KR100680290B1 - Routing method in wireless packet switching network system based on sip and the system thereof - Google Patents

Abstract

A termination routing method in a SIP(Session Initiation Protocol)-based wireless PS(Packet Switching) network system and a system thereof are provided to enable a single database server to offer destination terminal information for reception routing to a destination terminal of another company's network, so that a SIP proxy server can easily acquire the destination terminal information for reception routing. A SIP-based wireless PS network(100) comprises an SPS(SIP Proxy Server)(110), an SDS(SIP DB Server)(130), an SRS(SIP Relay Server)(150), and an SGS(SIP Gateway Server)(170). The SPS(110) covers an SIP proxy function, and executes a call processing function with a destination terminal according to a call request of an originating terminal. The SDS(130) executes a function of managing subscriber information. In case a destination terminal, corresponding to a call setup request from an originating terminal, doesn't exist in the PS network(100), the SDS(130), interworking with an HLR in a WCDMA network(10) or a CDMA network(30), judges whether the destination terminal is located in the WCDMA network(10) or the CDMA network(30). The SRS(150) controls a media session in accordance with data communication between terminals. The SGS(170) provides an interworking function with other networks, such as the WCDMA network(10), the CDMA network(30), etc.

Description

Routing method in wireless packet switching network system based on sip and system therefor {Routing method in wireless packet switching network system based on SIP and the system}

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

2 to 5 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.

6 is a diagram illustrating a process of performing a call connection between an originating PS terminal and a called PS terminal in a PS network according to an embodiment of the present invention.

7 is a diagram illustrating a call setup process between a PS network and a CS network according to an embodiment of the present invention.

8 is a diagram illustrating a process of recognizing destination terminal information for incoming routing in a database server of a packet switched network according to an embodiment of the present invention.

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

10: WCDMA Network

30: CDMA network

100: packet switched network system

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)

The present invention relates to incoming routing in a wireless packet-switched network, and more particularly, to a destination routing method and a system in a SIP-based wireless packet-switched 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 MEGACO.

Currently, not only Korea, but also many other countries have mixed circuit switching networks such as WCDMA, CDMA, GSM, and packet switching networks, so interworking with each network may be an important issue to provide smooth call services. In particular, the routing method used in the current IP network has a problem that the current communication network cannot be applied to the incoming routing for VoIP as the circuit switched network and the packet switched network coexist. That is, there is a need for a call routing method for establishing a call between an originating terminal and a called terminal for a call between a packet switching network and a destination switching network as well as a packet switching network for VoIP and leading to bearer traffic transmission.

Accordingly, the present invention has been made to solve the above-mentioned problems, SIP can be set up with the destination terminal of the circuit-switched network such as WCDMA network, CDMA network, SIP-based to provide smooth VoIP interworking with other networks An object of the present invention is to provide an incoming routing method and a system in a wireless packet switched network system.

Another object of the present invention is to provide the destination terminal information for the incoming routing to the destination terminal according to not only the circuit switching network but also the other company's network in one database server, so that the incoming routing is performed in the SIP proxy server that sets up the call. The present invention provides a method and system for receiving a routing in a SIP-based wireless packet switching network system capable of easily acquiring destination terminal information.

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, a method for routing incoming calls in a SIP-based wireless packet switching network system including a SIP proxy server and a database server, (a) the SIP proxy server from the originating terminal Receiving a call request including identification information of at least one called terminal; (b) requesting, by the SIP proxy server, the database server for called terminal information for call establishment using the identification information; (c) providing, by the database server, called terminal information corresponding to the identification information to a SIP proxy server; And (d) establishing, by the SIP proxy server, a call between the originating terminal and the called terminal using the called terminal information.

Here, the recognition information may be a telephone number.

And, the step (c), if the called terminal is a PS video terminal, comprising the step of recognizing whether or not the called terminal is logged in state information of the called terminal, the called terminal is logged in state In the case that the IP address of the called terminal may be included in the called terminal information. In addition, the step (c) includes the step of recognizing whether or not the destination terminal is registered in a location register of a preset circuit switching network when the destination terminal is a PS video terminal and logged out, the destination terminal The information may include information according to the recognized result.

And, the step (c), if the destination terminal is a CS video terminal, comprising the step of recognizing whether or not the destination terminal is registered in a location register of a circuit switching network, the destination terminal information is It may include information according to the recognized result. The preset circuit switched network may be at least one of a WCDMA network and a CDMA network. The SIP proxy server may establish a call between the calling terminal and the called terminal through a gateway server included in the wireless packet switching network when the called terminal is registered in the circuit switched network.

And, the step (c), if the destination terminal is not the PS video terminal and the CS video terminal, comprising the step of recognizing whether or not the destination terminal is its subscriber using a number portability system (NPS), The called terminal information may include information according to the recognized result. In addition, when the called terminal is its own subscriber, the called terminal is recognized as a CS-only terminal capable of only a voice call, so that the SIP proxy server is configured to allow the exchange of a circuit switched network corresponding to the calling terminal to set up the call. Information may be provided to the exchange.

According to another aspect of the present invention, a wireless packet switching network system comprising: a SIP proxy server for receiving a call request including identification information of at least one destination terminal from an originating terminal and establishing a call using the identification information; A database server that manages information of a subscribed terminal and state information of the subscribed terminal, and provides destination terminal information corresponding to the identification information requested by the SIP proxy server to set up the call to the SIP proxy server; And a gateway server 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, wherein the database server determines whether the called terminal is a PS network terminal or a CS network terminal. Provided is a SIP-based wireless packet switching network system which determines and transmits corresponding information to the SIP proxy server.

Here, the database server, the subscriber information management unit for managing the information of the subscribed terminal; And a subscriber state manager that manages state information of the subscribed terminal as to whether the subscribed terminal is logged in or logged out, wherein the subscriber information manager is the called terminal requested by the SIP proxy server when a call is set up. Information may be transmitted to the SIP proxy server.

The subscriber information management unit manages the IP address information of the PS video terminal that is logged in. When the called terminal is a PS video terminal and is in a logged in state, the IP address information of the called terminal is included in the called terminal information. Can be sent to the proxy server.

Further, when the called terminal is a PS video terminal and logged out, the subscriber information management unit acquires the called terminal information from at least one of a location register of a WCDMA network, a location register of a CDMA network, and a number portability system (NPS). It can be provided to the SIP proxy server.

And, when the called terminal is a CS video terminal, the subscriber information management unit recognizes whether the called terminal is registered in a preset location register of the circuit-switching network, and the information according to the result recognized in the called terminal information It can be provided including.

When the called terminal is not the PS video terminal and the CS video terminal, the subscriber information management unit recognizes whether the called terminal is its own subscriber using a number portability system (NPS), and the called terminal information is determined by the called terminal information. It may include information according to the recognized result. In addition, when the called terminal is its own subscriber, the called terminal is recognized as a CS-only terminal capable of only a voice call, so that the SIP proxy server is configured to allow the exchange of a circuit switched network corresponding to the calling terminal to set up the call. Information may be provided to the exchange.

Hereinafter, a method of routing an incoming call in a SIP-based wireless packet switching network system according to a preferred embodiment of the present invention, and a system thereof will be described in detail with reference to the accompanying drawings. The same or corresponding components will be given the same reference numerals and redundant description thereof will be omitted.

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

1, a WCDMA network 10 and a CDMA network 30 (hereinafter collectively referred to as a 'CS network') and a packet switching (PS) network (CS) Three communication networks are shown (100) (hereinafter referred to as 'PS networks'). PS network 100 is a SIP-based wireless packet network, PS network 100 according to the present invention is a SIP proxy server (SPS: SIP Proxy Server) 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 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.

Here, 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 may transmit a short message through the SMSC to enable the call when the called terminal is in a logout state (a state in which communication is not possible through the PS network such as a preset specific communication program is not executed).

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) 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, the unified HLR, and the location register of the WCDMA network 10 W-HLR, the location register of the CDMA network 30 will be referred to as C-HLR) in conjunction with the destination terminal WCDMA network (10) ) Or the CDMA network 30 is determined. 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, the information about the called terminal present in the CS network (including other companies' networks) as well as the PS network 100. Network information).

Here, the terminal according to the present invention can be called through the PS network 100 (also possible through the CS network) and the terminal is supported video (hereinafter referred to as "PS video terminal"), only through the CS network And a terminal capable of supporting video (for example, an existing WCDMA video terminal, hereinafter referred to as a "CS video terminal") and a terminal capable of only a voice call without a video support (a terminal capable of calling through a CDMA network only, hereinafter "CS"). -only terminal ". Accordingly, the SDS 130 of the PS network 100 according to the present invention may provide the SPS 110 with information about the called terminal according to the type and status (either login or logout) of the called terminal.

In addition, 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 a subscriber from a CDMA data network point of view (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 the application function is stored in the SDS 130. Here, a packet data serving node (PDSN) (not shown) that provides a Point-to-Point Protocol (PPP) interface with a 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 other networks (particularly, CS networks) such as CDMA and WCDMA, and is a PS terminal (PS network available terminal). And signaling between the CS terminal (terminals available only to the CS network) (interworking between the SIP of the PS network and the H.324M of the CS network). 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 100 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. will be.

2 to 5 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.

Referring to FIG. 2, 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. Here, the call processing unit 111 may receive a call request message from the originating terminal through a base station or an access point (AP).

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 about a subscriber who attempts to register a logged out terminal or deregister a logged in terminal in association with the SDS 130. 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. 3, the SDS 130 includes a subscriber information management unit 131, a subscriber authentication unit 132, a subscriber state management 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.

According to a preferred embodiment of the present invention, the SDS 130 may have a routing table as follows.

SDS W-HLR NPS Determination of destination terminal type / status and routing method accordingly PS video terminal CS video terminal Location registration Zaman login Log out O - - - - 1. PS Log-in status (video) → IP Routing by SPS X O - O - 2. PS Log-out & WCDMA Network (Voice) → No.7 SIG to SGS => 3G-MSC. X O - X - 3. PS Log-out & CDMA Network (Voice) → No.7 SIG to SGS => 2G-MSC. X X O O - 4. CS video & WCDMA network (video) → No.7 SIG by SGS => 3G-MSC. X X O X - 5. CS Video & CDMA Network (Voice) → No.7 SIG with SGS => 2G-MSC. X X X X X 6. Targeting (video or voice) → SGS => IP Routing to Third Party G / W X X X X O 7. NO SERVICE → NAK response to originating terminal

<Routing table>

According to the routing table, the routing method is set according to the type and status information of the called terminal. First, the SDS 130 determines whether the called terminal is a PS video terminal, and recognizes whether it is a logged in state or a logged out state through state information in the case of a PS video terminal. In the case of the PS video terminal in the logged-in state, the destination terminal information including the IP address of the corresponding destination terminal is transmitted to the SPS 110. If the called terminal is a PS video terminal and is in the logout state, the called terminal acquires the called terminal information from the W-HLR and provides it to the SPS 110 so as to perform a video and / or voice call through the WCDMA network 10. do. The SDS 130 determines that the destination terminal exists in the CDMA network 30 when there is no destination terminal information in the W-HLR, so that the SPS 110 exchanges the CDMA network 30 via the SGS 170 ( 2G-MSC) and transmits the relevant information (including network information associated with the destination terminal) to the SPS (110) to establish a call. Accordingly, the SPS 110 may perform signaling (converting SIP-based signaling to N.7 signaling of the CS network) through the SGS 170 to establish a call for a call in association with the corresponding CS network.

As shown in the routing table, if the destination terminal is a CS video terminal, in a similar manner, the SDS 130 determines whether the destination terminal is registered in the WCDMA network 10, and if so, the WCDMA 10 network. The call is established through the CDMA network 30 if the call is not registered.

In addition, the SDS 130 determines whether or not the destination terminal is a PS video terminal or a CS video terminal through its NPS, and if the destination terminal is a third party subscriber, the SPS 110 is a gateway of a third party network. To establish a call with the destination terminal through.

When the called terminal is its own subscriber, the called terminal recognizes it as a CS-only terminal capable of voice call only, and does not perform call setup through the SPS 110 (usually a CDMA network because only voice calls are possible). Voice calls over the bay are established. That is, the SDS 130 transmits the relevant information to the SPS 110 so that the SPS 110 sends information to the exchange of the CDMA network 30 via the SGS 170 (called terminal information such as a telephone number). ). Here, the SPS 110 may transmit a message indicating that the call connection through the PS network 100 is impossible to the calling terminal. In addition, the calling terminal receiving the message from the SPS 110 that the call connection is impossible through the PS network 100 may alternatively perform a call connection through the CDMA network 30.

As described above, the SDS 130 transmits the called terminal information according to the routing method corresponding to the type and status information of the called terminal to the SPS 110 when the receiving terminal information for the incoming routing request from the SPS 110; do. Accordingly, the SPS 110 may perform the incoming routing according to the type and state of the called terminal.

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 the 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. 4, 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. 5, the SGS 170 includes a signaling / bearer converter 171, a media converter 173, and an ISUP function unit 175.

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.

6 is a diagram illustrating a process of performing a call connection between an originating PS terminal and a called PS terminal in a PS network according to an embodiment of the present invention, and FIG. 7 is a PS network and a CS according to an embodiment of the present invention. A diagram illustrating a call establishment process between networks.

Referring to FIG. 6, in step 610, the SPS 110 receives a call connection request message from the originating PS terminal 50. Accordingly, the SPS 110 performs signaling in steps 620 through 650. That is, a step for setting up a call between the calling PS terminal 50 and the called PS terminal 60 is performed.

In step 620, the SPS 110 transmits the called terminal information inquiry request message including the identification information (for example, the telephone number) of the called terminal recognized from the call connection request message to the SDS 130, and in step 630 130 queries whether the called terminal is logged in or logged out through the called terminal state information. In step 640, the SDS 130 transmits the called terminal information including the IP address of the called terminal to the SPS 110 in the login state, and in step 650, the SPS 110 communicates with the calling PS terminal 50; A call between the called PS terminals 60 is set. Here, when the called PS terminal 60 is in the logout state, a call for a call is set up using a CS network, which will be described later with reference to the related drawings.

In steps 660 to 670, the SPS 110 requests the SRS 150 a media resource (channel) for a voice and / or video call, and the SRS 150 allocates the resource in response. Accordingly, in step 680, a session between the originating PS terminal 50 and the terminating PS terminal 60 is established and a voice and / or video call is performed via the SRS 150.

Hereinafter, a process of setting up a call in conjunction with a CS network when the called terminal is a logged out PS video terminal or a CS video terminal will be described with reference to FIG. 7.

Referring to FIG. 7, in step 710, the SPS 110 receives a call connection request message from the originating PS terminal 50. Accordingly, the SPS 110 performs signaling in steps 720 to 755.

That is, in step 715, the SPS 110 transmits the called terminal information inquiry request message including the identification information (for example, the telephone number) of the called terminal 70 recognized from the call connection request message to the SDS 130. . Accordingly, the SDS 130 inquires whether the called terminal 70 is a PS terminal or a CS terminal at step 720. In addition, when the called terminal 70 is a PS terminal, it is inquired through the called terminal state information whether it is logged in or logged out.

If the called terminal 70 is a PS terminal and is logged out or is a CS terminal, in step 725 to step 735, the called terminal 70 interworks with the HLR (W-HLR or C-HLR) 700 of the CS network. The called terminal information including information on the located network and the corresponding exchange is transmitted to the SPS 110. Here, the SPS 110 does not set up a call when the terminating terminal 70 is located in the CDMA network 30, but instead of a corresponding switch of the CDMA network 30 via the SGS 170 (calling terminal side exchange) ( Related information may be transmitted to the exchange receiving the call to set up the voice call. This is because when the terminating terminal 70 is located in the CDMA network 30, it is difficult to transmit and receive multimedia data such as a video call, so that the call side is not complicated by using the PS network 100 and the CDMA network 30 is also set by the calling party. This is because it is preferable to set the call by a simple method using. In addition, when the terminating terminal 70 is a third party subscriber, information about a network in which the terminating terminal 70 is located and a corresponding exchange (or an SPS 110 of the third party PS network 100 may be known) is linked to the NPS. It can also be acquired.

In steps 740 to 755, the SPS 110 establishes a call with the exchange 710 on the destination terminal 70 side through the SGS 170. In addition, the SPS 110 requests the SRS 150 media resources (channels) for voice and / or video calls, and the SRS 150 allocates the resources in response. Accordingly, a session between the originating PS terminal 50 and the called terminal 70 is established, and a voice call or a video call is performed via the SGS 170 in step 760. Here, the SGS 170 converts data (data according to signaling and data based on signaling) received from each network into a data format suitable for the PS network 100 and the CS network and transmits the data to each network.

Hereinafter, the process of recognizing information on the type of the called terminal for the incoming routing and the network where the called terminal is currently located in the SDS 130 will be described.

8 is a diagram illustrating a process of recognizing destination terminal information for incoming routing in a database server of a packet switched network according to an embodiment of the present invention.

Referring to FIG. 8, in step 810, the SDS 130 receives a signaling request including the telephone number of the called terminal from the SPS 110. In step 820, the subscriber information is retrieved using the telephone number of the called terminal, and the type of the called terminal is recognized.

In step 830 to step 870, the destination terminal information is acquired and transmitted to the SPS 110 in a manner corresponding to the type of the destination terminal. That is, when the called terminal is a PS video terminal, it checks whether or not to log in through the status information, and when logged in, provides the corresponding IP address to the SPS 110 as the called terminal information, and when logged out, registers with the W-HLR. The W-HLR checks and recognizes whether the information has been received and provides the information about the W-HLR to the SPS 110 as the called terminal information. If the destination terminal is a CS video terminal, the W-HLR is queried whether the destination terminal is registered to determine whether the WCDMA network 10 can be used, and information about the destination terminal is obtained and provided to the SPS 110. In addition, when the called terminal is not the PS video terminal and the CS video terminal, it recognizes whether the called terminal is the subscriber of the own network or the other network through the NPS and provides information on the information to the SPS 110 as the called terminal information.

Therefore, the step SPS 110 can perform the call setup between the calling terminal and the called terminal by obtaining the called terminal information from the SDS 130 in a manner corresponding to the type of the called terminal and / or whether or not to log in and the communication available network. have.

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, a SIP-based wireless packet switching network system capable of establishing a call with a destination terminal of a circuit switched network such as a WCDMA network or a CDMA network, and providing VoIP with smooth interworking with other networks. There is an effect that can provide an incoming routing method and the system in.

In addition, since one database server can provide the destination terminal information for the destination routing to the destination terminal according to the network of the other company as well as its circuit switching network, the destination terminal information for the destination routing in the SIP proxy server to establish a call There is an effect of providing an incoming routing method and a system in a SIP-based wireless packet-switching network system that can be easily obtained.

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 that it can be changed.

Claims (16)

A method for routing incoming calls in a SIP-based wireless packet switching network system including a SIP proxy server and a database server, (a) receiving, by the SIP proxy server, a call request including identification information of at least one called terminal from the calling terminal; (b) requesting, by the SIP proxy server, the database server for called terminal information for call establishment using the identification information; (c) providing, by the database server, called terminal information corresponding to the identification information to a SIP proxy server; And (d) the SIP proxy server establishing a call between the calling terminal and the called terminal by using the called terminal information; Incoming routing method in a SIP-based wireless packet switching network system comprising a. The method of claim 1, And the identification information is a telephone number. Incoming routing method in a SIP-based wireless packet switching network. The method of claim 1, wherein step (c) comprises: If the called terminal is a PS video terminal, comprising the step of recognizing whether the called terminal is in a login state through the status information of the called terminal, And when the called terminal is in a logged in state, providing the IP address of the called terminal in the called terminal information. The method of claim 3, wherein step (c) comprises: And if the called terminal is a PS video terminal and is in a logout state, recognizing whether the called terminal is registered in a preset location register of a circuit switched network, wherein the called terminal information is information based on the recognized result. Incoming routing method in a SIP-based wireless packet switching network comprising a. The method of claim 1, wherein step (c) comprises: And when the called terminal is a CS video terminal, recognizing whether the called terminal is registered in a preset location register of a circuit switched network, wherein the called terminal information includes information according to the recognized result. Incoming routing method in a SIP-based wireless packet switching network, characterized in that. The method according to claim 4 or 5, The preset circuit switched network is an incoming routing method in a SIP-based wireless packet switched network, characterized in that at least one of a WCDMA network and a CDMA network. The method according to claim 4 or 5, In the SIP-based wireless packet switching network, the SIP proxy server sets up a call between the calling terminal and the called terminal through a gateway server included in the wireless packet switching network when the called terminal is registered in the circuit switched network. Incoming routing method. The method of claim 1, wherein step (c) comprises: If the called terminal is not the PS video terminal and the CS video terminal, comprising the step of recognizing whether the called terminal is its own subscriber using a number portability system (NPS), the called terminal information is the result of the recognition Incoming routing method in a SIP-based wireless packet switching network comprising the information according to. The method of claim 8, When the called terminal is its own subscriber, the called terminal is recognized as a CS-only terminal capable of voice call only, and the SIP proxy server provides relevant information so that the switch of a circuit switched network corresponding to the calling terminal can establish the call. Incoming routing method in a SIP-based wireless packet switching network, characterized in that provided to the exchange. In a wireless packet switching network system, A SIP proxy server that receives a call request including identification information of at least one called terminal from an originating terminal and establishes a call using the identification information; A database server that manages information of a subscribed terminal and state information of the subscribed terminal, and provides destination terminal information corresponding to the identification information requested by the SIP proxy server to set up the call to the SIP proxy server; And A gateway server 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; And the database server determines whether the called terminal is a PS network terminal or a CS network terminal and transmits corresponding information to the SIP proxy server. The method of claim 10, wherein the database server, A subscriber information management unit managing information of the subscribed terminal; 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 11, The subscriber information management unit manages IP address information of the PS video terminal that is logged in. When the called terminal is a PS video terminal and is logged in, the IP address information of the called terminal is included in the called terminal information. SIP-based wireless packet switching network system, characterized in that transmitted to. The method of claim 11, When the called terminal is a PS video terminal and logged out, the subscriber information management unit acquires the called terminal information from at least one of a location register of a WCDMA network, a location register of a CDMA network, and a number portability system (NPS). SIP-based wireless packet switching network system, characterized in that provided by the SIP proxy server. The method of claim 11, When the called terminal is a CS video terminal, the subscriber information management unit recognizes whether the called terminal is registered in a preset location register of a circuit-switching network, and includes the information according to the recognized result in the called terminal information. SIP-based wireless packet switching network system, characterized in that provided by. The method of claim 11, When the called terminal is not the PS video terminal and the CS video terminal, the subscriber information management unit recognizes whether the called terminal is its own subscriber using a number portability system (NPS), and the called terminal information is the recognized terminal information. SIP-based wireless packet switching network system comprising the information according to the result. The method of claim 15, If the called terminal is its own subscriber, the called terminal is recognized as a CS-only terminal capable of a voice call only, and the SIP proxy server is configured to allow the exchange of a circuit switched network corresponding to the calling terminal to establish the call. SIP-based wireless packet switching network system, characterized in that for providing to the exchange.

Images