Abstract:
A method of establishing a communication in a mobile communication system includes two steps. The mobile communication system includes a core network and at least one user equipment connected thereto via a radio access network. The method includes providing at least two types of domain for carrying a predetermined datastream for the communication in the core network. The method also includes providing one of the types of domain for carrying the predetermined datastream in the radio access network.

Description:
BACKGROUND TO THE INVENTION  
       [0001]     1. Field of Invention  
         [0002]     The present invention relates to a method of establishing a communication to a user equipment, and particularly, but not exclusively, to a method of establishing voice communication.  
         [0003]     2. Background to the Invention  
         [0004]     Communication networks are commonplace today. Communication networks typically operate in accordance with a given standard or specification. For example, the standard or specification may define the communication protocols and/or parameters that shall be used for a connection. Examples of the different standards and/or specifications include, without limiting to these, PSTN (Public Switched Telephone Network), GSM (Global System for Mobile communications), other GSM based systems (such as GPRS: General Packet Radio Service), AMPS (American Mobile Phone System), DAMPS (Digital AMPS), WCDMA (Wideband Code Division Multiple Access) or 3rd generation (3G) UMTS (Universal Mobile Telecommunications System), IMT 2000 (International Mobile Telecommunications 2000) and so on.  
         [0005]     In a cellular communication system a base station serves user equipment (UE) such as mobile stations via a wireless interface, which may also be referred to as an air interface. An appropriate transceiver apparatus may serve each of the cells of the cellular system. The communication from the UE to a core network may be via the air interface and a radio access network, which typically comprises a base station and a radio access network controller. The radio access network controller may be connected to and controlled by another controller facility that is typically in the core network of the communication system. An example of the core network controller is a serving GPRS support node (SGSN). The controllers may be interconnected and there may be one or more gateway nodes for connecting the cellular network to other communication networks. For example, the SGSN may be connected to a Gateway GPRS support node (GGSN) for connecting the mobile network to the Internet and/or other packet switched networks.  
         [0006]     Communication can take place with the transmission of data between the user equipment and the radio access network during a call. An example of one type of data that may be transmitted during a call is speech or voice data. Other data types include multimedia data such as video and audio data. The communication between different user equipment may adopt one of two services: a circuit switched (CS) service or a packet switched (PS) service.  
         [0007]     User equipment adopting a circuit switched service can communicate with each other over a transmission channel that is reserved for the entire duration of the communication. This implies that the data is transmitted over a fixed route, or fixed datastream, and that the transmission time is fixed and predictable. Typically the transmission is continuous and lasts for the duration of the entire communication. As such, a circuit switched service is ideally suited for speech or voice calls to user equipment, and has been adopted by communication systems such as PSTN and GSM. In GSM, the quality of voice calls may be further enhanced by utilising an adaptive multirate (AMR) codec.  
         [0008]     User equipment may also adopt a packet switched service. In a packet switched service, the transmitted data is broken into sub-blocks known as data packets. Each data packet can be transmitted from source to destination independently of other data packets, and it is up to the network to route these packets from source to destination. Each data packet has a packet header, which contains information such as the source and destination addresses for the data packet. In general, a packet switched service provides only a so-called ‘best effort’ service: the data packets are transmitted from source to destination without any guarantees about the quality of service (QoS). Therefore, it is possible that some of the packets are lost during transmission, and the time required for the transmission from source to destination is generally unpredictable. Due to varying load in the network and possibly also due to different transmission paths of the packets, the transmission delay can vary from data packet to data packet within a datastream. These variations in the transmission qualities and to the QoS means that packet switched services are not generally as well suited for speech or voice calls as real-time services, such as circuit switched services, in general.  
         [0009]     An example of a protocol operating a packet switched service is the Internet Protocol (IP). An example of a network that operates a packet switched service is a UMTS (Universal Mobile Telecommunications System) network. A UMTS network may include at its core an IP Multimedia Subsystem (IMS). The IMS is an IP based system that can handle both voice or speech data and multimedia data.  
         [0010]     Today, GSM networks have evolved from a GSM circuit switched based core network to integrate new 3G services. One example of such a system is GSM/UMTS network, which includes a core network that combines features from both a GSM circuit switched based core network and a packet switched based UMTS core network.  
         [0011]     In a GSM/UMTS network, circuit switched services can be provided by the GSM part of the network and routed via the GSM MSC (mobile switching centre). The MSC may be connected to traditional circuit switched networks such a PSTN or ISDN (Integrated Services Digital Network). Packet switched services may be routed via the GPRS (general packet radio service) part of the network, and specifically via a SGSN (serving GPRS support node) and a GGSN (gateway GPRS support node). The GGSN may be connected to packet switched networks such as the IMS of a UMTS system. The IMS may in turn be connected to other networks such as the Internet, a PSTN or another GSM/UMTS network.  
         [0012]     Presently in a GSM/UMTS network voice calls from user equipment are transmitted via either the circuit switched GSM part of the network, often referred to as the circuit switched domain, or the packet switched GPRS part of the network, often referred to as the packet switched domain. The routing may be dependent on the destination of the call. For example, if the voice call terminates at a standard telephone connected to a PSTN network, then the voice call may be transmitted via the circuit switched domain via the MSC. However, if the voice call terminates at a device connected to the IMS or some other network connected to the IMS such as the Internet or the PSTN, then the voice call may be transmitted via the packet switched domain. Voice calls transmitted via the packet switched domain are commonly referred to as Voice over IP (VoIP) calls.  
         [0013]     Voice calls over the GSM circuit switched domain may utilise GSM call control (CC) for call establishment, call clearing, call information phase and other call control procedures. For GSM voice calls, the data bearer or data path that carries the datastream for the voice call will be established via the circuit switched domain.  
         [0014]     For VoIP calls, a Session Initiation Protocol (SIP) may be used for call control for session establishment, session release, session status and other call control related procedures. For VoIP calls, the data bearer or data path that carries the datastream for the voice call will be established via the packet switched domain.  
         [0015]     In a GSM/UMTS network, there are problems with running two separate call control mechanisms for voice calls over a circuit switched domain and VoIP calls over a packet switched domain. The user equipment has to maintain two different protocols, or protocol stacks, for call control purposes: one for circuit switched voice calls and one for packet switched VoIP calls. Both call control mechanisms are used for similar purposes, such as call establishment/release, but with the significant difference that each mechanism is used to establish a different type of data bearer type.  
         [0016]     Present VoIP techniques also specifically suffer various problems. Packet switched domains are not typically optimised for real time traffic, such as voice calls. Therefore, the utilisation of the air interface for VoIP calls may not be optimised in a packet switched domain. Packet switched domians are not as well suited to real time traffic, such as that of voice calls, compared to circuit switched domains. In a circuit switched domain, a dedicated channel or data bearer can be established for the duration of a voice call, and the voice data can be transmitted continuously over this data bearer. In VoIP, because the voice data needs to be sent as individual data packets, an overhead may be introduced when the voice data is transmitted over the air interface. These overheads may arise from the additional data of various data packet headers in each data packet such as RTP (real time protocol) headers, UDP (user datagram protocol) headers and IP headers. In a wired Internet network, such overheads may be acceptable due to the low costs of transmitting data. However, in a wireless system, such overheads may not be acceptable as the resource of the radio spectrum of the air interface is very valuable and efficiency is paramount.  
       SUMMARY OF THE INVENTION  
       [0017]     It is the aim of embodiments of the present invention to address one or more of the above-stated problems.  
         [0018]     According to the invention there is provided a method of establishing a communication in a mobile communication system comprising a core network and at least one user equipment connected thereto via a radio access network, the method comprising providing at least two types of domain for carrying a predetermined datastream for the communication in the core network, and providing one of said types of domain for carrying the predetermined datastream in the radio access network.  
         [0019]     The method is adapted such that the predetermined datastream is carried on the one type of domain in the radio access network irrespective of the domain on which it is carried in the core network.  
         [0020]     The communication may be a voice call and the predetermined datastream may comprise voice data. The at least two type of domains may include a packet switched domain and a circuit switched domain, wherein the circuit switched domain is provided in the radio access network.  
         [0021]     The user equipment may be connected to the radio access network via an air interface, and the circuit switched domain provided in the radio access network may also provided over the air interface.  
         [0022]     The voice call may be established using first and second call establishment methods for each of the packet and circuit switched domains. The first call establishment method for the packet switched domain may be a SIP based method. The SIP based method may be used to establish a PDP context. The PDP context may be initiated between a control element of the packet switched domain and a gateway element of the circuit switched domain. The PDP context may be initiated in the packet switched domain after an initial PDP context establishment for the voice call.  
         [0023]     The gateway element may establish a data bearer for the voice call.  
         [0024]     A second call establishment method for the circuit switched domain may be a circuit switched call control method.  
         [0025]     The gateway element may be a circuit switched media gateway.  
         [0026]     The voice call may be established using a single call establishment method for each of the packet and circuit switched domains. The call establishment method may be a circuit switched call control method.  
         [0027]     The voice data may be carried in the circuit switched domain in the core network. The voice call may be established using a single call establishment method for each of the packet and circuit switched domains. The call establishment method may be a SIP based method.  
         [0028]     The voice data may be carried in a circuit switched datastream over the radio access network and the core network. If the datastream for the voice data terminates at a packet switch enabled device, an interface may be provided to the core network to convert the circuit switched datastream. The circuit switched datastream may pass through a gateway element. The gateway element is a circuit switched media gateway.  
         [0000]     The user equipment may be a mobile terminal.  
         [0029]     In a further aspect the invention provides a mobile communication system comprising a core network and at least one user equipment connected thereto via a radio access network, the mobile communication system comprising means for providing at least two types of domain adapted for carrying a predetermined datastream for a communication in the core network, and means for providing one of said types of domain for carrying the predetermined datastream in the radio access network.  
         [0030]     The means for providing one of said types of domain for carrying the predetermined datastream in the radio access network may be adapted to direct the predetermined data stream to one domain in the radio access network irrespective of the domain in which the datastream is carried in the core network.  
         [0031]     The core network may be adapted to detect the predetermined data stream on either of the two domains, and responsive thereto to direct the datastream only to the one domain in the radio access network.  
         [0032]     The communication may be a voice call and/or the predetermined datastream may comprise voice data.  
         [0033]     The at least two type of domains may include a packet switched domain and a circuit switched domain, and the circuit switched domain may be provided in the radio access network.  
         [0034]     The user equipment may be connected to the radio access network via an air interface, and the circuit switched domain provided in the radio access network may be also provided over the air interface.  
         [0035]     The mobile communication system may further comprise means for establishing the voice call using first and second call establishment methods for each of the packet and circuit switched domains.  
         [0036]     The mobile communication system may further comprise means for establishing the voice call using a single call establishment method for each of the packet and circuit switched domains.  
         [0037]     The mobile communication system may further comprise means for carrying the voice data in the circuit switched domain in the core network.  
         [0038]     The mobile communication system may further comprise means for establishing the voice call using a single call establishment method for each of the packet and circuit switched domains.  
         [0039]     In a further aspect the invention provides a mobile communication system comprising a core network and at least one user equipment connected thereto via a radio access network, the mobile communication system comprising each of a packet switched domain and a circuit switched domain in the core network, wherein each of said packet switched and circuit switched domain are adapted to carry a predetermined datastream for a communication in the core network, wherein the predetermined datastream is carried only in the circuit switched domain in the radio access network. The predetermined datastream is preferably a voice datastream. As such, voice data transmitted in the packet switched domain in the core network is transmitted in the circuit switched domain in the radio access network.  
         [0040]     In a further aspect the invention provides a network element for a mobile communication system, wherein said mobile communication system comprises a core network and at least one user equipment connected thereto via a radio access network and at least a packet switched domain and a circuit switched domain for carrying a predetermined datastream for a voice call in the core network, said network element adapted for providing the circuit switched domain for carrying the predetermined datastream in the radio access network.  
         [0041]     In a further aspect the invention provides a network element for a mobile communication system, wherein said mobile communication system comprises a core network and at least one user equipment connected thereto via a radio access network, and at least a packet switched domain and a circuit switched domain for carrying a predetermined datastream for a voice call in the core network, said network element being adapted to route the predetermined datastream in the packet switched domain in the core network to the circuit switched domain in the radio access network.  
         [0042]     Said network element is preferably controlled responsive to a packet data protocol request as defined hereinafter.  
         [0043]     A first exemplary embodiment is described herein, in which a VoIP voice call is preferably routed in a core network of a mobile communication system in the packet switched domain, and preferably routed in the radio access network and/or the air interface between the core network and a user equipment in the circuit switched domain. The radio access network, and the air interface, are thus preferably configured with circuit switched data bearers. The VoIP voice call is preferably terminated at an IP enabled terminal connected to an external network accessed through the core network.  
         [0044]     The establishment of the call is preferably achieved using a PDP context set up using SIP for call control. This first embodiment thus ensures that the call is always routed in the circuit switched domain in the radio access network, and therefore the air interface, regardless of whether the call is handled in the circuit switched or packet switched domain in the core network.  
         [0045]     In this first embodiment, there is no specific requirement for how the set-up of circuit switched calls may be configured. Circuit switched calls may be set up using conventional circuit switched call-control mechanisms.  
         [0046]     Thus the first embodiment preferably provides for an arrangement in which calls are always routed in the circuit switched domain in the air interface, but packet switched calls may be established using a PDP context set up using a SIP session, and circuit switched calls may be set-up using conventional circuit switched call control methods. Preferably certain types of calls, such as voice calls, are routed in this way.  
         [0047]     Thus, in accordance with a first embodiment of the invention, there is generally provided a method of establishing a communication in a mobile communication system comprising a core network and at least one user equipment connected thereto via a radio access network, the method comprising providing at least two types of domain for carrying a predetermined datastream for the communication in the core network, and providing one of said types of domain for carrying the predetermined datastream in the radio access network.  
         [0048]     Preferably, the communication is a voice call and the predetermined datastream comprises voice data.  
         [0049]     The at least two type of domains may include a packet switched domain and a circuit switched domain, wherein the circuit switched domain is provided in the radio access network. The user equipment may be connected to the radio access network via an air interface, and the circuit switched interface provided in the radio access network may also be provided over the air interface.  
         [0050]     Preferably, the voice call is established using first and second call establishment methods for each of the packet and circuit switched domains.  
         [0051]     The first call establishment method for the packet switched domain may be a SIP based method. The SIP based method may also be used to establish a PDP context.  
         [0052]     The PDP context may be initiated between a control element of the packet switched domain and a gateway element of the circuit switched domain. Preferably the PDP context is initiated in the packet switched domain after an initial PDP context establishment for the voice call.  
         [0053]     The gateway element may establish a data bearer for the voice call.  
         [0054]     The second call establishment method for the circuit switched domain may be a circuit switched call control method.  
         [0055]     The gateway element may be a circuit switched media gateway.  
         [0056]     The first embodiment may provide, in an alternative arrangement, a method of routing a communication in a communication network, the communication network having at least two transport mechanisms for transferring a data stream to a terminal, wherein the data stream is routed through the network in dependence on the first transport mechanism, and selectively routed to the terminal on the second transport mechanism.  
         [0057]     The alternative arrangement of the embodiment preferably provides a gateway between the network and the terminal, said gateway transferring the data stream fro one transport mechanism to the other.  
         [0058]     The alternative arrangement of the embodiment preferably comprises a mobile communication system, in which the network comprises a mobile communication system having a packet switched domain supporting the first transport mechanism and a circuit switched domain supporting the second transport mechanism, and an air interface between the network and the terminal, the data stream being transported in the packet switched domain within the network, and being carried in a circuit switched transport mechanism in the air interface. The gateway is preferably an interface between the packet switched domain and the circuit switched domain.  
         [0059]     The data stream may preferably consist of voice data. The voice data may be transported as Voice over IP in the packet switched domain, and AMR speech in the circuit switched domain. The voce data may be transported as AMR speech in the Voice over IP packets. The communication between the network and the terminal is preferably established by way of a PDP context between the terminal and the network or a further network environment connected to the network, such as an IP environment.  
         [0060]     The selective routing is preferably responsive to an additional PDP context established between the terminal and a control element supporting the first transport mechanism. Such control element preferably communicates with the gateway.  
         [0061]     In the alternative arrangement of the first embodiment, the control of the selective routing is preferably by use of a SIP session, and specifically a PDP context.  
         [0062]     In the first embodiment of the invention there is preferably also provided a packet data protocol (PDP) context for establishing a circuit switched connection between a user equipment and a network element. The user equipment may be any device for connection in to a communication network, for example a mobile terminal. The network element may be an access network element such as an element of a radio access network. The network element may be a core network element. A core network element may be a gateway element. The packet data protocol context may be used to control a gateway element in the core network. The gateway element may be controlled to terminate packet switched communications directed toward the user equipment, and to terminate circuit switched communications directed toward the core network. The gateway element may provide an interface between a packet switched domain in the core network and a circuit switched domain in the access network. The packet data protocol may configure the gateway element. The packet data protocol may be initiated by the user equipment or the core network.  
         [0063]     A second embodiment differs from the first embodiment, in that control of the selective routing may preferably be by use of circuit switched call control as defined for circuit switched speech.  
         [0064]     As in the first embodiment, for the second embodiment there is ensured that the call is always routed in the circuit switched domain over the air interface, regardless of whether the call is handled in the circuit switched or packet switched domain in the core network.  
         [0065]     The second embodiment offers an advantage over the first embodiment, in that the calls are preferably set-up using a single technique. Specifically regardless of whether the calls are packet switched or circuit switched, a circuit switched call control technique as defined for circuit switched speech is preferably used for call set-up, and the calls are preferably all transmitted in the air interface in the circuit switched domain.  
         [0066]     In an alternative arrangement of the second embodiment there is provided a method of routing a communication in a communication network, the communication network having at least two transport mechanisms for transferring a data stream to a terminal, wherein the data stream is routed through the network in dependence on the first transport mechanism, and selectively routed to the terminal on the second transport mechanism.  
         [0067]     A third exemplary embodiment is described herein, in which a VoIP call is preferably routed in a core network of a mobile communication system, in the circuit switched domain, and preferably routed in the radio access network and over the air interface between the core network and a user equipment in the circuit switched domain. The radio access network and air interface are thus preferably configured with circuit switched data bearers. The VoIP voice call is preferably terminated at an IP enabled terminal connected to an external network accessed through the core network, and conversion may be required at the interface of that external network to the IP enabled terminal.  
         [0068]     The establishment of the call is preferably achieved using SIP for call control. This third embodiment thus preferably ensures that the call is always routed in the circuit switched domain in the air interface and in the core network, regardless of whether the call is a VoIP call or a circuit switched call.  
         [0069]     The third embodiment further preferably establishes the circuit switched call using a PDP context of an SIP. Thus calls may be established in the circuit switched domain using SIP for call control.  
         [0070]     A characteristic of each embodiment described herein is that, where a voice call is established, it is established in the radio access network and/or over the air interface between the core network and the user equipment using a circuit switched connection. Thus, all voice calls are preferably established in the circuit switched domain in the air interface, even if they are VoIP calls. More generally, for a given datastream, the datastream is carried in the radio access network and/or over the air interface by a predetermined domain, regardless of the domain carrying the datastream in the core network.  
         [0071]     Preferably the control method that establishes the connection in the domain over the air interface is the same regardless of the domain used the core network.  
         [0072]     The connection in the core network may further always be a circuit switched connection, regardless of whether the call terminates with a VoIP enabled terminal.  
     
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       [0073]     For a better understanding of the present invention reference will now be made by way of example only to the accompanying drawings, in which:  
         [0074]      FIG. 1  illustrates a communication system in which embodiments of the present invention can be applied;  
         [0075]      FIG. 2  illustrates an arrangement of the prior art;  
         [0076]      FIG. 3  illustrates a communications system in a first embodiment of the invention;  
         [0077]      FIG. 4  illustrates a flow chart in the first embodiment of the invention;  
         [0078]      FIG. 5  illustrates a message flow diagram in a second embodiment of the invention;  
         [0079]      FIG. 6  illustrates a further message flow diagram in a second embodiment of the invention;  
         [0080]      FIG. 7  illustrates a communications system in a third embodiment of the invention; and  
         [0081]      FIG. 8  is a message flow diagram for the third embodiment of the invention. 
     
    
     DESCRIPTION OF THE PREFERRED EMBODIMENTS  
       [0082]     The present invention is described herein with reference to particular examples. The invention is not, however, limited to such examples. In particular the invention is described by way of reference to an exemplary GSM/UMTS network.  
         [0083]      FIG. 1  illustrates an exemplary known GSM/UMTS network  100  that supports both circuit switched and packet switched services. The network  100  comprises various network elements including a base station (BS)  102 . The BS may communicate with user equipment (UE)  101  over an air interface  110 . Examples of UEs include mobile terminals, personal digital assistants (PDAs) and other suitably configured devices. The BS  102  is further connected to a radio network controller (RNC)  103 . The BS  102  and RNC  103  are generally referred to as a radio access network (RAN). The RNC is connected to other network elements, including a mobile switching centre (MSC)  104  and a serving GPRS support node (SGSN)  105 . The MSC  104  is connected to a home location register (HLR)  108 . The SGSN is connected to a gateway GPRS support node (GGSN)  106 . The elements of the BS  102 , RNC  103 , MSC  104 , HLR  108 , SGSN  105  and GGSN  106  together comprise a GSM UMTS public land mobile network (PLMN).  
         [0084]     The MSC  104  may communicate with external networks such as a public switched telephone network (PSTN)  109 . The GGSN  106  may communicate with external packet data networks such as an IMS network  107 . The MSC  104 , HLR  108  and PSTN  109  form part of a circuit switched (CS) domain  120 . The SGSN  105 , GGSN  106  and IMS  107  form part of a packet switched (PS) domain  122 .  
         [0085]     The PSTN  109  may further connect to standard telephones  110  and  111 . The IMS may further connect to other networks such as the Internet  116 , another PLMN  117  and a PSTN  118 . User equipment connected to each of these networks are able to communicate with the IMS. The user equipment may include a personal computer  112  and a SIP enabled device  113  connected to the Internet  116 , a mobile terminal  114  connected to the PLMN  117 , and a standard telephone  115  connected to the PSTN  118 .  
         [0086]     Reference is now made to  FIG. 2 , which illustrates examples of known arrangements for a voice, or speech, call in the network of  FIG. 1 .  
         [0087]      FIG. 2 ( a ) illustrates a voice call between a UE  201 , such as a mobile terminal, and a fixed line telephone  203 . The telephone  203  is connected to a  
         [0088]     CS domain  202  via a PSTN connection  205 . The UE  201  uses CS call control (CC) to establish a voice call between the UE  201  and the telephone  203  via the CS domain  202  and the PSTN connection  205 . The CS domain  202  may include all the elements of the CS domain  120  described in  FIG. 1  and may further include a RAN. Communications between the UE  201  and the CS domain  202  takes place over a user plane,  204 , defined in the air interface.  
         [0089]      FIG. 2 ( b ) illustrates a voice call between a UE  250 , such as a mobile terminal, and a Session Initiation Protocol (SIP) enabled device, such as a mobile terminal  256 , a desktop computer  257  or a laptop  258 . The SIP enabled device is connected to an IMS  252  via a PS connection such as may be provided by the Internet, which provides VoIP connectivity. The IMS  252  is connected to a PS domain  251 . The UE uses SIP based signalling to establish a voice call between the UE  250  and the SIP enabled device  256 ,  257  or  258 . This voice call between the IMS  252  and the SIP enabled device  256 ,  257 ,  258  may be in the form of a VoIP datastream. The PS domain  251  may include all the elements of the PS domain  122  described in  FIG. 1  and may further include a RAN. Communications between the UE  250  and the PS domain  251  takes place over the user plane  254  defined in the air interface.  
         [0090]     The datastream for a voice call over the user plane  254  between the UE  250  and the PS domain  251  for a VoIP call as shown in  FIG. 2 ( b ) may be larger than the datastream over the user plane  204  between the UE  201  and the CS domain  202  for a circuit switched voice call as shown in  FIG. 2 ( a ). This is partly due to the overhead of transmitting data packet headers that are present in data packets transmitted in a PS domain, especially over the air interface.  
         [0091]     Furthermore, in  FIG. 2 ( a ) CS call control is used to establish a voice call, whereas in  FIG. 2 ( b ), SIP based signalling is used to establish a voice call.  
         [0092]     Reference is now made to  FIG. 3 , which illustrates the establishment of a voice call in a first embodiment of the invention.  FIG. 3  illustrates a user equipment (UE)  401 , which establishes a call with a calling or called party  407 . A radio access network (RAN)  402  connects the UE  401  and a core network  400 . A packet switched domain  450  of the core network  400  includes a serving GPRS support node (SGSN)  403  and a gateway GPRS support node (GGSN)  404 . A circuit switched domain  452  of the core network  400  includes a circuit switched media gateway (CS MGW)  406 . An IP multimedia sub system (IMS)  405  is connected to the packet switched domain  450  of the core network  400 . The called/calling party  407  is connected to the IMS  405 . Each of the UE  401  and the CS MGW  452  include associated conversion entities  408  and  409  respectively.  
         [0093]     Only those elements of the core network  400  for understanding the described embodiment of the present invention are illustrated in  FIG. 3 . The RAN  402  and the CS MGW  406  may be considered to form part of the circuit switched infrastructure of the network.  
         [0094]     Referring further to  FIG. 3 , the UE communicates with the RAN  402  via communication link  456  over the Uu air interface. The RAN  402  communicates with the SGSN  403  via communication link  458 , and communicates with the CS MGW  406  via communication link  460 . The SGSN  403  communicates via communication link  462  with the GGSN  404 . The SGSN  403  communicates via communication link  464  with the CS MGW  406 . The GGSN  404  communicates with the IMS  405  via communication link  466 . The IMS  405  communicates via communication link  468  with the called or calling party  407 .  
         [0095]     The IMS  405  may communicate with the called/calling party  407  over a PSTN network if it is a fixed line telephone, or over a packet switched based network if it is an appropriately enabled device.  
         [0096]     The UE may establish a voice call with the called or calling party  407 , being either a telephone or a mobile terminal using SIP based signalling between the UE and the IMS in accordance with this embodiment of the invention.  
         [0097]     If the voice call is to a telephone (a circuit switched destination), then the call may be transmitted through a PSTN network. If the call is to a SIP enabled device (a packet switched destination), then the call may be transmitted through a PS network. In both cases, this embodiment of the invention enables the use of SIP based signalling to establish the voice call.  
         [0098]     In the following description of  FIG. 3 , it should be understood that the signalling and communications may occur in either direction between the UE  401  and the called or calling party  407 .  
         [0099]     If the called or calling party  407  is a packet switched device, such as a SIP enabled device, then the call may be a VoIP call, and the datastream carrying the voice data between the packet switched domain  450  and the party  407  is a VoIP datastream.  
         [0100]     However, in accordance with this embodiment of the invention, the VoIP datastream is terminated at a media gateway in the circuit switched domain. The VoIP datastream is converted at the media gateway to a circuit switched datastream carrying the voice data of the call.  
         [0101]     The circuit switched datastream may be AMR (adaptive multirate) coded speech. The circuit switched datastream is transmitted over the RAN  403  and the air interface to the UE  401 . The UE  401  may then convert the circuit switched datastream back to the audio of the original voice data. Alternatively, the UE  401  may convert the circuit switched datastream to a packet switched datastream if, for example, the UE  401  is itself a SIP enabled device adapted to operate with packet data.  
         [0102]     In this embodiment of the present invention, a new type of packet data protocol (PDP) context is established in the packet switched domain. The term ‘PDP context’ typically refers to the part of the data connection or data bearer that passes through the packet switched domain, for example the GPRS part of the UMTS network. The PDP context or data bearer can be seen as a logical connection or “pipe” from the UE to a gateway node, such as the GGSN. The new PDP context may be labelled “AMR Speech”, “Non Transparent IP Multimedia Stream” or any other suitable label.  
         [0103]     This embodiment of the invention is now described in detail, with reference to  FIG. 3  and the flow chart of  FIG. 4 .  
         [0104]     For the purposes of the described embodiment, it is assumed that a VoIP call is established between the UE  401  and the IMS  405  (an external network), to support a voice call between the UE  401  and the called/calling party  407 . The call may be initiated by either the UE  401  or the called/calling party  407 . This is represented by step  502  in  FIG. 4 . It is assumed that existing and established principles are used to establish the call, and the connection set-up logic is exactly as for normal VoIP calls.  
         [0105]     On establishing this call, it is detected that at least one of the data streams to be established may be realised in the circuit switched domain over the air interface. This is represented by step  504  in  FIG. 4 . More particularly, in this embodiment, it is determined that at least one of the datastreams is a voice datastream which may be realised as “AMR speech”. More generally, the data stream may be realised as a “non-transparent IP multimedia stream” rather than as a transparent IP stream between the UE  401  and the network  400 . This requires agreement between the UE  401  and the IMS  405  that at least one datastream will terminate in the CS MGW  406 , and be established as a non-transparent datastream. The notification to the CS MGW  406  may be made by either the UE  401  or the IMS  405 .  
         [0106]     In the described preferred embodiment, the UE  401  detects the characteristic of the datastream. The UE, after authorising such with the IMS  405 , then activates a new type of PDP context toward the SGSN  403 , being referred to herein (by way of example only) as the “AMR speech PDP context”. This is represented by step  506  in  FIG. 4 . The SGSN  403  then detects, responsive to the AMR speech PDP context, that an “AMR speech”-type of data bearer is required toward the UE  401 .  
         [0107]     The SGSN  403  maps the required data bearer to the CS MGW  406 , and determines that the CS MGW  406  offers a gateway to such data bearer. This is represented by step  508  in  FIG. 4 . The SGSN  403  then signals to the CS MGW  406 , and requests the desired data bearer information and user plane parameters from the CS MGW  406 .  
         [0108]     The CS MGW  406  allocates the necessary transcoding functions, and allocates the data bearer mapping in the RAN toward the Iu interface  456 . The CS MGW  406  prepares the mapping to the IP datastream to and from the external IMS network  405 , and provides the selected parameters, i.e. the user plane parameters, to the SGSN  403 . This is represented by step  510  in  FIG. 4 .  
         [0109]     The mapping of the IP datastream in the CS MGW  406  requires mapping of VoIP data to AMR speech toward the UE  401 , and mapping of AMR speech to VoIP data toward the IMS  405 .  
         [0110]     The SGSN thus establishes a circuit switched data bearer over the RAN  402  and Uu air interface, via the CS MGW  406 , toward the UE  401 . The circuit switched data bearer over the RAN and air interface may also be referred to as a circuit switched radio access bearer. Generally, the term radio access bearer refers to a data bearer established over the air interface for voice calls. The establishment of this circuit switched data bearer completes the AMR speech PDP context as represented by step  512  in  FIG. 4 . The circuit switched data bearer may utilise a circuit switched service over the Iu air interface and any AMR specific procedures such as time alignment and dynamic codec selection. It should be noted that this embodiment is directed to an example where an AMR datastream is directed towards and carried in an appropriate data bearer. More generally, any specific datastream may be directed toward an appropriate bearer in dependence on the type of datastream.  
         [0111]     After establishment of the AMR speech PDP context in  FIG. 3 , an IP datastream from the called/calling party  407  is directed to the CS MGW  406 , by SIP signalling between the SGSN  403  and the external IMS network  405 . The CS MGW  406  terminates the IP datastream, and utilising the conversion entity  409  converts the IP datastream to an encoded datastream suitable for the data bearer established toward the UE on the Iu interface  456 . In the present embodiment, this conversion is to a circuit switched datastream of AMR speech.  
         [0112]     The UE  401  may utilise the conversion entity  408  to re-converts the encoded AMR speech, or may use the AMR speech directly.  
         [0113]     The same principles work, in reverse, for communications from the UE  401  to the called/calling party  407 . The transmission of the datastream to and from the UE  401  in this way is represented by step  514  in  FIG. 4 .  
         [0114]     The conversion that takes place at the UE  401  may be dependent on the terminal type of the user equipment, or the data format required by the user. For a standard voice call, the conversion will typically be back to audio data, which may be based on standard speech decoding techniques.  
         [0115]     Thus, this embodiment of the invention enables a communication in the network to be transmitted over the air interface on the most appropriate data bearer for the type of data of the communication. Specifically, data is routed in the packet switched domain in the core network and is routed in a circuit switched domain in the RAN and over the air interface, where conversion between the packet switched domain and circuit switched domain is through a circuit switched media gateway.  
         [0116]     The first embodiment, illustrated with reference to  FIGS. 3 and 4 , preferably establishes a voice call in the circuit switched domain over the RAN and air interface, where the call is established using a PDP context set up using SIP (session initiation protocol) for call control. This differs from the prior art, where the establishment of a call using a PDP context results in the voice call being established in the packet switched domain over the air interface.  
         [0117]     Reference is now made to  FIGS. 5 and 6 , which illustrate the establishment of a voice call in a second embodiment of the invention.  
         [0118]     In this embodiment, the existing circuit switched call control (CC) signalling known in the art is reused to establish the user plane over the air interface between the UE and the CS MGW where the call is established in the core network in the packet switched domain.  
         [0119]     In this second embodiment, there is no requirement for the new type of PDP context described in the first embodiment, as it reuses the existing infrastructure and signalling of the GSM/UMTS network. This embodiment utilises the existing circuit switched call control as defined for circuit switched speech.  
         [0120]      FIG. 5  illustrates the second embodiment in the example scenario where the user equipment originates the call.  FIG. 6  illustrates the second embodiment in the example where the user equipment receives, or terminates, the call.  
         [0121]      FIG. 5  illustrates a message flow diagram in the second embodiment of the invention. The message flow is between the network elements of a terminal equipment (TE)  602 , a mobile terminal (MT)  604 , a serving mobile switching centre coupled to a visitor location centre (S-MSC/VLR)  606 , a gateway mobile switching centre (GMSC)  608 , a serving call sate control function (S-CSCF)  610 , and a home subscriber server (HSS).  
         [0122]     The TE  602  and MT  604  together comprise a user equipment. The TE  602  is a user plane entity and the MT  604  is a control plane entity.  
         [0123]     The S-MSC/VLR  606 , the GMSC  608 , the S-CSCF  610  and HSS  612  all form part of the core network. The S-CSCF  610  may be located in an IMS and form part of a packet switched domain.  
         [0124]     Only those network elements necessary for the understanding of the present embodiment are illustrated in  FIG. 5 . A person skilled in the art will appreciate that other network elements may be present that are not illustrated in  FIG. 5 .  
         [0125]     The user equipment comprising TE  602  and MT  604  attempts to establish a voice call to a called party (not illustrated). The TE triggers a call using SIP, denoted by block  650 . A SIP setup message “Setup” is transmitted from the TE  602  to the S-CSCF  610 . The SIP message contains information on the circuit switched capabilities of the TE  602  (denoted CS capability), the IP address of the called party, denoted B_IP, and the international mobile subscriber identity (IMSI) of the TE  602 , denoted A_IMSI?.  
         [0126]     The S-CSCF  610  then sends a MEGACO configuration message “Config”  654  to the GMSC  608 . The MEGACO message contains the IMSI and the B_IP transmitted by the TE  602 . The GMSC  608  allocates a mobile station roaming number (MSRN)  656  as denoted by block  565 , and sends a MEGACO response “Response” message  658  back to the S-CSCF  610 . The MEGACO response message  658  contains MSRN and B_IP.  
         [0127]     The S-CSCF  610  then sends a SIP message  660  towards the called party connected via a packet switched network to the S-CSCF  610 . The S-CSCF  610  also sends a SIP response message “Response”  662  back to the TE  602 . The SIP response message “Response”  662  contains a token corresponding to the MSRN (denoted token=MSRN).  
         [0128]     The TE  602  sends a setup message “Setup”  664  to the MT  604 , which includes the token indicating the MSRN.  
         [0129]     The MT  604  recognizes the MSRN, as denoted by block  606 , and initiates circuit switched call control. The MT  604  uses circuit switched call control to set up the call by sending a call control setup message “CC Setup”  668  to the S-MSC/VLR  606  containing information relating to the MSRN (denoted B=MSRN).  
         [0130]     The S-MSC/VLR  606  sends an ISUP initial address message (IAM)  670  to the GMSC  608 . The ISUP IAM message  670  contains the information relating to the MSRN (denoted B=MSRN). Then messaging  672  takes place between the GMSC  608  and the HSS  612 , exchanging routing information and to verify the TE.  
         [0131]     The GMSC  608  sends a routing information message “Send Routing Info”  674  comprising the mobile subscriber ISDN number for the TE  602  (denoted A_MSISDN) to the HSS  612 . The HSS sends a routing information response message “send Routing Info Resp”  676  comprising the IMSI of the TE  602  (denoted A_IMSI).  
         [0132]     The GMSC  608  can then verify, as denoted by block  678 , the IMSI received from the TE  602  with that received from the HSS to verify the TE  602 . The GMSC then sends an ISUP inquiry access code (IAC) message  680  to the S-MSC/VLR  606 .  
         [0133]     Once this message is received, call establishment is complete, and the S-MSC/VLR  606  sends a call establishment complete message “Complete”  682  using circuit switched call control to the MT  604 .  
         [0134]     Reference is now made to  FIG. 6  where a calling party attempts to establish a voice call with a user equipment. Note that references for like elements in  FIG. 5  are used in  FIG. 6 .  
         [0135]     A calling party (not illustrated) sends a SIP setup message  751  to the S-CSCF  610 .  
         [0136]     The S-CSCF  610  receives this message  750 , as denoted by block  759 , and sends a MEGACO configuration message “Config”  752  to the GMSC  608 . The MEGACO message  752  contains the IP address of the calling party (B_IP) and the international mobile subscriber identity (IMSI) of the calling party. The GMSC  608  allocates a mobile station roaming number (MSRN), as denoted by block  754 , and sends a MEGACO response message “Response”  756  back to the S-CSCF  610 . The MEGACO response message  756  contains the MSRN identity. The S-CSCF  610  then sends a SIP setup message “Setup”  758  towards the TE  602 . The SIP setup message  758  contains a CS capability field and a token=MSRN? field.  
         [0137]     The TE  602  sends a SIP response message “Response”  760  back to the S-CSCF  610 , and also a setup message “Setup”  762  to the MT  604 , which includes the token indicating the MSRN (token=MSRN).  
         [0138]     The MT  604  recognizes the MSRN and initiates circuit switched call control  764 , as denoted by block  764 . The MT  604  uses circuit switched call control to set up the call by sending a call control setup message “CC Setup”  766  to the S-MSC/VLR  606  containing information relating to the MSRN.  
         [0139]     The S-MSC/VLR  606  sends an ISUP initial address message (IAM)  768  to the GMSC  608 . The ISUP IAM message  768  contains the information relating to the MSRN (denoted B=MSRN). Messaging  770  then takes place between the GMSC  608  and the HSS  612  exchanging routing information and to verify the TE.  
         [0140]     The GMSC  608  sends a routing information message “Send Routing Info”  772  comprising the mobile subscriber ISDN number for the TE  602  (A_MSISDN) to the HSS  612 . The HSS sends a routing information response message “Send Routing Info Response”  774  comprising the IMSI of the TE  602  (denote A=IMSI).  
         [0141]     The GMSC  608  can then verify, as denoted by block  776 , the IMSI received from the TE  602  with that received from the HSS to verify the TE  602 . The GMSC then sends an ISUP inquiry access code (IAC) message “IAC”  680  to the S-MSC/VLR  606 .  
         [0142]     Once this message is received, call establishment is complete, and the S-MSC/VLR  606  sends a call establishment complete message “Complete”  682  using circuit switched call control to the MT  604 .  
         [0143]     In both the embodiments illustrated in  FIGS. 5 and 6 , the call establishment method is circuit switched call control as the messaging between the MT and the S-MSC/VLR (the core network) is done using circuit switched call control. Once the call is established, data is transmitted between the MT and the core network in the radio access network and air interface in the circuit switched domain. Within the core network, the call is handled in the packet switched domain.  
         [0144]     Thus as in the first embodiment of  FIGS. 3 and 4  a voice call is preferably only established in the CS domain in the air interface, and additionally the voice call is only established using CS techniques.  
         [0145]     Reference is now made to  FIGS. 7 and 8 , which illustrate the establishment of a voice call in a third embodiment of the invention.  
         [0146]      FIG. 7  illustrates an examplary network architecture for the third embodiment of the invention. In the example of the third embodiment, it is assumed that the user equipment is connected in a visited network. Referring to  FIG. 7 , a UE  802  is connected in to the visited network  804  via an air interface connection to a radio network controller (RNC)  806 . The visited network includes a plurality of GPRS support nodes (GSNs)  808 , each of which may include a SGSN and a GGSN (not shown). The RNC  806  connects the UE  802  to a selected one of the GSNs  808  when a call is established.  
         [0147]      FIG. 7  shows a proxy call state control function (P-CSCF)  810 , which controls the call state of the call to/from the UE  802  in the visited network  804 . The P-CSCF  810  therefore connects to the one of the GSNs  808  supporting the call to/from the UE  808 . The P-CSCF  810  in the visited network is connected to a serving call state control function (S-CSCF)  812  in a home network  814  with which the UE  802  is normally connected.  
         [0148]     In the home network  814 , the S-CSCF  812  is connected to a home subscriber server (HSS)  816  and an application server  818 . The S-CSCF  812  further connects in the home network  814  to a MRFC  820 , which in turn connects to a MRFP  822 .  
         [0149]     The S-CSCF  812  connects to a breakout gateway control function (BGCF)  824  in the home network, which connects to a media gateway control function (MGCF)  826  in the visited network. In accordance with this third embodiment of the invention, the MGCF  826  further connects to the RNC  806 . The MGCF  826  also connects to a circuit switched/IP multimedia sub system media gateway (CS/IMS MGW)  828 .  
         [0150]     The S-CSCF  812  of the home network further connects to other public land mobile networks (PLMNs)  830 , or external networks. The invention is described with reference to an example where a call is established between the UE  402  and a terminal connected to a PLMN  830 .  
         [0151]     The network illustrated in  FIG. 7  is a typical UMTS network arrangement as will be familiar to one skilled in the art. The arrangement is adapted in accordance with this third embodiment of the invention to provide the connection between the RNC  806  and the MGCF  826  as further described hereinbelow.  
         [0152]     The PLMN  830  may include the Internet or other communications networks in this embodiment. Various user equipment or terminals may be connected to the PLMN  830  such as mobile terminals, SIP enabled devices and personal computers.  
         [0153]     This third embodiment of the invention involves the use of a single call control mechanism, such as a SIP based call control mechanism, to establish both packet switched and circuit switched calls. This embodiment is now further described with reference to the message flow diagram of  FIG. 8 .  
         [0154]     In  FIG. 8 , for simplicity the message flow is shown as directly to the S-CSCF  812 , although in practice it would be via the P-CSCF  810 . A terminating network (to which a call is established with the UE  802 ) is denoted  830   n , being one of the PLMNs  830 .  
         [0155]     The UE attempts to establish a voice call to a party connected in a terminating network  830   n . The UE and the visited network  804  establish a PDP context for the voice call as represented by bi-directional signalling  902 . This signalling takes place before the establishment of a voice call and may be required to configure the various network elements and establish the data bearer for the voice call. The signalling may be based on SIP signalling and messages. Other suitable protocols may be used such as MEGACO, also known as H.248.  
         [0156]     During the bi-directional signalling, the IP address and port number of the CS MGW  828  may be transmitted to the UE. The IP address and port number of the CS MGW  828  may be determined in a discovery procedure similar to existing discovery procedures for determining the IP address and port number of a P-CSCF. The IP address and port number may be used by the terminating network  830   n  to direct voice data to the appropriate CS MGW, which can then be transmitted to the UE, rather than directly to the UE.  
         [0157]     The UE  802  then transmits an SIP INVITE message  904  to the S-CSCF  812 . The SIP INVITE message may include the IP address of the CS MGW obtained during the bi-directional signalling. This message may be routed via the P-CSCF  810  or transmitted directly to the S-CSCF  812 . Upon receipt of the SIP INVITE message  804 , the S-CSCF  812  performs SIP URI (universal resource indicator) address analysis in order to determine the destination of the call to be established.  
         [0158]     Once this destination is established, the S-CSCF forwards an SIP INVITE message  906 , which includes the IP address of the CS MGW, to the terminating network  830   n . The terminating network  830   n  may be, for example, one of: the same network, another network (PLMN), a PSTN, or the Internet.  
         [0159]     As represented by messages  908  and  910 , sequential SIP signalling occurs. Specifically, in message  908 , the terminating network  830   n  transmits a SIP  183  Session Progress message to the S-CSCF  812 .  
         [0160]     The terminating network  830   n  transmits a SIP  180  Ringing message  910  to the S-CSCF  812 , and then transmits the SIP  2000 K message  912  to the S-CSCF  812 .  
         [0161]     Responsive to the SIP  2000 K message  912 , the S-CSCF  812  returns a SIP  2000 K message  914  to the UE  802 . The UE  802  acknowledges the SIP  200  OK message  914  by transmitting a SIP ACK message  916  back to the S-CSCF  812 .  
         [0162]     In response to receiving the SIP ACK message  916 , the S-CSCF  812  transmits a SIP INVITE message  918  to the MGCF  826  that controls the RNC  806  serving the UE  802 . The SIP INVITE message  918  may provide all the information required by the MGCF  826  for initiating a RAB (radio access bearer) assignment procedure. The RAB assignment procedure is used to establish an Iu circuit switched (lu-CS) connection or data bearer between the RNC  806  and the CS MGW  828  in accordance with this embodiment of the invention.  
         [0163]     The MGCF  826  transmits a RAB Assignment Request message  920  to the RNC  806  serving the UE  802  with the appropriate parameters. The RNC  806  responds by transmitting an RAB Assignment Response (Successful) message  922  to the MGCF  826 . The response message may also include a RAB identifier and other parameters, such as transport layer information and the cause of failure if the RAB assignment unsuccessful.  
         [0164]     Whilst the RAB is being established between the RNC  806  and the CS MGW  828 , the CS MGW  828  also establishes a channel between the CS MGW  828  and the terminating network  830   n . The MGCF  826  transmits a H.248 Channel Setup message  919  to the CS MGW  828 .  
         [0165]     H.248 is an ITU-T standard, known as MEGACO under IETF. It is a protocol used between elements of a physically decomposed multimedia gateway e.g. a MGW and a MGCF, for the MGCF to tell the MGW when and how to establish a media channel for a call, and for the MGW to notify the MGCF of the status of the setup.  
         [0166]     The CS MGW  828  establishes a channel between the CS MGW and the terminating network  830   n  and transmits a H.248 Channel Setup Successful message  924  back to the MGCF  826 .  
         [0167]     Channel establishment is now complete at the CS MGW  828 . The MGCF transmits a SIP  2000 K message  926  to the S-CSCF  812  informing it of successful data bearer establishment. The S-CSCF  812  then transmits a SIP ACK message  928  to the terminating network  830   n  to activate the data bearer for the voice call.  
         [0168]     The voice call may then take place between the UE  802  and the terminating network  830   n . With reference to  FIG. 7 , the lines joining the UE  802  to the RNC  806 , the RNC  806  to the CS MGW  828  and the CS MGW  828  to the other PLMNs  830  represent the data bearer path for the voice call.  
         [0169]     In the example of this third embodiment, the voice data is transmitted as a circuit switched encoded datastream for the entire data bearer.  
         [0170]     The terminating network may use the IP address and port number of the CS MGW received during call establishment to transmit voice data to the CS MGW, which can then route the voice data onto the UE. Thus, a circuit switched encoded datastream can be maintained for the entire data bearer.  
         [0171]     This method is used for both circuit switched and packet switched based calls. There is no requirement for conversion, other than at the boundary between the PLMN and Internet, in case one end is a SIP device on the Internet, then circuit switched to packet switched VoIP conversion is needed.  
         [0172]     Three embodiments have thus been described for establishing—in preferred arrangements—a voice call in the circuit switched domain over the air interface, even when the call is routed in the packet switched domain in the core network. The first embodiment utilises SIP signalling to achieve this, the second embodiment utilises circuit switched call control to achieve this, and the third embodiment utilises SIP to establish all calls in the circuit switched domain. Thus the same control mechanism is used whether the core network carries the call in the circuit or packet switched domains.  
         [0173]     In the implementation of the third embodiment, session initiation protocol (SIP) is used as the call control method for both circuit switched and packet switched voice calls. The data bearers for the voice call are entirely in the circuit switched domain. The technique of this embodiment may be used in order for SIP to replace the GSM call control mechanism for a circuit switched call. Thus only one call control method is required for both circuit switched and packet switched calls.  
         [0174]     The above described methods result in several advantages over prior art methods.  
         [0175]     When SIP signalling is used as the call control mechanism to establish all voice calls, any suitably configured SIP device such as a mobile terminal or a laptop can readily make voice calls. Furthermore, the voice calls are also more efficient in their use of network resources than previous VoIP calls, as the existing circuit switched air interface is utilised in the transmission of the voice call without the need for packet switched overheads such as data packet headers. It is advantageous to reduce the data transmission over the air interface whenever possible due to capacity and cost restrictions of data transmission over the air interface. This may also increase capacity in the network and promote faster adoption of VoIP.  
         [0176]     Furthermore, the methods described above in embodiments do not require any compression or header removal techniques that have previously been suggested to reduce the data that needs to be transmitted over the air interface in a PS datastream. This makes the methods simpler to implement and cheaper to operate.  
         [0177]     By replacing previously separate circuit switched and packet switched call control mechanisms with a single mechanism such as one based on SIP described above, the call control protocol stacks that need to be employed in the UE may also be reduced, thus saving development costs and memory at the UE.  
         [0178]     Another significant advantage is that if both circuit switched and packet switched voice calls are handled in the manner as VoIP calls, then it may be possible to remove the MSC server present in existing circuit switched networks and save costs.  
         [0179]     Herein reference is made to the packet switched domain and the circuit switched domain. More generally, reference can be made to a first domain and a second domain, each of which domain carries or transports a respective first and second type of datastream. The first and second domains may alternatively be referred to as first and second transport platforms or transport mechanisms, being respective platforms or mechanisms for first and second datastreams.  
         [0180]     It is also noted herein that while the above describes exemplifying embodiments of the invention, there are several variations and modifications which may be made to the described embodiments without departing from the scope of the present invention as defined in the appended claims. One skilled in the art will recognise modifications to the described embodiments.