WO2008034385A1 - Dispositif utilisateur, serveur d'application de continuité d'appel et procédé de commutation de réseau - Google Patents

Dispositif utilisateur, serveur d'application de continuité d'appel et procédé de commutation de réseau Download PDF

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Publication number
WO2008034385A1
WO2008034385A1 PCT/CN2007/070666 CN2007070666W WO2008034385A1 WO 2008034385 A1 WO2008034385 A1 WO 2008034385A1 CN 2007070666 W CN2007070666 W CN 2007070666W WO 2008034385 A1 WO2008034385 A1 WO 2008034385A1
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WO
WIPO (PCT)
Prior art keywords
call
user equipment
network
session
description information
Prior art date
Application number
PCT/CN2007/070666
Other languages
English (en)
French (fr)
Inventor
Shuiping Long
Jie Xu
Yi Zhang
Fang You
Original Assignee
Huawei Technologies Co., Ltd.
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.)
Filing date
Publication date
Application filed by Huawei Technologies Co., Ltd. filed Critical Huawei Technologies Co., Ltd.
Publication of WO2008034385A1 publication Critical patent/WO2008034385A1/zh
Priority to US12/368,620 priority Critical patent/US8155084B2/en

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1083—In-session procedures
    • H04L65/1095—Inter-network session transfer or sharing
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M7/00—Arrangements for interconnection between switching centres
    • H04M7/12—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal
    • H04M7/1205—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal where the types of switching equipement comprises PSTN/ISDN equipment and switching equipment of networks other than PSTN/ISDN, e.g. Internet Protocol networks
    • H04M7/1225—Details of core network interconnection arrangements
    • H04M7/123—Details of core network interconnection arrangements where the packet-switched network is an Internet Protocol Multimedia System-type network
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W36/00—Hand-off or reselection arrangements
    • H04W36/0005—Control or signalling for completing the hand-off
    • H04W36/0011—Control or signalling for completing the hand-off for data sessions of end-to-end connection
    • H04W36/0022—Control or signalling for completing the hand-off for data sessions of end-to-end connection for transferring data sessions between adjacent core network technologies
    • H04W36/00224—Control 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]
    • H04W36/00226—Control 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]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/10—Architectures or entities
    • H04L65/1016—IP multimedia subsystem [IMS]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1069—Session establishment or de-establishment
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M3/00—Automatic or semi-automatic exchanges
    • H04M3/42—Systems providing special services or facilities to subscribers
    • H04M3/428—Arrangements for placing incoming calls on hold
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W80/00—Wireless network protocols or protocol adaptations to wireless operation
    • H04W80/08—Upper layer protocols
    • H04W80/10—Upper layer protocols adapted for application session management, e.g. SIP [Session Initiation Protocol]

Definitions

  • the present invention relates to the field of IP Multimedia Subsystem (IMS) technology, and more particularly to a method for handling voice call continuity (VCC) during ringing/ringback tone or during call hold.
  • IMS IP Multimedia Subsystem
  • VCC voice call continuity
  • Network switching method and related VCC Application Server VCC Application Server, VCC AS
  • VCC AS VCC Application Server
  • user equipment User Equipment
  • IMS is the subsystem supporting IP multimedia services proposed by the 3rd Generation Partnership Project (3GPP) in Release 5 (R5). Its core feature is the Session Initial Protocol. , SIP) protocol and its independence from access, IMS provides a common service platform for future multimedia applications, and it is an important step towards the provision of an all-IP network service system.
  • 3GPP 3rd Generation Partnership Project
  • R5 Release 5
  • SIP Session Initial Protocol
  • the IMS standard has evolved R5 and R6 versions with the 3GPP standard system. After the R6 version was basically stabilized in March 2005, 3GPP introduced the R7 version, which added some new functions while enhancing the original system functions.
  • the VCC function is a new function proposed by the R7 version to solve the problem.
  • VCC refers to the continuity of voice calls, that is, when users move between various access technologies, their calls remain contiguous.
  • users from traditional 2G networks such as GSM, CDMA, CDMA
  • 3G network such as Universal Mobile Telecommunications System UMTS, High Rate Packet Data HRPD
  • 3G services such as high-speed Internet access
  • CS Circuit Switched
  • 2G 2G network
  • CS Circuit Switched
  • the medium network element VCC AS can be used to control the VCC handover.
  • the network element that the user enters into the CS network through the IMS network is a media gateway control function unit (MGGW), and the MGCF also controls the media gateway (MGGW). Loaded route.
  • MGGW media gateway control function unit
  • the current VCC handover assumption is made in a call state, and the VCC handover is not considered during the ringing/ringback tone (master/called situation) and during the call hold, which may cause problems such as the user equipment leaving the IMS hotspot coverage and the user During the ringing/ringback tone or the call hold period of the device, the user equipment may be dropped due to the VCC network switching, which may affect the quality of the user's call.
  • the technical problem to be solved by the embodiments of the present invention is to provide a network switching method for a user equipment, a call continuity application server, and a user equipment, to support a network in which a user equipment performs call continuity during a ringing/ringback tone or during call hold. Switching to avoid user device dropped calls and improve user call quality.
  • an embodiment of the present invention provides a network switching method for a user equipment, where the method mainly includes: a session negotiation in which a user equipment that initiates handover performs a call continuity handover in a call hold state through a target network; After the negotiation is successful, the call is switched to the target network.
  • the embodiment of the present invention further provides a network switching method for a user equipment, where the method includes: receiving a network handover request from a user equipment that initiates handover, where the network handover request carries the user equipment session description information;
  • the call description is a call hold state, and the session description information is updated, and the session description information of the call hold state of the user equipment that initiates the handover is sent to the opposite end;
  • the corresponding call is switched to the target network according to the network handover request.
  • the embodiment of the present invention provides a user equipment, where the user equipment includes: a session negotiation processing unit, and a session negotiation for performing call continuity switching in a call hold state by using a target network;
  • the continuity switching unit is configured to switch the call to the target network after the session negotiation succeeds.
  • the embodiment of the present invention further provides a call continuity application server, including: a call continuity session negotiation processing unit during call hold, configured to control a call to switch to a target network during call hold Call negotiation; call continuity switching control unit during call hold, used to control network switching of call continuity during call hold after successful session negotiation.
  • a call continuity application server including: a call continuity session negotiation processing unit during call hold, configured to control a call to switch to a target network during call hold Call negotiation; call continuity switching control unit during call hold, used to control network switching of call continuity during call hold after successful session negotiation.
  • the embodiment of the present invention further provides a network switching method for a user equipment, where the method includes: a session negotiation in which a called user equipment performs a call continuity handover during a ringing period with a remote calling user equipment through the target network; After the negotiation succeeds, the call is switched to the target network, and the called user equipment sends an off-hook signal to the opposite calling user equipment.
  • the embodiment of the present invention provides a network switching method for a user equipment, where the to-be-switched network is an IP multimedia subsystem network, and the target network is a circuit domain network, wherein the method includes: the calling user equipment passes the target The network negotiates with the peer called user equipment for the call continuity switch during the ring back tone; after the session negotiation succeeds, the call is switched to the target network; the calling user equipment receives the off-hook signal sent by the opposite end.
  • the embodiment of the present invention further provides a user equipment, where the user equipment includes: a session negotiation processing unit, configured to perform session negotiation of call continuity during ringing through a target network with a remote calling user equipment; a switching unit, configured to perform network switching of call continuity during ringing after the session negotiation succeeds; an off-hook signal sending unit, configured to send, according to the handover result of the call continuity switching unit, to the opposite calling user equipment Send the off-hook signal.
  • a session negotiation processing unit configured to perform session negotiation of call continuity during ringing through a target network with a remote calling user equipment
  • a switching unit configured to perform network switching of call continuity during ringing after the session negotiation succeeds
  • an off-hook signal sending unit configured to send, according to the handover result of the call continuity switching unit, to the opposite calling user equipment Send the off-hook signal.
  • the embodiment of the invention further provides a user equipment, where the user equipment includes:
  • a session negotiation processing unit configured to perform a session negotiation of a call continuity switch during a ringback tone with a peer called user equipment by using a target network
  • a call continuity switching unit during a ringback tone configured to perform after the session negotiation is successful
  • the receiving unit is configured to receive an off-hook signal sent by the opposite party called user equipment after the network handover is completed.
  • the embodiment of the present invention further provides a call continuity application server, including: a call continuity session negotiation control unit during the ring/ringback tone, configured to control network switching during call ringing during ringing/ringback tone The session negotiation; the off-hook signal identification transmitting unit is configured to identify the off-hook signal of the called user equipment and transmit the signal to the calling user equipment.
  • one or more embodiments of the present invention have the following beneficial effects:
  • one party negotiates with the other party through the call continuity application server, and negotiates After passing, through the call continuity application server
  • the peer sends an off-hook signal to implement network handover of call continuity during ringing or ringback tone.
  • the user equipment that initiates the handover is in the target network through the call continuity application server and the pair.
  • the terminal performs the session negotiation in the call hold state, and switches to the target network after the session negotiation is completed, thereby realizing the call continuity switching during the call hold period, so that even during the ringing or ring back tone and during the call hold period, even the user equipment
  • the user equipment can also switch to another network through the call continuity application server, which can effectively prevent the user equipment from falling out during ringing or ring back tone and during call hold, thereby improving the quality of the user's call. .
  • FIG. 2 is a main flowchart of an embodiment of a method for a called user equipment to perform network switching during ringing according to the present invention
  • FIG. 3 is a main flowchart of an embodiment of a method for a network call of a calling user equipment during a ring back tone of the present invention
  • FIG. 4 is a flow diagram of an embodiment of a called user equipment switching from a circuit domain to an IMS domain during ringing of the present invention
  • FIG. 5 is a flow chart showing an embodiment of a method for a called user equipment to switch from an IMS domain to a circuit domain during ringing according to the present invention
  • FIG. 6 is a flow chart showing an embodiment of a method for a calling user equipment to switch from an IMS domain to a circuit domain during a ringback tone of the present invention
  • FIG. 7 is a flow chart of a first embodiment of a user equipment switching from a circuit domain network to an IMS domain during call hold of the present invention
  • Figure 8 is a flow chart showing a second embodiment of the user equipment switching from the circuit domain network to the IMS domain during call hold of the present invention
  • FIG. 9 is a flow chart showing a third embodiment of the user equipment switching from the circuit domain network to the IMS domain during call hold of the present invention.
  • FIG. 10 is a flow chart of a first embodiment of a user equipment switching from an IMS domain network to a circuit domain during call hold of the present invention
  • FIG. 11 is a second embodiment of the user equipment switching from the IMS domain network to the circuit domain during call hold of the present invention.
  • FIG. 12 is a schematic structural diagram of performing VCC network switching during ringing by a user equipment according to an embodiment of the present invention
  • FIG. 13 is a schematic structural diagram of performing VCC network switching during a ringback tone of a user equipment according to an embodiment of the present invention
  • FIG. 14 is a schematic structural diagram of performing VCC network switching control during ringing or ring back tone of a VCC application server according to an embodiment of the present invention
  • FIG. 15 is a schematic structural diagram of a VCC network handover performed by a user equipment during call hold according to an embodiment of the present invention.
  • FIG. 16 is a schematic structural diagram of performing VCC network switching control during call hold of a VCC application server according to an embodiment of the present invention.
  • the call continuity application server includes a voice, video and/or audio and video call continuity application server, which is described in the present invention.
  • the call continuity application server takes the VCC application server as an example, and performs the same session negotiation with the other party on the opposite end. After the negotiation is passed, the VCC application server sends an off-hook signal to the opposite end to implement ringing or back. VCC network switching during the ring tone.
  • the user equipment that initiates the handover negotiates with the peer in the call hold state through the voice call continuity application server, and switches to the target after the session negotiation is completed.
  • the internet the internet.
  • the network switching process and the network switching process during call hold are separately described.
  • An embodiment of the present invention provides a network switching method for a user equipment, where the method includes: a session negotiation in which a user equipment that initiates handover performs a call continuity handover in a call hold state through a target network; Switch the call to the target network after success.
  • the user equipment negotiates with the peer in the call hold state through the call continuity application server in the target network.
  • the method may further include: the user equipment connects to the call The continuation application server sends a network switching request that carries the session description information of the user equipment; the session negotiation further refers to: performing session negotiation according to the session description information; and switching the call to the target network after the session negotiation is successful. After the session negotiation is successful according to the session description information, the call corresponding to the network handover request is switched to the target network.
  • the session description information carries the call state holding information of the user equipment that initiates the handover.
  • the user equipment that initiates the handover When the network to be switched by the user equipment that initiates the handover is a circuit domain network, and the target network is an IP multimedia subsystem network, the user equipment that initiates the handover performs a call hold state with the opposite end through the call continuity application server in the target network.
  • the specific negotiation of the session includes: the user equipment that initiates the handover sends a network handover request carrying the session description information of the user equipment to the call continuity application server through the IP multimedia subsystem network.
  • the specific negotiation of the user equipment that initiates the handover in the target network through the call continuity application server and the peer end in the call hold state includes: The switched user equipment sends an invitation message to the call continuity application server through the circuit domain network, the mobile switching center, and the media gateway control function unit, where the invitation message includes: the session description information of the media gateway control function unit.
  • a network switching method of a user equipment includes: receiving a network handover request from a user equipment that initiates handover, where the network handover request carries the user equipment session description information; And the session description information is updated, and the session description information of the call holding state of the user equipment that initiates the handover is sent to the peer end; further, the user equipment and the peer end pass After the session description information session is successfully negotiated, the corresponding call is switched to the target network according to the network handover request.
  • the method may further include: receiving session description information from the peer end that carries the call hold status of the peer end; and sending, to the user equipment that initiates the handover, session description information that carries the call hold status of the opposite end.
  • the detected call hold status is the call hold initiated by the user equipment that initiates the handover; the session description information of the call hold of the user equipment that initiates the handover is only the sent session description information; The session description information of the call hold state is only the received session description. Information.
  • the call hold status detected by the call continuity application server is a call hold initiated by the peer end, and the session description information of the call hold of the user equipment that initiates the handover is only received session description information;
  • the session description information of the call hold is only the sent session description letter '&.
  • the only received session description information is added by the user equipment to be switched or the call continuity application server.
  • the method further includes: releasing a call leg of the network to be switched.
  • the media gateway control function unit receives an invitation message sent by the user equipment that initiates the handover through the circuit domain and the mobile switching center, where the invitation message includes:
  • the media gateway controls the session description information of the function unit; the session description information that is updated by the session description information and sent to the peer end to carry the call hold state of the user equipment that initiates the handover further refers to: generating, according to the session invitation message
  • the session describes an update message of the information, and sends an invitation or update message carrying the session description information of the call hold status of the media gateway control function unit to the peer end.
  • the session description information of the call hold status of the media gateway control function unit is only sent session description information;
  • the session description information of the call hold state of the peer end is only the received session description information.
  • the detected call hold status is a call hold initiated by the opposite end;
  • the session description information of the call hold status of the media gateway control function unit is only received session description information;
  • the session description information is only the sent session description information.
  • a network switching method of a user equipment includes: a session negotiation in which a called user equipment performs a call continuity handover during a ringing period with a peer calling user equipment through a target network; After successful, the call is switched to the target network, and the called user equipment sends an off-hook signal to the opposite calling user equipment.
  • the called user equipment passes the call continuity application service of the target network
  • the session negotiation of the call continuity switch during ringing with the peer calling user equipment.
  • the call is switched to the target network, and the process of the called user equipment sending the off-hook signal to the opposite-end calling user equipment includes: after the handover to the target network, the called user equipment sends the off-hook signal And the call continuity application server detects that the called user equipment is off-hook, and sends an off-hook signal to the opposite calling user equipment.
  • the process of the called user equipment sending an off-hook signal to the call continuity application server is: sending an off-hook signal to the call by using a session initiation protocol notification message or an unstructured supplementary service data service or a short message service Call the continuity application server.
  • the method further includes: the call continuity application server releasing the call leg of the network to be switched.
  • the called user equipment switches the call continuity during the ringing of the target network by the call continuity application server and the opposite calling user equipment.
  • the specific process of the session negotiation includes: the called user equipment sends a session invitation message carrying the session description information of the called user equipment to the call continuity application server through the IP multimedia subsystem network; the call continuity application server detects the location The call is in a non-call state, generates an update message according to the session invitation message, and sends the update message to the peer calling user equipment; the peer calling user equipment feeds back to the call continuity application server to carry the pair An update response message of the session description information of the calling user equipment;
  • the call continuity application server generates a temporary response message according to the update response message, and sends a temporary response message carrying the tongue description information of the opposite calling user equipment to the called user equipment.
  • the called user equipment negotiates with the peer calling user equipment during the ringing period of the call continuity switching by the calling continuity application server
  • the specific process includes: the called user equipment sends a start call message to the mobile switching center through the circuit domain network; the mobile switching center generates the initial message according to the initial call message.
  • the start address message is sent to the media gateway control function unit; the media gateway control function unit parses the initial address message, obtains the number of the called user equipment and the call continuity application server, feeds back the address full information to the mobile switching center, and continuously forwards the call
  • the sexual application server sends a session invitation message carrying the media gateway control function unit session description information, where the tongue invitation message includes the number of the called user equipment and the call continuity application server; the call continuity application server detects the call In the non-call state, the update message of the session description information is generated according to the session invitation message, and the update message is sent to the opposite calling user equipment; the opposite calling user equipment sends the call continuity application server to the call continuity application server.
  • the call continuity application server generates a temporary response message of the session description information of the opposite calling user equipment according to the update response message, and Transmitting a temporary response message to the media Off Control Function.
  • a network switching method for a user equipment where the to-be-switched network is an IP multimedia subsystem network, and the target network is a circuit domain network, and the method includes: the calling user equipment passes the target network.
  • the calling user equipment passes the target network.
  • the calling user equipment negotiates the session continuity call during the ring back tone by the call continuity application server of the target network and the peer called user equipment.
  • the method further includes: the peer called user equipment sends an off-hook signal to the call continuity application server after the session negotiation succeeds; the call continuity application The server sends the received off-hook signal to the calling user equipment.
  • the specific process of the session negotiation between the calling user equipment and the calling user equipment during the ring back tone call continuity handover includes: the calling user equipment sends a start call message to the mobile switching center through the circuit domain network; The mobile switching center generates initial address information according to the initial call message and sends the initial address information to the media gateway control function unit.
  • the media gateway control function unit parses the initial address message to obtain the number of the calling user equipment and the call continuity application server.
  • the call continuity application server sends the information to the mobile switching center, and transmitting the control function of the media gateway to the call continuity application server a session invitation message of the unit session description information, where the session invitation message includes a number of the calling user equipment and the call continuity application server; the call continuity application server detects that the call is a non-call state, according to the session invitation The message generates an update message of the session description information, and sends the update message to the peer called user equipment; the peer called user equipment feeds back to the call continuity application server to carry the peer called user equipment.
  • An update response message of the session description information; the call continuity application server generates a temporary response message of the session description information of the opposite calling user equipment according to the update response message, and sends the temporary response message to the media gateway Control function unit.
  • a network switching method of a user equipment includes: a session negotiation in which a user equipment that initiates handover performs a call continuity handover in a call hold state through a target network; after successful session negotiation Switch the call to the target network.
  • the user equipment negotiates with the peer in the call hold state through the call continuity application server in the target network.
  • the call continuity application server releases the call leg of the network to be switched.
  • the network to be switched is a circuit domain network
  • the target network is an IP multimedia subsystem network.
  • the specific process of the user equipment that initiates the handover through the call continuity application server in the target network and the peer in the call hold state The user equipment that initiates the handover sends a network handover request carrying the session description information of the user equipment to the call continuity application server through the IP multimedia subsystem network; if the call continuity application server detects that the call is a call hold state And generating, according to the invitation message, an update message; and sending, to the peer end, session description information that carries the call hold status of the user equipment that initiates the handover; and the peer end feedbacks that the call continuity application server carries the call hold of the opposite end
  • the call hold state detected by the call continuity application server is the call hold initiated by the user equipment that initiates the handover; the session description information of the call hold of the user equipment that initiates the handover is only the session description information sent; The session description information of the peer call hold state is only the received session description information.
  • the call hold status detected by the call continuity application server is held by the peer end; the session description information of the call hold of the user equipment that initiates the handover is only received session description information; the call hold of the opposite end The session description information is only the session description information sent.
  • the received session description information is added by the user equipment to be switched or the call continuity application server.
  • the specific process of the user equipment that initiates the handover in the target network through the call continuity application server and the peer in the call hold state includes: The switched user equipment sends a start message to the mobile switching center through the circuit domain network; the mobile switching center generates initial address information according to the initial call message and sends the information to the media gateway control function unit; the media gateway control function unit.
  • the call continuity application server sends an invitation message, where the invitation message includes: the session description information of the media gateway control function unit; the call continuity application server detects that the call is in a call hold state, and generates according to the session invitation message.
  • the call hold status detected by the call continuity application server is a call hold initiated by the user equipment that initiates the handover;
  • the session description information of the call hold status of the media gateway control function unit is only the sent session description information.
  • the session description information of the call hold state of the peer end is only the received session description information;
  • the user equipment that initiates the handover synchronizes with the mobile switching center to maintain the state of the call.
  • the call hold status detected by the call continuity application server is a call hold initiated by the opposite end;
  • the session description information of the call hold status of the media gateway control function unit is only received session description information;
  • the opposite end call The session description information of the hold state is only the session description information sent;
  • the method further includes: the call continuity application server and the circuit domain network are synchronized Called to keep the state.
  • the figure is a main flowchart of a method for performing network switching by a called user equipment during real ringing according to the present invention. Specifically, the method mainly includes the following steps:
  • Step s11 The called user equipment negotiates with the peer calling user equipment during the ringing period of the call continuity handover through the target network;
  • the called user equipment negotiates the session continuity call during the ringing of the target network by the voice call continuity application server and the opposite calling user equipment.
  • the session negotiation is performed according to the target network of the handover.
  • the specific processing flow is also different;
  • Step sl2 after the session negotiation is successful, the call is switched to the target network, and the called user equipment sends an off-hook signal to the opposite calling user equipment.
  • the called user equipment switches to the target network after the session negotiation succeeds, and sends an off-hook signal to the opposite calling user equipment through the voice call continuity application server, that is, the called user equipment sends the off-hook signal to
  • the voice call continuity application server sends the off-hook signal to the opposite calling user equipment when the voice call continuity application server detects that the called user equipment is off-hook.
  • the voice call continuity application server may further release the network call leg to be switched by the voice call continuity application server after detecting that the called user equipment is off-hook.
  • Other ways to remove the network call branch to be switched are not mentioned here.
  • the figure is a main flowchart of a method for performing network switching by a calling user equipment during a ring back tone of the present invention.
  • the network to be switched uses an IP multimedia subsystem network as an example, and the target network is a circuit domain network.
  • the calling user equipment performs network switching during the ring back tone mainly includes the following steps:
  • Step s21 The calling user equipment negotiates with the peer called user equipment to perform voice call continuity switching during the ring back tone period through the target network;
  • the calling user equipment negotiates a session of the voice call continuity switch during the ring back tone of the ringback tone by the voice call continuity application server in the circuit domain network;
  • Step s22 After the session negotiation succeeds, the call is switched to the target network; the calling user equipment receives the off-hook signal sent by the opposite end; That is, the peer called user equipment sends an off-hook signal, and the voice call continuity application server detects the off-hook signal and sends it to the calling user equipment to complete the handover.
  • Step s101 peer ( OTHER END POINT, OEP ) (including the calling terminal and the related network).
  • the Interrogating - Call Session Control Function sends a session invitation INVITE message to the IMS network to which the UE1 belongs.
  • Select a service call session control function S-CSCF
  • the INVITE message includes the Uniform Resource Identifier (URI) of the called UE 1.
  • Information such as Bob@huawei.com, and the Session Description Protocol (SDP) information of the peer (including the type of media that you want to establish, the encoding method, the IP address and port of the media, and so on).
  • SDP Session Description Protocol
  • the S-CSCF sends an INVITE message to the VCC AS, and the VCC AS acts as a
  • the SIP back-to-back user agent (B2BUA) establishes a SIP call dialog 1 with the OEP.
  • Steps s102 to s105 the VCC AS determines that the call should be delivered through the CS network, and obtains a Temporary Local Directory Number (TLDN) from the home location register HLR through the mobile application protocol.
  • TLDN Temporary Local Directory Number
  • the mobile application can be used.
  • the Mobile Application Protocol (MAP) message may also be in other corresponding protocols.
  • the MAP message includes: a Localizer Request (LOCREQ) and a Router Request (ROUTREQ).
  • Step si 06 the VCC AS modifies the previously received INVITE message, and modifies the request URI to an E.164 number that can be routed to the CS domain (such as the temporary local telephone number TLDN of the CDMA network, the mobile terminal routing number of the GSM/UMTS network) MSRN) is sent to the MGCF through the S-CSCF, and the establishment of the SIP call dialog 2 between the VCC AS and the MGCF is started by the INVITE message.
  • the CS domain such as the temporary local telephone number TLDN of the CDMA network, the mobile terminal routing number of the GSM/UMTS network
  • Step s107 to step si 08 the MGCF converts the INVITE message to the initial address message IAM, and sends the IAM message to the MSC/VLR, where the IAM message includes the bearer information TRK ( trunk ) between the allocated MGW and the MSC, MSC /VLR returns the address full message ACM.
  • step sl09 the UE 1 completes the call setup with the MSC/VLR, and generates a ringing tone locally.
  • Steps sll0 to slll the MGCF sends a 180 ringing message to the VCC AS through the S-CSCF, where the 180 ringing message includes the allocated MGW bearer (used to connect the peer end) information MGW SDP, and the VCC AS will The ringing message is forwarded to the peer end, that is, the VCC AS forwards the 180 ringing message to the I-CSCF through the S-CSCF, and the I-CSCF forwards the 180 ringing message to the opposite end.
  • MGW bearer used to connect the peer end
  • step s101-step sil l is a process for the incoming call to be sent through the circuit domain.
  • the present invention can also be used after the call is sent, and the UE according to some predetermined policies (such as the wireless condition changes, the call sent by the CS network has been established, and is waiting User response, etc.)
  • some predetermined policies such as the wireless condition changes, the call sent by the CS network has been established, and is waiting User response, etc.
  • Step sll2 the UE 1 sends an INVITE message to the VCC AS through the S-CSCF, where the INVITE message includes a Request URI (the Request URI is an E.164 number of the VCC AS), and may also include the SDP information of the UE 1;
  • the INVITE message includes a Request URI (the Request URI is an E.164 number of the VCC AS), and may also include the SDP information of the UE 1;
  • the VCC AS checks the call status to the non-call state, generates an update UPDATE message according to the received INVITE message (the request URI is modified to the address of the peer end in the message), and sends the response to the peer end, and the peer end responds.
  • a 200 OK response message where the 200 OK response message includes SDP information of the peer end response;
  • Step sll5 the 200 OK response message received by the VCC AS generates a temporary response message (183 message in this embodiment), where the temporary response message (ie, 183 message) includes SDP information of the peer end response, and the VCC AS will The temporary response message is sent to the UE1, and then the VCC AS waits for the user to go off-hook, and performs step si 16, or the UE decides that the CS network signal is unavailable, then performs step si 19; step sll6 ⁇ step sll8, UE1 detects that the user picks up the phone, passes Call direction established by CS network
  • the MSC/VLR sends an off-hook signal (such as sent by a CONNECT signaling message), and when the MSC/VLR detects the off-hook signal, it sends ANM information to the MGCF, and the MGCF converts the ANM information into a 200 OK response message (for the final step sl06) Response) sent to the VCC AS;
  • the UE1 sends a MESSAGE message to the VCC AS through the IP access network, where the MESSAGE message body indicates that the user picks up the phone, and after the VCC AS correctly parses, responds to the MESSAGE message with a 200 OK response message.
  • Step sl20 (including steps sl20a-sl20e), the VCC AS releases the call leg of the CS network side (including SIP call dialog 2);
  • the VCCAS may send a BYE message to the MGCF, and for the step si19, the CANCEL message shall be sent instead, and the MGCF responds to the 200 OK message; the MGCF sends the RLS (release message) to the MSC.
  • the MSC sends the RLS to the UE 1 to release the wireless connection, the UE returns to release the complete radio link control RLC message, and the MSC returns a release complete RLC message to the MGCF.
  • Step sl21 ⁇ step sl22 the VCC AS generates a 200 OK message (off-hook signal, the final response to the step slOl) is sent to the opposite end, and the opposite end responds with an ACK message;
  • Step sl23 ⁇ step sl24 the VCC AS generates a 200 OK message (the final response of step sll2) to the UE 1, and the UE 1 responds with an ACK message.
  • FIG. 5 is a flowchart of an embodiment in which the called user equipment is switched from the IMS domain to the circuit domain during the ringing of the present invention
  • the message in this embodiment is based on the SIP protocol, and the specific process is as follows:
  • Step s 201 The peer end (including the calling terminal and the related network) sends a session invitation INVITE message, and selects an S-CSCF and forwards the INVITE message to the I-CSCF, the I-CSCF of the IMS network to which the UE1 belongs.
  • the NVITE message includes the URI information of the called UE 1, such as Bob@huawei.com, and also includes the SDP information of the peer: the type of media to be established, the encoding mode, the IP address and port of the media, and the like.
  • the I-CSCF of the IMS network to which UE1 belongs, the I-CSCF selects an S-CSCF and forwards the INVITE message thereto.
  • the S-CSCF sends an INVITE message to the VCC AS, and the VCC AS acts as a SIP B2BUA to establish a SIP call dialog 1 with the OEP.
  • step s202 the VCC AS decides that the call needs to be sent through the IMS domain, and then returns an INVITE message to the S-CSCF, and the S-CSCF sends an INVITE message to the UE 1, and the SIP call dialog 2 between the UE 1 and the VCC AS starts to be established.
  • step s203 the UE 1 returns a 180 Ringing message to the S-CSCF and carries its own SDP information (received media type, encoding mode, media transceiving IP address and port, etc.) and the S-CSCF forwards to the VCC AS.
  • SDP information received media type, encoding mode, media transceiving IP address and port, etc.
  • Step s204 The VCC AS returns a 180 ringing message to the I-CSCF through the S-CSCF, and finally sends it to the OEP by the I-CSCF.
  • the UE 1 generates a local ringing tone, and then waits for the user of the UE 1 to go off.
  • the UE 1 decides to switch to the CS network according to some predetermined policies (for example, the wireless condition changes, the call has completed media negotiation), and then starts the VCC network. To switch, perform the following steps:
  • step s205 the UE 1 initiates a call through the CS network, and sends a Call Origination message to the MSC/VLR, where the Call Origination includes a CdPN (that is, the called number is (Service Code plus VCC AS E.164 number)) and CgPN (that is, the calling number is the number 1 of the UE 1 (Mobile Directory Number, such as 133XXXX5678));
  • CdPN that is, the called number is (Service Code plus VCC AS E.164 number)
  • CgPN that is, the calling number is the number 1 of the UE 1 (Mobile Directory Number, such as 133XXXX5678)
  • Step s206 The MSC/VLR generates an initial address message IAM and sends it to the MGCF, using the E.164 number of the VCC AS as the called number, the number of the UE 1 as the calling number, and the MSC/VLR discarding the Service Code value;
  • Step s207 The MSC/VLR sends a Traffic Channel Assignment message to the UE1, that is, the CS network allocates a traffic channel to the UE1, and the UE1 captures the voice channel;
  • Step s208 the MGCF sends an address full message ACM to the MSC/VLR;
  • Step s209 the MGCF generates an INVITE message (including bearer information allocated for the call) -
  • Step s210 ⁇ step s211 the VCC AS checks that the call state is non-calling state, generates an UPDATE message according to the received INVITE message (the request URI is changed to the address of the peer end in the message), and sends it to the peer end, and the peer end responds 200 OK.
  • the 200 OK message includes SDP information of the peer end response;
  • Step s212 the VCC AS generates a 183 message according to the received 200 OK message, sends it to the UE 1, and then waits for the user of the UE 1 to go off-hook;
  • the UE 1 detects that the user picks up the phone (may occur at any time after the step s215), sends an off-hook signal to the VCC AS through the 200 OK response message, and the VCC AS recognizes that the user picks up the phone; or if the UE 1 detects If the user picks up the phone, but the IP access network connection is not available, the off-hook information may be sent through the bearer mode of the unstructured supplementary service data (USSD) or the short message service (SMS) of the CS network in step s214. (such as encapsulating 200 OK message) sent to VCC AS;
  • USSD unstructured supplementary service data
  • SMS short message service
  • Step s215 ⁇ step s216, VCC AS releases SIP call dialog 2;
  • the VCC AS initiates the release of the BYE message sent by the SIP call dialog 2 for step s213.
  • the VCC AS may send a CANCEL message instead to initiate the release of the SIP call dialog 2;
  • Step s219 ⁇ step s220 the VCC AS generates a 200 OK response message (which is the final response to step s209) and sends it to the MGCF.
  • the MGCF can reply with an ACK message, and the MGCF can also send a CON/ANM (connection or address full message) to the MSC. Indicates that a call can be made.
  • FIG. 6 is a flowchart of an embodiment of a method for a calling user equipment to switch from an IMS domain to a circuit domain during a ringback tone according to the present invention.
  • the message in this embodiment is based on a SIP protocol, and the specific process is as follows:
  • Step s301 The UE 1 initiates a call through the IMS network, and sends an INVITE message to the S-CSCF of the IMS network to which the UE 1 belongs.
  • the INVITE message includes the address of the called user equipment of the peer end, such as Bob@huawei.com, including the SDP information of the UE 1: the type of the media to be established, the encoding mode, the IP address and port of the media, and the port.
  • the S-CSCF sends an INVITE message to the VCC AS, and the VCC AS establishes a SIP call dialog 1 with the UE 1 as a SIP B2BUA.
  • Step s302 the VCC AS returns an INVITE message to the S-CSCF, and then sends it to the peer to start establishment of the SIP call dialog 2 of the VCC AS and the peer.
  • Step s303 the peer returns a 180 Ringing message (indicating that the peer is ringing, waiting for the user to answer) to the S-CSCF and carries its own SDP information (received media type, encoding mode, media transceiver IP address and port, etc.), S - The CSCF is forwarded to the VCC AS.
  • step s304 the VCC AS returns a 180 ringing message to the S-CSCF, and the S-CSCF sends a 180 ringing message to the UE 1.
  • the UE 1 may generate a local ringback tone, and then waits for the peer user to go off-hook.
  • the UE 1 According to some predetermined policies (such as changes in wireless conditions, the call has completed media negotiation, etc.), the decision should be made to switch to the CS network, then start the VCC network switch, and perform the following steps:
  • Step s305 UE 1 initiates a call through the CS network, and sends an initial call message to the MSC.
  • the Call Origination includes a CdPN, where the called number is (Service Code + VCC AS E.164 number) and CgPN (ie, the calling number is the number MDN of the UE 1;
  • Step s306 the MSC generates an ISUP IAM (Initial Address Message) and sends it to the MGCF, using the E.164 number of the VCC AS as the called number, and the number of the UE 1 is used as the calling number, and the MSC/VLR discards the Service Code value;
  • ISUP IAM Initial Address Message
  • Step s307 the MSC/VLR sends a Traffic Channel Assignment message to the UE, that is, the CS network allocates a traffic channel for UE1, and UE1 captures a traffic channel;
  • Step s308 the MGCF sends an ACM message to the MSC/VLR;
  • the MGCF In step s309, the MGCF generates an INVITE message (including the bearer information allocated for the call - the IP address and the bandwidth, that is, the MGW SDP), and the Request URI in the INVITE message is the E.164 number of the VCC AS, and the message is I. -CSCF is sent to the VCC AS;
  • Step s310 ⁇ step s311 the VCC AS checks the call state to the non-call state, generates an UPDATE message according to the received INVITE message (where the Request URI is modified to the address of the peer end), and sends the response to the peer end, and the peer responds with a 200 OK response message.
  • the 200 OK response message includes the SDP information of the peer end response;
  • Step s312 The VCC AS generates a 183 message according to the received 200 OK message (response to step s309), and then sends the 183 message to the UE1, where the 183 message includes the response SDP information, and waits for the peer user.
  • the VCC AS generates a 183 message according to the received 200 OK message (response to step s309), and then sends the 183 message to the UE1, where the 183 message includes the response SDP information, and waits for the peer user.
  • Step s313 the peer detects that the user picks up the phone, sends a 200 OK response message (which is the final response to step s302) to the S-CSCF, and the S-CSCF forwards to the VCC AS;
  • Step s314 the VCC AS sends the 200 OK response message to the MGCF through the I-CSCF.
  • the MGCF converts to ANM/CON (connection or address full message) to the MSC, indicating that the call can be made, - step s315 ⁇ step s316, the MGCF responds with the received 200 OK response message by ACK, and the ACK message is forwarded by the VCC AS to the S- CSCF, finally arrived at the opposite end.
  • ANM/CON connection or address full message
  • Step s317 VCC AS releases SIP call dialog 1 : Released with 4XX/5XX/6XX error response message.
  • the present invention further provides a network switching method for a user equipment, where the method includes the following steps:: a session negotiation in which a user equipment that initiates handover performs a call continuity handover in a call hold state through a target network;
  • FIG. 7 is a flowchart of a first embodiment of the present invention, in which the user equipment is switched from the circuit domain network to the IMS domain, in this embodiment, the call is maintained by the UE1, and the UE 1 is initiated by the UE1.
  • the UE establishes a call and makes a call with the peer through the CS network.
  • the UE 1 decides to switch to the IMS network according to some policies (such as changes in wireless conditions).
  • the specific handover mainly includes the following steps:
  • Step s401 if the UE 1 does not perform IMS registration, registering
  • the UE 1 sends an INVITE invite message to the VCC AS (as a handover request) through the S-CSCF, and starts to establish a SIP call dialog 3, where the Request URI is an identity of the VCC AS (eg, Tel URI, SIPURI), and also includes the UE 1 SDP information, in addition, it should be noted that the SDP information of UE1 indicates that its media stream attribute is only sent;
  • the Request URI is an identity of the VCC AS (eg, Tel URI, SIPURI)
  • SDP information of UE1 indicates that its media stream attribute is only sent;
  • Step s403 to step s404 the VCC AS checks the call state to the call hold state, and generates a Re-INVITE message according to the received INVITE message (for example, the Request URI is changed to the address of the peer end, and the Contact is modified to the identity of the VCC AS), and sent to the pair.
  • the session negotiation is performed (for example, the media type, the coding mode, the IP address and the port of the media, and the port), where the SDP information indicates that the media stream attribute of the UE1 is only sent;
  • step s405 the peer end responds to the 200 OK message, and the 200 OK message includes the SDP information of the peer end response.
  • the SDP information of the peer end indicates that the media stream attribute is only received, and the VCC AS sends a 200 OK message.
  • step s401 is sent to the UE 1;
  • Step s406 as the three-way handshake mechanism of the SIP, the UE 1 returns an ACK message, and finally reaches the opposite end;
  • Step s407 the VCC AS sends a BYE message to the MGCF to release the CS side call;
  • Step s408 ⁇ step s411, the CS side call is released;
  • Step s412 after receiving the RLC message, the MGCF generates a BYE message received by the 200 OK message response step s407;
  • UE 1 talks to the peer through the IMS domain.
  • FIG. 8 is a flowchart of a second embodiment of the present invention, in which the user equipment is switched from the circuit domain network to the IMS domain, in this embodiment, the call is maintained by the peer end and the UE1 is actively initiated. Knowing the state of being held, the UE 1 establishes a call with the opposite end and makes a call through the CS network. The UE 1 decides to switch to the IMS network according to some policies (such as changes in radio conditions), and specifically includes the following steps:
  • Step s501 If UE 1 does not perform IMS registration, perform registration;
  • step s502 the UE 1 sends an INVITE invite message to the VCC AS (as a handover request) through the S-CSCF, and starts to establish a SIP call dialog 3, where the Request URI is the E.164 number of the VCC AS, and also includes the SDP information of the UE 1, and It should be noted that the SDP information of the UE1 indicates that the media stream is received only.
  • Step s503 the VCC AS checks the call state to the call hold state, generates a Re-INVITE message according to the received INVITE message (the Request URI is modified to be the address of the peer end), and sends the session negotiation to the peer end (eg, media type, The encoding mode, the IP address and port of the media, and the like, where the SDP information indicates that the media stream attribute of the UE1 is only received;
  • the VCC AS checks the call state to the call hold state, generates a Re-INVITE message according to the received INVITE message (the Request URI is modified to be the address of the peer end), and sends the session negotiation to the peer end (eg, media type, The encoding mode, the IP address and port of the media, and the like, where the SDP information indicates that the media stream attribute of the UE1 is only received;
  • step s505 the peer responds to the 200 OK message, and the 200 OK message includes the SDP information of the peer end response.
  • the SDP information of the peer end indicates that the media stream attribute is sent only, and the VCC AS will send the 200 OK message. (corresponding to step s501) is sent to the UE 1;
  • Step s506 as a three-way handshake mechanism of the SIP, the UE 1 returns an ACK message, and finally reaches the opposite end;
  • Step s507 the VCC AS sends a BYE message to the MGCF to release the CS side call;
  • Step s508 ⁇ step s511, the CS side call is released;
  • Step s512 After receiving the RLC message, the MGCF generates a 200 OK message response to the BYE message received in step s507;
  • UE 1 talks to the peer through the IMS domain.
  • FIG. 9 is a flowchart of a third embodiment of implementing handover of a user equipment from a circuit domain network to an IMS domain during call hold of the present invention
  • the SIP protocol is implemented, wherein the call is actively initiated by the peer end and UE1 is activated.
  • the UE 1 establishes a call with the opposite end and makes a call through the CS network.
  • the UE 1 decides to switch to the IMS network according to some policies (such as changes in radio conditions), and specifically includes the following steps:
  • Step s601 If UE 1 does not perform IMS registration, registering is performed;
  • step s602 the UE 1 sends an INVITE invite message to the VCC AS (as a handover request) through the S-CSCF, and starts to establish a SIP call dialog 3, where the Request URI is the E.164 number of the VCC AS, and also includes the SDP information of the UE 1, SDP. The information indicates that the media stream attribute is both sent and received.
  • Step s603 the VCC AS checks the call state to the call hold state, generates a Re-INVITE message according to the received INVITE message (the Request URI is modified to the address of the peer end), and sends
  • the SDP information set by the UE1 added by the VCC AS in the INVITE message indicates that the media stream attribute of the UE1 is only received, and the session is negotiated (for example, the media type, the encoding mode, and the IP address and port of the media).
  • Step s605 the peer responds 200 OK, which includes the SDP information of the peer end response, which needs to be described, where the SDP information of the peer end is sent only, and the VCC AS sends a 200 OK message (corresponding to step s501) to the UE l;
  • Step s606 as a three-way handshake mechanism of the SIP, the UE 1 returns an ACK message, and finally reaches the opposite end;
  • Step s607 the VCC AS sends a BYE message to the MGCF to release the CS side call.
  • Step s608 ⁇ step s611, the CS side call is released;
  • Step s612 After receiving the RLC message, the MGCF generates a 200 OK message response step s507 to receive the BYE message;
  • UE 1 can talk to the peer through the IMS domain network.
  • FIG. 10 the figure is a handover of a user equipment from an IMS domain network to a call during call hold of the present invention.
  • a flow chart of the first embodiment of the road domain is implemented in the embodiment according to the SIP protocol, wherein the call hold is initiated by the UE1, and the UE 1 establishes a call with the opposite end through the IMS network and performs a call, and the UE 1 according to some policies (such as a wireless condition) Change) decision to switch to the CS network, the specific switch mainly includes the following steps:
  • Step s701 The UE 1 initiates a call through the CS network, where the called number of the CdPN is (Service Code + VCC AS E.164 number), and the calling number of the CgPN is the number MDN of the UE 1 (Mobile Directory Number, such as 133xxxx5678);
  • Step s702 ⁇ step s705, if necessary, perform the authentication and registration process of the CS network; step s706, the MSC generates an ISUP IAM (Initial Address Message) and sends it to the MGCF, using the VCC AS E.164 number as the called number, and The number of UE 1 is used as the calling number, and the MSC discards the Service Code value;
  • ISUP IAM Initial Address Message
  • Step s707 The MSC/VLR sends a Channel Assignment message to the UE, that is, the CS network allocates a service channel to the UE.
  • Step s708 the MGCF may send the ACM (address full) to the MSC/VLR;
  • the MGCF In step s709, the MGCF generates an INVITE message (including the bearer information allocated for the call, that is, the MGW SDP includes the IP address and the bandwidth, the SDP information indicates that the media stream attribute of the MGW is both sent and received), and the Request URI is VCC AS. E.164 number, the message is sent to the VCC AS via the I-CSCF;
  • Step s710 ⁇ step s711 the VCC AS generates a Re-INVITE message according to the received INVITE message (the request URI is modified to be the address of the peer end), and is sent to the peer end through the S-CSCF, which needs to be explained, where the session description information indicates the MGW.
  • the media stream attribute is only sent;
  • Step s712 to step s713 the peer response 200 OK message arrives at the VCC AS via the S-CSCF, and includes the SDP information of the peer end response.
  • the SDP information of the peer end indicates that the media stream attribute is received only.
  • Step s714 The VCC AS sends the 200 OK message to the MGCF via the I-CSCF.
  • the SDP information of the peer end carried by the 200 OK message indicates that the media stream attribute of the peer end is both sent and received.
  • Step s715 correspondingly, the MGCF generates an ANM (Response message) and sends it to the VMSC; Step s716, as the SIP three-way handshake mechanism, the MGCF generates an ACK message and sends it to the VCC AS; Step s717 ⁇ Step s718.
  • the VCC AS sends the ACK message to the peer end through the S-CSCF; Step s719, in any step after step s709, the UE and The CS network completes the traffic channel capture. Steps s720 ⁇ s721.
  • the VYE AS can send a BYE message to the UE 1 to release the call leg of the IMS side, and the UE 1 returns a 200 OK message response.
  • the UE1 since the MSC does not know the call hold status of the UE1, the UE1 needs to synchronize the call hold status to the MSC, that is, in step s722, the UE1 sends a call hold request to the MSC (for example, sends a Flash with information/Hold Request message to the MSC);
  • the MSC After receiving the call hold request, the MSC sends a call hold request to the MGCF, for example, a Call Progress message (CPG), where the CPG message includes a peer call hold indication; the MGCF converts to generate a Re-INVITE/UPDATE
  • CPG Call Progress message
  • the MGCF converts to generate a Re-INVITE/UPDATE
  • FIG. 11 is a flowchart of a second embodiment of the user equipment switching from the IMS domain network to the circuit domain during the call hold of the present invention
  • the SIP protocol is implemented, wherein the call is actively initiated by the peer, UE 1
  • the UE establishes a call and makes a call with the peer end through the IMS network.
  • the UE 1 decides to switch to the CS network according to some policies (such as changes in wireless conditions).
  • the specific handover mainly includes the following steps:
  • Step s801 The UE 1 initiates a call through the CS network, where the CdPN called number is (Service Code + VCC AS E.164 number), and the CgPN calling number is the UE 1 number MDN (Mobile Directory Number, such as 133xxxx5678);
  • Step s802 ⁇ step s805, if necessary, perform the authentication and registration process of the CS network; step s806, the MSC generates an ISUP IAM (Initial Address Message) and sends it to the MGCF, using the VCC AS E.164 number as the called number, and The number of UE 1 is the calling number, and the MSC discards the Service Code.
  • ISUP IAM Initial Address Message
  • Step 807 The MSC/VLR sends a Channel Assignment message to the UE, that is, the CS network allocates a service channel to the UE.
  • Step s808 the MGCF may send the ACM (address full) to the MSC;
  • step s809 the MGCF generates an INVITE message (including the bearer information allocated for the call, that is, the MGW SDP includes the IP address and the bandwidth, the SDP information indicates that the media stream attribute of the MGW is both sent and received), and the Request URI is VCC AS. E.164 number, the message is sent to the VCC AS via the I-CSCF;
  • Step s810 the VCC AS generates a Re-INVITE message according to the received INVITE message (the request URI is modified to be the address of the peer end), and sends the message to the peer end through the S-CSCF, which needs to be explained, where the MGCF session description information indicates The media stream attribute of the MGW is only received;
  • Step s814 the VCC AS sends the 200 OK message to the MGCF via the I-CSCF.
  • the SDP information of the peer end carried by the 200 OK message indicates that the media stream attribute of the peer end is both sent and received.
  • Step s815 correspondingly, the MGCF generates an ANM (response message) and sends it to the VMSC;
  • Step s816 as the SIP three-way handshake mechanism, the MGCF generates an ACK message and sends it to the VCC AS; Step s817 ⁇ Step s818.
  • the VCC AS sends the ACK message to the peer end through the S-CSCF; Step s819, in any step after step s809, the UE and The CS network completes the traffic channel capture. Steps s820 ⁇ s821.
  • the VYE AS can send a BYE message to the UE 1 to release the call leg of the IMS side, and the UE 1 returns a 200 OK message response.
  • the VCC AS needs to be in this embodiment.
  • VCC AS uses the UPDATE message, there is no need to send an ACK message to the MGCF.
  • the figure is a function of performing VCC switching during ringing of the user equipment of the present invention.
  • the user equipment includes: a call processing unit, and the call processing unit is improved in the present invention, that is, the call processing unit is set in the present invention:
  • the voice call continuity switching processing unit 11 during ringing is used for network switching of voice call continuity during ringing.
  • the voice call continuity switching processing unit 11 during the ringing process in the embodiment specifically includes three logical function units, namely: a voice call continuity switching unit 111 during a ringing period, and a session negotiation processing unit. 112 and off-hook signal transmitting unit 113, wherein
  • the session negotiation processing unit 112 is configured to perform session negotiation of call continuity during ringing with the peer calling user equipment through the target network; that is, the session negotiation mainly used for performing voice call continuity switching during the ringing period,
  • the call is initiated to the VCC application server to perform session negotiation with the peer end, and the specific processing refers to the foregoing process description;
  • the ringing period continuity switching unit 111 is configured to perform network switching of call continuity during ringing after the session negotiation is successful;
  • the off-hook signal sending unit 113 is configured to send an off-hook signal to the opposite-end calling user equipment according to the switching result of the call continuity switching unit 111 during the ringing; that is, to the voice call after the network switching succeeds
  • the application server transmits the off-hook signal;
  • FIG. 13 is a schematic structural diagram of a function of performing VCC handover during a ringback tone of a user equipment according to the present invention, wherein the user equipment includes a call processing unit, and the call processing unit is improved in the present invention, that is, in the present invention.
  • the call continuity switching processing unit 12 during the ring back tone is used for network switching of voice call continuity during the ring back tone.
  • the voice call continuity switching processing unit during the ring back tone in the present invention specifically includes three logical function units, that is, the voice call continuity switching unit 121 and the session negotiation processing unit 122 during the ring back tone period. And receiving unit 123, wherein
  • the voice call continuity switching unit 121 during the ring back tone performs network switching of the call continuity during the ring back tone after the session negotiation is successful.
  • the session negotiation processing unit 122 is configured to perform session negotiation of the call continuity switch during the ring back tone by the target called user equipment through the target network; that is, mainly used to perform the ring back tone period.
  • the session negotiation of the voice call continuity switching is mainly to initiate a call to the VCC application server to perform session negotiation with the peer end. For specific processing, refer to the foregoing process description.
  • the receiving unit 123 is configured to receive an off-hook signal sent by the peer called user equipment after the network handover is completed.
  • the voice call continuity switching processing unit during the ring back tone in the present invention specifically includes two logical function units, that is, a voice call continuity switching initiation unit and session negotiation during the ring back tone.
  • a processing unit wherein the voice call continuity switching initiation unit during the ring back tone is mainly used to initiate a voice call continuity switch during a ring back tone according to a predetermined policy.
  • the session negotiation processing unit is mainly used for the session negotiation of the voice call continuity switching.
  • the call is initiated to the VCC application server to perform session negotiation with the peer end, and the specific processing refers to the foregoing process description.
  • FIG. 14 is a schematic structural diagram of a function of performing VCC handover control during ringing or ringback tone of a call continuity application server according to the present invention
  • the call continuity application server is improved in the present invention, that is, in the present invention
  • the call continuity switching control unit 13 during the ring/ringback tone is used to control network switching for voice call continuity during ringing or ring back tone.
  • the voice call continuity switching control unit during the ring/ringback tone in the present invention mainly includes two functional units, that is, a voice call continuity negotiation control unit and an off-hook between the ring/loopback tone.
  • Signal identification transmission unit wherein
  • the voice/continuous tone negotiation continuity session negotiation control unit 131 is configured to control the session negotiation of the network handover during the ringing/ringback tone call continuity; the specific processing refers to the foregoing process description, and details are not described herein again;
  • the off-hook signal identification transmitting unit 132 is mainly used for identifying the off-hook signal of the called user equipment and transmitting the signal to the calling user equipment to complete the handover.
  • the method may further include:
  • the releasing unit 133 is mainly configured to release the call leg in the network to be switched after the session negotiation succeeds.
  • the user equipment of the present invention implements VCC network switching during call hold.
  • the voice call continuity switching processing unit 14 during call hold is mainly used for network switching of voice call continuity during call hold.
  • the voice call continuity switching processing unit during the call hold period in the present invention specifically includes two logical function units, namely, a session negotiation processing unit 142 and a call hold period voice call continuity switching unit 141. among them
  • the session negotiation processing unit 142 is configured to perform a call negotiation of a call continuity switch in a call hold state between the target network and the opposite end; that is, mainly used to control session negotiation for performing a voice call handover to the target network during the call hold period;
  • the voice call continuity switching unit 141 is configured to switch the user equipment to the target network after successful session negotiation.
  • the voice call continuity switching processing unit during the call hold period in the present invention specifically includes two logical functional units, that is, a voice call continuity switching initiation unit and a session negotiation processing unit during call hold.
  • the voice call continuity session negotiation processing unit is mainly used to identify and initiate voice call continuity during call hold according to a predetermined policy; the session negotiation processing unit is mainly used to control switching to a target during call hold Session negotiation for the network.
  • the VCC application server is improved in the present invention, that is, the VCC application server in the present invention is provided with:
  • the voice call continuity switching control unit 15 during call hold is mainly used to control network switching for voice call continuity during call hold.
  • the voice call continuity switching control unit during the call hold period in the present invention mainly includes two functional units, namely, a session negotiation processing unit 152 and a call hold period call continuity switching control unit 151, wherein
  • the session negotiation processing unit 152 is configured to control session negotiation for switching to the target network during call hold; that is, mainly used for controlling session negotiation for performing voice call handover to the target network during the call hold period;
  • the call continuity switching control unit 151 during call hold is used to control network switching of call continuity during call hold after successful session negotiation.
  • the VCC application server may further include:
  • the releasing unit 154 is mainly configured to release the call leg in the network to be switched after the session negotiation succeeds.
  • the voice call continuity switching control unit during the call hold period in the present invention mainly includes two functional units, that is, a voice call continuity switching initiation unit and a session negotiation processing unit during call hold, according to the logical function division of the implementation thereof.
  • the voice call continuity switching initiation unit 151 is configured to identify and initiate voice call continuity during call hold according to a predetermined policy.
  • the session negotiation processing unit 152 is mainly used to control handover during call hold. Negotiate to the target network.
  • the VCC application server may further include:
  • the release unit 153 is mainly used to release the call leg in the network to be switched after the negotiation succeeds.
  • the above description is only a preferred embodiment of the present invention, and it should be noted that those skilled in the art can also make several improvements and retouchings without departing from the principles of the present invention. It is considered as the scope of protection of the present invention.

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Description

用户设备、 呼叫连续性应用服务器及网络切换方法
本申请要求于 2006年 9月 8、 2006年 10月 12日提交中国专利局、 申请 号为 200610122010.8、 200610139133.2、 发明名称分别为"用户设备、 语音呼 叫连续性应用服务器及网络切换方法 "的中国专利申请的优先权， 其全部内容 通过引用结合在本申请中。
技术领域
本发明涉及 IP多媒体子系统（ IP Multimedia Subsystem, IMS )技术领域， 更具体的说 , 本发明涉及一种振铃 /回铃音期间或呼叫保持期间处理语音呼叫 连续性（Voice Call Continuity, VCC )的网络切换的方法及相关的 VCC应用服 务器（ VCC Application Server, VCC AS ) 以及用户设备。
背景技术
IMS是第三代伙伴计划（ The 3rd Generation Partnership Project, 3 GPP )在 版本 5 ( Release 5, R5 ) 中提出的支持 IP多媒体业务的子系统， 它的核心特点 是采用会话初始协议 ( Session Initial Protocol, SIP )协议以及与接入的无关性， IMS为未来的多媒体应用提供一个通用的业务平台， 它是向全 IP网业务提供体 系迈进的重要一步。
IMS标准随着 3GPP标准体系衍生出 R5、 R6两个版本 ,继 2005年 3月 R6版本 基本稳定之后， 3GPP又推出了 R7版本， 在增强原有系统功能的同时， 补充了 一些全新的功能。
由于 IMS语音业务与电路域语音业务在一段时间内会同时存在，语音业务 之间的切换是急需解决的问题， VCC功能即是 R7版本提出解决所述问题的一 种新的功能。
按照上述标准定义， VCC指语音呼叫连续性， 即用户在各种接入技术之间 移动时，其通话保持连续性，举例来说，用户从传统 2G网络（例如全球通 GSM、 码分多址 CDMA )移动到 3G网络（例如通用移动电信系统 UMTS、 高速率分组 数据 HRPD ) 时， 其通话继续， 又可以发起其它 3G业务， 例如高速上网等。
参考图 1 , 为了支持 VCC, —个用户通过电路域（Circuit Switched, CS ) 网络（即 2G网络 )发起 /接收呼叫时 , 信令都要进入 IMS网络， 所述 IMS网络其 中一个网元 VCC AS可用于控制 VCC切换， 另外， 用户通过 IMS网络进入 CS网 络的网元是媒体网关控制功能单元（ MGW Control Function, MGCF ) , MGCF 也控制媒体网关（ Media Gateway, MGW )进行 载的路由。
目前的 VCC切换假设在通话状态下进行，没有考虑振铃 /回铃音（主 /被叫 情形）期间以及呼叫保持期间进行 VCC切换， 这可能产生一些问题， 例如用 户设备离开 IMS热点覆盖且用户设备处于振铃 /回铃音或呼叫保持期间 , 由于 用户设备不能进行 VCC网络切换， 则该用户设备可能产生掉话， 影响用户通 话质量。
发明内容
本发明实施例解决的技术问题是提供一种用户设备的网络切换方法、呼叫 连续性应用服务器以及用户设备 , 以支持用户设备在振铃 /回铃音期间或呼叫 保持期间进行呼叫连续性的网络切换，避免用户设备掉话，提高用户通话质量。
为解决上述问题，本发明实施例提供一种用户设备的网络切换方法，该方 法主要包括：发起切换的用户设备通过目标网络与对端进行呼叫保持状态下呼 叫连续性切换的会话协商； 在会话协商成功后将呼叫切换到目标网络。
本发明实施例还提供一种用户设备的网络切换方法， 所述方法包括：接收 来自发起切换的用户设备的网络切换请求，所述网络切换请求携带所述用户设 备会话描述信息； 若检测到所述呼叫为呼叫保持状态， 则对所述会话描述信息 更新，并向对端发送携带所述发起切换的用户设备的呼叫保持状态的会话描述 信息；
在所述用户设备与对端通过所述会话描述信息会话协商成功后，根据所述 网络切换请求将对应的呼叫切换到目标网络。
相应的， 本发明实施例提供一种用户设备， 所述用户设备包括： 会话协商 处理单元，用于通过目标网络与对端进行呼叫保持状态下的呼叫连续性切换的 会话协商； 呼叫保持期间呼叫连续性切换单元， 用于在会话协商成功后将呼叫 切换到目标网络。
本发明实施例还提供一种呼叫连续性应用服务器， 包括： 呼叫保持期间呼 叫连续性会话协商处理单元，用于控制进行呼叫保持期间切换到目标网络的会 话协商； 呼叫保持期间呼叫连续性切换控制单元， 用于在会话协商成功后控制 呼叫保持期间的呼叫连续性的网络切换。
本发明实施例再提供一种用户设备的网络切换方法， 所述方法包括： 被叫用户设备通过目标网络与对端主叫用户设备进行振铃期间呼叫连续 性切换的会话协商； 在所述会话协商成功后呼叫被切换到目标网络，被叫用户 设备向对端主叫用户设备发送摘机信号。
相应的，本发明实施例提供一种用户设备的网络切换方法，其中待切换网 络为 IP多媒体子系统网络， 目标网络为电路域网络， 其特征在于， 所述方法 包括：主叫用户设备通过目标网络与对端被叫用户设备进行回铃音期间呼叫连 续性切换的会话协商； 在所述会话协商成功后呼叫被切换到目标网络； 主叫用 户设备接收对端发送的摘机信号。
本发明实施例还提供一种用户设备，所述用户设备包括： 会话协商处理单 元，用于通过目标网络与对端主叫用户设备进行振铃期间呼叫连续性的会话协 商； 振铃期间呼叫连续性切换单元， 用于在所述会话协商成功后进行振铃期间 呼叫连续性的网络切换； 摘机信号发送单元， 用于根据所述呼叫连续性切换单 元的切换结果向对端主叫用户设备发送摘机信号。
本发明实施例再提供一种用户设备， 所述用户设备包括：
会话协商处理单元 ,用于通过目标网络与对端被叫用户设备进行回铃音期 间呼叫连续性切换的会话协商； 回铃音期间呼叫连续性切换单元， 用于在所述 会话协商成功后进行回铃音期间呼叫连续性的网络切换；接收单元， 用于接收 对端被叫用户设备在所述网络切换完成后发送的摘机信号。
相应的， 本发明实施例还提供一种呼叫连续性应用服务器， 包括： 振 /回 铃音期间呼叫连续性会话协商控制单元， 用于控制进行振铃 /回铃音期间呼叫 连续性的网络切换的会话协商； 摘机信号识别传送单元， 用于识别被叫用户设 备的摘机信号并传送给主叫用户设备。
与现有技术相比， 本发明一个或多个实施例具有以下有益效果： 本发明实施例中振铃或回铃音期间一方通过呼叫连续性应用服务器与对 端另一方进行会话协商，在协商通过后，还通过所述呼叫连续性应用服务器向 对端发送摘机信号， 实现了在振铃或回铃音期间的呼叫连续性的网络切换， 另 夕卜，在呼叫保持期间，发起切换的用户设备在目标网络通过呼叫连续性应用服 务器与对端进行呼叫保持状态下的会话协商，在会话协商完成后切换到目标网 络， 实现了在呼叫保持期间的呼叫连续性的切换，从而在振铃或回铃音期间以 及呼叫保持期间， 即使用户设备所处网络覆盖变差时，该用户设备也可通过呼 叫连续性应用服务器切换到另一网络，可有效避免用户设备在振铃或回铃音期 间以及呼叫保持期间发生掉话， 提高用户通话质量。
附图说明
图 1是现有技术 VCC结构参考模型；
图 2是本发明振铃期间被叫用户设备进行网络切换的方法实施例的主要 流程图；
图 3 是本发明回铃音期间主叫用户设备进行网络切换的方法实施例的主 要流程图；
图 4是本发明振铃期间被叫用户设备从电路域切换到 IMS域的实施例流 程图；
图 5是本发明振铃期间被叫用户设备从 IMS域切换到电路域的方法实施 例流程图；
图 6是本发明回铃音期间主叫用户设备从 IMS域切换到电路域的方法实 施例流程图；
图 7是本发明呼叫保持期间用户设备从电路域网络切换到 IMS域的第一 实施例流程图；
图 8是本发明呼叫保持期间用户设备从电路域网络切换到 IMS域的第二 实施例流程图；
图 9是本发明呼叫保持期间用户设备从电路域网络切换到 IMS域的第三 实施例流程图；
图 10是本发明呼叫保持期间用户设备从 IMS域网络切换到电路域的第一 实施例流程图；
图 11是本发明呼叫保持期间用户设备从 IMS域网络切换到电路域的第二 实施例流程图；
图 12是本发明实施例中用户设备实现振铃期间进行 VCC网络切换的结构 示意图；
图 13是本发明实施例中用户设备实现回铃音期间进行 VCC网络切换的结 构示意图；
图 14是本发明实施例中 VCC应用服务器振铃或回铃音期间进行 VCC网 络切换控制的结构示意图；
图 15是本发明实施例中用户设备实现呼叫保持期间进行 VCC网络切换的 结构示意图；
图 16是本发明实施例中 VCC应用服务器呼叫保持期间进行 VCC网络切 换控制的结构示意图。
具体实施方式
本发明具体实施例方式中，在振铃或回铃音期间一方通过呼叫连续性应用 服务器（所述呼叫连续性应用服务器包括语音、 视频和 /或音视频呼叫连续性 应用服务器， 本发明所述的呼叫连续性应用服务器以 VCC应用服务器为例， 下同）与对端另一方进行会话协商， 在协商通过后， 所述 VCC应用服务器向 对端发送摘机信号， 实现了在振铃或回铃音期间的 VCC网络切换， 另外， 在 呼叫保持期间，发起切换的用户设备在目标网络通过语音呼叫连续性应用服务 器与对端进行呼叫保持状态下的会话协商， 在会话协商完成后切换到目标网 络。 从而解决了振铃 /回铃音期间或呼叫保持期间网络覆盖不好， 而用户设备 不能进行 VCC切换导致用户通话质量不好的问题， 下面分别按照振铃期间的 网络切换流程、回铃音期间的网络切换流程以及呼叫保持期间的网络切换流程 分别进行说明。
本发明的一实施例中，提供一种用户设备的网络切换方法，所述方法包括： 发起切换的用户设备通过目标网络与对端进行呼叫保持状态下呼叫连续性切 换的会话协商； 在会话协商成功后将呼叫切换到目标网络。 该实施例中， 所述 用户设备通过目标网络中的呼叫连续性应用服务器与对端进行呼叫保持状态 下的会话协商。 本实施例中， 所述方法还可包括： 所述用户设备向所述呼叫连 续性应用服务器发送携带该用户设备的会话描述信息的网络切换请求；所述会 话协商进一步指：根据所述会话描述信息进行会话协商； 所述在会话协商成功 后将呼叫切换到目标网络进一步指： 根据所述会话描述信息会话协商成功后， 将对应所述网络切换请求的呼叫切换到目标网络。
本实施例中，所述会话描述信息中携带该发起切换的用户设备的呼叫状态 保持信息。
当发起切换的用户设备待切换的网络为电路域网络， 目标网络为 IP多媒 体子系统网络时，所述发起切换的用户设备通过目标网络中的呼叫连续性应用 服务器与对端进行呼叫保持状态下的会话协商的具体包括：发起切换的用户设 备通过 IP多媒体子系统网络向呼叫连续性应用服务器发送携带该用户设备的 会话描述信息的网络切换请求。
当待切换网络为 IP多媒体子系统网络， 目标网络为电路域网络时： 所述 发起切换的用户设备在目标网络通过呼叫连续性应用服务器与对端进行呼叫 保持状态的会话协商的具体包括：发起切换的用户设备通过电路域网络、移动 交换中心、媒体网关控制功能单元向呼叫连续性应用服务器发送邀请消息， 所 述邀请消息包括： 该媒体网关控制功能单元的会话描述信息。
本发明的另一实施例中， 一种用户设备的网络切换方法， 包括： 接收来自 发起切换的用户设备的网络切换请求，所述网络切换请求携带所述用户设备会 话描述信息；若检测到所述呼叫为呼叫保持状态，则对所述会话描述信息更新， 并向对端发送携带所述发起切换的用户设备的呼叫保持状态的会话描述信息； 进一步， 在所述用户设备与对端通过所述会话描述信息会话协商成功后， 根据所述网络切换请求将对应的呼叫切换到目标网络。
进一步， 所述方法还可包括：接收来自对端反馈的携带该对端的呼叫保持 状态的会话描述信息；向所述发起切换的用户设备发送携带对端的呼叫保持状 态的会话描述信息。
进一步，所述检测到的呼叫保持状态为所述发起切换的用户设备主动发起 的呼叫保持；所述发起切换的用户设备的呼叫保持的会话描述信息为只发送的 会话描述信息；所述对端的呼叫保持状态的会话描述信息为只接收的会话描述 信息。
进一步，所述呼叫连续性应用服务器检测到的呼叫保持状态为由对端发起 的呼叫保持，所述发起切换的用户设备的呼叫保持的会话描述信息为只接收的 会话描述信息； 所述对端的呼叫保持的会话描述信息为只发送的会话描述信 '&。
所述只接收的会话描述信息由所述待切换的用户设备或呼叫连续性应用 服务器添加。
该实施例中， 还包括： 释放待切换网络的呼叫支路。
当待切换网络为 IP多媒体子系统网络， 目标网络为电路域网络时： 从媒 体网关控制功能单元接收来自发起切换的用户设备通过电路域、移动交换中心 发送的邀请消息， 所述邀请消息包括： 该媒体网关控制功能单元的会话描述信 息； 对所述会话描述信息更新， 并向对端发送携带所述发起切换的用户设备的 呼叫保持状态的会话描述信息进一步指：根据所述会话邀请消息生成所述会话 描述信息的更新消息，并向对端发送携带媒体网关控制功能单元的呼叫保持状 态的会话描述信息的邀请或更新消息。
当所述检测到的呼叫保持状态为由所述发起切换的用户设备主动发起的 呼叫保持时：所述媒体网关控制功能单元的呼叫保持状态的会话描述信息为只 发送的会话描述信息；所述对端的呼叫保持状态的会话描述信息为只接收的会 话描述信息； 会话协商成功后还包括： 所述发起切换的用户设备与所述移动交 换中心同步呼叫保持状态。
进一步， 所述检测到的呼叫保持状态为由对端发起的呼叫保持； 所述媒体 网关控制功能单元的呼叫保持状态的会话描述信息为只接收的会话描述信息； 所述对端的呼叫保持状态的会话描述信息为只发送的会话描述信息；会话协商 成功后还包括： 所述呼叫连续性应用服务器与电路域网络同步呼叫保持状态。
本发明的一实施例中， 一种用户设备的网络切换方法， 包括： 被叫用户设 备通过目标网络与对端主叫用户设备进行振铃期间呼叫连续性切换的会话协 商； 在所述会话协商成功后呼叫被切换到目标网络，被叫用户设备向对端主叫 用户设备发送摘机信号。所述被叫用户设备通过目标网络的呼叫连续性应用服 务器与对端主叫用户设备进行振铃期间呼叫连续性切换的会话协商。所述在会 话协商成功后呼叫被切换到目标网络，被叫用户设备向对端主叫用户设备发送 摘机信号的过程具体包括： 在切换到目标网络后，被叫用户设备将摘机信号发 送给所述呼叫连续性应用服务器；所述呼叫连续性应用服务器检测到所述被叫 用户设备摘机， 将摘机信号发送给对端主叫用户设备。
所述被叫用户设备将摘机信号发送给所述呼叫连续性应用服务器的过程 为：
通过所述被叫用户设备所在的待切换网络将摘机信号的信令发送给所述 呼叫连续性应用服务器。
所述被叫用户设备将摘机信号发送给所述呼叫连续性应用服务器的过程 为：通过会话发起协议的通知消息或非结构化补充业务数据服务或短消息服务 将摘机信号发送给所述呼叫连续性应用服务器。
所述呼叫连续性应用服务器检测到被叫用户设备摘机之后还包括：所述呼 叫连续性应用服务器释放待切换网络的呼叫支路。
当待切换的网络为电路域网络， 目标网络为 IP多媒体子系统网络时： 所述被叫用户设备在目标网络通过呼叫连续性应用服务器与对端主叫用 户设备进行振铃期间呼叫连续性切换的会话协商的具体过程包括：被叫用户设 备通过 IP多媒体子系统网络向呼叫连续性应用服务器发送携带该被叫用户设 备的会话描述信息的会话邀请消息；所述呼叫连续性应用服务器检测到所述呼 叫为非通话状态，根据所述会话邀请消息生成更新消息， 并将所述更新消息发 送给对端主叫用户设备；对端主叫用户设备向所述呼叫连续性应用服务器反馈 携带该对端主叫用户设备的会话描述信息的更新响应消息；
所述呼叫连续性应用服务器根据所述更新响应消息生成临时响应消息，并 向被叫用户设备发送携带对端主叫用户设备的 ^舌描述信息的临时响应消息。
当待切换网络为 IP多媒体子系统网络， 目标网络为电路域网络时： 所述被叫用户设备通过呼叫连续性应用服务器与对端主叫用户设备进行 振铃期间呼叫连续性切换的会话协商的具体过程包括：被叫用户设备通过电路 域网络向移动交换中心发起始呼消息；移动交换中心才 据所述始呼消息生成初 始地址消息并发送到媒体网关控制功能单元；媒体网关控制功能单元解析初始 地址消息，获得待被叫用户设备和呼叫连续性应用服务器的号码，反馈地址全 信息给移动交换中心，并向呼叫连续性应用服务器发送携带媒体网关控制功能 单元会话描述信息的会话邀请消息 ,所述^舌邀请消息包括被叫用户设备和呼 叫连续性应用服务器的号码；所述呼叫连续性应用服务器检测到所述呼叫为非 通话状态，根据所述会话邀请消息生成所述会话描述信息的更新消息， 并将所 述更新消息发送给对端主叫用户设备；对端主叫用户设备向所述呼叫连续性应 用服务器反馈携带该对端主叫用户设备的会话描述信息的更新响应消息；所述 呼叫连续性应用服务器根据所述更新响应消息生成对端主叫用户设备的会话 描述信息的临时响应消息 ,并将所述临时响应消息发送给所述媒体网关控制功 能单元。
在本发明的另一实施例中，提供一种用户设备的网络切换方法，其中待切 换网络为 IP多媒体子系统网络， 目标网络为电路域网络， 所述方法包括： 主 叫用户设备通过目标网络与对端被叫用户设备进行回铃音期间呼叫连续性切 换的会话协商； 在所述会话协商成功后呼叫被切换到目标网络； 主叫用户设备 接收对端发送的摘机信号。
所述主叫用户设备通过目标网络的呼叫连续性应用服务器与对端被叫用 户设备进行回铃音期间呼叫连续性切换的会话协商。
所述在所述会话协商成功后将呼叫切换到目标网络后，还包括： 对端被叫 用户设备在所述会话协商成功后向呼叫连续性应用服务器发送摘机信号； 所述呼叫连续性应用服务器将接收到的所述摘机信号发送给主叫用户设 备。
所述的主叫用户设备与对端被叫用户设备进行回铃音期间呼叫连续性切 换的会话协商的具体过程包括：主叫用户设备通过电路域网络向移动交换中心 发起始呼消息；所述移动交换中心根据所述始呼消息生成初始地址信息并发送 到所述媒体网关控制功能单元； 所述媒体网关控制功能单元解析初始地址消 息，获得待主叫用户设备和呼叫连续性应用服务器的号码，反馈地址全信息所 述移动交换中心，并向呼叫连续性应用服务器发送携带所述媒体网关控制功能 单元会话描述信息的会话邀请消息 ,所述会话邀请消息包括主叫用户设备和呼 叫连续性应用服务器的号码；所述呼叫连续性应用服务器检测到所述呼叫为非 通话状态，根据所述会话邀请消息生成所述会话描述信息的更新消息， 并将所 述更新消息发送给对端被叫用户设备；对端被叫用户设备向所述呼叫连续性应 用服务器反馈携带该对端被叫用户设备的会话描述信息的更新响应消息；所述 呼叫连续性应用服务器根据所述更新响应消息生成对端主叫用户设备的会话 描述信息的临时响应消息 ,并将所述临时响应消息发送给所述媒体网关控制功 能单元。
在本发明的再一实施例中， 一种用户设备的网络切换方法， 包括： 发起切 换的用户设备通过目标网络与对端进行呼叫保持状态下呼叫连续性切换的会 话协商； 在会话协商成功后将呼叫切换到目标网络。
所述用户设备通过目标网络中的呼叫连续性应用服务器与对端进行呼叫 保持状态下的会话协商。
所述会话协商完成后还包括：所述呼叫连续性应用服务器释放待切换网络 的呼叫支路。
所述待切换网络为电路域网络， 目标网络为 IP多媒体子系统网络； 所述 发起切换的用户设备通过目标网络中的呼叫连续性应用服务器与对端进行呼 叫保持状态下的会话协商的具体过程包括： 发起切换的用户设备通过 IP多媒 体子系统网络向呼叫连续性应用服务器发送携带该用户设备的会话描述信息 的网络切换请求；若所述呼叫连续性应用服务器检测到所述呼叫为呼叫保持状 态， 则根据所述则邀请消息生成更新消息； 并向对端发送携带所述发起切换的 用户设备的呼叫保持状态的会话描述信息；对端反馈所述呼叫连续性应用服务 器携带该对端的呼叫保持状态的会话描述信息；所述呼叫连续性应用服务器向 所述发起切换的用户设备发送携带对端的呼叫保持状态的会话描述信息。
所述呼叫连续性应用服务器检测到的呼叫保持状态为所述发起切换的用 户设备主动发起的呼叫保持；所述发起切换的用户设备的呼叫保持的会话描述 信息为只发送的会话描述信息；所述对端的呼叫保持状态的会话描述信息为只 接收的会话描述信息。 所述呼叫连续性应用服务器检测到的呼叫保持状态为由对端发起的呼叫 保持；所述发起切换的用户设备的呼叫保持的会话描述信息为只接收的会话描 述信息； 所述对端的呼叫保持的会话描述信息为只发送的会话描述信息。
所述只接收的会话描述信息由所述待切换的用户设备或所述呼叫连续性 应用服务器添加。
待切换网络为 IP多媒体子系统网络， 目标网络为电路域网络时： 所述发 起切换的用户设备在目标网络通过呼叫连续性应用服务器与对端进行呼叫保 持状态的会话协商的具体过程包括：发起切换的用户设备通过电路域网络向移 动交换中心发起始呼消息；所述移动交换中心根据所述始呼消息生成初始地址 信息并将发送到媒体网关控制功能单元；所述媒体网关控制功能单元向呼叫连 续性应用服务器发送邀请消息， 所述邀请消息包括： 该媒体网关控制功能单元 的会话描述信息； 所述呼叫连续性应用服务器检测到所述呼叫为呼叫保持状 态，根据所述会话邀请消息生成所述会话描述信息的更新消息， 并向对端发送 携带媒体网关控制功能单元的呼叫保持状态的会话描述信息的邀请或更新消 息；对端向所述呼叫连续性应用服务器反馈携带该对端的呼叫保持状态的会话 描述信息；所述呼叫连续性应用服务器向所述媒体网关控制功能单元发送携带 对端的会话描述信息。
所述呼叫连续性应用服务器检测到的呼叫保持状态为由所述发起切换的 用户设备主动发起的呼叫保持；所述媒体网关控制功能单元的呼叫保持状态的 会话描述信息为只发送的会话描述信息；所述对端的呼叫保持状态的会话描述 信息为只接收的会话描述信息；
会话协商成功后还包括：所述发起切换的用户设备与所述移动交换中心同 步呼叫保持状态。
所述呼叫连续性应用服务器检测到的呼叫保持状态为由对端发起的呼叫 保持；所述媒体网关控制功能单元的呼叫保持状态的会话描述信息为只接收的 会话描述信息；所述对端的呼叫保持状态的会话描述信息为只发送的会话描述 信息；
^舌协商成功后还包括：所述呼叫连续性应用服务器与电路域网络同步呼 叫保持状态。
参考图 2， 该图是本发明实振铃期间被叫用户设备进行网络切换的一方法 实施例的主要流程图， 具体的， 主要包括以下步骤：
步骤 s 11， 被叫用户设备通过目标网络与对端主叫用户设备进行振铃期间 呼叫连续性切换的会话协商；
优选的 ,被叫用户设备在目标网络通过语音呼叫连续性应用服务器与对端 主叫用户设备进行振铃期间呼叫连续性切换的会话协商，在具体实现时，根据 切换的目标网络不同 , 会话协商具体处理流程也不同；
步骤 sl2， 在所述会话协商成功后呼叫被切换到目标网络， 被叫用户设备 向对端主叫用户设备发送摘机信号；
也就是说，被叫用户设备在所述会话协商成功后切换到目标网络，通过语 音呼叫连续性应用服务器向对端主叫用户设备发送摘机信号 ,即被叫用户设备 将摘机信号发送给语音呼叫连续性应用服务器；语音呼叫连续性应用服务器检 测到被叫用户设备摘机， 则将摘机信号发送给对端主叫用户设备。
需要说明的，本发明中所述语音呼叫连续性应用服务器检测到被叫用户设 备摘机之后还可由所述语音呼叫连续性应用服务器释放待切换的网络呼叫支 路， 具体实现时， 也可采用其他方式拆除待切换的网络呼叫支路， 这里不再赞 述。
参考图 3， 该图是本发明回铃音期间主叫用户设备进行网络切换的方法实 施例的主要流程图， 其中待切换的网络以 IP多媒体子系统网络为例， 目标网 络以电路域网络为例，具体的， 回铃音期间主叫用户设备进行网络切换主要包 括如下步骤：
步骤 s21， 主叫用户设备通过目标网络与对端被叫用户设备进行回铃音期 间语音呼叫连续性切换的会话协商；
优选的，主叫用户设备在电路域网络通过语音呼叫连续性应用服务器与对 端被叫用户设备进行回铃音期间语音呼叫连续性切换的会话协商；
步骤 s22, 在所述会话协商成功后呼叫被切换到目标网络； 主叫用户设备 接收对端发送的摘机信号； 也就是说，对端被叫用户设备发送摘机信号，语音呼叫连续性应用服务器 检测到所述摘机信号后发送给主叫用户设备完成切换。
下面结合具体的消息处理流程对本发明进行网络切换的方法进行详细说 明。
参考图 4，该图是本发明振铃期间被叫用户设备从电路域切换到 IMS域的 实施例流程图， 本实施例中的消息是基于 SIP协议， 具体处理流程如下： 步骤 slOl , 对端 （ OTHER END POINT, OEP ) (包括主叫终端和相关网络 ) 给 UE1归属的 IMS网络的查询呼叫会话控制功能(（ Interrogating - Call Session Control Function, I-CSCF )发送会话邀请 INVITE消息， I-CSCF选择一个服 务呼叫会话控制功能（ ( Service - Call Session Control Function, S-CSCF )并向 其转发该 INVITE消息。 其中所述 INVITE消息包括被叫 UE 1的统一资源标 识符 (Uniform Resource Identifier, URI)信息, 例如 Bob@huawei.com, 以及对 端的会话描述协议 ( Session Description Protocol, SDP )信息 （包括希望建立 的媒体类型， 编码方式， 媒体的收发 IP地址和端口等）。
基于过滤策略， S-CSCF发送 INVITE消息给 VCC AS, VCC AS作为一个
SIP背对背用户代理（ back-to-back user agent , B2BUA )建立与 OEP之间的 SIP call dialog 1。
步骤 sl02 ~步骤 sl05, VCC AS决策到呼叫应该通过 CS网络下发， 则通 过移动应用协议向归属位置寄存器 HLR获取临时本地路由号码（Temporary Local Directory Number , TLDN ),本实施例中可以采用移动应用协议 ( Mobile Application Protocol , MAP )消息， 也可以采用其它相应的协议， 其中， 所述 MAP消息包括： 定位请求消息（ Localizer request , LOCREQ )和路由请求消 息 ( Router request, ROUTREQ )。
步骤 si 06， VCC AS修改之前收到的 INVITE消息， 将请求 URI修改为一 个可以路由到 CS域的 E.164号码（如 CDMA网络的临时本地电话号码 TLDN, GSM/UMTS网络的移动终端路由号码 MSRN )，并通过 S-CSCF发送到 MGCF, 通过 INVITE消息开始了 VCC AS和 MGCF之间 SIP call dialog 2的建立。 步骤 sl07 ~步骤 si 08， MGCF转换 INVITE消息为初始地址消息 IAM, 并将 IAM消息发送给 MSC/VLR, 其中 , 所述 IAM消息包括分配的 MGW和 MSC之间的承载信息 TRK ( trunk )， MSC/VLR返回地址全消息 ACM。
步骤 sl09， UE 1与 MSC/VLR完成呼叫建立， 本地产生振铃音。
步骤 sll0 ~ slll , MGCF通过 S-CSCF将 180振铃消息发送给 VCC AS, 其中所述 180振铃消息包括分配的 MGW承载（用来连接对端）信息 MGW SDP, VCC AS将所述 180振铃消息转发给对端，也就是说 VCC AS通过 S-CSCF 将所述 180振铃消息转发给 I-CSCF,所述 I-CSCF再将 180振铃消息转发给对 端。
上述步骤 slOl-步骤 sil l为来电通过电路域下发的流程， 本发明也可以在 来电下发后， UE根据一些预定策略（如无线条件变化， CS网络下发的呼叫已 经建立完毕，正在等待用户应答等）决策到应切换到 IMS网络时，则启动 VCC 网络切换， 并执行下面的步骤：
步骤 sll2， UE 1通过 S-CSCF发送 INVITE消息到 VCC AS, 其中所述 INVITE消息中包括 Request URI(所述 Request URI为 VCC AS的 E.164号码） , 也可以包括 UE 1的 SDP信息；
步骤 si 13 ~步骤 si 14. VCC AS检查呼叫状态为非通话状态， 根据收到的 INVITE消息生成更新 UPDATE消息 （该消息中将 Request URI修改为对端的 地址）， 发送到对端， 对端应答 200 OK响应消息， 所述 200 OK响应消息包括 对端应答的 SDP信息；
步骤 sll5, VCC AS 收到的 200 OK响应消息生成临时响应消息（本 实施例中为 183消息）， 所述临时响应消息（即 183消息） 中包括对端应答的 SDP信息， VCC AS将所述临时响应消息发送到 UE1 , 然后 VCC AS等待用户 摘机，执行步骤 si 16，或者 UE决策到 CS网络信号不可用时，则执行步骤 si 19; 步骤 sll6 ~步骤 sll8， UE1检测到用户摘机，通过 CS网络建立的呼叫向
MSC/VLR发送摘机信号（如通过 CONNECT信令消息发送）， MSC/VLR检测 到摘机信号时向 MGCF发送 ANM信息， MGCF将所述 ANM信息转换成 200 OK响应消息 （针对步骤 sl06的最终响应）发送到 VCC AS; 步骤 sll9， UE1通过 IP接入网络向 VCC AS发送 MESSAGE消息， 所述 MESSAGE消息正文指示用户摘机， VCC AS正确解析后， 以 200 OK响应消 息应答 MESSAGE消息；
步骤 sl20 (包括步骤 sl20a-sl20e )， VCC AS释放 CS网络侧的呼叫支路 (包括 SIP call dialog 2 );
需要说明的，本实施例中针对步骤 sll6 ~步骤 sll8， VCCAS可向 MGCF 发送 BYE消息，针对步骤 si 19则应改为发送 CANCEL消息， MGCF响应 200 OK消息； MGCF发送 RLS (释放消息 )给 MSC以释放 MSC侧的呼叫， MSC 发送 RLS到 UE 1以释放无线连接， UE返回释放完成无线链路控制 RLC消息， MSC向 MGCF返回释放完成 RLC消息。
步骤 sl21 ~步骤 sl22, VCC AS生成 200 OK消息（摘机信号， 针对步骤 slOl的最终响应 )发送到对端， 对端以 ACK消息应答；
步骤 sl23 ~步骤 sl24， VCC AS生成 200 OK消息（步骤 sll2的最终响应 ) 发送到 UE 1 , UE 1以 ACK消息应答。
参考图 5，该图是本发明振铃期间被叫用户设备从 IMS域切换到电路域的 实施例流程图， 本实施例中的消息基于 SIP协议， 具体流程如下：
步骤 s 201 , 对端 （包括主叫终端和相关网络）发送会话邀请 INVITE 消 息， 给 UE1归属的 IMS网络的 I-CSCF, I-CSCF选择一个 S-CSCF并向其转 发该 INVITE消息。 其中所述的 NVITE 消息包括被叫 UE 1的 URI信息， 如 Bob@huawei.com, 另外， 还包括对端的 SDP信息： 希望建立的媒体类型、 编 码方式、 媒体的收发 IP地址和端口等）给 UE1归属的 IMS网络的 I-CSCF, I-CSCF选择一个 S-CSCF并向其转发该 INVITE消息。
基于过滤策略， S-CSCF发送 INVITE消息给 VCC AS, VCC AS作为一个 SIP B2BUA建立与 OEP之间的 SIP call dialog 1。
步骤 s202， VCC AS决策到呼叫需要通过 IMS域下发， 则返回 INVITE消 息给 S-CSCF, S-CSCF发送 INVITE消息给 UE 1， UE 1和 VCC AS之间的 SIP call dialog 2开始建立。 步骤 s203 , UE 1返回 180 Ringing消息给 S-CSCF并携带自身 SDP信息 (接受的媒体类型、 编码方式、 媒体的收发 IP地址和端口等）和 S-CSCF转 发给 VCC AS
步骤 s204, VCC AS通过 S-CSCF返回 180振铃消息给 I-CSCF, 最后由 I-CSCF发送给 OEP。
UE 1产生本地振铃音， 然后等待 UE 1的用户摘机， 本发明中 UE 1根据 一些预定策略（例如无线条件变化， 呼叫已经完成媒体协商）决策到应切换到 CS网络， 则启动 VCC网络切换， 执行下面的步骤：
步骤 s205， UE 1通过 CS网络发起呼叫， 向 MSC/VLR发送始呼消息 (Call Origination),其中所述 Call Origination包括 CdPN(即被叫号码为（ Service Code 加 VCC AS E.164号码 ) )和 CgPN (即主叫号码为 UE 1的号码 MDN ( Mobile Directory Number, 如 133XXXX5678 ) );
如果需要， 就执行 CS网络的鉴权 &注册流程。
步骤 s206， MSC/VLR生成初始地址消息 IAM并发送到 MGCF,使用 VCC AS的 E.164号码作为被叫号码， UE 1的号码作为主叫号码， 且 MSC/VLR丢 弃 Service Code值；
步骤 s207, MSC/VLR向 UE1发送 Traffic Channel Assignment消息， 即 CS网络为 UE1分配业务信道， UE1捕获业 言道；
步骤 s208, MGCF发送地址全消息 ACM到 MSC/VLR;
步骤 s209， MGCF生成 INVITE消息（包括为本次呼叫分配的承载信息 -
IP地址和带宽等，即 MGW SDP )，其中该 INVITE消息中 Request URI为 VCC AS的 E.164号码， 该消息经 I-CSCF发送到 VCC AS;
步骤 s210 ~步骤 s211 , VCC AS检查到呼叫状态为非通话状态，根据收到 的 INVITE消息生成 UPDATE消息（该消息中 Request URI修改为对端的地址）， 发送到对端， 对端应答 200 OK应到消息， 所述 200 OK消息中包括对端应答 的 SDP信息；
步骤 s212， VCC AS根据收到的 200 OK消息生成 183消息，发送到 UE 1 , 然后等待 UE 1的用户摘机； 步骤 s213， UE 1检测到用户摘机 (可以是在步骤 s215之后任一时间发生 ) , 通过 200 OK应答消息将摘机信号发送到 VCC AS , VCC AS识别到用户摘机； 或者如果 UE 1检测到用户摘机，但 IP接入网连接已不可用， 则可在步骤 s214,通过 CS网络的非结构式辅助业务数据（ unstructured supplementary service data, USSD )或短消息业务 SMS等承载方式将摘机信息（如封装 200 OK消 息）发送到 VCC AS;
步骤 s215 ~步骤 s216, VCC AS释放 SIP call dialog 2;
需要说明的， 本实施例中 VCC AS发起释放 SIP call dialog 2发送的 BYE 消息是针对步骤 s213的， 针对步骤 s214所述 VCC AS可改为发送 CANCEL 消息发起释放 SIP call dialog 2;
步骤 s217 ~步骤 s218, VCC AS发送 200 OK响应消息（是对步骤 s201 的最终应答）到对端， 对端以 ACK消息确认；
步骤 s219 ~步骤 s220, VCC AS生成 200 OK响应消息（是对步骤 s209 的最终应答）发送到 MGCF, MGCF可以以 ACK消息应答， MGCF也可以发 送 CON/ANM (连接或地址全消息 )到 MSC, 指示可以进行通话。
图 6是本发明回铃音期间主叫用户设备从 IMS域切换到电路域的的方法 实施例流程图， 本实施例中的消息基于 SIP协议， 具体流程如下：
步骤 s301 , UE 1通过 IMS网络发起呼叫， 发送^舌邀请 INVITE 消息到 UE 1归属的 IMS网络的 S-CSCF。 所述 INVITE 消息包括对端被叫用户设备 的地址， 如 Bob@huawei.com, 包括 UE 1的 SDP信息： 希望建立的媒体类型、 编码方式、 媒体的收发 IP地址和端口等。
基于过滤策略， S-CSCF发送 INVITE消息给 VCC AS, VCC AS作为一个 SIP B2BUA建立与 UE 1之间的 SIP call dialog 1。
步骤 s302, VCC AS返回 INVITE消息到 S-CSCF, 然后发送到对端， 开 始 VCC AS和对端的 SIP call dialog 2的建立。
步骤 s303 , 对端返回 180 Ringing消息（指示对端正在振铃， 等待用户应 答）给 S-CSCF并携带自身 SDP信息（接受的媒体类型、 编码方式、 媒体的 收发 IP地址和端口等）， S-CSCF转发给 VCC AS。 步骤 s304， VCC AS返回 180振铃消息给 S-CSCF, S-CSCF将 180振铃消 息发送给 UE 1， UE 1可能产生本地回铃音， 然后等待对端用户摘机， 本发明 中 UE 1根据一些预定策略（例如无线条件变化， 呼叫已经完成媒体协商等） 决策到应切换到 CS网络， 则启动 VCC网络切换， 执行下面的步骤：
步骤 s305, UE 1 通过 CS 网络发起呼叫， 向 MSC发送始呼消息 Call
Origination, 其中所述 Call Origination包括 CdPN, 所述 CdPN即被叫号码为 ( Service Code + VCC AS E.164号码）和 CgPN (即主叫号码为 UE 1的号码 MDN;
如果需要， 就执行 CS网络的鉴权 &注册流程。
步骤 s306, MSC生成 ISUP IAM (初始地址消息 ) 并发送到 MGCF, 使 用 VCC AS 的 E.164号码作为被叫号码， 而 UE 1的号码作为主叫号码， 且 MSC/VLR丢弃 Service Code值；
步骤 s307， MSC/VLR向 UE发送 Traffic Channel Assignment消息 , 即 CS 网络为 UE1分配业务信道， UE1捕获业务信道；
步骤 s308 , MGCF发送 ACM消息到 MSC/VLR;
步骤 s309， MGCF生成 INVITE消息（其中包括为本次呼叫分配的承载信 息 - IP地址和带宽等， 即 MGW SDP )， 所述 INVITE消息中 Request URI为 VCC AS的 E.164号码， 该消息经 I-CSCF发送到 VCC AS;
步骤 s310 ~步骤 s311 , VCC AS检查呼叫状态为非通话状态，根据收到的 INVITE消息生成 UPDATE消息（其中 Request URI修改为对端的地址），发送 到对端， 对端应答 200 OK响应消息， 所述的 200 OK响应消息中包括对端应 答的 SDP信息；
步骤 s312, VCC AS根据收到的 200 OK消息生成 183消息 （应答步骤 s309 ), 然后将所述的 183消息发送到 UE 1 , 其中所述的 183消息中包括应答 SDP信息， 并等待对端用户的摘机；
步骤 s313 , 对端检测到用户摘机， 发送 200 OK响应消息（是对步骤 s302 的最终应答）到 S-CSCF, S-CSCF前转到 VCC AS;
步骤 s314, VCC AS将该 200 OK响应消息通过 I-CSCF发送给 MGCF, MGCF转换成 ANM/CON (连接或地址全消息 )到 MSC， 指示可以进行通话、- 步骤 s315 ~步骤 s316， MGCF以 ACK应答收到的 200 OK响应消息 , ACK 消息由 VCC AS前转给 S-CSCF, 最后到达对端。
步骤 s317, VCC AS释放 SIP call dialog 1 : 以 4XX/5XX/6XX错误响应消 息来释放。
另外， 本发明还提供一种用户设备的网络切换方法， 所述方法包括步骤： 发起切换的用户设备通过目标网络与对端进行呼叫保持状态下呼叫连续 性切换的会话协商；
在会话协商成功后将呼叫切换到目标网络。
对于该方法的具体实现过程如图 7至 11所述。
参考图 7， 该图是本发明呼叫保持期间实现用户设备从电路域网络切换到 IMS域的第一实施例流程图， 本实施例中基于 SIP协议实现， 其中呼叫保持由 UE1主动发起， UE 1通过 CS网络与对端建立呼叫并进行通话， UE 1根据一 些策略（如无线条件变化）决策到应切换到 IMS 网络， 具体切换主要包括以 下步骤：
步骤 s401 , 如果 UE 1没有进行 IMS注册， 则进行注册；
步骤 s402, UE 1通过 S-CSCF发送 INVITE邀请消息到 VCC AS (作为切 换请求），开始建立 SIP call dialog 3，其中 Request URI为 VCC AS的身份标识 (如 Tel URI, SIPURI ), 也包括 UE 1的 SDP信息， 另外， 需要说明的， 这里 UE1的 SDP信息指示其媒体流属性为只发送；
步骤 s403 ~步骤 s404, VCC AS检查呼叫状态为呼叫保持状态， 根据收到 的 INVITE消息生成 Re-INVITE消息（如 Request URI修改为对端的地址， Contact修改为 VCC AS的身份标识）， 发送到对端进行会话协商（如， 媒体类 型、 编码方式、 媒体的收发 IP地址和端口等）， 其中 SDP信息指示 UE1的媒 体流属性为只发送；
步骤 s405，对端应答 200 OK消息，所述 200 OK消息包括对端应答的 SDP 信息， 需要说明的， 这里对端应答的 SDP信息指示其媒体流属性为只接收， VCC AS将 200 OK消息 （与步骤 s401对应）发送到 UE 1; 步骤 s406, 作为 SIP的三次握手机制， UE 1返回 ACK消息， 最终到达对 端；
步骤 s407, VCC AS发送 BYE消息到 MGCF以释放 CS侧呼叫；
步骤 s408 ~步骤 s411 , CS侧呼叫被释放；
步骤 s412, MGCF收到 RLC消息后， 生成 200 OK消息应答步骤 s407收 到的 BYE消息；
此时 UE 1通过 IMS域与对端通话。
参考图 8， 该图是本发明呼叫保持期间实现用户设备从电路域网络切换到 IMS域的第二实施例流程图， 本实施例中基于 SIP协议实现， 其中呼叫保持由 对端主动发起且 UE1知晓被保持状态， UE 1通过 CS网络与对端建立呼叫并 进行通话， UE 1根据一些策略（如无线条件变化）决策到应切换到 IMS网络， 具体包括以下步骤：
步骤 s501 , 如果 UE 1没有进行 IMS注册， 则进行注册；
步骤 s502, UE 1通过 S-CSCF发送 INVITE邀请消息到 VCC AS (作为切 换请求）， 开始建立 SIP call dialog 3 , 其中 Request URI为 VCC AS的 E.164号 码， 也包括 UE 1的 SDP信息， 另外， 需要说明的， 这里 UE1的 SDP信息指 示其媒体流为只接收；
步骤 s503 ~步骤 s504, VCC AS检查呼叫状态为呼叫保持状态， 根据收到 的 INVITE消息生成 Re-INVITE消息（ Request URI修改为对端的地址），发送 到对端进行会话协商（如， 媒体类型、， 编码方式、 媒体的收发 IP地址和端口 等）， 其中 SDP信息指示 UE1的媒体流属性为只接收；
步骤 s505, 对端应答 200 OK消息， 所述 200 OK消息中包括对端应答的 SDP信息， 需要说明的， 这里对端应答的 SDP信息指示其媒体流属性为只发 送， VCC AS将 200 OK消息 （与步骤 s501对应 )发送到 UE 1；
步骤 s506, 作为 SIP的三次握手机制， UE 1返回 ACK消息， 最终到达对 端；
步骤 s507, VCC AS发送 BYE消息到 MGCF以释放 CS侧呼叫；
步骤 s508 ~步骤 s511 , CS侧呼叫被释放； 步骤 s512, MGCF收到 RLC消息后， 生成 200 OK消息应答步骤 s507收 到的 BYE消息；
此时 UE 1通过 IMS域与对端通话。
参考图 9， 该图是本发明呼叫保持期间实现用户设备从电路域网络切换到 IMS域的第三实施例流程图， 本实施例中基于 SIP协议实现， 其中呼叫保持由 对端主动发起且 UE1不知晓被保持状态， UE 1通过 CS网络与对端建立呼叫 并进行通话， UE 1根据一些策略（如无线条件变化）决策到应切换到 IMS网 络， 具体包括以下步骤：
步骤 s601 , 如果 UE 1没有进行 IMS注册， 则进行注册；
步骤 s602, UE 1通过 S-CSCF发送 INVITE邀请消息到 VCC AS (作为切 换请求）， 开始建立 SIP call dialog 3 , 其中 Request URI为 VCC AS的 E.164号 码， 也包括 UE 1的 SDP信息， SDP信息指示其媒体流属性为既发送也接收； 步骤 s603 ~步骤 s604, VCC AS检查呼叫状态为呼叫保持状态， 根据收到 的 INVITE消息生成 Re-INVITE消息（ Request URI修改为对端的地址），发送 到对端进行会话协商 （如， 媒体类型、 编码方式、 媒体的收发 IP地址和端口 等）， 其中 VCC AS添加的 UE1在 INVITE消息中设置的 SDP信息指示 UE1 的媒体流属性为只接收；
步骤 s605, 对端应答 200 OK, 其中包括对端应答的 SDP信息， 需要说明 的，这里对端应答的 SDP信息为只发送， VCC AS将 200 OK消息（与步骤 s501 对应）发送到 UE l;
步骤 s606, 作为 SIP的三次握手机制， UE 1返回 ACK消息， 最终到达对 端；
步骤 s607, VCC AS发送 BYE消息到 MGCF以释放 CS侧呼叫；
步骤 s608 ~步骤 s611 , CS侧呼叫被释放；
步骤 s612, MGCF收到 RLC消息后， 生成 200 OK消息应答步骤 s507收 到的 BYE消息；
此时 UE 1可通过 IMS域网络与对端通话。
参考图 10， 该图是本发明呼叫保持期间用户设备从 IMS域网络切换到电 路域的第一实施例流程图， 本实施例中基于 SIP协议实现， 其中呼叫保持由 UE1主动发起， UE 1通过 IMS网络与对端建立呼叫并进行通话， UE 1根据一 些策略（如无线条件变化）决策到应切换到 CS网络， 具体切换主要包括以下 步骤：
步骤 s701 , UE 1通过 CS网络发起呼叫， 其中 CdPN被叫号码为 （ Service Code + VCC AS E.164号码）， CgPN主叫号码为 UE 1的号码 MDN ( Mobile Directory Number, 如 133xxxx5678 );
步骤 s702 ~步骤 s705, 如果需要， 就执行 CS网络的鉴权 &注册流程； 步骤 s706, MSC生成 ISUP IAM (初始地址消息）并发送到 MGCF, 使用 VCC AS E.164号码做被叫号码， 而 UE 1的号码做主叫号码， 且 MSC丢弃 Service Code值；
步骤 s707 , MSC/VLR向 UE发送 Channel Assignment消息 , 即 CS网络为 UE分配业务信道；
步骤 s708, MGCF可能发送 ACM (地址全 )到 MSC/VLR;
步骤 s709, MGCF生成 INVITE消息（包括为本次呼叫分配的承载信息， 即 MGW SDP -包括 IP地址和带宽等， SDP信息指示 MGW的媒体流属性为 既发送也接收）， Request URI为 VCC AS的 E.164号码， 该消息经 I-CSCF发 送到 VCC AS;
步骤 s710 ~步骤 s711 , VCC AS才 据收到的 INVITE消息生成 Re-INVITE 消息（Request URI修改为对端的地址），经 S-CSCF发送到对端， 需要说明的， 这里会话描述信息指示 MGW的媒体流属性为只发送；
步骤 s712〜步骤 s713 , 对端应答 200 OK消息经 S-CSCF到达 VCC AS, 其 中包括对端应答的 SDP信息， 需要说明的， 这里对端应答的 SDP信息指示其 媒体流属性为只接收；
步骤 s714， VCC AS将 200 OK消息经 I-CSCF发送到 MGCF,需要说明的， 这里所述 200 OK消息携带的对端的 SDP信息指示对端的媒体流属性为既发送 也接收；
步骤 s715, 相应的， MGCF生成 ANM (应答消息）发送到 VMSC; 步骤 s716，作为 SIP三次握手机制， MGCF生成 ACK消息发送到 VCC AS; 步骤 s717 ~步骤 s718. VCC AS将 ACK消息经 S-CSCF发送到对端； 步骤 s719, 在步骤 s709后任一步骤， UE和 CS网络完成业务信道捕获； 步骤 s720 ~ s721 , VCC AS发送 ACK消息到对端后， 即可以发送 BYE消 息到 UE 1 , 以释放 IMS侧的呼叫支路， UE 1返回 200 OK消息应答。
另外， 由于 MSC不知晓 UE1的呼叫保持状态， 这里 UE1还需要向 MSC 同步呼叫保持状态， 即在步骤 s722, UE1向 MSC发送呼叫保持请求（例如发 送 Flash with information/Hold Request消息到 MSC );
而 MSC收到所述呼叫保持请求后， 向 MGCF发送呼叫保持请求， 例如发 送呼叫进行消息（ Call Progress message , CPG ) , 所述 CPG消息包括对端呼叫 保持指示； MGCF转换生成 Re-INVITE/UPDATE消息发送到 VCC AS (步骤 s723 ~步骤 s726 )， 其中 SDP内容包括 a=sendonly属性行， 需要说明的， 若 MGCF使用 UPDATE消息， 不需要向 VCC AS发送 ACK消息。
参考图 11， 该图是本发明呼叫保持期间用户设备从 IMS域网络切换到电 路域的第二实施例流程图， 本实施例中基于 SIP协议实现， 其中呼叫保持由对 端主动发起， UE 1通过 IMS网络与对端建立呼叫并进行通话， UE 1根据一些 策略（如无线条件变化）决策到应切换到 CS网络， 具体切换主要包括以下步 骤：
步骤 s801 , UE 1通过 CS网络发起呼叫， 其中 CdPN被叫号码为 （ Service Code + VCC AS E.164号码）， CgPN主叫号码为 UE 1的号码 MDN ( Mobile Directory Number, 如 133xxxx5678 );
步骤 s802 ~步骤 s805, 如果需要， 就执行 CS网络的鉴权 &注册流程； 步骤 s806, MSC生成 ISUP IAM (初始地址消息）并发送到 MGCF, 使用 VCC AS E.164号码做被叫号码， 而 UE 1的号码做主叫号码， 且 MSC丢弃 Service Code Κ
步骤 807 , MSC/VLR向 UE发送 Channel Assignment消息 , 即 CS网络为 UE分配业务信道；
步骤 s808, MGCF可能发送 ACM (地址全）到 MSC; 步骤 s809, MGCF生成 INVITE消息（包括为本次呼叫分配的承载信息， 即 MGW SDP -包括 IP地址和带宽等， SDP信息指示 MGW的媒体流属性为 既发送也接收）， Request URI为 VCC AS的 E.164号码， 该消息经 I-CSCF发 送到 VCC AS;
步骤 s810 ~步骤 s811 , VCC AS才 据收到的 INVITE消息生成 Re-INVITE 消息（Request URI修改为对端的地址），经 S-CSCF发送到对端， 需要说明的， 这里 MGCF的会话描述信息指示 MGW的媒体流属性为只接收；
步骤 s812〜步骤 s813 , 对端应答 200 OK消息经 S-CSCF到达 VCC AS, 其 中包括对端应答的 SDP信息， 需要说明的， 这里对端应答的 SDP信息指示其 媒体流属性为只发送；
步骤 s814， VCC AS将 200 OK消息经 I-CSCF发送到 MGCF,需要说明的， 这里所述 200 OK消息携带的对端的 SDP信息指示对端的媒体流属性为既发送 也接收；
步骤 s815, 相应的， MGCF生成 ANM (应答消息）发送到 VMSC;
步骤 s816，作为 SIP三次握手机制， MGCF生成 ACK消息发送到 VCC AS; 步骤 s817 ~步骤 s818. VCC AS将 ACK消息经 S-CSCF发送到对端； 步骤 s819, 在步骤 s809后任一步骤， UE和 CS网络完成业务信道捕获； 步骤 s820 ~ s821 , VCC AS发送 ACK消息到对端后， 即可以发送 BYE消 息到 UE 1 , 以释放 IMS侧的呼叫支路， UE 1返回 200 OK消息应答。
另外， 由于 MSC不知晓呼叫保持状态， 本实施例中 VCC AS还需要向
UE1所在的电路域网络同步呼叫保持状态， 即切换后 VCC AS向 MGCF发送 Re-INVITE/UPDATE消息将呼叫保持 (步骤 s822 ~步骤 s826, 其中 SDP内容 包括 a=sendonly属性行）， MGCF转换生成 CPG消息，包括 Remote hold指示， 然后发送到 MSC/VLR, MSC/VLR可能向 UE 1发送呼叫被保持指示消息（如 GSM/UMTS 网 络下 的 消 息 为 Facility(Invoke = NotifySS (HOLD, CallOnHold-Indicator))。
若 VCC AS使用 UPDATE消息， 不需要向 MGCF发送 ACK消息。
参考图 12，该图是本发明用户设备实现振铃期间进行 VCC切换的功能结 构示意图， 其中所述用户设备包括： 呼叫处理单元， 本发明中改进所述呼叫处 理单元， 即本发明中在所述呼叫处理单元设置：
振铃期间语音呼叫连续性切换处理单元 11， 用于在振铃期间进行语音呼 叫连续性的网络切换。
按照其实现的逻辑功能划分，本实施例中所述振铃期间语音呼叫连续性切 换处理单元 11具体包括 3个逻辑功能单元， 即： 振铃期间语音呼叫连续性切 换单元 111、 会话协商处理单元 112以及摘机信号发送单元 113， 其中
会话协商处理单元 112, 用于通过目标网络与对端主叫用户设备进行振铃 期间呼叫连续性的会话协商；也就是说主要用于进行所述振铃期间语音呼叫连 续性切换的会话协商， 本发明中其主要是发起到 VCC应用服务器的呼叫， 以 与对端进行会话协商， 具体处理参考前述流程说明；
所述振铃期间呼叫连续性切换单元 111 , 用于在所述会话协商成功后进行 振铃期间呼叫连续性的网络切换；
所述摘机信号发送单元 113， 用于根据所述振铃期间呼叫连续性切换单元 111的切换结果向对端主叫用户设备发送摘机信号； 也就是说在网络切换成功 后向语音呼叫连续性应用服务器传送摘机信号；
参考图 13，该图是本发明用户设备实现回铃音期间进行 VCC切换的功能 结构示意图，其中所述用户设备包括呼叫处理单元，本发明中改进所述呼叫处 理单元， 即本发明中在所述呼叫处理单元设置：
回铃音期间呼叫连续性切换处理单元 12， 用于在回铃音期间进行语音呼 叫连续性的网络切换。
按照其实现的逻辑功能划分，本发明中所述回铃音期间语音呼叫连续性切 换处理单元具体包括有三个逻辑功能单元 ,即回铃音期间语音呼叫连续性切换 单元 121、 会话协商处理单元 122和接收单元 123， 其中
回铃音期间语音呼叫连续性切换单元 121， 在所述会话协商成功后进行回 铃音期间呼叫连续性的网络切换。
会话协商处理单元 122， 用于通过目标网络与对端被叫用户设备进行回铃 音期间呼叫连续性切换的会话协商； 也就是说， 主要用于进行所述回铃音期间 语音呼叫连续性切换的会话协商， 本发明中其主要是发起到 VCC应用服务器 的呼叫， 以与对端进行会话协商， 具体处理参考前述流程说明。
接收单元 123 , 用于接收对端被叫用户设备在所述网络切换完成后发送的 摘机信号。
此外，按照其实现的逻辑功能划分，本发明中所述回铃音期间语音呼叫连 续性切换处理单元具体包括有两个逻辑功能单元，即回铃音期间语音呼叫连续 性切换启动单元和会话协商处理单元，其中所述回铃音期间语音呼叫连续性切 换启动单元， 主要用于根据预定策略启动回铃音期间语音呼叫连续性切换。所 述会话协商处理单元， 主要用于进行语音呼叫连续性切换的会话协商，本发明 中其主要是发起到 VCC应用服务器的呼叫， 以与对端进行会话协商， 具体处 理参考前述流程说明。
参考图 14， 该图是本发明呼叫连续性应用服务器振铃或回铃音期间进行 VCC切换控制的功能的结构示意图， 本发明中改进所述呼叫连续性应用服务 器， 即本发明中在所述呼叫连续性应用服务器设置：
振 /回铃音期间呼叫连续性切换控制单元 13， 用于在振铃或回铃音期间控 制进行语音呼叫连续性的网络切换。
按照其实现的逻辑功能划分， 本发明中所述振 /回铃音期间语音呼叫连续 性切换控制单元主要包括 2个功能单元， 即振 /回铃音间语音呼叫连续性协商 控制单元和摘机信号识别传送单元， 其中
振 /回铃音间语音呼叫连续性会话协商控制单元 131， 用于控制进行振铃 / 回铃音期间呼叫连续性的网络切换的会话协商； 具体处理参考前述流程说明 , 这里不再赘述；
摘机信号识别传送单元 132， 主要用于识别被叫用户设备的摘机信号并传 送给主叫用户设备， 完成切换。
优选的， 还可以包括：
释放单元 133， 主要用于在所述会话协商成功后释放待切换网络中的呼叫 支路。
参考图 15，该图是本发明用户设备实现呼叫保持期间进行 VCC网络切换 的功能结构示意图，其中所述用户设备包括呼叫处理单元，本发明中改进所述 呼叫处理单元， 即本发明中在所述呼叫处理单元设置：
呼叫保持期间语音呼叫连续性切换处理单元 14， 主要用于在呼叫保持期 间进行语音呼叫连续性的网络切换。
按照其实现的逻辑功能划分，本发明中所述呼叫保持期间语音呼叫连续性 切换处理单元具体包括有两个逻辑功能单元，即会话协商处理单元 142和呼叫 保持期间语音呼叫连续性切换单元 141， 其中
会话协商处理单元 142 用于在目标网络与对端进行呼叫保持状态下的呼 叫连续性切换的^舌协商；即主要用于控制进行所述呼叫保持期间的语音呼叫 切换到目标网络的会话协商；
呼叫保持期间语音呼叫连续性切换单元 141， 用于在会话协商成功后将所 述用户设备切换到目标网络。
此外，按照其实现的逻辑功能划分，本发明中所述呼叫保持期间语音呼叫 连续性切换处理单元具体包括有两个逻辑功能单元，即呼叫保持期间语音呼叫 连续性切换启动单元和会话协商处理单元，其中所述呼叫保持期间语音呼叫连 续性会话协商处理单元主要用于根据预定策略识别并启动呼叫保持期间的语 音呼叫连续性；所述会话协商处理单元主要用于控制进行呼叫保持期间切换到 目标网络的会话协商。
参考图 16, 该图是本发明 VCC应用服务器呼叫保持期间进行 VCC网络 切换控制的功能结构示意图， 本发明中改进所述 VCC应用服务器， 即本发明 中在所述 VCC应用服务器设置有：
呼叫保持期间语音呼叫连续性切换控制单元 15 , 主要用于在呼叫保持期 间控制进行语音呼叫连续性的网络切换。
按照其实现的逻辑功能划分，本发明中所述呼叫保持期间语音呼叫连续性 切换控制单元主要包括 2个功能单元，即会话协商处理单元 152和呼叫保持期 间呼叫连续性切换控制单元 151， 其中 所述会话协商处理单元 152, 用于控制进行呼叫保持期间切换到目标网络 的会话协商；即主要用于控制进行所述呼叫保持期间的语音呼叫切换到目标网 络的会话协商；
呼叫保持期间呼叫连续性切换控制单元 151 , 用于在会话协商成功后控制 呼叫保持期间的呼叫连续性的网络切换。
优选的， 在本实施例中， 所述 VCC应用服务器还可包括：
释放单元 154， 主要用于在所述会话协商成功后释放待切换网络中的呼叫 支路。
此外，按照其实现的逻辑功能划分，本发明中所述呼叫保持期间语音呼叫 连续性切换控制单元主要包括 2个功能单元，即呼叫保持期间语音呼叫连续性 切换启动单元和会话协商处理单元，其中， 所述呼叫保持期间语音呼叫连续性 切换启动单元 151， 主要用于根据预定策略识别并启动呼叫保持期间的语音呼 叫连续性； 所述会话协商处理单元 152, 主要用于控制进行呼叫保持期间切换 到目标网络的 ^舌协商。
与前面类似的， 本实施里中， 所述 VCC应用服务器还可包括：
释放单元 153， 主要用于^舌协商成功后释放待切换网络中的呼叫支路。 以上所述仅是本发明的优选实施方式，应当指出，对于本技术领域的普通 技术人员来说， 在不脱离本发明原理的前提下， 还可以作出若干改进和润饰， 这些改进和润饰也应视为本发明的保护范围。

Claims

权 利 要 求
1、 一种用户设备的网络切换方法， 其特征在于， 包括：
发起切换的用户设备通过目标网络与对端进行呼叫保持状态下呼叫连续 性切换的会话协商；
在会话协商成功后将呼叫切换到目标网络。
2、 根据权利要求 1所述的方法， 其特征在于， 所述用户设备在目标网络 通过呼叫连续性应用服务器与对端进行呼叫保持状态下的会话协商。
3、 根据权利要求 2所述的方法， 其特征在于， 所述方法还包括： 所述用 户设备向所述呼叫连续性应用服务器发送携带该用户设备的会话描述信息的 网络切换请求；所述会话协商进一步指：根据所述会话描述信息进行会话协商； 所述在会话协商成功后将呼叫切换到目标网络进一步指：根据所述会话描述信 息会话协商成功后， 将对应所述网络切换请求的呼叫切换到目标网络。
4、 根据权利要求 3所述的方法， 其特征在于， 所述会话描述信息中携带 该发起切换的用户设备的呼叫状态保持信息。
5、 根据权利要求 3或 4所述的用户设备的网络切换方法， 其特征在于， 发起切换的用户设备待切换的网络为电路域网络， 目标网络为 IP多媒体子系 统网络；所述发起切换的用户设备通过目标网络中的呼叫连续性应用服务器与 对端进行呼叫保持状态下的会话协商的具体包括：
发起切换的用户设备通过 IP多媒体子系统网络向呼叫连续性应用服务器 发送携带该用户设备的会话描述信息的网络切换请求。
6、 根据权利要求 3或 4所述的方法， 其特征在于， 待切换网络为 IP多媒 体子系统网络， 目标网络为电路域网络；
所述发起切换的用户设备在目标网络通过呼叫连续性应用服务器与对端 进行呼叫保持状态的会话协商的具体包括：发起切换的用户设备通过电路域网 络、移动交换中心、媒体网关控制功能单元向呼叫连续性应用服务器发送邀请 消息， 所述邀请消息包括： 该媒体网关控制功能单元的会话描述信息。
7、 一种用户设备的网络切换方法， 其特征在于， 包括：
接收来自发起切换的用户设备的网络切换请求，所述网络切换请求携带所 述用户设备会话描述信息；
若检测到所述呼叫为呼叫保持状态， 则对所述会话描述信息更新， 并向对 端发送携带所述发起切换的用户设备的呼叫保持状态的会话描述信息；
在所述用户设备与对端通过所述会话描述信息会话协商成功后，根据所述 网络切换请求将对应的呼叫切换到目标网络。
8、 根据权利要求 7所述的方法， 所述方法还包括： 接收来自对端反馈的 携带该对端的呼叫保持状态的会话描述信息；向所述发起切换的用户设备发送 携带对端的呼叫保持状态的会话描述信 , 。
9、 根据权利要求 7所述的方法， 其特征在于， 所述检测到的呼叫保持状 态为所述发起切换的用户设备主动发起的呼叫保持；
所述发起切换的用户设备的呼叫保持的会话描述信息为只发送的会话描 述信息； 所述对端的呼叫保持状态的会话描述信息为只接收的会话描述信息。
10、根据权利要求 7所述的方法， 其特征在于， 所述呼叫连续性应用服务 器检测到的呼叫保持状态为由对端发起的呼叫保持；
所述发起切换的用户设备的呼叫保持的会话描述信息为只接收的会话描 述信息； 所述对端的呼叫保持的会话描述信息为只发送的会话描述信息。
11、 根据权利要求 9或 10所述的方法， 其特征在于， 所述只接收的会话 描述信息由所述待切换的用户设备或呼叫连续性应用服务器添加。
12、根据权利要求 7 - 11中任一项所述的用户设备的网络切换方法， 其特 征在于， 还包括： 释放待切换网络的呼叫支路。
13、 根据权利要求 7 - 10所述的方法， 其特征在于， 待切换网络为 IP多 媒体子系统网络， 目标网络为电路域网络；
从媒体网关控制功能单元接收来自发起切换的用户设备通过电路域、移动 交换中心发送的邀请消息， 所述邀请消息包括： 该媒体网关控制功能单元的会 话描述信息；
对所述会话描述信息更新，并向对端发送携带所述发起切换的用户设备的 呼叫保持状态的会话描述信息进一步指：
根据所述会话邀请消息生成所述会话描述信息的更新消息 ,并向对端发送 携带媒体网关控制功能单元的呼叫保持状态的会话描述信息的邀请或更新消 息。
14、 根据权利要求 13所述的方法， 其特征在于， 所述检测到的呼叫保持 状态为由所述发起切换的用户设备主动发起的呼叫保持；
所述媒体网关控制功能单元的呼叫保持状态的会话描述信息为只发送的 会话描述信息；
所述对端的呼叫保持状态的会话描述信息为只接收的会话描述信息； 会话协商成功后还包括：所述发起切换的用户设备与所述移动交换中心同 步呼叫保持状态。
15、 根据权利要求 13所述的方法， 其特征在于， 所述检测到的呼叫保持 状态为由对端发起的呼叫保持；
所述媒体网关控制功能单元的呼叫保持状态的会话描述信息为只接收的 会话描述信息；
所述对端的呼叫保持状态的会话描述信息为只发送的会话描述信息； ^舌协商成功后还包括：所述呼叫连续性应用服务器与电路域网络同步呼 叫保持状态。
16、 一种用户设备， 其特征在于， 包括：
会话协商处理单元，用于通过目标网络与对端进行呼叫保持状态下的呼叫 连续性切换的会话协商；
呼叫保持期间呼叫连续性切换单元，用于在会话协商成功后将呼叫切换到 目标网络。
17、 一种呼叫连续性应用服务器， 其特征在于， 包括：
呼叫保持期间呼叫连续性会话协商处理单元，用于控制进行呼叫保持期间 切换到目标网络的会话协商；
呼叫保持期间呼叫连续性切换控制单元，用于在会话协商成功后控制呼叫 保持期间的呼叫连续性的网络切换。
18、 根据权利要求 17所述的呼叫连续性应用服务器， 其特征在于， 所述 呼叫连续性应用服务器还包括： 释放单元， 用于在所述会话协商成功后释放待切换网络中的呼叫支路。
19、 一种用户设备的网络切换方法， 其特征在于， 包括：
被叫用户设备通过目标网络与对端主叫用户设备进行振铃期间呼叫连续 性切换的会话协商；
在所述会话协商成功后呼叫被切换到目标网络，被叫用户设备向对端主叫 用户设备发送摘机信号。
20、 根据权利要求 19所述的用户设备的网络切换方法， 其特征在于， 所 述被叫用户设备通过目标网络与对端主叫用户设备进行振铃期间呼叫连续性 切换的会话协商进一步指：所述被叫用户设备通过目标网络的呼叫连续性应用 服务器与对端主叫用户设备进行振铃期间呼叫连续性切换的会话协商。
21、 根据权利要求 20所述的用户设备的网络切换方法， 其特征在于， 所 述被叫用户设备向对端主叫用户设备发送摘机信号进一步指：所述被叫用户设 备通过所述呼叫连续性应用服务器向对端主叫用户设备发送摘机信号。
22、 根据权利要求 21所述的用户设备的网络切换方法， 其特征在于， 所 述被叫用户设备将摘机信号发送给所述呼叫连续性应用服务器为：
通过所述被叫用户设备所在的待切换网络将摘机信号的信令发送给所述 呼叫连续性应用服务器。
23、 根据权利要求 21所述的用户设备的网络切换方法， 其特征在于， 所 述被叫用户设备将摘机信号发送给所述呼叫连续性应用服务器为：
通过会话发起协议的通知消息或非结构化补充业务数据服务或短消息服 务将摘机信号发送给所述呼叫连续性应用服务器。
24、 一种用户设备的网络切换方法， 其中待切换网络为 IP多媒体子系统 网络， 目标网络为电路域网络， 其特征在于， 所述方法包括：
主叫用户设备通过目标网络与对端被叫用户设备进行回铃音期间呼叫连 续性切换的会话协商；
在所述会话协商成功后呼叫被切换到目标网络；主叫用户设备接收对端发 送的摘机信号。
25、 根据权利要求 24所述的用户设备的网络切换方法， 其特征在于， 所 述主叫用户设备通过目标网络的呼叫连续性应用服务器与对端被叫用户设备 进行回铃音期间呼叫连续性切换的会话协商。
26、 一种用户设备， 其特征在于， 包括：
会话协商处理单元，用于通过目标网络与对端主叫用户设备进行振铃期间 呼叫连续性的会话协商；
振铃期间呼叫连续性切换单元，用于在所述会话协商成功后进行振铃期间 呼叫连续性的网络切换；
摘机信号发送单元，用于根据所述呼叫连续性切换单元的切换结果向对端 主叫用户设备发送摘机信号。
27、 一种用户设备， 其特征在于， 包括：
会话协商处理单元 ,用于通过目标网络与对端被叫用户设备进行回铃音期 间呼叫连续性切换的会话协商；
回铃音期间呼叫连续性切换单元，用于在所述会话协商成功后进行回铃音 期间呼叫连续性的网络切换；
接收单元，用于接收对端被叫用户设备在所述网络切换完成后发送的摘机 信号。
28、 一种呼叫连续性应用服务器， 其特征在于， 包括：
振 /回铃音期间呼叫连续性会话协商控制单元， 用于控制进行振铃 /回铃音 期间呼叫连续性的网络切换的会话协商；
摘机信号识别传送单元，用于识别被叫用户设备的摘机信号并传送给主叫 用户设备。
29、 根据权利要求 15所述的呼叫连续性应用服务器， 其特征在于， 所述 呼叫连续性应用服务器还包括：
释放单元， 用于在所述会话协商成功后释放待切换网络中的呼叫支路。
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