WO2011011922A1 - 非稳定VoIP呼叫中的终端切换方法、网络单元、设备及系统 - Google Patents

非稳定VoIP呼叫中的终端切换方法、网络单元、设备及系统 Download PDF

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Publication number
WO2011011922A1
WO2011011922A1 PCT/CN2009/073023 CN2009073023W WO2011011922A1 WO 2011011922 A1 WO2011011922 A1 WO 2011011922A1 CN 2009073023 W CN2009073023 W CN 2009073023W WO 2011011922 A1 WO2011011922 A1 WO 2011011922A1
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WIPO (PCT)
Prior art keywords
handover
call
network element
network
terminal
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PCT/CN2009/073023
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English (en)
French (fr)
Inventor
雷正雄
Original Assignee
上海贝尔股份有限公司
阿尔卡特朗讯公司
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
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Application filed by 上海贝尔股份有限公司, 阿尔卡特朗讯公司 filed Critical 上海贝尔股份有限公司
Priority to JP2012521925A priority Critical patent/JP5727477B2/ja
Priority to EP09847708.6A priority patent/EP2461627B1/en
Priority to KR1020127004993A priority patent/KR20120047979A/ko
Priority to US13/388,125 priority patent/US9374740B2/en
Priority to CN200980159990.0A priority patent/CN102474789B/zh
Priority to PCT/CN2009/073023 priority patent/WO2011011922A1/zh
Publication of WO2011011922A1 publication Critical patent/WO2011011922A1/zh
Priority to US15/090,034 priority patent/US9807650B2/en

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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04WWIRELESS COMMUNICATION NETWORKS
    • H04W36/00Hand-off or reselection arrangements
    • H04W36/0005Control or signalling for completing the hand-off
    • H04W36/0055Transmission or use of information for re-establishing the radio link
    • H04W36/0058Transmission of hand-off measurement information, e.g. measurement reports
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04WWIRELESS COMMUNICATION NETWORKS
    • H04W36/00Hand-off or reselection arrangements
    • H04W36/0005Control or signalling for completing the hand-off
    • H04W36/0011Control or signalling for completing the hand-off for data sessions of end-to-end connection
    • H04W36/0022Control or signalling for completing the hand-off for data sessions of end-to-end connection for transferring data sessions between adjacent core network technologies
    • H04W36/00224Control or signalling for completing the hand-off for data sessions of end-to-end connection for transferring data sessions between adjacent core network technologies between packet switched [PS] and circuit switched [CS] network technologies, e.g. circuit switched fallback [CSFB]
    • H04W36/00226Control or signalling for completing the hand-off for data sessions of end-to-end connection for transferring data sessions between adjacent core network technologies between packet switched [PS] and circuit switched [CS] network technologies, e.g. circuit switched fallback [CSFB] wherein the core network technologies comprise IP multimedia system [IMS], e.g. single radio voice call continuity [SRVCC]
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04WWIRELESS COMMUNICATION NETWORKS
    • H04W36/00Hand-off or reselection arrangements
    • H04W36/0005Control or signalling for completing the hand-off
    • H04W36/0011Control or signalling for completing the hand-off for data sessions of end-to-end connection
    • H04W36/0022Control or signalling for completing the hand-off for data sessions of end-to-end connection for transferring data sessions between adjacent core network technologies
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04WWIRELESS COMMUNICATION NETWORKS
    • H04W36/00Hand-off or reselection arrangements
    • H04W36/14Reselecting a network or an air interface

Definitions

  • Terminal switching method network unit, device and system in unstable VoIP call
  • the present invention relates to the field of communications, and in particular to techniques for switching terminals in an unstable state of a VoIP call. Background technique
  • FIG. 1 exemplarily shows the SRVCC network architecture specified in 3GPP TS 23.216.
  • the UE accesses the IMS through an E-UTRAN (Evolved Universal Terrestrial Radio Access Network) and an S-GW/PDN GW.
  • E-UTRAN also known as LTE (Long Term Evolution)
  • LTE Long Term Evolution
  • the EPS is simplified into two network elements, eNodeB and EPC, by functionally integrating the NodeB, RNC, and CN of the existing WCDMA and TD-SCDMA systems.
  • the so-called EPS which includes EPC (Evolved Packet Core, Core Network), E-UTRAN (also known as LTE) and UE.
  • the EPC includes: MME (Mobility Management Entity), which serves as a control node and is responsible for signaling processing of the core network; S-GW (Service Gateway) / PDN-GW (Packet Data Network Gateway), which is responsible for data processing of the core network.
  • MME Mobility Management Entity
  • S-GW Service Gateway
  • PDN-GW Packet Data Network Gateway
  • Figure 1 also shows the interface between the network elements suggested by the specification.
  • E-UTRAN and EPC are connected through S1 (similar to Iu) interface
  • E-UTRAN can be connected through X2 (similar to lur) interface (not shown)
  • UE and E-UTRAN through LTE-Uu interface. connection.
  • UTRAN UMTS Terrestrial Radio Access Network
  • GERAN GSM/EDGE Radio Access Network
  • Circuit switched domain In UTRAN/GERAN, the UE accesses the IMS network via a base station, an MSC (Mobile Switching Center) server.
  • MSC Mobile Switching Center
  • UTRAN is a new access network for UMTS. It has become an important access method for UMTS. It can include NodeB (Node B), RNC (Radio Network Controller), CN (Core). Network) etc.; GERAN is a key part of GSM developed and maintained by 3GPP, also included in the UMTS/GSM network, which includes the base station BS and base station controller (BSC) and their interfaces (eg Ater interface, Abis) Interface, A interface, etc.). Typically, the mobile operator's network consists of multiple GERANs and is combined with UTRAN in the UMTS/GSM network.
  • NodeB Node B
  • RNC Radio Network Controller
  • CN Core
  • Network Network
  • GERAN is a key part of GSM developed and maintained by 3GPP, also included in the UMTS/GSM network, which includes the base station BS and base station controller (BSC) and their interfaces (eg Ater interface, Abis) Interface, A interface, etc.).
  • Figure 2 exemplarily shows the related call flow from E-UTRAN to the target GERAN.
  • the voice call needs to be anchored in advance in, for example, an SCC AS (Service Centralization and Continuity Application Server) in the IMS.
  • SCC AS Service Centralization and Continuity Application Server
  • the source E-UTRAN decides to perform handover from the packet domain to the circuit domain for the VoIP call that is being performed by the local UE according to the measurement report received from the local/source UE
  • the handover request is sent to the local MME, and then
  • the source MME divides the bearer (for subsequent transfer of the voice service), and sends a corresponding handover request from the packet domain to the circuit domain to the MSC server or the media gateway that can currently cover the local UE.
  • the corresponding MSC/media gateway performs handover preparation and establishes a circuit, a session transfer is initiated.
  • the session transfer process (steps 11 through 12) of the IMS layer (on the remote UE, that is, the party that establishes the VoIP session with the local UE). Specifically, the session transfer process is performed by the SCC AS in the IMS, the remote UE is updated with the SDP (session description protocol) of the target CS access leg, and the source EPC PS access leg is released. The above steps will cause the voice portion of the ongoing session to be transformed from EPC to MGW at the user level.
  • SDP session description protocol
  • Handover procedure from layer 2 to the target cell switching from E-UTRAN to GSM performed on the local UE and the access network, steps 15-21), which is between the local UE and the RAT performed at the access network Switch.
  • 3GPP TS 23.216 does not specify how to support SRVCC in an unstable state of the session.
  • An unstable state can refer to an early state of a session, such as a caller receiving a temporary SIP response message (except for a 100 Trying message) instead of receiving a 200 OK message.
  • a method for switching a terminal from a packet switched domain to a circuit switched domain in an unsteady state of a call including: a first network element Anchoring the call based on a call setup request received from the terminal, and saving a temporary response returned to the terminal and a status of the call; the terminal detecting that a handover is required and transmitting a measurement report to the access network; The access network determines that the handover is requested to the second network element as long as the signaling bearer is detected, and the second network element receives the handover from the access network.
  • a network unit for managing the concentration and continuity of services which is capable of switching a terminal from a packet switched domain to a circuit switched domain in an unsteady state of a call, the network element comprising: And means for saving a temporary response returned to the terminal and a status of the call after the call is anchored based on a call setup request received from the terminal; and, a temporary response check and transmission device, for Upon receiving a handover request from another network element, the temporary response saved before the call is detected to be in an unstable state and the temporary response is sent to yet another network element.
  • an access device which is capable of switching a terminal from a packet switched domain to a circuit switched domain in an unsteady state of a call, the access device comprising determining means for the determining means When the measurement report received from the terminal satisfies the handover condition, it is determined that the handover is requested to another network element as long as the signaling bearer is detected.
  • a network element for managing mobility of a terminal which is capable of switching the terminal from a packet switched domain to a circuit switched domain in an unstable state of a call, the network element including a determining device
  • the determining means is configured to, after receiving the request for handover from the access network, determine a handover request to send the packet switched domain to the circuit switched domain to another network element whenever the signaling bearer is detected.
  • a communication system comprising at least: at least one terminal; a network unit for managing concentration and continuity of services according to the present invention; an access device according to the present invention; and The mobility of the network unit that manages the terminal.
  • Figure 1 exemplarily shows the network architecture for SRVCC specified in TS 23.216;
  • Figure 2 shows an example of the current technology from E-UTRAN to
  • Figure 3 exemplarily shows from E-UTRAN to according to the present invention
  • Figure 4 is a flow diagram of a method for SRVCC handover in an unsteady call state, in accordance with one embodiment of the present invention
  • FIG. 5 is a block diagram of a network element in accordance with one embodiment of the present invention.
  • FIG. 6 is a block diagram of an access device according to an embodiment of the present invention
  • FIG. 7 is a block diagram of a network element according to an embodiment of the present invention
  • QCI Quality of Service Level Identifier
  • MSC Mobile Switching Center
  • the MSC Server Upon receiving the SRVCC PS to CS request, the MSC Server will initiate a session handover procedure to the IMS and SCC AS.
  • the SCC AS then responds to the MSC server based on the status of the access leg and the last response sent from the SCC AS to the local terminal (the response was previously saved by the SCC AS). After sending a response to the MSC server, the SCC AS will complete the session switch, update the remote end and release the initial access leg.
  • the MSC Server can connect to the IMS/SCC in the following two ways.
  • AS the MSC Server can connect to the IMS/SCC in the following two ways.
  • the MSC server supports the 12/11 interface to the CSCF/SCC AS, the MSC server connects directly to the IMS/SCC AS via the 12/11 interface;
  • the MSC Server connects to the IMS/SCC AS via the MGCF/MGW.
  • the present invention is applicable to either mode.
  • only the first mode will be described below.
  • the present invention can also be applied to the second mode and the call flow of the second mode is similar to the first mode.
  • the SCC AS should be enhanced to preserve the status of the access leg and the last response message sent from the SCC AS to the calling party. If the SCC AS sends a 180 Ringing SIP message to the calling party in the access leg, it should save the response. In addition, upon receiving a session switch initiation request from the MSC server, the SCC AS shall send an appropriate response to the MSC server based on the access leg status and the pre-saved response;
  • the MME shall enhance to detect the IMS signalling bearer and send an SRVCC request to the MSC server based on the local policy;
  • - E-UTRAN should be enhanced to initiate an SRVCC handover when an IMS signaling bearer is detected.
  • the MSC server should have the ability to command the MGW to play the ring tone to the calling party.
  • Figure 3 shows the signaling flow for SRVCC handover of a call/session in an unstable state.
  • the session handover request sent by the MGW to the SCC AS may be an ISUP message or a SIP message according to the capabilities of the MSC server.
  • the invention is applicable to both ISUP and SIP scenarios.
  • SIP SIP The interest is described as an example.
  • step 1 the local UE sends an INVITE request to the IMS and SCC AS.
  • the request is forwarded to the S-CSCF (not shown) in accordance with the normal IMS session establishment process.
  • the request is forwarded to the SCC AS using iFC (Initial Filtering Criteria) service logic.
  • iFC Intelligent Filtering Criteria
  • step 2 the SCC AS anchors the session in order to implement the handover.
  • step 3 the SCC AS sends a temporary response to the UE.
  • the example of the temporary response may be 180 Ringing or 183 Session Progress SIP.
  • step 4 the local UE detects that a handover from E-UTRAN to GERAN-UTRAN may be required and sends a measurement report to the E-UTRAN.
  • the source E-UTRAN determines the SRVCC handover that triggers to GERAN. This is different from the signaling procedure shown in Figure 2.
  • the triggering of the handover can be determined based on the existence of the IMS signaling bearer, thereby implementing the SRVCC handover procedure of the call/session in the unstable state.
  • step 6 the source E-UTRAN sends a Handover Required message to the source MME and indicates to the source MME that this is an SRVCC handover operation.
  • the MME will send an SRVCC PS to CS request to the MSC server.
  • Steps 8 to 12 are the same as steps 5 to 9 in Fig. 2, and a detailed description thereof can be referred to 3GPP TS 23.216, which is not described here.
  • step 13 the MSC server initiates session switching by transmitting an INVITE message with STN-SR (Session Handover Number - Single Radio) and MGW SDP information to the IMS.
  • STN-SR Session Handover Number - Single Radio
  • MGW SDP information MGW SDP information
  • step 14 the SCC AS discovers the anchored session based on the STN-SR. Then, the SCC AS detects that the access leg is in an unstable state, that is, the call is in an early state. After that, the SCC AS checks the last saved SIP temporary response and sends a corresponding response to the MSC server. For example, if the saved response is 180 Ringing, the SCC AS will send a 180 Ringing message to the MSC server. If the saved response is a 183 Session Progress message, the SCC AS will send a 183 Session Progress message to the MSC server. However, it should be clear that if the SCC AS detects that the access leg is in a steady state, it operates in accordance with the original SRVCC procedure.
  • Step 15 illustrates two exemplary choices, 183 Session Progress message and 180 Ringin message.
  • the MSC server will play a ring tone to the calling party.
  • the present invention is not limited to these two types of messages, and other response messages may be used for the unsteady state.
  • step 16 the SCC AS continues to complete the session handoff process and updates the remote end.
  • the remote end can be updated using an UPDATE or re-INVITE message. It should be noted, however, that in the present invention only UPDATE messages can be used due to the session being in an unsteady state.
  • step 17 since the initial access leg is in an unstable state, the SCC AS sends a CANCEL message instead of a BYE message to release the access leg.
  • FIG. 3 is only for the case where the local UE is the calling party, The invention is also applicable to the case where the local UE is the called party, and the signaling flow is similar to that of FIG. For the sake of simplicity, it will not be described here.
  • the SRVCC handover procedure can now also be implemented when the call/session is in an unstable state. This is undoubtedly an enhancement to the original SRVCC program.
  • a method for handover of a terminal from an E-UTRAN (PS domain) to a UTRAN/GERAN (CS domain) in an unstable state of a VoIP call will now be described with reference to FIG.
  • the method of this embodiment can be applied to, for example, the network architecture shown in Fig. 1, and the description of the network architecture will not be repeated here.
  • the first network element anchors the call based on a call setup request received from the terminal, and saves the temporary response returned to the terminal and the status of the call.
  • the first network unit may be, for example, the SCC AS in FIG. 3
  • the terminal is, for example, the local UE in FIG. 3
  • the call setup request may be, for example, an INVITE message conforming to the SIP protocol
  • the temporary The response may be, for example, a 180 Ringing message conforming to the SIP protocol or a 183 Session Progress message.
  • the state of the call in this embodiment refers to the early state of the call, that is, the state when the call has not been fully established.
  • the local UE sends an INVITE message to the SCC AS to request to establish a call.
  • the SCC AS anchors the call and sends a 180 Ringin message or 183 Session Progress message to the local UE and saves the sent temporary response message and the status of the access leg.
  • the terminal detects that a handover is required and sends a measurement report to the access network.
  • the access network is, for example, the source E-UTRAN in FIG.
  • the local UE detects that SRVCC handover from E-UTRAN to GERAN-UTRAN may be required, thus transmitting a measurement report to the source E-UTRAN.
  • the access network detects the signaling bearer on the premise that the measurement report satisfies the handover condition, it is determined to request the handover to the second network element.
  • the place The second network element is, for example, the source MME shown in FIG.
  • the source E-UTRAN can determine the trigger to GERAN's SRVCC handover.
  • the source E-UTRAN will also determine the SRVCC handoff that is triggered to the GERAN, and the handoff at this time conforms to the standard SRVCC handoff procedure.
  • voice bearer i.e., voice
  • the source E-UTRAN will also determine the SRVCC handoff that is triggered to the GERAN, and the handoff at this time conforms to the standard SRVCC handoff procedure.
  • 3GPP TS 23.216 3GPP TS 23.216, which will not be repeated here.
  • the second network unit determines to send the PS to the first network unit as long as the signaling bearer is detected.
  • the handover request is, for example, an INVITE message having an information indicating a continuity of handover using a single wireless voice call and a call parameter in accordance with the SIP protocol, wherein the information indicating the continuity handover using a single wireless voice call is, for example, indicated to
  • the PS to CS handover uses the STN-SR of the SRVCC procedure, and the call parameters are, for example, the SDP from the media gateway.
  • step 404 can be performed, for example, as follows: The MME first sends an SRVCC PS to CS request to the MSC server.
  • the MSC server performs handover preparation and establishes a circuit, and then initiates a session handover by transmitting an INVITE message with SDP information of the STN-SR and the MGW to the SCC AS.
  • the present invention is not limited to the above process, and those skilled in the art The administrator can imagine any interaction between the MME and the SCC AS that can implement the handover.
  • step 405 after the first network unit receives the handover request, if the call is detected to be in an unstable state, the temporary response saved before is checked and the temporary response is sent to the third network. Unit, thus continuing the session switching process and updating the remote end.
  • the third network element is, for example, the MSC server shown in FIG.
  • the SCC AS finds an anchored session based on the STN-SR. Then, the SCC AS detects that the access leg is in an unstable state, that is, an early state of the call. After that, the SCC AS checks the last saved SIP temporary response and sends a corresponding response to the MSC server. For example, if the saved response is 180 Ringing, the SCC AS will send a 180 Ringing message to the MSC server; if the saved response is a 183 Session Progress message, the SCC AS will send a 183 Session Progress message to the MSC Server. In the case of a 180 Ringing message, the MSC server will play a ring tone to the calling party. However, it should be understood that the present invention is not limited to these two types of messages, and other response messages may be used for the unsteady state. The SCC AS then proceeds through the session handoff process and updates the remote end.
  • the SCC AS sends a CANCEL message instead of a BYE message to release the access leg.
  • the present embodiment is also applicable to the case where the local UE is the called party, and the method flow is similar to that described above. For the sake of simplicity, it will not be described here.
  • the SRVCC handover procedure can now also be implemented when the call/session is in an unstable state, which enhances the original SRVCC scheme.
  • a management service is provided A centralized and continuous network element capable of switching a terminal from a packet switched domain to a circuit switched domain in an unsteady state of a call. This will be described below in conjunction with the drawings.
  • FIG. 5 illustrates a network element 500 that manages the concentration and continuity of services in accordance with one embodiment of the present invention.
  • the network unit 500 includes a saving device 501 and a temporary response checking and transmitting device 502.
  • the saving device 501 is configured to save the temporary response returned to the terminal and the state of the call after the call is anchored based on the call setup request received from the terminal. For example, when receiving the INVITE message sent by the UE, the SCC AS anchors the call and sends a 180 Ringing or 183 Session Progress message to the UE, and then the saving device 501 saves the message sum sent to the UE and the access leg status.
  • the temporary response check and transmission means 502 is configured to check the previously saved temporary response and detect the temporary response when it is detected that the call is in an unstable state when receiving a handover request from another network element (source MME)
  • the response is sent to yet another network element (MSC server).
  • MSC server network element
  • the SCC AS receives an INVITE message from the source MME with the SDP of the STN-SR and the MGW and detects that the access leg is in an unstable state
  • the temporary response check and transmission device 502 checks the last saved 180 Ringing or 183.
  • the Session Progress response message and the corresponding response message is sent to the MSC server.
  • the network unit 500 of the present embodiment, and the storage device 501 and the temporary response check and transmission device 502, which are included in the embodiment, may be implemented in software, hardware or a combination of software and hardware.
  • those skilled in the art are familiar with a variety of devices that can be used to implement these components, such as processors, microcontrollers, application specific integrated circuits (ASICs), programmable logic devices (PLDs), and/or field programmable gate arrays (FPGAs). Wait.
  • the various components of the network element of this embodiment can be physically separated and operatively connected to each other.
  • the network unit 500 of the centralized and continuous management service of the embodiment described above in connection with FIG. 5 can implement the foregoing method for switching the call from the PS to the CS in an unstable state of the VoIP call.
  • the SRVCC handover procedure can be implemented when the call/session is in an unstable state, thereby enhancing the original SRVCC scheme.
  • an access device is provided that is capable of switching a terminal from a packet switched domain to a circuit switched domain in an unstable state of a call.
  • FIG. 6 shows an access device 600 including decision device 601, in accordance with one embodiment of the present invention.
  • the determining means 601 is configured to determine to request handover to another network element (source MME) as long as the signaling bearer is detected before the measurement report received from the terminal satisfies the handover condition.
  • source MME another network element
  • the access device 600 of the present embodiment and the determining device 601 included therein may be implemented in software, hardware, or a combination of software and hardware.
  • those skilled in the art are familiar with a variety of devices that can be used to implement these components, such as microprocessors, microcontrollers, application specific integrated circuits (ASICs), programmable test devices (PLDs), and/or field programmable gates. Array (FPGA), etc.
  • the access device 600 of the embodiment described above in connection with FIG. 6 can implement the previously described method for switching the call from the PS to the CS in an unstable state of the VoIP call.
  • the SRVCC handover process can be implemented while the call/session is in an unstable state, thereby enhancing the original SRVCC scheme.
  • a network element for managing mobility of a terminal capable of switching the terminal from a packet switched domain to a circuit switched domain in an unstable state of a call.
  • FIG. 7 shows a network element 700 for managing the mobility of a terminal, comprising decision means 701, in accordance with one embodiment of the present invention.
  • the network unit 700 of the embodiment and the determining device 701 included therein may Implemented in software, hardware, or a combination of software and hardware.
  • those skilled in the art are familiar with a variety of devices that can be used to implement these components, such as microprocessors, microcontrollers, application specific integrated circuits.
  • ASIC can be edited, device (PLD) and / or field programmable gate array (FPGA).
  • PLD device
  • FPGA field programmable gate array
  • the network unit 700 of the embodiment described above in connection with Fig. 7 can implement the previously described method for switching the call from the PS to the CS in an unstable state of the VoIP call.
  • the SRVCC handover procedure can be implemented while the call/session is in an unstable state, thereby enhancing the original SRVCC scheme.
  • a communication system comprising at least one terminal, a network unit 500 for centralized and continuous management services described in the foregoing embodiments, The access device 600 and the network unit 700 that manages the mobility of the terminal.
  • the communication system may also include other network elements, such as an MSC server or the like.
  • the PSVCC switching of the PS to CS of the call can be realized in an unstable state of the VoIP call.
  • a specific operation procedure reference may be made to the above description of a method for switching the call from the PS to the CS in an unstable state of the VoIP call according to an embodiment of the present invention, which is not described herein.
  • the network element, the access device, and the management terminal that manage the service concentration and continuity in the unstable state of the VoIP call are transmitted by some exemplary embodiments.
  • a network element, and a communication system including at least one terminal, a network unit that manages service concentration and continuity, an access device, and a network unit that manages terminal mobility are described in detail, but the above embodiments are not exhaustive Various changes and modifications can be made by those skilled in the art within the spirit and scope of the invention. Therefore, the invention is not limited by the embodiments, and the scope of the invention is limited only by the appended claims.

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  • Computer Networks & Wireless Communication (AREA)
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Description

非稳定 VoIP呼叫中的终端切换方法、 网络单元、 设备及系统 技术领域
本发明涉及通信领域, 具体地涉及在 VoIP呼叫的非稳定状态下 切换终端的技术。 背景技术
在 3GPP TS 23.216中已经定义了如何在呼叫 /会话处于稳定状态时支 持单一无线语音呼叫连续性 ( SRVCC, Single Radio Voice Call Continuity ) , 也就是说用户在发起至电路交换 ( CS )域的切换时在使用 分组交换 ( PS )域的媒体流的 IMS会话中处于激活状态。
图 1示例性示出了 3GPP TS 23.216中规定的 SRVCC网络架 构。 如图 1所示, UE通过 E-UTRAN (演进通用陆地无线接入网 络) 、 S-GW/PDN GW接入 IMS。 E-UTRAN也称为 LTE (长期 演进技术) , 包含若干个 E-Node B, 负责无线接入网部分。 EPS 通过对现有的 WCDMA和 TD-SCDMA系统的 NodeB、 RNC、 CN 进行功能上的整合, 简化为 eNodeB和 EPC两种网元。
而所谓的 EPS , 其包括 EPC (演进分组核心, 即核心网 ) , E-UTRAN (也称为 LTE ) 和 UE。 EPC 包括: MME (移动性管 理实体), 用于充当控制节点, 负责核心网的信令处理; S-GW (服 务网关) /PDN-GW (分组数据网络网关) , 负责核心网的数据处 理。 其中, 非 3GPP无线接入网可通过 PDN-GW接入 EPC, 3GPP 无线接入网可通过 S-GW接入 EPC。
此外, 图 1 还示出了该规范建议的网络单元之间的接口。 例 如, E-UTRAN与 EPC通过 S1 (类似于 Iu )接口连接, E-UTRAN 之间则可通过 X2(类似于 lur )接口连接(未示出 ),UE与 E-UTRAN 则通过 LTE-Uu接口连接。 在图 1所示的环境中, 当 UE处于 E-UTRAN的覆盖边缘或覆 盖区域外时,可以决定切换到 UTRAN( UMTS地面无线接入网络) /GERAN ( GSM/EDGE 无线接入网络) 提供的电路交换域。 在 UTRAN/GERAN 中, UE 经由基站、 MSC (移动交换中心) J! 务 器接入到 IMS网络。
其中, UTRAN是一种较新的用于 UMTS的接入网, 目前已成 为 UMTS较重要的一种接入方式, 其可包括 NodeB (节点 B ) 、 RNC (无线网络控制器) 、 CN (核心网络) 等; 而 GERAN则是 由 3GPP 制定和维护的 GSM 的一个关键部分, 也包括在 UMTS/GSM 网络中, 它包括基站 BS和基站控制器 ( BSC ) 以及 它们的接口 (例如 Ater接口、 Abis接口、 A 接口等) 。 通常, 移 动运营商的网络由多个 GERAN组成, 在 UMTS/GSM的网络中则 与 UTRAN组合。
关于图 1 中的其他网络单元以及各网元之间通信方式等详细 信息, 可以参考 TS 23.216。
图 2则示例性示出了从 E-UTRAN到目标 GERAN的相关呼叫 流程。 为了完成语音会话的转移, 需要预先在 IMS中的例如 SCC AS (服务集中和连续性应用服务器) 上锚定该语音呼叫。
如图 2所示, 当源 E-UTRAN根据接收自本地 /源 UE的测量 报告, 决定对本地 UE正在进行的 VoIP呼叫进行从分组域到电路 域的切换时, 向本地 MME发送切换请求, 随后源 MME对承载进 行划分 (用于后续对语音服务的转移) , 并向目前能够覆盖本地 UE 的 MSC J! 务器或媒体网关发送相应的从分组域到电路域的切 换请求。 相应的 MSC/媒体网关进行切换准备并建立电路后, 发起 会话转移。 这里, 需要注意的是, 如果该本地 UE将要切换到的目 标 MSC与收到来自 MME的切换请求的 MSC为同一个时, 则虚 线部分的步骤 6至 9可以省略 (步骤 20、 21也是如此) 。 随后的步骤中 (即步骤 10之后) , 图 2示出了两个用户层面 的切换过程:
IMS层 (在远程 UE, 也就是与本地 UE建立 VoIP会话的对 方)的会话转移过程(步骤 11至 12 )。 具体地, 由 IMS中的 SCC AS执行会话转移过程, 用目标 CS接入腿( access leg )的 SDP (会 话描述协议) 更新远程 UE , 并释放源 EPC PS 接入腿。 上述步骤 将导致正在进行的会话的语音部分在用户层面从 EPC 变换到 MGW。
在层 2向目标小区的切换过程(在本地 UE和接入网上执行的 从 E-UTRAN到 GSM的切换, 步骤 15-21 ) , 这是在本地 UE和 接入网络处执行的 RAT之间的切换。
关于图 2的各步骤的细节, 可以参考 3GPP TS 23.216。
然而, 3GPP TS 23.216中并未指明如何在会话的非稳定状态 下支持 SRVCC。 非稳定状态可以是指会话的早期状态, 例如主叫 方收到临时 SIP响应消息(除了 100 Trying消息)而不是收到 200 OK消息。
因此, 需要一种能够在会话的非稳定状态下支持从 PS 域至 CS域的 SRVCC切换的机制。 发明内容
为了解决上述现有技术中的问题, 根据本发明的一个方面, 提出了一 种用于在呼叫的非稳定状态下将终端从分组交换域切换到电路交换域的方 法, 包括: 第一网络单元基于接收自所述终端的呼叫建立请求而锚定该呼 叫, 并且保存返回给所述终端的临时响应和该呼叫的状态; 所述终端检测 到需要进行切换并且向接入网发送测量报告; 只要所述接入网在该测量报 告满足切换条件的前提下只要检测到信令承载, 就判定向第二网络单元请 求切换; 所述第二网络单元在收到来自所述接入网的对切换的请求之后, 只要检测到所述信令承载, 就判定向所述第一网络单元发送分组交换域至 电路交换域的切换请求; 在所述第一网络单元收到该切换请求后, 如果检 测到该呼叫处于非稳定状态, 则检查之前保存的所述临时响应并且将所述 临时响应发送至第三网络单元, 由此继续完成会话切换过程并更新远程端。
根据本发明的另一个方面, 提出了一种管理服务的集中和连续性的网 络单元, 其能够在呼叫的非稳定状态下将终端从分组交换域切换到电路交 换域, 该网络单元包括: 保存装置, 用于在该呼叫基于接收自所述终端的 呼叫建立请求而被锚定后、 保存返回给所述终端的临时响应和该呼叫的状 态; 和, 临时响应检查和发送装置, 用于当收到来自另一网络单元的切换 请求时、 在检测到该呼叫处于非稳定状态的情况下检查之前保存的所述临 时响应并将该临时响应发送至又另一网络单元。
根据本发明的另一个方面, 提出了一种接入设备, 其能够在呼叫的非 稳定状态下将终端从分组交换域切换到电路交换域, 该接入设备包括判定 装置, 该判定装置用于在接收自所述终端的测量报告满足切换条件的前提 下只要检测到信令承载, 就判定向另一网络单元请求切换。
根据本发明的另一个方面,提出了一种管理终端的移动性的网络单元, 其能够在呼叫的非稳定状态下将所述终端从分组交换域切换到电路交换 域, 该网络单元包括判定装置, 该判定装置用于在收到来自接入网的对切 换的请求之后, 只要检测到信令承载就判定向另一网络单元发送分组交换 域至电路交换域的切换请求。
根据本发明的另一个方面, 提出了一种通信系统, 其至少包括: 至少 一个终端; 根据本发明的管理服务的集中和连续性的网络单元; 根据本发 明的接入设备; 和根据本发明的管理终端的移动性的网络单元。 附图说明
通过阅读下面结合附图对本发明具体实施例的说明, 本发明的上述及 其他特征和优点将变得更加明显。 其中: 图 1示例性示出了 TS 23.216中规定的用于 SRVCC的网络架 构;
图 2 示 例 性 示 出 了 现 有 技 术 中 从 E-UTRAN 到
GERAN/UTRAN的 SRVCC切换的信令流程;
图 3 示例 性 示 出 了 根据本发 明 的 从 E-UTRAN 到
GERAN/UTRAN的非稳定呼叫状态下的 SRVCC切换的信令流程; 图 4 是根据本发明一个实施例的用于在非稳定呼叫状态下进 行 SRVCC切换的方法的流程图;
图 5是根据本发明一个实施例的网络单元的框图;
图 6是根据本发明一个实施例的接入设备的框图; 以及 图 7是是根据本发明一个实施例的网络单元的框图; 具体实施方式
本发明的目的是能够在 VoIP呼叫仍处于非稳定状态下的情况 下实现从 E-UTRAN到 GERAN/UTRAN的 SRVCC切换, 其中所 述非稳定状态的一个例子是呼叫的早期状态。
因此, 本发明的基本思想是使得移动性管理实体 ( MME ) 和 eNodeB能够在检测到 IMS信令承载( QCI(服务质量等级标识符) =5 )的存在时发起 SRVCC切换过程。换句话说, MME当发现 IMS 信令承载存在时,即使不存在语音承载( QCI=1 ),也将发送 SRVCC PS至 CS切换请求至移动交换中心( MSC )服务器。在收到 SRVCC PS至 CS请求后, MSC服务器将向 IMS和 SCC AS发起会话切换 过程。 然后, SCC AS将基于接入腿( access leg ) 的状态和上一次 从 SCC AS发送给本地终端的响应 (该响应由 SCC AS事先保存) 来对 MSC服务器作出响应。 发送响应至 MSC服务器后, SCC AS 将完成会话切换、 更新远程端并释放初始接入腿。
应当指出, MSC J! 务器可以以下面两种方式连接到 IMS/SCC AS:
1. 如果 MSC服务器支持至 CSCF/SCC AS的 12/11接口, 则 MSC服务器直接经由 12/11接口连接到 IMS/SCC AS;
2. 如果 MSC J! 务器不支持至 CSCF/SCC AS的 12/11接口, 则 MSC服务器经由 MGCF/MGW连接到 IMS/SCC AS。
然而, 本发明适用于任一种方式。 为了简化, 下面仅描述第 一种方式。 然而, 本发明也可以用于第二种方式并且第二种方式 的呼叫流程类似于第一种方式。
应当指出,在本发明中, 需要稍微地对 eNB、 MME和 SCC AS 进行如下的改进:
- SCC AS 应当增强, 以便保存接入腿的状态以及上一次从 SCC AS发送到主叫方的响应消息。 如果 SCC AS在接入腿中发送 了 180 Ringing SIP消息至主叫方, 则它应当保存这个响应。 此外, 在收到来自 MSC服务器的会话切换发起请求后, SCC AS应当基 于接入腿状态和事先保存的响应而发送合适的响应至 MSC 服务 器;
MME应当增强, 以便检测 IMS信令承载并且基于本地策略 发送 SRVCC请求至 MSC J! 务器; 以及
- E-UTRAN应当增强, 以便当检测到存在 IMS信令承载时发 起 SRVCC切换。
可选地, 如果 SCC AS发送的临时响应是 180 Ringing SIP消 息,则 MSC服务器应当具有命令 MGW播放铃音给主叫方的能力。
图 3示出了非稳定状态下的呼叫 /会话的 SRVCC切换的信令 流程。
应当指出, 在这里, 由 MGW发送至 SCC AS的会话切换请 求按照 MSC服务器的能力可以是 ISUP消息或 SIP消息。 本发明 既适用于 ISUP情形也适用于 SIP情形。 为了简化, 这里以 SIP消 息为例进行描述。
在步骤 1中, 本地 UE发送 INVITE请求至 IMS和 SCC AS。 该请求按照正常的 IMS 会话建立过程而被转发至 S-CSCF (图中 未显示) 。 利用 iFC (初始过滤准则) 服务逻辑而将该请求转发 至 SCC AS。
在步骤 2中, SCC AS为了实现切换而锚定该会话。
在步骤 3中, SCC AS发送临时响应至 UE。 在这里, 所述临 时响应的例子可以是 180 Ringing或 183 Session Progress SIP消 在步骤 4 中, 本地 UE检测到可能需要进行从 E-UTRAN至 GERAN-UTRAN的切换并发送测量报告至 E-UTRAN。
在步骤 5中,基于该 UE的测量报告、以及可能地基于" SRVCC operation possible"指示、并且当检测到针对该 UE而建立的 QCI=5 承载( IMS信令)或 QCI=1承载( IMS语音)的存在时,源 E-UTRAN 判定触发至 GERAN的 SRVCC切换。这与图 2所示的信令流程不 同, 在这里可以基于 IMS信令承载的存在而判定触发切换, 由此 实现非稳定状态下的呼叫 /会话的 SRVCC切换过程。
在步骤 6中,源 E-UTRAN发送切换请求( Handover Required ) 消息至源 MME并指示该源 MME这是 SRVCC切换操作。
在步骤 7中, 源 MME检查有关的 QCI以及通用容器的类型。 如果存在语音承载 ( QCI=1 ) 并且容器类型指示了 SRVCC切换, 则该 MME执行与图 1中的步骤 4相同的过程(即 MME将语音承 载从非语音承载中划分出来并且准备向 MSC服务器发起 PC至 CS 切换过程)。如果容器类型指示了 SRVCC切换但不存在语音承载, 则根据本发明, MME 将进一步检查是否存在 IMS 信令承载 ( QCI=5 ) 。 如果存在 IMS信令承载, 则该 MME可以基于存储 于 MME中的运营商本地策略来准备发起 PS至 CS切换过程。 在 这里, 假设运营商本地策略允许 MME 当存在 IMS信令承载但不 存在语音承载时发起 PS至 CS切换过程。 因此, 该 MME将发送 SRVCC PS至 CS请求给 MSC服务器。
步骤 8至 12与图 2中的步骤 5至 9相同, 其具体描述可以参 考 3GPP TS 23.216, 在这里不再勢述。
在步骤 13中, MSC服务器通过发送具有 STN-SR (会话切换 号-单一无线)和 MGW的 SDP信息的 INVITE消息至 IMS , 来发 起会话切换。 STN-SR指示了针对 PS至 CS切换使用 SRVCC过程。
在步骤 14中, SCC AS基于 STN-SR而发现了被锚定的会话。 然后, SCC AS检测到接入腿处于非稳定状态, 即呼叫处于早期状 态。 之后, SCC AS检查上一次保存的 SIP临时响应并发送相应的 响应至 MSC J! 务器。 例如, 如果所保存的响应是 180 Ringing , 则 SCC AS将发送 180 Ringing消息至 MSC服务器。 如果所保存 的响应是 183 Session Progress消息,则 SCC AS将发送 183 Session Progress消息至 MSC服务器。 然而应当清楚, 如果 SCC AS检测 到接入腿处于稳定状态, 则按照原始的 SRVCC过程操作。
步骤 15说明了两种示例性选择,即 183 Session Progress消息 和 180 Ringin 消息。 在 180 Ringin 消息的情况下, MSC服务器 将对主叫方播放铃音。 然而应当理解, 本发明不限于这两种消息, 也可以针对非稳定状态而使用其他响应消息。
在步骤 16中, SCC AS继续完成会话切换过程并更新远程端。 在 3GPP TS 23.237中, 可以使用 UPDATE或 re-INVITE消息来 更新远程端。 然而应当指出, 在本发明中由于会话处于非稳定状 态而只能使用 UPDATE消息。
在步骤 17中, 由于初始接入腿处于非稳定状态, 因此 SCC AS 发送 CANCEL消息而不是 BYE消息来释放该接入腿。
应当指出, 尽管图 3只针对本地 UE是主叫方的情况, 然而本 发明也适用于本地 UE为被叫方的情况,并且信令流程与图 3相似。 为了简化, 在这里不再对其进行描述。
这样, 通过使用本发明的方案, 现在在呼叫 /会话处于非稳定 状态时也可以实现 SRVCC切换过程。 这无疑是对原始 SRVCC方 案的增强。
下面参照图 4来描述根据本发明一个实施例的用于在 VoIP呼叫的非 稳定状态下将终端从 E-UTRAN ( PS域)切换到 UTRAN/GERAN ( CS域) 的方法。 本实施例的方法可以适用于例如图 1所示的网络架构, 关于该网 络架构的描述在这里不再重复。
如图 4所示, 首先, 在步骤 401中, 第一网络单元基于接收自所述终 端的呼叫建立请求而锚定该呼叫, 并且保存返回给所述终端的临时响应和 该呼叫的状态。 在这里, 所述第一网络单元可以例如是图 3中的 SCC AS, 所述终端例如是图 3中的本地 UE, 所述呼叫建立请求可以例如是符合 SIP 协议的 INVITE消息, 并且所述临时响应例如可以是符合 SIP协议的 180 Ringing消息或 183 Session Progress消息。 此外, 应当指出, 呼叫的状态 在本实施例中是指呼叫的早期状态, 也就是说呼叫还未完全建立成功时的 状态。 具体地, 根据本实施例并参考图 3, 本地 UE发送 INVITE消息至 SCC AS以请求建立呼叫。 收到该 INVITE消息后, SCC AS锚定该呼叫, 并向本地 UE发送 180 Ringin 消息或 183 Session Progress消息并且保存 所发送的临时响应消息以及接入腿的状态。
接着, 在步骤 402中, 所述终端检测到需要进行切换并且向接入网发 送测量报告。 在这里, 所述接入网例如是图 3中的源 E-UTRAN。 具体地, 根据本实施例并参考图 3,本地 UE检测到可能需要进行从 E-UTRAN 至 GERAN-UTRAN 的 SRVCC 切换, 因此发送测量报告至源 E-UTRAN。
然后, 在步骤 403中, 只要所述接入网在该测量报告满足切换条件 的前提下检测到信令承载, 就判定向第二网络单元请求切换。 在这里, 所 述第二网络单元例如是图 3所示的源 MME。 此外, 在本实施例中, 检测 到信令承载是指针对该 UE 而建立的 QCI=5承载。 具体地, 根据本 实施例并参考图 3,基于该 UE的测量报告、以及可能地基于" SRVCC operation possible"指示, 并且当检测到信令承载的存在时, 源 E-UTRAN就可以判定触发至 GERAN的 SRVCC切换。
应当指出, 当检测到语音承载(即) 时, 源 E-UTRAN也将判 定触发至 GERAN 的 SRVCC 切换, 而此时的切换符合标准的 SRVCC切换过程。 具体流程可以参考 3GPP TS 23.216, 在这里不 再赘述。
接着, 在步骤 404中, 所述第二网络单元在收到来自所述接入网的 对切换的请求之后, 只要检测到所述信令承载, 就判定向所述第一网络单 元发送 PS至 CS的切换请求。 在这里, 所述切换请求例如是符合 SIP协议 的、 具有指示使用单一无线语音呼叫连续性切换的信息和呼叫参数的 INVITE 消息, 其中指示使用单一无线语音呼叫连续性切换的信息例如是 指示了针对 PS至 CS切换使用 SRVCC过程的 STN-SR,而所述呼叫 参数例如是来自媒体网关的 SDP。 具体地, 根据本实施例并参考图 3, 当 源 MME收到来自源 E-UTRAN的对切换的请求后, 检查其存储的通用容 器类型, 如果该容器类型指示了 SRVCC切换, 则只要能检测到信令承载 ( QCI=5 ) , 就可以基于运营商本地策略而发起 PS至 CS的 SRVCC切换 过程,也就是说向 SCC AS发送具有 STN-SR和 MGW的 SDP的 INVITE 消息。 在本发明中假设运营商本地策略允许 MME 当存在 IMS信令 承载但不存在语音承载时发起 PS至 CS的 SRVCC切换过程。
应当指出, 在现有技术中, 步骤 404 的具体过程可以例如如 下进行: MME首先发送 SRVCC PS至 CS请求给 MSC J! 务器。
MSC服务器进行切换准备并建立电路,然后通过发送具有 STN-SR 和 MGW的 SDP信息的 INVITE消息至 SCC AS来发起会话切换。 然而, 应当理解, 本发明并不限于上述过程, 并且本领域技术人 员可以设想任何能够实现切换的 MME与 SCC AS之间的交互。 最后, 在步骤 405中, 所述第一网络单元收到该切换请求后, 如果 检测到该呼叫处于非稳定状态, 则检查之前保存的所述临时响应并且将所 述临时响应发送至第三网络单元, 由此继续完成会话切换过程并更新远程 端。 在这里, 所述第三网络单元例如是图 3所示的 MSC服务器。 具体地, 根据本实施例并参考图 3, SCC AS基于 STN-SR而发现了被锚定的 会话。 然后, SCC AS检测到接入腿处于非稳定状态, 即呼叫的早 期状态。 之后, SCC AS检查上一次保存的 SIP临时响应并发送相 应的响应至 MSC服务器。例如,如果所保存的响应是 180 Ringing , 则 SCC AS将发送 180 Ringing消息至 MSC服务器; 如果所保存 的响应是 183 Session Progress消息,则 SCC AS将发送 183 Session Progress消息至 MSC J! 务器。 如果是 180 Ringing消息的情况, 则 MSC服务器将对主叫方播放铃音。 然而应当理解, 本发明并不 限于这两种消息, 也可以针对非稳定状态而使用其他响应消息。 然后, SCC AS继续完成会话切换过程并更新远程端。
应当指出 , 尽管在 3GPP TS 23.237 中指明 了可以使用 UPDATE或 re-INVITE消息来更新远程端, 然而在本发明中由于 会话处于非稳定状态而只能使用 UPDATE消息。
还应当指出, 由于初始接入腿处于非稳定状态, 因此 SCC AS 发送 CANCEL消息而不是 BYE消息来释放该接入腿。
尽管上面只针对本地 UE是主叫方的情况进行了描述,然而本 实施例也适用于本地 UE为被叫方的情况,并且方法流程与上面描 述的相似。 为了简化, 在这里不再对其进行描述。
通过以上描述可知, 通过使用本发明的方案, 现在在呼叫 /会 话处于非稳定状态时也可以实现 SRVCC切换过程,这增强了原始 SRVCC方案。
在同一发明构思下, 根据本发明的另一个方面, 提供了一种管理服务 的集中和连续性的网络单元, 其能够在呼叫的非稳定状态下将终端从分组 交换域切换到电路交换域。 下面就结合附图对其进行说明。
图 5示出了根据本发明一个实施例的管理服务的集中和连续性的网络 单元 500。 该网络单元 500包括保存装置 501和临时响应检查和发送装置 502。 其中, 保存装置 501用于在该呼叫基于接收自终端的呼叫建立请求而 被锚定后、 保存返回给所述终端的临时响应和该呼叫的状态。 例如, 当收 到 UE发送的 INVITE消息时, SCC AS锚定该呼叫并且发送 180 Ringing 或 183 Session Progress消息至该 UE, 然后保存装置 501保存发 送给该 UE 的消息和以及接入腿状态。 临时响应检查和发送装置 502 用于当收到来自另一网络单元(源 MME ) 的切换请求时、 在检测到该呼 叫处于非稳定状态的情况下检查之前保存的所述临时响应并将该临时响应 发送至又另一网络单元( MSC服务器) 。 例如, 当 SCC AS收到来自 源 MME的具有 STN-SR和 MGW的 SDP的 INVITE消息并且检测到接 入腿处于非稳定状态时, 临时响应检查和发送装置 502检查上一次保 存的 180 Ringing或 183 Session Progress响应消息并将相应的响 应消息发送至 MSC服务器。
在实施上, 本实施例的网络单元 500以及其包含的保存装置 501和临 时响应检查和发送装置 502,可以以软件、硬件或软件和硬件组合的方式来 实现。 例如, 本领域技术人员熟悉多种可用来实现这些部件的设备, 诸如 理 器、 微控制器、 专用集成电路 (ASIC )、 可编程逻辑设备 (PLD )和 /或现场可 编程门阵列(FPGA )等。 本实施例的网络单元的各个组成部分可以物理地分 开实现而操作上地相互连接。
在操作上, 上述结合图 5说明的实施例的管理服务的集中和连续性的 网络单元 500, 可以实现前面描述的用于在 VoIP呼叫的非稳定状态下将该 呼叫从 PS切换至 CS的方法。 通过使用该网络单元, 可以在呼叫 /会话处 于非稳定状态时实现 SRVCC切换过程, 从而增强原始的 SRVCC 方案。 在同一发明构思下, 根据本发明的另一方面, 提供了一种接入设备, 其能够在呼叫的非稳定状态下将终端从分组交换域切换到电路交换域。
图 6示出了根据本发明的一个实施例的接入设备 600,其包括判定装置 601。 其中, 判定装置 601用于在接收自终端的测量报告满足切换条件的前 提下只要检测到信令承载, 就判定向另一网络单元(源 MME )请求切换。 例如, 源 E-UTRAN收到 UE发送的测量报告, 如果该测量报告满足切换 条件并且只要能检测到信令承载, 即 QCI=5, 则判定装置 601 判定向源 MME发出切换请求。
在实施上,本实施例的接入设备 600以及其包含的判定装置 601,可以 以软件、 硬件或软件和硬件组合的方式来实现。 例如, 本领域技术人员熟悉 多种可用来实现这些部件的设备, 诸如微处理器、 微控制器、 专用集成电路 (ASIC )、 可编考 ϋ£辑设备 (PLD )和 /或现场可编程门阵列 (FPGA )等。
在操作上,上述结合图 6说明的实施例的接入设备 600,可以实现前面 描述的用于在 VoIP呼叫的非稳定状态下将该呼叫从 PS切换至 CS的方法。 通过使用该接入设备, 可以在呼叫 /会话处于非稳定状态时实现 SRVCC切换过程, 从而增强原始的 SRVCC方案。
在同一发明构思下, 根据本发明的另一方面, 提供了一种管理终端的 移动性的网络单元, 其能够在呼叫的非稳定状态下将所述终端从分组交换 域切换到电路交换域。
图 7示出了根据本发明的一个实施例的管理终端的移动性的网络单元 700, 其包括判定装置 701。 其中, 判定装置 701用于在收到来自接入网的 对切换的请求之后, 只要检测到信令承载就判定向另一网络单元发送分组 交换域至电路交换域的切换请求。例如, 当源 MME收到来自源 E-UTRAN 的切换请求后, 只要能检测到信令承载, 即 QCI=5, 判定装置 701就判定 向 SCC AS发送包含 STN-SR和 MGW的 SDP的 INVITE消息以请求进行 PS至 CS的 SRVCC切换过程。
在实施上,本实施例的网络单元 700以及其包含的判定装置 701,可以 以软件、 硬件或软件和硬件组合的方式来实现。 例如, 本领域技术人员熟悉 多种可用来实现这些部件的设备, 诸如微处理器、 微控制器、 专用集成电路
(ASIC )、 可编考 ϋ£辑设备 (PLD )和 /或现场可编程门阵列 (FPGA )等。
在操作上,上述结合图 7说明的实施例的网络单元 700,可以实现前面 描述的用于在 VoIP呼叫的非稳定状态下将该呼叫从 PS切换至 CS的方法。 通过使用该网络单元 700, 可以在呼叫 /会话处于非稳定状态时实现 SRVCC切换过程, 从而增强原始的 SRVCC方案。
在同一发明构思下, 根据本发明的又另一方面, 还提出了一种通信系 统, 该通信系统至少包括至少一个终端、 前面实施例中描述的管理服务的 集中和连续性的网络单元 500、接入设备 600以及管理终端的移动性的网络 单元 700。 此外, 所述通信系统还可以包括其他网络单元, 例如 MSC服务 器等。
例如,在本实施例的通信系统中,可以在 VoIP呼叫的非稳定状态下实 现该呼叫的 PS至 CS的 SRVCC切换。 具体的操作流程可以参见上文对根 据本发明实施例的用于在 VoIP呼叫的非稳定状态下将该呼叫从 PS切换至 CS的方法的描述, 在此不对其进行勢述。
以上虽然通过一些示例性的实施例对本发明的用于在 VoIP呼叫的非 稳定状态下将该呼叫从 PS切换至 CS的方法、管理服务集中和连续性的网 络单元、 接入设备、 管理终端移动性的网络单元、 以及包括至少一个终端、 管理服务集中和连续性的网络单元、 接入设备、 管理终端移动性的网络单 元的通信系统进行了详细的描述, 但是以上这些实施例并不是穷举的, 本 领域技术人员可以在本发明的精神和范围内实现各种变化和修改。 因此, 本发明并不限于这些实施例, 本发明的范围仅由所附权利要求为准。

Claims

权利要求
1.一种用于在呼叫的非稳定状态下将终端从分组交换域切换到电路交 换域的方法, 包括:
第一网络单元基于接收自所述终端的呼叫建立请求而锚定该呼叫, 并 且保存返回给所述终端的临时响应和该呼叫的状态;
所述终端检测到需要进行切换并且向接入网发送测量报告;
只要所述接入网在该测量报告满足切换条件的前提下只要检测到信令 承载, 就判定向第二网络单元请求切换;
所述第二网络单元在收到来自所述接入网的对切换的请求之后, 只要 检测到所述信令承载, 就判定向所述第一网络单元发送分组交换域至电路 交换域的切换请求; 以及
在所述第一网络单元收到该切换请求后, 如果检测到该呼叫处于非稳 定状态, 则检查之前保存的所述临时响应并且将所述临时响应发送至第三 网络单元, 由此继续完成会话切换过程并更新远程端。
2.根据权利要求 1 所述的方法, 其中, 所述切换是单一无线语音呼叫 连续性 SRVCC切换, 和 /或所述呼叫的非稳定状态是呼叫还未完全被建立 时的状态。
3.根据权利要求 1 或 1 所述的方法, 其中, 所述分组交换域是由 E-UTRAN网络提供的, 而所述电路交换域是由 GERAN或 UTRAN网络 提供的。
4.根据权利要求 1至 3中任一项所述的方法, 其中, 所述第一网络单 元是服务集中和连续性应用服务器 SCC AS , 和 /或所述接入网是源 E-UTRAN, 和 /或所述第二网络单元是源移动性管理实体 MME , 和 /或所述第三网络单元是移动交换中心 MSC服务器。
5.根据权利要求 1至 4中任一项所述的方法, 其中, 所述呼叫建立请 求是 INVITE消息, 和 /或所述临时响应是 180 Ringin 消息或 183 Session Progress消息,和 /或所述切换请求是具有指示使用单一无线语音呼叫连续 性切换的信息和呼叫参数的 INVITE消息。
6.根据权利要求 5所述的方法, 其中, 所述指示使用单一无线语音呼 叫连续性切换的信息是会话切换号-单一无线 STN-SR,和 /或所述呼叫参数 是会话描述协议 SDP。
7.根据权利要求 1至 6中任一项所述的方法, 其中, 检测到所述信令 承载是指检测到服务盾量等级标识符 QCI的值等于 5。
8.—种管理服务的集中和连续性的网络单元, 其能够在呼叫的非稳定 状态下将终端从分组交换域切换到电路交换域, 包括:
保存装置, 用于在该呼叫基于接收自所述终端的呼叫建立请求而被锚 定后、 保存返回给所述终端的临时响应和该呼叫的状态; 和
临时响应检查和发送装置, 用于当收到来自另一网络单元的切换请求 时、 在检测到该呼叫处于非稳定状态的情况下检查之前保存的所述临时响 应并将该临时响应发送至又另一网络单元。
9.根据权利要求 8所述的网络单元, 其中, 所述另一网络单元是源移 动性管理实体 MME , 和 /或所述又另一网络单元是移动交换中心 MSC服务器。
10. 根据权利要求 8或 9所述的网络单元, 其中, 所述切换是单一无 线语音呼叫连续性 SRVCC切换, 和 /或所述呼叫的非稳定状态是呼叫还未 完全被建立时的状态。
11. 根据权利要求 8至 10中任一项所述的网络单元,其中,所述分组 交换域是由 E-UTRAN 网络提供的, 而所述电路交换域是由 GERAN或 UTRAN网络提供的。
12. 根据权利要求 8至 11中任一项所述的网络单元,其中,所述呼叫 建立请求是 INVITE消息, 和 /或所述临时响应是 180 Ringin 消息或 183 Session Progress消息, 和 /或所述切换请求是具有指示使用单一无线语音 呼叫连续性切换的信息和呼叫参数的 INVITE消息。
13. 根据权利要求 12所述的网络单元,其中,所述指示使用单一无线 语音呼叫连续性切换的信息是会话切换号-单一无线 STN-SR,和 /或所述呼 叫参数是 ^^描述协议 SDP。
14. 一种接入设备, 其能够在呼叫的非稳定状态下将终端从分组交换 域切换到电路交换域, 包括:
判定装置, 用于在接收自所述终端的测量报告满足切换条件的前提下 只要检测到信令承载, 就判定向另一网络单元请求切换。
15. 根据权利要求 14 所述的接入设备, 其中, 所述接入设备位于 源 E-UTRAN 中, 和 /或所述另一网络单元是源移动性管理实体 MME。
16. 根据权利要求 14或 15所述的接入设备, 其中, 所述切换是单一 无线语音呼叫连续性 SRVCC切换, 和 /或所述呼叫的非稳定状态是呼叫还 未完全被建立时的状态。
17. 根据权利要求 14至 16中任一项所述的接入设备, 其中, 所述分 组交换域是由 E-UTRAN网络提供的, 而所述电路交换 由 GERAN或 UTRAN网络提供的。
18. 根据权利要求 14至 17中任一项所述的接入设备, 其中, 检测到 所述信令承载是指检测到服务盾量等级标识符 QCI的值等于 5。
19. 一种管理终端的移动性的网络单元, 其能够在呼叫的非稳定状态 下将所述终端从分组交换域切换到电路交换域, 包括:
判定装置, 用于在收到来自接入网的对切换的请求之后, 只要检测到 信令承载就判定向另一网络单元发送分组交换域至电路交换域的切换请 求。
20. 根据权利要求 19所述的网络单元,其中,所述网络单元是源移动 性管理实体 MME , 和 /或所述接入网是源 E-UTRAN, 和 /或所述另一 网络单元是服务集中和连续性应用服务器 SCC AS。
21. 根据权利要求 19或 20所述的网络单元, 其中, 所述切换是单一 无线语音呼叫连续性 SRVCC切换, 和 /或所述呼叫的非稳定状态是呼叫还 未完全被建立时的状态.
22. 根据权利要求 19至 21中任一项所述的网络单元, 其中, 所述分 组交换域是由 E-UTRAN网络提供的, 而所述电路交换 由 GERAN或 UTRAN网络提供的。
23. 根据权利要求 19至 22中任一项所述的网络单元, 其中, 所述切 换请求是具有指示使用单一无线语音呼叫连续性切换的信息和呼叫参数的 INVITE消息。
24. 根据权利要求 23所述的网络单元,其中,所述指示使用单一无线 语音呼叫连续性切换的信息是会话切换号-单一无线 STN-SR,和 /或所述呼 叫参数是 ^^描述协议 SDP。
25. 根据权利要求 19至 24中任一项所述的网络单元, 其中, 检测到 所述信令承载是指检测到服务盾量等级标识符 QCI的值等于 5。
26. 一种通信系统, 至少包括:
至少一个终端;
根据权利要求 8至 13所述的网络单元;
根据权利要求 14至 18所述的接入设备; 和
根据权利要求 19至 25所述的网络单元。
PCT/CN2009/073023 2009-07-31 2009-07-31 非稳定VoIP呼叫中的终端切换方法、网络单元、设备及系统 WO2011011922A1 (zh)

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EP09847708.6A EP2461627B1 (en) 2009-07-31 2009-07-31 Terminal handover method, network element, device and system in non-stable voice over ip (voip) calling,
KR1020127004993A KR20120047979A (ko) 2009-07-31 2009-07-31 불안정 보이스 오버 ip(voip)호에서의 단말 핸드오버 방법, 네트워크 요소, 디바이스 및 시스템
US13/388,125 US9374740B2 (en) 2009-07-31 2009-07-31 Method for handing over a terminal during a non steady VoIP call, network element, device and system
CN200980159990.0A CN102474789B (zh) 2009-07-31 2009-07-31 非稳定VoIP呼叫中的终端切换方法、网络单元、设备及系统
PCT/CN2009/073023 WO2011011922A1 (zh) 2009-07-31 2009-07-31 非稳定VoIP呼叫中的终端切换方法、网络单元、设备及系统
US15/090,034 US9807650B2 (en) 2009-07-31 2016-04-04 Method for handing over a terminal during a non steady VoIP call, network element, device and system

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