WO2007025950A1

WO2007025950A1 - Unified centrex services via access code

Info

Publication number: WO2007025950A1
Application number: PCT/EP2006/065733
Authority: WO
Inventors: Jian Fen Huang; Yi Jun Hu
Original assignee: Nokia Siemens Networks Gmbh & Co. Kg
Priority date: 2005-08-31
Filing date: 2006-08-28
Publication date: 2007-03-08
Also published as: CN1925630A

Abstract

The present invention relates to a method for implementing VPBX (Virtual Private Branch Exchange), which is applied to a kind of communication network that comprises at least one application server and at least one soft switch. The method includes steps as follows: (1) defining an attribute for users in a BG (Business Group); (2) mapping said attribute to an AC (Access Code); (3) inserting said AC in front of a dialed number; and (4) routing said soft switch to said application server based on said AC. By using this method, the various users in the same BG can all use Service Logic of the application server, and can all dial the small numbers in the group and show the small numbers, which significantly reduces the cost of calls between different types of users in one group. Meanwhile, it is only needed to provide Attendant Service at the application server. The present invention would meet the needs of a united Fixed-Mobile network in the future.

Description

Method and device for implementing VPBX

Field of the invention

The present invention relates to a method for implementing a virtual private branch exchange (VPBX) , and more particularly, to a method and device for implementing the VPBX in a fixed network.

Background of the invention The VPBX (also known as central office exchange service,

Centrex) is a function that a digital stored-program control exchange (SPC exchange) offers. The Centrex is capable of providing functions like mini PBX for Business Group (BG) users. The BG users do not have to buy a physical PBX, but only need to classify a portion of the local telephone users on the exchange of the telecommunication office as a user group according to their requirements.

Users in said user group have two telephone numbers with different length: one of which is the large number (also known as Long Number) i.e. direct outward dialing number; and the other is the small number (also known as Short number) , i.e. "Extension" number in the user group, and the two numbers co-exist and are used together. The "Extension" users of the same group can be distributed at anywhere within the same local exchange office without limitations of "operator" equipments and region. Hence the Centrex is applicable for group (or company) users. The capacity of the telephone may range from 10+ to 100, 000+ depending on the needs of the company.

At present, some SPC exchanges are able to implement the Centrex, but these SPC exchanges only implement the Centrex for users who are based on some specific types of protocol (also known as Centrex based on specific fields) . For example, the Centrex is now available for users using the Integrated Access Device (IAD) based on the Media Gateway Control Protocol (MGCP) and the Integrated Access Media Gateway (AG) based on MGCP H.248; meanwhile, for users based on the Session Initiation Protocol (SIP) , and users using the Public Switched Telephone Network (PSTN) as well.

However, said Centrex in the prior art only implements the

Centrex for users based on a specific type of protocol (users based on a specific field) respectively. There is no unified Centrex that can be used by users in all fields . Problems for Centrex implemented individually based on such specific users are as follows:

First, when a BG contains different types of users, since different types of user could not be placed in the same group, if a user in a user group needs to call a user of a different user group in a same business group, the user has to dial the long number. And, the cost of dialing the Long Number is much higher than that of dialing the Short one (oftentimes dialing Short number is free of charge) . Therefore, the BG users have to pay additional fees for calling a different type of user even if they are in the same group, which increases the financial burden of a BG.

Second, since each type of user has a separate Centrex, it is necessary to provide the operator function for each Centrex, which leads to the need of a plurality of operators as well as inconvenience when the number is queried in a same BG due to the fact that these operators are located separately and cannot be unified. Besides, not only operators, but the management platform and the billing system of these Centrex should also be provided separately, which complicates the management platform and the billing system in a BG.

Furthermore, such Centrex implemented individually based on the specific user is hard to be extended to that of multi- device suppliers, so currently it can not meet the requirements, proposed by domestic operators, of the WAC (Wide Area Centrex) consisting of the devices which provided by various suppliers, used for different users and distributed over wide areas.

Summary of the invention The major object of the present invention is to provide a method for implementing VPBX, which enables different types of users in a same BG to be placed in one user group, thereby allowing the users in the BG to communicate with each other by calling a small number and providing a unified operator, management platform and billing system in the same BG.

The above object of the present invention will be achieved through the following technical solution: a method for implementing the VPBX which is applied to a communication network that comprises at least one application server and at least one soft switch, said method comprises steps as follows :

(1) defining an attribute for users in a BG;

(2) mapping said attribute to an Access Code (AC) ; (3) inserting said AC in front of a dialed number; and

(4) routing said soft switch to said application server based on said AC .

Wherein said application server has a VPBX function.

In step 4, after said call is routed to said application server, said application server inserts a second AC in front of the dialed number. Said second AC is mapped to a caller category by said soft switch.

Said soft switch may be a narrow-band soft switch, or a broad-band soft switch which usually used to be connected to an SIP (Session Initial Protocol) user. A Media Gateway (MG) controlled by said soft switch is usually used to couple said soft switch to a server in a PSTN.

The users using SIP may be connected to said application server via a safe gateway. The users using SIP may also be connected to said application server via a broad-band soft switch or an SIP server.

The present invention further provides a communication system employing the said method. The communication system comprises at least one application server and at least one soft switch, wherein said switch has a detection unit that determines whether an AC is in front of the dialed number, and if it is, the call is routed to said application server, wherein said application server has a Service Logic unit which is used by the call routed from said soft switch to said application server .

It can be seen that the method for implementing the Centrex in accordance with the present invention has the following advantages and features:

(1) A unified Centrex function is able to be implemented by using the method of the present invention. Therefore, the unified Centrex features can be used by all access types of users, such as SIP users, MGCP users, H.248 users, ISDN users and analog users POTS, i.e. various users in the same BG can all use the Service Logic in the application server, can all dial the small numbers in the group and can all show the small numbers, which significantly reduces the cost of calls between different types of users in one group.

(2) The unified Centrex function can be implemented by using the method of the present invention. Rather than providing a respective operator for each type of the Centrex, only one operator is needed for an application server to use the Attendant Service thereof, which is good for centralized maintenance and management of the numbers in a BG. There is also no need to arrange separate management platforms and billing systems for each type of users, and one unified management platform and billing system based on the application server would work.

(3) The present invention not only applies to users in fixed networks, but also to those in Fixed-Mobile Convergence networks based on the IP Multimedia Subsystem (IMS) . It will facilitate the introduction of new technology, new functions, and network upgrading.

(4) For various types of users the present invention facilitates extension to the Centrex of multi-device suppliers, which would meet the requirements, proposed by domestic operators, of the WAC consisting of the devices provided by various suppliers, used for different users and distributed over wide areas, and the users could make full use of the existing functions in the Hosted Office.

Brief Description of the drawings

Figure 1 is a structure diagram depicting an embodiment of the present invention;

Figure 2 is a signal flow diagram of one embodiment in accordance with the present invention, depicting the process of one user at EX EWSD (Exchange EWSD) calling another user using a MGCP-based soft switch (MGCF hiE9200) or a third- party user; Figure 3 is a signal flow diagram of another embodiment depicting the process of one user at EX EWSD calling another user using SIP.

Detailed description of the embodiments

The present invention will be described in detail with reference being made to the accompanying drawings, and these specific embodiments are illustrative rather than limiting.

Figure 1 is a structure diagram depicting an embodiment of the present invention. In the structure of figure 1, hiQ4200 is an application server for implementing a unified Centrex and connected directly to hiE9200 via an SIP interface. Since there is no interface connecting hiQ4200 to EWSD directly, hiE9200 couples EWSD to hiQ4200, and thus the EWSD user could use Service Logic in hiQ4200. In PSTN, EWSD could be connected to a digital telephone ISDN, or an analog telephone POTS. IAD and AG are connected to soft-switching hiE9200 via MGCP protocol and H.248 respectively. A SIP server connects an SIP-based client to an application server. Figure 2 is a signal flow diagram of the first embodiment in accordance with the present invention, depicting a process of one user at EX EWSD in PSTN calling another user using an MGCP-based soft switch (MGCF hiE9200) or a third-party user. According to the method of the present invention, when EWSD user and hiE9200 user are registered at hiQ4200 to become unified Centrex users, adding an attribute to each of them by a human-machine command enables the EWSD user and hiE9200 to be a virtual user of hiQ4200 respectively and the caller and the said called user of MGCF hiE9200 belong to the same BG.

Since the Centrex is used, each user is characterized by one telephone with two numbers, i.e. an actual PSTN number and virtual PNP (Private Numbering Plan) "Extension" number. The caller has a large number (A-PSTN-No, herein using A to represent the caller) as well as a small number (A-PNP-No) . Similarly, the called subscriber also has a large number (B- PSTN-No, herein using B to represent the called subscriber) as well as a small number (B-PNP-No) . When an EXl EWSD user initiates a call as a calling party, he usually uses an ISUP (ISDN User Part) protocol in the SS7 (Signal System 7) . The ISUP protocol provides the signal function for ISDN voice, data, text and image service. When using the ISUP protocol, the EXl EWSD first initiates an IAM (initial Address Message) to MGCF hiE9200. The IAM message comprises a Calling Party Number (CgPN) and a Called Party Number (CdPN) . Here the CgPN is a large number A-PSTN-No and the CdPN is a small number B- PNP-No. As mentioned above, after the EWSD user and the hiE9200 user have been added an attribute respectively by the human-machine command, the calling party device EWSD adds a Prefix (also known as Access Code, AC) in front of the CdPN B-PNP-No and hence the CdPN becomes prefix B-PNP-No.

After hiE9200 receives said IAM message, according to the prefix, it routes the call to hiQ4200 first and then performs signal conversion, i.e. SS7 signal is converted to SIP signal . After hiQ4200 receives said signal, first it should determine which BG the caller belongs to according to the A-PSTN-No, then determine whether the called subscriber is the user in the BG according to the B-PNP-No. If it is determined that the caller is in a BG and the called subscriber also belongs to the same BG, the Centrex number conversion service is performed at the hiQ4200 according to its existing Service Logic, i.e. A-PSTN-No is converted to A-PNP-No and B-PNP-No is converted to B-PSTN-No, then the other Service Logic of the Centrex can be performed in hiQ4200, such as call forwarding, call waiting or other services. Subsequently, hiQ4200 adds another prefix in front of B-PSTN-No identifying that the call is triggered by hiQ4200 and hence preventing the called subscriber from triggering it again. Then hiQ4200 sends an SIP signal, in which the CdPN is a B-PSTN-No and the CgPN is an A-PNP-No, to hiE9200. In addition, a parameter P- Ass-Id in the SIP signal should also carry a B-PSTN-No of the CgPN.

If the called subscriber is an MGCF hiE9200 user, an A-PNP-No will be shown in the called subscriber telephone. If the called subscriber is a third party user, such as a user using another EWSD, the SIP signal will be converted to SS7 signal again in hiE9200. hiE9200 maps the prefix before B-PSTN-No of said CdPN to a caller category when the ISUP protocol in SS7 is used, and Terminal Office will route the call to the called subscriber instead of inserting an AC before CdPN after receiving this special caller category. And then, hiE9200 sends an IAM message to a third party EWSD user. In the IAM message, the B-PSTN-No of the called subscriber is stored in CdPN domain, the A-PSTN-No of the caller is stored in CgPN domain, and the AdCgPN (Additional Calling Party Number) of the caller stored in GN (Generic Number) parameters is a A-PNP-No.

Figure 3 is a signal flow diagram of the second embodiment depicting the process of one user at EX EWSD calling another user using an SIP protocol in accordance with the present invention .

In Figure 3, firstly, the EWSD user and the SIP user are respectively defined an attribute. Media from EWSD are processed by MG hiG1200 controlled by MGCF hiE9200, while signals from EWSD are centralized to be processed by hiE9200. The EWSD user data usually use a TDM (Time Division Multiplexing) Frame Relay network and the TDM network will be converted to an IP-based network by hiG1200. The signal process in the figure 3 is as follows: first, EWSD uses the ISUP protocol to initiate a call by sending an IAM message to hiE9200, and then hiE9200 controls hiG1200 via the MGCP protocol and sends a Create Connection command (MGCP CRCX) to hiG1200 with the purpose of creating a speech path connection between the EWSD user and hiG1200. In this connection, hiG1200 is required to send some caller parameters to hiE9200, thus hiG1200 sends to hiE9200 an answer message MGCP 200 OK /w Session Description Protocol (SDP) based on a session parameter description in SDP, and the answer message carries some caller parameters, such as IP address of the caller, encoding and decoding information of the caller and so on .

After hiE9200 receives said MGCP 200 OK /w SDP message, it first routes the call to hiQ4200 according to the prefix, then it performs the signal conversion, i.e. SS7 signal is converted to SIP signal. hiE9200 initiates an SIP INVITE message (SIP INVITE /w SDP) to hiQ4200, and then hiQ4200 returns an SIP 100 Tying message as an answer. hiQ4200 performs the Centrex number conversion service according to its existing Service Logic, i.e. A-PSTN-No is converted to A- PNP-No and B-PNP-No is converted to B-PSTN-No, then the other Service Logic of the Centrex, such as call forwarding, call waiting or other services, could be performed in hiQ4200.

When hiQ4200 looks for the called subscriber according to the called number carried in the INVITE message. Since herein the called subscriber is an SIP user, hiQ4200 initiates an SIP INVITE message (SIP INVITE /w SDP) to the SIP server to which the SIP user belongs, and then the SIP server returns an SIP 100 Tying message as an answer. If the SIP user is idle, the SIP server will return a ringing message SIP 180 Ringing to hiQ4200 which returns the same ringing message SIP 180 Ringing to hiE9200 again, and then hiE9200 continues to return ISUP Address Complete Message (ISUP ACM) in the SS7 protocol to the EWSD user.

hiE9200 sends a Modify Connection command (MGCP MDCX) to hiG1200 again to request hiG1200 to provide the caller with a ring back tone, and hiG1200 returns an answer message MGCP 200 OK to hiE9200 immediately. The EWSD user then would hear the ring back tone.

Subsequently, the called SIP subscriber picks up the phone, the SIP server sends an SIP 200 OK message to notify hiQ4200, and hiQ4200 returns an SIP ACK message to the SIP user for confirmation. Then hiQ4200 sends another SIP 200 OK message to notify hiE9200, and hiE9200 also returns an SIP ACK message to hiQ4200 for confirmation. hiE9200 sends an Answer Message (ANM) in the SS7 protocol to the EWSD caller again. Then hiE9200 sends a MGCP MDCX command to hiG1200 again to request hiG1200 to stop providing the caller with the ring back tone, and at the same time, parameters of the called subscriber, such as IP address of the called subscriber, encoding and decoding information of the called subscriber and so on, are notified to hiG1200 by the command. hiG1200 connects the caller based on circuit switching to the called subscriber based on IP according to the acquired IP addresses of the caller and the called subscriber. Then hiG1200 returns an answer message MGCP 200 OK to hiE9200 immediately.

At this time the call between the caller and the called subscriber could begin. After calling, if the called SIP user hangs up, the SIP server will send an SIP BYE message to notify hiQ4200, and hiQ4200 will return an SIP 200 OK message to the SIP server for confirmation. Subsequently hiQ4200 sends another SIP BYE message to notify hiE9200, and hiE9200 also returns an SIP 200 OK message to hiQ4200 for confirmation. Then hiE9200 sends Release Message (REL) in the SS7 protocol to the EWSD caller again. hiE9200 sends a Delete Connection command (MGCP DLCX) to hiG1200 for the purpose of releasing the speech path between the EWSD user and hiG1200 thereby making it available for other calls. Then hiG1200 sends an answer message MGCP 250 OK to hiE9200 immediately. After the EWSD caller releases the path, EWSD sends Release Complete Message (RLC) in the SS7 protocol to hiE9200.

The foregoing two examples are both in full-trigger mode, i.e. for the caller and the called subscriber coming from different domains, the call is routed to the application server hiQ4200 provided that the caller picks up the phone and dials the small number of the called subscriber, and this process is known as full-trigger. Furthermore, in the foregoing two examples, there is only one application server and the caller and the called subscriber are in the same BG

(known as Group for short, i.e. in-group call), and the dial- up mode of dialing the small number is used, and the caller triggers the call to the application server by inserting a prefix (AC) before CdPN so as to use Service Logic in the application server. Then the application server inserts the second prefix (AC) in front of the CdPN to indicate that the call has triggered the application server and prevent the called subscriber from triggering it again after the caller has triggered it, and then the call is routed to the called subscriber via the application server.

Besides the foregoing two full-trigger examples, more embodiments in full-trigger mode in accordance with the present invention will be described below.

In the embodiment 3, there is only one application server, and an in-group user calls an ex-group user (ex-group call out) using the dial-up mode of dialing 9 + large number. In this embodiment, the call is also triggered to the application server by the caller to use the Service Logic in the application server.

In the embodiment 4, there is only one application server, and an ex-group user calls an in-group user (extra-group call in) using the dial-up mode of dialing the large number. With a prefix (AC) being inserted in front of the called number, the call is triggered to the application server by the called subscriber, instead of being triggered in the process of being sent from the caller to the called subscriber, and Service Logic in the application server is used. Then the application server inserts the second prefix (also known as AC) in front of the CdPN to indicate that the call has been triggered to the application server and prevent the called subscriber from triggering it repeatedly.

In the embodiment 5, there is only one application server, and the caller and the called subscriber are in different BGs (inter-group call) and the dial-up mode of dialing 9 + large number is used. With a prefix (AC) being inserted in front of the called number, the call is triggered to the application server by the caller, and Service Logic in the application server is used. And then, the application server inserts the second prefix (also known as AC) in front of the CdPN to indicate that the call has been triggered to the application and prevent the called subscriber from triggering it again, then the call is routed to the called subscriber.

In the embodiment 6, there are two application servers and the caller and the called subscriber are in different BGs (an inter-call between two application servers), which is equivalent to the combination of the embodiment 3 and the embodiment 4. Firstly, the caller triggers the call to the application server, using the dial-up mode of dialing 9 + large number, and uses Service Logic in the application server 1. With a prefix (AC) being inserted in front of the called number, the call is triggered to the application server 2 by the called subscriber instead of being triggered in the process of being sent from the caller to the called subscriber, and Service Logic in the application server 2 is used. Then the application server 2 inserts the second prefix (also known as AC) in front of the CdPN to indicate that the call has triggered to the application server 2 and prevent the called subscriber from triggering it repeatedly.

Considering the EWSD user has been used widely, a half- trigger mode may also be used when the unified Centrex is implemented practically. The half-trigger is that the call does not need to trigger to the application server hiQ4200 and the EWSD Service Logic is still used in the case that a group contains multi-domain users and the caller and the called subscriber are both the EWSD users; only when either the caller or the called subscriber is not an EWSD user, the call needs to be triggered to the application server hiQ4200 and the hiQ4200 Service Logic is used. Because the application server is not always triggered using the method, it is known as half-trigger. Whether the full-trigger or the half-trigger should be used in a particular application depends on the needs of the user. Some half-trigger embodiments are described as follows.

The embodiment 7 is the case that the caller and the called subscriber are both the EWSD user in the embodiment 1. Herein the EWSD Centrex Service can be used, but when either the caller or the called subscriber is not an EWSD user, the process and the signal flow process as the embodiment 1 described are still to be used.

The embodiment 8 is the case of an in-group call. For example, when one EWSD user dials another EWSD user, using the dial-up mode of dialing a small number, the Centrex Service Logic in EWSD will be used and the call does not need to be triggered to the application server. For another example, when an EWSD user dials a soft-switching hiE9200 user, using the dial-up mode of dialing a small number, the EWSD caller triggers the call to the application server in which Service Logic is used. For still another example, when an EWSD user dials an operator, using the dial-up mode of dialing "0", the Centrex operator logic in EWSD will be used and the call does not need to be triggered to the application server, while all operator functions can be maintained. Alternatively, when an EWSD user dials an operator in the application server, using the dial-up mode of dialing "10", the caller will trigger the call to the application server in which the operator logic is used, and the operator in the application server can provide some services, such as Call Transfer (CT) , Automatic Call Design (ACD) , Address Query (AQ) etc, and the range of Number Query provided is significantly wider than that provided by the Centrex operator.

The embodiment 9 is the case that an in-group EWSD user dials an extra-group user (extra-group call out) , using the dial-up mode of dialing a large number 9+, the Centrex Service Logic in EWSD is used and the call doesn't need to be triggered to the application server.

The embodiment 10 is the case that an ex-group EWSD user dials an in-group user (ex-group call in) , using the dial-up mode of dialing a large number. One case is that the EWSD user dials the EWSD user directly; another case is that the EWSD user dials the operator first, and then the operator transfers the call to the corresponding user. In the above two cases, the calls both use the Centrex Service Logic in EWSD and do not need to be triggered to the application server .

The embodiment 11 is the case that some special services are provided by using EWSD, such as Call Restriction, Call Screening which is set by the operator through the human- machine command. Call Transfer uses the dial-up mode of dialing a number as "R*12*transfer number#"; Call forwarding uses the dial-up mode of dialing a number as "*57*forwarding number#"; Group Call Pick UP uses the dial-up mode of dialing a number as "*11#". Such servers are provided by using the Service Logic in EWSD.

In the above embodiments, it will be recognized that a unified virtual Centrex function is able to be implemented by using the present method. The unified Centrex features can be used by all access types of users, such as SIP users, MGCP users, H.248 users, ISDN users and analog users POTS, i.e. the various users in the same BG can all use the Service

Logic in the application server, can all dial small numbers in the group and show the small numbers, which significantly reduces the cost of calls between different types of users in one group. At the same time, rather than providing a respective operator for each type of the Centrex, only one operator is needed for an application server to use the Attendant Service thereof, which is good for centralized maintenance and management of the numbers in a BG.

Claims

1. A method for implementing VPBX, which is applied to a communication network, that comprises at least one application server and at least one soft switch, said method comprising steps as follows:

(1) defining an attribute for users in a BG;

(2) mapping said attribute to an AC;

(3) inserting said AC in front of a called number;

(4) routing said soft switch to said application server based on said AC.

2. The method for implementing VPBX of claim 1, wherein said application server has a VPBX function.

3. The method for implementing VPBX of claim 1, wherein in step (4), after said call is routed to said application server, said application server inserts a second AC in front of the called number.

4. The method for implementing VPBX of claim 3, wherein said second AC is mapped to a caller category by said soft switch.

5. The method for implementing VPBX of claim 1, wherein said soft switch is a narrow-band or a broad-band soft switch .

6. The method for implementing VPBX of claim 1, wherein said soft switch controls an MG (Media Gateway) .

7. The method for implementing VPBX of claim 1, wherein the users using Session Initiation Protocol (SIP) are connected to said application server via a safe gateway.

8. The method for implementing VPBX of claim 1, wherein the users using Session Initiation Protocol (SIP) are connected to said application server via a broad-band soft switch, or an SIP server.

9. A communication system employing said method, which comprises at least one application server and at least one soft switch, wherein said soft switch has a detection unit that determines whether an AC is in front of the dialed number, and if it is, the call is routed to said application server .

10. The communication system of claim 9, wherein said application server has a Service Logic unit which is used when the call is routed from said soft switch to said application server.