WO2007093116A1

WO2007093116A1 - A method and system for realizing the simulating service and the access signaling adaptive entity

Info

Publication number: WO2007093116A1
Application number: PCT/CN2007/000441
Authority: WO
Inventors: Shibi Huang; Lingzhi Mao; Peng Wang
Original assignee: Huawei Technologies Co., Ltd.
Priority date: 2006-02-15
Filing date: 2007-02-08
Publication date: 2007-08-23
Also published as: CN101022471A; CN100571299C

Abstract

A method and system for realizing the PSTN simulating service, which are used in the packet-based SIP packet network and perform PSTN service of the simulating user line signaling interaction in the call session, includes: the access signaling adaptive entity performs the convert between the simulating user line signaling and the SIP signaling to the simulating user line signaling information in the interaction of the call session process, realizes the transmission of the said simulating user line signaling information between the access signaling adaptive entity and the SIP application server by using SIP protocol, and the SIP application server performs the related processing according to the analysis of the said simulating user line signaling information to realize the PSTN simulating service at the same time. The invention satisfies the protocol that the service realization is separated from the corn control in the IMS construction, and reduces the cost of network construction, management, performance and maintenance, and has deep influence to the social economic sense.

Description

Method, system and access signaling adaptation entity for realizing simulation service

The present invention relates to a public switched telephone network (PSTN) emulation service technology, and more particularly to a method, system and access signaling for implementing a public switched telephone network emulation service for simulating subscriber line signaling interaction in a session. Adapt the entity. Background of the invention

The analog subscriber line signaling is the signaling between the analog telephone terminal and the switching equipment transmitted on the analog subscriber line. The analog subscriber line signaling mainly includes the following four types of signaling: 1) user state signaling describing the user status, The user status signaling reflects the state of the off-hook of the analog telephone terminal. The switching device can detect whether the user picks up the hook by detecting the presence or absence of the current on the subscriber line. 2) The digital signaling is dialed by the user sent by the analog telephone terminal to the switching device. The number information; 3) ringing tone and signal tone, is the signaling sent by the switching device to the analog telephone terminal to notify the user of the result of the call connection, such as a dial tone, a busy tone, a ring back tone, and the like; 4) The flashing signaling is a signaling sent by the analog telephone terminal to the switching device, and is generally used to perform analog subscriber line signaling information interaction between the user and the switching device during the triggering of the call session.

In general, the analog subscriber line signaling interaction between the analog telephone terminal and the switching device occurs during the call session setup or call session teardown. However, part of the PSTN supplementary service needs to be implemented during the call session, that is, the user's call process. Simulate user line signaling interactions. Taking the call waiting service as an example, during the conversation between the analog telephone user A and the telephone user B, if a third party telephone user C calls the analog telephone user A, the switching device will send a call waiting tone signal to the analog telephone user A to notify that there is a Three parties, The telephone user C hears the ring back tone, and the analog telephone user A can have the following three options:

1) If the incoming call of the telephone user C is rejected, the analog telephone user A does not perform any operation, and after a period of time, the switching device will stop transmitting the waiting tone signal;

2) If the telephone user B is reserved and the telephone user C is called instead, the analog telephone user A presses the R key or the flashing fork on the telephone, and then presses the number key 2 after hearing the dial tone;

3) If the call with the telephone user B ends and the call is made to the telephone user C, the analog telephone user A presses the R key or the flashing fork on the telephone, and then presses the number key 1 after hearing the dial tone.

It can be seen from the above example that during the call session between the analog telephone user A and the telephone user B, in order to implement the call waiting service, the analog telephone terminal and the switching device need to perform analog subscriber line signaling interaction, for example, the switching device sends a waiting to the analog telephone terminal. The audio signal and the analog telephone terminal send digital signals such as an R key, a numeric key 2, and a numeric key 1 to the switching device.

Other services for simulating subscriber line signaling interaction during the call session include call forwarding, three-party calling, finding malicious calls, busy callbacks, and busy parked calls, and other information about the definition and operation of these services. Refer to the relevant standards of the national communications industry, and will not go into details here.

At present, the core switching device service processing software can be regarded as a finite state automaton, which is dependent on analog subscriber line signaling and whether it is a narrowband switch device with bearer control and a softswitch device with phase separation based on bearer control. The input and output events of the network side signaling are driven to perform various PSTN services. For the service processing that needs to perform analog subscriber line signaling interaction during the call session, it is also necessary to rely on the event driver of the analog subscriber line signaling in the call session to implement the corresponding service.

With the gradual maturity of packet technology, traditional telecommunication networks based on circuit switching are developing towards packet-switched broadband telecommunication networks. Many operators have prepared or have established packet telecommunication networks, although users use packet communication terminals to access packet telecommunication networks. Future development Trends, but operators need to gradually transform the existing PSTN/Integrated Service Digital Network (ISDN) network to implement the next PSTN/ISDN network to the next generation network (NGN). Smooth evolution of the Next Generation Network), after the existing PSTN/ISDN core network is replaced by the packet telecom core network, the existing PSTN/ISDN terminal users cannot perceive the network change, and can retain the terminals and user network interfaces of the existing network. , business experience and so on. In the current research of relevant standards organizations, the application of the above packet telecommunication network to the transformation and replacement of the PSTN/ISDN core network may also be referred to as PSTN/ISDN emulation (PSTN/).

ISDN Emulation).

As a packet-switched network, the IP Multimedia Subsystem (IMS) is the subsystem of the 3rd Generation Partnership Project (3GPP) proposed in Rel 5 to support IP multimedia services. The core feature of IMS is the use of Session Initiation Protocol (SIP) and the independence of access. IMS is a multimedia control/call control platform on the packet domain. It supports both session and non-session multimedia services. Future multimedia applications provide a common business platform. Under the trend of network convergence, many international and domestic organizations are studying IMS-based network convergence solutions, with the goal of making IMS a common platform for SIP-based sessions, while supporting multiple mobile and fixed access methods. As the research on network convergence has just begun and the technology is still immature, its standardization work has become the focus of current research. In the framework of NGN, the terminal and access network are various, and the core network based on SIP session has only one IMS network, and IMS provides services for both fixed and mobile terminals. As a network with IP packet technology as the core, IMS network is a fusion scheme of network evolution and a major trend in the future development.

Since the wideband multimedia domain in NGN adopts the IMS architecture, if the PSTN/ISDN emulation subsystem (PES, PSTN/ISDN Emulation Subsystem) also adopts IMS-based Network architecture, then many network entity functions will be integrated and shared. Therefore, relevant standard projects have been established in the International Telecommunication Union-Telecommunication Standards Department (ITU-T) and the European Telecommunications Standards Institute (ETSI) organization. research work. The functional structure of the IMS-based PSTN/ISDN emulation subsystem is given in the draft ETSI standard TS 02030 VI .2.7. Figure 1 is a schematic diagram of the IMS-based PES functional architecture. As shown in Figure 1, the PES architecture uses the access gateway control. The access signaling adaptation entity such as the function (AGCF, Access Gateway Control Function) and the media gateway (MG, Media Gateway) implements the access adaptation of the traditional PSTN terminal to the IMS network. According to the standard draft TS 02030 V1.2.7, the access signaling adaptation entity such as AGCF can complete the conversion between analog subscriber line signaling and SIP signaling, but does not implement PSTN service logic, and moves PSTN service logic control to IMS. The application server (AS) of the network, such as the PES AS, that is, the IMS-based PES architecture should conform to the principle of separation of service implementation and core control. In addition, the ETSI draft standard TS 3044 also gives some specific process definitions for implementing PSTN simulation services based on IMS. For the specific research results of the IMS-based PSTN simulation subsystem in the ETSI standard research, please refer to the above two draft standards, which will not be described in detail in this paper.

Currently, in ETSI TS 3044 V0.1.0, the specific process definition of PSTN emulation service based on IMS gives the conversion rules between part of analog subscriber line signaling and SIP signaling during call session establishment or call session demolition, such as After the user picks up the phone and dials the called number, the AGCF determines the number termination according to the configuration data, and converts the analog subscriber line signaling such as the user's off-hook and called number into a SIP session initial request (INVITE) message, and the SIP INVITE message is sent to The PES AS that performs PSTN service logic control. After the AGCF implements the corresponding analog subscriber line signaling and SIP signaling conversion, although the expression of the analog subscriber line signaling changes in the SIP protocol, the information that the original analog subscriber line signaling needs to transmit may be The conversion is obtained in the SIP protocol, so the information required to simulate the subscriber line signaling is not lost, so that the PES AS can be based on the SIP signaling. The simulated subscriber line signaling information carried carries out the corresponding PSTN service logic control.

In ETSI TS 3044 V0.1.0, the analog subscriber line signaling that interacts during the call session cannot be implemented and simulated like other analog subscriber line signaling that occurs during call session setup or call session teardown. The conversion between the AGCF and the PES AS is passed through the SIP protocol. In the existing process, the analog subscriber line signaling for the interaction during the call session is interpreted and processed in the AGCF, and is not converted to the SIP protocol and sent to the PES AS. For example, for the service of the flashing fork, it is detected by the AGCF. After the user's flashing fork simulates the subscriber line signaling, according to the state of the existing call in the AGCF, the meaning expressed by the service code information of the flashing signal and the subsequent call is explained, and the call session process such as three-party call and call waiting is completed. The business logic control and media control processing of the analog subscriber line signaling interaction service are required. In the existing solution, for a service involving a multi-party session such as call waiting, the AGCF needs to maintain multiple SIP session states, for example, the AGCF needs to determine which session is currently activated according to the simulated subscriber line signaling, and which session is currently maintained and other session state information; For a service involving multiple session media, such as a three-party call service, the AGCF also needs to control the MG to implement the three-party media conference bridge connection control.

In summary, the processing method of the analog subscriber line signaling that is exchanged during the call session given in ETSI TS 3044 V0.1.0, for the analog subscriber line signaling during the call session, is through the access signaling adaptation entity. To be interpreted and processed, it is not converted to the SIP protocol and sent to the PES AS. As a result, the services that rely on these analog subscriber line signaling can only be placed on the access signaling adaptation entity and cannot be moved up to the PES AS. achieve. The design principles of the IMS-based PES architecture are not fully compliant, causing the following problems:

1) Some PSTN service logic control needs to be implemented in the AGCF, which violates the design principle of the IMS-based PES architecture in TS 02030 V1.2.7 and moves to the PES AS.

2) A large number of industries involved in the analog subscriber line signaling interaction during the call session It must be implemented on the AGCF, which makes the control software on the AGCF more complicated.

For example, in the AGCF, the crossover analog subscriber line signaling during the session is terminated, so that all services related to the analog subscriber line signaling must be implemented on the AGCF, while ETSI TS 3044 V0.1.0 only gives call waiting. Service descriptions such as three-party calling, and other services that rely on the analog-to-clip analog subscriber line signaling, such as call forwarding, finding malicious calls, busy callbacks, busy parked calls, etc., cannot be obtained by PES AS. User line signaling information, these services can only be implemented on the AGCF. In this way, the AGCF needs to perform a large number of services simulating subscriber line signaling interaction during the implementation of the call session; for services involving multiple parties, the AGCF also needs to maintain multiple SIP session states; the AGCF also controls the MG to complete multiple session media. The need for media control and the like of the service increases the complexity of the control software on the AGCF.

3) The management and maintenance of the PES service is complicated.

From the perspective of the existing solution, some PSTN services are implemented in the PES AS, and some PSTN services are implemented in the access signaling adaptation entity. This distributed service implementation method makes it difficult to solve the problems of cooperation and conflict between services. Moreover, service data configuration needs to be performed in the access signaling adaptation entity such as the AGCF, which results in complicated management and maintenance of services. Summary of the invention

In view of this, the main object of the present invention is to provide a method for realizing a public telephone switching network emulation service, which can conform to the principle of separation of service implementation and core control in an IMS-based PES architecture.

Another object of the present invention is to provide a system for implementing a public switched telephone network emulation service that is capable of complying with the principle of separation of service implementation and core control in an IMS-based PES architecture.

It is still another object of the present invention to provide an access signaling adaptation entity that can be based on The principle of separation of service implementation and core control in the PES architecture of IMS.

In order to achieve the above object, the technical solution of the present invention is specifically implemented as follows:

A method for implementing a public telephone switching network PSTN emulation service, the method comprising: in a session initial protocol SIP packet network, an access signaling adaptation entity performs an analog subscriber line for an analog subscriber line signaling information exchanged during a call session Transmitting between signaling and SIP signaling, implementing the transmission of the analog subscriber line signaling information between the access signaling adaptation entity and the SIP application server by using a SIP protocol; and simultaneously simulating subscriber line signaling according to the SIP application server The information is processed accordingly to implement the PSTN simulation service.

The method for performing the conversion between the analog subscriber line signaling and the SIP signaling by the access signaling adaptation entity is: carrying the analog subscriber line signaling information in different SIP messages and transmitting the information to the SIP application server, or The one or more analog subscriber line signaling information is carried in the same SIP message and transmitted to the SIP application server.

And the method for the access signaling adaptation entity to perform the conversion between the analog subscriber line signaling and the SIP signaling is: the access signaling adaptation entity parses the function information corresponding to the analog subscriber line signaling information, and then The parsed function information is delivered to the SIP application server through the SIP protocol.

The method for the access signaling adaptation entity to perform the conversion between the analog subscriber line signaling and the SIP signaling is:

The SIP subscription and response mechanism is used, and the user-key digital signaling information in the analog subscriber line signaling information is transmitted to the SIP application server by using the key markup language KPML.

The SIP signaling is a SIP Info message, or a SIP Message message, or a SIP Invite message.

The SIP signaling includes a message body of a MIME media type carrying the analog subscriber line signaling information, where the MIME media type includes: A MIME media type field for identifying a MIME media type category; an encoding mode field of an encoding method used to identify the analog subscriber line signaling information carried in the message body of the MIME media subtype.

The value of the MIME media type field is an application application; the value of the MIME media subtype field is analog subscriber line signaling analog-subsci, iber-signal.

The coding mode field takes the value of the text extension Bacchus paradigm ABNF mode, or the extended markup language XML mode.

The analog subscriber line signaling includes: user tap signal signaling, and/or digital signaling, and/or ringing, and/or tone information, and/or billing pulse signaling, and/or reverse polarity Signaling, and/or frequency shift keying signaling.

The SIP packet network is an IP Multimedia Subsystem IMS;

The access signaling adaptation entity is an access gateway control function AGCF;

The SIP application server is a PSTN emulation service application server PES AS.

The access signaling adaptation entity is an integrated access device with an access signaling adaptation and an access bearer adaptation function.

The session process includes a first user and a second user, and the PSTN service is a call waiting emulation service, and the specific implementation of the method includes:

After receiving the request from the third user to call the first user, the PES AS carries the call waiting tone in the SIP signaling and sends the call waiting tone to the AGCF, where the AGCF controls the connection in the SIP packet network. The ingress gateway plays a waiting tone to the first user;

The AGCF carries the information of the flashing signal from the first user to the PES AS, and the PES AS carries the dial tone in the SIP signaling and sends the information to the AGCF.

The AGCF controls the access gateway in the SIP packet network to disconnect the first user Connect to the second user's media, and send a dial tone to the first user; the ACGF carries the service code information from the first user in the SIP signaling and sends the information to the PES AS; The AS parses the received service code information to learn that the current service is a call waiting emulation service and the second user is a hold party;

Transmitting, by the media resource server in the SIP packet network, the PSE to the sitter, and updating, by the AGCF, a session description protocol SDP between the first user and a third user, where the AGCF uses an update. The latter SDP establishes a media path between the first user and the third user.

After receiving the request from the third user to call the first user, the PES AS carries the call waiting tone in the SIP signaling and sends the call waiting tone to the AGCF, where the AGCF controls the connection in the SIP packet network. The incoming gateway plays a waiting tone to the first user;

After receiving the flashing signal from the first user, the AGCF controls an access gateway in the SIP packet network to disconnect the media connection of the first user and the second user, and sends the media connection to the first user Send a dial tone;

The ACGF carries the service code information and the flashing signal information from the first user in the SIP signaling and sends the information to the PES AS. The PES AS parses the received information to learn that the current service is a call waiting emulation. The service and the second user is a hold party;

The session process includes a first user and a second user, and the PSTN service is a three-party call emulation service, and the specific implementation of the method includes: After receiving the flashing signal from the first user, the AGCF controls an access gateway in the SIP packet network to disconnect the media connection of the first user and the second user, and sends the media connection to the first user Sending a dial tone; the first user dialing a number of the third user;

The ACGF carries the third user number information and the flashing signal information from the first user to the PES AS in the SIP signaling; the PES AS parses the received message to learn that the current service is Three-party call simulation service;

The PES AS and the AGCF update a session description protocol SDP between the first user and the third user, and the AGCF establishes a media path between the first user and the third user by using the updated SDP.

After the PES AS parses the received information and learns that the current service is a three-party call emulation service, before the PES AS and the AGCF update the session description protocol SDP between the first user and the third user, the method further includes:

The P£S AS changes the second user to receive only the non-transmitted media stream through the SIP protocol.

The session process includes a first user and a second user, and the PSTN service is an incoming transfer simulation service, and the specific implementation of the method includes:

After the AGCF receives the flashing signal from the first user, controlling an access gateway in the SIP packet network to disconnect the media connection of the first user and the second user, and to the first The user sends a dial tone;

The ACGF carries the service code from the first user and the information of the flashing signal in the SIP signaling to the PES AS; the PES AS parses the received information to learn that the current service is an incoming call. Simulating the service and determining that the transfer party is a third user;

The PES AS negotiates a session description protocol SDP between the second user and the third user, and establishes a media path between the second user and the third user by using the updated SDP; and simultaneously releases the first user Conversation. The first user is an analog user.

A system for implementing a public switched telephone network PSTN emulation service, in a session initial protocol SIP packet network, the system at least includes: an access signaling adaptation entity, a SIP application server;

The access signaling adaptation entity further includes: a conversion unit that performs conversion between the analog subscriber line signaling and the SIP signaling for the simulated user line signaling information that is exchanged during the call session;

Transmitting the analog subscriber line signaling information by using the SIP protocol between the converting unit and the SIP application server; the SIP application server performs corresponding processing according to the simulated subscriber line signaling information to implement a PSTN emulation service.

The converting unit respectively carries the received analog subscriber line signaling information in different

The SIP message is delivered to the SIP application server, or more than one analog subscriber line signaling information is carried in the same SIP message and transmitted to the SIP application server.

The converting unit parses the function information corresponding to the analog subscriber line signaling information, and then transmits the parsed function information to the SIP application server by using a SIP protocol.

The conversion unit uses SIP's subscription and response mechanism and applies a key markup language

KPML delivers the user button digital signaling information in the analog subscriber line signaling information to the fans

SIP application server.

The SIP packet network is an IP Multimedia Subsystem IMS;

An access signaling adaptation entity, where the access signaling adaptation entity includes: a conversion unit that simulates conversion of subscriber line signaling and SIP signaling between simulated subscriber line signaling information during a call session .

As can be seen from the above technical solution, the present invention provides a SIP application server such as PES.

On the AS, PSTN imitation of analog subscriber line signaling interaction is required during the call session. The method of the real service, the method of the invention conforms to the principle of separating the service implementation and the core control in the IMS architecture. The IMS-based architecture can realize the 100% inheritance of the PSTN service by using the IMS-based architecture, and the unified method is adopted. The IMS core network provides services for PSTN emulation users and IP multimedia users, which reduces the network construction cost and management operation and maintenance cost of operators, and has far-reaching social and economic significance.

At the same time, compared with the existing mode in which the PSTN emulation service is implemented locally in the AGCF, the method and system for providing the PSTN emulation service by the POST AS of the service provider are provided by the access operator's AGCF. The PES AS in the home domain implements the PSTN emulation service, supports the service mobility, and implements the centralized and unified PSTN emulation service on the PES AS in the home domain, which conveniently solves the equal access, billing, and service of the PSTN emulation service. Issues such as conflicts and service delivery simplify the management and operation and maintenance of the PS N simulation service. BRIEF DESCRIPTION OF THE DRAWINGS

Figure 1 is a schematic diagram of a IMS-based PES functional architecture;

2 is a schematic diagram of a related entity connection of an analog telephone terminal using an access signaling adaptation entity to access a SIP-based packet network;

3 is a flow chart of an embodiment of the call waiting emulation service of the present invention;

4 is a flowchart of an embodiment of the call waiting emulation service optimization process of the present invention; FIG. 5 is a flowchart of an embodiment of the three-party call emulation service of the present invention;

6 is a flow chart of an embodiment of the incoming call emulation service of the present invention. Mode for Carrying Out the Invention The core idea of the present invention is: In a packet-based SIP packet network, for a PSTN service that performs analog subscriber line signaling interaction in a call session, the access signaling adaptation entity pairs the call. The simulated subscriber line signaling information that is exchanged during the session is used to simulate the conversion between the subscriber line signaling and the SIP signaling, and the analog subscriber line signaling information is implemented between the access signaling adaptation entity and the SIP application server by using the SIP protocol. The delivery is performed by the SIP application server according to the analog subscriber line signaling information, thereby implementing the PSTN simulation service.

The present invention will be further described in detail below with reference to the accompanying drawings and preferred embodiments.

The present invention is applicable to an application scenario in which an analog telephone terminal uses an access signaling adaptation entity to access a SIP-based packet network, and implements a PSTN emulation service in which a simulated user signaling interaction between a user and a network is performed in a session, such as an IMS-based PSTN. The simulation service and the application scenario in which the SIP integrated access device is used to access the analog telephone terminal and implement the PSTN simulation service on the SIP application server. For convenience of explanation, FIG. 1 is simplified to FIG. 2, which is a schematic diagram of an associated physical connection of an analog telephone terminal using an access signaling adaptation entity to access a SIP-based packet network.

As shown in FIG. 2, the analog telephone terminal is a telephone terminal that accesses a SIP-based packet network such as an IMS-based PES; it should be noted that at least one party in the call session process of the present invention is an analog telephone terminal. And performing PSTN services simulating subscriber line signaling interaction during the call session.

The access signaling adaptation entity is a network side entity that performs functions such as conversion between the analog subscriber line signaling and the SIP signaling. In an actual application, for example, in an IMS-based PES, the access signaling adaptation entity function may be It is an AGCF in the IMS-based PES functional architecture; it may also be an integrated access device in which the access signaling adaptation and the access bearer adaptation function are combined;

The PES AS is a functional entity that performs PSTN service logic control and implements PSTN supplementary services.

The L interface between the analog telephone terminal and the access signaling adaptation entity is an interface for transmitting analog subscriber line signaling, such as in an IMS-based PES, analog telephone terminal and access. The signaling adaptation entity can transmit analog subscriber line signaling through the MG shown in FIG. 1, and the media gateway control protocol such as H.248 can be applied between the MG and the access signaling adaptation entity; The I interface between the adaptation entity and the PES AS can use the SIP signaling protocol. For example, in the IMS-based PES, the access signaling adaptation entity and the PES AS can pass other network entities in the IMS core network, such as A network entity such as an S-CSCF or an I-CSCF transmits SIP signaling.

In the present invention, the analog subscriber line signaling that is exchanged during the call session is converted into SIP signaling, and is transmitted between the access signaling adaptation entity and the PES AS through the I interface, so that the PES AS obtains sufficient analog subscriber line. Signaling information and corresponding service processing conform to the principle of separation of service implementation and core control of IMS-based PES architecture. Specifically, the access signaling adaptation entity of the present invention further includes: a conversion unit that simulates conversion of the subscriber line signaling and the SIP signaling between the simulated subscriber line signaling information during the call session; The transmitting of the analog subscriber line signaling information is implemented by the SIP protocol between the unit and the PES AS. The PES AS performs corresponding processing according to the analog subscriber line signaling information to implement the PSTN simulation service.

The conversion unit implementation converts to:

The conversion unit respectively carries the received analog subscriber line signaling information to the SIP application server in different SIP messages, or carries one or more analog subscriber line signaling information in the same SIP message and transmits the information to the PEA AS;

Alternatively, the converting unit parses the function information corresponding to the received analog subscriber line signaling information, and then transmits the parsed function information to the PES AS through the SIP protocol;

Alternatively, the conversion unit uses the SIP subscription and response mechanism and applies the key markup language KPML to deliver the user button digital signaling information in the analog subscriber line signaling information to the PES AS.

In the prior art, analog subscriber line signaling in the call origination or call termination phase is conveniently Converted to SIP messages with similar meanings, for example, the on-hook user signaling and the BYE message in the SIP signaling all indicate the meaning of the end of the call, so the on-hook analog subscriber line signaling at the end of the call can be easily converted into SIP signaling. BYE message; The analog subscriber line signaling interacting during the call session is difficult to directly convert to the SIP message with corresponding meaning, and the appropriate SIP message is needed to transmit the simulated subscriber line signaling information during the call session. After the call session is established, according to the SIP protocol specification, a message in the SIP protocol, such as a SIP Info message, may be used to deliver the interaction information in the session. Therefore, the present invention may use the SIP Info message to deliver the interaction simulation during the call session. User line signaling information; In addition, the present invention may also use other SIP messages, such as SIP Message, Invite, etc., to communicate the simulated subscriber line signaling information during the call session, but only the SIP session identifier of the Messgae or Invite message at this time. The SIP session identifier of the established call session is different, and the Message or Invite message is required to carry other information to associate the two different SIP session identifiers.

In order to implement the simulated subscriber line signaling information that is exchanged during the call session in SIP Info, Message, Invite, etc., the format of the analog subscriber line signaling information in the SIP message must be set so that both parties can correctly understand Corresponding information. The analog subscriber line signaling interacting during the call session involves signaling such as cross-cutting signaling, digital signaling, ringing, and tone in the analog subscriber line signaling, such as the service code dialed by the user after the flashing fork or the flashing fork. , dial tone, busy tone, call waiting tone, ringing signal, and more. In order to

The SIP message transmits the simulated subscriber line signaling information during the call session. The present invention sets a new multi-purpose Internet Mail Extensions (MIME) media type, that is, analog subscriber line signaling.

( analog-subscriber-signal ) MIME media type, this new analog subscriber line signaling MIME media type includes the following fields:

MIME media type: used to identify MIME media types The field of the category, in the present invention, the value is an application (application);

MIME subtype: A field used to identify a subtype of a MIME media type. The subtype of the MIME media type is analog-subscriber-signal used in the present invention to simulate a subscriber line signaling media subtype;

Required parameters: This field is not used in the present invention; Optional parameters: This field is not used in the present invention; Encoding scheme: used to identify the MIME media subtype in the message body The coding mode of the analog subscriber line signaling information carried may be an ABNF (Augmented Backus-Naur Form) method or an extended markup language XML method;

The analog subscriber line signaling information carried in the message body of the MIME media subtype may include but is not limited to the following information:

a. The user shoots the signaling information;

b. Digital signaling information simulating subscriber line signaling: including number information dialed by the user;

c ringing stream and tone information: used for ringing and tone information delivered by the PES AS to the access signaling adaptation entity; information for notifying the user of the result of the call connection, such as dial tone, busy tone, ring back tone Isophonic signal and ringing signal;

d. billing pulse signaling information;

e. frequency shift keying signaling information;

f. Reverse polarity signaling information.

According to the analog-subscriber-signal MIME media type set by the above invention, the SIP message may include a message body of the analog-subscriber-signal MIME media type, for example, the content of the SIP message body carrying the information of the caller's subscriber line signaling may be expressed as follows:

Content-Type: application/analog-subscriber-signal Hook: hook-flash

Wherein, the Content-Type: application/analog-subscriber-signal indicates that the information carried in the message body belongs to the MIME media type, and the MIME media subtype of the MIME media type is analog-subscriber-signal MIME media type;

The hook-flash indicates that the information carried in the message body is the flashing signaling information. It should be noted that this is just an example. Other characters can also be used to indicate the flashing fork, as long as the user negotiates well.

The following is an example of call waiting, three-way calling, and incoming call. The following describes the analog subscriber line signaling information that is exchanged between the access signaling adaptation entity and the PES AS by using the SIP protocol to implement PSTN simulation. Business approach.

3 is a flowchart of an embodiment of the call waiting emulation service of the present invention. In FIG. 3, it is assumed that the access signaling adaptation entity is an AGCF, and the analog subscriber line signaling is used in the process of transmitting a call session between the AGCF and the PES AS by using the SIP protocol. The interactive user line signaling information is simulated, and the simulated user A is an analog telephone terminal, and the user B and the user C are SIP terminals. The method for implementing the call waiting simulation service of the present invention includes the following steps:

Step 300 to step 306: A basic session call is established between the simulated user A and the user B.

After receiving the dialing information from the simulated user A, the AGCF sends a session initial request (Invite) SIP message of the user B corresponding to the dialing information to the PES AS; the PES AS forwards the received Invite B SIP message to the user B, and the user B will 180 Ringing SIP ringing signaling is sent to the AGCF via the PES AS; after receiving the 180 Ringing signaling, the AGCF controls the access gateway to send a ringback tone to the analog user, and receives 200 from the user B forwarded via the PES AS. After the OK SIP acknowledges the signaling, the ACK SIP response message is sent to the user B via the PES AS, and the user A and the user B are simulated to communicate.

In this step, according to the existing IMS-based PSTN simulation technology, the AGCF performs conversion between the analog subscriber line signaling and the SIP signaling, and the specific implementation can be referred to the relevant draft standard. I won't go into details here.

Step 307 ~ Step 308: During the call between the user A and the user B, the PES AS receives the call initial request (Invite) from the call of the user C, and the PES AS determines that the session between the simulated user A and the user B already exists. It is assumed that the analog subscriber A subscribes to the call waiting service according to the pre-configured service data, and the PES AS transmits the call waiting tone analog subscriber line signaling information to the AGCF through the SIP Info message.

In this step, the SIP Info message carries a message body indicating the call waiting tone to simulate the subscriber line signaling information, which can be expressed as follows:

Content-Type: application/analog-subscriber-signal

Tone: call-waiting

The tone: call-waiting indicates that the information carried in the message body is call waiting tone information.

Step 309: Step 310: The PES AS returns a 182 SIP message indicating that the call is being queued to the user C. After receiving the SIP Info message from the PES AS carrying the call waiting tone analog subscriber line signaling information, the AGCF controls the access gateway to the analog user. A inserts a call waiting tone to simulate a subscriber line signal.

Step 311: Simulate the user A flashing fork and send it to the AGCF through analog subscriber line signaling.

Step 312: The AGCF transmits the flashing analog user line signaling information to the PES AS through the SIP Info message.

In this step, the SIP Info message carries a message body indicating the information of the analog-to-be-simulated subscriber line signaling, which can be expressed as follows:

Content-Type: application/analog-subscriber-signal

Hook: hook-flash

The hook: hook-flas indicates that the information carried in the message body is the flash information. Step 313: The PES AS carries the dial tone analog subscriber line signaling information in the SIP Info message and sends it to the AGCF.

Step 314 to step 315: After receiving the SIP Info message from the PES AS carrying the dial tone analog subscriber line signaling information, the AGCF controls the access gateway to disconnect the media connection between the simulated user A and the user B, and sends the media connection to the simulated user A. Dial tone simulates subscriber line signaling.

In this step, the specific implementation of the process of controlling the access gateway to disconnect the media connection between the simulated user A and the user B is well known to those skilled in the art, and may be referred to the relevant standards, and details are not described herein again.

Step 316: The analog user A dials a service code such as the number key 2, and sends it to the AGCF through analog subscriber line signaling.

Step 317: The AGCF carries the received service line analog subscriber line signaling in the SIP Info message, and transmits the service code dialed by the simulated user A to the PES AS to simulate the subscriber line signaling information. The PES AS receives the analog subscriber A dialed. After the service code information is analyzed, it is learned that the simulated user A chooses to keep the user B and connects the call waiting user C.

Step 318 ~ Step 323: The PES AS applies for the corresponding voice resource to the Media Resource Server (MRS) and plays the voice to the hold party, User B.

The MRS is a physical implementation of the media resource control function (MR C ) and the media resource processing function (MRFP ) in FIG. 1 , and is used to provide media resources such as playback and conference bridge control for the network. In the prior art, the MRS product of the media resource is provided. In the present invention, the MRS can be separated into the control layer MRFC entity and the media layer MRPP entity. In this embodiment, only a service implementation example is used, which indicates that the media resource is used for the call holder. Waiting for music. In actual implementation, this step is optional because the holder knows that it is to be held, such as a voice prompt during a call, so that no prompt tone is needed. The use of MRS is not relevant to the present invention, but is a specific embodiment given for business process integrity.

Step 324 ~ Step 325: PES AS uses simulated user A and user B to establish a meeting. The session description protocol (SDP) information of the simulated user A negotiated, and returns a 200 OK message to the user C to accept the call; after receiving the 200 OK message, the user C returns an acknowledgement ACK message.

Steps 326 to 329: The PES AS initiates a SDP update to the user A by using the SDP information of the user C that is carried in the Invite A SIP message of the user A in step 307, and the AGCF initiates the SDP update. According to the updated SDP, the media stream connection control of the IP media endpoint of the user A is simulated, and a connection is established between the IP media endpoint of the simulated user A and the IP media endpoint of the user C, thereby connecting the Han direction of the simulated user A and the user C. Media access.

In the process shown in Figure 3, the AGCF converts each analog subscriber line signaling information that is exchanged during the call session into a SIP Info message and then passes it to the PES AS for processing. In order to reduce the transmission of the analog subscriber line between the AGCF and the PES AS. The number of interactions of the command, without affecting the service logic control processing of the PES AS, the AGCF can optimize the transmission of the analog subscriber line signaling information, that is, collect multiple analog subscriber line signaling from the analog user terminal. Then, it is sent to the PES AS at one time through the SIP Info message. 4 is a flowchart of an embodiment of the call waiting emulation service optimization process of the present invention. As shown in FIG. 4, the analog user of the beat-to-cross analog subscriber line signaling in step 311 and the service code dialed by the simulated user A in step 316 are simulated. The line signaling information is carried in a SIP message and sent to the PES AS for processing. '

It should be noted that, for the AGCF to implement the above optimization process, the AGCF needs to have the ability to determine when the service code dialed by the user is terminated. The implementation of the capability can be determined by presetting the dialing rule of the service code on the AGCF. When to terminate.

Steps 411 to 416 in FIG. 4 implement the optimization processing of steps 311 to 317 in FIG. 3, and assume that the dialing rule of the service code is preset in the AGCF. Said as follows:

Step 411: Simulate the user A flashing fork and send it to the AGCF through analog subscriber line signaling.

Step 412 ~ Step 413: After receiving the analog subscriber line signaling of the flashing fork, the AGCF controls the access gateway to disconnect the media connection between the simulated user A and the user B, and sends a dial tone analog subscriber line signaling to the analog user A;

Step 414: The analog user A dials a service code such as the number key 2, and sends it to the AGCF through analog subscriber line signaling.

Step 415 to step 416: After the AGCF determines that the service code is dialed according to the preset service code dialing rule, the AGCF carries the service code analog subscriber line signaling information dialed by the flashing and analog user A in the SIP Info message, to the PES. The AS passes the analog user A's flashing fork and the dialed service code to simulate the subscriber line signaling information. After receiving the flashing fork and the dialed service code information, the PES AS analyzes and knows that the simulated user A chooses to keep the subscriber B, and connects the call waiting. User C.

Steps 400 to 410 in FIG. 4 are completely consistent with the implementations of steps 300 to 310 in FIG. 3. The subsequent steps after step 416 in FIG. 4 are completely identical to steps 318 to 329 in FIG. Retelling.

This embodiment emphasizes that multiple analog subscriber line signaling from the analog user terminal is sent to the PES AS through the same SIP Info message, which reduces the number of interactions between the AGCF and the PES AS to transmit analog subscriber line signaling.

5 is a flowchart of an embodiment of the three-party call emulation service according to the present invention. In FIG. 5, it is assumed that the access signaling adaptation entity is an AGCF, and the analog subscriber line signaling is used in the process of transmitting a call session between the AGCF and the PES AS by using the SIP protocol. The interactive user line signaling information is simulated, and it is assumed that the simulated user A is an analog telephone terminal, the user B and the user C are SIP terminals, and the simulated user A and the user B are in communication, and the present invention implements a three-party calling simulation. The method of true business includes the following steps:

Step 500: Simulate the user A flashing fork and pass it to the AGCF through analog subscriber line signaling.

Step 501 to step 502: After receiving the analog subscriber line signaling of the flashing fork, the AGCF controls the access gateway to disconnect the media connection between A and B, and sends a dial tone analog user line signaling to the user A.

Step 503: The simulated user A dials the number of the third party, that is, the user C, and transmits the number information of the user C to the AGCF through the analog subscriber line signaling.

Step 504 to step 505: After the AGCF determines that the user dials the terminal according to the preset dialing rule, the AGCF carries the analog subscriber line signaling information of the number of the tap and the dialed user C in the SIP Info message, and sends the information to the PES AS. After receiving the number information of the flashing fork and the user C, the PES AS analyzes and knows that the simulated user A needs to keep the user B and connects the user C.

Step 506 to step 508: The PES AS changes the IP media endpoint of the user B to receive and not send the IP address through the SIP protocol. The specific implementation is known to those skilled in the art. For details, refer to related protocols, and details are not described herein.

It should be noted that, in the actual implementation, this step is optional. If this step is omitted, User B may also send the media stream, but since User B has been held by the call, the simulated user A is also unable to receive the media from User B. Streaming, this will result in a waste of IP network bandwidth.

Step 509 - Step 510: The PES AS initiates a call session request to the user C using the SDP of the simulated user A; the user C returns 180 Ringing SIP ringing signaling to the PES AS.

Step butterfly 511 ~ Step 513: The PES AS changes the media connection mode of the IP media endpoint of the simulated user A to receive only the IP media endpoint from the user C according to the SDP description of the IP media endpoint of the user C carried in the SIP ringing signaling. Media stream. Step 515: The simulated user A receives the ring back tone signaling from the user C.

Step 516 ~ Step 517: User C returns a 200 OK SIP response signal to the PES AS to answer the call, and the PES AS returns a response ACK message to User C.

Steps 518 to 521: The PES AS changes the media connection mode of the IP media endpoint of the simulated user A according to the SDP description of the IP media endpoint of the user C carried in the SIP ringing signaling of the user C in the steps 511 to 513. The media stream from the IP media endpoint of user C is sent and received, and the AGCF controls the access gateway to implement corresponding media connection control, thereby connecting the two-way media path simulating user A to user C.

6 is a flowchart of an embodiment of an incoming call emulation service according to the present invention. In FIG. 6, it is assumed that an access signaling adaptation entity is an AGCF, and analog subscriber line signaling uses a SIP protocol to transfer a call session between an AGCF and a PES AS. The simulated subscriber line signaling information is interactively in the process, and assumes that the simulated user A is an analog telephone terminal, the user B and the user C are SIP terminals, and the simulated user A and the user B are in communication, and the present invention implements the incoming call. The method of connecting the simulation service includes the following steps:

Step 600: Simulate the user A flashing fork and pass it to the AGCF through analog subscriber line signaling.

Step 601 ~ Step 602: After receiving the caller line signaling, the AGCF controls the access gateway to disconnect the media connection between the simulated user A and the user B, and sends a dial tone analog subscriber line signaling to the analog user A.

Step 603: The simulated user A dials a service code, such as *12*user C's number #, and transmits it to the AGCF through analog subscriber line signaling.

Step 604 ~ Step 605: After the AGCF learns that the user dials the terminal according to the preset dialing rule, the AGCF carries the received flashing fork and the dialed *12* user C number #impersonized subscriber line signaling information in the SIP Info message. And sent to the PES AS, the PES AS receives the number of the fork and the number of the * 12* user C dialed, and then analyzes and learns the simulation. User A chooses to transfer the user B call to the caller, User C.

This step can also be implemented as follows: The AGCF first parses the specific analog subscriber line signaling meaning, and then transmits the function information of the transfer operation to the PES AS through a protocol such as SIP, where the number of the user C can be used as a parameter of the transfer operation. For example, in the SIP protocol, an operation field is extended to carry a handover operation, and another number field is extended to carry the number of the user C. After the PES AS receives the SIP message carrying the parameters of the handover operation and the corresponding number of the transferred user, The meaning of the operation field and the corresponding parameters implement business logic control.

Step 606: The PES AS initiates a call session request Invite SIP message to the user C by using the SDP description of the IP media endpoint of the user B.

Step 607: User C returns 180 Ringing signaling to the PES AS, and carries the SDP description of the IP media endpoint of User C in the ringing signaling.

Step 608 to step 610: The PES AS changes the media connection mode of the IP media endpoint of the user B to simultaneously send and receive the media stream from the IP media endpoint of the user C according to the SDP description of the IP media endpoint of the user C carried in the 180 Ringing. .

Step 611 ~ Step 612: User C returns a 200 OK response response to the PES AS. After receiving the 200 OK, the PES AS returns an ACK confirmation message to User C.

So far under the control of the PES AS, User B and User C have established a two-way media path. Step 613 to Step 614: After the PES AS transfers the user B call to the user C, the SIP Bye process is used to release the session simulating the user A.

It should be noted that, in this paper, only the application flow of the PSTN simulation service that needs to perform the analog subscriber line signaling interaction during the typical call session is not limited, and the method of the present invention is not limited to the services, and the method of the present invention. The same can be applied to other services that need to perform analog subscriber line signaling interaction during the call session. This is an example of implementing the access signaling adaptation entity on the AGCF in the IMS-based PES functional architecture. The method of the invention is only applicable to such a network architecture, and the method of the invention is equally suitable Other network functional architectures for conforming to the logical architecture shown in FIG. 2, such as using a SIP integrated access device that implements the function of the access signaling adaptation entity to access the analog telephone terminal and implement PSTN service emulation on the SIP application server Application scenarios and the like; the flowchart illustrations and text descriptions made in the present invention are merely illustrative of the key techniques of the present invention, and do not represent a complete call and service control process, nor do they exhaust all possible branching processes; The format and name of the event packet carried by the SIP message only highlights the analog user signaling information that it must carry, and does not mean that this is the only way to describe it.

In summary, the solution of the present invention implements a PSTN emulation service that requires user line signaling interaction during a call session on an application server of an IMS network, and conforms to the principle of separation of service implementation and core control in the IMS architecture; The invention can combine the existing technology to realize 100% inheritance of the PSTN service by using the IMS-based architecture, and provide services for the PSTN emulation user and the IP multimedia user by using the unified IMS core network, thereby reducing the network construction cost and management operation and maintenance of the operator. Cost has far-reaching socio-economic significance.

At the same time, the solution of the present invention provides access to the AGCF of the access operator but provides the PSTN emulation service by the PES AS of the service provider, which ensures that the PSTN of the home domain implements the PSTN emulation service, supports the service mobility, and implements The PSTN emulation service is implemented centrally and uniformly in the PES AS of the home domain, which conveniently solves the problems of equal access, service conflict, and service provision of the PSTN emulation service, and manages the management and operation of the PSTN emulation service by the operator. dimension.

In addition, a SIP event packet draft of the IETF key interaction can also be applied between the access signaling adaptation entity and the PES AS ( draft-ietf-sipping-kpml: A Session Initiation Protocol (SIP) Event Package for Key Press Stimulus The SIP subscription and response mechanism is given, and the key press markup language (KPML, Key Press Markup Language) is used to transfer the number of user buttons in the analog subscriber line signaling information. Word signaling information. The specific implementation includes: the PES AS uses the SIP protocol subscription (SUBSCRIBE) message to carry the analog user button event subscription to the access signaling adaptation entity by using the dialing rule described by the KPML, and the access signaling adaptation entity detects the user button information. If the user case information meets the dialing rule of the PES AS subscription, the access signaling adaptation entity sends a SIP notification (Notify) message to the PES AS to simulate the number information dialed by the user.

The above is only the preferred embodiment of the present invention, and is not intended to limit the scope of the present invention. Any modifications, equivalents, improvements, etc., which are made within the spirit and principles of the present invention, should be included. It is within the scope of the invention.

Claims

Claim

A method for implementing a PSTN emulation service of a public switched telephone network, the method comprising:

In the session initiation protocol SIP packet network, the access signaling adaptation entity performs the conversion between the simulated subscriber line signaling and the SIP signaling for the simulated subscriber line signaling information exchanged during the call session, and the simulation is implemented by the SIP protocol. The user line signaling information is transmitted between the access signaling adaptation entity and the SIP application server; and the SIP application server performs corresponding processing according to the simulated subscriber line signaling information to implement the PSTN simulation service.

The method according to claim 1, wherein the method for performing the conversion between the analog subscriber line signaling and the SIP signaling by the access signaling adaptation entity is: separately performing analog subscriber line signaling information The bearer is carried in the SIP message to the SIP application server, or the one or more analog subscriber line signaling information is carried in the same SIP message and transmitted to the SIP application server.

The method according to claim 1, wherein the method for performing the conversion between the analog subscriber line signaling and the SIP signaling by the access signaling adaptation entity is: the access signaling adaptation entity Parsing the function information corresponding to the analog subscriber line signaling information, and transmitting the parsed function information to the SIP application server by using a SIP protocol.

The method according to claim 1, wherein the method for performing the conversion between the analog subscriber line signaling and the SIP signaling by the access signaling adaptation entity is:

The method according to claim 1, wherein the SIP signaling is a SIP Info message, or a SIP Message message, or a SIP Invite message.

The method according to claim 1, wherein the SIP signaling includes a message body of a MIME media type carrying the analog subscriber line signaling information, where the MIME media type includes:

a MIME media type field for identifying a MIME media type category; a MIME media subtype field for identifying a subtype of the MIME media type; and an analog subscriber line signaling information carried in the message body for identifying the MIME media subtype The encoding mode field of the encoding method.

The method according to claim 6, wherein the value of the MIME media type field is an application application; and the value of the MIME media subtype field is analog subscriber line signaling analog-subscriber-signal.

The method according to claim 6, wherein the encoding mode field takes a text extension Bacchus normal mode ABNF mode, or an extended markup language XML mode.

9. The method according to claim 1, wherein the analog subscriber line signaling comprises: user tap signal signaling, and/or digital signaling, and/or ringing, and/or tone information. , and/or billing pulse signaling, and/or reverse polarity signaling, and/or frequency shift keying signaling.

10. The method according to claim 1, wherein: the SIP packet network is an IP Multimedia Subsystem IMS;

The SIP application server is a simulation service server PES AS.

The method according to claim 1, wherein: the access signaling adaptation entity is an integrated access device with an access signaling adaptation and an access bearer adaptation function.

The method according to claim 10, wherein the session includes a first user and a second user, and the PSTN service is a call waiting emulation service, and the specific implementation of the method includes: After receiving the request from the third user to call the first user, the PES AS carries the call waiting tone in the SIP signaling and sends the call waiting tone to the AGCF, where the AGCF controls the connection in the SIP packet network. The ingress gateway plays a waiting tone to the first user;

The AGCF controls an access gateway in the SIP packet network to disconnect the media connection of the first user and the second user, and sends a dial tone to the first user;

The ACGF carries the service code information from the first user in the SIP signaling and sends the information to the PES AS. The PES AS parses the received service code information to learn that the current service is a call waiting emulation service and the The second user is a holder;

The method according to claim 10, wherein the session includes a first user and a second user, and the PSTN service is a call waiting emulation service, and the specific implementation of the method includes:

The ACGF will send service code information from the first user and the flash signal information The information is sent to the PES AS in the SIP signaling; the PES AS parses the received information to learn that the current service is a call waiting emulation service and the second user is a hold party;

The method according to claim 10, wherein the session includes a first user and a second user, and the PSTN service is a three-party call emulation service, and the specific implementation of the method includes:

After receiving the flashing signal from the first user, the AGCF controls an access gateway in the SIP packet network to disconnect the media connection of the first user and the second user, and sends the media connection to the first user Sending a dial tone; the first user dialing a number of the third user;

The ACGF sends the third user number information and the flashing signal information from the first user to the PES AS in the SIP signaling; the PES AS parses the received information to learn that the current service is a three-way Call simulation service;

The method according to claim 14, wherein, after the PES AS parses the received information to learn that the current service is a three-party call emulation service, the PES AS and the AGCF update the first user and the first Before the three-user session description protocol SDP, the method further includes:

The PES AS changes the second user to receive only the wood sending media stream through the SIP protocol.

16. The method according to claim 10, wherein: during the session The first user and the second user are included, and the PSTN service is an incoming call emulation service, and the specific implementation of the method includes:

The PES AS negotiates a session description protocol SDP between the second user and the third user, and establishes a media path between the second user and the third user by using the updated SDP; and simultaneously releases the first user Conversation.

The method according to any one of claims 12 to 16, wherein the first user is an analog user.

A system for implementing a public switched telephone network PSTN emulation service, characterized in that, in a session initial protocol SIP packet network, the system at least includes: an access signaling adaptation entity, a SIP application server;

The system according to claim 18, wherein the converting unit respectively carries the received analog subscriber line signaling information in different SIP messages and transmits the information to the SIP application server, or more than one analog The subscriber line signaling information is carried in the same SIP message and delivered to the SIP application server.

The system according to claim 18, wherein the conversion unit parses the function information corresponding to the analog subscriber line signaling information, and then transmits the parsed function information to the SIP application server by using a SIP protocol. .

The system according to claim 18, wherein the converting unit uses a SIP subscription and response mechanism, and applies a key markup language KPML to transmit user button digital signaling information in the analog subscriber line signaling information to the system. Said SIP application server.

22. The system according to claim 18, wherein the SIP packet network is an IP Multimedia Subsystem IMS;

An access signaling adaptation entity, where the access signaling adaptation entity includes: simulating subscriber line signaling and SIP signaling for analog subscriber line signaling information exchanged during a call session A conversion unit that converts between commands.

The access signaling adaptation entity according to claim 23, wherein the conversion unit respectively carries the received analog subscriber line signaling information in different SIP messages and transmits the information to the SIP application server. Or transmitting one or more analog subscriber line signaling information in the same SIP message to the SIP application server.

The access signaling adaptation entity according to claim 23, wherein the conversion unit parses the function information corresponding to the analog line signaling information, and then transmits the parsed function information through the SIP protocol. Give the SIP application server.

The access signaling adaptation entity according to claim 23, wherein the conversion unit uses a SIP subscription and response mechanism, and applies a key markup language KPML to transmit a user button in the analog subscriber line signaling information. Digital signaling information to the SIP application server.

27. The access signaling adaptation entity of claim 23, wherein The access signaling adaptation entity is an access gateway control function AGCF.