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WO2004039097A1 - Procede de communication permettant d'appeler sur le domaine a commutation de circuits de reseaux centraux du gsm/wcdma - Google Patents

Procede de communication permettant d'appeler sur le domaine a commutation de circuits de reseaux centraux du gsm/wcdma Download PDF

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Publication number
WO2004039097A1
WO2004039097A1 PCT/CN2003/000784 CN0300784W WO2004039097A1 WO 2004039097 A1 WO2004039097 A1 WO 2004039097A1 CN 0300784 W CN0300784 W CN 0300784W WO 2004039097 A1 WO2004039097 A1 WO 2004039097A1
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WO
WIPO (PCT)
Prior art keywords
message
sip
server
call
msc server
Prior art date
Application number
PCT/CN2003/000784
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English (en)
Chinese (zh)
Inventor
Jie Wang
Jiongjiong Gu
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
Priority to AU2003271017A priority Critical patent/AU2003271017A1/en
Publication of WO2004039097A1 publication Critical patent/WO2004039097A1/fr

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W76/00—Connection management
    • 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
    • H04L65/102—Gateways
    • H04L65/1043—Gateway controllers, e.g. media gateway control protocol [MGCP] controllers
    • 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
    • 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
    • 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
    • H04L65/1101—Session protocols
    • H04L65/1104—Session initiation protocol [SIP]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W80/00—Wireless network protocols or protocol adaptations to wireless operation
    • 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 communication technologies, and in particular, to a communication method for making a call in a GSM / WCDMA core network circuit domain.
  • the WCDMA core network is mainly divided into the CS domain and the packet PS domain.
  • the CS domain provides voice, fax, and circuit data services
  • the PS domain provides packet data services.
  • the WCDMA standard currently has three versions, R99, R4, and R5, and subsequent versions are under development.
  • the CS domain consists of entities such as MSC, GMSC, and HLR.
  • the network architecture is basically the same as that of the GSM network.
  • the TMSC bears the call signaling and TUP / ISUP between MSCs.
  • Figure 1 depicts the existing 3GPP R4BICSCN network architecture.
  • the R4 specification proposes the FICS concept of the BICSCN (Bearer Independent Circuit Swi tching Core Network), that is, the CS domain that has nothing to do with ⁇ . It is mainly composed of MSC Server (MSC Server), GMSC Server (GMSC Server), MGW, HLR, etc. It consists of entities.
  • MSC Server MSC Server
  • GMSC Server GMSC Server
  • MGW Mobility Management Entity
  • HLR Home Location Register
  • 3GPP R4 selects the BICC (Bearer Independent Call Control) protocol developed by ITU-T as the (G) call signaling between MSC servers.
  • the BICC was developed on the basis of ISUP and inherited most of the traditional signaling design. Thought, support TDM / ATM (AAL1, AAL2) / IP various bearers of MGW. Taking into account the R4 CS domain network
  • the network architecture is consistent with the current fixed network NGN and 3GPP2 (the standardization organization of cdma2000) LMSD (Legency MS Domain, traditional mobile terminal domain) network architecture, that is, the server and MGW structure are used to separate control and bearer.
  • SIP Session Initiation Protocol
  • Session Initiation Protocol is an IP telephone signaling protocol proposed by the IETF (Interne Engineering Task Force). As its name implies, SIP is used to initiate sessions.
  • It can control the establishment and termination of multimedia sessions in which multiple participants participate, and it can dynamically adjust and modify session attributes such as session bandwidth requirements, the type of media being transmitted (voice , Video and data, etc.), media codec formats, support for multicast and unicast, etc.
  • SIP is designed with full consideration of extended adaptability to other protocols. It supports many kinds of address description and addressing, including: user name @host address, called number @? 3, gateway address, and description of ordinary E. 164 phone number such as Tel: 010-62281234. In this way, the SIP caller can identify whether the called party is on the traditional telephone network according to the called address, and then initiate and establish a call to the called party through a gateway connected to the traditional telephone network.
  • SIP itself contains the function of registering with the registration server, and it can also use other positioning servers such as DNS, LDAP and other positioning servers to enhance its positioning function.
  • SIP-T is an application of SIP.
  • the payload part of SIP encapsulates the ISUP information unit or translates the ISUP information unit.
  • the SIP protocol call control process is used to simplify the ISUP control signaling. Because different standardization organizations adopt different protocols, it will bring difficulties in interconnection and interoperability between different networks in the future ALL IP phase.
  • the protocol stack for IP bearer BICC signaling defined by 3GPP is BICC / STCP / IP / L2 / L1.
  • the STCP link is equivalent to a traditional SS7 link and is generally a static link.
  • BICC signaling and TUP and ISUP signaling use the same segment-by-segment routing instead of end-to-end routing.
  • the control layer of the wireless core network maintains a hierarchical structure, that is, the TMSC server is required to complete the long-distance routing function. This routing method and network structure in the IP network should be more optimized.
  • An object of the present invention is to provide a communication method that can implement calling in a GSM / WCDMA communication system, and is used to overcome the insufficiency of interconnection and interworking among different systems.
  • the object of the present invention is achieved by the following method, a communication method for realizing a call in a GSM / WCDMA communication system
  • the GSM communication system includes (G) MSC, BSC, HLR, VLR, UE, the WCDMA
  • the communication system includes (G) MSC server, MGW, HLR, VLR, RNC, UE, and the method includes steps: applying SIP / SIP-T protocol between (G) MSC or (G) MSC server to establish call signaling and perform each other Communication.
  • the method further includes carrying the IP network between the MGWs, and the (G) MSC server controls the MGW to perform message processing by using the H.248 protocol.
  • the specific method includes at least one of the following calling procedures:
  • Calling call called call; release of call initiated by UE; release of call initiated by network side.
  • the method further includes establishing a UE-to-UE call procedure in the mobile network.
  • the calling process includes the following steps:
  • the MSC server responds to the request and constructs a SIP-T INVITE message template, and sends the INVITE message to the next hop address.
  • the GMSC server receives the INVITE message from the MSC server and passes the INFO message back to the MSC server to establish the terminal on MG1.
  • Bearer plane service media flow between terminal T3 on T2 and MG2;
  • the MSC server returns a SIP-T AC message to the GMSC server; the end-to-end bearer of the calling process is set up.
  • the step of the calling process further includes modifying the ISUP IAM message template by the address message in the SIP-T message header of the GMSC server, selecting the office route and the idle circuit, and sending the final IAM message to SIGTRN / SG or SS7.
  • PSTN PSTN
  • the GMSC server receives the full ACM address message from the PSTN, encapsulates the message into the 180 ringing indication message of S IP-T, and sends the 180 message back to the MSC server;
  • the MSC server After receiving the message, the MSC server parses the encapsulated ACM information from the 180 message to notify the calling UE that the called user has been connected;
  • the GMSC server receives the ANM response message from the PSTN, encapsulates the message into the 200K instruction message of SIP-T, and sends the 200K message to the MSC server in the reverse direction;
  • the MSC server After receiving the message, the MSC server parses the encapsulated ANM information from the 200 message to notify the calling UE that the called user has answered.
  • the calling process further includes a step in which the MSC server internally transforms the call-related domain in the calling Setup call setup message from the Iu interface into an IAM message of ISUP.
  • the calling process further includes the step of, after the GMSC server receives the INVITE message from the MSC server and finds that it contains an ISUP IAM encapsulation, and directly uses the message content as a template for the ISUP IAM message to be sent to the PSTN. .
  • the step of constructing an INVITE message template includes filling a SIP address field according to the called number, and filling an SDP message of the MG1 relay-side terminal T1 into the SIP INVITE message template.
  • the called process includes the following steps:
  • the GMSC server receives an ISUP IAM message from the PSTN;
  • the GMSC server constructs a SIP-T INVITE message template, and sends the S IP-T INVITE message forward with the roaming number as the address information;
  • the MSC server sends the SIP-T INFO message back to the GMSC server to establish a bearer plane service media stream between different MG2 terminal T2 and MG1 terminal T3.
  • the GMSC server returns a SIP-T ACK message to the MSC server; the end-to-end bearer establishment of the called process is completed.
  • the called process further includes the following steps:
  • the MSC server receives the Alerting ringing message from the Iu interface, constructs the corresponding ISUP ACM message, encapsulates the message into the 180 ringing indication message of SIP-T, and sends the 18G message to the GMSC server in the reverse direction;
  • the GMSC server After receiving the SIP-T 180 message from the MSC server, the GMSC server parses the encapsulated ACM information from the message, and uses this message as a template to make necessary changes to the 180 header information and passes the final ACM message through SIGTRAN / SG or SS7 sent to PSTN;
  • the MSC server receives a Connect response message from the Iu interface, constructs the ANM message of ISUP according to the message, and encapsulates the message into the 200K instruction message of SIP-T, and sends the 200K message back to the GMSC server. ;
  • the GMSC server After receiving the SIP-T 200 message from the MSC server, the GMSC server parses the encapsulated A picture information from the message, uses this message as a template, and makes necessary modifications according to the 200 message header information, and passes the final Alice message.
  • SIGTRAN / SG or SS7 is sent to PSTN.
  • the GMSC server controls the MG2 to allocate ISUP circuit terminal T1 and context C1 resources through H.248 / MGCP commands, analyzes the number attributes, and initiates a routing query to the HLR.
  • the constructed SIP-T INVITE message template includes the SDP message of the terminal T2, and the incoming IAM message is encapsulated into the message body part.
  • the call release process initiated by the UE includes the following steps:
  • the MSC server receives a disconnection (Di sconnec t) command from the UE;
  • the MSC server constructs a SIP-T BYE message
  • the GMSC server After receiving the BYE message from the MSC server, the GMSC server fills in the ISUP REL message and sends the message to the PSTN through SIGTRAN / SG or SS7;
  • the MSC server After the MSC server receives a 200 000K message from the GMSC server, it issues the H.248 / MGCP command to request that MG1 release the terminal T2 on the relay side of the R4 core network;
  • the MSC server initiates a release signaling process to Iu to release the air interface and ground link resources. After receiving the Iu release response, it requests MG1 to release the terminal T1 on the radio access network side by issuing an H.248 / MGCP command.
  • the step of constructing the S IP-T BYE message includes encapsulating the I SUP REL call release message in the message, filling in the BYE message header according to the established peer address, and sending it to the destination GMSC server.
  • the process of initiating a call release by the network side includes the following steps:
  • the GMSC server receives the REL disconnection command from the PSTN;
  • GMSC server constructs S IP- T BYE message
  • the MSC server After receiving the BYE message from the GMSC server, the MSC server fills in the Di sconnec t message of the Iu interface and sends the message to the UE;
  • the MSC server issued the H.248 / MGCP command to request MG1 to release the terminal T2 on the relay side of the R4 core network, and after receiving the response from MG1, it returned the S IP-T 200 0 message to the GMSC server, which encapsulated the ISUP RLC message;
  • GMSC After receiving the 200K message from the MSC server, the server parses the ISUP RLC message encapsulated therein, and issues an H.248 / MGCP command to request MG1 to release the terminal T3 of the R4 core network relay side;
  • the MSC server then initiates a release signaling process to the Iu to release the air interface and ground link resources. After receiving the Iu release response, it requests the MG1 to release the wireless access network terminal T1 by issuing an H.248 / MGCP command;
  • the step of constructing a BYE message includes encapsulating the I SUP REL call release message from the PSTN in the BYE message, filling in the BYE message header according to the peer address of the established call, and sending it to the destination MSC. server.
  • the inter-office call from the UE to the UE in the mobile network is equivalent to merging the UE calling and called processes, and deleting the GMSC server entity and its corresponding S IP-T / ISUP interworking conversion function.
  • the method further includes changing the layered network into a flat network by using ENUM or a routing server solution, replacing end-to-end routing in the BICC protocol with segment-by-segment routing.
  • the method further includes adding an extension field to the INVITE and INFO messages of the SIP-T to transmit Codec messages supported by the mobile calling and called terminal or the media gateway to support the TrFO out-of-band Codec negotiation capability.
  • the present invention introduces SIP / SIP-T call signaling between the wireless core network MSC server and the (G) MSC server, the interworking between the WCDMA core network and the fixed NGN and cdma2000 core network in the ALL IP network is well solved. At this time, all servers use unified SIP / SIP-T signaling.
  • Figure 1 is the existing 3GPP R4BICSCN network structure
  • FIG. 2 is a SIP-T based CS domain network structure according to an embodiment of the present invention
  • FIG. 3 is a calling call (UE initiated call) process according to an embodiment of the present invention
  • FIG. 4 is a called call process according to an embodiment of the present invention ( Call initiated by the network side)
  • Figure 5 is a call release process initiated by the UE according to the embodiment of the present invention
  • FIG. 6 is a call release process initiated by a network side according to an embodiment of the present invention.
  • the general WCDMA communication system includes (G) MSC server, MGW, HLR, VLR, RNC, UE and other devices.
  • FIG. 2 depicts a SIP-T-based CS domain network structure according to an embodiment of the present invention.
  • the CS domain network architecture is mainly composed of an MSC server, a GMSC server, an MGW, and a BSS.
  • the invention proposes to use SIP / SIP-T call signaling between the MSC server and the (G) MSC server, while the other parts of the control still follow the R4 specification, that is, the MSC server / GMSC server uses the H. 248 protocol to control MGW, GW Available between IP bearers.
  • the caller call process includes the following steps:
  • step 1 the UE initiates a call setup to the MSC server (Setup), the MSC server returns the call process (Cal l proceeding), and instructs MG1 to add a new R4 core network relay-side terminal via the H.248 / MGCP instruction ( T2) and context (C1) resources.
  • step 2 the MSC server internally converts the call-related fields (calling number, called number, user attribute, bearer capacity, service indication, etc.) in the calling call setup (Setup) message from the Iu interface to ISUP. IAM message.
  • step 3 the MSC server constructs a SIP-T INVITE message template, fills in the SIP address field according to the called number, fills in the SDP information of the MG1 relay-side terminal T1 into the INVITE message body, and uses the ISUP IAM message constructed in step 2 as The message body extension is filled in the SIP INVITE message template, and INVITE is sent to the next hop address.
  • step 4 after the (G) MSC server receives the INVITE message from the MSC server (possibly after several proxy forwarding or redi rect redirection) and finds that it contains the I SUP IAM package, it directly uses the message content as pending Template for ISUP IAM message to PSTN.
  • step 5 the GMSC server instructs MG2 to newly allocate R4 core network side terminal (T3) and context (C3) resources through the H.248 / MGCP protocol, and the command carries the SDP information of the remote terminal T2.
  • T3 core network side terminal
  • C3 context
  • step 6 the GMSC server transmits the SDP information of the terminal T3 back to the MSC server through the INFO message.
  • the MSC server modifies the attributes of the terminal T2 on the MG1 according to the information, and sends the SDP information of the remote terminal T3 to the terminal T2. In this way, a bearer plane service media stream is established between T2 of MG1 and T3 of MG2.
  • step 7 the MSC server requires the MG1 to further add the terminal T1 on the radio access network side in the C1 context through the H.248 / MGCP protocol, and then issues an RAB assignment to allocate the air interface and Iu interface bearer plane resources (the The message carries the bearer association information of the terminal T1).
  • Step 7 can be parallel to steps 4, 5, and 6 in time.
  • the GMSC server modifies the ISUP IAM message template according to the address information in the SIP-T message header, selects the office route and idle circuit, and sends the final IAM message to the PSTN through SIGTRAN / SG or SS7;
  • step 9 the GMSC server receives the full ACM address message from the PSTN, encapsulates the message into the 180 ringing indication message of SIP-T, and sends the 180 message to the MSC server in the reverse direction;
  • the MSC server parses the packet from the 180 message.
  • the installed ACM information which reversely converts the relevant protocol domain information into the Alerting message of the Iu interface to notify the calling UE that the called user has been connected;
  • the GMSC server receives an A-band response message from the PSTN, encapsulates the message into a 200K instruction message of SIP-T, and sends the 200K message to the SC server in the reverse direction;
  • step 12 after receiving the message, the MSC server parses the encapsulated A wake-up information from the 200 message, reversely converts the relevant protocol domain information into a connection message of the Iu interface, and notifies the calling UE that the called user has answered, At the same time, the MG1 is instructed to modify the back unidirectional connection between T1 and T2 to a bidirectional connection through the H. 248 / MGCP protocol, and finally returns a SIP-T ACK message to the GMSC server; the end-to-end bearer of the calling process is completed.
  • FIG. 4 shows a called call process (a call initiated by a network side) according to an embodiment of the present invention.
  • the called call process includes the following steps:
  • the GMSC server receives an ISUP IAM message from the PSTN, and controls the MG2 to allocate ISUP circuit terminal (T1) and context (C1) resources through H.248 / MGCP commands. After analyzing the number attributes, it initiates a routing query to the HLR. ;
  • step 12 while the GMSC server is taking the route, it can control the MG2 through the H.248 / MGCP command to allocate the R4 core network side terminal (T2) resources in the context C1;
  • the GMSC server constructs a SIP-T INVITE message template, which contains the SDP information of the terminal T2, encapsulates the incoming IAM message into the message body, and sends the roaming number as the address information forward.
  • SIP-T INVITE message in step 14, the MSC server receives an INVITE message from the GMSC server After (possibly after several proxy forwarding or redi rect redirection), it finds that it contains the I SUP IAM package, parses the called number information (roaming number), and queries the VLR to initiate the paging and authentication process of the called UE;
  • step 15 after the called pager and authentication pass, the MSC server according to the call related domain (calling number, called number, user attributes, (Bearing capacity, service indication, etc.) are internally converted into a Setup message sent by the Iu interface to the called UE, and a Ca ll Conf i rmed response is received from the called UE;
  • the call related domain calling number, called number, user attributes, (Bearing capacity, service indication, etc.
  • the MSC server controls the MG1 to allocate R4 core network side terminal (T3) and context (C2) resources through the H.248 / MGCP command, which carries the SDP information of the remote terminal T2 contained in the INVITE, and then T3
  • the local SDP information is transmitted back to the GMSC server through the SIP-T INFO message.
  • the GMSC server modifies the attributes of terminal T2 on MG2 according to the information, and sends the SDP information of its remote terminal T3 to terminal T2, so that MG2's T2 and A bearer plane service media stream is established between T3 of MG1; this step may be performed in parallel with step 15;
  • step 17 the MSC server requires the MG1 to further add the terminal T4 on the radio access network side in the C2 context through the H.248 / MGCP protocol, and then issues an RAB assignment to allocate the air interface and Iu interface bearer plane resources (the The message carries the bearer association information of the terminal T4).
  • the MSC server receives the A 1 er ti ng called ringing message from the Iu interface, constructs a corresponding ISUP ACM message, and encapsulates the message into a 180 ringing indication message of SIP-T, and sends the 180 message.
  • the MSC server controls the MG1 to play the in-band ringback tone to the terminal T3 on the R4 core network side through the H.248 / GCP protocol;
  • step 20 after receiving the SIP-T 180 message from the MSC server, the GMSC server parses the encapsulated ACM information from the message, uses the message as a template, and makes necessary modifications according to the 180 message header information.
  • ACM message is sent to PSTN via SIGTRAN / SG or SS7;
  • the MSC server receives the connection response message from the Iu interface, constructs the ANM message of ISUP according to the message, and encapsulates the message into the 2000K instruction message of SIP-T, and sends the 200K message back to GMSC.
  • the MSC server controls the MG1 through the H.248 / MGCP protocol to stop playing the ringback tone on the terminal T3 on the R4 core network side, and modify the wireless access side terminal T4. For two-way connection with T3;
  • step 23 after receiving the SIP-T 200 message from the MSC server, the GMSC server parses the encapsulated A picture information from the message, and uses the message as a template to make necessary modifications according to the 200 message header information.
  • the final A wake-up message is sent to the PSTN through SIGTRAN / SG or SS7, and a SIP-T ACK message is returned to the GMSC server; the end-to-end bearer of the called process is established.
  • FIG. 5 shows a call release process initiated by a UE according to an embodiment of the present invention.
  • the release process includes the following steps:
  • step 31 the MSC server receives the disconnect command Disconnect from the UE, sends a Release command to the UE, requests it to release call control resources, and receives the call control resources from the UE. Response to completion of control resource release.
  • step 32 the MSC server constructs a SIP-T BYE message, encapsulates an ISUP REL call release message therein, fills in a BYE message header according to the address of the opposite end of the established call, and sends it to the destination server;
  • step 33 after receiving the BYE message from the MSC server, the GMSC server finds the ISUP REL package contained therein, parses the release reason and other information, fills in the ISUP REL message accordingly, and sends the message through S IGTRAN / SG or SS7.
  • the GMSC server requests the MG2 to release the terminal T3 on the relay side of the R4 core network by issuing the H.248 / MGCP command, and returns a 200K message of SIP-T to the MSC server after receiving the response from MG2, where Encapsulates the ILC RLC message; after receiving the ISUP RLC message from the PSTN, the GMSC server requests the MG2 to release the PSTN-side terminal T4 by issuing an H.248 / MGCP command;
  • step 35 after receiving the 200 OK message from the GMSC server, the MSC server requests the MG1 to release the terminal T2 on the relay side of the R4 core network by issuing an H.248 / MGCP command.
  • step 36 the MSC server initiates to Iu The release signaling process releases the air interface and ground link resources. After receiving the Iu release response, it issues a H.248 / MGCP command to request MG1 to release the terminal T1 on the radio access network side.
  • FIG. 6 shows a call release process initiated by a network side according to an embodiment of the present invention.
  • the release process includes:
  • the GMSC server receives the REL disconnection command from the PSTN, and issues an H.248 / MGCP command to request that MG2 release the terminal T4 on the PSTN side and the terminal T3 on the R4 core network side;
  • the GMSC server constructs a SIP-T BYE message, which encapsulates an ISUP REL call release message from the PSTN, fills in a BYE message header according to the address of the opposite end of the established call, and sends it to the destination MSC server;
  • step 43 after receiving the BYE message from the GMSC server, the MSC server finds the I SUP REL package contained therein, parses the release reason and other information therein, and fills in the disconnection message (Di s connec t) of the Iu interface accordingly, And send the message to the UE;
  • the MSC server requests the MG1 to release the terminal T2 on the relay side of the R4 core network by issuing the H.248 / MGCP command.
  • the MSC server After receiving the response from the MG1, it returns a SIP-T 200 0 message to the GMSC server, which encapsulates ISUP. RLC message; after receiving the 200K message from the MSC server, the GMSC server parses the ISUP RLC message encapsulated in it, and then issues an H.248 / MGCP command to request that MG1 release the terminal T3 on the relay side of the R4 core network;
  • step 45 the MSC server releases the call-related resources after receiving a release request from the UE, and returns a release completion response to the UE;
  • step 46 the MSC server then initiates a release signaling process to Iu to release the air interface and ground link resources. After receiving the Iu release response, it requests MG1 to release the radio access network side by issuing an H.248 / MGCP command. Terminal T1; steps 45 and 46 may be parallel with step 44 in time.
  • the interface between the (G) MSC server and the MG needs to be appropriately extended based on the H.248 / MGCP protocol;
  • an extended field needs to be added to the INVITE and INFO messages of SIP-T for transmitting Codec information supported by the mobile calling and called terminal or media gateway, such as FR, EFR, FR- AMR, etc .;
  • the network architecture separated from control and bearer can also be applied to GSM network or WCDMA network.
  • the present invention introduces SIP / SIP-T signaling as a call control signal between the MSC server and the (G) MSC server in a GSM or WCDMA wireless core network that carries voice over IP. Since (G) MSC server and MGW are logical functional entities, they can also be implemented on the same physical entity in implementation, similar to MSC and (G) MSC providing IP voice bearer interface, so this patent includes both MSC and (G) SIP / SIP-T signaling between MSCs.
  • the above process describes the normal process of the basic caller and called party of the present invention.
  • applications and processes of the SIP protocol in each supplementary service, intelligent service, and abnormal process refer to the above basic process implementation principles and ideas, and TS 23.218 of 3GPP.
  • IETF's draf t-ietf-s ipping-i sup-06 ISUP and SIP mapping specification
  • draf t-ietf-s ipping-s ipt-04 SIP-T specification.

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Abstract

L'invention se rapporte à un procédé de communication permettant d'appeler sur le domaine à commutation de circuits de réseaux centraux du GSM/WCDMA. Ledit procédé établit une signalisation d'appel et assure la communication entre des commutateurs (G)MSC ou des serveurs (G)MSC par l'intermédiaire du protocole SIP/SIP-T, ce qui résout le problème de communication entre les réseaux centraux du WCDMA et du grand NGN cdma2000 fixé dans un réseau TOUT IP. Conformément à l'invention, tous les serveurs utilisent la signalisation unifiée SIP/SIP-T, et l'acheminement de terminal à terminal remplace l'acheminement pas-à-pas antérieur grâce à une solution d'acheminement du type serveur d'acheminement ENUM, etc., pour transformer un réseau hiérarchique en un réseau planaire, ce qui permet de simplifier l'architecture réseau et de bénéficier des avantages du réseau IP.
PCT/CN2003/000784 2002-10-28 2003-09-17 Procede de communication permettant d'appeler sur le domaine a commutation de circuits de reseaux centraux du gsm/wcdma WO2004039097A1 (fr)

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AU2003271017A AU2003271017A1 (en) 2002-10-28 2003-09-17 A communication method for calling on the circuit switched domain of core networks of gsm/wcdma

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CN02145689.5 2002-10-28
CN02145689 2002-10-28

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Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN100396112C (zh) * 2004-09-29 2008-06-18 华为技术有限公司 通信系统中实现移动数据业务统一路由的方法
CN100403795C (zh) * 2004-12-31 2008-07-16 华为技术有限公司 一种实现ngn网络与移动网络视频互通的方法
CN103108298A (zh) * 2011-11-10 2013-05-15 北京信威通信技术股份有限公司 一种移动通信系统中全网呼叫的实现方法

Families Citing this family (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN100456847C (zh) * 2006-03-08 2009-01-28 华为技术有限公司 一种选择被叫路由的方法、网络及设备
CN101577724B (zh) * 2006-06-09 2016-04-13 Sk电信有限公社 提供基于会话发起协议的早期媒体服务的方法

Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2001063947A1 (fr) * 2000-02-21 2001-08-30 Telefonaktiebolaget Lm Ericsson (Publ) Procede de regulation des appels permettant de reduire les delais a un minimum lors de l'etablissement d'appels multimedia ou vocaux dans un reseau de telecommunications a commutation de paquets
US20010043577A1 (en) * 2000-02-22 2001-11-22 Peter Barany System and method for controlling a wireless packet switched voice call
WO2002015598A1 (fr) * 2000-08-16 2002-02-21 Nokia Corporation Systeme et procede permettant la fourniture de services sur differents reseaux
CN1351790A (zh) * 1999-05-17 2002-05-29 艾利森电话股份有限公司 用于提供无线接入载体业务的方法和设备

Patent Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1351790A (zh) * 1999-05-17 2002-05-29 艾利森电话股份有限公司 用于提供无线接入载体业务的方法和设备
WO2001063947A1 (fr) * 2000-02-21 2001-08-30 Telefonaktiebolaget Lm Ericsson (Publ) Procede de regulation des appels permettant de reduire les delais a un minimum lors de l'etablissement d'appels multimedia ou vocaux dans un reseau de telecommunications a commutation de paquets
US20010043577A1 (en) * 2000-02-22 2001-11-22 Peter Barany System and method for controlling a wireless packet switched voice call
WO2002015598A1 (fr) * 2000-08-16 2002-02-21 Nokia Corporation Systeme et procede permettant la fourniture de services sur differents reseaux

Non-Patent Citations (2)

* Cited by examiner, † Cited by third party
Title
DURUCHUAN, ZHONGJIAQI: "Research of ALL IP Network Architecture of WCDMA.", GUANGDONG COMMUNICATION TECHNOLOGY, vol. 22, no. 1, January 2002 (2002-01-01), pages 19 - 22 *
ZHANGHUAYAN, WANGWEI.: "3GPP", TELECOMWORLD, vol. 92, July 2002 (2002-07-01), pages 72 - 73 *

Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN100396112C (zh) * 2004-09-29 2008-06-18 华为技术有限公司 通信系统中实现移动数据业务统一路由的方法
CN100403795C (zh) * 2004-12-31 2008-07-16 华为技术有限公司 一种实现ngn网络与移动网络视频互通的方法
CN103108298A (zh) * 2011-11-10 2013-05-15 北京信威通信技术股份有限公司 一种移动通信系统中全网呼叫的实现方法

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AU2003271017A1 (en) 2004-05-13
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