WO2004039097A1 - A communication method for calling on the circuit switched domain of core networks of gsm/wcdma - Google Patents

Abstract

The invention provides a communication method for calling on the circuit switched domain of core networks of GSM/WCDMA. The method sets up a call signaling and communicates between (G)MSCs or (G)MSC Servers through SIP/SIP-T protocol, which solves the problem of communication between the core networks of WCDMA and fixed NGN, cama2000 in ALL IP network. According to the invention, all servers use unite SIP/SIP-T signaling, and terminal to terminal route takes the place of former hop by hop route by the route solution, such as, ENUM, route server, etc, to make hierarchical network into planar network, which reduces the network architecture and takes the advantage of IP network.

Description

 Communication method for calling in circuit domain of GSM / WCDMA core network TECHNICAL FIELD

 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.

Background technique

 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, and the PS domain provides packet data services. The WCDMA standard currently has three versions, R99, R4, and R5, and subsequent versions are under development.

 From the perspective of the WCDMA CS domain network structure, R4 and R99 have changed greatly. R99 core network

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. (G) Call signaling BICC is used between MSC servers, and TDM / ATM / IP multiple bearers are supported between MGWs. (G) The MSC server controls the MGW through the 3GPP extended H. 248 protocol.

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. Fixed network NGN At present, most manufacturers and operators have adopted SIP / SIP-T as the call control signaling between softswitch Sof tswi tch, and 3IP2's ALL IP specification has selected SIP / SIP-T as the inter-MSCe For call control signaling, the MSCe function is similar to the MSC server in WCDMA. 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.

One of the advantages of SIP is the user positioning function. 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. At the same time, 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.

 On the other hand, 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. In this mode, BICC signaling and TUP and ISUP signaling use the same segment-by-segment routing instead of end-to-end routing. At this time, 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.

Summary of the Invention

 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.

 Further, 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.

In addition, the method further includes establishing a UE-to-UE call procedure in the mobile network. Specifically, the calling process includes the following steps:

 UE initiates a call request;

 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.

 Further, 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;

 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;

 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;

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.

Preferably, 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. Preferably, 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. .

 Specifically, 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.

 In addition, 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.

 Further, 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;

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. ;

 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.

 Preferably, after receiving the ISUP IAM message from the 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.

 Specifically, 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.

 In addition, 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;

 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;

 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. Specifically, 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.

 In addition, 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;

 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;

 Specifically, 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.

In addition, 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. Further, 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.

 Preferably, 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.

 Since 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.

 After adopting SIP / SIP-T signaling, other routing solutions such as ENUM and routing server can be used to replace the original segment-by-segment routing method with end-to-end routing. Changing the layered network into a flat network will help simplify Network structure and IP network advantages.

BRIEF DESCRIPTION OF THE DRAWINGS

 Figure 1 is the existing 3GPP R4BICSCN network structure;

 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.

detailed description

In order for those skilled in the art to better understand the present invention, the following describes the present invention with reference to the accompanying drawings. Embodiments of the 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. In a WCDMA communication system, 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 following further describes the present invention with reference to a call flow. As shown in FIG. 3, a caller call (call initiated by a UE) according to an embodiment of the present invention is described. The caller call process includes the following steps:

 First, in 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. Then, in 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.

In 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. In 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.

 In 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.

 In 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.

 In 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.

 In step 8, 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;

 In 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;

In step 10, after receiving the message, 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;

 In step 11, 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;

 In 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:

 First, in step 11, 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. ;

 Then, in 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;

Then, in step 13, 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;

 Then, according to the result of the above steps, in 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;

 In step 16, 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;

 In 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).

In step 18, 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. Reverse to GMSC server; In step 19, after receiving the Iu interface Alerting 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;

 In 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;

 In step 21, 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. Server

 In step 22, after receiving the Iu interface connection message, 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;

 In 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:

In 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.

 In 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;

 In 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. Sent to PSTN; In step 34, 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;

 In 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. In 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:

In step 41, 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; In step 42, 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;

 In 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;

 In step 44, 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. 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;

 In 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;

 In 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 call flow examples described above describe the S IP-T flow between the end office MSC server and the GMSC server in the case of UE-to-PSTN and PSTN-to-UE calls. Merge the UE calling and called processes, and delete the GMSC server entity and its corresponding S IP_TnSUP interworking conversion function), in This is not repeated.

 In order to implement the above-mentioned calling process, the interface between the (G) MSC server and the MG needs to be appropriately extended based on the H.248 / MGCP protocol;

 In addition, in order to support the TrFO out-of-band Codec negotiation capabilities, 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. For the 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. (Call flow), IETF's draf t-ietf-s ipping-i sup-06 (ISUP and SIP mapping specification) and draf t-ietf-s ipping-s ipt-04 (SIP-T specification).

 Although the present invention is described through the embodiments, those skilled in the art should know that those skilled in the art can make improvements and modifications without departing from the essential spirit of the present invention. The entitlement should therefore include these improvements and changes.

Claims

Rights request 1. A communication method for calling in a circuit domain of a GSM / WCDMA core network, the GSM communication system includes (G) MSC, BSC, HLR, VLR, UE, and the WCDMA communication system includes (G) MSC server ( (G) MSC Server), MGW, HLR, VLR, RNC, UE, the method includes the steps: applying SIP / SIP-T protocol between (G) MSC or (G) MSC server to establish call signaling and communicate with each other.  . 2. The communication method for calling in a circuit domain of a GSM / WCDMA core network according to claim 1, wherein the MGW is carried by an IP network, and (G) the MSC server controls the MGW for message processing by using the H.248 protocol.  3. The communication method for making a call in the circuit domain of the GSM / WCDMA core network according to claim 1, further comprising using an EN circle or a routing server solution to replace the BICC protocol with an end-to-end routing solution. The segment-by-segment routing method changes a hierarchical network into a flat network.  4. The communication method for calling in the circuit domain of the GSM / WCDMA core network according to claim 1, further comprising adding an extension field to the INVITE and INFO messages of the SIP-T for transmitting the mobile calling and called terminal or the media gateway. Supported Codec messages to support TrFO out-of-band Codec negotiation capabilities.  5. The communication method for making a call in the GSM / WCDMA core network circuit domain according to one of the preceding claims, comprising at least one of the following calling procedures:  Calling call (call initiated by UE); Called call (call initiated by the network side); Release of call initiated by UE;  The call initiated by the network side is released.  6. The communication method for calling in the circuit domain of the GSM / WCDMA core network according to claim 5, further comprising establishing a UE-to-UE call process in the mobile network.  7. The communication method for performing call processing in a circuit domain of a GSM / WCDMA core network according to claim 6, wherein the inter-office call from UE to UE in the mobile network is equivalent to a combination of UE calling and called processes, And delete (G) MSC server entity and its corresponding SIP-T / ISUP interworking conversion function.  8. The communication method for calling in a circuit domain of a GSM / WCDMA core network according to claim 7, wherein the calling process comprises the following steps:  UE initiates a call request;  The MSC server responds to the request and constructs a SIP-T INVITE message, 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 bearer plane service media flow between the terminal T2 on MG1 and the terminal T3 on 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 communication method for performing call processing in a GSM / WCDMA core network circuit domain according to claim 8, further comprising the step The GMSC server modifies the ISUP I AM message template according to the address message in the SIP-T message header, selects the office route and idle circuit, and passes the final IAM message through SIGTRN / SG or SS7 sent to PSTN;  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 back to 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 back to the MSC server;  After receiving the message, the MSC server parses the encapsulated A picture information from the 200 message, and notifies the calling UE that the called user has answered.  10. The communication method for making a call in a circuit domain of a GSM / WCDMA core network according to claim 9, wherein the calling process further comprises that the MSC server sends the calling related domain in the calling Setup call setup message from the Iu interface to Steps to internalize IAM messages into ISUP.  11. The communication method for making a call in the circuit domain of the GSM / WCDMA core network according to claim 9, wherein the calling process further comprises, after the GMSC server receives the INVITE message from the MSC server, it finds that the Encapsulating, then directly using the message content as a template for an ISUP IAM message to be sent to the PSTN.  12. The communication method for making a call in a circuit domain of the GSM / WCDMA core network according to claim 9, wherein the step of constructing an INVITE message template comprises filling in a SIP address domain according to the called number, and relaying the MG1 relay-side terminal The SDP message of T1 is filled into the SIP INVITE message template. 13. Calling in the circuit domain of a GSM / WCDMA core network according to claim 5. Call communication method, the called process includes the following steps: The GMSC server receives the ISUP IAM message from the PSTN;  The GMSC server constructs a SIP-T INVITE message template, and sends the SIP-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 communication method for calling in a circuit domain of a GSM / WCDMA core network according to claim 13, further comprising the following steps:  The SC server receives the Alerting called ringing message from the Iu interface, constructs the corresponding I SUP ACM message, and encapsulates the message into the 180 ringing indication message of SIP-T, and sends the 180 message to the GMSC server in the reverse direction. ;  After receiving the SIP-T 180 message from the MSC server, the GMSC server parses the encapsulated ACM information from the message, and uses the message as a template to make necessary changes based on the 180 message header information. After the modification, the final ACM The message is sent to PSTN through SIGTRAN / SG or SS7;  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 200 message back to the GMSC server. ; After receiving the SIP-T 200 message from the MSC server, the GMSC server parses the encapsulated A wake-up information from the message, using this message as a template, according to the 200 message header information After making necessary modifications, the final Alice message is sent to PSTN through SIGTRAN / SG or SS7.  15. The communication method for making a call in the circuit domain of the GSM / WCDMA core network according to claim 14, wherein the constructed S IP-T INVITE message template includes an SDP message of the terminal T2, and the incoming side IAM The message is encapsulated in the message body.  16. The communication method for calling in a circuit domain of a GSM / WCDMA core network according to claim 5, wherein 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;  MSC server constructs S IP-T BYE message;  The GMSC server fills in the ISUP REL message after receiving the BYE message from the MSC server and sends the message to the PSTN through SIGTRAN / SG or SS7;  After receiving the 200K message from the GMSC server, the MSC server sends 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 the Iu to release the air interface and ground link resources. After receiving the Iu release response, it issues a H.248 / MGCP command to request the MG1 to release the terminal T1 on the radio access network side.  17. The communication method for performing call processing in a GSM / WCDMA core network circuit domain according to claim 16, wherein the step of constructing a SIP-T BYE message includes encapsulating an I SUP REL call release message in the message, and Fill in the BYE message header according to the established peer address and send it to the destination GMSC server. 18. The communication method for making a call in a circuit domain of a GSM / WCDMA core network according to claim 5, wherein 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;  The GMSC server constructs a SIP-T BYE message;  After receiving the BYE message from the GMSC server, the MSC server fills in a Dis connect message of the Iu interface and sends the message to the UE;  The MSC server issued the H. 248 / MGCP command to request that MG1 release the terminal T2 on the relay side of the R4 core network, and after receiving the response from MG1, return the 200K message of SIP-T to the GMSC server, which encapsulated the ISUP RLC message; After receiving the 200 message from the MSC server, after parsing the ISUP RLC message encapsulated in it, the H.248 / MGCP command is issued 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 issues a H.248 / MGCP command to request the MG1 to release the wireless access network terminal T 1;  19. The communication method for calling in a circuit domain of a GSM / WCDMA core network according to claim 18, wherein the step of constructing a BYE message comprises encapsulating the BYE message from a PSTN ISUP REL call release message, And> Fill in the BYE message header according to the opposite address of the established call and send it to the destination MSC server.