CN104009988B - Call control method based on VoIP service system - Google Patents

Abstract

The invention provides a call control method based on the VoIP service system. SIP terminals and heterogeneous fusion gateways only need to support the common standard of RFC3261, and users only need to use the numeric keypad when making dial-up calls for various services, and all complex analysis and processing are performed by each functional entity using this call control method Finish. All client terminals, including SIP service terminals and heterogeneous fusion gateways, are used as thin clients, which greatly enhances compatibility and versatility, and reduces investment in the number and types of service terminals and user training in use.

Description

A Call Control Method Based on VoIP Service System

technical field

The present invention relates to a call control method based on a VoIP service system, in particular to a call control method applicable to a variable-length number (referring to a number whose length is The call control method of the VoIP service system analyzed by the service type, the attribution of the called user, etc.).

Background technique

At present, the service system is developing and evolving towards an all-IP technology system, and the VoIP system based on SIP signaling has become the most mainstream technology of the multimedia service system under the IP technology system. However, the evolution of technology is a long-term process, and the network and service system of the traditional circuit switching system will exist for a long time. Therefore, the multimedia service system based on the VoIP technology system needs to be able to realize interconnection and interoperability with the service system based on circuit switching. Although the current call control method based on the VoIP service system can control the calls of various services and can also support the interconnection and intercommunication with the circuit switching service system (the interconnection and intercommunication of the two systems requires the deployment of heterogeneous Converged gateways perform two-way translation and adaptation for differences in control signaling and media transmission methods, because heterogeneous converged gateways serve as terminal agents for each phone in the circuit switching system in the VoIP system, and heterogeneous converged gateways can be regarded as The role of a terminal in the VoIP system), but there are still deficiencies and shortcomings, the main shortcomings are as follows:

(1) Inconsistent user experience when dialing calls for common services.

Ordinary services generally refer to voice services carried out between two users, that is, traditional telephone services.

In the service system of the traditional circuit switching system, the user identity is a telephone number. For example, the telephone number in the civilian field is an E.164 number, and the telephone number in the military field is a number with certain rules. Its characteristic is that such variable-length numbers are generally divided into "numbers representing area information" + "numbers representing user identities". Therefore, when a user dials to call a local user, he can dial "the number representing the area information" + "the number representing the user's identity", or he can only dial the "number representing the user's identity" (for example, dialing a Chengdu local user in Chengdu If you need to dial a number to call a user outside the city, you must dial "the number representing the area information" + "the number representing the user's identity" (for example, when calling a user in Beijing from Chengdu, you need to dial 010xxxxxxxx).

However, in the service system based on the VoIP technical system, the user identity is a SIP URI, which is in the form of "username@home area information". The "username" here is a universal character string, which can be a variable-length number, any letter, or even a combination of letters, numbers and special symbols; "attribution area information" is the domain name information representing the area, and can also be It is the IP address + port number representing the region. Therefore, when a user dials a local user, he or she can dial the complete SIP URI information, or just dial the "username" (for example, when dialing a local user in Chengdu, you can dial the complete SIP URI information similar to xxx@chengdu.com SIP URI or xxx, the latter is automatically added to a complete SIP URI by the SIP phone); if you need to dial to call a user outside the city, you must dial the complete SIP URI (for example, when dialing a Beijing user in Chengdu, you need to dial something like xxx Full SIP URI of @beijing.com).

It can be seen that the experiences of making dial-up calls for common services in the service systems of the two technical systems are completely different. In addition, if you want to use the telephone terminal of the circuit switching service system to call the SIP telephone terminal through the heterogeneous fusion gateway (for example, dial a Beijing user xxxxxxxxx@ in the VoIP system from the telephone of the circuit switching service system in Chengdu) beijing.com, the call will first be sent to the heterogeneous fusion gateway in Chengdu. The heterogeneous fusion gateway extracts the user name of the called user as "xxxxxxxx", but for the gateway itself, it is equivalent to the user dialing only carrying The "user name" information does not carry the "home area information", so the heterogeneous fusion gateway will automatically add the local area information, and finally generate "xxxxxxxx@chengdu.com", which will cause the call to fail).

(2) Inconsistent user experience when dialing calls for special services.

Special business is represented by teleconference. Generally speaking, there are two modes of teleconference: scheduled conference and temporary conference. A scheduled conference is a conference that has been configured with conference member information, while an ad hoc conference is a temporary conference member that is assigned when a conference is held.

In the business system of the traditional circuit switching system, when a user initiates a conference reservation, he needs to dial "the service function code representing the conference reservation" + "the number representing the area information" + "the conference room number", where "the number representing the area information " is optional, depending on whether the configuration information of the meeting room is local (for example, to hold a scheduled conference in Chengdu in Chengdu, you can dial abc028xxx or abcxxx; if you hold a scheduled conference in Beijing in Chengdu, you must dial abc010xxx, where "abc" is the business function code information and business mode information, which means the reserved conference mode of special business). When a user initiates an ad hoc conference, he needs to dial consecutively "the service function code representing the ad hoc conference" + "the number representing the area information of user 1" + "the number representing the identity of user 1" + "*"+... where "represents the user area Information number" is optional, depending on whether the user is a local user (for example, to hold an ad hoc conference in Chengdu, and the other two conference members are Chengdu user xxxxxxxx and Beijing user yyyyyyyy, you need to dial abdxxxxxxxx*010yyyyyyyy, where "abd " is the business function code information and business mode information, which means the temporary conference mode of special business).

However, in the service system based on the VoIP technology system, the relevant information of the reserved conference is configured on the conference server. Therefore, when the user initiates the reserved conference, he needs to dial the "SIP URI representing the reserved conference" (for example, the conference held in Chengdu) For a scheduled conference in Chengdu, you can dial conference001@chengdu.com or conference001; if you hold a scheduled conference in Beijing in Chengdu, you must dial conference001@beijing.com). When a user initiates an ad hoc conference, he needs to configure the conference members on the SIP phone that supports the conference function. When making a dial-up call, the INVITE request message sent by the SIP phone will carry the recipient-list list that will contain the conference member information. xml file.

It can be seen that the experience of making dial-up calls for special services in the service systems of the two technical systems is also completely different. In addition, it is impossible to use the telephone terminal of the circuit switching service system to conduct a conference call through the heterogeneous fusion gateway (the above-mentioned conference-related operations cannot be performed on ordinary telephone terminals).

(3) There are requirements for SIP service terminals when performing special services.

Here we still take the teleconferencing service as an example. Whether it is a scheduled meeting or an ad hoc meeting, the SIP service terminal needs to be able to identify and parse the "isfocus" parameter of the Contact message header, the "recipient-list-invite" parameter of the Require message header, and the Require-Disposition The "recipient-list" parameter of the message header. For ad hoc conferences, it is also required that the SIP service terminal must be able to support the configuration of the list of invited conference members. The operation button of "initiate a conference" must support the REFER message type, the "recipient-list-invite" parameter of the Require message header, and the Require- INVITE message with the "recipient-list" parameter in the Disposition message header.

It can be seen that ordinary SIP service terminals and heterogeneous fusion gateways that only support RFC3261 cannot carry out special services.

Therefore, how to reduce the weight (that is, simplify, including simplifying the two aspects of SIP signaling itself and SIP service terminals) call control method, to ensure that the most common standard SIP service terminals and heterogeneous fusion gateways can carry out various services (including ordinary services and Special services), and how to simplify the user's operation and ensure that the user has exactly the same experience as using an ordinary phone is an urgent problem to be solved.

Contents of the invention

The technical problem to be solved by the present invention is to provide a call control method based on the VoIP service system which simplifies the user's operation and is exactly the same as using an ordinary telephone.

The technical scheme that the present invention adopts is as follows: a kind of call control method based on VoIP service system, it is characterized in that, its method step is:

Step 1. The service controller functional entity extracts the user name part from the called SIP URI (user name@home area information) in the received SIP message, and judges whether there is service function code information according to the user name, otherwise it is normal Business, identify the user ID number information and user area number information according to the user name part, if it is a special service, go to step 4;

Step 2, perfecting the attribution area information of the SIP URI according to the user identity number information and the user area number information;

Step 3, routing the SIP request signaling of the call to the home area of the called terminal (for a foreign user) or the called terminal (for a local user);

Step 4, perform service trigger processing, and route the SIP request signaling of the call to the corresponding application server functional entity;

Step 5. The application server functional entity extracts the user name part (the part in front of "@") from the called SIP URI in the received SIP message, that is, the variable-length number string dialed by the user. In the long number string information, the service function code information, service mode information, and service member information are identified according to the local configuration information, and the corresponding follow-up processing of special services is performed based on these information.

Preferably, the method steps further include: determining whether the user has the corresponding service authority after determining the service type requested by the user, otherwise rejecting service access, and performing service access if yes.

Preferably, the method further includes: after judging the service type requested by the user, judging the service system to which the user number belongs; The message is routed to the regional heterogeneous fusion gateway, and the latter performs service adaptation; if the destination user is a user of the VoIP system in the region, complete the URI information of the INVITE message and route the SIP message to the SIP used by the destination user Service terminal; if the target user is a user in another area, complete the URI information of the INVITE message and route the SIP message to the area to which the target user belongs; if the target user is a service number of a special service, complete the URI message of the INVITE message And route the SIP message to the application server processing the special service or the area where the application server is located (when the application server does not belong to the local area).

Compared with the prior art, the beneficial effect of the present invention is that: the SIP terminal and the heterogeneous fusion gateway only need to support the common standard of RFC3261, and the user only needs to use the numeric keypad when making dial-up calls of various services, and all complex analysis All the processing and processing are completed by each functional entity using this variable-length number-based lightweight call control method. All client terminals, including SIP service terminals and heterogeneous fusion gateways, are used as thin clients, which greatly enhances compatibility and versatility, and reduces investment in the number and types of service terminals and user training in use.

Description of drawings

FIG. 1 is a schematic diagram of the principle of one embodiment of the present invention.

detailed description

In order to make the object, technical solution and advantages of the present invention clearer, the present invention will be further described in detail below in conjunction with the accompanying drawings and embodiments. It should be understood that the specific embodiments described here are only used to explain the present invention, not to limit the present invention.

Any feature disclosed in this specification (including any appended claims, abstract and drawings), unless expressly stated otherwise, may be replaced by alternative features which are equivalent or serve a similar purpose. That is, unless expressly stated otherwise, each feature is one example only of a series of equivalent or similar features.

In this specific embodiment, the functional entities using the variable-length number-based lightweight call control method include service controllers and conference servers in two regions (the conference server is an example of a service server); the general thin client also It includes common SIP terminals and heterogeneous fusion gateways in two areas, as shown in Figure 1.

As shown in Figure 1, in this embodiment, the area number of area 1 is 028, the domain name information is chengdu.com, the number of SIP terminal 1 is 3030111, the number of ordinary phone 1 is 6060111; the area number of area 2 is 010, the domain name information is beijing.com, the number of SIP terminal 2 is 3030222, and the number of ordinary phone 2 is 6060222.

The service controller receives and processes the SIP message generated by the dial-up call of the service controller from the local client and other areas, identifies and triggers the service according to the variable-length number information carried in the signaling.

The conference server receives and processes the SIP message routed from the local service controller, conducts conference mode identification and conference member identification according to the variable-length conference phone number information carried in the signaling, and performs corresponding follow-up actions according to the identified mode and conference members Conference control processing.

Universal thin clients refer to ordinary standard SIP service terminals and heterogeneous fusion gateways that only support RFC3261, and do not need to support other SIP extension protocols.

Specific embodiment 1: Common service dial-up call service

When a user uses an ordinary phone 1 to call a user of the SIP service terminal 2, he needs to dial the "area code" + "user number" of the destination user, that is, 0103030222.

Heterogeneous fusion gateway 1 converts the dialing signaling generated by the circuit switching system of ordinary telephone 1 into SIP signaling, generates a SIP message with 0103030222@chengdu.com as the called SIP URI, and sends it to its local service control Device 1.

In this specific embodiment, after receiving the SIP message, the service controller 1 first judges the authority of the calling user (the calling user is 6060111@chengdu.com, which is a valid local user), and then configures the Dial the called variable length number (0103030222) for analysis. Since there is no service function code, it is judged that the call is a normal service dial-up call.

The service controller 1 analyzes the variable-length number of the called user, and judges the location of the called user according to the local policy and location attribution information: if the called user number belongs to other areas, then improve the URI information of the SIP message and Route the SIP message to the destination area; if the destination user number is the connected circuit switching network user in this area, then improve the URI information of the SIP message and route the SIP message to the heterogeneous fusion gateway, and the latter will carry out the business Adaptation; if the destination user is a packet switching network user in the area, then improve the URI information of the INVITE message and route the SIP message to the SIP service terminal used by the destination user. For example, when the called number is 0103030222, the called user belongs to area 2 by analyzing the "area number" 010, so the called SIP URI (0103030222@beijing.com) of the SIP message is routed to the representative area 2 The business controller 2.

Specific embodiment two: Conference dial-up call service

For example, the user initiates a conference reservation using the ordinary phone 1, and needs to dial the corresponding "conference reservation function code" 901 and the "meeting room number" 001.

Heterogeneous fusion gateway 1 converts the dialing signaling generated by the circuit switching system of ordinary telephone 1 into SIP signaling, generates a SIP message with 901001@chengdu.com as the called SIP URI, and sends it to its local service control Device 1.

After receiving the SIP message, service controller 1 first judges the authority of the calling user (6060111@chengdu.com), and then analyzes the dialed variable-length number (901001) according to the local configuration information. If the service function code is 901, it is judged that the call is a dial-up call to reserve a conference, and the SIP message is routed to the conference server 1.

After receiving the SIP message, the conference server 1 analyzes the variable-length number dialed by the user, and determines the conference mode, conference function and conference-related configuration information according to the local policy. In this example, the conference is a call to reserve a conference, and the conference server 1 analyzes the conference members by checking the conference configuration information according to the pre-configured conference room information and assigns the conference room, and then immediately and automatically initiates a dial-up call to all other conference members .

When a user initiates an ad hoc conference using an ordinary phone 1, he needs to dial the corresponding "temporary conference function code" 902 and the conference member number list "3030111*0106060222".

Heterogeneous fusion gateway 1 converts the dialing signaling generated by the circuit switching system of ordinary telephone 1 into SIP signaling, generates a SIP message with 9023030111*0106060222@chengdu.com as the called SIP URI, and sends it to its local Business Controller 1.

After receiving the SIP message, service controller 1 first judges the authority of the calling user (6060111@chengdu.com), and then analyzes the dialed variable-length number (9023030111*0106060222) according to the local configuration information. If the service function code of the service is 902, it is judged that the call is a dial-up call of an ad hoc conference, and the SIP message is routed to the conference server 1.

After receiving the SIP message, the conference server 1 analyzes the variable-length number dialed by the user, and determines the conference mode, conference function and conference-related configuration information according to the local policy. In this specific embodiment, the conference is a temporary conference call, and the conference server 1 analyzes the numbers of other conference members (3030111 and 0106060222) according to the variable-length number, and immediately initiates a dial-up call to all other conference members automatically.

Claims (3)

1. a kind of calling-control method based on VoIP operation system is it is characterised in that its method and step is：

Step one, service controller functional entity extract user name portion in the called SIP URI from the sip message receiving Point, business function code information is determined whether according to user name, otherwise identifies that user identity number is believed according to a username portion Breath and user area number information, are then to enter step 4；

Step 2, improve the attributed region information of SIP URI according to user identity number information and user area number information；

Step 3, by calling SIP request signalling route to terminal called attributed region or terminal called；

Step 4, carry out business triggering process, and the SIP request signalling route that this is called is to corresponding application server functionality Entity；

Step 5, application server functionality entity extract user name portion in the called SIP URI from the sip message receiving Point, from the elongated number character string information that user is dialled, business function code information, business are identified according to locally configured information Pattern information and business information about firms.

2. the calling-control method based on VoIP operation system according to claim 1 is it is characterised in that methods described walks Suddenly also include：Judge after the type of service determining user's request whether user has corresponding service authority, otherwise refuse service access, Have, carry out service access.

3. the calling-control method based on VoIP operation system according to claim 1 and 2 is it is characterised in that described side Method also includes：After judging the type of service of user's request, judge the operation system that Subscriber Number belongs to, if purpose user number Code is the user of one's respective area circuit switching system, then improve the URI information of sip message and be routed to the sip message after improving One's respective area isomery Convergence gateway, is carried out the adaptation of business by the latter；If purpose user is the user of one's respective area VoIP system, Improve the URI information of sip message and the sip message after improving is routed to the SIP service terminal that purpose user is used；If Purpose user is the user in other regions, then improve the URI information of sip message and the sip message after improving is routed to purpose The region that user is belonged to；If purpose user is the service number of particular service, improves the URI message of sip message and incite somebody to action Sip message after improving is routed to the region that the application server processing this particular service or its application server are located.