KR100813687B1 - SAPI Media Gateway Controller - Google Patents

Abstract

The present invention relates to a SIP media gateway controller capable of interworking various types of VoIP media gateways based on different protocols with a SIP-based network. This invention, SIP PSM for operating the SIP protocol; A call converter which performs an intermediate setup and association function of a call between the SIP PSM and the PSM to be described later; A PSM having input / output associated with a related MG and the call converter, and storing basic data necessary for a protocol; A call list module which holds local call list index information, basic identification information, and protocol type information in the PSM and abstracts and transmits information of a counterpart PSM in a desired form through each call converter; An MG status manager which stores the state of the MG and the state of a user terminal connected to the MG and periodically reports the state of the MG so as to manage the state of the MG managed by the SIP MGC; This is accomplished by a call router to allow the SIP MGC to operate its own fixed route without a proxy server.

Description

SIP MEDIA GATEWAY CONTROLLER}

1 is a block diagram of an H.323 based telephone network according to the prior art.

2 is a block diagram of a MGCP based telephone network according to the prior art;

3 is a block diagram of a MEGACO based telephone network according to the prior art.

4 is a block diagram of a SIP media gateway controller in accordance with the present invention.

5 is a block diagram of a SIP-based VoIP system using SIP MGC.

Figure 6 is a block diagram showing the structure and operation of the call list.

7 is a block diagram showing a virtual call intermediary operating example in the present invention.

*** Explanation of symbols for the main parts of the drawing ***

40: SIP MGC 41: SIP PSM

43: PSM 44: Call List Module

45: MG Status Manager 46: Call Router

The present invention relates to a SIP media gateway controller capable of interfacing various types of VoIP media gateways based on different protocols with a Session Initiation Protocol (SIP) -based network, and in particular, various types based on different protocols. Defines the structure of the SIP media gateway controller that can interoperate VoIP media gateways with the SIP-based network, the next-generation VoIP protocol, and determines the call handling method between the controller and the gateway. The present invention relates to an SIP media gateway controller capable of receiving various additional service functions supported by a SIP call processing server.

The reason why a separate telecommunications company can offer a much lower rate than existing international call rates is because the international telephone base network is based on a dedicated Internet line. A typical one-to-one line network system allocates a certain bandwidth to one call, which is expensive. However, when a packet network transmission line is used as a packet-based transmission line, a number of calls can be shared by a certain bandwidth. As a result, a service can be provided at a low cost because a service for many calls is always possible with a high bandwidth share.

In a packet network based on this leased line, two terminals ultimately maintaining one call must have a physical connection point with the packet network. This is because the protocol between the packet network and the terminal is different, because TCP / IP, which is mainly used in the Internet, is applied to the packet network, but the public communication network (PSTN) is applied to the terminal. The gateway between these two different protocols is called a gateway, and the equipment used for this purpose is called a Voice over IP (VoIP) gateway.

The VoIP gateways are classified according to the characteristics of the data transmission medium. The VoIP gateways are often referred to as VoIP media gateways (hereinafter referred to as "VoIP MG"). VoIP MG usually operates based on TCP / IP. In addition, TCP / IP refers to the most basic infrastructure network, and a telephone-related protocol that substantially runs a gateway exists over TCP / IP. Frequently referring to the protocol of MG, it means this telephony related protocol.

Currently, the most commonly used standard protocol for telephony is H.323, the protocol discussed in ITU-T (see Figure 1). Since this protocol is the first popularized VoIP gateway protocol, it has been adopted as the gateway device of most communication companies. When adding a new gateway device, H.323 gateway device is adopted to maintain compatibility with existing devices.

However, since the H.323 protocol is the result of combining several protocols of ITU-T, there are many factors that consider other packet network environments in addition to the Internet, which is a large or complicated protocol. Thus, H.323 gateways are now adopted by universality, but will shift to other small, low complexity, highly scalable protocols that can flexibly operate calls as more users of packet telephony grow in the future. It is expected to be. In addition, if the bandwidth of the Internet, which is widely used around the world, is sufficiently secured, it is expected that a protocol will be adopted that can provide more call processing and better call quality at low cost without establishing a dedicated line.

Currently, MGCP (see IETF RFC 2705) MEGACO (IETF RFC 3015, see ITU-T H.248), a gateway-specific protocol, is the next generation protocol for media gateways (see FIGS. 2 and 3). MGCP (MGCP) and MEGACO (MEGACO: MEdia GAteway Control) adopt a structure consisting of MG and MGC to control MG, and adopt a small or low complexity protocol. In particular, MEGACO is a new version of MGCP that is also compatible with other protocols, making it more widely available than MGCP. As MEGACO considers the compatibility with other protocols, it is possible to use the existing H.323 gateway as it is and to interwork the gateway-specific protocol with existing Internet terminals or other services. It can reduce the burden.

As another next-generation protocol, SIP (see IETF RFC 2543), an inter-terminal protocol reflecting the characteristics of the Internet, is drawing attention (see FIG. 2). While MGCP and MEGACO are gateway-only protocols, SIP is available to general terminals, so SIP is likely to be used universally in All-IP.

However, MEGACO, which has a small or low complexity protocol, improved media handling of MG, and compatibility with other protocols, can only support where MG supporting MEGACO always exists, so MGC and MG must always be purchased together. There is a problem of cost burden, and there is a problem that cannot be completely applied to an Internet phone terminal or a call processing server that can provide an additional service because it has a limitation and a protocol applied only to a VoIP gateway. This means that even if the call processing server provides intelligent services, service support is not possible without the MEGACO command, which can be one-to-one correspondence with MGC. After all, the key factor in determining the success of the VoIP market is the low communication cost, system construction cost, better call quality, and various additional service support, and all-IP IP address is given to all communication terminals. When predicting that the system is the communication system of the future, MEGACO has a disadvantage that is not future-oriented except for the advantages of interworking with existing telephone systems.

In addition, since the SIP according to the prior art is a general terminal center and the standard is not officially completed yet, there is a shortage of operating a VoIP gateway capable of interworking with an existing telephone network such as MGCP. Accordingly, there is a situation in which a SIP-t protocol capable of interfacing SIP with an existing telephone network is separately discussed.

Accordingly, an object of the present invention is to define a structure of a SIP media gateway controller, which is a next generation VoIP protocol, for several types of VoIP media gateways based on different protocols, and to newly determine a call processing method between the controller and the gateway, and to design a next generation protocol. It is to provide an SIP media gateway controller that can provide various additional service functions supported by SIP call processing server while complementing VoIP gateway which is a weak point of SIP.

A first feature of the present invention is to allow SIP-based SIP MGC to work with various VoIP MGs that support different protocols.

The second aspect of the present invention is to separate the call list in charge of the global call processing in the SIP MGC and the local call list (Local Call List) operated by the corresponding protocol in each protocol stack module to call between the SIP terminal and the MG. The global call list consists of only simple information that specifies the index of the internal call and the type of the corresponding protocol in the local call list of the protocol stack associated with a particular call.

According to a third aspect of the present invention, a SIP MGC uses a call list, which is a call processing list, a virtual MG when connecting a call from a SIP terminal to an MG, and a virtual SIP terminal when a call is connected from a MG to a SIP terminal. By inserting into the protocol, call association between call intermediary and two different protocols of both real call terminals is established to operate.

Figure 4 is a block diagram showing an embodiment of the SIP media gateway controller according to the present invention, as shown therein, SIP PSM 41 operating the SIP protocol in the SIP MGC (40); A call converter 42 for performing an intermediate setting and linking function between the SIP PSM 41 and the PSMs 43A-43K to be described later; Input / output is associated with the associated MG and the call converter 42, and each of the PSMs 43A-43K includes basic data storage necessary for the protocol; It holds only local call list index information, basic identification information, and protocol type information in each of the PSMs 43A-43K, and through the call converter 42, abstracts and transmits the information of the counterpart PSM in a desired form. Call list module 44; An MG status manager 45 which stores the state of the MG and the state of the user terminal connected to the MG so as to manage the state of the MG managed by the SIP MGC 40 and periodically reports the state of the MG; The SIP MGC 40 is configured as a call router 46 for operating a fixed route independently without a proxy server, which will be described in detail with reference to FIGS. 5 to 7 attached to the operation of the present invention. As follows.

The SIP MGC controlling various MGs supporting different protocols according to the present invention has a structure as shown in FIG. 4. That is, the SIP MGC has a call processing module with various MGs supporting different protocols such as SIP, MGCP, H.323, MEGACO, WAP, and the like. When a call establishment request is received from the external SIP to the UE connected to the H.323 gateway, the SIP MGC sends a call establishment request composed of the SIP protocol to the internal H.323 call processing module for processing. First, a detailed operation of each module of the SIP MGC will be described.

The SIP PSM (SIP Protocol Stack Method) 41 is a part of the SIP MGC 40 that operates the SIP protocol. When the MGC communicates with an external network, it always communicates only with the module and the SIP protocol.                     

The call converter 42 performs an intermediate setting and association function of the call between the SIP PSM 41 and the PSMs 43A to 43K to be described later. The call converter 42 does not directly convert a message of each protocol, but analyzes the corresponding protocol in each PSM 43A-43K to update the call-related information of the call list module 44, based on the call processing signal. Call association is handled according to the type of call list and the information of the module 44.

Each of the PSMs 43A-43K is a one-to-one processing module corresponding to the VoIP MG protocol. The input / output of the module is always connected only to the related MG and the call converter 42, and has a basic data storage necessary for the protocol.

Each PSM 43A-43K has a local call list internally for calls that the protocol can handle. However, it is difficult to analyze this information in other PSMs because the information contained here is in the format required for a particular protocol. The PSMs 43A-43K may share information for analyzing call lists of different protocols, but this results in a weakening of the independence of the protocols of the PSMs 43A-43K and the PSMs 43A-43K. It acts as a factor to increase the complexity. Therefore, the call list module 44 shown in FIG. 4 has only local call list index information, basic identification information, and protocol type information in each PSM 43A-43K, and each PSM through the call converter 42. (43A-43K) abstracts and conveys information from the partner PSM in the desired format.

The SIP MGC 40 may set a call route only through a Sip Proxy server according to SIP characteristics. However, due to the nature of the gateway, since a fixed route can be operated independently without a proxy server, the call router 46 is embedded.

The MG status manager 45 manages the state of the MG managed by the corresponding SIP MGC 40. Since the SIP proxy server can manage only the SIP MGC 40, the setting of the MG managed at the bottom of the MGC is entirely. This is because it is the share of the SIP MGC 40. It also stores the state of the MG and the state of the user terminal connected to the MG and periodically receives the status report of the MG as an audit (audition) function.

SIP MGC 40 as described above is configured with a SIP system as shown in Figure 5 to support the VoIP service network operation. The SIP proxy server is responsible for call routing and arbitration in the SIP network. After the call setup is completed, the SIP MGC 40 or the SIP terminal directly transmits and receives media. By applying the structure as shown in FIG. 5, all the MGs of other protocols that are used in the past can be recycled to the SIP system, thereby overcoming interworking with the gateway pointed out as the greatest weakness of the SIP.

The call list module 44 configured in the SIP MGC 40 is a call processing list that can be shared for each protocol module. The call list module 44 corresponds to the global call processing list from the SIP MGC 40 position, and the index of the call of the local call processing list embedded in each PSM (C), the basic information of the call, and the PSM The protocol type of (43A-43K) is recorded.

The PSMs 43A-43K internally maintain call processing information and local call processing lists in a format that the protocol can recognize. This is a method for improving independence and lowering complexity in a module by abstracting information shared between each PSM 43A-43K and the SIP PSM 41.                     

For example, if these call lists and local call processing lists are not used, when the PSM-1 43A makes a call to the SIP PSM 41, the PSM-1 43A provides a data format called psm1CallID in the protocol. If this is used as information required for call processing, the SIP PSM 41 should be aware of this data format. However, if both the format and data format of the protocol of the PSM-1 43A to the SIP PSM 41 need to be known, it means that the SIP PSM 41 must be newly implemented even if the internal format of the psmCallID is partially modified. This applies not only to the SIP PSM 41 but also to the PSM-1 43A, increasing the complexity of each module and its dependence on other protocols.

The call list module 44 has an internal local call processing list for each of the PSMs 43A-43K and the SIP PSM 41 to overcome this point, and in the local call processing list associated with only one call. Only indexes are stored. If the PSM-1 43A has information about the SIP PSM 41, the local call processing module of the PSM-1 43A resolves itself. If the PSM-1 43A transmits a signal to the SIP PSM 41 side, the call converter 42 transforms the signal of the PSM-1 43A into SIP format and the SIP PSM associated with the call list module 44. The index in the local call processing module of 41 is found and passed to the SIP PSM 41. 7 shows the structure of the call list module 44 as described above.

In the above description, the purpose of operating the local call list was to improve independence and reduce complexity in the module through an abstraction of information shared between the PSMs 43A-43K. We explained that the information of the counterpart PSM is obtained from the local call list itself. However, since the information of the counterpart PSM must be abstracted into a data type that can be recognized by the current PSM 43A-43K, the protocol-based terminal of the counterpart PSM is based on the protocol-based protocol of the current PSM 43A-43K in the local call list. Should be abstracted to the terminal. In this description, the terminal thus abstracted is defined as a virtual call intermediary.

7 shows how a virtual call intermediary operates in a local call list. That is, in the PSM as shown in FIG. 7, since the other party of the MG is a virtual MG instead of an actual SIP terminal, the contents of the PSM that are already implemented are used almost without modification, but the message is transmitted to the virtual MG. Ask the call converter to convert to SIP. In this case, since the SIP PSM processes the message of the MG as if it is transmitted from the virtual SIP terminal, the advantages obtained by using the virtual MG in the counterpart PSM are equally obtained. Whether or not the virtual call intermediary is distinguished from the actual terminal using a flag variable that can specify the type of the terminal.

As described in detail above, the present invention newly defines a structure of a SIP media gateway controller, which is a next generation VoIP protocol, for various types of VoIP media gateways based on different protocols, and newly determines a call processing method between the controller and the gateway. Gateway equipment that supports the SIP system and guarantees the stability of hardware system performance by providing various additional service functions supported by SIP call processing server while complementing VoIP gateway, which is a weak point of SIP, a protocol. This has the effect of reducing the burden of new development.

In addition, from the aspect of securing system stability, development power can be focused on SIP MGC, which is relatively easy to implement, stability test, and scalable, and has no influence on the gateway side, thereby reducing overall system development cost. It can be effective.

In addition, there is an effect that the SIP MGC can provide a variety of SIP supplementary services to the end user by manipulating a general MG that does not support SIP as an intermediate agent.

Claims (4)

A SIP PSM operating a SIP protocol; A call converter which performs an intermediate setup and association function of a call between the SIP PSM and the PSM to be described later; A PSM having input / output associated with a related MG and the call converter, and storing basic data necessary for a protocol; A call list module which holds local call list index information, basic identification information, and protocol type information in the PSM and abstracts and transmits information of a counterpart PSM in a desired form through each call converter; An MG status manager which stores the state of the MG and the state of a user terminal connected to the MG and periodically reports the state of the MG so as to manage the state of the MG managed by the SIP MGC; The SIP MGC has a call router for operating a fixed route independently without a proxy server, and the SIP-based SIP MGC is configured to interoperate with various VoIP MGs supporting different protocols. Media gateway controller. The method according to claim 1, wherein the call list module in charge of global call processing in the SIP MGC is separated from a local call list module operated by a corresponding protocol in each protocol stack module to manage calls between SIP terminals and MGs. SPI media gateway controller, comprising a simple information that specifies the index of the internal call and the type of the protocol in the local call list of the protocol stack associated with the particular call. The method of claim 1, wherein the SIP MGC inserts a virtual MG when connecting a call from the SIP terminal to the MG through a call list, which is a call processing list, and inserts a virtual SIP terminal as a call intermediary when connecting a call in the opposite direction. SPI media gateway controller, characterized in that for establishing the association of the call. The SIP media gateway controller of claim 1, wherein the SIP MGC is connected between the SIP MG and a packet network in a SIP-based VoIP system.

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