KR100539732B1 - Call server structure for lan telephony - Google Patents

Abstract

The present invention relates to a structure of a call server for a LAN telephony, and in particular, in the conventional technology, since the voice data is transmitted in a unicast manner, when a multi-party call or a multi-call becomes frequent, the traffic of the network increases rapidly. There was a problem that caused loss and disconnection. Also, as the number of users increased, the call server was placed in various places to perform the LAN telephony system service. There is a problem in user database management, signaling, and quality of service (QoS) management among them, and it is difficult for the call server to obtain the status information of another subnetwork, Also, we need to go through a router to find the call server. There was a problem that caused a lot of network overhead.

Therefore, the present invention solves the problems caused by the gradual network traffic increase even if the number of calls increases because multiple calls or 1: 1 calls generate the same network traffic, and due to the LAN telephony system existing in different subnetworks, The problem of using cast is to solve the problem of unicast transmission due to network location problem by delivering voice data to LAN telephony systems in other subnetworks using multicast method using RTP multicast tunneling technique through call server. This ensures overall reliability, flexibility and scalability, which in turn provides a better voice service for the user.

Description

Call Server Structure for LAN Telephony {CALL SERVER STRUCTURE FOR LAN TELEPHONY}

The present invention relates to a structure of a call server for a LAN telephony, and more particularly, to a structure of a call server for a LAN telephony to efficiently manage multicast voice data transmission and multiple call servers in a LAN telephony environment.

Recently, as the expansion of the data network due to the development of the high speed computer network and the multimedia having the continuous and explosive nature occupy the communication channel, the systems and the computer network have developed a new area of integration of the computer network according to various needs of users.

Among them, voice and data networks are representative examples of various computer networks such as packet network, integrated information network (ISDN), integrated telephone network (GSTN), telephone, fax, 700 voice information service, wireless calling service, Internet phone service, phone banking service, and call center. Voice services, such as dedicated UnPBXs, are now available, and this network integration allows the phone to emerge as another application terminal device on the network, just as a computer functions as a server in a typical office environment. Through this, it is possible to search various information resources of the network and convert it to voice or fax format and output it.

In other words, TCP / IP-based LAN was selected as an environment that satisfies these factors, which is not only most suitable for extending the interworking and functioning with the Internet, but also for applying the concept of PC-based telephone private network. Therefore, as voice data is transmitted on LAN and various voice services are provided, the service area of call server for LAN telephony has gradually expanded, and thus, a large number of call servers are needed. There is a need for an interface technology that can be used.

Currently, the LAN telephony system adopts a method of loading voice data in an RTP packet to provide a real time service while maintaining the characteristics of the voice data when transmitting voice data, and unicast without passing through a call server. Terminals send and receive data directly.

Therefore, if multi-party or multi-call is frequent due to the characteristics of the unicast method, the traffic of the network may increase rapidly, causing loss of voice data and disconnection. In order to manage all users, there are difficulties in user registration, address conversion, and user information management, and there is a bottleneck in service provision. Therefore, users have a call server in various places and users register with a call server suitable for them.

On the other hand, in the multi-call server system environment as described above, regardless of the registered call server, a seamless interface must be formed between call servers to receive all services provided by each call server.

However, in the conventional general LAN telephony system, since the operation method for the multiple call servers is ambiguous and only partially sets and uses the interface, problems in user database management, signaling, and quality of service (QoS) management are revealed among them.

That is, when different users with different subnetwork addresses try to find a call server or receive a service in a different area, or when the call server understands the status of the network of the user's system in another subnetwork and determines the quality of service. Since it is difficult to obtain status information and the user has to go through a router to find a call server, the broadcasting technique has a lot of network overhead, which makes it difficult to manage.

1 is a diagram illustrating a conceptual configuration of a conventional LAN telephony system. As shown in FIG. 1, when a user makes a call to an arbitrary terminal connected to a LAN network, a signal transmitted through the telephone network is transmitted through a gateway. It is converted and delivered to the call server, and the call server searches for the other terminal to be connected and connects to the unicast method if the subnet area is managed by itself.

However, if the other terminal is a subnet area managed by another call server, the other terminal obtains information on the terminal through the other side call server and connects directly through a router.

As described above, in the conventional technology, since the voice data is transmitted in a unicast manner, when the multi-party call or the multi-call is frequent, the traffic of the network increases rapidly, causing loss of voice data and disconnection. In addition, LAN telephony system services are performed by placing call servers in various places as the number of users increases, and the operation method for these multiple call servers is ambiguous and only partially sets the interface. There is a problem in signaling and quality of service (QoS) management, and it is difficult to obtain the service of a call server in another area, or it is difficult for the call server to obtain status information of another subnetwork, and also has to go through a router to find the call server. The problem that causes network overhead there was.

Accordingly, the present invention has been made to solve the above-mentioned problems, and improves the performance of the system by reducing network traffic through a transmission method using multicast of voice data in a LAN telephony environment, and in a multi-call server environment. By providing smooth communication between call servers, LAN telephony users can receive all call server services, and provide a call server structure for LAN telephony that performs integrated and efficient management of each call server area. There is a purpose.

The present invention for achieving the above object is a distributed component unit that provides a basis for performing integrated management of distributed call servers; A call server operating unit for operating a call server using the distributed component unit; A distributed interface unit for interfacing mutual resources between distributed component units to control a client or a server in the call server operating unit; A database for storing user management information; A tunneling unit performing a multicast tunneling procedure for data transmission between distributed call servers; Characterized in that consisting of the operating interface unit for the protocol interface with the system operating system.

Hereinafter, an embodiment according to the present invention will be described in detail with reference to the accompanying drawings.

Once the LAN telephony system is functionally divided, it can be divided into signaling, voice transmission, and supplementary services, and structurally, integrated management between the call server operation unit under the management of the telephone network and the call server and distributed call servers It can be divided into areas for.

First, interworking with a general telephone network is performed through a gateway, which is controlled by a call server, and a network area under the management of a call server is called a zone, and terminals in the zone support the H.323 standard. If the other party asks the call server of the system that applied for the voice service, whether the other user is under the management of any call server, the management call server asks for the location of the other user in signaling or performs signaling for the voice service request. It relies on distributed components to search for user databases in multiple locations and inform you of the results.

In other words, through distributed components, integrated management of call servers distributed in various places is performed and the status of the entire LAN telephony system is managed.

In addition, the use of distributed components allows new call servers to be added to the distributed components only through the registration process, making system expansion very easy.

Figure 2 is a block diagram showing the structure of a call server according to the present invention, a distributed component unit 10 that provides a basis for performing integrated management of distributed call servers as shown therein; A call server operating unit 20 for operating a call server using the distributed component unit 10; A distributed interface unit 30 for interfacing mutual resources between the distributed component units 10 so as to control a client or a server in the call server operating unit 20; A database 40 for storing user management information; A tunneling unit 50 for performing a multicast tunneling procedure for data transmission between distributed call servers; It consists of the operating interface unit 60 for the protocol interface with the system operating system (OS), the following describes the operation and operation of the present invention configured as described above.

First, when the distributed component unit 10 requests the address of the user who is called in the LAN telephony system, the call server operating unit 20 calls the procedure of the client user interface 10a to call the address of the called user. Internal server interstub 10b which is responsible for requesting and retrieving an address from each distributed object database by a procedure for finding a counterpart address internally. The procedure of Server Inter Stub) is called through RPC channel 10c, and the procedure of Server Interstub 10b executes a command to find an address with server interstubs of several call servers distributed and obtains the result. It comes to play.

Next, the COM library, RPC channel, SCM (s), registery, Class factory, Object, and Class object table which are not described above are basically used for interworking with the client interfacing unit 10a and the server intertubing unit 10b. Functions provided by distributed components.

Next, the distributed interface unit 30 includes a client control unit 30a for interlocking client components; It consists of a server control unit (30b) to enable the server component to interwork.

Next, the call server operation unit 20 is responsible for all the interworking between the client control unit 30a and the server control unit 30b, and the call server control unit 20a for performing call information management or address change or conference management; A registration processing unit 20b for processing registration and authorization or status messages of the call server; A media control channel manager 20c for collecting, analyzing, and generating signaling that specifies and promises a compression method, a session state, and a transmission rate of media to receive voice service; A network manager 20d for service operation and quality management according to a user level; The call routing unit 20e analyzes and collects the signaling message and delivers the message about the result to the user system. The other user server is managed by another call server, so that voice data can be delivered through a multicast transmission method. If not, the call server operator 20 sends the multicast voice data sent by the caller by the call server controller 20a to the tunneling unit 50 to tunnel to the call server of the counterpart user. The voice data is received through the tunneling unit 50 to transmit voice data through multicast to its zone.

More specifically, the call server controller 20a is a place where all control operations of the call server are performed. Address translation control, additional service provision, call table management, multipoint conference management, service operation management according to user level, zone management, and RTP Tunneling control, and the like, the registration processing unit 20b receives the call for registration of users under the management of the call server itself, a request for permission according to the voice service request, and an information message about the system status. It delivers to the server control module, receives the result, generates a message, and delivers it to the requester.

In addition, the call routing unit 20e analyzes and collects the signaling message requested by the user system, delivers it to the call server controller 20a, generates a message about the result, and delivers the message to the user system. When the call server is managed, it performs the function of delivering the requested signaling to the counterpart user.

3 is an exemplary view showing a conceptual configuration of a system according to the present invention. As shown in FIG. 3, the most important difference is that other calls may be made in a multicast manner in which a network traffic is not burdened in the prior art. Since all the terminals are connected in a unicast manner because the router for connecting with the terminal managed by the server does not support multicast, in the present invention, the terminal transmits in multicast with the terminal in its own network managed by each call server. This is possible by transmitting data through RTP tunneling between the call servers without direct connection with terminals in the network managed by other call servers, and by transmitting voice data in a multicast manner to the terminals managed by the call servers. It was.

FIG. 4 is a flowchart illustrating a processing process when a user makes a registration request to a call server according to the present invention. When a user A makes a registration request, the call server controller receives and analyzes a request message from the registration processing unit 20b. If it passes to 20a to check whether the maximum number of user registrations has been exceeded, the registration rejection command is transmitted to the registration processing unit 20b to transmit a registration rejection message to the user A.

However, if the maximum allowable number of registrations has not been exceeded yet, it is checked whether the requested user A is a suitable user for the call server. In this case, it is determined whether or not it is already registered, and if not yet assigned, a user class is assigned, stored in a database, and then a registration permission command is sent to the registration processing unit 20b to deliver a registration permission message to the user A for registration. To complete.

5 is a flowchart illustrating a service execution process when a call service is requested according to the present invention. When user A requests a call from user B, registration processing unit 20b analyzes the request message and executes a call server controller ( 20a).

Accordingly, the call server control unit 20a requests the client control unit 30a to check whether the information about the call request is valid, and if the user B executes a command of the client interrogation 10a, the RPC channel. Through 10c, the server interstub 10b instructs the call server control unit 20a to search the database and executes a search command to other call servers registered in registery to perform a search command of the user B.

Accordingly, the information about the searched result is transmitted to the call server controller 20a through the server controller 30b, the server intertub 10b, the client interface 10a, and the client controller 30a. If it is determined whether the user is a service allowable or not, the service rejection command is transmitted to the registration processing unit 20b to transmit a rejection message to the user A.

However, if the user B is a serviceable user, this time, it is determined whether two users use the same call server. If the user B is the same call server, the call related information is stored in the service related information table, and the tunneling unit 50 is disabled. The permission message is transmitted to the user A through the registration processing unit 20b. If another call server is used, the RTP tunneling information is transmitted to the tunneling unit 50 and the permission message is transmitted to the user A.

Next, the call routing unit 20e receives the call signal message, analyzes it, and delivers it to the call server controller 20a to determine whether it is internal routing or external routing.

Accordingly, in the internal routing, the call routing unit 20e delivers the call signal message to the user B, and the media control channel manager 20c receives and analyzes the media related signal information and analyzes the call server controller 20a and the user B. ) And exchange related information, then perform voice service by multicast.

On the other hand, in the case of external routing, the call server control unit 20a transmits the external routing command of the call signal and the information on the call related table to the client control unit 30a, and instructs the client interrogation command to route the signaling message to the destination call server. Get off.

On the other hand, the client inter-proxy instruction routes the actual call signaling message of the server interstub 10b through the RPC channel to issue a simple call instruction to the destination call server through the call related table information.

Accordingly, the destination call server receives a command through the server control unit and transmits a command to the call routing unit to deliver a call signal message to the user B.

Accordingly, the call server for each user (A, B) exchanges various media-related information while storing and managing session information on the current call service request, and transmits data to each other's call server in a tunneling manner and manages itself. The service is performed by transmitting voice data to each user in a multicast manner.

As described above, in the LAN telephony system composed of a call server according to the present invention, since multiple calls or 1: 1 calls generate the same network traffic by using a multicast transmission method, even if the number of calls increases, the network traffic increases gradually. Problems that occur are solved, and the problem of using unicast due to the LAN telephony system existing in other subnetworks is that the multicast method is used for the LAN telephony system existing in other subnetworks using RTP multicast tunneling technique through a call server. By transmitting the voice data, the unicast transmission problem caused by the network location problem is solved, thereby ensuring the reliability, flexibility, and scalability of the entire system, thereby providing a better voice service to the user.

1 is an exemplary view showing a conceptual configuration of a conventional general LAN telephony system.

Figure 2 is a block diagram showing the structure of a call server according to the present invention.

3 is an exemplary view showing a conceptual configuration of a system according to the present invention.

Figure 4 is a flow chart showing a process when a user makes a registration request to a call server according to the present invention.

5 is a flowchart illustrating a service performing process when a call service is requested according to the present invention.

*** Description of the symbols for the main parts of the drawings ***

10: distributed component unit 20: call server operation unit

30: distributed interface unit 40: database

50: tunneling unit 60: operating interface unit

Claims (2)

A distributed component unit providing a base for performing integrated management of distributed call servers; A call server operating unit for operating a call server using the distributed component unit; A distributed interface unit for interfacing mutual resources between distributed component units to control a client or a server in the call server operating unit; A database for storing user management information; A tunneling unit performing a multicast tunneling procedure for data transmission between distributed call servers; A call server architecture for a LAN telephony, comprising an operating interface unit for a protocol interface with a system operating system. According to claim 1, wherein the call server operating unit is responsible for all the interworking between the client control unit and the server control unit and the call server control unit for performing call information management or address change or conference management; A registration processing unit for processing registration and authorization or status messages of the call server; A media control channel manager for collecting, analyzing, and generating signaling that specifies and promises a compression scheme, a session state, and a transmission rate of media to receive voice service; A network management unit for service operation and quality management according to a user level; Call server for LAN telephony, characterized in that consisting of a call routing unit for analyzing and collecting the signaling message and delivers the message about the result to the user system.