CN1695385A - Communication method for calling in GSM/WCDMA core network circuit domain - Google Patents

Abstract

The invention discloses a communication method for calling in GSM/WCDMA core network circuit domain, which applies SIP/SIP-T protocol to establish calling signaling and communicate with each other between MSC or MSC Server, solving the problem of intercommunication between WCDMA core network and fixed NGN, cdma2000 core network in ALL IP network. At this time, all the servers adopt uniform SIP/SIP-T signaling, and can replace the original section-by-section routing mode with end-to-end routing through other routing solutions such as ENUM, routing Server, etc., so as to change the layered network into a plane network, which is helpful to simplify the network structure and bring the advantages of IP network into play.

Description

Communication method for calling in GSM/WCDMA core network circuit domain Technical field

The invention relates to the field of communication technology, in particular to a communication method for calling in the circuit domain of the GSM/WCDMA core network.

Background technique

The WCDMA core network is mainly divided into a circuit CS domain and a packet PS domain. The CS domain provides voice, fax and circuit data services, and the PS domain provides packet data services. There are currently three versions of the WCDMA standard, R99, R4, and R5, and subsequent versions are being formulated.

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

The CS domain is composed of entities such as MSC, GMSC, and HLR. The network architecture is basically the same as that of the GSM network. The MSCs are based on TDM bearer and call signaling TUP/ISUP. Figure 1 depicts the existing 3GPP R4BICSCN network structure. The R4 specification proposes the concept of BICSCN (Bearer Independent Circuit Switching Core Network), that is, a CS domain that has nothing to do with the load, and is mainly composed of MSC servers (MSC Server), GMSC servers (GMSC Server), MGW, HLR, etc. (G) MSC servers use call signaling BICC, and MGWs support multiple bearers of TDM/ATM/IP. (G) The MSC server controls the MGW through the H.248 protocol extended by 3GPP.

3GPP R4 selects the BICC (Bearer Independent Call Control) protocol developed by ITU-T as the call signaling between (G) MSC servers, mainly because BICC is developed on the basis of ISUP and inherits most of the traditional signaling design The idea is to support various TDM/ATM (AAL1, AAL2)/IP bearers of the MGW. Considering the R4 CS domain network The network architecture is consistent with the current fixed network NGN and the LMSD (Legency MS Domain, traditional mobile terminal domain) network architecture of 3GPP2 (cdma2000 standardization organization), that is, the Server+MGW structure that separates control and bearer is adopted. At present, most manufacturers and operators of fixed network NGN have adopted SIP/SIP-T as the call control signaling between Softswitches, and the 3GPP2 ALL IP specification has selected SIP/SIP-T as the signaling between MSCes. Call control signaling, MSCe function is similar to MSC server in WCDMA. SIP (Session Initiation Protocol, Session Initiation Protocol) is an IP telephony signaling protocol proposed by 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 participated by multiple participants, and can dynamically adjust and modify session attributes, such as session bandwidth requirements, transmitted media types (voice , video and data, etc.), media codec formats, support for multicast and unicast, etc.

SIP fully considers the expansion adaptability to other protocols in design. It supports many kinds of address description and addressing, including: user name@host address, called number@?3 gateway address, description of common E. 164 telephone number such as Tel: 010-62281234, etc. In this way, the SIP caller can identify whether the called party is on the traditional telephone network according to the called party 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 location function. SIP itself has the function of registering with the registration server, and it can also use the location servers provided by other location servers such as DNS and LDAP to enhance its location function.

SIP-T is an application of SIP, which encapsulates ISUP information units or translates ISUP information units in the payload part of SIP, and uses the call control process of SIP protocol to simplify ISUP control signaling. Since different standardization organizations adopt different protocols, it will bring difficulties in interconnection and intercommunication between different networks in the future ALL IP stage.

On the other hand, the protocol stack for IP carrying BICC signaling defined by 3GPP is BICC/STCP/IP/L2/L1, and the STCP link is equivalent to a traditional SS7 link, which is generally a static link. In this way, BICC signaling and TUP and ISUP signaling adopt the same segment-by-segment routing instead of end-to-end routing. At this time, the wireless core network control layer maintains a hierarchical structure, that is, the TMSC server is required to complete the long-distance routing function. This routing method and network structure should be more optimized in the IP network.

Contents of the invention

The purpose of the present invention is to provide a communication method that can realize calling in the GSM/WCDMA communication system, and is used to overcome the above-mentioned deficiency that different systems cannot be interconnected and intercommunicated.

The object of the present invention is achieved by following method, a kind of communication method that realizes calling in GSM/WCDMA communication system, described GSM communication system comprises (G) MSC, BSC, HLR, VLR, UE, described WCDMA The communication system includes (G) MSC server, MGW, HLR, VLR, RNC, UE, and the method includes the steps of: applying SIP/SIP-T protocol between (G) MSCs or (G) MSC servers to establish call signaling and perform mutual communication.

Further, the method also includes that the MGWs are carried by the IP network, and the (G) MSC server uses the H.248 protocol to control the MGW to process messages.

The specific described method includes at least one of the following calling procedures:

Calling call; Called call; Call release initiated by UE; Call release initiated by network side.

In addition, the method also includes establishing a UE-to-UE call process in the mobile network. Specifically, the calling process of the calling party consists of 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 transmits the INFO message to the MSC server in reverse to establish a terminal on MG1 Service media flow on the bearer plane between T2 and 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 established.

Further, the step of the calling process also includes that the GMSC server modifies the ISUP IAM message template based on the address message in the SIP-T message header, selects an office route and an idle circuit, and sends the final IAM message through SIGTRN/SG or SS7 to PSTN;

The GMSC server receives the full message of the ACM address 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 reverse;

After receiving the message, the MSC server parses the encapsulated ACM information from the 180 message, and notifies 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 a SIP-T 200 OK indication message, and sends the 200 OK message to the MSC server in reverse;

After receiving the message, the MSC server parses the encapsulated ANM information from the 200 message, and notifies the calling UE that the called user has answered.

Preferably, the calling process further includes a step of the MSC server internally converting the call-related field in the calling Setup message from the Iu interface into an ISUP IAM message. Preferably, the calling process also includes the step of, after the GMSC server receives the INVITE message of the MSC server and finds that it contains the package of ISUP IAM, the content of the message is directly used as the template of the ISUP IAM message to be sent to the PSTN .

Specifically, the step of constructing the INVITE message template includes filling in the SIP address field according to the called number, and filling in the SDP message of the MG1 trunk terminal T1 into the SIP INVITE message template.

In addition, the called call process includes the following steps:

The GMSC server receives the ISUP IAM message from PSTN;

The GMSC server constructs a SIP-T INVITE message template, and sends a SIP-T INVITE message forward with the acquired roaming number as address information;

The MSC server reversely transmits the SIP-T INFO message back to the GMSC server to establish different bearer plane service media streams between the terminal T2 of MG2 and the terminal T3 of MG1.

The GMSC server returns a SIP-T ACK message to the MSC server; the end-to-end bearer of the called process is established.

Further, the called call process also includes the following steps

The MSC server receives the Alerting called ringing message from the Iu interface, constructs a corresponding ISUP ACM message, encapsulates the message into a SIP-T 180 ringing indication message, and sends the 18G message to the GMSC server in reverse;

After the GMSC server receives the SIP-T 180 message from the MSC server, it parses the encapsulated ACM information from the message, uses the message as a template, and after necessary modification of the 180 message header information, passes the final ACM message through SIGTRAN/SG or SS7 sent to PSTN; The MSC server receives the connection (Connect) response message from the Iu interface, constructs the ANM message of ISUP according to the message, encapsulates the message into the 200 OK indication message of SIP-T, and sends the 200 OK message to the GMSC server in reverse ;

After the GMSC server receives the SIP-T 200 message from the MSC server, it parses out the encapsulated A-picture information from the message, uses the message as a template, and after making necessary modifications according to the 200 message header information, passes the final A-picture message through SIGTRAN/SG or SS7 sent to PSTN.

Preferably, after receiving the PSTN ISUP IAM message, the GMSC server controls MG2 to allocate ISUP circuit terminal T1 and context C1 resources through the H. 248/MGCP command, and initiates a routing query to the HLR after analyzing the number attributes.

Specifically, the constructed SIP-T INVITE message template includes the SDP message of the terminal T2, and encapsulates the incoming IAM message into the message body.

In addition, the call release process initiated by the UE includes the following steps:

The MSC server receives a Disconnect 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 PSTN via SIGTRAN/SG or SS7;

After receiving the 200 0K message from the GMSC server, the MSC server requests 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 initiates a release signaling process to Iu to release the air interface and ground link resources, and after receiving the Iu release response, the MSC server sends an H.248/MGCP command to request MG1 to release the terminal T1 on the radio access network side. Specifically, the step of constructing the SIP-T BYE message includes encapsulating the ISUP REL call release message in the message, and 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 the network side initiating call release includes the following steps:

The GMSC server receives the REL disconnection command from PSTN;

GMSC server constructs SIP-T BYE message;

After receiving the BYE message from the GMSC server, the MSC server fills in the disconnection (Disconnect) message of the Iu interface, and sends 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, and returns a SIP-T 200 0 message to the GMSC server after receiving a response from MG1, in which the RLC message of the ISUP is encapsulated; GMSC After receiving the 200 OK message from the MSC server, the server parses the encapsulated ISUP RLC message, and requests MG1 to release the terminal T3 on the relay side of the R4 core network by issuing an H. 248/MGCP command;

The MSC server then initiates a release signaling process to Iu to release the air interface and ground link resources, and after receiving the Iu release response, 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 BYE message includes, encapsulating the ISUP REL call release message from the PSTN in the BYE message, and 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 cross-office call from UE to UE in the mobile network is equivalent to the merging of UE calling and called procedures, and the GMSC server entity and its corresponding SIP-T/ISUP interworking conversion function are deleted. Further, the method also includes using the ENUM or routing server solution to replace the segment-by-segment routing in the BICC protocol with end-to-end routing, so as to change the layered network into a flat network.

Preferably, the method further includes adding an extension field in the INVITE and INFO messages of SIP-T to transmit the Codec messages supported by the mobile calling and called terminals or media gateways, so as 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, it can well solve the intercommunication between the WCDMA core network and the fixed NGN and cdma2000 core network in the ALL IP network. At this time, unified SIP/SIP-T signaling is adopted between all servers.

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 with end-to-end routing, changing the hierarchical network into a flat network, which will help simplify Network structure and the advantage of IP network.

Description of drawings

Fig. 1 is the existing 3GPP R4BICSCN network structure;

Fig. 2 is the CS domain network structure based on SIP-T of the embodiment of the present invention; Fig. 3 is the process of the calling call (the call initiated by the UE) of the embodiment of the present invention; Fig. 4 is the process of the called call of the embodiment of the present invention ( A call initiated by the network side); FIG. 5 is a call release process initiated by the UE according to the embodiment of the present invention;

Fig. 6 is a flow of call release initiated by the network side according to the embodiment of the present invention.

Detailed ways

In order for those skilled in the art to better understand the present invention, the present invention will be described below in conjunction with the accompanying drawings. Embodiment of the invention.

A general WCDMA communication system includes (G) MSC server, MGW, HLR, VLR, RNC, UE and other devices.

Figure 2 depicts the SIP-T-based CS domain network structure of the embodiment of the present invention. In the WCDMA communication system, the CS domain network architecture is mainly composed of MSC server, GMSC server, MGW, and BSS. Considering the future interconnection and intercommunication, this The invention proposes the use of SIP/SIP-T call signaling between the MSC server and the (G) MSC server, while the control of other parts still follows the R4 specification, that is, the MSC server/GMSC server uses the H. 248 protocol to control MGW, GW IP bearer can be used between.

The present invention will be further described below in conjunction with the call flow. As shown in FIG. 3, the flow of the calling call (the call initiated by the UE) of the embodiment of the present invention is shown. The calling flow includes the following steps:

First, in step 1, the UE initiates call setup (Setup) to the MSC server, and the MSC server returns the call procedure (Call proceeding), and instructs MG1 to newly allocate the R4 core network relay side terminal ( 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 capability, service indication, etc.) in the calling call setup (Setup) message from the Iu interface into ISUP's IAM messages.

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 is extended to fill in the SIP INVITE message template, and the INVITE is sent to the next hop address. In step 4, after the (G) MSC server receives the INVITE message from the MSC server (maybe forwarded by several proxy or redirected by redirect), and finds that it contains the package of ISUP IAM, it will directly take the content of the message as the message to be sent 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 sends the SDP information of the terminal T3 back to the MSC server through the INFO message, and the MSC server modifies the attributes of the terminal T2 on 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 flow is established between T2 of MG1 and T3 of MG2.

In step 7, the MSC server requests 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 RAB assignment to allocate air interface and Iu interface bearer plane resources (the The message carries bearer association information of the terminal T1). Step 7 can be paralleled with 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 an office route and an idle circuit, and sends the final IAM message to PSTN through SIGTRAN/SG or SS7;

In step 9, the GMSC server receives the full message of the ACM address from the PSTN, encapsulates the message into a 180 ringing indication message of SIP-T, and sends the 180 message to the MSC server in reverse;

In step 10, after receiving the message, the MSC server parses the packet from the 180 message The ACM information installed, reversely converts the relevant protocol domain information into the Alerting message of the Iu interface, and notifies the calling UE that the called user has been connected;

In step 11, the GMSC server receives the AMR response message from the PSTN, encapsulates the message into a SIP-T 200 OK indication message, and sends the 200 OK message to the SC server in reverse;

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, command MG1 to modify the unidirectional connection between T1 and T2 to a bidirectional connection through the H. 248/MGCP protocol, and finally return a SIP-T ACK message to the GMSC server; the end-to-end bearer of the calling process is established.

Figure 4 shows the called call flow (call initiated by the network side) of the embodiment of the present invention. The called call flow includes the following steps:

First, in step 11, the GMSC server receives the ISUP IAM message from the PSTN, controls MG2 to allocate ISUP circuit terminal (T1) and context (C1) resources through the H. 248/MGCP command, and initiates a routing query to the HLR after analyzing the number attributes ;

Then, in step 12, while fetching the route, the GMSC server can control MG2 to allocate R4 core network side terminal ( T2 ) resources in the context C1 through the H. 248/MGCP command;

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 it forward with the obtained roaming number as the address information SIP-T INVITE message; In step 14, the MSC server receives the INVITE message from the GMSC server Afterwards (possibly through several proxy forwarding or redirect redirection), it is found that it contains the package of ISUP IAM, and the called number information (roaming number) is parsed, and the paging and authentication process of the called UE is initiated after querying the VLR;

Then, according to the results of the above steps, in step 15, after the called party is paged and authenticated, the MSC server, according to the call-related fields (calling number, called number, user attribute, bearer capacity, service indication, etc.) are internally converted into a Setup message sent by the Iu interface to the called UE, and receive a Ca l l Confirmed response from the called UE;

In step 16, the MSC server controls MG1 to allocate R4 core network side terminal (T3) and context (C2) resources through H. The local SDP information of the remote terminal T3 is sent back to the GMSC server through the SIP-T INFO message, and the GMSC server modifies the attributes of the terminal T2 on the MG2 according to the information, and sends the SDP information of the remote terminal T3 to the terminal T2, so that T2 of MG2 and The bearer plane service media flow is established between T3 of MG1; this step can be paralleled with step 15;

In step 17, the MSC server requests MG1 to further add the terminal T4 on the side of the wireless access network in the C2 context through the H.248/MGCP protocol, and then issues RAB assignments to allocate air interface and Iu interface bearer plane resources (the The message carries bearer association information of the terminal T4).

In step 18, the MSC server receives the A 1erting called ringing message from the Iu interface, constructs a corresponding ISUP ACM message, and encapsulates the message into a SIP-T 180 ringing indication message, and sends the 180 message Reversely sent to the GMSC server; In step 19, after receiving the Iu interface Alerting message, the MSC server controls MG1 to play the in-band ringback tone backwards on the terminal T3 on the R4 core network side through the H.248/GCP protocol;

In step 20, after the GMSC server receives the SIP-T 180 message from the MSC server, it parses out the encapsulated ACM information from the message, takes the message as a template, and after performing necessary modifications according to the 180 message header information, the final ACM messages are sent to PSTN via SIGTRAN/SG or SS7;

In step 21, the MSC server receives the connection response message from the Iu interface, constructs an ISUP ANM message according to the message, encapsulates the message into a SIP-T 2000K indication message, and sends the 200 OK message to the GMSC in reverse server;

In step 22, after receiving the Iu interface connection message, the MSC server controls MG1 to stop playing the ringback tone on the terminal T3 on the R4 core network side through the H.248/MGCP protocol, and modifies the terminal T4 on the wireless access side For bidirectional 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, uses the message as a template, and performs necessary modifications according to the 200 message header information, Send the final A wake-up message to PSTN via SIGTRAN/SG or SS7, and return a SIP-T ACK message to the GMSC server at the same time; the end-to-end bearer of the called process is established.

FIG. 5 shows the call release process initiated by the UE according to the embodiment of the present invention, and the release process includes the following steps:

In step 31, the MSC server receives the disconnection command Disconnect from the UE, sends a release (Release) command to the UE to request it to release the call control resources, and receives from the UE Response to control resource release completion.

In step 32, the MSC server constructs a SIP-T BYE message, encapsulates the ISUP REL call release message, and fills in the BYE message header according to the peer address 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 therein, fills in the ISUP REL message accordingly and sends the message through SIGTRAN/SG or SS7 Sent to PSTN; In step 34, the GMSC server requests 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 200 OK message of SIP-T to the MSC server after getting a response from MG2, wherein Encapsulate the RLC message of ISUP; After receiving the ISUP RLC message from PSTN, the GMSC server requires MG2 to release the terminal T4 on the PSTN side by issuing an H. 248/MGCP command;

In step 35, after receiving the 200 OK message from the GMSC server, the MSC server requires 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 a Release the signaling process to release the air interface and ground link resources, and after receiving the Iu release response, request MG1 to release the terminal T1 on the radio access network side by sending an H.248/MGCP command.

Figure 6 shows the call release process initiated by the network side of the embodiment of the present invention, and the release process includes:

In step 41, the GMSC server receives the REL disconnection command from the PSTN, and requests MG2 to release the terminal T4 on the PSTN side and the terminal T3 on the R4 core network side by issuing an H. 248/MGCP command; In step 42, the GMSC server constructs a SIP-T BYE message, encapsulates the ISUP REL call release message from the PSTN, and fills in the BYE message header according to the peer address 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 ISUP REL package contained therein, analyzes the release reason and other information therein, and fills in the disconnection message (Dis connect) of the Iu interface accordingly, and send the message to the UE;

In step 44, the MSC server sends an H.248/MGCP command to request MG1 to release the terminal T2 on the relay side of the R4 core network, and returns a SIP-T 2000 message to the GMSC server after receiving a response from MG1, which encapsulates the ISUP After receiving the 200 OK message from the MSC server, the GMSC server parses the ISUP RLC message encapsulated therein, and requests MG1 to release the terminal T3 on the relay side of the R4 core network by issuing an H. 248/MGCP command;

In step 45, the MSC server releases the call-related resources after receiving the release (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, and after receiving the Iu release response, the MSC server sends an H.248/MGCP command to request MG1 to release the resources on the radio access network side. Terminal T1; Steps 45, 46 can be parallelized with step 44 in time.

The above describes the call flow examples respectively describing the SIP-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 of UE calling and called procedures, and delete the GMSC server entity and its corresponding SIP_TnSUP interworking conversion function), in This will not be repeated.

In order to realize the above call process, the interface between the (G) MSC server and the MG needs to be appropriately extended on the basis of the H.248/MGCP protocol;

In addition, in order to support the TrFO out-of-band Codec negotiation capability, it is necessary to add an extension field in the INVITE and INFO messages of SIP-T to transmit the Codec information supported by the mobile calling and called terminals or media gateways, such as HD, FR, EFR, WB- AMR, etc.;

The network architecture due to the separation of control and bearer can also be applied to GSM network or WCDMA network. The present invention introduces SIP/SIP-T signaling as the call control signaling between the MSC server and the (G)MSC server in the GSM or WCDMA wireless core network with voice over IP. Since the (G) MSC server and the MGW belong to logical functional entities, they can also be implemented on the same physical entity in terms of implementation, similar to the MSC and (G) MSC that provide IP voice bearer interfaces, so this patent includes MSC and (G) Situation of using 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 application and process of the SIP protocol in various supplementary services, intelligent services and abnormal processes, the implementation principles and ideas of the above basic process can be referred to, and TS 23.218 of 3GPP (call flow), IETF's draft t-ietf-s ipping-i sup-06 (ISUP-SIP mapping specification) and draft t-ietf-s ipping-s ipt-04 (SIP-T specification).

Although the present invention has been described through the embodiments, those skilled in the art should know that improvements and modifications can be made by those skilled in the art without departing from the essence and spirit of the present invention. The claims should therefore cover these improvements and modifications.

Claims (18)

1. Claim

   1st, a kind of communication means called in GSM/WCDMA core net circuit domains, the gsm communication system includes（G) MSC, BSC, HLR, VLR, UE, the WCDMA communication system include（G) MSC servers（（G) MSC Server), MGW, HLR, VLR, RNC, UE, the method comprising the steps of：（G) between MSC or（G) set up call signaling using SIP/SIP-T agreements between MSC server and communicate with each other.

   Carried between the 2, communication means according to claim 1 called in GSM/WCDMA core net circuit domains, the MGW with IP network,（G) MSC server carries out Message Processing using the protocol integrated test system MGW of H. 248.

   3rd, the communication means according to claim 1 called in GSM/WCDMA core net circuit domains, described method is also included by using EN circles or routing server solution, replaced, using piecewise routing mode during Bearer Independent Call Control Protocol, hierarchical network being changed into planar network with end-to-end route.

   4th, the communication means according to claim 1 called in GSM/WCDMA core net circuit domains, being additionally included in increases the Codec message that extension field is used to transmit mobile calling and called terminal or WMG support in SIP-T INVITE and INFO message, to support TrFO with outer codec negotiation ability.

   5th, the communication means called in GSM/WCDMA core net circuit domains according to one of the claims, at least including one of following calling procedure：

   Calling（The calling that UE is initiated）；

   Called calling（The calling that network side is initiated）； The calling release that UE is initiated；

   The calling release that network side is initiated.

   6th, the communication means according to claim 5 called in GSM/WCDMA core net circuit domains, in addition to set up the calling procedure of UE to UE in mobile network.

   7th, UE to UE across office calling, equivalent to UE callers and the merging of called flow, and is deleted in the communication means according to claim 6 that call treatment is carried out in GSM/WCDMA core net circuit domains, the mobile network（G) MSC server entities and its corresponding SIP-T/ ISUP intercommunication translation functions.

   8th, the communication means according to claim 7 called in GSM/WCDMA core net circuit domains, the calling process is made up of following steps：

   UE initiates call request；

   MSC server respond request simultaneously constructs SIP-T INVITE messages, and INVITE message is sent into next hop address；

   GMSC servers receive the INVITE message from MSC server, and by back transfer INFO message to MSC server to set up the loading end service media stream on MG1 on terminal T2 and MG2 between terminal T3；

   MSC server returns to SIP- T AC message to GMSC servers；Calling procedure carries foundation and finished end to end.

   9. the communication means according to claim 8 that call treatment is carried out in GSM/WCDMA core net circuit domains, in addition to step

   Address message modification ISUP I AM source template of the GMSC servers in SIP- T message headers, selection office to and idle circuit, and by final IAM message by SIGTRN/SG or SS7 is sent to PSTN;

   GMSC servers receive ACM Address Complete Messages from PSTN, the message are encapsulated into SIP-T 180 ringing indication messages, and 180 message is reversely sent into MSC server；

   MSC server is received after the message, and the ACM information of encapsulation is parsed from 180 message, notifies calling UE called subscriber to have been turned on；

   GMSC servers receive ANM response messages from PSTN, and 200 0K that the message is encapsulated into SIP-T indicate message, and the 200 0K message is reversely sent into MSC server;

   MSC server is received after the message, and the A that encapsulation is parsed from 200 message draws information, notifies calling UE called subscriber to reply.

   10th, the communication means according to claim 9 called in GSM/WCDMA core net circuit domains, the calling process, which also includes MSC server, the step of domain of dependence internally fills the IAM message for turning to ISUP is called in the caller Setup call setup messages from Iu interfaces.

   11st, the communication means according to claim 9 called in GSM/WCDMA core net circuit domains, the calling process is additionally included in after the INVITE message that GMSC servers receive MSC server, it was found that wherein include ISUP IAM encapsulation, then the step of directly using the message content as the template to be sent to PSTN ISUP IAM message.

   12nd, the communication means according to claim 9 called in GSM/WCDMA core net circuit domains, the step of construction INVITE message template, fills in sip address domain including just blunt according to called number, and the MG1 SDP message for relaying lateral terminal T1 is filled in into SIP INVITE source templates.

   13rd, it is according to claim 5 to be exhaled in GSM/WCDMA core net circuit domains The communication means cried, described called calling procedure comprises the following steps：

   GMSC servers receive the ISUP IAM message from PSTN;

   GMSC server constructs SIP- T INVITE message templates, and before getting roaming number as address information to sending SIP-T INVITE messages；

   MSC server reversely passes SIP-T INFO messages back GMSC servers, the loading end service media stream between terminal T3 to set up different MG2 terminal T2 and MG1

   GMSC servers return to SIP- T ACK messages to MSC server；Called flow carries foundation and finished end to end.

   14. the communication means according to claim 13 called in GSM/WCDMA core net circuit domains, further comprising the steps of

   SC servers receive Alert ing called terminal ringing message from Iu interfaces, construction correspondence I SUP ACM message, and the message is encapsulated into S I P- T 180 ringing indication messages, and 180 message is reversely sent into GMSC servers；

   GMSC servers are received after the message of SIP- T 180 from MSC server, the ACM information of encapsulation is parsed from the message, using the message as template, it is just blunt to carry out that after necessity ^ i change, final ACM message is sent into PSTN by SIGTRAN/SG or SS7 according to 180 message header informations;

   MSC server receives connection from Iu interfaces（Connect) response message, according to message constructing ISUP answer message, and 200 0K that the message is encapsulated into SIP-T indicate message, and 200 0 message is reversely sent into GMSC servers；

   GMSC servers are received after the message of SIP- T 200 from MSC server, and the awake information of A of encapsulation is parsed from the message, using the message as template, ~ according to 200 message header informations Carry out after necessary modification, the final beautiful message of A is sent to PSTN by S IGTRAN/SG or SS7.

   15th, the communication means according to claim 14 called in GSM/WCDMA core net circuit domains, the S IP- T INVITE message templates of described construction, terminal T2 SDP message is wherein included, and Incoming side IAM message is encapsulated into message body portion.

   16th, the communication means called in GSM/WCDMA core net circuit domains described in claim 5 is occupied, the calling release process that described UE is initiated comprises the following steps：

   MSC server receives taking out stitches from UE（Di sconnec t) order；

   MSC server constructs S IP-T BYE message；

   GMSC servers, which are received, to be filled in ISUP REL message and the message is sent into PSTN by S IGTRAN/SG or SS7 after the BYE message from MSC server;

   MSC server is received after the 200 0K message from GMSC servers, and the terminal T2 of R4 core net trunk sides is discharged by issuing H. 248/MGCP order requests MG1;

   MSC server initiates release signaling flow to discharge air interface and terrestrial links resource to Iu, receives after Iu release responses, the terminal Tl of wireless access net side is discharged by issuing H. 248/MGCP order requests MG1.

   17th, the communication means according to claim 16 that call treatment is carried out in GSM/WCDMA core net circuit domains, the step of construction SIP-T BYE message, includes encapsulating I SUP REL call release messages in the message, and the opposite end address that just blunt evidence has been set up fills in BYE message headers and is sent to purpose GMSC servers.

   18th, the communication means called in GSM/WCDMA core net circuit domains described in claim 5 is just occupied, the process that described network side initiates calling release comprises the following steps： GMSC servers receive the REL from PSTN and taken out stitches order；

   GMSC server constructs SIP- T BYE message；

   MSC server fills in taking out stitches for Iu interfaces after receiving the BYE message from GMSC servers（Di s connect) message, and the message is sent to UE;

   MSC server discharges the terminal T2 of R4 core net trunk sides by issuing H. 248/MGCP order requests MG1, obtains MG1 and responds the 200 0K message that backward GMSC servers return to SIP- T, wherein encapsulating ISUP RLC message；GMSC servers are parsed after the ISUP RLC message wherein encapsulated after 200 0 message from MSC server are received, and the terminal T3 of R4 core net trunk sides is discharged by issuing H. 248/MGCP order requests MG1;

   MSC server initiates release signaling flow to discharge air interface and terrestrial links resource then to Iu, receives after Iu release responses, the terminal T 1 of wireless access net side is discharged by issuing H. 248/MGCP order requests MG1；

   19th, the communication means according to claim 18 called in GSM/WCDMA core net circuit domains, the step of described construction BYE message, includes, the ISUP REL call release messages from PSTN are encapsulated in the BYE message, and>BYE message headers are filled according to the opposite end address for the calling set up and are sent to purpose MSC server.