• 专利附录
  • 专利说明
  • 权利要求
  • 类似技术
  • 同族专利
  • 引用文献
  • 法律状态
  • 优先权
    • 专利摘要:
    The present invention relates to a communication system comprising a network PSTN 1 comprising an STP 7 and an IP network 2 comprising a media gateway 13 and a media gateway switch 15 connected to said media gateway 13 and having a point code. The STP 7 and the media gateway switch 15 are connected to each other via a signaling link, in particular a signaling link 16 of the SS7 ISUP protocol. The present invention also relates to a method for routing an incoming call in said PSTN network 1 to a dual-mode telephone 3 that can be connected to both said PSTN network 1 and said IP network 2..
    公开号:BE1019820A3
    申请号:E201000704
    申请日:2010-11-24
    公开日:2013-01-08
    发明作者:Burlin Bernard Noel De；Joeri Uyttendaele；Davy Vandemoere
    申请人:Mondial Telecom；
    IPC主号:
    专利说明:

    "Communication system and method for routing an incoming call"
    TECHNICAL AREA
    The present invention relates to a communication system comprising a public switched telephone network (PSTN) and a network using the Internet Protocol (IP), as well as a method for routing an incoming call in a PSTN network, to a two-mode telephone which can be independently connected to both PSTN and IP networks.
    Context of the invention
    Voice communications have recently made rapid progress in two parallel areas. The first area is cellular telephony, which has increased the mobility of telephone users. A second, more recent area is Voice Communications using the Internet Protocol (VoIP), in which voice communications are routed over IP networks.
    While undeniable advantages have been provided both by cellular telephony and by VoIP communications, their parallel development has also led to certain disadvantages for a user wishing to exploit both systems.
    Increasingly, dual mode phones are being proposed that can be independently connected to both PSTN and IP networks. In particular, dual-mode telephones are provided that include both a mobile transmitter / receiver for wirelessly connecting the dual-mode telephone to a PSTN cellular network, such as, for example, a GSM network or a UMTS network. , and including a wireless LAN (LAN) transceiver for connection to an IP network via a wireless LAN. The existence of such dual-mode telephones, coupled with the growing popularity of VoIP communications, has led to the emergence of the Fixed / Mobile Convergence (FMC) concept offering a simple communication system comprising both a PSTN cellular network and a network. IP, so that the two-mode phone can maintain communications with this communication system, using one or the other of its connections.
    International Patent Application WO 00/79814 A1 discloses such a communication system comprising both a PSTN network and an IP network. In this communication system, the first of these networks includes a home location register in which the two-mode telephone is registered, and the second network includes a service location recorder. When the dual-mode telephone is connected to this second network, the service location recorder, located in the second network, sends updated location information to the first network. The most significant disadvantage of this system lies in the fact that location data must be exchanged between the two networks, whenever the dual-mode telephone is connected to and disconnected from the second network, even if, in fact , no call occurs. In addition, this document of the prior art does not provide for the transfer of a call in progress when the dual-mode telephone goes from one network to another.
    The US patent application US 2005/0096024 A1 discloses a cellular communication system in which, when the dual-mode telephone is connected to the IP network, the IP network must transmit its IP address to a location recorder placed in the PSTN network. Therefore, the system has the same disadvantages as those of WO 00/79814 A1.
    US patent application US 2009/0131045 A1 discloses another communication system comprising a PSTN network and an IP network. In this system, incoming calls to the dual-mode telephone are always routed through the IP network. If the dual-mode phone is not connected to the IP network at this time, incoming calls are forwarded from the IP network to the PSTN. Although this system no longer requires sending a location update, from the IP network to the PSTN network, whenever the dual-mode phone is connected to or disconnected from the IP network, this system requires however, a considerable bandwidth between the two networks, especially when the dual-mode telephone is not connected to the IP network, in which case the call must be transferred from the PSTN network to the IP network, and then transmitted again to the PSTN network. Although this document describes the determination of the presence of a two-mode telephone in an environment in which it can be connected to an IP network or a cellular network (or other terrestrial network), this document does not address the problem of the transfer. call. Some publications have addressed the problem of transferring an ongoing call when a dual-mode telephone is switched from an IP network to a PSTN network. More specifically, this problem has been addressed 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).
    US patent application US 2006/0198360 A1 discloses a communication system and a method for transferring a call when a two-mode telephone goes from an IP network to a PSTN. In this communication system, the IP network comprises a call manager that triggers, via the PSTN network, a continuity call to the two-mode telephone if the connection, via the IP network, is disturbed. However, this also requires additional resources to be provided in the IP network, said resources being in the form of an apparatus comprising said call manager, a quality monitor, a threshold selector and a target selector.
    European patent application EP2018014A1 discloses a similar communication system and a method in which a continuity call is by the two-mode telephone either manually or automatically if the connection via the IP network is disturbed. A dual-mode telephone can be connected to the IP network and to a terrestrial network. Here, a private branch exchange (PBX) is connected to the outside world through an Integrated Services Digital Network (ISDN) and calls are routed through it. When a call is to be transferred from an IP network to a terrestrial network, the dual-mode telephone initiates the placing of the first call (the current call between the two-mode telephone and a remote device) waiting by sending a message to the PBX. The PBX then transfers this message to the remote device. A second call is stable between the P interface of the dual-mode telephone, through the PBX, to the cellular interface of the dual-mode telephone, that is to say that the dual-mode telephone is called itself. The VoIP branch of this call is then transferred from the PBX to the remote device. The first call is completed when the communication between the dual mode phone and the remote device is established through the PBX.
    US patent application US 2006/0121902 A1 also discloses a communication system and method for transferring a call when a dual mode telephone is switched from an IP network to a PSTN network. In this method, the continuity call is triggered by the dual-mode telephone, rather than by a call manager placed in the IP network. However, this document does not disclose how the two-mode telephone manages the connection to various PSTN networks and, in particular, the end of roaming to exit the IP network and switch to a PSTN network other than a "national" network. the dual-mode phone.
    US patent application US 2008/0273505 A1 also discloses a communication system and a method for transferring a call when a two-mode telephone goes from an IP network to a PSTN network. However, this document does not disclose how the two-mode telephone manages the connection to various PSTN networks and, in particular, the end of roaming to exit the IP network and switch to a PSTN network other than the "internal" network of the PSTN. dual-mode phone.
    SUMMARY OF THE INVENTION
    An object of the present invention is to provide an appropriate communication system for providing call connectivity to a two-mode telephone, via both a PSTN network and an IP network with, between these two networks, a comparatively small bandwidth.
    In addition, another object of the present invention is to provide a communication system that is integrated with a mobile operator such that the IP network and the cellular network cooperate with each other to provide integrated coverage to a network. user.
    For this purpose, at least one embodiment of the present invention comprises a communication system comprising a PSTN network having a signaling transfer point, a network using the Internet protocol having a media gateway (MG) connected to a gateway switch media (MGC) and a two-mode telephone that can be connected to both said PSTN and said IP network. This media gateway switch comprises a signaling point code, in particular a national signaling point code (NSPC) and is connected to said signaling transfer point, in the PSTN, via at least one signaling link, in particular a link of the signaling point. SS7 ISDN protocol ISUP. This provides integration between the networks.
    In communication systems, common channel signaling is the transmission of signaling information, that is, control information inherent to the routing of communications, through a channel separate from the communications themselves. . For example, in a PSTN, a channel of a communication line is typically used for the sole purpose of providing signaling for the establishment and release of calls. In most cases, a single 64 kbit / s channel is sufficient to handle call setup and call clearing for many telephone and data channels.
    SS7 (Signaling System No. 7) is the most commonly used set of signaling protocols. The basic international protocol SS7 is defined by the ITU-T standardization body in its Q.700 series of recommendations. The national SS7 protocols are developments of this basic international protocol. Built into the SS7 system, ISUP is the user part for making calls.
    In the field of communications, a signaling transfer point (STP) is understood to be a router that relays SS7 messages between signaling network endpoints (SEPs) and other signaling transfer points (STPs), in order to forward them to the appropriate signaling link. SEP and STP are identified by unique point codes.
    Assigning a point code to the MGC and connecting it to a STP in the PSTN via a signaling link which may take the form of a dedicated signaling link in a communication line such as a line E1 or T1 and, in particular, an SS7 ISUP protocol channel, it becomes possible for the PSTN network to test, via the IP network, the availability of a two-mode telephone, before actually transferring the call to the IP network, without having to update the home location register (HLR) of the PSTN network, whenever the dual-mode telephone is connected to or disconnected from the IP network. The necessary bandwidth of the link between the PSTN and the IP network is thus reduced, and fewer changes to the PSTN network are required, resulting in a less complex and more reliable system.
    Advantageously, said IP network may further comprise a routing server connected to said NGC. The routing server routes calls over the IP network and can be polled by the MGC to determine whether or not the dual-mode phone is connected to the IP network. In a preferred embodiment, said routing server may be a server using the Session Initiation Protocol (SIP), but, of course, other variants may be considered by those skilled in the art.
    The dual-mode telephone may further comprise a mobile radio transmitter / receiver for wirelessly connecting said two-mode telephone to said PSTN network via at least one base station of a public land mobile network (PLMN) connected to said network. PSTN. The mobile radio transceiver may be, for example, a GSM transceiver, a CDMA transceiver, a PDC transceiver, a PDMA transceiver, a UMTS transceiver, a Tetra transceiver, or a transmitter. / CDMA2000 receiver. The dual mode phone can be mobile.
    The dual-mode telephone may also include a wireless transceiver for wirelessly connecting said two-mode telephone to said IP network. Said wireless transmitter / receiver may be, for example, a transmitter / receiver of a wireless local area network (LAN) according to an IEEE 802.11 standard (also known as WiFi ™ technology). The dual-mode telephone can thus be used appropriately, such as a cordless phone on an area covered by a wireless LAN. However, the wireless transceiver may also be any other type of transceiver suitable for wireless connection to an IP network, such as, for example, a WiMAX ™ transceiver, or well, even, the mobile transmitter / receiver itself when it is appropriate, also, for connection to an IP network.
    Advantageously, said PSTN network may further comprise a mobile switching center (MSC) for routing calls in the PSTN network.
    The PSTN may further include a Network Service Control Point (SCP), wherein the MSC is configured to direct an incoming call directed to said two-mode telephone to be processed by said SCP, and said SCP is configured to route the call to said first STP. Incoming calls are thus necessarily routed to this signaling gateway to the IP network.
    Said dual-mode telephone may include a unique identifier, and said PSTN may further include a home location register (HLR) listing said identifier and connected to said MSC, and said MSC is configured to determine, by controlling, in said HLR, whether a incoming call is directed to said two-mode telephone. A virtual mobile network operator (MVNO) using said MSC can thus provide the dual-mode telephone, all mobile services.
    Another object of the present invention is to provide an appropriate communication system for providing call connectivity to a two-mode telephone, via both a PSTN and an IP network, in which call continuity can be provided. even when the dual-mode phone during a call in progress changes from a connection to an IP network to a connection to a PSTN network.
    The IP network may further include a Voice Call Continuity (VCC) server connected to said MG, to provide call continuity even when the dual-mode telephone switches from one type of connection to another.
    The dual-mode telephone may be configured to call the VCC server over a PSTN network using a VCC service number if an existing connection to said IP network is disrupted. The continuity call is thus established by the two-mode telephone itself, making the necessary resources in the IP network even smaller.
    In particular, said dual-mode telephone may be configured to select said VCC service number, from an array of available telephone numbers, depending on the location of the two-mode telephone. In particular, different numbers may be selected based on whether the dual-mode telephone is directly connected to a national PSTN network, or is roaming on a PSTN network of a third party. This can be determined, for example, on the basis of a country code of the mobile subscriber and / or on the basis of a subscriber network code of a mobile subscriber of a base station. a PLMN network, to which the dual-mode telephone is connected, wirelessly, to said PSTN network, or on the basis of a global positioning signal. Continuity can thus be ensured by the most economical routing, the most reliable routing and / or the delivery offering the best service.
    Said VCC server may be configured to check, upon receipt of a continuity call from said two-mode telephone, using said VCC service number, if a call in progress is routed to said two-mode telephone, via said IP network and, in this case, in this case, said server switches part of the IP network of the current call, with at least part of the continuity call, to provide call continuity via the connection to the PSTN.
    Another object of the present invention is to provide a method for routing an incoming call in a PSTN to a dual-mode telephone connectable to both said PSTN and said IP network.
    In at least one embodiment of the invention, this method comprises the steps of: - sending a call connection request, from an STP located in said PSTN network, to an MGC comprising a code semaphore point in said IP network; determining whether said dual-mode telephone is connected to said IP network; if the dual-mode telephone is connected to the IP network, to route the call to the two-mode telephone via said IP network; and if the dual-mode telephone is not connected to or responding to the IP network, responding to said connection request from the STP and addressed to the MGC, with a release message passing from the MGC to the STP, and forwarding the call to the two-mode telephone via said PSTN network.
    Said connection request and said release message may be in the form of SS7 ISUP messages.
    This method may also include a step of sending a question, from said MGC to a routing server located in said IP network, to determine whether said dual-mode telephone is connected to said IP network.
    If, after the incoming call has been routed to the two-mode telephone via the IP network, the connection of the two-mode telephone to the IP computer network is disrupted, the dual-mode telephone can call a VCC server in said IP network via a PSTN network. , using a VCC service number.
    When the VCC server receives a call from the dual-mode telephone, using a VCC service number, if a current call is routed via said IP network, to said two-mode telephone, the current call can be routed back through the network. PSTN, preferably by exchanging part of the IP network of the current call, with at least part of the call from the dual-mode telephone, to the media gateway, as stipulated via the signaling information transmitted by the server VDC.
    The present invention provides the integration of communication systems using SIP and SS7, and solves the problem of single pricing. In addition, it allows a user to make a call from a number and then to be called each time on the same number, whether the user is connected to a cellular network (GSM) or to a network. IP, when making the call. In addition, the present invention provides an economical solution that allows easy integration to a mobile network operator (MNO), which is both resilient, and substantially transparent to the MNO.
    BRIEF DESCRIPTION OF THE DRAWINGS
    These and other objects of the present invention will become more apparent upon reading the following detailed description and referring to the accompanying drawings in which:
    Figure 1 shows an embodiment of a communication system according to the invention;
    FIGS. 2a, 2b and 2c show how, in the communication system of FIG. 1, an incoming call, which is routed to the two-mode telephone, via the IP network, is processed if the two-mode telephone is connected to the IP network;
    FIGS. 3a, 3b and 3c show how, in the communication system of FIG. 1, an incoming call is routed to the two-mode telephone, via the PSTN network, if the dual-mode telephone is not connected to the IP network or else is connected but does not answer;
    FIGS. 4a and 4b show how, in the communication system of FIG. 1, a call in progress is routed again via the PSTN network if the connection of the two-mode telephone to the IP network is disturbed.
    DETAILED DESCRIPTION OF THE INVENTION
    The switched telephone networks use a signaling protocol called Common Channel Signaling System # 7 or Semaphore Network Number 7 (more commonly known as SS7 or C7). In the switched telephone network signaling network endpoints (SEPs) send and receive SS7 signaling messages. There are three types of signaling transfer points: service signaling points (SSPs), signaling transfer points (STPs), and control transfer points (SCPs).
    In SS7 networks, user ISDN semaphore (ISUP) messages are used to establish, manage, and release circuits that carry voice calls between telephone exchanges. ISUP messages also carry information about the caller's identity, such as the call number and the name of the caller. ISUP is used for both ISDN and non-ISDN calls between telephone exchanges.
    Transaction Capabilities Application Part (TCAP) signaling messages support telephony services such as free calls, calling cards, local number portability, and cellular roaming and authentication services. Mobile services are made possible by the mobile application part (MAP) of a TCAP message. TCAP supports non-circuit-related information exchange between signaling points using the connectionless services of the Signaling Connection Control Part (SCCP).
    Voice over Internet Protocol (VoIP) networks carry SS7 over IP using protocols defined by the SIGTRAN working group of the Internet Engineering Task Force (IETF), the international organization responsible for recommending Internet standards. Sigtran protocols support the strict conditions of SS7 / C7 as defined by the International Telecommunications Union (ITU)
    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 links of the public switched network, compresses and packages the voice data and delivers the compressed voice data packets to the IP network. For voice calls from the IP network, the media gateway performs the same actions in reverse. For ISDN calls from the public switched network, the Q.931 signaling information is transported from the media gateway to the media gateway switch for call processing.
    The media gateway switch governs the registration and resource management of the media gateway (s) and exchanges ISUP messages with the central switches through a signaling gateway. Because media gateway switch providers use standard computers, a media gateway switch is sometimes called a "softswitch."
    The signaling gateway provides seamless interoperability between a switched network and IP networks, and can terminate SS7 signaling or translate and relay messages over an IP network to a media gateway switch or other signaling gateway. Because of their critical role in voice integrated networks, signaling gateways are sometimes deployed in groups of two or more to ensure high availability.
    Media gateways, Media Gateway gateway switches, and signaling gateways may be separate or integrated physical devices in any combination.
    A particular embodiment of a communication system according to the invention is illustrated in FIG. 1. This communication system comprises a PSTN network 1, an IP 2 network, such as the Internet network, and a dual-mode telephone 3.
    The PSTN network 1 is a PLMN network comprising an MSC 4 connected to an HLR 5, an SCP 6, and an STP 7. Although only one of each of these components is shown in the illustrated embodiment, these components may be doubled for redundancy purposes, as is usual in this technical field. The PSTN network 1 also includes base stations 8, each of which provides a coverage area for a mobile telephone cell.
    The IP network 2 comprises a routing server 9 which, in this particular embodiment, is a SIP server, and an MG 13 media gateway connected to a MGC 15. This MGC 15 is identified with a signaling point code, a so that it can exchange call signaling information via STP 7, using SS7 ISUP protocol signaling. The MG 13 is connected, via said IP network 2, to a wireless LAN router, providing a coverage area to a wireless LAN, and connected to a VCC server 12. The server VCC 12 has a current calling table (CPT) 10. Just like STP 7, MG 13 and MGC 15 can be doubled for redundancy purposes.
    In this embodiment, the MSC4 and the MG 13 are connected via a line E1 or T1 14. With this line 14, a signaling link 16 of the SS7 ISUP protocol provides signaling communication between the STP 7 and the MGC 15. The dual-mode telephone 3 comprises both a mobile transmitter / receiver such as, for example, a GSM transceiver, a CDMA transceiver, a PDC transceiver, a PDMA transceiver, a UMTS transceiver or a CDMA2000 transceiver and, in this embodiment, a wireless LAN transmitter / receiver, such as an IEEE 802.11 (WiFi ™) compliant transceiver, allowing the dual mode to be connected, independently, to both the PSTN network 1 and the IP network 2. The dual-mode telephone 3 further comprises a SIP client which can be implemented by a generic programmable data processor which makes it to implement a specific computer program. The dual-mode telephone 2 has a unique identifier which is stored in the HLR 5 as part of a provider of an FMC service.
    Referring now to the state diagram shown in Fig. 2a, when an incoming call to this unique identifier is received via the PSTN 1, the MSC 4 checks whether this unique identifier is registered in the HLR 5 as belonging to a subscriber of this FMC service. If this is the case, the MSC 4 addresses the call to the SCP 6 which applies a forced routing program via the STP 7. The STP 7 sends a connection request in the form of a message ΊΑΜ ", via the signaling link 16 of the ISUP SS7 protocol, to the media gateway switch 15 which then interrogates the routing server 9 by sending a SIP message "Invite." If the dual-mode telephone 3 is connected to the IP network 2, its SIP client will have sent a message of registration to the routing server 9. If the dual-mode telephone 3 is not yet registered at the routing server 9 when the MGC 15 queries the routing server 9, the routing server 9 will then transmit the SIP message "Invite" on the two-mode telephone 3, and send back to the MGC 15 a SIP message with the code "100" ("Trial"), if it is available, the dual-mode telephone 3 will then reply to the SIP message "Invite" of the routing server 9, first by a SIP message with the code "100" ("Ess ai "), followed by SIP messages with codes" 180 "(" ringing tone ")," 183 "(" media ") and" 200 "(" OK "). These three messages are transmitted to the MGC 15, by the routing server 9, as a positive response to his question. The MGC 15 transcode them to the STP 7, via channel 16 of the SS7 ISUP protocol, as "ACM", "CPG" and "ANM" messages, accepting the connection request from the STP 7 and carrying out the routing of the incoming call 17, as a VoIP call, to the dual-mode telephone 3, via the IP network 2, as shown in FIG. 2b. The MGC 15 also sends an acknowledgment SIP message "ACK" to the routing server 9 which will transmit it to the two-mode telephone 3. If the dual-mode telephone 3 is busy, as shown in FIG. 2c, it will respond to the SIP message "Invite" of the routing server 9, by a SIP message with a code "486" ("busy") which will then be transmitted to the MGC 15 which transcode to the STP 7, in "REL" message SS7 ISUP protocol , with the release cause code "17" ("busy"). The STP 7 will then acknowledge the busy signal by a signal "RLC" SS7 ISUP protocol sent to the MGC 15 which will transcode as a SIP message "ACK" to the dual-mode phone 3 via the routing server 9 .
    Referring now to FIG. 3a, if the dual-mode telephone 3 is not (or no longer) registered at the routing server 9, the routing server 9, after its SIP message including the code "100" ("Test"), will return a negative answer to the question from MGC 15, in the form of a SIP message with a trouble code in the "4xx" series (other than the "486" code), in the series "5xx" or in the series "6xx". The MGC 15 will then, in turn, respond to the connection request with an "REL" message from the SS7 ISUP protocol, including a release cause code "20". The STP 7 will then acknowledge receipt of this message, by an "RLC" message of the SS7 ISUP protocol, addressed to the MGC 15 which will transcode it as a SIP message "ACK", to the routing server 9. In a similar scenario illustrated in FIG. 3b, the dual-mode telephone 3, although connected to the IP network 2 (FIG.1), does not respond to the SIP repeated "Invite" messages sent by the routing server 9. In this case, after a delay, the Routing server will send to the dual-mode telephone 3, a SIP message "cancel", and send to the MGC 15, a SIP message with the code "408" ("request timeout"). As in the previous scenario, the MGC 15 will then respond to the connection request 15 with a "REL" message of the ISUP SS7 protocol, including a release cause code "20". The STP 7 will then acknowledge receipt of this message, by an "RLC" message from the SS7 ISUP protocol, sent to the MGC 15 which will transcode it as a SIP message "ACK" sent to the routing server 9. In both scenarios, the call 55 is then routed to the dual-mode telephone 3, via the PSTN network 1, without entering the IP network 2, as illustrated in FIG. 3c.
    Referring again to the example illustrated in FIGS. 2a and 2b, once there is a call in progress 19, via the IP network 2 addressed to the dual-mode telephone 3, the dual-mode telephone 3 will control the quality of the call. connection, for example by controlling packet loss, wireless signal amplitude, current variation and / or size of timing buffer, etc.
    Referring to FIG. 4a, if the connection of the dual-mode telephone 3 to the IP network 2 is significantly disrupted, for example due to the fact that the dual-mode telephone 3 moves towards the outside range of the router 11 of the local area network ( LAN), the SIP client, in the dual-mode telephone 3, will trigger, via a PSTN network, a continuity call 56 addressed to the VCC server 12, using a VCC service number.
    This VCC service number can be selected from an array of available VCC service numbers, depending on the location that can be determined from global positioning data and / or the subscriber's country code. to a mobile (MNC) and / or the mobile subscriber network identifier (MCC) of the base station 8 of the PLMN network, to which the two-mode telephone 3 is connected. Thus, if this MNC corresponds to that of a "national" PLMN network to which the dual-mode telephone 3 is subscribed, the selected VCC service number will usually be a short specific number. If, however, the MNC and / or MCC do not match a national "PLMN" network indicating that the dual-mode telephone 3 is connected to a PLMN network of a third-party operator, the selected VCC service number may be a Generic number, where applicable a green number which will be routed via another VolP line. The VCC service number may even have a different country prefix if the MCC also differs from that of the "national" PLMN network, indicating that the dual-mode telephone is roaming abroad, making the selection even more important. an economical VCC service number. Although in the illustrated embodiment, the continuity call is routed via the same PSTN network 1 as the incoming call 55 from it, it can also be routed via a different telephone network and enter the IP network via a gateway. different media.
    When the VCC server 12 receives a continuity call 56 directed to such a VCC service number and coming from a two-mode telephone 3 of the subscriber, the VCC server 12 will search, in the CPT 10, if a call in progress, with this dual-mode telephone, occurs on the IP network 2. If the answer is positive, the VCC server 12 will swap the continuity call 56, with the IP network of the current call 55, and then close the network part IP. As illustrated in FIG. 4b, this completes the transfer of the call in progress 55 from the IP network 2 to the PSTN network.
    1.
    Only examples of incoming calls have been shown to the dual-mode telephone 3, but it should be understood that the dual-mode telephone 3 can also, itself, trigger outgoing calls, both via its transmitters / receivers. mobile and wireless telephony. In particular, the call transfer illustrated in Figures 4a and 4b can be realized regardless of whether the current call (55) is an incoming or outgoing call. While only calls between the dual-mode telephone 3 and a subscriber to a PSTN network have been shown, it should also be understood that the dual-mode telephone 3 can also send and receive calls to subscribers to a network. IP and from these subscribers. Although it has been described, in this particular embodiment of the invention, only a transfer initiated by the client, the transfer could, ultimately, also be performed by the VCC server, with a continuity call from from the MGC to the two-mode telephone as specified via the signaling information transmitted by the VCC server. Although the media gateway 13 and the media gateway switch 15 are illustrated here as two different entities, it should also be understood that they can be combined and form a single set.
    Although the present invention relates mainly to voice calls, it should also be understood that other mobile services, such as text messaging and voice messaging, can also be accessed both via the PSTN network 1 and the Internet. IP network
    2.
    The present invention is capable of various modifications and other alternative forms, the specific embodiments of which have been shown by way of example, in the drawings, and are here described in detail. But, it must be understood that the intention is not to limit the invention to particular forms that have been disclosed but, on the contrary, the intention is to cover all modifications, equivalents and variations within the scope of the invention. the applicability of the invention as expressed in the appended claims.
    权利要求:
    Claims (19)
    [1]
    A communication system comprising: - a public switched telephone network (1) comprising a signal transfer point (7); a network using the Internet protocol (2) comprising a media gateway (13) connected to a media gateway switch (15); and - a two-mode telephone (3) connectable to both said public switched telephone network (1) and said network using the Internet protocol (2); and characterized in that said media gateway switch (15) comprises a signaling point code and is connected to said signaling transfer point (7) via a signaling link.
    [2]
    The communication system of claim 1, wherein said signaling link is a signaling link (16) of the ISUP SS7 protocol extended over a communication line (14).
    [3]
    The communication system of any one of the preceding claims, wherein said network (2) using the Internet protocol further comprises a routing server (9) connected to said media gateway switch (15).
    [4]
    A communication system according to any one of the preceding claims, wherein said dual-mode telephone (3) comprises a mobile telephone transmitter / receiver for wireless connection of said dual-mode telephone (3) to said public switched telephone network ( 1) via at least one base station (8) of a public land mobile network connected to said public switched telephone network (1).
    [5]
    A communication system according to any one of the preceding claims, wherein said dual-mode telephone (3) comprises a wireless transceiver for wireless connection of said dual-mode telephone (3) to said network (2) using the Internet protocol.
    [6]
    The switching system of any preceding claim, wherein said public switched telephone network (1) further comprises a mobile switching center (4).
    [7]
    The communication system of claim 6, wherein a public switched telephone network (1) further comprises a network service control point (6), said mobile switching center (4) being configured to direct an incoming call (55), directed to said dual mode telephone to be processed by said network service control point (6), and said network service control point (6) is configured to signal the call (55) via said first network service point (6). transfer signaling (7).
    [8]
    The communication system of claim 7, wherein said dual-mode telephone (3) has a unique identifier, and said public switched telephone network (1) further comprises a home location register (5) listing said unique identifier and connected to said mobile switching center (4), and said mobile switching center (4) is configured to determine whether an incoming call (55) is directed to said two-mode telephone, by checking in said home location register (5).
    [9]
    The communication system of any preceding claim, wherein said network (2) using the Internet protocol further comprises a voice call continuity server (12) connected to said media gateway (13).
    [10]
    The communication system of claim 9, wherein said dual-mode telephone (3) is configured to call the voice call continuity server (12) via a public switched telephone network (1), using a service number of voice call continuity if an existing connection to said network (2) using the Internet protocol is disrupted.
    [11]
    The communication system of claim 10, wherein said dual-mode telephone (3) is configured to select said voice call continuity service number, from an array of available numbers depending on the location of the dual-mode telephone. (3).
    [12]
    The communication system according to one of claims 9 or 10, wherein said voice call continuity server (12) is configured to control, upon receipt of a voice continuity call (56) from said two-mode telephone ( 3), using said voice call continuity service number, if a call in progress (55) is routed to said two-mode telephone (3) via said network (2) using the Internet protocol and, in this case, is configured to switch a portion (20a) of the IP network of the current call (55), with at least a portion of the voice continuity call (56).
    [13]
    A method for routing an incoming call (55) in a public switched telephone network (1) to a dual mode telephone (3) connectable to both said public switched telephone network (1) and to a network (2). ) using the Internet protocol, comprising the steps of: - sending a connection request from a signaling transfer point (7) in said public switched telephone network (1) to a media gateway switch (15). ) located in said network (2) using the Internet protocol, but identified by a point code; - if the dual-mode telephone (3) is connected to said network (2) using the Internet protocol, to route the call (55) to the dual-mode telephone (3) via said network (2) using the Internet protocol; and if the dual-mode telephone (3) is not connected to said network (2) using the Internet protocol, or does not respond, to respond to said connection request from the signaling transfer point (7) to the switch media gateway (15), by a clearing message from the media gateway switch (15) to the signaling transfer point (7), and forwarding the call (55) to the dual-mode telephone (3) via said public switched telephone network (1).
    [14]
    The method of claim 13, wherein said connection request and said release message are in the form of SS7 ISUP protocol messages.
    [15]
    The method of one of claims 13 or 14, further comprising sending a request from said media gateway switch (15) to a routing server (9) located in said network (2). using the Internet protocol, to determine whether said dual-mode telephone (3) is connected to said network (2) using the Internet protocol.
    [16]
    The method of any one of claims 13 to 15, wherein if, after the incoming call (55) has been routed to the two-mode telephone (3), via the network (2) using the Internet protocol, the connection of the two-mode telephone (3) to the network (2) using the Internet protocol is disturbed, the dual-mode telephone (3) calls a voice call continuity server (12) placed in said network (2) using the Internet protocol, via a public switched telephone network (1), using a voice call continuity service number.
    [17]
    The method of claim 16, wherein said voice call continuity service number is selected from an array of available numbers based on the location of said dual mode telephone (3).
    [18]
    The method of one of claims 16 or 17, wherein, when the voice call continuity server (12) receives a call (56) from the dual mode telephone (3) using a service continuity number of voice call, if a call in progress (55) is routed via said network (2) using the Internet protocol, to said two-mode telephone (3), the call in progress (55) is routed again via the public telephone network switched (1).
    [19]
    The method of claim 18, wherein said current call (20) is routed back to the media gateway (13) via the public switched telephone network (1), swapping part of the IP network of the call. in progress (55), with at least part of the call (56) coming from the two-mode telephone (3), as stipulated by the signaling information transmitted by the voice call continuity server (12).
    类似技术:
    公开号 | 公开日 | 专利标题
    US9363384B2|2016-06-07|Systems for delivering calls on dual-mode wireless handsets
    US8655353B2|2014-02-18|Method and system for international roaming and call bridging
    EP1894372B1|2017-01-11|Inter-domain call routing
    US9078174B2|2015-07-07|Call transfer for an integrated wireline and wireless service
    US7920551B2|2011-04-05|Multiple access service convergence
    US9363370B2|2016-06-07|Methods of delivering calls on dual-mode wireless handsets
    US9148288B2|2015-09-29|Conditional telecommunications
    WO2007103121A2|2007-09-13|Mobile application gateway for connecting devices on a cellular network with individual enterprise and data networks
    WO2007103217A1|2007-09-13|System and method for enabling vpn-less session setup for connecting mobile data devices to an enterprise data network
    KR101089046B1|2011-12-02|System and method for providing multimedia caller information display service
    US9184978B2|2015-11-10|Operating and supporting dual mode user equipment
    US20060135166A1|2006-06-22|Selective redirection of cellular calls to a local wireless endpoint
    BE1019820A3|2013-01-08|COMMUNICATION SYSTEM AND METHOD FOR ROUTING AN INCOMING CALL.
    WO2005120115A1|2005-12-15|Method for switching between two telephony services
    US20220078216A1|2022-03-10|Method and system for initiating a mobile voice call
    BE1021396B1|2015-11-16|IMPROVEMENTS IN VOICE QUALITY CONTROL
    BE1021478B1|2015-11-27|IMPROVEMENTS IN LOCALIZATION REGISTERS
    EP2214375B1|2014-07-30|Method for controlling voice communications in the absence of an IMS core network, and associated interoperation agent
    FR2941832A1|2010-08-06|Operators e.g. mobile network operators, interconnecting system for e.g. general packet radio service networks, has server whose selection is determined to authenticate user whose device houses software, during launching of software
    同族专利:
    公开号 | 公开日
    WO2011064278A1|2011-06-03|
    EP2505022A1|2012-10-03|
    EP2326126A1|2011-05-25|
    引用文献:
    公开号 | 申请日 | 公开日 | 申请人 | 专利标题
    DE60030273T2|1999-04-27|2007-08-30|Tekelec, Calabasas|METHOD AND SYSTEMS FOR CONTROLLING CHARACTER EMBODIMENTS IN A COMMUNICATION NETWORK USING CONTACT ADDRESS INFORMATION|
    WO2000079814A1|1999-06-21|2000-12-28|Nokia Networks Oy|Mobility between ip telephony networks and cellular networks|
    US7577427B2|2003-11-05|2009-08-18|At&T Intellectual Property I, L.P.|System and method of transitioning between cellular and voice over internet protocol communication|
    US7539492B2|2004-12-03|2009-05-26|Cisco Technology, Inc.|System and method for providing a handoff leg associated with a preexisting leg in a network environment|
    US7830863B2|2005-03-07|2010-11-09|Solacom Technologies Incorporated|Voice over internet protocol call continuity|
    US20080273505A1|2007-05-01|2008-11-06|Robert Lee Hollingsworth|Providing Handover/Handoff for Dual Mode Mobile Terminals in a GSM Network Using A Three-Way Calling Mechanism|
    DE602007004103D1|2007-07-17|2010-02-11|Research In Motion Ltd|A method for passing sessions from a VOIP interface to a mobile phone interface in a dual-mode device|
    US8825058B2|2007-09-10|2014-09-02|Net2Phone, Inc.|Single number services for fixed mobile telephony devices|EP2515498A1|2011-04-20|2012-10-24|Mondial Telecom|Improvements in or relating to voice quality control|
    BE1021396B1|2012-10-24|2015-11-16|Mondial Telecom|IMPROVEMENTS IN VOICE QUALITY CONTROL|
    法律状态:
    2018-01-31| MM| Lapsed because of non-payment of the annual fee|Effective date: 20161130 |
    优先权:
    申请号 | 申请日 | 专利标题
    EP09176908|2009-11-24|
    EP09176908A|EP2326126A1|2009-11-24|2009-11-24|Communications system and method for routing an incoming call|
    [返回顶部]