WO2007025436A1

WO2007025436A1 - A method for calling identity delivery when calling in the communication system

Info

Publication number: WO2007025436A1
Application number: PCT/CN2006/001682
Authority: WO
Inventors: Youzhu Shi
Original assignee: Huawei Technologies Co., Ltd.
Priority date: 2005-09-01
Filing date: 2006-07-14
Publication date: 2007-03-08

Abstract

A method for calling identity delivery when calling in the communication system is disclosed. By using the method, even in a packet core network, in which the call control signaling is an SIP form, a user can acquire the calling display of a new calling during the communicating state. In the present invention, when the INVITE message is received, the service control unit judges whether the called user terminal is in a communicating state and meets the CID service transferring condition of the communicating state, if yes, sends a message carrying the CID service identifier in the communicating state to the called user terminal, the called user terminal or the SIP user agent point applies the CID service in the communicating state according to the identifier. The CID service identifier in the communicating state can be carried by the MIME in the SIP message, or by the new head domain expand-defined in the SIP message, or by the new parameter expand-defined in the existed head domain of the SIP message.

Description

Method for calling party identification display during communication in communication system

TECHNICAL FIELD The present invention relates to caller identification display technology, and more particularly to a method for implementing caller identification display in a packet core network using a Session Initiation Protocol ("SIP") as call control signaling.

Background technique

With the rapid development of microelectronics and computer technology, switching technology has been unprecedentedly developed. Traditional telecommunications networks based on circuit-switched technology are developing toward packet-switched broadband telecommunications networks.

This is because circuit-switching technology is mainly used for transport and voice-related services. This type of network switching has great limitations for data services. Packet switching technology is an exchange method proposed for the characteristics of data communication services. Its basic feature is that it uses storage and forwarding for connectionless, and divides the data to be transmitted into many small pieces of data according to a certain length. The header field of the corresponding function for routing and verifying data is added before the data as a basic unit of data transmission, that is, a packet. With packet switching technology, no connection needs to be established before communication. Each node first collects and saves the packet sent by the previous node in the buffer, and then selects the appropriate link according to the address information in the packet header. Send to the next node, so that bandwidth can be dynamically allocated according to user requirements and network capabilities during communication.

In packet switching technology, used by the Internet Engineering Task Force (INTERNET)

The TASK FORCE, called "IETF";), launched the Session Initiation Protocol (SIP) as the call control signaling for the packet telecom core network, which is one of the current technological development trends.

According to the standards of the International Telecommunication Union Telecommunication Standardization Sector ("ITU-T") and the "European Standards Institute" (ETSI), In the packet telecommunication network architecture of SIP as the call control signaling of the core network, SIP separates the network nodes of service control, call control, and access control to implement services and calls. The complete separation of controls is a clear trend.

The Call Waiting service and the Calling Identity Delivery (Calling Identity Delivery) call service of the present invention will be described below.

Call waiting service refers to a service that can hold the first call while receiving a second call while the user is on a call, and implements a service to switch between two calls. For example, when A is talking to B, if he encounters a call to A, A can hear the incoming waiting tone in the microphone. At this time, A can ask B to wait a little, and then talk to C; or C, please continue to talk to B. Free choice of the caller will not delay the handling of important things. For a detailed description of the call waiting service, please refer to the national standard “General Technical Specifications for Telephone Exchange Equipment of the Ministry of Posts and Telecommunications (Appendix)”.

The CID service is a new service widely used in telecommunication networks to provide the called subscribers. Caller identification information such as calling number, calling name, calling time, time, etc. can be displayed on the called user terminal device. For details, refer to the national standard "Technical requirements and test methods for telephone caller identification information transmission and display function". In this document, the caller identification information is sent to the called user terminal device in two states: one is the called user terminal on-hook state; the other is the called user terminal off-hook state.

The CID service in the off-hook state of the called user terminal is: when the user B with the CID function is already in the call state with the user A, and the third party user C is calling the user B, the user B is displayed on the user B terminal device. User C's identification information business. The CID service in the off-hook state of the called user must be based on the call waiting service described in the above. Therefore, the CID service in the call state is sometimes called the CID service in the call waiting.

In the traditional telecommunication network, the network control device to which the called user terminal device belongs provides service control, call control, and access control functions to the user, and the network control device determines that the called user B meets the CID service calling condition in the call state. After that, the calling party identification information can be sent to the called user B terminal device directly through appropriate user signaling, such as Frequency Shift Keying ("FSK").

On the other hand, the above-mentioned called user terminal is off-hook and the CID service in the on-hook state, as a telecommunications popular service, needs to be provided to the access registered user of the packet telecommunication network.

As described above, in the packet telecommunication network architecture in which SIP is used as the call control signaling of the core network, the network nodes of the service control, the call control, and the access control are separated. In this telecommunications grid The CID service under construction has encountered problems in its implementation. Specifically, after determining that the called user B meets the CID service calling condition in the call state, the service control node sends the calling identification information to the called terminal device. In the SIP protocol, for a call, the caller identification information is carried by the SIP INVITE message and sent to the called user, but if in this scenario, the service control node sends a SIP INVITE message to the terminal device, the terminal device is already in the When the call is in progress, the SIP INVITE message may be rejected, so that the CID service application in the call state fails.

The reason for this situation is: The SIP INVITE message indicates the initial request of a call session, and the called terminal device does not know the service subscription and application of the user B, that is, it is not clear whether the user B has activated the call. Waiting for the service, whether it has the CID service right in the call state, so when it receives a new incoming call (SIP INVITE message) for User B, and the terminal device is already in the call state, it will The incoming call may be rejected because it cannot be processed (ie, it is not known how to handle it), resulting in a user experience that is inferior in business inheritance.

Summary of the invention

In view of this, the main object of the present invention is to provide a method for calling party identification display during a call in a communication system, so that in a packet core network using SIP as call control signaling, the user can still get a new call in a call state. The caller's display of incoming calls.

To achieve the above object, the present invention provides a method for caller identification display during a call in a communication system. The packet core network of the communication system uses the SIP protocol as call control signaling, and provides various services for the user by using the service control unit. The logic control function includes the following steps:

When the service control unit receives the first message indicating the call request, if the called user terminal of the call is in a call state and meets the caller identification display service calling condition in a preset call state, the called user is called to the called user. The terminal or its SIP user agent node sends a second message, where the first identifier indicating the caller identification display service in the call state is carried;

The called user terminal or its SIP user agent node receives and parses the second message, and completes the caller identification display service application in the call state according to the first identifier.

The first message is an "INVITE" message in the SIP protocol.

Further in the method, the second message may be one of the following SIP protocol messages: "MESSAGE" message, "NOTIFY" message, "INPO" message, or "INVITE" Message.

In addition, in the method, the first identifier includes a call waiting service identifier.

Further in the method, the second message carries the first identifier by a new header field defined by the extension.

Further in the method, the second message carries the first identity by extending a new parameter defined in an existing header field.

In addition, in the method, the existing header field carrying the first identifier by extending the definition new parameter may be a "P-Asserted-Identity" header field.

In addition, in the method, if the called user terminal is a legacy terminal and accesses the packet core network through the SIP user agent node, and the SIP user agent node and the user media conversion node are the same network entity, Then, the SIP user agent node directly sends the caller identification display information in the call state to the legacy terminal.

Further in the method, if the called user terminal is a legacy terminal and passes the

The SIP user agent node accesses the packet core network, and the SIP user agent node and the user media conversion node are different network entities, and the SIP user agent node sends a call state to the legacy terminal by forwarding the user media conversion node. The next caller identification display information.

In addition, in the method, the method further includes the following steps:

The call control node in the packet core network triggers the message to the service control unit when receiving the first message;

The service control unit sends a second message to the called user terminal or its SIP user agent node by the delivery of the call control node.

By comparison, it can be found that the main difference between the technical solution of the present invention and the prior art is that, when receiving the INVITE message, the service control unit determines whether the called user is in a call state and satisfies the CID service calling condition in the call state, and if so, Sending a message carrying the CID service application identifier in the call state to the called user terminal, and the called user terminal or the SIP user agent node applies the CID service in the call state according to the identifier.

The CID service application identifier in the call state may be carried by the MIME body in the SIP message body, or carried by the new header field defined by the extension in the SIP message, or the SIP message has been carried by the new parameter defined by the extension in the header field. The difference in the technical solution brings about a more obvious beneficial effect, that is, because the service control unit grasps the current state and the subscription information of the user, it can make an accurate judgment of whether to allow the CID service in the application call state, the called user. All the terminal has to do is apply the CID service in the call state according to the indication of the service control unit, so that in the packet core network with SIP as the call control signaling, the user can still get a new incoming call in the call state. The caller shows a good experience in business inheritance.

DRAWINGS

1 is a network logical structure diagram of a packet core network in which SIP is used as call control signaling; FIG. 2 is a signaling diagram of a method for caller identification display during a call in a communication system according to a first embodiment of the present invention;

Fig. 3 is a signaling diagram showing a method of caller identification display during a call in a communication system according to a second embodiment of the present invention.

DETAILED DESCRIPTION OF THE INVENTION In order to make the objects, technical solutions and advantages of the present invention more comprehensible, the present invention will be further described in detail with reference to the accompanying drawings.

2 shows a CID method during a call in a communication system according to a first embodiment of the present invention, which is applicable to a packet core network 1 using SIP as call control signaling, and the logical structure of the packet core network is shown in FIG. The main control service node 11, the call control node 12, the access control node 13, and the SIP user agent node 2, the user media conversion node 3, the legacy terminal 4, and the SIP terminal 5. It can be seen that the packet core network with SIP as the call control signaling supports the traditional terminal 4 such as the PLAIN OLD TELEPHONE SERVICE ("PoTS") terminal and the integrated service digital network (Integrated) in addition to the SIP terminal 5. The Services Digital Network, called "ISDN", is connected to terminals.

The following describes each component in the packet core network to which the present invention is applied. The user media conversion node 3 is used to provide mutual conversion of circuit voice and packet voice between the legacy terminal 4 and the packet domain, and supports the codec function of the packet voice.

The SIP user agent node 2 is a SIP User Agent (SIP UA) that accesses the packet core network with SIP as call control signaling, and provides direct connection between user signaling and SIP signaling of the legacy terminal. Or indirect translation. Wherein, the user media conversion node 3 and When the SIP user agent node 2 is a different network entity, the E1 interface between the two is a media gateway control protocol interface such as H.248; when the user media conversion node 3 and the SIP user agent node 2 are the same network entity, the E1 interface For a custom internal interface.

The access control node 13 is a network node that provides the legacy terminal 4 and the SIP terminal 5 with functions such as registration authentication authentication of the access packet core network. The terminal is registered to the call control node 12, and when the terminal initiates a call, the access control node 13 routes the call to its home call control node 12. When the access control node 13 and the SIP user agent node 2 are different network entities, the E2 interface between the two is a SIP interface; when the access control node 13 and the SIP user agent node 2 are the same network entity, the E2 interface is SIP or a custom internal interface. The E3 interface between the SIP terminal 5 and the access control node 13 is a SIP interface.

The call control node 12 is configured to provide call control, routing connection, and the like for the registered terminal accessing the packet core network, which can trigger the call to the service control node 11. The E6 interface between the two call control nodes 12 is a SIP interface. When the call control node 12 and the access control node 13 are different network entities, the E4 interface between the two is a SIP interface; when the call control node 12 and the access control node 13 are the same network entity, the E4 interface is a SIP interface. Or a custom internal interface.

The service control node 11 provides various service logic control functions for the registered terminals accessing the packet core network, and is a host execution environment for various services. When the service control node 11 and the call control node 12 are different network entities, the E5 interface between the two is a SIP interface; when the service control node 11 and the call control node 12 are the same network entity, the E5 interface is a SIP interface or The internal interface defined. There may be multiple service control nodes that handle different services.

In an embodiment, the service control node 11 is a service control node capable of handling CID traffic in a call state.

As shown in FIG. 2, in the CID method during a call in the communication system of the present embodiment, first, in step 210: when the user B is using the legacy terminal to talk with another user A, the call control node to which the registration belongs is received. Upon an incoming call, a SIP INVITE message is received. In order to make the text concise, the flow of the incoming call in its outgoing section (calling side) is not mentioned.

Thereafter, in step 220: the call control node triggers the INVITE message to the service control node according to a certain manner, such as the called user number, if the service control node can process the call state The CID service in the state satisfies the CID service calling condition in the preset call state. In step 230, the service control node determines that the user B is in the call state, can apply the call waiting service, and subscribes to the CID service in the call state, then enables the CID service processing flow in the call state, and sends a SIP MESSAGE message to the call control node. The message carries the CID service application identifier and the caller identification information in the call state. Generally, the SIP message body includes a Multipurpose Internet Mail Extensions ("MIME") media type body to implement For this feature (eg CWCID Media Type ), the MIME media type can be defined as follows:

Media type name: application (media type name: application)

Media subtype name: CallWaiting-CallerlD (media subclass name: Call Waiting - Caller ID).

Required parameters: version (required parameters: version)

Optional parameters: base (optional parameter: basic)

Encoding scheme: XML (encoding type: XML)

The MIME body may include a specific parameter of the caller identification information, and the parameter content may be directly taken from the caller identification information carried in the incoming SIP INVITE message received by the service control node. Generally, the caller identification The information is carried in the P-Asserted-Identity header field.

Another method is that the SIP message carries only the MIME body of the CID service application identifier in the call state, and the caller identification information is still carried by the P-Asserted-Identity header field.

In addition, the encoding format of the MIME body can also be binary:

Encoding scheme: binary (encoding type: binary)

At step 240: the call control node delivers a SIP MESSAGE message to the legacy terminal. In step 250, the SIP user agent node of the traditional terminal receives and parses the SIP MESSAGE message, and sends the caller identification information extracted from the message to the traditional manner according to the CID service application identifier in the call state carried in the message. terminal. It should be noted that if the user media switching node and the SIP user agent node are the same network entity, the SIP user agent node can directly send the CID information in the call state to the traditional terminal, such as the FSK mode. For the specific process, refer to the national standard "telephone". Technical requirements and test methods for caller identification information transmission and display functions; if the user media conversion node and the SIP user agent node are different networks The SIP user agent node first sends the CID information in the call state to the user media conversion node through the El interface, such as the event packet of the H.248 protocol, and the latter sends the CID information to the traditional terminal, such as the FSK mode.

Thereafter, at step 260: the SIP user agent node returns a 200 OK response code to the SIP MESSAGE message to the call control node.

At step 270: the call control node passes the 200 OK response code to the SIP User Agent node.

It should be noted that, in this embodiment, after the service control node processes the CID service in the call state, the sent SIP message may be either a MESSAGE message or any SIP capable of carrying the CID service application identifier in the call state. Messages, such as SIP NOTIFY messages, SIP INFO messages, and SIP INVITE messages.

On the other hand, regardless of the specific SIP message, there are many options for carrying the application identifier in the call state through the SIP message. One is to specify the CID service application identifier in the call state through the MIME body, or directly define a new SIP header field, and the value of the header field indicates the CID service application identifier in the call state, or directly in the P- The Asserted- Identity header field is extended with a new parameter. The value of this parameter indicates the CID service application identifier in the call state.

It should be noted that the flowchart illustrations and texts 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. Therefore, in the above description of the embodiment, the access control node is omitted, and when the access control node and the SIP user agent node are different network entities, the call will pass through the access control node.

3 is a diagram showing a CID method during a call in a communication system according to a second embodiment of the present invention, and is also applicable to a packet core network using SIP as call control signaling, and the difference from the embodiment of FIG. 2 is that the called user The terminal of B is a SIP terminal, so there is no SIP user agent node.

As shown in FIG. 3, in the CID method during a call in the communication system of the present embodiment, first, in step 310: when user B is using a SIP terminal to talk to another user A, the call control node to which the registration belongs is received. Upon an incoming call, a SIP INVITE message is received. Thereafter, in step 320: the call control node according to a certain manner, such as the called subscriber number, will The INVITE message is triggered to the service control node. If the service control node can process the CID service in the call state, the CID service call condition in the preset call state is met. In step 330, the service control node determines that the user B is in the call state, can apply the call waiting service, and subscribes to the CID service in the call state, and then enables the CID service processing flow in the call state, and sends a SIP MESSAGE message to the call control node. The message carries the CID service application identifier and the caller identification information in the call state. As described above, a MIME media type body is included in the SIP message body to implement this function (for example: CWCID Media Type) „ The MIME body can contain specific parameters of the calling party identification information, and the parameter content can be directly taken from the service control. The caller identification information carried in the incoming SIP INVITE message received by the node. Generally, the caller identification information is carried in the P-Asserted-Identity header field. The other method is that the SIP message carries only the call. The CID service in the state applies the MIME body of the identity, and the caller identification information is still carried by the P-Asserted-Identity header field. Thereafter, in step 340: the call control node delivers the SIP MESSAGE message to the SIP terminal.

Different from the first embodiment, in this embodiment, the call control node directly sends a SIP MESSAGE message to the SIP terminal. After receiving the message, the SIP terminal parses it, obtains the CID application identifier in the call state, and completes the CID service application in the call state.

In this embodiment, similarly, regardless of the specific SIP message, there are various options for carrying the CID service application identifier in the call state through the SIP message. One is to specify the CID service application identifier in the call state through the MIME body, or directly define a new SIP header field, and the value of the header field indicates the CID service application identifier in the call state, or directly in the P- A new parameter is extended in the Asserted-Identity header field, and the value of the parameter indicates the CID service application identifier in the call state.

In addition, as described above, the CID service in the call state is premised on the call waiting service. Therefore, if the call waiting service application identifier is carried in the SIP message, and the parameter indicating the CID service in the SIP message is matched, The object of the invention can also be achieved. When the service control node determines that the CID service in the call state can be used, the SIP message is sent, and the message carries an application identifier of the call waiting service. Similarly, the call waiting service application identifier can be transmitted through a MIME body or a new extension. The SIP header field, or a new extended parameter in an existing SIP header field, etc., to indicate the call waiting service application identifier, and in the SIP protocol The Privacy header field indicates whether the calling number is allowed to be displayed to the called party, that is, the CID service application identifier in the call state is indicated by the cooperation of the call waiting service and the normal CID service. For example, the SIP message does not carry the Privacy header field to indicate the permission. The called party displays the calling number, and carries the Privacy header field to indicate that the calling number cannot be displayed to the called party. The specific SIP protocol indicates whether to allow the calling party to display the calling number. For details, refer to the relevant IETF standard and ETSI standard. The invention is not described in detail.

Although the present invention has been illustrated and described with reference to the preferred embodiments of the present invention, those skilled in the art The spirit and scope of the invention.

Claims

Rights request

A method for caller identification display during a call in a communication system, wherein a packet core network of the communication system uses a SIP protocol as call control signaling, and provides a user with various service logic control functions by a service control unit, wherein , with the following steps:

The called user terminal or its SIP user agent node receives and parses the second message, and the first identifier completes the caller identification display service in the call state.

2. The method for caller identification display during a call in a communication system according to claim 1, wherein the first message is an "INVITE" message in a SIP protocol.

3. The method for caller identification display during a call in a communication system according to claim 1, wherein the second message is one of the following SIP protocol messages:

"MESSAGE" message, "NOTIFY" message, "INFO" message, or "INVITE" message.

4. The method for caller identification display during a call in a communication system according to claim 1, wherein the first identifier comprises a call waiting service identifier. .

The method of claim 1, wherein the second message carries the first identifier by using a new header field defined by the extension.

6. The method for caller identification display during a call in a communication system according to claim 1, wherein the second message carries the first identifier by a new parameter defined by extension in an existing header field.

The method for displaying a caller ID during a call in a communication system according to claim 6, wherein the existing header field carrying the first identifier by defining a new parameter is

Asserted-Identity" header field.

The method for caller identification display during a call in a communication system according to any one of claims 1 to 7, wherein if the called user terminal is a legacy terminal and is connected through the SIP user agent node Into the packet core network, and the SIP user agent node and user The media switching node is the same network entity, and the SIP user agent node sends the caller identification display information in the call state to the legacy terminal.

The method for caller identification display during a call in a communication system according to any one of claims 1 to 7, wherein if the called user terminal is a legacy terminal and is connected through the SIP user agent node Entering the packet core network, and the SIP user agent node and the user media conversion node are different network entities, and the SIP user agent node sends the caller identification in the call state to the legacy terminal by forwarding the user media conversion node. Display information.

The method for caller identification display during a call in a communication system according to any one of claims 1 to 3, further comprising the steps of: