Services and System Aspects, Codec for CircuitSwitched Multimedia Telephony Service: General Description (3G TS 26.110 version 3.0.1),1999.

[6] ITU-T H.324, Terminal for Low Bitrate Multimedia Communication, 1998.[7] 3GPP, Service Requirements for the IP Multimedia (3GPP TS 22.228), 2002.[8] 3GPP, IP Multimedia Subsystems (3GPP TS 23.228), 2002.[9] Handley, M., et al., SIP: Session Initiation Protocol, RFC2543, IETF, 1999.

[10] Honko, H., Internet Video Prestudy, 1997.[11] 3GPP, Transparent end-to-end packet switched streaming service (3GPP TS 26.233), 2001.[12] 3GPP, Technical Specification Group (TSG) RAN, Working Group 2 (WG2), Stage 2 Func-

tional Specification of Location Services in URAN (3G TR 25.923 version 1.4.0), 1999.[13] 3GPP, Technical Specification Group Services and System Aspects, Location Services

(LCS), Service description, Stage 1 (3G TS 22.071 version 3.1.0), 1999.[14] 3GPP, Multimedia Messaging Service (3GPP TS 22.140), 2001.[15] 3GPP, MMS Architecture and functionality (3GPP TS 23.140), 2001.[16] WAP-205-MmsArchOverview, 2001.[17] WAP-206-MmsMessagingService, 2001.[18] WAP-207-MmsInetInterworking, 2001.[19] WAP-208-MmsRelayRelayProtocol, 2001.[20] 3GPP, Technical Specification Group (TSG) RAN, Working Group 2 (WG2), UE Radio

Access Capabilities, 3G TS 25.306 version 3.0.0, 2000.[21] Johnson, C., Joshi, H. and Khalab, J. “WCDMA Radio Network Planning for Location Ser-

vices and System Capacity”, IEE 3G2002 conference in London, 9th May.

umts services

Technology

umts bearer service

bearer negotiation

bearer class

bearer parameters

bearer properties

bearer characteristics

bearer handling

umts services