BE1021396B1 - IMPROVEMENTS IN VOICE QUALITY CONTROL - Google Patents

Abstract

The invention described here relates to a multimode terminal system comprising at least an internet network (540, 545), a mobile network (560), a platform (505) and a multimode terminal (575). The platform (505) and the terminal (575) can both be connected to one of the internet networks (540, 545) and to the mobile network (560). A voice quality control module (525) is provided to determine whether the call voice quality of a call made to or from the terminal (575) via the Internet network (540, 545 ) meets a predetermined minimum voice quality level. The handover between the internet network (540, 545) and the mobile network (560) is initiated when the voice quality of the call drops below the predetermined minimum voice quality level.

Description

IMPROVEMENTS IN QUALITY CONTROL

VOICE

The present invention relates to improvements in voice quality control and more specifically, although not exclusively, relates to the quality of voice calls for the transfer between modes in multimode telecommunications systems.

Voice communications have recently made rapid progress in two parallel areas. The first area is cellular telephony, which has increased the mobility of telephony users. A second, and more recent, area is communications using the Voice Over Internet Protocol (VoIP), in which voice communications are routed over IP networks. Although cellular telephony and VoIP communications have both led to undeniable advantages, their parallel development has also led to certain disadvantages for a user wishing to exploit them simultaneously.

Dual-mode terminals that can be independently connected to both a Public Switched Telephone Network (PSTN) and Internet Protocol (IP) networks have been developed. In particular, two-mode terminals of this type have been developed, including both a mobile telephone transceiver for wirelessly connecting the dual-mode terminal to a cellular PSTN network, such as for example a GSM network (Global Telecommunications System for mobile phones). Mobiles), or a Universal Mobile Telecommunications System (UMTS), and a wireless Local Area Network (LAN) transceiver for connections to an IP network via a wireless LAN. The existence of these two-mode terminals, associated with the growing popularity of VoIP, led to the appearance of the concept of Convergence

Fixed-Mobile Convergence (FMC), providing a single communications system encompassing both a cellular PSTN and an IP network, so that the two-mode terminal can continue to communicate with this communication system using one or the other of its connections.

WO-A-00/79814 discloses a communication system of this type comprising both a PSTN and an IP network. In this communications system, a first of the networks includes a home location register on which the dual mode terminal is registered, and the second network includes a server location register. When the two-mode terminal connects to this second network, the server location register sends the updated location information to the first network.

Document US-A-2005/0096024 discloses a communication system similar to that of WO-A-00-079184, wherein, when the two-mode terminal connects to the IP network, the IP network must transmit its IP address to an IP network. PSTN network location register.

Document US-A-2009/0131045 discloses another communication system comprising a PSTN network and an IP network. In this system, incoming calls to the dual-mode terminal are always forcibly routed through the IP network. If the dual mode terminal is not connected to the IP network at this time they are forwarded from the IP network to the PSTN network. Although this system no longer requires an IP network location update to be sent to the PSTN network each time the two-mode terminal connects to or disconnects from the IP network, a considerable bandwidth is required between the two networks, especially when the two-mode terminal is not connected to the IP network. In this case, the call must be transmitted from the PSTN network to the IP network and then sent back to the PSTN network.

The problem of transferring a call in progress when a dual-mode terminal changes from an IP network to a PSTN network has, however, been described in numerous documents, for example by the 3GPP ™ project in its Technical Specification 3GPP TS 23.206. However, to perform call forwarding with voice call continuity, this technical specification also requires additional resources in the form of an IP Multimedia Subsystem (IMS).

Document US-A-2006/0198360 discloses a communication system and method for transferring a call when a two-mode terminal goes from an IP network to a PSTN network. The IP network then comprises a call manager that issues a continuity call to the two-mode terminal via the PSTN network if the connection via the IP network is degraded.

EP-A-2018014 discloses a communications system and method according to which a continuity call is issued by a dual-mode handset that can connect to both an IP network and a fixed network, either manually or automatically, if the connection through the IP network is degraded. A Private Branch Exchange (PBX) is then connected to the outside world through an Integrated Services Digital Network (ISDN) and all calls are routed through that network. . When a call is to be transferred between an IP network and a fixed network, the handset initiates a call waiting (the current call between the handset and a remote device) by sending a message to the switch. The switch forwards the message to the remote device. A second call is established between the handset and the switch by means of a third call that is established between the switch and the remote device. The first call is interrupted when the connection between the handset and the remote device has been routed through the second and third calls.

US-A-2006/0121902 and US-A-2008/0273505 also disclose communication systems and methods for transferring a call when a two-mode terminal goes from an IP network to a PSTN.

It is therefore an object of the present invention to provide a system and method for determining a transfer between networks in a multimode terminal system based on the voice quality of the call on one of the networks.

According to a first aspect of the present invention, this relates to a multimode communication system comprising at least one internet protocol network, at least one mobile telecommunications network, a telecommunications platform and at least one multimode terminal, said communication platform. telecommunications and said at least one multimode terminal can both be connected to said at least one internet protocol network and said mobile telecommunications network, said system comprising: at least one voice continuity control module for providing call continuity for a call involving at least one multimode terminal when connected to said at least one internet protocol network; characterized in that said system further comprises at least one voice quality control module for determining the voice quality of the call of a call involving said at least one multimode terminal when connected to said at least one internet protocol network; and in that said at least one voice continuity control module has the function of initiating a transfer between said at least one internet protocol network and said at least one mobile telecommunications network according to a predetermined minimum speech quality level. determined by said at least one voice quality control module. The voice continuity control module advantageously consists of two components respectively residing on the telecommunication platform and the multimode terminal and cooperating entre'eux. The voice quality control module is also advantageously constituted by two components residing respectively on the telecommunication platform and on the multimode terminal and cooperating with each other.

The present invention may more particularly apply to communications systems that operate in association with a mobile-to-mobile convergence (MMC) platform. In such a system, the voice quality can be determined by both the MMC platform and a multimode terminal including a MMC application and can be adapted according to specific requirements of a subscriber or operator of the system.

The system further comprises a call manager associated with said at least one voice continuity control module, said call manager having the function of triggering a transfer between said at least one internet protocol network and said at least one telecommunications network. mobile.

The predetermined level of minimum voice quality may be determined depending on the availability of the internet protocol, the availability of the mobile telecommunications network and / or a real-time transport protocol associated with the voice traffic of an ongoing call which is provided by a real-time transport control protocol module.

The term "availability" as used herein refers to the ability to detect the presence of the network concerned, the determination of whether it is possible to connect to the network concerned, and whether the network concerned is responding. the operating criteria in the system, for example whether the signal strength is at a level sufficient to ensure a satisfactory connection.

A real-time protocol analyzer module may also provide real-time protocol data relating to the voice traffic of the call, the predetermined minimum speech quality level being determined in accordance with the real-time protocol. A database containing data relating to at least one user of the system may also be provided, the predetermined minimum speech quality level being determined in accordance with said user data. Multimode terminal data relating to the terminals associated with the system may also be used to determine the predetermined minimum speech quality.

In one embodiment, the platform includes a first voice quality control module and a first voice continuity control module and the multimode terminal associated with the platform includes a second voice quality control module and a second voice quality control module. voice continuity, said pre-defined minimal speech quality of an ongoing call being determined by at least one of said first and second voice quality control modules and said handoff being triggered by at least said first voice continuity control module.

According to another aspect of the present invention, this relates to a multimode terminal for use in a multimode terminal system such as that described above, wherein said terminal comprises a voice quality control module having a speech manager. an internet protocol for detecting the availability of said at least one internet protocol network and determining whether the voice quality meets said predetermined minimum speech quality associated with said at least one internet protocol network and triggering a transfer between said at least one protocol network internet and said mobile telecommunications network.

It is preferable that the multimode terminal connects to an internet protocol network in preference to a mobile telecommunications network. The terminal initiates a transfer of said internet protocol network to the mobile telecommunications network when said voice quality of the IP network falls below a predetermined minimum voice quality, the transfer being triggered according to the availability of the internet protocol network and / or the availability of the mobile telecommunications network.

It is preferred that the multimode terminal include a voice continuity control module for initiating a handover between said at least one internet protocol network and said mobile telecommunications network when said voice quality falls below said predetermined minimum speech quality. . The multimode terminal may further include a real-time transport control protocol module constituting a real-time transport control protocol for the voice traffic of said call and / or a real-time protocol analyzer module providing real-time protocol data relating to the voice traffic of said call, said predetermined minimum speech quality level being determined in accordance with said real-time transport control protocol and / or said real-time protocol.

According to another aspect of the present invention, this relates to a method for implementing a transfer of a voice call between an internet protocol network and a mobile telecommunications network in a multimode terminal system, said system having least one internet protocol network, at least one mobile telecommunications network, one telecommunications platform and at least one multimode terminal, the telecommunications platform and said at least one multimode terminal both being connectable to said at least one internet protocol network and auditing at least one mobile telecommunications network, said method comprising the steps of: a) assigning a predetermined minimum speech quality level to a voice call connected to said at least one internet protocol network; b) determining the voice quality of the call for said at least one internet protocol network; c) comparing said voice quality of the determined call with said predetermined minimum speech quality level; and d) initiating a transfer of said voice call between said at least one internet protocol network and said at least one mobile telecommunications network when said voice quality of the determined call falls below said predetermined minimum speech quality level.

It is preferable that step d) consists in triggering a transfer according to the coverage of the internet protocol network and / or the availability of the mobile telecommunications network. Step a) may consist in assigning said predetermined minimum speech quality level according to one of: user data; multi-mode terminal data; voice traffic with real-time protocol; and real-time transport control protocol information.

In addition, step a) may consist in establishing a voice call via an internet protocol network in preference to a mobile telecommunications network.

For a better understanding of the present invention, reference will now be made, by way of example, to the accompanying drawings, in which: FIG. 1 illustrates a communications system in which a call can be transferred between an IP network and a network GSM; Figure 2 illustrates the routing of an incoming call over the IP network when the dual-mode terminal is connected to the IP network in the communications system shown in Figure 1; Fig. 3 illustrates the routing of an incoming call through the PSTN to the dual mode network when the dual mode terminal is not connected to the IP network or does not respond in the communications system shown in Fig. 1; FIGS. 4a and 4b illustrate a rerouting via the PSTN network of a call in progress if the connection of the two-mode terminal to the IP network is degraded; Figure 5 illustrates an MMC architecture according to the present invention; Figure 6 illustrates a speech quality control architecture of the MMC call in accordance with the present invention; Fig. 7 illustrates an incoming call for a two-mode terminal when the call is routed over an IP network; Figure 8 illustrates an incoming call for a two-mode terminal after transfer to GSM; FIG. 9 illustrates an outgoing call from a two-mode terminal via an IP network; FIG. 10 illustrates an outgoing call from a two-mode terminal after transfer to GSM; Fig. 11 is similar to Fig. 7 and illustrates an evolved ISUP anchor; Figure 12 is similar to Figure 8 and illustrates ISUP and INAP anchoring.

The present invention will be described with respect to particular embodiments and with reference to certain drawings although the invention is not limited thereto. The drawings described are shown schematically and are not limiting. In the drawings, the size of some of the elements may be exaggerated and may not be scaled for illustration purposes.

Circuit-switched telephone networks use a signaling protocol referred to as the Common Channel Signaling System # 7 (more commonly referred to as SS7 or C7). On the Public Switched Telephone Network (PSTN), signaling endpoints send and receive SS7 signaling messages. There are three types of signaling endpoints: a Service Switch Point (SSP); a Signal Transfer Point (STP); and a Service Control Point (SCP).

In SS7 networks, Service Integration Digital Network (ISDN) signaling to User Part (ISUP) messages are used to establish, manage and release trunk circuits that route voice calls between telephone exchanges. ISUP messages also carry caller ID information such as the phone number and name of a caller. ISUP is used for both ISDN and non-ISDN calls between exchanges.

Transaction Capabilities Application Part (TCAP) signaling messages support telephony services, such as free calls (free phone), calling cards, number portability local and mobile roaming (wireless) and authentication services. Mobile services are enabled by information routed through the Mobile Application Part (MAP) of a TCAP message. The TCAP protocol supports the exchange of non-circuit related information between signaling points using the connectionless service of the Signaling Connection Control Part (SCCP).

Voice over Internet Protocol (VoIP) networks carry SS7 over IP using protocols defined by the Signal Transport Working Group (SigTran) of the Internet Engineering Task Force (IETF), the international organization responsible for the recommendation of the internet standards. SigTran protocols support the stringent requirements of SS7 / C7 signaling as defined by the International Telecommunication Union.

In IP telephony networks, signaling information is exchanged between the following functional elements: the media gateway (MG), the Media Gateway Controller (MGC), and the signaling gateway.

The media gateway terminates voice calls on inter-central trunks from the public switched telephone network, compresses and packages the voice data, and delivers compressed voice packets to the IP network. For voice calls from an IP network, the media gateway performs these functions in the reverse order. For ISDN calls from the PSTN, Q.931 signaling information is carried from the media gateway to the media gateway controller for call processing.

The media gateway controller processes the recording and the management of resources at the multimedia gateway (s) and exchanges ISUP messages with the telephone exchanges via a signaling gateway. Because manufacturers of media gateway controllers often use out-of-the-box computing platforms, a media gateway controller is sometimes referred to as a "softswitch".

The signaling gateway provides transparent interworking of signaling between circuit-switched and IP networks and can interrupt SS7 signaling or convert and relay messages over an IP network to a media gateway controller or another signaling gateway. Because of their critical role in integrated voice networks, signaling gateways are often deployed in groups of two or more to ensure high availability.

Multimedia gateways, signaling gateways or media gateway controllers ("softswitch") may be separate physical devices or integrated as any combination.

Although the present invention is described with reference to a two-mode terminal that can be connected to a GSM or IP network, it should be noted that the terminal can be connected to more than two types of networks and can therefore include a multimode terminal. In addition, the term "mobile terminal" as used herein should be considered as covering dual-mode terminals and multimode terminals.

FIG. 1 represents a communications system 100 comprising a PSTN network 110, an IP network 120, such as the Internet, and a dual mode terminal 260. The dual mode terminal 260 may be any suitable multimode device, for example a smart phone, a tablet or other portable device with the necessary connectivity. Although the terminal 260 is described as being a two-mode terminal, it should be noted that it may be a multimode terminal operating on more than two modes.

The PSTN network 110 is a public land mobile network (PLMN) comprising a mobile switching center (MSC) 140 connected to a home location register (HLR) 150, an SCP 160 and STP 170. It should be noted that although only one of these elements has been represented, they can be duplicated for redundancy purposes as is commonly done in this technical field. The PSTN network 110 also comprises base stations 175, 180, 185, each base station providing coverage for a mobile telephone cell.

The IP network 120 comprises a routing server 190 which, in this particular embodiment, is a Signal Initiation Protocol (SIP) server, and an MG 130 connected to an MGC 200 controller. MGC controller 200 is identified by a signaling point code, so that it can exchange call signaling information via STP 170 using SS7 ISUP signaling. The MG 130 is connected, via the IP network 120, to a wireless local area network (WLAN) router 210 providing coverage of a wireless LAN, and to a call continuity server. The VCC 220 has a Call In Progress Table (CPT) 230. Like the STP 170, the MG 130 and the MGC 200 can be doubled. for redundancy purposes.

In this embodiment, the MSC controller 140 and the MG 130 are connected through an E1 or T1 junction 240. In this junction 240, an ISUP signal channel SS7 250 provides signaling communications between the point. STP 170 and the MGC 200 controller. The dual mode terminal 260 includes both a mobile radio transceiver, such as a GSM transceiver, a CDMA transceiver, a PDC transceiver, a PDMA transceiver. , a UMTS transceiver or a CDMA2000 transceiver and, in this embodiment, a WLAN transceiver, such as an IEEE 802.11 (Wi-Fi ™) type transceiver, enabling the dual-mode terminal 260 to connect independently to both the PSTN 110 network and the IP 120 network.

The dual-mode terminal 260 further comprises a SIP client which may be implemented by a generic programmable data processor executing a specific computer program. The dual-mode terminal 260 has a unique identifier which is stored in the HLR register 150 as belonging to a subscriber to an FMC service.

FIGS. 2 and 3 are similar to FIG. 1 and illustrate the routing of an incoming call 300 to a two-mode terminal 260. In FIG. 2, the connection is established via the IP network 120 and in FIG. , the connection does not use the IP network 120.

When an incoming call 300 is received via the PSTN network 110, the MSC controller 140 checks whether the unique identifier associated with the dual mode terminal 260 is registered in the HLR register 150 as belonging to a subscriber to this FMC service. If this is the case, the MSC 140 returns the call to SCP 160, which enforces a forced routing routine through STP 170. STP 170 sends a connection request through from the SS7 ISUP signaling channel to the MGC controller 200 which then queries the routing server 190.

If the dual-mode terminal 260 is connected to the IP network 120, its SIP client will have sent a registration message to the routing server 190. If the dual-mode terminal 260 is still registered on the routing server 190 when the MGC 200 interrogates the server In routing 190, the routing server 190 thus transmits the SIP invitation message "Prompt" to the two-mode terminal 260, and returns to the MGC controller 200 a SIP message "Trying". If it is available, the two-mode terminal 260 then responds to the SIP message "Invite" of the routing server 190, first by a SIP message "Trying", then by SIP messages "Ringing" ("Ringing") and / or "Early Media" and "OK". These two or three messages, for example "Ringing" and "OK" or "Early Media" and "OK" referring to two messages, and "Ringing", "Early Media" and "OK" when it comes to three messages are transmitted by the routing server 190 to the MGC controller 200 as a positive response to its request. The MGC 200 converts and retransmits them to the STP 170 through the SS7 ISUP channel 250, accepting the connection request from the STP 170 and establishing the incoming call, as indicated at 300, so that it is routed to the dual mode terminal 260 as a VoIP call through the IP network 120 to the dual mode terminal 260 via the WLAN router 210.

The MGC controller 200 also sends an acknowledgment SIP message "ACK" to the routing server 190, which will transmit it to the two-mode terminal 260. If the two-mode terminal 260 is busy, it responds to the SIP message "Invite" of the server routing 190 by a SIP message by a message "Busy" ("Busy"), which will then be transmitted to the MGC controller 200, the latter converting it into a SS7 ISUP message "REL" with the release cause code "Busy" for the STP point 170. The STP 170 then acknowledges the busy signal to the MGC 200 which converts it into a SIP message "ACK" for the two-mode terminal 260 via the routing server 190.

If the dual-mode terminal 260 is not (or no longer) registered on the routing server 190, the routing server 190 returns, after its SIP message "Trying", a negative response to the request from the MGC controller 200 , in the form of a SIP message containing a failure code. The MGC 200 controller then responds to the connection request with an SS7 ISUP "REL" message. STP 170 then acknowledges this message with an SS7 ISUP "RLC" message for the MGC 200, which converts it into a SIP message "ACK" for the routing server 190.

If the dual-mode terminal 260, although connected to the IP network 120, does not respond to repeated "Prompt" SIP messages sent by the routing server 190, beyond a time limit, the routing server 190 sends a message SIP "Cancel" at the dual-mode terminal 260, and a SIP message "Request Timeout" to the MGC controller 200. The MGC 200 responds to the connection request with a message SS7 ISUP "REL". The STP point 170 then acknowledges it by an SS7 ISUP message "RLC" for the MGC 200 which converts it into a SIP message "ACK" for the routing server 190. In both cases, the incoming call 300 is routed to the dual-mode terminal 260 via the PSTN network 110, without entering the IP network 120, as illustrated at 300 'in FIG.

Once there is a call in progress 300 (FIG. 2) via the IP network 120 to the dual mode terminal 260, the dual mode terminal 260 monitors the quality of the connection, for example by monitoring the packet losses. , the wireless signal strength, the current jitter and / or the jitter buffer size, etc., and when the quality of the connection drops below a predetermined threshold, a transfer may be triggered by a network to another, for example from an IP network to a mobile telecommunications network and vice versa. The transfer is described in more detail below and is specific to the platform, that is to say that it is controlled by the telecommunications platform on which the two-mode terminal 260 operates.

Figure 4a is similar to Figure 2 and illustrates part of the transfer sequence. If the connection of the dual-mode terminal 260 to the IP network 120 is significantly degraded, for example due to the fact that the two-mode terminal 260 moves within range of the WLAN router 210, the SIP client of the dual-mode terminal 260 triggers a continuity call 400 via a PSTN network to the VCC server 220, using a VCC service number.

This VCC service number can be selected from a table of location-dependent VCC service numbers that can be determined from global positioning data and / or the mobile network indicator (MNC). and / or the mobile country code (MCC) of the base station 185 of the PLMN network to which the two-mode terminal 260 is connected. Therefore, if this MNC is the same as that of a PLMN of "Home") to which the dual-mode terminal 260 subscribes, the selected VCC service number is usually a specific short number. However, if the MNC and / or MCC do not correspond to a "departure" PLMN network, indicating that the dual-mode terminal 260 is connected via a third party PLMN network, the VCC service number selected may be a generic number, possibly a toll-free number, which will be routed through another VolP junction. The VCC service number may even be a different country prefix if the MCC code also differs from that of the "out" PLMN network, indicating that the dual mode terminal 260 is roaming abroad, making it even more important selecting an inexpensive VCC service number. Although, in the illustrated embodiment, the continuity call 400 is routed through the same PSTN network 110 as that from which the incoming call 300 originates, it can therefore be routed through a network. phone, and enter the IP network via a different media gateway.

When the VCC server 220 receives a continuity call 400 intended for a VCC service number of this type and originating from a two-mode subscriber terminal 260, the VCC server 220 checks in the CPT table 230 whether there is a call in 300 with this two-mode terminal via the IP network 120. If the answer is positive, the VCC server 220 switches the continuity call 400 with the IP network of the current call 300, then closes the network path IP. As illustrated in FIG. 4b, this puts an end to the transfer of the current call 300 from the IP network 120 to the PSTN network 110.

Although only examples of incoming calls for the two-mode terminal 260 have been illustrated, it should be noted that the dual-mode terminal 260 itself can also send outgoing calls, both via its mobile and wireless transceivers. Indeed, the call transfer illustrated in Figures 4a and 4b can be achieved regardless of whether the current call 300 is an incoming call or an outgoing call. It should also be noted that although only calls established between the dual-mode terminal 260 and the PSTN subscriber have been illustrated, the dual-mode terminal 260 may also make and receive calls to and from subscribers of a network. IP.

Although, in this particular embodiment of the invention, only a transfer initiated by a client has been described, the transfer can also be performed by the VCC server in the case of a continuity call from the controller. MGC at the two-mode terminal, following an instruction obtained via signaling information transmitted by the VCC server. Note also that although the MG 130 and the MGC 200 are shown here as two separate entities, they can be combined into a single unit.

It will also be appreciated that although the present invention relates primarily to voice calls, other mobile services, such as text messages and voice mail, may also be accessible both through the PSTN 110 and the network. IP 120.

Figure 5 illustrates a block diagram of a communications system 500 according to the present invention. The architecture 500 comprises an MMC architecture 505, a fixed network 510 and an MSC / HLR server 515. The MMC convergence is a derivative of the FMC convergence according to which a two-mode terminal operates using software and specialized hardware for connect voice calls and other applications by voice over WLAN (VoWLAN) and / or cellular service. A WLAN is used to route calls over the internet and uses a wireless carrier network if the WLAN is not present. It differs from FMC convergence in that a classification is used so that WLAN connections are the primary connection and the wireless carrier is the secondary connection.

It should be noted that the present invention is not limited to the WLAN and may be a 3G, EDGE or other IP network. The term Wi-Fi as used herein refers to a connection to any suitable IP network, including 3G, EDGE, VoWLAN, and so on. The architecture 505 includes an SI P 520 core connected to the VCC MMC server and Voice Quality Control (VQC) 525. There are provided first and second MG 530, 535 gateways which make connections to the Fixed network 510 and MSC / HLR server 515 from the SIP heart 520. The MMC VCC and VQC 525 can be connected to a public Wi-Fi zone (public zone) 540 and to a controlled Wi-Fi zone (dots). wireless access) 545.

The fixed network 510 is connected to a first PSTN network 550 and the MSC / HLR server 515 is connected to a second PSTN network 555. The first and second PSTN networks 550, 555 can be connected to each other or, in an embodiment, can be the same PSTN network. The MSC / HLR server 515 is also connected to a Mobile Radio Access Network (RAN) 560 via a mobile MSC / HLR server 565. The MSC / HLR server 515 is also connected to an 570 mobile voice mail system.

A dual-mode terminal 575 may be connected to the public area 540, to the wireless access points 545 and to the mobile network RAN 560. The dual-mode terminal 575, which may for example be the handset of a smart phone, comprises an application MMC client that controls the network connection so that Wi-Fi zones, public area 540, and wireless access points 545 are prioritized, with the mobile RAN 560 being used only when Wi-Fi is not available. The MMC 510 architecture supports the highest possible call voice quality on the most appropriate network based on availability. The architecture 505 has two main elements, namely an MMC platform corresponding to the SIP core 520 and the VCC and VQC525 MMC server, and an MMC client (not shown) that is installed on the dual-mode terminal 575.

The current technology implemented on smartphone handsets in combination with the performance of Wi-Fi, the voice quality of the call (VCQ) enabled by Wi-Fi calls (VoIP over Wi-Fi) is greater than the average quality of GSM calls. A high-definition coded on the MMC client allows this high VCQ, as will be described in more detail below. In accordance with the present invention, when the VCQ Wi-Fi of a Wi-Fi call drops below the quality of an average GSM call, a transfer is triggered between the Wi-Fi call and a GSM cellular network .

Figure 6 illustrates a VCQ MMC control system 600. The system 600 includes a MMC platform 610 comprising a Voice Call Continuity Module (VCC) 615, a voice quality control module (VQC) 620 and a call manager module 625. The VCC 615 controls the process of the transfer between Wi-Fi and GSM. The VQC 620 module analyzes voice quality based on real-time voice traffic (RTP, Real-Time

Protocol) and the Real-Time Transport Control Protocol (RTCP).

The RTP protocol defines a standardized packet for delivering audio and video over IP networks and is used in all communications systems that use the streaming of multimedia data, including telephony and video conferencing and web-based instant messaging features. RTP is used in combination with the RTCP protocol. While RTP carries multimedia streams, such as audio and video, RTCP is used to monitor transmission statistics and quality of service (QoS) and help synchronize multiple streams. In addition, the RTP protocol is one of the technical underpinnings of VoIP and is often used in conjunction with the signaling protocol to establish connections across the network.

The call manager module 625 communicates with first and second Session Border Controllers (SBCs) 630, 640. The first SBC 630 forms an interface to the SIP trunk (not shown) and the second SBC 640 forms an interface with a SIP stack 670 in a two-mode terminal 650, as will be described in more detail below. A first MG 635 also interfaces with the SBC 630 and provides communications to a PSTN and Mobile Network Operator (MNO) (none of which represent). The first MG 635 gateway also handles GSM call forwarding. A public internet 645 is available between the MMC 610 platform and the dual mode 650 terminal.

In the dual-mode terminal 650, a Wi-Fi transceiver 655 and a GSM transceiver 680 are provided. The Wi-Fi 655 transceiver is connected to a VCC 660 module, a VQC 665 module and the SIP 670 battery. The VCC 660 corresponds to the VCC 615 module of the MMC 610. The VQC 665 module analyzes the voice quality based on RTCP and Wi-Fi coverage. The SIP 670 stack is optimized for the particular MMC 610 platform with which the dual-mode terminal 650 is associated, and can be connected over Wi-Fi, as indicated by transceiver 675, to provide SIP signaling (SIP SIG) to the SBC controller 640 associated with the MMC platform 610. Although the Wi-Fi transceiver 655 and the Wi-Fi transceiver 675 are represented in the form of two separate modules, it should be noted that they can be implemented in the form of a single transceiver module.

In addition, although the transceiver 675 is shown in Figure 6 as being located outside the mobile terminal 650 for clarity, it should be noted that it is nevertheless located inside the body of the body. mobile terminal 650.

VCQ quality is managed by integrated MMC components to ensure end-to-end call quality. According to the present invention, this is achieved by the VQC module 620 and the VCC module 615 on the MMC platform 610 and by the VQC module 665 and the VCC module 660 on the dual mode terminal 650. The VQC modules 620, 665 verify indicators voice call quality for all current IP voice calls. These verifications may include: 1) correct exchanges of SIP signaling information; 2) the presence of RTP traffic, that is, an RTP protocol carrying VoIP traffic; 3) end-to-end delay of IP packets; 4) delay jitter, i.e., variations in the delay of IP packets; 5) loss of IP packets, ie the loss of RTP information; 6) Wi-Fi coverage level (this includes availability, access, signal strength, etc.); and 7) the level of coverage of the GSM network (this includes availability, access, etc.).

The VCC modules 615 and 660 together perform the transfer of a call that is in progress to another network when necessary. The transfer runs in the background and is transparent to the subscriber or user of the dual-mode terminal 650.

The dual-mode terminal 650 can trigger a Wi-Fi quality call-emitting process each time it connects to a Wi-Fi hotspot. This will depend on whether or not the dual-mode terminal 650 is already connected to a Wi-Fi call, whether the subscriber's preferences allow it or not, etc. During this process, the dual-mode terminal 650 checks the VCQ quality, once it has connected to Wi-Fi, by initiating a voice test call. This test call is sent in the background and the subscriber or the user of the dual mode terminal 650 is not aware that this is done, if a predetermined minimum quality is not reached at the time of the call. test call, the dual-mode handset 650 is not allowed to handle calls over Wi-Fi.

Based on historical data, the MMC 610 platform is able to predict the times when it is necessary to trigger a call transfer in order to avoid any disturbance of the VCQ quality for the subscriber or user of the dual-mode terminal 650. This prediction is based on an algorithm that incorporates a mix of quality-related parameters as well as external variables related to the subscriber, such as terminal hardware, software, and the Wi-Fi hotspot. The conditions enabling the triggering of a transfer are determined according to the MMC service provider, during the start-up of the service, and / or by a cursor provided on the dual-mode terminal 650, which is operated by the user and intended to adjust the sensitivity. These conditions may be defined according to personal preferences or specific marketing conditions.

As described above, a transparent transfer process is of paramount importance for the MMC service is required for cost optimization, ie when the Wi-Fi call is more cost effective than a call GSM, it is necessary to preserve the voice quality of the Wi-Fi call and to ensure the continuity of the voice call beyond the reach of the Wi-Fi access point. The transparent transfer process will be triggered either by the two-mode terminal or by the VCC server MMC and will be implemented without intervention of the subscriber or the user of the mobile terminal.

Figure 7 is similar to Figure 5 and like reference numerals are used to denote like components. In Figure 7, there is shown a user 580 establishing a call. It should be noted that the user may be another dual-mode terminal, a landline or a mobile phone. An incoming Wi-Fi call is shown as a dashed line between user 580 and the 575 dual-mode terminal. The transmitted call connects to the PSTN 555 and is forwarded to the MSC / HLR 515 where it is routed. through the media gateway 530 to the SIP heart 520, to reach the VQC server 525, and through the public Wi-Fi zone 540 to the dual mode terminal 575. The VQC server 525 determines that the voice quality of the call in the public Wi-Fi zone 540 respects the predetermined minimum level and retransmits the call to the two-mode terminal via the public Wi-Fi zone 540. The dual-mode terminal 575 is therefore connected to a VoIP call intended for the user 580.

Figure 8 is similar to Figure 7 and illustrates an incoming Wi-Fi call after transfer. In a manner similar to FIG. 7, the incoming Wi-Fi call is illustrated by a dashed line between the user 580 and the dual-mode terminal 575. However, in this case, the call reaches the VQC server. 525 but the voice quality of the call in the public Wi-Fi zone 540 does not respect the predetermined minimum level. As a result, the VQC server 525 reroutes the call through the SIP heart 520, which sends it back through the media gateway 530, to the MSC / HLR module 515 and the mobile RAN 560, so that it is transmitted. to the dual-mode terminal 575 via a GSM network. The VoIP call becomes a GSM call and the dual mode terminal 575 is no longer connected to the public Wi-Fi zone 540.

Figures 9 and 10 are similar to Figures 7 and 8 and respectively illustrate outgoing calls before and after transfer. In FIG. 9, the dual-mode terminal 575 issues a Wi-Fi call according to the MMC protocol. The Wi-Fi call meets the minimum call quality and is routed to the VQC 525 server, SIP 520 core, through the first MG 530, to the fixed network 510, and through the PSTN 550 network. , up to user 580, as indicated by a dashed line.

In FIG. 10, the outgoing call is routed via the mobile RAN 560 to the MSC / HLR module 515 via the media gateway 515, from the SIP heart 520, to the VQC server 525. As call does not respect the minimum level of Wi-Fi, it is sent back to the SIP 520 core, through the first MG 530, to the fixed network 510, to the PSTN 550 network and to the user 580, as illustrated by a dotted line.

The transfer between the networks has been described with reference to Figures 4a and 4b above.

The Wi-Fi to GSM transfer feature is supported by ISUP voice trunks connecting the first MG 635 gateway (Figure 6) of the MMC 610. The GSM to Wi-Fi transfer functionality is more Advanced and can be supported by a GSM network connected to the MMC platform to which mobile calls are in fact "anchored" in the voice platform. This means that the 610 platform is informed of all mobile calls in progress on the connected GSM network. The anchoring of mobile calls to the MMC voice platform can be activated by physically anchoring all current mobile calls to accept all incoming mobile calls and returning them to the MSC or by performing the anchoring by means of a more advanced SS7 signaling. In the first case, the anchoring can be achieved using existing conventional ISUP voice trunks, but since all mobile calls must be routed through the MMC voice platform, the infrastructure costs can be significantly increased. In the second case, the anchoring is obtained by integration of intelligent network platforms (IN, Intelligent Network) that support the Intelligent Network Application Part (INAP) of the signaling protocol that allows a advanced processing of mobile calls.

The IN network is a network architecture that can be used simultaneously for fixed and mobile telecommunications networks. It relies on the SS7 protocol between telephone switching exchanges and other network nodes belonging to network operators. This architecture allows operators to differentiate themselves by offering value-added services in addition to traditional telecommunications services, for example, PSTN, ISDN and GSM services. The intelligence consists of network nodes on the service layer of the core network, which is different from the core network switch layer. IN nodes are generally the property of telecommunications operators. The INAP part is a signaling protocol that is part of the SS7 protocol suite and is usually a layer above the TCAP part. It may be considered logical to control telecommunications services that have migrated from traditional switch points to service-independent computing platforms.

Figure 11 is similar to Figures 9 and 10 and illustrates an anchoring performed using existing conventional ISUP voice interfaces. A call from the fixed or mobile user 580 is here connected to the PSTN network 555 and transits to the MSC / HLR module 515. From the MSC / HLR module 515, the call passes through the second MG 535 gateway to the SIP heart 520. The call is sent back to the MSC / HLR 515 because the dual mode terminal 575 is not available on an IP network, ie Wi-Fi. The call is anchored via two call paths on the voice trunk and is connected to the mobile RAN network 560 and the dual mode terminal 575 in GSM. The path of the call is illustrated by the dashed line. Once the dual-mode 575 terminal becomes available over Wi-Fi, the transfer of GSM to Wi-Fi is triggered, provided that the minimum call quality requirements are met.

FIG. 12 is similar to FIGS. 9 and 10 and comprises IN architecture modules 590, 595 respectively associated with the MSC / HLR modules 515 and with the SIP core 520. Here, IN integration is required between the IN 590 architecture modules and 595, as indicated by a dashed line. SS7 INAP signaling is provided via the IN integration. When a call is sent from the fixed or mobile user 580, it is established through an ISUP and INAP SS7 implementation and through an IN integration, as illustrated by a dashed line . Since the dual-mode terminal 575 is not connected in Wi-Fi, the signaling of the call is sent back to the MSC / HLR module 515 and connects via the dual-mode terminal 575 in the manner already described above. with reference to FIG. 11. The path of the call is illustrated by the dashed line, as before.

As described above, the VQC control is used to trigger a transfer between an IP network and a GSM network or other network based on a predetermined minimum VCQ level. This minimum level of VCQ can be established using VCQ indicators, as previously described. However, VCQ indicators can also be set according to subscriber (user) preferences, for example cost, acceptable VCQ quality below the recommended level, terminal type, number of terminals at the end of the network. provision of the user, the position of the terminal or terminals and their availability.

As described above, as part of the MMC architecture, a VQC module is provided on both the platform and the multimode terminal. For the transfer of a voice call from one IP network to another network, the VQC module that is part of the multimode terminal or part of the MMC platform, as previously described, determines whether the voice quality of the call is lower. at a voice quality level of the predetermined minimum call required for connection to an IP network. In some cases, the two VQCs can together perform the determination that the voice quality of the call has dropped below a voice quality level of the predetermined minimum call.

Having determined that the voice quality level of the call was below the predetermined minimum set according to the preferences of the user and / or the system provider, a transfer is initiated by the VCC module either on the platform or on the the multimode terminal. In some cases, both VCCs can trigger a call transfer together. The call manager of the platform establishes a new call path while retaining the existing call path. A transfer is made when the new call path has been established, and the old call path is interrupted once the transfer is complete. This transfer is transparent vis-à-vis the user of the multimode terminal.

When it is determined that the transfer of an existing call is necessary, the server, for example the MMC platform, contacts the client, for example the multimode terminal, to inform it that a transfer is necessary. The client responds that it is happy to perform the transfer based on the near environment, for example when no other IP network is available nearby, to which the client can connect. Once the server has received a customer acknowledgment, the call manager is instructed to establish a new call path while retaining the old call path and make a one-time transfer. that the new calling path has been established and then interrupting the old call path as described above.

It should be noted that the present invention is not limited to the particular embodiments described above, but covers many variations and / or many additions.

Claims (22)

A multimode communication system (500) comprising at least one internet protocol network (540, 545), at least one mobile telecommunications network (560), a telecommunications platform (505; 610) and at least one multimode terminal ( 575; 650), said two telecommunications platforms (505; 610) and said at least one multimode terminal (575; 650) connectable to said at least one internet protocol network (540, 545; 645) and said network. mobile telecommunications system (560), said system comprising: at least one voice continuity control module (615, 660) for providing call continuity for a call involving said at least one multimode terminal (650) when it is connected to said at least one internet protocol network (540, 545, 645); characterized in that said system (500) further comprises at least one voice quality control module (620, 665) for determining the voice quality of the call of a call involving said at least one multimode terminal (575 650) when connected to said at least one internet protocol network (540, 545; 645); and in that said at least one voice continuity control module (620, 665) functions to trigger a transfer between said at least one internet protocol network (550, 545; 645) and said at least one telecommunications network. mobile (560) based on a predetermined minimum speech quality level determined by said at least one voice quality control module (620, 665). 2. The system of claim 1, further comprising a call manager (625) associated with said at least one voice continuity control module (620, 665), said call manager (625) having as function to trigger a transfer between said at least one internet protocol network (550, 545, 645) and said at least one mobile telecommunications network (560). 3. System according to claim 1 or 2, wherein the transfer is triggered according to the availability of the internet protocol network. 4. System according to any one of the preceding claims, wherein the transfer is triggered according to the availability of the mobile telecommunications network. A system according to any one of the preceding claims, further comprising a real-time transport control protocol module providing a real-time transport control protocol relating to the voice traffic of said call, said predetermined minimum speech quality level being determined according to said real-time transport protocol. The system of any of the preceding claims, further comprising a real-time protocol analyzer module providing real-time protocol data relating to voice traffic of said call, said predetermined minimum speech quality level being determined in accordance with said transport time protocol. real. A system according to any one of the preceding claims, further comprising a database (515) for storing data relating to at least one user of the system, said predetermined minimum speech quality level being determined in accordance with said user data. . A system according to any one of the preceding claims, further comprising a location register (515) for storing terminal data relating to multimode terminals (575; 650) associated with said system (500), said minimum voice quality predetermined value being determined in accordance with said terminal data. A system according to any one of the preceding claims, wherein said telecommunications platform (610) comprises a first voice quality control module (620) and a first voice continuity control module (615) and said multimode terminal ( 650) comprises a second minimum voice quality control module (665) and a second voice continuity control module (660), said predetermined minimum speech quality of a current call being determined by at least one of said first and second voice quality control modules (620, 665) and said transfer being triggered by at least said first voice continuity control module (615). The system of claim 9, wherein said second voice quality control module (665) comprises an internet protocol manager for detecting the availability of said at least one internet protocol network (540, 545, 645) and determining if the voice quality meets said predetermined minimum speech quality associated with said at least one internet protocol network (540, 545; 645). A multimode terminal (575; 650) for use in a multimode terminal system (500) according to any one of the preceding claims, said multimode terminal (575; 650) comprising a voice quality control module (665) comprising an internet protocol manager for detecting the availability of said at least one internet protocol network (540, 545; 645) and determining whether the voice quality complies with said predetermined minimum speech quality associated with said at least one internet protocol network (540, 545; 645). Multimode terminal according to claim 10, wherein a transfer is triggered according to the availability of an internet protocol network. 13. Multi-mode terminal according to claim 10 or 11, wherein a transfer is triggered according to the availability of a mobile telecommunications network. The multimode terminal of any one of claims 10 to 12, further comprising a voice continuity control module (660) for initiating a transfer between said at least one internet protocol network (540, 545, 645) and said mobile telecommunications network (560) when said voice quality falls below said predetermined minimum speech quality. A multimode terminal according to any one of claims 10 to 13, wherein said multimode terminal connects to an internet protocol network (540, 545, 645) preferably to a mobile telecommunications network (560). A multimode terminal according to any one of claims 10 to 14, further comprising a real-time transport control protocol module providing a real-time transport control protocol relating to the traffic of said call, said predetermined minimum speech quality level. being determined in accordance with said real-time transport control protocol. A multimode terminal according to any one of claims 10 to 15, further comprising a real time protocol analyzer module providing real time protocol data relating to voice traffic of said call, said predetermined minimum speech quality level being determined according to said auditory protocol module. real time protocol. 18. A method of executing the transfer of a voice call between an internet protocol network (540, 545; 645) and a mobile telecommunications network (560) in a multimode terminal system (500), said system having at least one an internet protocol network (540, 545; 645), at least one mobile telecommunications network (560), a telecommunications platform (505; 610) and at least one multimode terminal (575; 650), said telecommunications platform (505); 610) and said at least one multimode terminal (575; 650) can both be connected to said at least one internet protocol network (540, 545; 650) and said at least one mobile telecommunications network (560), said method comprising the steps of: a) assigning a predetermined minimum speech quality level to a voice call connected to said at least one internet protocol network (540, 545; 645); b) determining voice quality of the call for said at least one internet protocol network (540, 545, 645); c) comparing said voice quality of the determined call with said predetermined minimum speech quality level; and d) initiating a transfer of said voice call between said at least one internet protocol network (540, 545, 645) and said at least one mobile telecommunications network (560) when said voice quality of the given call decreases in below said predetermined minimum speech quality level. The method of claim 17, wherein step d) comprises triggering a transfer based on the availability of an internet protocol network. The method of claim 17 or 18, wherein step d) comprises initiating a transfer based on the availability of a mobile telecommunications network. The method of any one of claims 17 to 19, wherein step a) comprises assigning said predetermined minimum speech quality level in accordance with one of: user data; multi-mode terminal data; voice traffic with real-time protocol; and real-time transport control protocol information. The method of any one of claims 17 to 20, wherein step a) comprises establishing a voice call over an internet protocol network (540, 545; 645) preferably to a mobile telecommunications network (560).