WO2012126321A1 - 一种实现单接入系统语音连续性的方法及系统 - Google Patents

一种实现单接入系统语音连续性的方法及系统 Download PDF

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Publication number
WO2012126321A1
WO2012126321A1 PCT/CN2012/072186 CN2012072186W WO2012126321A1 WO 2012126321 A1 WO2012126321 A1 WO 2012126321A1 CN 2012072186 W CN2012072186 W CN 2012072186W WO 2012126321 A1 WO2012126321 A1 WO 2012126321A1
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Prior art keywords
voice
srvcc
network
srtp
data packet
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PCT/CN2012/072186
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English (en)
French (fr)
Inventor
田甜
韦银星
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中兴通讯股份有限公司
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Publication of WO2012126321A1 publication Critical patent/WO2012126321A1/zh

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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1083In-session procedures
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1083In-session procedures
    • H04L65/1095Inter-network session transfer or sharing
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04WWIRELESS COMMUNICATION NETWORKS
    • H04W12/00Security arrangements; Authentication; Protecting privacy or anonymity
    • H04W12/03Protecting confidentiality, e.g. by encryption
    • H04W12/033Protecting confidentiality, e.g. by encryption of the user plane, e.g. user's traffic

Definitions

  • the present invention relates to communication network security technologies, and in particular, to a method and system for implementing Single Radio Voice Call Continuity (SRVCC). Background technique
  • the Evolved Packet System (EPS) network evolution is proposed in the R8 version of the 3rd Generation Partnership Project (3GPP).
  • 3GPP 3rd Generation Partnership Project
  • the wireless network is an Evolved Universal Terrestrial Radio Access Network (E-UTRAN), which uses Long Term Evolution (LTE) access.
  • E-UTRAN Evolved Universal Terrestrial Radio Access Network
  • LTE Long Term Evolution
  • the SRVCC solution mainly solves the problems in the LTE network when deploying Voice over Internet Protocol (VoIP) services.
  • VoIP Voice over Internet Protocol
  • the ongoing VoIP service will face the problem of voice continuity after leaving the LTE range.
  • the service-call session control function anchors the signaling path to the service set and continuity in the IP Multimedia Subsystem (IMS) network based on the user's subscription to the Home Subscriber Server (HSS).
  • IMS IP Multimedia Subsystem
  • HSS Home Subscriber Server
  • SCC Service-call session control function
  • the SRVCC service can switch voice to Circuit Switched (CS) to ensure uninterrupted voice calls.
  • SRVCC currently only has LTE to CS switching requirements, and there is no need for reverse switching.
  • the SRVCC solution is implemented based on IMS. Therefore, IMS needs to be deployed on the network. After the VoIP service is enabled on the EPS network, SRVCC needs to be used in specific scenarios.
  • FIG. 1 is a schematic diagram of a voice continuity system architecture of a conventional single access system, as shown in FIG.
  • voice continuity system In a single access system voice continuity system:
  • CS network is mainly used for 3G/2G CS domain network access
  • the Enhanced Mobile Switch Center is used to help the UE complete the voice continuity between the CS domain and the Packet Switch (PS).
  • PS Packet Switch
  • the eMSC includes the eMSC server (eMSC). Server and enhanced media gateway (eMGW), where eMSC Server is mainly responsible for signaling plane control, and eMGW is mainly responsible for media plane connection;
  • the EPS network is mainly responsible for LTE network access and LTE network resource management;
  • the Call Session Control Function (CSCF) is responsible for IMS network session control;
  • the SCC AS is used to guarantee the continuity of the IMS session service.
  • the UE first accesses the IMS network through the EPS, establishes a SIP session with the remote end (the remote end), and the CSCF anchors the session to the SCC AS.
  • the eMSC notifies the SCC AS to pass the original.
  • the call branch of the EPS network is switched to connect to the eMSC.
  • the SDES solution is based on the mature IETF standard RFC4568 "Session Description Protocol (SDP) Security Descriptions for Media Streams", which expands a special key attribute in SDP to transmit the media key to be used directly in the SDP.
  • SDP Session Description Protocol
  • the scheme is very simple. For example, user A calls user B, and carries the key K1 in the SDP key attribute of the INVITE message, and K1 is used to encrypt the media stream sent from user A to user B. After receiving the K1, B will save the key to decrypt the encrypted media stream received from User A; the call is successfully established, and the SDP key attribute carried by User B in the 200 OK message returned to User A carries the key. K2, K2 is used to encrypt the media stream sent by the user to the user.
  • FIG. 2 is a schematic diagram of a KMS-based IMS media security solution architecture in the prior art, where user A and user B are senders and receivers of media information, respectively; KMS acts as a trusted third party to implement key management and distribution.
  • Function Proxy-Call Session Control Function (P-CSCF) and Service-Call Session Control Function (S-CSCF) are IMS network elements.
  • P-CSCF Proxy-Call Session Control Function
  • S-CSCF Service-Call Session Control Function
  • each user equipment can establish a trust relationship with the KMS through the GBA mode.
  • Each user equipment and the KMS establish a shared key. If the GBA cannot be used, the user equipment can obtain the shared secret with the KMS by using other methods such as a pre-shared key. key.
  • FIG. 3 is a basic flow chart of the KMS scheme, which is briefly described as follows:
  • the session caller first sends a REQUEST_INIT message to the KMS requesting a ticket containing the relevant key material, in which the caller obtains the relevant key from the KMS request. Will be encrypted and included in it.
  • the KMS returns a REQUEST_RESP message to the calling party, which contains the ticket requested by the calling party.
  • the calling party After the calling party obtains the ticket containing the relevant key, the calling party sends the ticket to the called party in the TRANSFER_INIT message; since the called party cannot decrypt the ticket to obtain the information contained therein, the called party then places the ticket in the The RESOLVE-INIT message continues to be sent to the KMS.
  • the KMS decrypts the ticket and returns the relevant key to the called party in the RESOLVE-RESP message. After the called party obtains the relevant key material, it sends the TRANSFER-RESP to the calling party. Message.
  • the UE 1 and the UE 2 establish an IP bearer connection in the IMS network by using SIP signaling, encrypt the user voice data through the media encryption layer, and the key material required for the encryption/decryption of the UE 1 and UE 2 media encryption layers according to FIG. 3
  • the process acquisition specifically, is obtained by the UE 1 and the UE 2 through the IMS media plane security scheme negotiation during the call setup process;
  • the voice data layer is mainly used for transmitting voice data between the UE 1 and the UE 2, and the layer is an end-to-end protocol transparently transmitted in the IP bearer network; Secure Real-Time Transport Protocol (SRTP) is mainly used to encrypt the voice data layer;
  • SRTP Secure Real-Time Transport Protocol
  • the IP layer and the UDP layer are IP network underlying link transmission protocols.
  • FIG. 5 is a schematic flowchart of a UE initiating an SRVCC service.
  • the UE accesses the IMS network through the EPS network, and establishes a SIP session with the remote node in the IMS network.
  • UE1 switches from the LTE network to the traditional CS network.
  • the process specifically includes:
  • Step 501 The EPS network finds that the radio quality of the UE is deteriorated, and determines to trigger the SRVCC handover, and sends a handover request message to the eMSC, where the message carries the identity identifier of the UE and the SRVCC handover number STN-SR;
  • Step 502. The eMSC notifies the CS network to prepare CS wireless and wired resources;
  • Step 503. After the preparation is completed, return a handover response message to the EPS network, where the message carries the destination radio resource access parameter.
  • Step 504. The EPS network sends a handover request message to the UE.
  • Step 505. The UE accesses the destination CS network according to the destination radio resource access parameter in the handover request message.
  • the CS network After the UE access is completed, the CS network returns a handover complete message to the eMSC, and the eMSC simultaneously notifies the EPS network of the handover completion message;
  • Step 508 After receiving the handover complete message, the EPS may choose to release the resources originally accessed by the UE.
  • Step 509 The eMSC returns a handover response message to the EPS network, and sends a session invitation message (Invite) to the SCC AS in the IMS network.
  • a session invitation message Invite
  • Step 510 The SCC AS associates with the previously anchored session according to the user identity, and sends a re-invite request or an Update request message to the Remote End to report the peer IP address information in the Remote End.
  • the IP address information of the UE is changed to eMSC address information;
  • Step 511 After the remote update of the SCC AS is successful, return a 200 OK message to the eMSC, from The communication between the UE and the Remote End is switched from the original LTE network to the CS network.
  • 6 is a schematic diagram of a prior art media plane protocol stack after the UE switches to the CS network.
  • the UE and the eMSC transmit voice data packets through the CS bearer, and the eMSC acts as a relay node, and the CS bearer Convert to IP bearer and then communicate with Remote End over IP network.
  • the CS bearer protocol is mainly used for transmitting a voice packet underlying transport protocol in the CS network, such as an AAL2 protocol, an STM or an RTP protocol.
  • the main purpose of the present invention is to provide a method and system for implementing voice continuity of a single access system, which can provide an end-to-end media security protection mechanism of the CS network in an SRVCC scenario, and improve the user communication security level.
  • a method for implementing voice continuity SRVCC in a single access system comprising:
  • the user terminal UE adopts an encryption mechanism for the voice media of the session, and after the SRVCC handover is completed, the UE continues to encrypt the voice data by using the encryption mechanism, and uses the encrypted voice data stream as a voice data packet, and performs CS switching through the circuit. network transmission.
  • the method further includes: the UE acquiring the CS network to support the SRVCC media security capability.
  • the UE acquires the CS network to support the SRVCC media security capability: the UE acquires the CS network to support the SRVCC media security capability by using the handover command message in the pre-switched EPS network, or obtains the CS network by using the message in the CS network after the handover.
  • the method further includes: acquiring, by the UE, the CSVCC media security capability of the CS network, and acquiring the media security capability of the remote node SRVCC.
  • the UE After the SRVCC handover is completed, the UE continues to use the encryption mechanism before the handover to encrypt the voice data, and uses the key material negotiated by the same session before the SRVCC handover for protection; or uses the regenerated key material for protection.
  • the encryption mechanism is an encryption mechanism based on the Secure Real-Time Transport Protocol (SRTP).
  • SRTP Secure Real-Time Transport Protocol
  • the method further includes: after the UE completes the SRVCC handover, when the eMSC receives the voice encrypted media stream transmitted by the UE through the CS network, the eMSC encapsulates the voice encrypted media stream into an SRTP packet and transmits the same to the remote node; When the SRTP packet of the remote node is encapsulated, the SRTP packet is encapsulated into a voice data packet and transmitted to the UE through the CS network.
  • the method further includes: determining, by the eMSC, that the current voice media is a voice encrypted media stream, specifically, when the SRVCC is switched, by the EPS network or
  • the service continuity server SCC AS in the IMS network informs the eMSC whether the current voice media is a voice encrypted media stream.
  • the eMSC determines whether the current voice media is a voice encrypted media stream according to the media stream detection.
  • the eMSC encapsulates the voice encrypted media stream that is transmitted by the UE through the CS network into an SRTP packet and delivers the packet to the remote node: when the eMSC has the SRVCC media security capability, the SRTP packet is directly encapsulated on the UDP layer for transmission. When the eMSC does not have the SRVCC media security capability, the SRTP packet is encoded according to the RTP protocol, and the generated RTP data packet is encapsulated on the UDP layer for transmission;
  • the eMSC encapsulates the SRTP packet from the remote node into a voice data packet, and the UE transmits the SRTP packet to the CS bearer protocol layer as a user voice data packet by the CS network.
  • the method further includes: the remote node determines that the UE has initiated an SRVCC handover, and the eMSC does not have the SRVCC media security capability, and the remote node receives the data of the UE.
  • the transmission control layer protocol in the data packet is subjected to secondary decoding processing, and when the voice data packet is transmitted to the UE, the voice data packet is subjected to secondary encoding processing and then transmitted.
  • the secondary decoding process is: first parsing the RTP message, and further parsing according to the SRTP layer protocol to obtain a voice data packet.
  • the secondary encoding process is then sent as: encoding the voice data packet according to the SRTP protocol, generating the SRTP data packet, and continuing to encode the SRTP data packet according to the RTP protocol, and finally generating the RTP data packet and then the RTP.
  • the data packet is encapsulated at the UDP protocol layer and passed to the UE.
  • the remote node determines that the UE has initiated the SRVCC handover: the remote node receives the re-session invitation message sent by the SCC AS or the update message carries the SRVCC indication, or the remote node receives the RTP report of the opposite end.
  • the type of encoding indicated in the text has changed,
  • the SRVCC indication is indicated by a message header parameter or by a specific media type indication in the message body.
  • a system for implementing voice continuity of a single access system including a UE,
  • the UE is configured to, after the SRVCC handover is completed, complete the SRVCC handover after the voice media of the session is in use, and continue to use the encryption mechanism to encrypt the voice data, and use the encrypted voice data stream as a voice data packet, and pass the circuit.
  • Exchange CS network transmissions
  • the UE is further configured to acquire the CS network to support the SRVCC media security capability after completing the SRVCC handover.
  • the UE is further configured to acquire the SRVCC media security capability of the remote node while acquiring the CS network supporting the SRVCC media security capability.
  • the encryption mechanism adopted by the UE is an SRTP-based encryption mechanism.
  • the system also includes an eMSC,
  • the eMSC is configured to encapsulate the voice encrypted media stream into an SRTP packet transmission when the UE receives the voice encrypted media stream transmitted by the UE through the CS network after the UE completes the SRVCC handover. Giving the remote node; and receiving the SRTP packet of the remote node, encapsulating the SRTP packet into a voice data packet, and transmitting the same to the UE through the CS network.
  • the eMSC is further configured to: before the voice encrypted media stream is encapsulated into an SRTP packet and sent to the remote node, determine whether the current voice media is a voice encrypted media stream, and determine that the current voice media is a voice encrypted media stream, and then the voice is The encrypted media stream is encapsulated and passed to the remote node as an SRTP packet.
  • the system also includes a remote node,
  • the remote node is configured to perform secondary decoding processing on the transmission control layer protocol in the data packet from the UE when determining that the UE has initiated the SRVCC handover, and the eMSC does not have the SRVCC media security capability, and When the UE transmits a voice data packet, the voice data packet is subjected to secondary encoding processing and then transmitted.
  • the secondary decoding process is: first parsing the RTP message, and further parsing according to the SRTP layer protocol to obtain a voice data packet.
  • the secondary encoding process is then sent as: encoding the voice data packet according to the SRTP protocol, generating the SRTP data packet, and continuing to encode the SRTP data packet according to the RTP protocol, and finally generating the RTP data packet and then the RTP.
  • the data packet is encapsulated at the UDP protocol layer and passed to the UE.
  • the UE adopts an encryption mechanism for the voice media of the session, and the UE continues to use the encryption mechanism to encrypt the voice data after the SRVCC handover is completed.
  • the encrypted voice data stream is transmitted as a voice data packet through the CS network. Therefore, in the embodiment of the present invention, if the UE originally uses the media security protection in the IMS network communication, after the UE is switched to the CS network, the secure session can still be performed, thereby improving the user communication security level.
  • FIG. 1 is a schematic diagram of a voice continuity system architecture of a conventional single access system
  • 2 is a schematic diagram of a KMS-based IMS media security solution architecture in the prior art
  • FIG. 3 is a basic flowchart of a KMS solution
  • FIG. 4 is a schematic diagram of an IMS media security protocol stack
  • FIG. 5 is a schematic flowchart of a UE initiating an SRVCC service
  • FIG. 6 is a schematic diagram of a prior art media plane protocol stack after the UE switches to the CS network;
  • FIG. 7 is a schematic diagram of the media plane protocol stack after the UE completes the SRVCC handover according to the first embodiment of the present invention;
  • FIG. 8 is a schematic flowchart of implementing voice continuity of a single access system according to Embodiment 1 of the present invention
  • FIG. 9 is a schematic diagram of a media plane protocol stack after a UE completes SRVCC handover according to Embodiment 2 of the present invention
  • FIG. 10 is a schematic flow chart of a method for implementing voice continuity of a single access system according to Embodiment 2 of the present invention. detailed description
  • the basic idea of the present invention is: the UE is using an encryption mechanism for the voice media of the session, and after the SRVCC handover is completed, the UE continues to encrypt the voice data by using the encryption mechanism, and uses the encrypted voice data stream as a voice data packet. , transmitted through the CS network.
  • the encryption mechanism is an encryption mechanism based on the SRTP protocol.
  • the method further includes: the UE acquiring the CS network to support the SRVCC media security capability, and then transmitting the encrypted voice data stream as a voice data packet through the CS network.
  • the handover command message in the EPS network and the message sent to the UE in the CS network during registration, session establishment, and the like may indicate that the CS network supports the SRVCC media security capability.
  • the UE may switch the handover command in the pre-EPS network.
  • the message acquisition CS network supports the SRVCC media security capability, and the CS network can also be used to support the SRVCC media security capability by using a message in the CS network after handover.
  • the remote node SRVCC media security capability is indicated, and the remote node SRVCC media security capability may be provided by the remote node during the session establishment process.
  • the key material negotiated by the same session before the SRVCC handover may be used for protection.
  • the regenerated key may also be used. The material is protected, including protection and integrity protection.
  • the eMSC receives the voice encrypted media stream transmitted by the UE (the handover UE) through the CS network, and encapsulates the voice encrypted media stream into an SRTP packet and transmits the packet to the remote end.
  • the node specifically, when the eMSC has the SRVCC media security capability, the SRTP packet is directly encapsulated on the UDP layer for transmission.
  • the SRTP packet is encoded according to the RTP protocol, and then the RTP data packet is encapsulated.
  • the eMSC receives the SRTP packet of the remote node, and encapsulates the SRTP packet into a voice data packet, which is transmitted to the UE through the CS network. Specifically, the eMSC uses the SRTP packet as the user voice. The data packet is encapsulated into the CS bearer protocol layer according to the prior art, and is transmitted as a user voice data packet.
  • the eMSC server includes the eMSC server and the eMGW
  • the eMSC Server needs to notify the eMGW when it is aware that the currently switched media is an encrypted media stream.
  • the eMSC can detect whether the currently switched voice media is a voice encrypted media stream: when the SRVCC is switched, the eMSC is currently notified by the service continuity server (SCC AS) in the EPS network or the IMS network. Whether the voice media stream is encrypted or not; the eMSC may also determine whether the current media is a voice encrypted media stream according to the media stream detection. The eMSC determines that the current voice media is a voice encrypted media stream, and then encapsulates the voice encrypted media stream as an SRTP packet and transmits the same to the remote node.
  • SCC AS service continuity server
  • the remote node in the media plane encryption mode, the remote node senses that the peer end (the UE) has initiated the SRVCC handover, and the eMSC does not have the SRVCC media security. Capabilities, when the remote node receives the data packet of the peer end, performing secondary decoding processing on the transmission control layer protocol in the data packet; meanwhile, the remote node sends voice data to the opposite end At the time of the packet, the voice data packet is subjected to a transmission control protocol secondary encoding process.
  • the remote node notifies that the peer end has initiated the SRVCC handover, and the remote node receives the re-session invitation message sent by the SCC AS or the update message carries the SRVCC indication, or the remote node receives the The type of encoding indicated in the RTP message to the peer changes.
  • the SRVCC indication may be indicated by a message header parameter or by a specific media type in the message body.
  • the remote node receives the peer data packet, and performs secondary decoding processing on the transmission control layer protocol of the data packet, where the remote node first parses the peer data packet.
  • the RTP message is further parsed according to the SRTP layer protocol to obtain a voice data packet.
  • the remote node When the eMSC does not have the SRVCC media security capability, the remote node performs a transmission control protocol secondary encoding process on the transmitted voice data packet, where the remote node encodes the voice data packet according to the SRTP protocol to generate an SRTP data packet. Then, the SRTP data packet is further encoded according to the RTP protocol, and finally the RTP data packet is generated, and then the RTP data packet is encapsulated in the UDP protocol layer and transmitted to the opposite end.
  • the present invention also correspondingly proposes a system for implementing voice continuity of a single access system, the system including a UE,
  • the UE is configured to, after the SRVCC handover is completed, complete the SRVCC handover after the voice media of the session is in use, and continue to use the encryption mechanism to encrypt the voice data, and use the encrypted voice data stream as a voice data packet, and pass the circuit.
  • Exchange CS network transmissions
  • the UE is further configured to acquire the CS network to support the SRVCC media security capability after completing the SRVCC handover.
  • the UE is further configured to acquire the media security capability of the remote node SRVCC while acquiring the CS network supporting the SRVCC media security capability.
  • the encryption mechanism adopted by the UE is an SRTP-based encryption mechanism.
  • the system also includes an eMSC,
  • the eMSC is configured to: when the UE completes the SRVCC handover, and receives the voice encrypted media stream that is transmitted by the UE through the CS network, the voice encrypted media stream is encapsulated into an SRTP packet and transmitted to the remote node; When the SRTP packet of the remote node is encapsulated, the SRTP packet is encapsulated into a voice data packet and transmitted to the UE through the CS network.
  • the eMSC is further configured to: before the voice encrypted media stream is encapsulated into an SRTP packet and sent to the remote node, determine whether the current voice media is a voice encrypted media stream, and determine that the current voice media is a voice encrypted media stream, and then the voice is The encrypted media stream is encapsulated and passed to the remote node as an SRTP packet.
  • the system also includes a remote node,
  • the remote node is configured to perform secondary decoding processing on the transmission control layer protocol in the data packet from the UE when determining that the UE has initiated the SRVCC handover, and the eMSC does not have the SRVCC media security capability, and When the UE transmits a voice data packet, the voice data packet is subjected to secondary encoding processing and then transmitted.
  • the secondary decoding process is: first parsing the RTP message, and further parsing according to the SRTP layer protocol to obtain a voice data packet.
  • the secondary encoding process is then sent as: encoding the voice data packet according to the SRTP protocol, generating the SRTP data packet, and continuing to encode the SRTP data packet according to the RTP protocol, and finally generating the RTP data packet and then the RTP.
  • the data packet is encapsulated at the UDP protocol layer and passed to the UE.
  • FIG. 7 is a schematic diagram of a media plane protocol stack after the UE completes the SRVCC handover in the first embodiment of the present invention. After the handover occurs, the terminal finds that if the switched voice media has been encrypted by using the SRTP protocol, the UE continues to use the session. The key material and protocol before switching protect the voice data.
  • the UE transmits the SRTP and the voice data layer as a traditional voice data packet on the CS bearer protocol layer. Since the eMSC does not have the SRVCC media security capability, the SRTP and Voice Data data are transparent as voice packets in the CS network. Transmission, when the eMSC receives the encrypted data packet, the eMSC transmits the encrypted data packet to the IP bearer network based on the RTP protocol; the remote node receives the encrypted data packet, performs secondary transmission control protocol analysis on the data packet, and parses the RTP basis first. Then, continue to parse the SRTP protocol, and finally obtain voice data.
  • FIG. 8 is a schematic flowchart of implementing voice continuity of a single access system according to Embodiment 1 of the present invention. As shown in FIG. 8, the process includes:
  • Step 801 The EPS network triggers the SRVCC handover according to the radio measurement report of the UE.
  • Step 803 The UE determines whether the switched media has adopted SRTP encryption. If yes, the UE acquires the network SRVCC capability.
  • the UE may obtain the network SRVCC capability by using the following methods:
  • the EPS access network adds an SRVCC capability indication to the CS network capability through the SRVCC handover command, or indicates the CS network capability when the UE is attached to the EPS network; or
  • the UE and the remote node acquire the session through the session negotiation during the IMS establishment session; or the UE switches to the target CS network, and monitors whether the CS network supports the SRVCC capability by monitoring the downlink media data packet.
  • Step 804 The UE switches the current media to the target network according to the target network access parameter in the SRVCC handover request.
  • Step 805 The eMSC returns a handover response message to the EPS network, and sends a session invitation message to the SCC AS in the IMS network.
  • Step 806 The SCC AS receives the eMSC session invitation message, and associates with the source call branch according to the UE user identifier in the message, and determines that the currently switched media stream is an encrypted medium.
  • Step 807 The SCC AS sends a re-session invitation message or an update message to the remote node according to the remote node address information saved by the source call branch, where the message carries an SRVCC indication, where the SRVCC indication is indicated by the message header parameter or passes through the message body. Indicate the specific media type;
  • Step 808 - Step 809 If the remote node supports the SRVCC security protection capability, return a 200 OK message, and the SCC AS returns a 200 OK message to the eMSC after receiving the message;
  • Step 810 The UE protects the voice media by using the session key information before the handover, and then uses the SRTP packet as the user data of the CS media plane bearer protocol, and transmits the data through the CS bearer network.
  • Step 811 The remote node receives the uplink data packet, parses the RTP layer message, and further parses the message according to the SRTP layer protocol, and performs decryption according to the session key information before the handover, and finally obtains the user data packet, and the remote node
  • the downlink data packet is sent, the user media stream is protected by the session key information before the handover, and is encapsulated into an SRTP packet, and then encapsulated according to the RTP protocol, and the downlink data is sent to the eMSC through the IP network, and the downlink data protocol stack is sent.
  • /SRTP/RTP/UDP/IP Such as /SRTP/RTP/UDP/IP.
  • the eMSC receives the downlink data packet to strip the data stack header of the RTP layer, and uses the Voice Data/SRTP as the voice data, and encapsulates the CS media plane bearer protocol to the UE;
  • Step 812 The UE acquires the SRTP packet through the CS network, decrypts the encrypted data by using the session key information before the handover, and finally obtains the user data.
  • Example 2 The UE acquires the SRTP packet through the CS network, decrypts the encrypted data by using the session key information before the handover, and finally obtains the user data.
  • This embodiment describes the case where the eMSC has the SRVCC media security capability and implements the voice continuity of the single access system.
  • FIG. 9 is a schematic diagram of a media plane protocol stack after a UE completes SRVCC handover according to Embodiment 2 of the present invention.
  • the encrypted media stream packet is directly encapsulated as an SRTP packet on the UDP layer for transmission.
  • the eMSC encapsulates the SRTP into a voice data packet and transmits the packet to the handover terminal through the CS bearer network.
  • FIG. 10 is a schematic flowchart of a method for implementing voice continuity of a single access system according to Embodiment 2 of the present invention. As shown in FIG. 10, the method includes:
  • Step 1001 The EPS network triggers the SRVCC handover according to the radio measurement report of the UE.
  • Step 1003 The UE determines whether the switched media has adopted SRTP encryption. If yes, the UE acquires the network SRVCC capability.
  • the UE may obtain the network SRVCC capability by using the following methods:
  • the EPS access network adds an SRVCC capability indication to the CS network capability through the SRVCC handover command, or indicates the CS network capability when the UE is attached to the EPS network; or
  • the UE and the remote node acquire the session through the session negotiation during the IMS establishment session; or the UE switches to the target CS network, and monitors whether the CS network supports the SRVCC capability by monitoring the downlink media data packet.
  • Step 1004 The UE switches the current media to the target network according to the target network access parameter in the SRVCC handover request.
  • Step 1005 The eMSC notifies the SCC AS to update the remote media, and notifies the EPS network that the handover is complete, and the step is the same as the steps 506-511;
  • Step 1006 The UE uses the session key information before the handover to protect the voice media, and then uses the SRTP packet as the user data of the CS media plane bearer protocol to be transmitted through the CS bearer network.
  • Step 1007 The eMSC determines whether the currently switched media is Encrypted media;
  • the manner in which the eMSC determines whether the currently switched media is the encrypted media may be: when the EPS network initiates the handover, the eMSC is notified; or, when the eMSC sends the session invitation message to the IMS network element SCC AS, the SCC AS informs; or, the eMSC is The uplink or downlink media data is detected for judgment. If the eMSC Server includes the eMSC Server and the eMGW in the 3G network, the eMSC Server detects the current switching medium as the encrypted medium, and the eMSC Server notifies the eMGWo.
  • Step 1008 The eMSC receives the uplink CS voice data, strips the CS media plane bearer protocol layer data header, and encapsulates the SRTP packet on the UDP as the UDP layer protocol user data, and sends the data to the remote node through the IP bearer network;
  • Step 1009 The eMSC receives the downlink encrypted data stream of the remote node, strips the data header under the UDP protocol layer, and encapsulates the SRTP packet on the CS media plane bearer protocol, as the CS media plane bearer protocol user data, through the CS network. Passed to the UE;
  • Step 1010 The UE acquires the SRTP packet through the CS network, decrypts the encrypted data by using the session key information before the handover, and finally obtains the user data.
  • the present invention provides a method of implementing a secure SRVCC session that provides a key negotiation mechanism that enables a secure session to be made after the SRVCC handover.

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Abstract

本发明公开了一种实现单接入系统语音连续性(SRVCC)的方法,包括:用户终端(UE)正在会话的语音媒体采用了加密机制,则所述UE在SRVCC切换完成后,继续使用所述加密机制对语音数据进行加密,并把加密语音数据流作为语音数据包,通过电路交换CS网络传输。本发明还相应地公开了一种实现单接入系统语音连续性的系统。通过本发明,如果UE原先在IMS网络通信采用媒体安全保护时,切换到CS网络后,仍然能够进行安全会话,从而能够提高用户通信安全等级。

Description

一种实现单接入系统语音连续性的方法及系统 技术领域
本发明涉及通信网络安全技术, 尤其涉及一种实现单接入系统语音连 续性( Single Radio Voice Call Continuity , SRVCC ) 的方法及系统。 背景技术
基于移动运营商对无线宽带高速率和低时延的需求, 第三代合作伙伴 计划 ( 3rd Generation Partnership Project, 3 GPP ) 的 R8版本标准中提出了 演进分组系统(Evolved Packet System, EPS ) 网络演进架构, 包括无线网 络和核心网络。 无线网络为演进的通用陆地无线接入网络 ( Evolved Universal Terrestrial Radio Access Network, E-UTRAN ),采用长期演进( Long Term Evolution, LTE )接入。 SRVCC方案主要解决 LTE网络部署网络电 话 ( Voice over Internet Protocol, VoIP )业务时存在的问题。
当 LTE在没有达到全网覆盖范围时,随着用户的移动,正在进行的 VoIP 业务会面临离开 LTE范围后的语音连续性的问题,这时,当用户签约 SRVCC 业务,服务 -呼叫会话控制功能( S-CSCF )根据用户在归属签约服务器( Home Subscriber Server, HSS ) 的签约, 将信令路径锚定在位于 IP多媒体子系统 ( IP Multimedia Subsystem, IMS ) 网络中的业务集中和连续性 (Service Centralization and Continuity, SCC )服务器上, SRVCC服务可以将语音切 换到电路交换( Circuit Switched, CS ), 从而保证语音通话不中断。
SRVCC 目前只有 LTE到 CS 的切换需求, 暂没有反向切换的需求。
SRVCC方案基于 IMS实现, 因此网络上需要部署 IMS, 且 EPS网络开通 VoIP业务后, 在特定场景需要使用 SRVCC。
图 1为现有单接入系统语音连续性系统架构示意图, 如图 1所示, 现 有单接入系统语音连续性系统中:
CS网络主要用于 3G/2G CS域网络接入;
增强移动交换中心( enhanced Mobile Switch Center, eMSC )用于帮助 UE完成 CS域和分组交换域( Packet Switch, PS )间语音连续性, 在 3G网 络中信令和承载分离, eMSC包含 eMSC服务器(eMSC Server )和增强媒 体网关( eMGW ), 其中 eMSC Server主要负责信令面控制, eMGW主要负 责媒体面接续;
EPS网络主要负责 LTE网络接入, 以及 LTE网络资源管理;
呼叫会话控制功能( CSCF ) 负责 IMS网络会话控制;
SCC AS用于保证 IMS会话业务连续性。
在 SRVCC 场景下, UE先通过 EPS接入到 IMS 网络, 与远端节点 ( Remote End )建立 SIP会话, CSCF把会话锚点到 SCC AS, 当 UE切换 到 CS网络, eMSC通知 SCC AS把原先通过 EPS网络的呼叫分支切换到与 eMSC相连。
在 3GPP TS33.328中, IMS媒体面安全有两个解决方案,一种叫做 SDES 方案, 另一种叫做密钥管理服务器 (KMS )方案, 下面对这两种方案作简 要介绍:
SDES 方案是基于成熟的 IETF 标准 RFC4568 "Session Description Protocol (SDP) Security Descriptions for Media Streams", 即在 SDP中扩展了 一个特殊的密钥属性, 用来直接在 SDP中传输所要使用的媒体密钥。 该方 案非常简单, 例如, 用户 A呼叫用户 B, 在发起呼叫的邀请(INVITE ) 消 息的 SDP密钥属性中携带了密钥 Kl , K1用于从用户 A发往用户 B的媒体 流加密, 用户 B收到该 K1后会将这个密钥保存下来, 用于解密从用户 A 收到的加密媒体流;呼叫建立成功, 用户 B在回给用户 A的 200OK消息中 的 SDP密钥属性携带密钥 K2, K2用于用户 Β发往用户 Α的媒体流的加密, 用户 A收到该消息后, 保存 K2, 用于解密从用户 B收到的加密媒体流。 图 2为现有技术中基于 KMS的 IMS媒体安全解决方案架构示意图, 其中, 用户 A和用户 B分别是媒体信息的发送方和接收方; KMS作为可信 任的第三方实现密钥的管理和分发功能;代理 -呼叫会话控制功能( P-CSCF ) 和服务 -呼叫会话控制功能(S-CSCF ) 为 IMS网络网元, 图 2中的其它网 元的功能介绍请参考 3GPP TS23.228。 图 2中每个用户设备均可与 KMS通 过 GBA方式建立信任关系,每个用户设备和 KMS建立共享密钥,如果 GBA 无法使用, 用户设备可以使用预共享密钥等其它方式和 KMS 获得共享密 钥。
图 3为 KMS方案基本流程图, 简述如下: 会话呼叫方首先向 KMS发 送 REQUEST— INIT消息请求包含相关密钥材料的票据(ticket ), 在此票据 中, 呼叫方向 KMS请求得到的相关密钥会被加密后包含在其中。 KMS向 呼叫方返回 REQUEST— RESP消息,该消息中包含主叫方所申请的票据。主 叫方得到包含相关密钥的票据后, 呼叫方在 TRANSFER— INIT消息中将票 据发给被叫方; 由于被叫方无法解密票据从而获得其中包含的信息, 被叫 方则将此票据在 RESOLVE— INIT消息中继续发送给 KMS, 由 KMS解密票 据并将其中的相关密钥在 RESOLVE— RESP消息中返回给被叫方,被叫方获 得相关密钥材料后, 向呼叫方发送 TRANSFER— RESP消息。
图 4为 IMS媒体安全协议栈示意图, 其中,
UE 1和 UE 2在 IMS网络中使用 SIP信令建立 IP承载连接, 通过媒体 加密层对用户语音数据进行加密, UE 1和 UE 2媒体加密层的加密 /解密所 需的密钥材料根据图 3流程获取, 具体的, 由 UE 1和 UE 2在呼叫建立过 程中通过 IMS媒体面安全方案协商获取;
语音数据层( Voice data )主要用于传递 UE 1和 UE 2之间的语音数据, 该层是端到端协议, 在 IP承载网络透传的; 安全实时传输协议 ( SRTP )主要用于对语音数据层进行加密;
IP层和 UDP层是 IP网络底层链路传输协议。
图 5为 UE发起 SRVCC业务的流程示意图, UE通过 EPS网络接入到 IMS网络, 并在 IMS网络与远端节点建立 SIP会话, 会话过程中, UE1从 LTE网络切换到传统的 CS网络, 如图 5所示, 该流程具体包括:
步驟 501. EPS网络发现 UE无线质量变差, 决定触发 SRVCC切换, 向 eMSC发送切换请求消息, 消息中携带 UE的身份标识和 SRVCC切换号 码 STN-SR;
步驟 502. eMSC通知 CS网络准备 CS无线和有线资源;
步驟 503. 准备完成后, 向 EPS 网络返回切换应答消息, 消息中携带 目的无线资源接入参数;
步驟 504. EPS网络向 UE发送切换请求消息;
步驟 505. UE根据切换请求消息中的目的无线资源接入参数, 接入到 目的 CS网络;
步驟 506-507. UE接入完成后, CS网络向 eMSC返回切换完成消息, eMSC同时把这个切换完成消息通知 EPS网络;
步驟 508. EPS收到切换完成消息后, 可以选择释放 UE原先接入的资 源;
步驟 509. eMSC向 EPS网络返回切换应答消息, 同时向 IMS网络中 的 SCC AS发送会话邀请消息 ( Invite );
步驟 510. SCC AS 根据用户身份标识关联到原先锚定的会话, 并向 Remote End发送重会话邀请( re-Invite )请求或者更新 ( Update )请求消息 把 Remote End中对端 IP地址信息由所述 UE的 IP地址信息改为 eMSC地 址信息;
步驟 511. SCC AS远端更新成功后, 向 eMSC返回 200 OK消息, 从 而使 UE和 Remote End之间的通信由原来的的 LTE网络切换到 CS网络。 图 6为当 UE切换到 CS网络后, 现有技术媒体面协议栈示意图, 根据 图 6, 在 CS网络, UE与 eMSC之间通过 CS承载传递语音数据包, eMSC 作为中继节点,把 CS承载转换为 IP承载, 然后通过 IP网络与 Remote End 进行语音通信。 其中, CS承载协议, 主要用于在 CS网络传递语音数据包 底层传输协议, 例如 AAL2协议、 STM或者 RTP协议等。
但是, 在现有的 CS网络中, 还没有端到端 CS媒体安全机制, 所以, 当用户终端 (UE )原先在 IMS网络通信采用媒体安全保护时, 切换到 CS 网络后, 由于 CS网络不支持端到端媒体面媒体安全保护, 会导致切换完成 后, 无法继续进行安全的通信, 从而使用户通信安全降级。 发明内容
有鉴于此, 本发明的主要目的在于提供一种实现单接入系统语音连续 性的方法及系统, 能够在 SRVCC场景下提供 CS网络端到端媒体安全保护 机制, 提高用户通信安全等级。
为达到上述目的, 本发明实施例的技术方案是这样实现的:
一种实现单接入系统语音连续性 SRVCC的方法, 包括:
用户终端 UE 正在会话的语音媒体采用了加密机制, 则所述 UE 在 SRVCC切换完成后, 继续使用所述加密机制对语音数据进行加密, 并把加 密语音数据流作为语音数据包, 通过电路交换 CS网络传输。
在所述 UE完成 SRVCC切换之后, 该方法还包括: 所述 UE获取 CS 网络支持 SRVCC媒体安全能力。
所述 UE获取 CS网络支持 SRVCC媒体安全能力为: 所述 UE通过切 换前 EPS网络中的切换命令消息获取 CS网络支持 SRVCC媒体安全能力, 或者, 通过切换后 CS网络中的消息获取所述 CS网络支持 SRVCC媒体安 全能力。 该方法还包括: 所述 UE获取 CS网络支持 SRVCC媒体安全能力的同 时, 获取远端节点 SRVCC媒体安全能力。
所述 UE在 SRVCC切换完成后, 继续使用切换前的加密机制对语音数 据进行加密时, 使用 SRVCC切换前同一会话协商出的密钥材料进行保护; 或使用重新生成的密钥材料进行保护。
所述加密机制为基于安全实时传输协议 SRTP的加密机制。
该方法还包括: 所述 UE完成 SRVCC切换后, eMSC收到所述 UE通 过 CS网络传输的语音加密媒体流时, 将所述语音加密媒体流封装为 SRTP 包传递给远端节点; eMSC收到远端节点的 SRTP包时, 把所述 SRTP包封 装为语音数据包, 通过 CS网络传递给所述 UE。
所述 eMSC将语音加密媒体流封装为 SRTP包传递给远端节点之前,该 方法还包括: 所述 eMSC判定当前的语音媒体为语音加密媒体流, 具体的, 在 SRVCC切换时, 由 EPS网络或者 IMS网络中的业务连续性服务器 SCC AS告知 eMSC当前的语音媒体是否为语音加密媒体流; 或者, 由 eMSC根 据媒体流检测判断当前的语音媒体是否为语音加密媒体流。
所述 eMSC将所述 UE通过 CS网络传输的语音加密媒体流封装为 SRTP 包传递给远端节点为: 所述 eMSC具备 SRVCC媒体安全能力时, 将所述 SRTP包直接封装在 UDP层上进行传递;所述 eMSC不具备 SRVCC媒体安 全能力时, 将所述 SRTP包按照 RTP协议进行编码, 再将生成的 RTP数据 包封装在 UDP层上进行传递;
所述 eMSC将来自远端节点的 SRTP包封装为语音数据包,通过 CS网 络传递给所述 UE为:所述 eMSC把所述 SRTP包作为用户语音数据包封装 到 CS承载协议层并进行传递。
该方法还包括: 所述远端节点判定所述 UE 已发起 SRVCC切换, 且 eMSC不具备 SRVCC媒体安全能力, 则所述远端节点收到所述 UE的数据 包时, 对所述数据包中的传输控制层协议进行二次解码处理, 以及在向所 述 UE发送语音数据包时, 对所述语音数据包进行二次编码处理后再发送。
所述二次解码处理为: 先解析 RTP消息, 再按照 SRTP层协议作进一 步解析, 获取语音数据包,
所述二次编码处理后再发送为: 对语音数据包按照 SRTP协议进行编 码, 生成 SRTP数据包, 再继续把所述 SRTP数据包按照 RTP协议进行编 码, 最终生成 RTP数据包之后把所述 RTP数据包封装在 UDP协议层, 传 递给所述 UE。
所述远端节点判定 UE已发起 SRVCC切换为:所述远端节点收到 SCC AS发送过来的重会话邀请消息或者更新消息中携带 SRVCC指示, 或者, 所述远端节点收到对端的 RTP报文中指示的编码类型发生改变,
在重会话邀请消息或者更新消息中, SRVCC指示通过消息头参数中指 示或通过消息体中特定媒体类型指示。
一种实现单接入系统语音连续性的系统, 包括 UE,
所述 UE, 设置为在正在会话的语音媒体采用了加密机制的情况下, 完 成 SRVCC切换后, 继续使用所述加密机制对语音数据进行加密, 并把加密 语音数据流作为语音数据包, 通过电路交换 CS网络传输。
所述 UE,还设置为在完成 SRVCC切换之后,获取 CS网络支持 SRVCC 媒体安全能力。
所述 UE, 还设置为在获取 CS网络支持 SRVCC媒体安全能力的同时, 获取远端节点 SRVCC媒体安全能力。
所述 UE采用的加密机制为基于 SRTP的加密机制。
该系统还包括 eMSC,
所述 eMSC, 设置为在 UE完成 SRVCC切换后, 收到所述 UE通过 CS 网络传输的语音加密媒体流时,将所述语音加密媒体流封装为 SRTP包传递 给远端节点; 以及在收到远端节点的 SRTP包时, 把所述 SRTP包封装为语 音数据包, 通过 CS网络传递给所述 UE。
所述 eMSC,还设置为在将语音加密媒体流封装为 SRTP包传递给远端 节点之前, 判断当前的语音媒体是否为语音加密媒体流, 判定当前的语音 媒体为语音加密媒体流,则将语音加密媒体流封装为 SRTP包传递给远端节 点。
该系统还包括远端节点,
所述远端节点, 设置为在判定 UE已发起 SRVCC切换, 且 eMSC不具 备 SRVCC媒体安全能力时, 对来自所述 UE的数据包中的传输控制层协议 进行二次解码处理, 以及在向所述 UE发送语音数据包时,对所述语音数据 包进行二次编码处理后再发送,
所述二次解码处理为: 先解析 RTP消息, 再按照 SRTP层协议作进一 步解析, 获取语音数据包,
所述二次编码处理后再发送为: 对语音数据包按照 SRTP协议进行编 码, 生成 SRTP数据包, 再继续把所述 SRTP数据包按照 RTP协议进行编 码, 最终生成 RTP数据包之后把所述 RTP数据包封装在 UDP协议层, 传 递给所述 UE。
本发明实施例实现单接入系统语音连续性的方法及系统, UE正在会话 的语音媒体采用了加密机制, 则所述 UE在 SRVCC切换完成后, 继续使用 所述加密机制对语音数据进行加密, 并把加密语音数据流作为语音数据包, 通过 CS网络传输。 所以, 通过本发明实施例, 如果 UE原先在 IMS网络通 信采用媒体安全保护时, 切换到 CS网络后, 仍然能够进行安全会话, 从而 能够提高用户通信安全等级。 附图说明
图 1为现有单接入系统语音连续性系统架构示意图; 图 2为现有技术中基于 KMS的 IMS媒体安全解决方案架构示意图; 图 3为 KMS方案基本流程图;
图 4为 IMS媒体安全协议栈示意图;
图 5为 UE发起 SRVCC业务的流程示意图;
图 6为当 UE切换到 CS网络后, 现有技术媒体面协议栈示意图; 图 7为本发明实施例 1中, UE完成 SRVCC切换后的媒体面协议栈示 意图;
图 8为本发明实施例 1实现单接入系统语音连续性的流程示意图; 图 9为本发明实施例 2中 UE完成 SRVCC切换后的媒体面协议栈示意 图;
图 10 为本发明实施例 2 实现单接入系统语音连续性的方法流程示意 图。 具体实施方式
本发明的基本思想是: UE正在会话的语音媒体采用了加密机制, 则所 述 UE在 SRVCC切换完成后,继续使用所述加密机制对语音数据进行加密, 并把加密语音数据流作为语音数据包, 通过 CS网络传输。
优选的, 所述加密机制为基于 SRTP协议的加密机制。
优选的, UE在 SRVCC切换完成之后, 该方法还包括: 所述 UE获取 CS网络支持 SRVCC媒体安全能力, 之后, 才把加密语音数据流作为语音 数据包, 通过 CS网络传输。 这里, EPS网络中的切换命令消息以及注册、 会话建立等过程中 CS 网络中发送给 UE的消息中均可指示 CS 网络支持 SRVCC媒体安全能力, 换言之, UE可以通过切换前 EPS网络中的切换命 令消息获取 CS网络支持 SRVCC媒体安全能力, 也可以通过切换后 CS网 络中的消息获取所述 CS网络支持 SRVCC媒体安全能力。 可选的, 在 EPS 网络或 CS网络中指示 CS网络支持 SRVCC媒体安全能力的同时, 还可以 指示远端节点 SRVCC媒体安全能力, 所述远端节点 SRVCC媒体安全能力 可以在会话建立过程中由远端节点提供。
优选的, UE使用 SRVCC切换前的加密机制对语音数据进行加密时, 可以使用 SRVCC 切换前同一会话协商出的密钥材料进行保护, 可选的, SRVCC切换后, 也可以使用重新生成的密钥材料进行保护, 所述保护包括 加密和完整性保护。
对于 eMSC, 需要说明的是, SRVCC切换后, eMSC收到所述 UE (发 生切换的 UE )通过 CS网络传输的语音加密媒体流时, 将所述语音加密媒 体流封装为 SRTP包传递给远端节点, 具体的, eMSC具备 SRVCC媒体安 全能力时,将 SRTP包直接封装在 UDP层上进行传递, eMSC不具备 SRVCC 媒体安全能力时, 将 SRTP包按照 RTP协议进行编码, 再将生成 RTP数据 包封装在 UDP层上进行传递; eMSC收到远端节点的 SRTP包时, 把所述 SRTP包封装为语音数据包, 通过 CS网络传递给所述 UE, 具体的, eMSC 把所述 SRTP包作为用户语音数据包按照现有技术封装到 CS承载协议层, 作为用户语音数据包进行传递。 这里, 如果 eMSC包含分设的 eMSC Server 和 eMGW,则所述 eMSC Server感知到当前被切换的媒体为加密媒体流时, 需要通知所述 eMGW。
优选的, eMSC感知到当前被切换的语音媒体是否为语音加密媒体流可 以为: 在 SRVCC切换时, 由 EPS网络或者 IMS网络中的业务连续性服务 器(SCC AS )告知 eMSC当前被切换的语音媒体是否为语音加密媒体流; 也可以由 eMSC根据媒体流检测判断当前的媒体是否为语音加密媒体流。 eMSC判定当前的语音媒体为语音加密媒体流,再将所述语音加密媒体流封 装为 SRTP包传递给远端节点
对于远端节点, 需要说明的是, 在媒体面加密模式下, 远端节点感知 对端 (所述 UE ) 已发起 SRVCC切换, 且 eMSC不具备 SRVCC媒体安全 能力, 则所述远端节点收到所述对端的数据包时, 对所述数据包中的传输 控制层协议进行二次解码处理; 同时, 所述远端节点向所述对端发送语音 数据包时, 对所述语音数据包进行传输控制协议二次编码处理。
优选的, 所述远端节点感知对端已发起 SRVCC切换可以为: 所述远端 节点收到 SCC AS发送过来的重会话邀请消息或者更新消息中携带 SRVCC 指示, 或者, 所述远端节点收到对端的 RTP报文中指示的编码类型发生改 变。在 re-Invite或 Update消息中, SRVCC指示具体可以通过消息头参数中 指示, 也可以通过消息体中特定媒体类型指示。
优选的, 所述远端节点收到所述对端数据包, 对所述数据包的传输控 制层协议进行二次解码处理具体为: 所述远端节点收到对端数据包后, 先 解析 RTP消息, 再按照 SRTP层协议作进一步解析, 获取语音数据包。
eMSC不具备 SRVCC媒体安全能力时, 所述远端节点对发送的语音数 据包进行传输控制协议二次编码处理具体为: 所述远端节点对语音数据包 按照 SRTP协议进行编码, 生成 SRTP数据包, 再继续把所述 SRTP数据包 按照 RTP协议进行编码, 最终生成 RTP数据包,之后把所述 RTP数据包封 装在 UDP协议层, 传递给所述对端。
本发明还相应地提出一种实现单接入系统语音连续性的系统, 系统包 括 UE,
所述 UE, 设置为在正在会话的语音媒体采用了加密机制的情况下, 完 成 SRVCC切换后, 继续使用所述加密机制对语音数据进行加密, 并把加密 语音数据流作为语音数据包, 通过电路交换 CS网络传输。
所述 UE,还设置为在完成 SRVCC切换之后,获取 CS网络支持 SRVCC 媒体安全能力。
所述 UE, 还设置为在获取 CS网络支持 SRVCC媒体安全能力的同时, 获取远端节点 SRVCC媒体安全能力。 所述 UE采用的加密机制为基于 SRTP的加密机制。
该系统还包括 eMSC,
所述 eMSC, 设置为在 UE完成 SRVCC切换后, 收到所述 UE通过 CS 网络传输的语音加密媒体流时,将所述语音加密媒体流封装为 SRTP包传递 给远端节点; 以及在收到远端节点的 SRTP包时, 把所述 SRTP包封装为语 音数据包, 通过 CS网络传递给所述 UE。
所述 eMSC,还设置为在将语音加密媒体流封装为 SRTP包传递给远端 节点之前, 判断当前的语音媒体是否为语音加密媒体流, 判定当前的语音 媒体为语音加密媒体流,则将语音加密媒体流封装为 SRTP包传递给远端节 点。
该系统还包括远端节点,
所述远端节点, 设置为在判定 UE已发起 SRVCC切换, 且 eMSC不具 备 SRVCC媒体安全能力时, 对来自所述 UE的数据包中的传输控制层协议 进行二次解码处理, 以及在向所述 UE发送语音数据包时,对所述语音数据 包进行二次编码处理后再发送,
所述二次解码处理为: 先解析 RTP消息, 再按照 SRTP层协议作进一 步解析, 获取语音数据包,
所述二次编码处理后再发送为: 对语音数据包按照 SRTP协议进行编 码, 生成 SRTP数据包, 再继续把所述 SRTP数据包按照 RTP协议进行编 码, 最终生成 RTP数据包之后把所述 RTP数据包封装在 UDP协议层, 传 递给所述 UE。
下面通过具体实施方式对本发明的技术方案作进一步详细说明。
实施例 1
本实施例描述 eMSC不具备 SRVCC媒体安全能力时,实现单接入系统 语音连续性的情况。 图 7为本发明实施例 1中, UE完成 SRVCC切换后的媒体面协议栈示 意图,终端发生切换后,发现如果切换前,被切换的语音媒体已经采用 SRTP 协议进行加密,则 UE继续使用该会话切换前的密钥材料和协议对语音数据 进行保护。
在本实施例中, UE把 SRTP和语音数据层当成传统语音数据包在 CS 承载协议层上进行传输, 由于 eMSC不具备 SRVCC媒体安全能力, SRTP 和 Voice Data数据在 CS网络作为语音数据包进行透明传输, eMSC在收到 终端加密数据包时, 使加密数据包基于 RTP协议向 IP承载网络传输; 远端 节点收到加密数据包,对数据包进行二次传输控制协议解析,在先解析 RTP 基础上, 再继续解析 SRTP协议, 最终获取语音数据。
图 8为本发明实施例 1 实现单接入系统语音连续性的流程示意图, 如 图 8所示, 该流程包括:
步驟 801 : EPS网络根据 UE的无线测量报告, 触发 SRVCC切换; 步驟 802、 UE收到 EPS网络的 SRVCC切换命令;
步驟 803、 UE判断所切换的媒体是否已经采用了 SRTP加密, 如果是, UE获取网络 SRVCC能力。
这里, UE获取网络 SRVCC能力可以通过以下几个途径: EPS接入网 络通过 SRVCC切换命令中添加 SRVCC能力指示, 指示 CS网络能力; 或 者, UE在附着到 EPS网络时, EPS网络告知 UE; 或者, UE与远端节点 在 IMS建立会话过程中,通过会话协商获取;或者 UE切换到目标 CS网络, 通过对下行媒体数据包进行监测判断 CS网络是否支持 SRVCC能力。
步驟 804、 UE根据 SRVCC切换请求中目标网络接入参数, 把当前媒 体切换到目标网络;
步驟 805: eMSC向 EPS网络返回切换应答消息, 同时向 IMS网络中 的 SCC AS发送会话邀请消息; 步驟 806、 SCC AS收到 eMSC会话邀请消息, 根据消息中 UE用户标 识, 关联到源呼叫分支, 判断当前所切换的媒体流为加密媒体;
步驟 807、 SCC AS根据源呼叫分支保存的远端节点地址信息, 向远端 节点发送重新会话邀请消息或者更新消息, 消息中携带 SRVCC指示, 其中 SRVCC指示通过消息头参数进行指示或者通过消息体中特定媒体类型进行 指示;
步驟 808-步驟 809、如果远端节点支持 SRVCC安全保护能力,返回 200 OK消息, SCC AS收到后向 eMSC返回 200 OK消息;
步驟 810: UE使用切换前的会话密钥信息对语音媒体进行保护, 然后 将该 SRTP包作为 CS媒体面承载协议的用户数据,通过 CS承载网络传递。
步驟 811、 远端节点收到上行数据包, 先解析 RTP层消息, 然后按照 SRTP层协议对消息进行进一步解析, 并根据切换前的会话密钥信息进行解 密, 最终获取用户数据包, 远端节点发送下行数据包时, 采用切换前的会 话密钥信息对用户媒体流进行保护, 封装为 SRTP 包, 再此基础上再按照 RTP协议进行封装, 通过 IP网络向 eMSC发送下行数据, 下行数据协议栈 如 /SRTP/RTP/UDP/IP。 其中 eMSC收到下行数据包把剥离 RTP层以下协议 栈数据头, 把 Voice Data/SRTP当做语音数据, 封装在 CS媒体面承载协议 传递给 UE;
步驟 812: UE通过 CS网络获取 SRTP包,采用切换前的会话密钥信息, 对加密数据进行解密, 最终获得用户数据。 实施例 2
本实施例描述 eMSC具备 SRVCC媒体安全能力时,实现单接入系统语 音连续性的情况。
图 9为本发明实施例 2中 UE完成 SRVCC切换后的媒体面协议栈示意 图, 与图 7相比, SRVCC切换后, eMSC收到切换终端的加密媒体流时, 直接把加密媒体流包作为 SRTP包封装在 UDP层上进行传递, 同时收到远 端节点的 SRTP包时, eMSC把 SRTP封装为语音数据包, 通过 CS承载网 络传递给切换终端。
图 10 为本发明实施例 2 实现单接入系统语音连续性的方法流程示意 图, 如图 10所示, 该方法包括:
步驟 1001、 EPS网络根据 UE的无线测量报告, 触发 SRVCC切换; 步驟 1002、 UE收到 EPS网络的 SRVCC切换命令;
步驟 1003、 UE判断所切换的媒体是否已经采用了 SRTP加密,如果是, UE获取网络 SRVCC能力。
这里, UE获取网络 SRVCC能力可以通过以下几个途径: EPS接入网 络通过 SRVCC切换命令中添加 SRVCC能力指示, 指示 CS网络能力; 或 者, UE在附着到 EPS网络时, EPS网络告知 UE; 或者, UE与远端节点 在 IMS建立会话过程中,通过会话协商获取;或者 UE切换到目标 CS网络, 通过对下行媒体数据包进行监测判断 CS网络是否支持 SRVCC能力。
步驟 1004、 UE根据 SRVCC切换请求中目标网络接入参数, 把当前媒 体切换到目标网络;
步驟 1005、 eMSC通知 SCC AS更新远端媒体, 并通知 EPS网络切换 完成, 该步驟同步驟 506-511 ;
步驟 1006、 UE使用切换前的会话密钥信息对语音媒体进行保护, 然后 将该 SRTP包作为 CS媒体面承载协议的用户数据,通过 CS承载网络传递; 步驟 1007、 eMSC判断当前切换的媒体是否为加密媒体;
这里, eMSC判断当前切换的媒体是否为加密媒体的方式可以为: EPS 网络发起切换时候, 告知 eMSC; 或者, eMSC向 IMS网元 SCC AS发送会 话邀请消息时, 由 SCC AS告知; 或者, eMSC根据检测上行或者下行媒体 数据进行判断。 如果在 3G网络, 承载和信令分离, eMSC包含 eMSC Server和 eMGW 时, 当 eMSC Server检测到当前切换媒体为加密媒体, 由 eMSC Server通知 eMGWo
步驟 1008、 eMSC收到上行 CS语音数据, 剥离 CS媒体面承载协议层 数据头, 把 SRTP包封装在 UDP之上, 作为 UDP层协议用户数据, 通过 IP承载网络发送给远端节点;
步驟 1009、 eMSC收到远端节点的下行加密数据流, 剥离 UDP协议层 之下的数据头, 把 SRTP包封装在 CS媒体面承载协议之上, 作为 CS媒体 面承载协议用户数据, 通过 CS网络传递给 UE;
步驟 1010、 UE通过 CS网络获取 SRTP包, 采用切换前的会话密钥信 息, 对加密数据进行解密, 最终获得用户数据。
可以看出本发明提供了实现安全 SRVCC会话的方法,能够提供密钥协 商机制, 使其能够在 SRVCC切换后仍然能够进行安全会话。
以上所述, 仅为本发明的较佳实施例而已, 并非用于限定本发明的保 护范围, 凡在本发明的精神和原则之内所作的任何修改、 等同替换和改进 等, 均应包含在本发明的保护范围之内。

Claims

权利要求书
1、 一种实现单接入系统语音连续性 SRVCC的方法, 其中, 该方法包 括:
用户终端 UE 正在会话的语音媒体采用了加密机制, 则所述 UE 在 SRVCC切换完成后, 继续使用所述加密机制对语音数据进行加密, 并把加 密语音数据流作为语音数据包, 通过电路交换 CS网络传输。
2、 根据权利要求 1所述的方法, 其中, 在所述 UE完成 SRVCC切换 之后, 该方法还包括: 所述 UE获取 CS网络支持 SRVCC媒体安全能力。
3、根据权利要求 2所述的方法,其中,所述 UE获取 CS网络支持 SRVCC 媒体安全能力为: 所述 UE通过切换前 EPS网络中的切换命令消息获取 CS 网络支持 SRVCC媒体安全能力, 或者, 通过切换后 CS网络中的消息获取 所述 CS网络支持 SRVCC媒体安全能力。
4、 根据权利要求 3所述的方法, 其中, 该方法还包括: 所述 UE获取 CS网络支持 SRVCC媒体安全能力的同时,获取远端节点 SRVCC媒体安全 能力。
5、 根据权利要求 1所述的方法, 其中, 所述 UE在 SRVCC切换完成 后, 继续使用切换前的加密机制对语音数据进行加密时, 使用 SRVCC切换 前同一会话协商出的密钥材料进行保护; 或使用重新生成的密钥材料进行 保护。
6、 根据权利要求 1至 5任一项所述的方法, 其中, 所述加密机制为基 于安全实时传输协议 SRTP的加密机制。
7、 根据权利要求 6所述的方法, 其中, 该方法还包括: 所述 UE完成 SRVCC切换后, eMSC收到所述 UE通过 CS网络传输的语音加密媒体流时, 将所述语音加密媒体流封装为 SRTP包传递给远端节点; eMSC收到远端节 点的 SRTP包时, 把所述 SRTP包封装为语音数据包, 通过 CS网络传递给 所述 UE。
8、 根据权利要求 7所述的方法, 其中, 所述 eMSC将语音加密媒体流 封装为 SRTP包传递给远端节点之前,该方法还包括: 所述 eMSC判定当前 的语音媒体为语音加密媒体流, 具体的, 在 SRVCC切换时, 由 EPS网络 或者 IMS网络中的业务连续性服务器 SCC AS告知 eMSC当前的语音媒体 是否为语音加密媒体流; 或者, 由 eMSC根据媒体流检测判断当前的语音 媒体是否为语音加密媒体流。
9、 根据权利要求 7所述的方法, 其中,
所述 eMSC将所述 UE通过 CS网络传输的语音加密媒体流封装为 SRTP 包传递给远端节点为: 所述 eMSC具备 SRVCC媒体安全能力时, 将所述 SRTP包直接封装在 UDP层上进行传递;所述 eMSC不具备 SRVCC媒体安 全能力时, 将所述 SRTP包按照 RTP协议进行编码, 再将生成的 RTP数据 包封装在 UDP层上进行传递;
所述 eMSC将来自远端节点的 SRTP包封装为语音数据包,通过 CS网 络传递给所述 UE为:所述 eMSC把所述 SRTP包作为用户语音数据包封装 到 CS承载协议层并进行传递。
10、 根据权利要求 9所述的方法, 其中, 该方法还包括: 所述远端节 点判定所述 UE已发起 SRVCC切换, 且 eMSC不具备 SRVCC媒体安全能 力,则所述远端节点收到所述 UE的数据包时,对所述数据包中的传输控制 层协议进行二次解码处理, 以及在向所述 UE发送语音数据包时,对所述语 音数据包进行二次编码处理后再发送。
11、 根据权利要求 10所述的方法, 其中,
所述二次解码处理为: 先解析 RTP消息, 再按照 SRTP层协议作进一 步解析, 获取语音数据包,
所述二次编码处理后再发送为: 对语音数据包按照 SRTP协议进行编 码, 生成 SRTP数据包, 再继续把所述 SRTP数据包按照 RTP协议进行编 码, 最终生成 RTP数据包之后把所述 RTP数据包封装在 UDP协议层, 传 递给所述 UE。
12、 根据权利要求 10所述的方法, 其中, 所述远端节点判定 UE已发 起 SRVCC切换为: 所述远端节点收到 SCC AS发送过来的重会话邀请消息 或者更新消息中携带 SRVCC指示, 或者, 所述远端节点收到对端的 RTP 报文中指示的编码类型发生改变,
在重会话邀请消息或者更新消息中, SRVCC指示通过消息头参数中指 示或通过消息体中特定媒体类型指示。
13、 一种实现单接入系统语音连续性的系统, 其中, 该系统包括 UE, 所述 UE, 设置为在正在会话的语音媒体采用了加密机制的情况下, 完 成 SRVCC切换后, 继续使用所述加密机制对语音数据进行加密, 并把加密 语音数据流作为语音数据包, 通过电路交换 CS网络传输。
14、 根据权利要求 13所述的系统, 其中,
所述 UE,还设置为在完成 SRVCC切换之后,获取 CS网络支持 SRVCC 媒体安全能力。
15、 根据权利要求 14所述的系统, 其中,
所述 UE, 还设置为在获取 CS网络支持 SRVCC媒体安全能力的同时, 获取远端节点 SRVCC媒体安全能力。
16、根据权利要求 13至 15任一项所述的系统, 其中, 所述 UE采用的 加密机制为基于 SRTP的加密机制。
17、 根据权利要求 16所述的系统, 其中, 该系统还包括 eMSC, 所述 eMSC, 设置为在 UE完成 SRVCC切换后, 收到所述 UE通过 CS 网络传输的语音加密媒体流时,将所述语音加密媒体流封装为 SRTP包传递 给远端节点; 以及在收到远端节点的 SRTP包时, 把所述 SRTP包封装为语 音数据包, 通过 CS网络传递给所述 UE。
18、 根据权利要求 17所述的系统, 其中,
所述 eMSC,还设置为在将语音加密媒体流封装为 SRTP包传递给远端 节点之前, 判断当前的语音媒体是否为语音加密媒体流, 判定当前的语音 媒体为语音加密媒体流,则将语音加密媒体流封装为 SRTP包传递给远端节 点。
19、 根据权利要求 17所述的系统, 其中, 该系统还包括远端节点, 所述远端节点, 设置为在判定 UE已发起 SRVCC切换, 且 eMSC不具 备 SRVCC媒体安全能力时, 对来自所述 UE的数据包中的传输控制层协议 进行二次解码处理, 以及在向所述 UE发送语音数据包时,对所述语音数据 包进行二次编码处理后再发送,
所述二次解码处理为: 先解析 RTP消息, 再按照 SRTP层协议作进一 步解析, 获取语音数据包,
所述二次编码处理后再发送为: 对语音数据包按照 SRTP协议进行编 码, 生成 SRTP数据包, 再继续把所述 SRTP数据包按照 RTP协议进行编 码, 最终生成 RTP数据包之后把所述 RTP数据包封装在 UDP协议层, 传 递给所述 UE。
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