KR101015538B1 - VoIP Access Gateway and inter-Local Call Processing Method thereof - Google Patents

Abstract

Disclosed are a Voice over Internet Protocol (VoIP) Access Gateway and a call processing method between local subscribers thereof. When the access gateway of the present invention determines that the origination of the local subscriber or the call to the local subscriber is for the call connection between the local subscribers, the destination IP address of the Session Initialization Protocol (SIP) message is substituted for the IP address of the proxy server. By including your own IP address in the SIP message, SIP messages are processed internally without being sent to the proxy server. Through this, the access gateway of the present invention can reduce the traffic load occurring on the proxy server and the IP network, and can handle calls between at least local subscribers even when the proxy server or the IP network fails.

VoIP access gateway, SIP message, proxy server, local call connection

Description

VoIP Access Gateway and inter-Local Call Processing Method

According to the present invention, SIP can perform a call connection between a local terminal without using the proxy server 140 while performing call processing for a Voice over Internet Protocol (VoIP) between a proxy server and an analog terminal. (Session Initialization Protocol) based VoIP Access Gateway and its local subscribers call processing method.

It is the Voice over Internet Protocol (VoIP), that is, the Internet telephone, which uses a conventional packet network for data communication for voice calls. VoIP telephone systems use the Internet to enable many services not available in traditional telephone networks.

1 is a block diagram of a conventional SIP-based VoIP system.

Referring to FIG. 1, general analog phones 111 and 113 access an access gateway 120 to access an IP network 130 that provides VoIP service. The access gateway 120 generates a SIP (Session Initial Protocol) based signaling message for call processing and transmits the signaling message to the counterpart through the proxy server 140.

The access gateway 120 may provide a call service using SIP to various types of local terminals including the analog phones 111 and 113 in cooperation with the proxy server 140.

Proxy server 140 is a server for the call connection request of the client to route the SIP message to the other party. The proxy server 140 determines a routing destination of the SIP message in conjunction with a location server or a DNS server.

In the conventional SIP-based access gateway 120 as shown in FIG. 1, the proxy server 140 is essential for delivering SIP messages related to call processing.

For example, even when the first subscriber 111 connected to the access gateway 120 connects the phone to the second subscriber 113, the access gateway 120 connects the IP network 130 and the proxy server 140. Send and receive SIP messages. Therefore, according to FIG. 1, the access gateway 120 sends the message "INVITE 1002@192.168.1.48" to the proxy server 140 and receives the message "INVITE 1002@192.168.1.153" from the proxy server 140. Will be.

Therefore, if a failure occurs in the proxy server 140 or the IP network 130, all call processing related SIP messages cannot be delivered. Therefore, analog call subscribers 111 and 113 connected to the access gateway 120 as well as normal call processing are not provided. Even the handling of local calls, which are telephone connections between them, becomes impossible.

An object of the present invention can perform a call connection between a local terminal while not performing a call processing for a Voice over Internet Protocol (VoIP) between a proxy server and an analog terminal without using the proxy server 140. The present invention provides a SIP (Session Initialization Protocol) based VoIP Access Gateway and its local subscriber call processing method.

In order to achieve the above object, according to the present invention, a call processing method between local subscribers of a VoIP access gateway connected to a proxy server through an IP network and providing VoIP service to a plurality of local subscribers is provided. 1, receiving an incoming call number for outgoing call connection from the subscriber; If the called phone number is a phone number of a second subscriber who is one of the local subscribers, the destination IP address of the SIP message sent to the second subscriber side for the outgoing call of the first subscriber is set as its IP address. Performing a SIP-based outgoing procedure for an outgoing call of the first subscriber; Receive an INVITE message of SIP from the IP network according to the originating process, and if the outgoing telephone number included in the INVITE message is the first subscriber telephone number, send it to the first subscriber side for the incoming call of the second subscriber. Setting a destination IP address of the SIP message as its IP address and performing a SIP-based incoming call process for the second subscriber; And opening a voice call path through the calling process and the incoming call process.

According to an embodiment of the present disclosure, the opening of the voice call path may be provided by internally switching the voice signals of the first and second subscribers without generating the RTP packets.

According to another embodiment of the present invention, a VoIP access gateway that is connected to an external proxy server through an IP network and has a subscriber interface unit for connecting to a plurality of local subscribers, provides a VoIP service to the local subscriber. It includes a call processing unit for connecting the call by generating and sending and receiving a SIP message for the origination or reception of the local subscriber to the local subscriber and the proxy server.

If the call processor determines that the origination of the local subscriber or the call to the local subscriber is for a call connection between the local subscribers, the call processing unit replaces the destination IP address of the SIP message with its own IP address instead of the IP address of the proxy server. By including, the SIP message may be processed internally without being transmitted to the proxy server.

According to an embodiment, the subscriber interface unit may include a switching module for switching voice signals between local subscribers when a call is connected between local subscribers by an operation of the call processor.

The VoIP access gateway according to the present invention processes SIP messages for call processing between local subscribers without passing them to the proxy server, thereby reducing the traffic load on the proxy server and the IP network, and reducing the traffic load on the proxy server or IP network. Even if a failure occurs, at least the call between local subscribers can be handled.

The call processing method of the VoIP access gateway of the present invention changes and processes only the destination address (proxy server address) according to the local call and the general call, and thus does not change the internal processing flow or SIP message processing procedure of the conventional VoIP access gateway. It can be easily applied to conventional equipment.

Through this, in an environment without a proxy server, a standalone SIP communication network that performs call processing between local subscribers may be configured.

BEST MODE FOR CARRYING OUT THE INVENTION The present invention will be described in more detail with reference to the drawings.

2 is a block diagram schematically illustrating a VoIP system including a VoIP gateway according to the present invention.

Referring to FIG. 2, the VoIP system 200 of the present invention includes a SIP Proxy Server 140 connected through a VoIP Access Gateway 210 and an Internet Protocol (IP) network 130. do. Hereinafter, the 'VoIP Access Gateway' is simply referred to as 'AGW (Access Gateway)'.

An analog telephone is connected to the AGW 100. 2 illustrates an example in which a first subscriber 111 and a second subscriber 113 are connected, and will be described with reference to the first subscriber 111 and the second subscriber 113 described below.

The proxy server 140 controls the AGW 210 to perform packet routing for SIP signaling messages for call connection with other nodes.

The AGW 210 is connected to the IP network 130 through a Real-Time Transport Protocol (RTP) channel for voice data and a Session Initiation Protocol (SIP) channel for signaling, to transmit and receive voice data packets, and to transmit and receive a voice data packet. 140) send and receive signaling messages.

The AGW 210 includes a subscriber interface 211, a call processor 213, a digital signal processor 215, an Ethernet connection 217, and an operation information manager 219. And provide a VoIP service to the second subscriber 113.

Additionally, in the case of a call between a local subscriber (hereinafter referred to as an 'local call') (such as a call connection between the first subscriber 111 and the second subscriber 113), the AGW 100 ) May process the call directly without going through the proxy server 140.

The subscriber interface unit 211 provides a physical interface for connecting an analog phone, detects a hook on / off, a hook flash, and the like of the analog phone and provides it to the call processor 213.

The subscriber interface unit 211 may include a plurality of subscriber cards, each of the subscriber cards may include a plurality of subscriber ports (line), and the subscriber interface unit 211 has a hook off or the like of any subscriber card. Physical location information on which port is provided is provided to the call processor 213 together.

The subscriber interface 211 provides an incoming call transmitted from the call processor 213 to the first subscriber 111 or the second subscriber 113.

The call processing unit 213 receives an event such as HOOK ON / OFF, HOOK FLASH from the subscriber interface unit 211, allocates system resources for the call, and generates a SIP control message for signaling for generating a proxy server. It is responsible for call processing such as generating a SIP session by transmitting and receiving with the counterpart subscriber side through the 140.

In the incoming process, when the call processing unit 213 receives an INVITE message, which is a SIP signaling message, from the Ethernet access unit 217, the call processing unit 213 inquires the subscriber information corresponding to the incoming phone number included in the message to the operation information management unit 219. Check the subscriber's physical location. The call processing unit 213 requests the digital signal processing unit 213 to generate an incoming signal corresponding to the incoming call, and causes the digital signal processing unit 213 to send a ringing signal to the called subscriber through the subscriber interface unit 211. Provide it.

In the case of an origination, the call processor 213 determines which subscriber is the originating event based on the physical location information of the line provided by the subscriber interface 211.

The call processing unit 213 determines whether the originating or incoming call requested for the call connection is a local call connection, and processes the call differently from a general call in the case of the local call connection. This will be described again below.

The digital signal processing unit 215 converts the voice data received from the subscriber interface unit 211 into an IP packet and provides it to the Ethernet connection unit 217 according to the opening of the voice call, or from the IP packet received from the Ethernet connection unit 217. The voice data is decoded and provided to the first subscriber 111 or the second subscriber 113 through the subscriber interface unit 211 to enable the voice call.

In the case of outgoing call, the digital signal processor 215 detects DTMF using a digital signal processor (DSP) and provides the call to the call processor 213. A call tone and call tone for call processing is requested by the call processor 213. It generates a ring tone and ringing signal (ring) such as a ringing tone and provides it to the subscriber.

The Ethernet connection unit 217 connects the AGW 210 to the proxy server 140 via the IP network 130 as a connection interface to the IP network 130.

The operation information management unit 219 stores and manages various system information, parameters, registered subscriber information, and the like in a storage medium (not shown). The system information includes the IP address and port number of the AGW 100, the IP address and port number of the proxy server 140, and the subscriber information includes the subscriber's unique number assigned to each line and the physical location of the line. Information (card number, port number, etc.) and the like.

The configuration of the AGW 210 of the present invention has been described above.

Here, the normal outgoing call processing of the AGW 100 will be briefly described. When the first subscriber 111 is hooked off for transmission, the subscriber interface 211 generates a hook off signal and provides the hook off signal to the call processor 213. The call processor 213 receiving the hook off signal requests the digital signal processor 215 to provide a dial tone to the first subscriber 111. After the dial tone is provided, when the first subscriber 111 inputs the other party's telephone number, the corresponding DTMF signal is converted into numeric information by the digital signal processor 213 and transmitted to the call processor 213. The call processor 213 generates an INVITE message including the other party's telephone number and transmits the INVITE message to the proxy server 140 through the Ethernet access unit 217 to request a call connection. In other words, the destination address of the INVITE message becomes the address of the proxy server 140 (in FIG. 2, 192.168.1.48).

The proxy server 140 checks the other party's IP address using the location server and the DNS server, and routes the INVITE message to the other party.

Thereafter, when the call processor 213 receives the OK (INVITE) message (status code 200) provided by the counterpart according to the SIP protocol, the call processor 213 sends an ACK message to open the RTP voice call path.

As such, the general VoIP service provided by the AGW 100 to the first subscriber or the second subscriber 113 is performed through the proxy server 140. However, the connection between the local calls where the first subscriber 111 calls the second subscriber 113 does not go through the proxy server 140.

Hereinafter, a method of handling a call between local subscribers will be described in more detail with reference to FIG. 3. For convenience of explanation, a situation in which the first subscriber 111 makes a call to the second subscriber 113 will be described as an example. Assume that the phone number of the first subscriber 111 is 1001 and the phone number of the second subscriber 113 is 1002. Therefore, the outgoing phone number is 1001 and the incoming phone number is 1002. In addition, it is assumed that the IP address of the AGW 100 is 192.168.1.153 and the IP address of the proxy server 140 is 192.168.1.48.

In addition, since the call processing method of the present invention described below follows a normal call processing process according to the normal VoIP, a part of the following description that is not specifically mentioned about call generation, connection, and sending and receiving of a message is conventional. Protocols and algorithms related to SIP signaling and RTP data transmission and reception regarding VoIP are used as they are.

FIG. 3 is a flowchart provided to explain a method for handling a call between local subscribers of a VoIP gateway according to the present invention. In FIG. 3, a SIP message sent by a called party to a calling party is represented by a double solid line, and is sent by the calling party. Distinguish from.

The call processor 213 receives the Hook Off signal of the first subscriber 111 from the subscriber interface 211 and requests the digital signal processor 215 to provide the dial tone to the first subscriber 111. Accordingly, after the dial tone is provided, when the first subscriber 111 inputs the incoming telephone number (ie, the second subscriber telephone number 1002), the digital signal processor 213 recognizes the incoming telephone number and the call processor 213. Provided to (S301).

When the call processing unit 213 receives the incoming telephone number from the digital signal processing unit 213, the call processing unit 213 queries the operation information management unit 219 whether there is a local subscriber having the same telephone number as the incoming telephone number, and whether the incoming telephone number is the local subscriber. Determine (S303).

If the called telephone number is determined to be the telephone number of the local subscriber, the call processor 213 sets a 'local mode' for the corresponding outgoing call. The 'local mode' setting is to set the address information (IP address, Port number, etc.) of the proxy server 140 related to the outgoing call processing to the address information (IP address, Port number, etc.) of the AGW 100. . Therefore, for the processing of the outgoing call, the destination address of the SIP message sent by the call processing unit 213 is used instead of the IP address of the proxy server 140, the IP address (192.168.1.153) of the AGW (100) (S305).

The call processing unit 213 transmits the INVITE message of the SIP that sets the destination IP address as its IP address to the outgoing call connection of the first subscriber 111 to the Ethernet access unit 217. 2 and 3, the destination IP address of the SIP message is set to 192.168.1.153, which is different from the destination address of the SIP message transmitted by the access gateway 120 in the prior art. The message will not be transmitted to the IP network 130 from the Ethernet connection 217 will be received and processed by itself (S307).

When the call processing unit 213 receives the INVITE message returned to itself, the call processing unit 213 inquires the local subscriber having the same telephone number as the calling telephone number to the operation information management unit 219 to determine whether the calling telephone number is the local subscriber ( S309).

If the calling telephone number is determined to be the telephone number of the local subscriber, the call processor 213 sets the 'local mode' for the incoming call. The 'local mode' setting is to set the address information (IP address, Port number, etc.) of the proxy server 140 related to the incoming call processing to the address information (IP address, Port number, etc.) of the AGW 100. Therefore, for the processing of the incoming call, the destination address of the SIP message sent by the call processing unit 213 is used instead of the IP address of the proxy server 140, the IP address (192.168.1.153) of the AGW 100 is used (S311).

The call processing unit 213 inquires the subscriber information about the called phone number included in the corresponding INVITE message to the operation information management unit 219, and sends the signal to the physical location of the called phone number. ). Accordingly, the digital signal processing unit 213 provides the slave signal to the second subscriber 113 (S313).

The call processing unit 213 provides a ringing message (status number 180) of SIP to the calling party after providing the name signal. The destination IP address of the SIP message in this case is also set to the IP address 192.168.1.153 of the AGW 100 itself (S315).

When the call processing unit 213 receives a ringing message sent by the called party, the call processing unit 213 requests the digital signal processing unit 213 to provide a ringback tone to the first subscriber 111, and the digital signal processing unit 213 The ringback tone is provided to the first subscriber (S317).

When the second subscriber 113 is hooked off according to the slave signal (S319), the subscriber interface 211 provides the hook off signal to the call processor 213, and the call processor 213 generates an OK message of SIP. To the calling party. Here again, the destination IP address of the SIP message is set to the IP address 192.168.1.153 of the AGW 100 itself (S321).

When the call processor 213 receives the OK message provided by the called party, the call processor 213 requests the digital signal processor 213 to stop the ringback tone, and provides the called party with an ACK message in response to the OK message. Here again, the destination IP address of the SIP message is set to the IP address 192.168.1.153 of the AGW 100 itself (S323 and S325).

If it is determined in step S303 or step S309 that the incoming or outgoing telephone number is not a local subscriber and is not determined to be a local call, the destination IP address of the SIP message sent by the call processor 213 is the proxy server 140. ) Is used as it is, and proceeds in the same manner as a general SIP call processing procedure (S327, S329).

In response to the sending of the ACK message by the calling party, the voice call path between the calling party and the called party is opened, so that local call processing of the AGW 100 is performed without passing through the proxy server 140.

The voice message is transmitted to the first subscriber (111) ↔ subscriber interface (211) ↔ digital signal processor (213) ↔ Ethernet connection (217) ↔ digital signal processor (213) ↔ subscriber interface (211) ↔ second subscriber (113) It will be sent and received via the path of.

However, according to an exemplary embodiment, the digital signal processing unit 215 has a switching function to switch the RTP packet without providing the Ethernet connection unit 217, or the subscriber interface unit 211 includes a switching module to provide the first subscriber ( The voice signal between 111 and the second subscriber 113 may be switched.

According to an embodiment, the operation information manager 219 may map and manage a destination address (IP address of a proxy server in normal mode) of a SIP message to each local subscriber information. In this case, in step S305 or S311, the call processing unit 213 may perform the local mode setting by changing the SIP destination address mapped to the corresponding subscriber information to the IP address of the AGW 100 itself.

Although the above has been illustrated and described with respect to preferred embodiments of the present invention, the present invention is not limited to the above-described specific embodiments, it is usually in the technical field to which the invention belongs without departing from the spirit of the invention claimed in the claims. Various modifications can be made by those skilled in the art, and these modifications should not be individually understood from the technical spirit or the prospect of the present invention.

1 is a block diagram of a conventional SIP-based VoIP system,

2 is a configuration diagram briefly showing a VoIP system including a VoIP gateway of the present invention; and

3 is a flowchart provided to explain a method for call processing between local subscribers of the VoIP gateway of the present invention.

Claims (4)

In the method of call processing between local subscribers of VoIP access gateway connected to proxy server through IP network and providing VoIP service to a plurality of local subscribers, Receiving an incoming call number for outgoing call connection from a first subscriber, one of the local subscribers; If the called phone number is a phone number of a second subscriber which is one of the local subscribers, the IP address of the VoIP access gateway owns a destination IP address of a SIP message sent to the second subscriber side for an outgoing call of the first subscriber. Setting to an address and performing a SIP-based outgoing procedure for an outgoing call of the first subscriber; Receive an INVITE message of SIP from the IP network according to the originating process, and if the outgoing telephone number included in the INVITE message is the first subscriber telephone number, send it to the first subscriber side for incoming call of the second subscriber. Setting a destination IP address of a SIP message as an IP address of the VoIP access gateway and performing a SIP-based incoming call process for incoming call of the second subscriber; And And a method of opening a voice call path through the originating process and the receiving process. The method of claim 1, The opening of the voice call path, The method of processing a call between local subscribers of a VoIP access gateway, wherein the voice signals of the first subscriber and the second subscriber are internally switched without being generated as an RTP packet. A VoIP access gateway connected to an external proxy server through an IP network and having a subscriber interface unit for connecting to a plurality of local subscribers to provide VoIP services to the local subscribers. A call processing unit which connects a call by generating a SIP message for origination or reception of the local subscriber and sending and receiving with the proxy server, If the call processor determines that the origination of the local subscriber or the call to the local subscriber is for a call connection between the local subscribers, the VoIP access gateway replaces the destination IP address of the SIP message with the IP address of the proxy server. By including its IP address, the SIP access gateway to be processed internally without being sent to the proxy server. The method of claim 3, The subscriber interface unit includes a switching module for switching voice signals between the local subscribers when a call is connected between local subscribers by an operation of the call processor.

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