Completion of fax and analog data call

Technical field of the invention

The invention relates to a method, a session control en ^¬ tity, and a computer program product for modifying a speech call to a fax or data call in a communication network .

Background of the invention

Despite the telecommunication technology is moving to ^¬ wards Internet Protocol (IP) based solutions and con ^¬ verged networks, the role of analogue modem data and es- pecially facsimile telephony is still assumed to be im ^¬ portant in the future due to the fact that in existing fixed telecom equipments are connected to an IP Multime ^¬ dia Subsystem (IMS) defined by 3rd Generation Partnership Project (3GPP) via intelligent access device (IAD) or analogue telephone adapter (ATA) like adapter equip ^¬ ment. An ATA is an adapter that is used to connect a standard telephone to a high-speed modem to facilitate voice over IP (VoIP) and fax calls over the Internet.

Additionally facsimile has special legal status as for ^¬ mal undersigning or signature in some countries, which therefore need to be offered by service providers. Hence, facsimile service is expected to be needed also in IMS networks that are Session Initiation Protocol (SIP) based and provide telephony services also for fixed subscribers. SIP is defined by Internet Engineer ^¬ ing Task Force (IETF) and it is an application-layer control protocol for creating, modifying, and terminat ^¬ ing sessions with one or more participants. These ses-

sions may include Internet multimedia conferences, Internet telephone calls, and multimedia distribution. Members in a session can communicate via multicast or via a mesh of unicast relations, or a combination of these. A call state control function (CSCF) performs call control functions, service switching functions, ad ^¬ dress translation functions, and vocoder negotiation functions in the IMS using SIP.

A media gateway control function (MGCF) acts as an in- terworking point between a circuit switched (CS) network and an IP network in the control plane of the network. The MGCF controls the parts of the call state related to connection control for media channels in a media gateway (MGW) , communicates with call state control, and per ^¬ forms protocol conversion between the call control pro ^¬ tocols, such as SIP and integrated services digital net ^¬ work (ISDN) user part (ISUP) . ISUP is a call control component of the signaling system No. 7 (SS7) protocol that determines the procedures for setting up, coordi ^¬ nating, and taking down trunk calls in the SS7 network. A network element acting as MGCF is responsible for sig ^¬ nalling level interworking between CS and IMS domains. A mobile switching center (MSC) is a mobile network ele- ment in the CS domain which performs the switching func ^¬ tions in its area of operation and controls the inter ^¬ working with other networks. A home location register (HLR) is a register of a CS mobile network where all subscriber parameters of a mobile subscriber are perma- nently stored.

European Telecommunications Standards Institute (ETSI) Telecommunications and Internet converged Services and Protocols for Advanced Networking (TISPAN) has defined

number of different ways to support facsimile within IMS architecture. Such mechanisms are e.g. International Telecommunication Union (ITU) - Telecommunication Standardization Sector (ITU-T) T.38 (Real time facsimile service), ITU-T V.152 as well as support for ITU-T T.30 in-band over G.711 codec. All mechanisms are supported in varying degree within different equipments and solu ^¬ tions. Importance of T.38 real-time facsimile has been recently raised due to the fact that it will guarantee sending and receiving of facsimile documents in poor quality of service (QoS) conditions, which is clear benefit over other possible mechanisms.

Analogue data (modem) similarly also can use ITU-T V.152, ITU-T V.151 (Modem over IP) or even in-band transfer of data over G.711 codec. So far analogue fac ^¬ simile and modem services have not been included in the IMS by the 3GPP.

There is a problem if an analogue data or facsimile te ^¬ lephony call is made by the IMS user towards CS mobile subscriber by using a phone number that cannot identify the real nature of call, i.e. when single numbering scheme is in use within mobile CS network, and the IMS user wants to modify the session on fly, e.g. switch from speech call to fax during the session.

The object of the invention is to overcome the above drawbacks .

Summary of the invention

The invention solves the problem of prior art by provid ^¬ ing a method and a session control entity, comprising

controlling a first session comprising a first media type towards a first user by means of a control protocol of a packet switched network, controlling the first ses ^¬ sion towards a second user by means of a control proto- col of a circuit switched network, receiving a request from the first user to modify the first session by changing media type from the first media to a second me ^¬ dia type, releasing the first session towards the second user only, establishing a new session comprising the second media type towards the second user, modifying the first session towards the first user by changing the me ^¬ dia type from the first media type to the second media type, and associating the new session towards the first user with the modified first session towards the second user.

The method or session control entity may further include controlling conversion of the first media type and the second media type between the packet switched network and the circuit switched network. The controlling the conversion of the first media type and the second media type may comprise controlling a media gateway entity. The method or session control entity may further com ^¬ prise charging the first session and the new session, and may comprise charging the first session and the new session as a single session. The control protocol of the packet switched network may be a session initiation pro ^¬ tocol (SIP) . The control protocol of the circuit switched network may be an integrated services digital network (ISDN) user part (ISUP) . The request to modify the first session may be a re-INVITE or an UPDATE request of a session initiation protocol (SIP) . The first media type may be for example a speech call, a facsimile or an analogue data call, and the second media type may

be another one of the speech call, the facsimile and the analogue data call. The request to modify the first session may be a request to change the media type to T.38 fax. Establishing the new session towards the sec- ond user may comprise indicating group 3 fax in a high layer compatibility (HLC) parameter of an initial ad ^¬ dress message (IAM) of integrated services digital net ^¬ work (ISDN) user part (ISUP) .

In the method or the session control entity, controlling the first session may comprise establishing the session originating from the packet switched network and terminating to the circuit switched network, or establishing the session originating from the circuit switched net- work and terminating to the packet switched network.

The present invention has the advantage that it provides a solution for routing a call originated by IMS sub ^¬ scriber as a voice-only session and dynamically upgrad- ing the call into corresponding data call without re ^¬ initiating the call by end users.

Description of drawings

Figure 1 presents an overview of prior art network architecture relevant for this invention.

Figure 2 presents a signaling flow of an embodiment of the invention.

Figure 3 illustrates an internal structure and functions of a session control element implementing aspects of the invention .

Detailed description of the invention

Figure 1 presents an overview of prior art network architecture relevant for this invention. A first terminal device 1 is located in a packet switched communication network 8. The first terminal device 1 may be a mobile or fixed line terminal and supports communication using at least one packet switched control protocol, such as SIP, for establishing and controlling communication ses- sions 11. The packet switched communication network 8 may be for example, an IMS network or another internet or SIP based network, and a user of the first terminal device 1 may be a subscriber to the communication network 8.

A second terminal device 7 is located in a circuit switched communication network 9. The second terminal device 7 may be a mobile or fixed line terminal and sup ^¬ ports communication using circuit switched control pro- tocol, such as ISUP, for establishing and controlling communication sessions 11. The circuit switched communi ^¬ cation network 9 may be for example, a public switched telephone network (PSTN) or a public land mobile network (PLMN) , and a user of the second terminal device 7 may be a subscriber to the communication network 9.

An MGCF 3 is interfacing the packet switched communica ^¬ tion network 8 and the circuit switched communication network 9, and is implementing protocol conversion be- tween control protocols of the communication networks 8 and 9, for example between SIP and ISUP, so that it is possible to establish communication 11 between the first terminal device 1 and the second terminal device 7 even though the first terminal device 1 and the second termi-

nal device 7 do not support the same (single) protocol for controlling the communication 11. The MGCF 3 is further communicating with a media gateway (MGW) 10, using an appropriate media gateway control protocol (MEGACO, MGCP) or H.248 protocol specified by ITU-T. These proto ^¬ cols are used to support separation of call control en ^¬ tities (e.g. MGCF, MSC) from user plane entities (e.g. MGW) . The user plane is a layer in which user data (e.g. modem or fax data) or speech coming from e.g. a radio network controller is converted to pulse code modulation (PCM) user data and vice versa. Tasks of the user plane are e.g. processing transcoding of speech channels and data format conversion for data. The user plane traffic 12 associated with control plane communication 11 is routed through the MGW 10 between the packet switched communication network 8 and the circuit switched commu ^¬ nication network 9. Only the network elements relevant for interworking between CS and PS session are shown in Fig 1. In practice there are many other network elements (e.g. MSC, CSCF) involved in both control path 11 and user plane 12 but those are omitted for simplicity.

Various ways exist to initiate analogue data calls, i.e. modem or facsimile, by an IMS subscriber. A facsimile or modem call may be initiated so that it is immediately known or recognized to be a data call. In this case, a session description protocol (SDP) within an INVITE message of SIP protocol may contain directly information about the nature of the call e.g. about T.38, and no speech codec.

Alternatively, a facsimile call may also be initiated at first by a voice call (using e.g. G.729 codec) but then after a called party has answered, the call may be modi-

fied to be the facsimile call by sending re-INVITE re ^¬ quest with a new session description. Also other ways exist, such as for instance both voice and facsimile (T.38) is negotiated from the beginning of a call and then both may be used without any further codec modifi ^¬ cation but that is less common way of behavior in prac ^¬ tical implementations.

When a facsimile or analogue data call is completed it may be possible that already during a call establishment phase it is known that the call is an analogue data call. User equipment (UE) of calling IMS subscriber sends an initial INVITE request and may include a re ^¬ quest for facsimile service in an SDP offer, for example request for T.38 transfer. The called subscriber may be a circuit switched subscriber and the call is routed to an MGCF by using standardized IMS procedures. The MGCF examines the received SDP offer and if it supports re ^¬ quested media type (e.g. T.38) then the MGCF will con- tinue a normal call establishment towards the CS net ^¬ work. The MGCF in this case sends an Initial Address Message (IAM) with Transmission Medium Requirement (TMR) parameter indicating 3.1 kHz audio as well as High Layer Compatibility (HLC) parameter encoded with information stating that the call is Group 3 Fax call. Also called party' s address (B number) is sent by the MGCF towards CS network. The CS network may use either both the called party number (B number) information or received HLC parameter to determine that the call is a facsimile or an analogue modem call. IP Multimedia media Gateway (IM-MGW) handles required interworking in user plane (media) between the IMS and the CS domains.

Basic service group Basic service (1) (2)

NOTE 1: Basic services are defined in 3GPP TS 22.002 [3] and 3GPP TS

22.003 [4] .

NOTE 2: TS: Teleservice BS: Bearer service. NOTE 3: The transparent facsimile services apply to A/Gb mode of GERAN only. The non-transparent facsimile services apply to UTRAN only. NOTE 4: The Voice Group Services, TS 91 and TS 92 apply to A/Gb mode of GERAN only.

The above table shows basic service groups defined by the 3GPP in the specification TS 22.004 "General on sup ^¬ plementary services".

According to the invention a model is provided that may be implemented and standardized to support also calls or sessions that begin by using a voice codec (speech call) but are afterwards switched into e.g. T.38 real-time

facsimile call. An example call flow is seen in Figure 2.

UE 1 of a calling IMS user sends an initial INVITE 201 request indicating only voice codecs in an SDP offer and the call is routed (possible via call state control function CSCF 2) to an MGCF 3. Then normal IMS-CS inter- working procedures apply so far. The MGCF 3 is interfac ^¬ ing a packet switched domain (SIP UE 1, CSCF 2) and a circuit switched domain (gateway-MSC 4, HLR 5, Visited- MSC (VMSC) 6, CS UE 7) and based on the information re ^¬ ceived in the INVITE request, the MGCF 3 sends an IAM 202 message indicating 3.1 kHz audio call towards the called party. The gateway-MSC 4, after receiving the IAM 202 message, queries 203 the HLR 5. The HLR enquiry 203 is resulting the selection of TSIl service (speech) . The gateway-MSC 4 forwards the IAM 204 to the VMSC 6 which completes the call signaling to the CS UE 7 in step 205. The call is also completed at user plane 206. For sim- plicity, in Fig 2 example, part of the message flow is not shown, e.g. the CS/PS answer message flow from the CS UE 7 towards the SIP UE 1. Also there may exist addi ^¬ tional network elements involved in the signaling path, e.g. further CSCFs, border gateway control function (BGCF) etc.

It is to be noted that in this example the called party (CS UE 7) is a mobile subscriber, however, the called party may equally well be a fixed line subscriber, in which case the mobile specific parts of the CS network (gateway-MSC 4, HLR 5, VMSC 6) are to be replaced by corresponding fixed network elements, e.g. by telecommu ^¬ nication switches.

After the called subscriber, i.e. the user of CS UE 7, has answered the call and possibly talked with the call ^¬ ing subscriber, i.e. the user of SIP UE 1, it is possi ^¬ ble that the calling subscriber wants to initiate a fax transfer by using the same IMS session to the called subscriber. Traditionally this would require that both subscribers clear the ongoing call and the calling party establish a new call, with fax characteristics for the fax transmission, unless alternative speech & fax ser- vice (TS61) is available in the network. This invention does not require alternative fax & speech to be sup ^¬ ported which service also in practice is not available in e.g. CS mobile networks or supported by terminals.

According to the invention, the SIP UE 1 sends a re-

INVITE 207 request to the same SIP dialogue that has al ^¬ ready been established towards the CS UE 7. The MGCF 3, after receiving the re-INVITE 207, detects is step 208 that a media type (here fax) in the re-INVITE is differ- ent from the originally negotiated (here speech) and thereby generates a normal call clearing in release 209 message towards the CS UE 7 and a re-establishment of a new facsimile call towards the CS UE 7 with a new IAM 211 by using HLC indicating Group 3 Fax and using the same called party address that was used in the original call attempt, i.e. INVITE 201. In step 212 the HLR en ^¬ quiry results in selection of TS61 service (facsimile) . Steps 213 and 214 complete the call establishment for the new call leg and corresponds to steps 204 and 205. In step 215 the CS UE 7 responds towards the SIP UE 1 by sending CONNECT message which is transmitted to the MGCF 3 in ISUP as ANSWER or CONNECT message in step 216. The MGCF 3 may then determine that the establishment of the new call leg to the CS UE 7 was successful and acknowl-

edges the re-INVITE 207 with a 200 OK SIP response in step 217. The response may include information confirm ^¬ ing the change of the media or the service, e.g. T.38 indication. This way the CS UE 7 see the incoming fax call as a new call coming from the same calling party, however from SIP UE 1 point of view the same session has been modified to facsimile call. The MGCF 3 handles the logic on behalf of the IMS subscriber.

According to aspects of the invention a new capability for the MGCF is introduced to modify a CS side connec ^¬ tion by clearing an old speech connection and establishing a new facsimile call in order to create better end user experience for the IMS subscriber. This procedure may be repeated more than one time during the same IMS session, meaning that the IMS subscriber could after a facsimile transfer continue the call as a voice call again, which require that the MGCF clears the fax call and re-establishes a new speech call towards the CS sub- scriber. The procedure does not use alternative fax & speech service (TS61) but may be used and deployed with more traditional facsimile services (e.g. G3 fax) avail ^¬ able in today's CS mobile networks.

In one aspect of the invention, a timer is introduced in an MGCF and the MGCF is trying to establish a new call leg only after a certain time has expired. This may be required because termination of the first connection to ^¬ wards the CS UE may take time, meaning that if the MGCF would establish the new session immediately, the MGCF may get busy tone.

An SIP UE may try to update or upgrade a session, no matter which party initiated the session. This means

that even if a CS UE has initiated the call/session, the IMS party may send a re-INVITE to swap from speech to fax or vice versa, and an MGCF may release the existing CS call and try to establish a new fax call to the CS UE. Typically both parties (end users) know when they want to switch from one service to the other service, therefore it should not matter which party initiated the call/session originally. The charged party may then change. However to be fair in charging, the situation, (e.g. which party initiated the original session) may be indicated in the charging data records (CDR) and thereby the originally calling party may be charged for the whole call/session.

Figure 3 illustrates the internal structure and func ^¬ tions of a session control element, e.g. MGCF, imple ^¬ menting this invention. The MGCF 3 includes first ses ^¬ sion control unit 31 configured to control a call/session with a first media type (e.g. speech) to- wards a SIP UE 1. The first session control unit 31 may control the call/session by means of a control protocol of a packet switched network, such as SIP. A second ses ^¬ sion control unit 32 is configured to control the call/session towards a CS UE 7. The second session con- trol unit 32 may control the call/session by means of a control protocol of a circuit switched network, such as ISUP. The first and the second session control units 31 and 32 may control both originating and terminating sessions. A receiving unit 33 is configured to receive, during the active call/session, a request to modify the call/session by changing the media type or a service from the current media/service (e.g. speech, facsimile, analog data call) to another media/service (e.g. speech, facsimile, analog data call) . The receiving unit 33 may

receive the request in a SIP re-INVITE or in a SIP UPDATE requests. The receiving unit 33 may first check with the session control units 31, 32 which media or service is currently used. The receiving unit 33 and the session control units 31, 32 may determine whether change of media or service is possible for the call/session. A releasing unit 34 may be configured to release the call/session towards the CS UE 7 only if the change of media or service is supported. The releasing unit 34 may release the call/session by sending a REL message of ISUP. An establishing unit 35 is configured to establish a new call/session with the other media type or service towards the CS UE 7. A modifying unit 36 is configured to modify the remaining part of the origi- nal call/session towards the SIP UE 1. The modifying unit 36 may change the media type or service from the original media type or service (controlled by the first session control unit 31) to the other media type or ser ^¬ vice indicated in the request received by the receiving unit 33. The modifying unit 36 may send an indication of the changed/new media type or service to the SIP UE 1 in a 200 OK SIP response. An associating unit 37 is configured to associate the new call/session towards the CS UE 7 with the modified call/session towards the SIP UE 1. A controlling unit 38 may control conversion of a media stream of the first media and/or a media stream of the second media between the packet switched network and the circuit switched network. The controlling unit 38 may control a media gateway entity 10 through which the stream of the first and the second media type traverse at a user plane. The controlling may be implemented us ^¬ ing for example MGCP, MEGACO or H.248 protocol. Addi ^¬ tionally the MGCF 3 may have a charging unit 39a that is collecting charging related information for communica-

tion controlled by the MGCF 3. The charging unit 39a may charge the first session and the second session as a single session. If the first session is initiated by the CS UE 7, a user of the CS UE 7 may be charged for the whole session if the SIP UE 1 initiates the change of media type and the new call/session towards the CS UE 7 is established by the establishing unit 35. A charged party may be changed by the charging unit 39a when the type of media is changed so that the party initiating the change of media type will be charged. Charging tar ^¬ iff or rate may be changed according to a media type in use. A monitoring unit 39b may be configured to monitor that the CS connection is fully released before the es ^¬ tablishing unit 35 is trying to establish the new call/session. The monitoring unit 39b may be implemented as a timer and the establishing unit 35 may start estab ^¬ lishing the new connection only after the timer has expired.

All units described above may be implemented e.g using microprocessors and/or other electrical components and/or by software.

The invention is not limited to mobile networks, IMS networks and to interworking of them, but may also be applied in other type of networks having similar session gateway entity role as MGCF and supporting plurality of communication services to enable switching from one service to another service during the session. Therefore, the MGCF is only used here as an example of a session control entity. Functions of the session control entity (MGCF) described above may be implemented by code means, as software, and loaded into memory of a computer.