KR100727069B1 - Method for connecting with other network in wireless packet switching network system based on SIP and the system thereof - Google Patents

Abstract

The present invention relates to a method and system for interworking with other networks in a SIP-based wireless packet switching network system. According to an embodiment of the present invention, a method for interworking in a wireless packet switching network system coupled to a STUN server via a NAT, the method comprising: receiving a call request message from an originating terminal; (b) recognizing that the called terminal is located in another network by using information of the called terminal corresponding to the call request message stored in the database server; Driving a STUN client to obtain information about a public IP and a port to be allocated in a NAT from the STUN server; Changing the private IP and port of Layer 7 in the call request message to be the same as the public IP and port; And another network interworking method in a SIP-based wireless packet switching network system comprising the step of establishing a call with the destination terminal using the changed call request message is provided. Therefore, there is an effect that can provide a method and system that can smoothly provide SIP-based services in a network structure using NAT.

STUN, VoIP, SIP, public network, NAT, other network

Description

Interworking method and system in SIP-based wireless packet switching network system {Method for connecting with other network in wireless packet switching network system based on SIP and the system}

1 is a diagram illustrating a SIP-based service flow according to the prior art.

2 is a diagram illustrating a SIP signaling flow using NAT according to the prior art.

Figure 3 is a diagram showing the configuration of a schematic system for performing the STUN method according to the prior art.

4 is a diagram schematically showing an entire system for interworking with other networks in a SIP-based wireless packet switched network system according to an embodiment of the present invention.

5 is a schematic diagram showing an entire system of a network for performing packet-based wireless communication according to an embodiment of the present invention.

6 to 9 is a functional block diagram showing the configuration according to each function of the SPS, SDS, SRS and SGS of the PS network according to an embodiment of the present invention.

10 is a diagram illustrating a process of setting up a call with a destination terminal located in another network in a wireless packet switching network system according to an exemplary embodiment of the present invention.

<Description of the symbols for the main parts of the drawings>

9, 330, 420: NAT

10: WCDMA Network

30: CDMA network

100: packet switching network

110: SIP proxy server (SPS: SIP Proxy Server)

130: database server (SDS: SIP DB Server)

150: relay server (SRS: SIP Relay Server)

170: gateway server (SGS: SIP Gateway server)

400: magnetic net

410: base station

430: STUN server

440: perception

The present invention relates to other network interworking in a wireless packet switching network system, and more particularly, to a method and system for interworking with other networks in a SIP-based wireless packet switching network system.

In recent years, a voice or video call service can be provided using a wireless packet switching network such as a packet network for data communication as well as a wireless communication service using a circuit switched network. Voice / video communication technology using a wireless packet-switched network called Voice over Internet Protocol (VoIP) is currently being developed and researched due to the advantages that it can be used simultaneously by multiple users through a cable, it is scalable, and the cost is much lower than that of a conventional telephone. It is becoming. Commonly used protocols in VoIP include H.323, SIP, MGCP and GACO.

The packet-switched network performs communication using the IP address of the terminal. Here, many service providers use private IP through NAT (Network Address Translation) to secure their own network and save insufficient public IP address to protect their private network connected to the public network from the invaders. . In the wireless mobile communication, the private IP is allocated to the terminal for the same reason as described above, and when the terminal uses another network (for example, a public network), the public IP is allocated to the private IP initially allocated by NAT. The service is provided by allocation. However, Session Initiation Protocol (SIP) -based services use private IPs by using NAT according to separation of signaling traffic for establishing a session and data traffic, which is data transmission and reception according to actual calls. There is a problem.

1 is a diagram illustrating a SIP-based service flow according to the prior art, and FIG. 2 is a diagram illustrating a SIP signaling flow using a NAT according to the prior art.

Referring to FIG. 1, a SIP setup for providing a session for exchanging a service between a calling terminal 1 and a called terminal 2 as shown by a dotted arrow starting from the calling terminal 1 requesting communication. The process is carried out. The originating terminal 1 uses the proxy server 5 and the DNS (Domain Name Service) server 8 to access the IP of the other terminal (the receiving terminal) 2 via the other proxy server 2 (6). It finds and routes the SIP message, and sets the IP and port for exchanging actual media traffic with the destination terminal 2. Afterwards, each terminal 10 or 11 transmits and receives actual media traffic (for example, by using a Real-Time Transport Protocol (RTP)) by using an IP and a port.

With reference to Figure 2 will be described the problem when using the NAT (9) in the SIP-based service. First, there is a problem in the process of routing the SIP messages 60, 61, 62, 63 and 64 by the terminal 1 using the proxy server 1 (5) and the DNS server (8) to find the IP of the other terminal (2). Occurs. As shown in the figure, the SIP messages 60, 61, 62, 63 and 64 pass through the proxy servers 5 and 6 in the process of visiting the opposite terminal 2 and respond to the SIP request. Whenever you go through a proxy server to get the same path as the request, the address of the proxy server is recorded in a specific field (for example, Via field). When passing through NAT (9), due to the basic nature of NAT (9), private IPs and ports in Layers 3 and 4 are replaced with public IPs and ports assigned by NAT (9). 3.4 80: 10.20.30.40 81) You cannot change the IP recorded in Layer 7. Therefore, the message sent to the proxy server 2 (6) is recorded in the private IP of the proxy server 1 (5), which is sent to the private IP of the proxy server 1 (5) that does not exist in the public network when sending the response Proxy server 1 (5) cannot receive a response and SIP setup cannot be successful.

Second, a problem occurs in the process of setting an IP and a port for exchanging actual media traffic between the terminals 10 and 11. As shown in the figure, the SIP message 60, 61, 62, 63, 64 is negotiated by enclosing IP and port information for media transmission and reception with the called terminal (c = IN IP4 1.2.3.4, m = audio 49170). RTP / AVP 0). Even in this case, since only the IPs and ports of the layers 3 and 4 are replaced with the public IPs and ports allocated by the NAT 9 in the NAT 9, the IPs and ports for transmitting media traffic cannot be changed. Accordingly, even if the other 6789 terminal 2 is set up with the SIP, the other terminal 2 transmits the media traffic to the private IP of the 1234 terminal 1, so the 1234 terminal 1 cannot receive it.

In order to compensate for this problem, a STUN (Simple Traversal of UDP through NATs) scheme is generally used to solve the above-mentioned problems by raising a STUN client on a terminal and installing a STUN server behind a NAT 9. Hereinafter, the STUN method will be described with reference to the drawings.

3 is a diagram showing the configuration of a schematic system for performing the STUN method according to the prior art.

Referring to FIG. 3, the system for performing the STUN method includes a terminal 300 equipped with STUN, a NAT 330, and a STUN server 350. The STUN method will be described with reference to the drawings. When the STUN client mounted on the terminal 300 transmits a specific message to the private network by driving the STUN client before transmitting the SIP message to an external communication network, the STUN server 350 outside the NAT 330 may perform the NAT 330. The NAT 330 finds information on the IP and port to be allocated to the terminal 300 through the specific message passed through. Thereafter, the STUN server 350 transmits information on the found IP and port to the terminal 300, and the terminal 300 generates a SIP message using the corresponding IP and port and transmits it to an external communication network. Therefore, it is possible to solve the problem of using the above-described NAT.

However, this requires a change of the terminal and makes the terminal heavy according to a separate device and / or program (STUN client) mounted on the terminal, and has a problem in that a separate STUN server must be provided. In addition, the terminal may not know whether the destination terminal for performing communication is located in the other network, there is also a problem that must always run the STUN client.

Accordingly, the present invention has been made to solve the above problems, and provides a method and a SIP-based wireless packet network system using the method that can solve the problem of using NAT in providing a SIP-based service. There is a purpose.

Another object of the present invention is to provide a method and system for interworking in a SIP-based wireless packet switching network system that does not add a separate device and / or program to a terminal according to a STUN client in using the STUN method. have.

It is still another object of the present invention to provide a method and system for interworking in a SIP-based wireless packet switching network system capable of simplifying a procedure according to SIP setup by running a STUN client only when a called terminal exists in another network. There is this.

Other objects of the present invention will become more apparent through the preferred embodiments described below.

According to an aspect of the present invention for achieving the above object, the other network interworking method in a wireless packet switching network system coupled to the STUN server via NAT, the method comprising the steps of: (a) receiving a call request message from the originating terminal; (b) recognizing that the called terminal is located in the other network by using information of the called terminal corresponding to the call request message stored in a database server; (c) driving a STUN client to obtain information about a public IP and a port to be allocated in the NAT from the STUN server; (d) changing the private IP and port of Layer 7 in the call request message to be the same as the public IP and port; And (e) establishing a call with the called terminal using the changed call request message. The other network interworking method in the SIP-based wireless packet switching network system is provided.

The step (b) may include: requesting, by the SIP proxy server receiving the call request message from the terminal, information of the called terminal to the database server; And transmitting information of the called terminal including the IP information of the called terminal from the database server to the SIP proxy server, wherein the SIP proxy server is located in the other network through the IP information. Can be recognized. In addition, the STUN client may be driven by the SIP proxy server.

The method may further include generating, by the relay server, a session for transmitting media traffic between the calling terminal and the called terminal according to a request of the SIP proxy server.

According to another aspect of the present invention, in a wireless packet-switching network system, a call is established (signaled) with at least one called terminal in response to a call request message transmitted from an originating terminal, and a session between the originating terminal and the called terminal. a SIP proxy server controlling a session; A database server that manages information of a subscribed terminal and state information of the subscribed terminal, and provides destination terminal information requested by the SIP proxy server for signaling to the SIP proxy server; And a relay server for allocating at least one channel so that the session is generated in response to a channel request from a SIP proxy server that sets up the call, wherein the SIP proxy server includes a STUN client when the called terminal is located in another network. client) to obtain information about the public IP and the port to be allocated in the NAT from the STUN server coupled via NAT, and the private IP and the port of the layer 7 in the call request message to the public IP and the port. A SIP-based wireless packet switching network system is provided which interoperates with another network, wherein a call is established with the called terminal using the changed call request message.

Here, the SIP proxy server may recognize that the called terminal is located in the other network by using the called terminal information.

The SIP proxy server may further include: a call processing unit configured to set up and release the call, drive the STUN client, and perform the channel request to the relay server; And a session controller managing session information of the calling terminal and the called terminal participating in the session.

The database server may include a subscriber information manager configured to manage information of a subscribed terminal; And a subscriber state management unit that manages state information of the subscribed terminal as to whether the subscribed terminal is logged in or logged out, wherein the subscriber information management unit requests the incoming call requested by the SIP proxy server when establishing a call. Terminal information may be transmitted to the SIP proxy server.

The relay server may further include: a media session controller configured to allocate the channel to the SIP proxy server; A media controller configured to analyze, copy, and distribute media according to communication between the calling terminal and the called terminal when there are a plurality of called terminals; And a charging information generator for generating a charging CDR according to communication between the calling terminal and the called terminal.

If the called terminal exists in the circuit switched network, the gateway server may further include a gateway server that relays communication between the calling terminal and the called terminal in association with the circuit switched network. The gateway server may include: a signaling / bearer converter configured to convert signaling and bearers between the calling terminal and the called terminal existing in the circuit switched network; A media converter configured to perform real-time conversion of a media format when communicating between the calling terminal and a called terminal existing in the circuit switched network; And an ISUP function unit that performs a routing and call distribution function according to the call setup.

Hereinafter, another network interworking method and a system thereof in a SIP-based wireless packet switched network system according to an embodiment of the present invention will be described in detail with reference to the accompanying drawings.

4 is a diagram schematically showing an entire system for interworking with other networks in a SIP-based wireless packet switched network system according to an embodiment of the present invention.

Referring to FIG. 4, the entire system communicates with a terminal 1, a base station 410, a network 400 including a packet switched network 100, a NAT 420, a STUN server 430, and a public IP. Public network or self network 400 to perform and other network 440, such as other private network at a location via the public network.

The terminal 1 is not only a mobile communication terminal including a mobile phone, a PCS, a PDA, a smart phone, and the like, but also a wireless LAN (LAN) terminal, a Bluetooth terminal, a portable Internet terminal, and a PMP (Portable Multimedia). Internet-enabled terminals such as a player) are included. Hereinafter, a mobile communication terminal such as a mobile phone will be described by way of example, but it will be apparent through the following description that all the terminals capable of the above-described Internet can be equally applied.

The magnetic network 400 refers to a communication network operated by a specific company or an individual and using a private IP and a port. Accordingly, the terminal 1 may use a private IP and a port through its own network 400 and receive a shared IP and a port from the NAT 420 through a packet-switched network 100 through a public network (for example, the Internet network) or Other networks 440, such as other private networks via a public network, can be used. The network 400 may include a circuit switched network such as a WCDMA network, a CDMA network, as well as a packet switched network. For example, the magnetic network 400 may include a circuit switched network such as a WCDMA network, a CDMA network, or the like operated by a mobile communication company such as KTF, SKT, or LGT.

The base station 410 performs wireless communication with the terminal 1 and communication with a specific device (for example, the SPS 110 (described later in detail) as shown in the drawing) in the self network 400. By performing the function of relaying the communication between the self network and the terminal, the base station may be obvious to those skilled in the art, so a detailed description thereof will be omitted. However, although the base station and the SPS are shown to be directly connected, it is obvious that a separate device (for example, a switch, a PDSN, etc.) may be coupled between them.

The NAT 420 provides a public IP to a corresponding terminal 1 located in the magnetic network 400 to provide a service to a private IP and a port used in the magnetic network 400 to another network using a public IP and a port such as an Internet network. And assigning ports.

The packet switched network 100 is a SIP-based communication network system that provides Voice over Internet Protocol (VoIP), and includes a SIP proxy server (SPS) 110, a database server (SDS: SIP DB server) 130, A relay server (SRS: SIP Relay Server) 150 and a gateway server (SGS: SIP Gateway server) 170 are included. The packet switching network according to the present invention performs a SIP setup according to the communication request of the terminal 1, and generates a session for transmitting and receiving media traffic according to communication between the terminals. do.

The SIP proxy server 110 of the packet switched network 100 performs the functions of a STUN client function and an application layer gateway (ALG). Here, the ALG is an apparatus that improves the problem of NAT that fails to change the IP and port of Layer 7 of the SIP message. That is, when receiving the SIP message for call setup from the terminal 1, the STUN client is driven to obtain information on the public IP and port to be allocated in the NAT from the STUN server 430. In addition, the IP and the port of the layer 7 of the SIP message are changed to be the same as the public IP and the port obtained from the STUN server 430. Therefore, according to the present invention, compared to the conventional STUN method, it is not necessary to mount a STUN client for each terminal 1 using the SIP proxy server 110 of the packet-switching network 100 that sets up a call and controls a session. It is possible to obtain the effect that the change of and need not be equipped with an additional device (or program) in the terminal. In addition, since the terminal 1 does not know whether the called terminal is located in the other network 440, according to the prior art, the terminal should always drive the STUN client every time a call is requested. Since the SIP proxy server 110 can determine whether or not it is located at 440, it is not necessary to run the STUN client every time the SIP setup is performed, thereby reducing the procedure for call setup.

Hereinafter, the magnetic network 400 and the packet-switched network 100 of the magnetic network 400 will be described in detail.

5 is a block diagram schematically illustrating an entire system of a network for performing packet-based wireless communication according to an embodiment of the present invention.

Referring to FIG. 5, a WCDMA network 10 and a CDMA network 30 (hereinafter, collectively referred to as a 'CS network') and a packet switching (PS) network (CS), which are circuit switching (CS) networks, Three communication networks 100 (hereinafter referred to as 'PS networks') are shown as self networks. The PS network 100 is a SIP-based wireless packet network. The PS network 100 according to the present invention is a SIP proxy server 110, a database server 130, a relay server 150, and a gateway server 170 as described above. ).

The SIP Proxy Server (SPS) 110 (hereinafter referred to as 'SPS') of the PS network 100 is a standard defined in SIP (Session Initiation Protocol: IETF RFC3261) and is a call used in 3GPP or 3GPP2. Control Protocol) It is a server in charge of proxy function and performs call processing function (set, change, release) of call terminal with call terminal. In addition, the SPS 110 controls a session according to data communication after a call is established between the calling terminal and the called terminal. In addition, the SPS 110 registers the subscriber with the SDS 130 in association with the SDS 130 and performs a function of authenticating the subscriber.

In particular, when the SPS 110 establishes a call with the destination terminal existing in the other network 440, since the SIP message passes through the NAT 420, the SPS 110 and the STUN server 430 located across the NAT 420; By interworking, the public IP and port information of the calling terminal to be allocated through the operation of the STUN client are acquired to change the private IP and the port of the calling terminal in the SIP message (particularly, the IP and the port of the layer 7). Accordingly, the PS network 100 according to the present invention has the problem of using the NAT 420 as described above (the recipient terminal is unable to change the IP and port of the SIP message layer 7 to the call setup to the calling terminal). Failing to send the response message and the media traffic).

In addition, the SPS 110 may be coupled to a short message service center (SMSC) (not shown) that provides a short message service to a subscribed terminal and system. The SPS 110 transmits a short message through the SMSC to enable the reception when the called terminal is in a logout state (a state in which communication is not possible through the PS network 100 such as a state in which a predetermined communication program is not executed in advance). Can be.

Database server (SDS) 130 (hereinafter referred to as 'SDS') is a server that performs the function of managing the subscriber's information, the subscriber's authentication, subscriber status (log-in status or logout ( It manages the log-out status), provides additional services, collects billing CDRs (Call Detail Record), and transfers billing CDR files. In particular, the SDS 130 is a location register (HLR or VLR, below) of the WCDMA network 10 or the CDMA network 30 when the called terminal does not exist in the PS network 100 according to the call establishment request of the calling terminal. For convenience of explanation, the unit is integrated into the HLR, and the location register of the WCDMA network 10 is referred to as a W-HLR, and the location register of the CDMA network 30 is referred to as a C-HLR. 10) or whether it exists in the CDMA network 30. In addition, it interoperates with the Number Portability System (NPS) to determine whether the called terminal is a third party subscriber. Here, the order in which the SDS 130 requests the called terminal information to the CS network is set in the order of W-HLR, C-HLR, and if the called terminal information does not exist in each CS network, it is preferable to request the NPS. . Further, more preferably, the SDS 130 determines only whether the called terminal exists in the WCDMA network 10 by interworking with the W-HLR only, and if the called terminal does not exist in the WCDMA network 10, its CDMA network ( 30) Alternatively, it may be determined that it exists in a communication network of another company, and it may be determined whether the called terminal is its own subscriber or a third party subscriber in association with the NPS. If the called terminal is located in the CDMA network 30, call setting may be performed again only through the CDMA network 30 without using the PS network 100. That is, the voice call is set up through the CDMA network 30 so that only the voice call is performed.

Therefore, the SDS 130 is the SPS 110 to establish a call between the calling terminal and the called terminal information about the called terminal present in the CS network (including other companies' networks) as well as the PS network 100 (currently belonging to) Network information). Here, the SDS 130 may be combined with an IP Location Server (IPLS) (not shown), which is a system that manages the location and status of the subscriber from the CDMA data network viewpoint (or 1xEV-DO system).

The relay server (SRS) 150 (hereinafter referred to as 'SRS') controls a media session according to data communication between terminals, and analyzes and copies media during a 1: n call such as a video conference call. , Distribution, etc. In addition, the SRS 150 is a video conferencing call service, video conferencing call-based technology, video conferencing call media control, intelligent network prepaid billing processing, billing information transmission, voice control function when performing Push-To-Talk (PTT) function, Application functions such as announcement and ring-back-tone functions may be performed. Specific data related to application functions is stored in the SDS. Here, the Packet Data Serving Node (PDSN) (not shown) that provides a Point-to-Point Protocol (PPP) interface with the terminal may be coupled between the SPS 110 and the SRS 150. In addition, a differentiated services code point (DSCP) (not shown), which is a service logic that manages processing of data services among intelligent network services, may be combined.

The gateway server (SGS) 170 (hereinafter, referred to as 'SGS') provides an interworking function with a CS network such as a CDMA network and a WCDMA network, and includes a PS terminal (PS network available terminal) and a CS terminal ( It performs the function of signaling (interworking between SIP of PS network and H.324M of CS network) between CS networks. That is, the SGS 170 is combined with a second generation exchange (2G-MSC) 31 of the CDMA network 30 and a third generation exchange (3G-MSC) 13 of the WCDMA network 10, so that the PS terminal and the CS terminal are combined. Relay communication between Here, the STPs 13 and 33 of the CDMA network 30 and the WCDMA network 10 are devices that provide a backbone function for No. 7 signaling. In addition, the SGS 170 may perform functions such as bearer (communication network or transmission method connecting a gateway and a terminal), media conversion, DTMF digit collection / processing, and billing information transmission. Accordingly, the PS network 100 according to the present invention may perform communication between the PS terminal and the CS terminal by interworking with the WCDMA network 10 and / or the CDMA network 30 and the SGS 170 that are circuit switching networks.

Hereinafter, the configuration of each node (SPS, SDS, SRS and SGS) of the PS network according to the present invention will be described in detail, and the interworking function between each node will become more apparent through the following description.

6 to 9 are functional block diagrams showing the configuration according to each function of the SPS, SDS, SRS and SGS of the PS network according to an embodiment of the present invention.

Referring to FIG. 6, the SPS 110 includes a call processor 111, a session controller 113, and a subscriber register 115.

The call processing unit 111 checks the called terminal through the message of the calling terminal requesting the call, and obtains information on the called terminal from which the call is requested from the SDS 130. A call is established between the calling terminal and the called terminal by using the obtained information about the called terminal. Here, the called terminal may be plural. That is, the call processing unit 111 may set and control a call for a 1: n call as well as a 1: 1 call. In addition, the call request message from the originating terminal may be received through the base station of the network 400, etc. may be received via the NAT.

In addition, the call processing unit 111 performs the function of the application layer gateway (ALG) to convert the function of the STUN client and the address information in the SIP message. That is, when the call processing unit 111 establishes a call with the destination terminal existing in the other network 440, since the SIP message passes through the NAT 420, the STUN server 430 located across the NAT 420. It acquires the public IP and port information of the originating terminal to be allocated through the operation of the STUN client in conjunction with the, and changes the private IP and port (particularly, IP and port of the layer 7) of the originating terminal in the SIP message. Therefore, the call with the called terminal existing in the other network 440 can be established using the public IP and the port.

In addition, the call processor 111 requests the SRS 150 to allocate a media resource (channel) to generate a call session with respect to the set call and receives the media resource. Therefore, the calling terminal can make a call (voice or video) using the created call session with the called terminal (s).

In addition, the call processing unit 111 provides a function to add the member to the session and participate in the call session when the subscriber who participated in the call session additionally invites a specific member to the call. Here, the call processor 111 may notify all the participants who participated in the call session information (eg, a phone number, an IP address, etc.) of the members who participated in the call session.

In addition, the call processing unit 111 may notify the destination terminal of the call connection notification message by using a message transmission method such as SMS when the state of the called terminal requested for call is logged out. Therefore, the called terminal user who is in the logged out state may know that a call has been made to him or her, and the call may be connected according to the response of the called terminal user. In this case, the call processing unit waits for a response of the called terminal for a predetermined time (for example, 10 seconds), sets up a call when receiving a response within a certain time, and stops connection of the call or receives a CS network if no response is received. Connect via

The session controller 113 manages session information of subscribers participating in each call session. In addition, the session controller 113 may additionally join a particular subscriber to a call session already created or withdraw from the state in which the call session is maintained.

The subscriber registration unit 115 performs a function of registering or releasing subscriber information on a subscriber who has attempted to register a logged out terminal or deregister a logged in terminal in association with an SDS. Here, the subscriber register 115 performs service authentication through authentication of a subscriber when registering a subscriber. The subscriber's authentication may be HTTP Digest authentication. In addition, the subscriber register 115 is interworking via the IP Location Server (IPLS) and the SDS 130 that manages the location and status of the terminal in the 1xEV-DO system for authentication of illegal subscribers.

Referring to FIG. 7, the SDS 130 includes a subscriber information managing unit 131, a subscriber authenticating unit 132, a subscriber state managing unit 133, an additional service providing unit 134, and a charging information processing unit 135.

The subscriber information management unit 131 performs a function of receiving, storing and managing subscriber information from a specific server of a mobile communication company coupled through a communication network when the terminal is initially registered with the mobile communication company. The subscriber information manager 131 manages the attributes of the subscriber (eg, general / intelligence network classification, WCDMA / CDMA classification, etc.), and may update them in real time when requested by the subscriber. In addition, the subscriber information management unit 131 manages the IP address information of the logged in subscriber, and sends the IP address information of the called terminal to the SPS 110 (or SGS 170 to transmit to another network) when the call is set up. to provide. In addition, the subscriber information management unit 131 interoperates with the W-HLR or the C-HLR to determine whether the called terminal is in the WCDMA network 10 or the CDMA network 30, and determines whether the called terminal is a company or a third party subscriber. It works with NPS to do this.

Subscriber authentication unit 132 interworks with IPLS for authentication of illegal subscribers, and performs the subscriber's authentication function when registering subscribers. In this case, it is preferable to manage a list of subscribers that have failed authentication, so as not to provide a service to subscribers that have failed authentication.

The subscriber state management unit 133 manages subscriber state information on whether the terminal of the subscriber is in a logged in state (executing a specific communication program) state or a logged out state (ends a specific communication program) state. When the terminal executes a specific communication program, the subscriber state management unit 133 automatically transmits state information (login state) of the terminal from the terminal and receives it through the SPS 110 without the user's separate input. It may be registered by operation. Similarly, status information for logout can be registered in the same way.

Here, the subscriber status information may further include information on call status by subscriber and service, as well as information on login / logout of the subscriber.

The supplementary service providing unit 134 includes a video mailbox service, an image monitoring service, an image 2-ring service, an additional service described in H.450 (Call Transfer), CD (Call Diversion), and CH (Call Hold). ), CP (Call Park), CW (Call Waiting), CC (Call Completion), CO (Call Offering), CI (Call Intrusion, etc.) and provides additional services such as (). The additional service provider 134 interworks with a corresponding database and a sound source server to provide each additional service.

The billing information processing unit 135 collects the billing CDR from the SRS 150 (at the time of the call between the PS terminals) or SGS 170 (at the time of the call between the PS terminals and the CS terminals) for every call at the end of the call between terminals, Stored and managed. Here, the collected charging CDR is preferably stored in a disk.

In addition, the billing information processing unit 135 makes the collected billing CDR into a file according to a predetermined condition (number of CDRs, file size, time period), and transmits the file to a billing processing device (not shown) coupled through a communication network.

Referring to FIG. 8, the SRS 150 includes a media session controller 151, a media controller 153, a charging information generator 155, and an application unit 157.

When the SPS 110 requests resource allocation in the process of generating a call session, the media session controller 151 performs a function of accommodating the resource session and allocating the corresponding resource. Resources allocated to the initial call session can be expanded or reduced as needed. In addition, the media session controller 151 performs a media session upgrade (upgrade) in real time with respect to the addition or withdrawal of a specific member among the members participating in the call session.

The media control unit 153 classifies and analyzes the media (in general, referred to as "bearer traffic" as information of voice, data, moving picture, etc.) that has entered the SRS 150 according to its attributes. In other words, it analyzes, copies, and distributes media in real time. Real-Time Transport Protocol (RTP) / User Datagram Protocol (UDP) is used for bearer traffic transmission. The RTP packet entering the SRS 150 is copied as much as the user who participated in the call session except the talker and distributed to the destination IP of each user. The media controller 153 preferably provides a real-time conversion function between different audio / video codecs of session participants. Therefore, voice / video call is possible even between terminals using different codecs.

The charging information generation unit 155 generates a charging CDR for the call between PS terminals. The billing information generation unit 155 generates a billing CDR immediately after the call is terminated, and the generated CDR is transmitted to the SDS 130. The CDR content includes billing policies (packets, times, incidents), origination and destination identifiers (for example, International Mobile Station / Subscriber Identity (IMSI), Mobile Directory Number (MDN), etc.), call volume (packet volume and calls). Time, etc.), service information, network information, as well as information necessary for billing, information necessary for processing complaints, and network-related analysis information. Accordingly, the SRS 150 may be used to process billing for all packets transmitted and received in the call session.

The application unit 157 performs an application function according to ringback tone, DTMF tone detection, PTT, chat service (IM), video conference call service, and the like. Therefore, the PS network 100 according to the present invention may perform various application functions through the application unit 157 of the SRS 150 and the additional service providing unit 134 of the SDS 130. The application unit 157 and the additional service providing unit 134 of the SDS 130 has an effect that can be added to the various application functions without additional server simply by adding and changing the service logic.

Referring to FIG. 9, the SGS 170 includes a signaling / bearer converter 171, a media converter 173, an ISUP function unit 175, and an IP converter 177.

The signaling / bearer converter 171 performs a function of converting signaling and bearers between the PS terminal and the CS terminal. The signaling transformation from the PS network 100 to the CS network is converted from SIP to ISUP No. 7, and the bearer transformation is converted from RTP to TDM (Time Division Multiplexer), and vice versa.

The media converter 173 performs a function of converting in real time into a media format suitable for the PS network 100 and the CS network during communication between the PS network 100 and the CS network.

An integrated service digital network (ISDN) user part (ISP) user part 175 performs a phone number translation function. That is, the phone number (identification information for recognizing the terminal) of the terminal of the ITU-T E.164 / E.165 type is converted into the IPv4 (or IPv6) type IP address used in the PS network 100. In addition, the ISUP function unit 175 performs a routing and call distribution function.

It will be apparent that the SGS 170 may perform all functions other than those necessary for relaying the interworking between the CS network and the PS network 100.

Hereinafter, a process for providing a communication service between a called terminal and a calling terminal according to a call request from a calling terminal of a self network 400 using a private IP and a port in the PS network 100 according to the present invention will be described. .

10 is a diagram illustrating a process of setting up a call with a destination terminal located in another network in a packet switched network system according to an exemplary embodiment of the present invention.

Referring to FIG. 10, in step 910, the SPS 110 of the PS network 100 receives a call request message for data communication such as a video and / or voice call from a calling terminal. That is, the SIP message is received from the calling terminal. In step 920, the SPS 110 determines whether the called terminal exists in its own network (a network using the same private IP and port). The SPS 110 receives the called terminal information from the SDS 130, and the IP address of the called terminal included in the received called terminal information is a private IP used in the own network or other private network via the public network, or the like. It is determined whether or not the IP used in the other network 440.

If the IP of the called terminal is not the IP in the same network 400, in step 930 to step 940, the SPS 110 acquires information on the public IP and port to be allocated later using the STUN method. That is, the SPS 110 drives the STUN client to transmit a specific message (eg, a UDP-based binding request message) to the STUN server 430 via the NAT 420. The STUN server 430 obtains information on the IP and port of the originating terminal in a specific message received via the NAT 420 and transmits the information to the SPS 110.

In step 950, the SPS 110 changes the IP and port of the originating terminal of the layer 7 in the SIP message to be the same as the public IP and the port obtained in step 940. Here, the IPs and ports of the layers 3 and 4 may also be changed. However, since the IPs and ports of the layers 3 and 4 are changed in the NAT 420, it is preferable not to perform a separate change in the SPS 110.

In step 960, the SPS 110 establishes a call with the called terminal located in the other network 440 through a gateway (not shown) of the other network 440, and in step 970, the SRS 150 establishes a call with the SPS 110. Create a session for communication such as video and / or voice call by allocating channels according to the request.

In steps 980 to 990, the SPS 110 determines whether the call is terminated and releases the call and session when the call ends.

The present invention is not limited to the above embodiments, and many variations are possible by those skilled in the art within the spirit of the present invention.

As described above, according to the present invention, there is an effect of providing a method and system capable of smoothly providing a SIP-based service in a network structure using NAT.

In addition, in using the STUN method, there is an effect of providing another network interworking method and a system in a SIP-based wireless packet switching network system that does not add a separate device and / or program to a terminal according to the STUN client.

In addition, there is an effect that can simplify the procedure according to SIP setup by running the STUN client only when the destination terminal exists in the other network.

Although described above with reference to a preferred embodiment of the present invention, those skilled in the art that various modifications of the present invention within the scope without departing from the spirit and scope of the invention described in the claims below And can be changed.

Claims (11)

In the other network interworking method in a wireless packet switching network system combined with a STUN server via NAT, (a) receiving a call request message from an originating terminal; (b) recognizing that the called terminal is located in the other network by using information of the called terminal corresponding to the call request message stored in a database server; (c) driving a STUN client to obtain information about a public IP and a port to be allocated in the NAT from the STUN server; (d) changing the private IP and port of Layer 7 in the call request message to be the same as the public IP and port; (e) establishing a call with the called terminal using the changed call request message; And (f) a relay server interworking with each other in a SIP-based wireless packet switching network system including generating a session for transmitting media traffic between the calling terminal and the called terminal according to a request of the SIP proxy server; Way. The method of claim 1, wherein step (b) comprises: Requesting, by the SIP proxy server, the call request message from the terminal, the information on the called terminal to the database server; And And transmitting the information of the called terminal including the IP information of the called terminal from the database server to the SIP proxy server, wherein the SIP proxy server is located in the other network through the IP information. The other network interworking method in a SIP-based wireless packet switching network system characterized in that the recognition. The method of claim 2, The STUN client is another SIP interworking method in a SIP-based wireless packet switching network system characterized in that the SIP proxy server is driven. delete In a wireless packet switching network system, A SIP proxy server for establishing (signaling) a call with at least one called terminal and controlling a session between the calling terminal and the called terminal according to a call request message transmitted from the calling terminal; A database server that manages information of a subscribed terminal and state information of the subscribed terminal, and provides destination terminal information requested by the SIP proxy server for signaling to the SIP proxy server; And A relay server for allocating at least one channel so that the session is created in response to a channel request from a SIP proxy server that has established the call, The SIP proxy server acquires information on the public IP and port to be allocated in the NAT from the STUN server coupled via a NAT by driving a STUN client when the called terminal is located in another network, and the call request SIP-based wireless packet-switching network system interworking with another network, wherein the call is set up with the called terminal using the changed call request message by changing the private IP and port of the layer 7 in the message to the same as the public IP and port. . The method of claim 5, And the SIP proxy server recognizes that the called terminal is located in the other network by using the called terminal information. The method of claim 5, wherein the SIP proxy server A call processing unit for setting up and releasing the call, driving the STUN client, and performing the channel request to the relay server; And And a session controller for managing session information of the calling terminal and the called terminal participating in the session. The method of claim 5, wherein the database server, A subscriber information management unit managing information of a subscribed terminal; And Includes a subscriber status management unit for managing the status information of the subscribed terminal whether the subscribed terminal is logged in or logged out state, And the subscriber information manager transmits the called terminal information requested by the SIP proxy server to set up a call to the SIP proxy server. The method of claim 5, wherein the relay server, A media session controller for allocating the channel to the SIP proxy server; A media controller configured to analyze, copy, and distribute media according to communication between the calling terminal and the called terminal when there are a plurality of called terminals; And And a billing information generation unit for generating a billing CDR according to communication between the originating terminal and the called terminal. The method of claim 5, And a gateway server for relaying communication between the calling terminal and the called terminal in association with the circuit switching network when the called terminal exists in the circuit switched network. The method of claim 10, wherein the gateway server, A signaling / bearer converter for converting signaling and bearers between the calling terminal and the called terminal existing in the circuit switched network; A media converter configured to perform real-time conversion of a media format when communicating between the calling terminal and a called terminal existing in the circuit switched network; And SIP-based wireless packet-switching network system interworking with the other network, characterized in that it comprises an ISUP function for performing a routing (routing) and call distribution function according to the call setup.

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