KR101119673B1 - Method and apparatus for supporting VoIP service based on soft switch - Google Patents

Abstract

The present invention relates to a mobile Internet system, and more particularly, to a method and apparatus for supporting VoIP service.

The present invention performs call processing according to a call request from a terminal and a VoIP service request from a peer in a soft switch-based portable Internet network, and sets up QoS triggering and service class (BE, UGS) according to the requested VoIP service. Can support / release This enables VoIP services in soft switch-based portable Internet networks.

Soft Switch (SSW), Internet Multimedia Subsystem (IMS), Voice over Internet Protocol (VoIP), Quality of Service (QoS), Unsolicited Granted Service (UGS), Session Initiation Protocol (SIP), Application Programming Interface (API), ISF (Initial Service Flow), Snooper

Description

Method and apparatus for supporting VoIP service based on soft switch

The present invention relates to a mobile Internet system, and more particularly, to a method and apparatus for supporting VoIP service.

The portable Internet system combines the advantages of high-speed data transmission provided by a wired network and the mobility provided by a wireless network, and is a next-generation communication system that enables a variety of contents and services to be provided at high speed anytime, anywhere, even when stopped or on the move.

In the portable Internet system, since data is transmitted and received between the portable Internet terminal (hereinafter referred to as a "terminal") and a base station by using a radio, there may be a large change in the data transmission rate according to the characteristics of the wireless channel and the environment. Services that require latency, jitter, and bandwidth guarantees, such as voice calls and video services, must support quality of service (QoS) to support service flow (eg, UGS) setup and QoS triggering. Here, QoS refers to efficiently providing a limited resource (traffic, bandwidth) by differentiating a service quality level according to the importance of a subscriber or a provided service.

In the portable Internet system, technology development for providing a Voice over Internet Protocol (VoIP) service that enables voice calls by converting voice data into IP (Internet Protocol) data packets among multimedia services is being progressed.

In order to provide VoIP service in the portable Internet system, call control with terminals subscribed to the same network as well as other communication networks (PSTN, PLMN, wired and wireless networks) must be controlled. For this, Session Initiation Protocol Call control protocol is used.

In addition, in order to provide a VoIP service in a portable Internet system, a hardware-based Internet Multimedia Subsystem (IMS) or a software-based soft switch that supports data transmission in a network is used.

First, IMS was developed to guarantee the quality of multimedia in all IP environments. It supports the transmission of multimedia data in current 2G, 3G, and WiMAX systems, including multiple hardware configurations to provide multimedia services. .

Such IMS does not have enough user products for each of the various multimedia services, and has to have hardware for each multimedia service including VoIP. Therefore, the complexity of the system is high and the maintenance is difficult. In addition, there is a disadvantage that a large cost to implement the system.

On the other hand, the soft switch supports the function of the exchange means such as the IMS method through software, and is widely used due to the advantage of reducing the complexity and cost of the system when implemented compared to the IMS method.

However, the soft switch method according to the IEEE 802.16e and WiMAX standards does not consider the creation of the UGS (Unsolicited Granted Service) service class and QoS setting / release according to the VoIP service request. Therefore, there is a difficulty in providing VoIP service that should apply the UGS service class.

SUMMARY OF THE INVENTION The present invention has been made in view of the above-described problems, and provides a method and apparatus for supporting VoIP service in a soft switch-based portable Internet network.

An object of the present invention is to provide a VoIP service support method and apparatus capable of allocating and releasing QoS by a terminal or a network for a VoIP service in a soft switch-based portable Internet network.

VoIP service support method according to an embodiment of the present invention for achieving the above object in the soft switch-based wireless communication network, after the network entry and SIP (Session Initiation Protocol) registration procedure of the terminal and the terminal and Delivering a SIP signaling message to establish a session for VoIP service between the soft switches; Transmitting VoIP data over the session; snooping the SIP signaling message during the delivery, and QoS triggering to allocate QoS resources for the session based on information extracted from the SIP signaling message. It further comprises.

VoIP service support method according to an embodiment of the present invention for achieving the above object in the soft switch-based wireless communication network, the VoIP service between the terminal and the soft switch after the network entry and SIP registration procedure of the terminal is performed Sending a SIP signaling message to establish a session for the message; Transmitting VoIP data through the session after establishing the session; detecting the SIP signaling message in the terminal and assigning QoS resources for the session based on information extracted from the SIP signaling message. QoS triggering further comprises.

In an apparatus for supporting VoIP service according to an embodiment of the present invention for achieving the above object, in a soft switch-based wireless communication network, a VoIP service between a terminal and the soft switch after a network entry of the terminal and a SIP registration procedure are performed An ASN-GW for delivering a SIP signaling message and for transmitting VoIP data over the session to establish a session for the; The ASN-GW includes a snooper that snoops the SIP signaling message during the delivery and QoS triggers to allocate QoS resources for the session based on information extracted from the SIP signaling message. .

An apparatus for supporting VoIP services according to an embodiment of the present invention for achieving the above object includes a modem for transmitting and receiving a SIP signaling message according to a VoIP service request with a soft switch-based wireless communication network; A VoIP client establishing a session for the VoIP service with the soft switch based on identification information, QoS information, and classification information of the terminal extracted from the SIP signaling message, and performing QoS triggering so that QoS resources for the session are allocated; And an application programming interface (API) for converting and relaying the SIP signaling message between the modem and the VoIP client.

According to an embodiment of the present invention, in a soft switch-based portable Internet system, a VoIP call session may be created / released according to a call request from a terminal and a VoIP service request from a peer, thereby providing a VoIP service.

According to an embodiment of the present invention, in a soft switch-based portable Internet system, a VoIP service can be provided by supporting QoS triggering and service flow (UGS) allocation / release according to call requests from a terminal and a network.

Hereinafter, with reference to the accompanying drawings will be described embodiments of the present invention.

Prior to the detailed description with reference to the drawings, the IEEE 802.16e, WiMAX standard according to the QoS requirements of each service, the service provided by the Unsolicited Grant Service (UGS), Real-time Polling Service (rtPS), Extended Real-time (ertPS) Polling Service), nrtPS (Non Real-time Polling Service), and BE Service (Best Effort Service).

Among them, BE is a non-real time service, which is used for general data transmission that is not time sensitive, and UGS is a real time service, which is sensitive to time, delay, and bandwidth, such as VoIP. Used for

In the case of VoIP, if the delay constraints and jitter on the transmitted data cannot be satisfied, the data cannot be used even if the data is successfully transmitted. Thus, VoIP requires transmission through UGS, which can guarantee latency, jitter, and bandwidth for the data being transmitted. Therefore, in order to support VoIP service in portable Internet system, not only BE but also Quality of Service (QoS) for UGS should be supported.

According to an embodiment of the present invention, in order to support a VoIP service, the portable Internet system, as shown in FIG. 1, has a terminal 100, a base station 200, and an ASN-GW (300, Access Service Network-GateWay). And a CSN (Core Service Network) 400 and a VoIP CSN 500.

The terminal 100 includes an application programming interface (API) 110, a voice over internet protocol client (VoIP) client 120, and a portable internet modem 130 (hereinafter referred to as an IEEE 802.11 modem, hereinafter referred to as a "modem").

The ASN-GW 300 detects and analyzes call signaling between the terminal 100 and the VoIP CSN 500 and generates a service flow according to a call request. To this end, it includes a SIP snooper 310 (Snap Initiation Protocol Snooper) for analyzing the SIP signaling message, and an Anchor Service Flow Administrator (AFA) 320 for generating a service flow according to the call request. Here, the detection of the SIP signaling message is made in the data plane of the ASN-GW 300.

The VoIP CSN 500 performs call processing based on a call request from the terminal 100 and call processing based on a call request from the Peer 600 to the terminal 100, and performs a soft switch 510. Soft Switch), AS (520, Application Server), MGW (530, Media GateWay), SGW (540, Signaling GateWay), MRCF (550, Media Resource Control Function).

The portable Internet system according to the embodiment of the present invention including the above-described configuration can support VoIP service in two aspects. The first is to support the VoIP service by performing QoS triggering according to the VoIP service request from the terminal 100, and the second is to perform the QoS triggering according to the VoIP service request from the network to support the VoIP service.

In the following description, the API 110, the SIP snooper 310, and the soft switch 510 for supporting the VoIP service will be described in detail, and other components will be omitted to clarify the gist of the present invention. Can be. In addition, for details not described above, the WiMAX NWG standard document and the 3GPP standard document may be referred to.

In the portable Internet system of FIG. 1, an R1 interface is connected between the terminal 100 and the base station 200, and an R6 interface is connected between the base station 200 and the ASN-GW 300, and the ASN-GW 300 and the CSN ( 400) is connected to the R3 interface. Although not shown in the figure, the base stations are connected by an R4 interface.

The terminal 100 receives downlink (DL) data transmitted from the base station 200 and transmits uplink (UL) data to the base station 200.

The modem 130 transmits the VoIP data and the SIP signaling message received from the VoIP client 120 to the base station 200 through the API 110. In addition, it receives the VoIP data and the SIP signaling message from the base station 200 and transmits the received VoIP data and the SIP signaling message to the VoIP client 120 through the API 110. The modem supports the IEEE 802.16e standard as an example.

Here, the uplink (UL) SIP signaling message is transmitted from the VoIP client (Client) 120 to the modem 130 via the API 110, and then through the base station 200 and the ASN-GW 300 of the network. Is transmitted to the VoIP CSN 500.

The downlink SIP signaling message is transmitted from the VoIP CSN 500 of the network to the terminal 100 through the ASN-GW 300 and the base station 200. Then, it is transmitted from the modem 130 to the VoIP client 120 via the API 110.

The SIP signaling message is for supporting multimedia communication in the network. In the case of a VoIP service, the SIP signaling message includes classification information and QoS information for QoS allocation / release according to a call request from a terminal and a VoIP call request from a network. Such QoS information and classification information are included in the SIP signaling message in the API 110 of the terminal 100 and the SGW 540 of the VoIP CSN 500.

In the process of QoS allocation, the classification information of the uplink (UL) and the downlink (DL) includes a source IP, a source port number, a protocol, a destination IP, and a destination port number. In addition, the QoS information of the uplink (UL) and the downlink (DL) includes a guaranteed data rate and a maximum data rate.

In the QoS release process, the classification information of the uplink (UL) includes a source IP, a source port number and a protocol, and the classification information of the downlink (DL) includes a protocol, a destination IP, and a destination port number.

The VoIP client 120 is implemented in the form of an application program in the terminal 100 to provide a VoIP service to the subscriber. The VoIP client 120 may have various forms according to the service characteristics of the subscriber and the service provided by the service provider. It can be implemented as.

In order to transmit the VoIP data and the SIP signaling message output from the VoIP client 120 to the modem 130, and to transmit the VoIP data and the SIP signaling message received through the modem 130 to the VoIP client 120, the VoIP client ( An interface for connecting 120 and modem 130 is required.

To this end, the terminal 100 converts a SIP signaling message for relaying and QoS triggering of VoIP data between the VoIP client 120 and the modem 130 and transmitting the SIP signaling message to the VoIP client 120 and the modem 130. It includes an API (110).

The API 110 is driven by calling a connection function to perform a connection between heterogeneous entities, and performs an interface function for connecting heterogeneous entities such as the VoIP client 120 and the modem 130.

Specifically, the API 110 converts the VoIP data and the SIP signaling message from the VoIP client 120 into a form suitable for the modem 130 and relays the data to the modem 130. In addition, the VoIP data and the SIP signaling message from the modem 130 are converted into a form suitable for the VoIP client 120 and relayed to the VoIP client 120. Here, the conversion and relaying of the SIP signaling message performed in the AIP 100 may be equally applied in the uplink (UL) and the downlink (DL).

The base station 200 transmits data including SIP and SIP signaling messages (control signaling) between the terminal 100 and the ASN-GW 300 to the terminal 100 and the ASN-GW 300. The base station 200 may include a PHS 210 (Personal Handy System).

The CSN 400 generates rules related to the subscriber's network service policy and billing, and provides subscribers with IP connectivity and network services. To this end, CSN 400 includes AAA (Authentication-Authorization-Accounting) 410 that provides control and information based on user authentication, authorization verification and charging.

The AAA 410 has the subscriber's service subscription information, and provides the subscriber's service subscription information and QoS information of the service to be provided to the ASN-GW 300 when the subscriber wants to access the mobile Internet network. Here, the AAA 410 may provide, for example, two types of QoS information according to the service class of the subscriber.

The first type is for low class of service, and can provide QoS according to the BE service flow when the subscriber is subscribed to a general data service.

The second type is for a high class of service. When a subscriber is subscribed to a general data service and a multimedia service, QoS according to BE service flow and UGS service flow may be applied when initial subscriber authentication of a subscriber connected to ASN-GW. Can provide.

In addition, in the charging service support according to the VoIP service execution, the AAA 410 supports the charging service based on time and the data transmission amount when the dynamic service flow is executed. At this time, a procedure for charging is performed between the AAA 410 and the ASN-GW. The charging start procedure is performed before the dynamic service flow is generated. If the dynamic service flow is deleted, the charging stop procedure is performed to use the VoIP service. Support billing services.

The ASN-GW 300 performs an SIP Snooper 310 analyzing the detected SIP signaling message and an Anchor SFA 320 generating a service flow according to a call request from the terminal 100 and the VoIP CSN 500. It includes. In addition, the ASN-GW 300 may provide a command line interface (CLI) for enabling or disabling the SIP snooping function.

The dynamic QoS generation procedure initiated at the terminal 100 and the execution of SIP snooping performed at the ASN-GW 300 may be simultaneously performed. Here, if SIP snooping is performed, the serving SFA may reject at least one of the R6 Path-Reg-Req message from the base station 200 or the R4 Path-Reg-Req message from the serving ASN-GW.

When the ASN-GW 300 makes a VoIP call request from the terminal 100 to the terminal of the Peer 600, or when a VoIP call request is made from the Peer terminal to a terminal in the portable Internet network, the terminal 100 and the Peer terminal Detect and analyze the SIP signaling message between 600.

The ASN-GW 300 performs QoS assignment according to the UGS service class based on the detected SIP signaling message. Here, the allocation of QoS is performed in the SFA 320. In addition, the VoIP data and the SIP signaling message from the terminal 100 connected to the base station 200 in the uplink (UL) are transmitted to the VoIP CSN 500 (VoIP Core Service Network), and the VoIP in the downlink DL The VoIP data and the SIP signaling message from the CSN 500 are transmitted to the terminal 100 through the base station 200.

In addition, the ASN-GW (specifically, SFA) uses the subscriber information received from the AAA 410 of the CSN 400 and the QoS information of the service provided to the subscriber to provide QoS according to the class of service to be provided to the subscriber. Can be assigned.

As an example, in order to support the setting of dynamic QoS according to the VoIP service request from the terminal 100, the ASN-GW 300 and the base station 200 interwork with each other to add a dynamic service using a DSA-Req message and R6 Path-Reg-Req supports the data path setting procedure.

In this case, when the terminal 100 does not perform the handover, the Anchor ASN-GW that performed the initial network entry may be the same as the serving ASN-GW currently providing the service. The Anchor Data Path Function (DPF), Anchor SFA 320, Serving DPF and Serving SFA may be located together in the ASN-GW 300.

In addition, the ASN-GW 300 transmits to the subscriber terminal 100 based on the subscriber information received from the AAA 410 of the CSN 400 and the service class information provided to the subscriber, that is, the QoS class of the subscriber. Alternatively, QoS may be allocated according to a service flow of data to be received from the terminal 100. The ASN-GW 300 may provide the PHS classification information to the base station through the PHS 210.

In order to provide the VoIP service to the terminal 100, the ASN-GW 300 establishes and releases a SIP session between the terminal 100 and the peer terminal using QoS information and classification information included in the SIP signaling message. Support. The ASN-GW 300 may provide a function of detecting a SIP signaling message from the soft switch 510 and the terminal 100 of the VoIP CSN 500 and a function of analyzing the detected SIP signaling message.

Here, the detection of the SIP signaling message is performed in the data plane DP, and the analysis of the SIP signaling message is performed in the SIP snooper 310. The SIP snooper 310 transmits the information obtained through the analysis of the detected SIP signaling message to the terminal 100 in the downlink (DL) and to the VoIP CSN 500 in the uplink (UL).

Specifically, when the SIP signaling message is transmitted from the terminal 100 to the ASN-GW 300, the SIP signaling message is detected in the data plane of the ASN-GW 300, and the data plane is detected SIP signaling. Send a message to the SIP snooper 310.

In addition, when the SIP signaling message is received from the soft switch 510 to the ASN-GW 300, classification information is detected through the SIP signaling message in the data plane of the ASN-GW, and the detected SIP signaling message and By sending the classification information to the SIP Snooper 310, detection of the SIP signaling message is made.

The SIP snooper 310 analyzes the SIP signaling message to obtain detailed QoS information and classification information through multimedia session parameters defined in the Session Description Protocol (SDP) of the SIP signaling message. Thereafter, the obtained detailed QoS information and classification information are transmitted to the Anchor SFA 320 of the ASN-GW 300. Here, detailed QoS information and classification information are transmitted to Anchor SFA 320 to determine the grant of dynamic QoS request for UGS.

Once the QoS information and the classification information are obtained, the Anchor SFA 320 allocates a service flow, that is, a BE or UGS service class according to general data or VoIP data requested for transmission. Then, data is transmitted through the assigned BE or UGS service class.

With reference to FIG. 2, QoS allocation and data transmission of a BE or UGS service class will be described.

VoIP traffic must be distinguished from other traffic (general data transfer) to provide Point to Point Service Flow (PPSF) over UGS. To this end, in the present invention, when a data packet of VoIP traffic is received by the ASN-GW, the ASN-GW classifies the received data packet. Here, the VoIP traffic may be made through UGS or initial service flow BE.

After a call request from the peer terminal connected to the VoIP CSN 500, if traffic is transmitted in the network, the ASN-GW 300 receives the SIP signaling message from the soft switch 510, and the SIP snooper Analyze the SIP signaling message received via 310.

In case of downlink, the source IP address, sour port number, destination IP address, and protocol type are identified from the classification information through analysis of SIP signaling message, and the service class of the traffic (voice & data) to be transmitted from QoS information. do.

In the uplink, the destination IP address and protocol type are identified from the classification information through analysis of the SIP signaling message, and the service class of the traffic (Voice & Data) to be transmitted from the QoS information.

The Anchor SFA 320 of the ASN-GW 300 classifies the received traffic (Voice & Data) based on a preset packet classification table, and generates a service flow according to a classification result. For example, if the received traffic is VoIP, it may be classified as UGS, and if it is general data, it may be classified as BE.

Subsequently, a GRE key is generated between the ASN-GW 300 and the base station 200 using a classification result, and a GRE tunnel is generated using the GRE key. Thereafter, traffic is transmitted from the ASN-GW 300 to the base station 200 using the generated GRE tunnel.

Thereafter, the base station 200 transmits the traffic from the ASN-GW 300 to the terminal 100 by using a radio resource, and the terminal 100 receives the traffic from the base station 200 through the modem 130. do. The received traffic is transmitted to the VoIP client 120 through the API 110 to provide a VoIP service.

Here, the dynamic QoS request for the UGS may be made based on the subscriber class information from the AAA 410, where the Anchor SFA 320 may be located together with the Anchor Authenticator.

In this case, the SIP snooper 310 may determine whether to perform snooping of the SIP signaling message in order to maintain an identification list of the terminal for the anchor data path function (DPF). In this case, the Serving Service Flow Administrator (SFA), the Anchor DPF, and the SIP Snooper 310 may be located together in the ASN-GW 300, and the current DPF may be required when performing snooping of the SIP signaling message. Becomes Anchor DPF. Here, if the identification information of the terminal is not included in the identification list of the anchor DPF, it may be discarded without using the SIP signaling message.

Based on the subscriber information from the AAA 410, the Anchor SFA 320 may generate a service flow according to a BE service class for general data transmission and a dynamic service flow according to a UGS service class for providing VoIP service. . Here, the generation of the dynamic service flow may be classified into QoS allocation according to the dynamic service request from the terminal 100 and QoS allocation according to the dynamic service request from the network.

In addition, the Anchor SFA 320 may be allowed to the subscriber using information about the QoS grade of the service provided to the subscriber from the AAA 410 and the SIP signaling message snooped at the SIP snooper 310. The following two QoS, that is, QoS according to the service request from the network or QoS according to the service request from the terminal 100 may be determined.

Specifically, the Anchor SFA 320 recognizes the VoIP call request from the terminal 100 or the network through the SIP information from the SIP snooper 310, and the subscriber uses the subscriber information from the AAA 410 to enable the subscriber to make a VoIP call. Any subscriber who can use the service will allow the creation of dynamic QoS for the UGS. In this way, the VoIP service can be provided to subscribers.

The Anchor SFA 320 may perform both QoS assignment according to the dynamic service request from the terminal 100 and the network. That is, the QoS allocation according to the VoIP service request from the terminal 100 and the QoS allocation according to the VoIP service request from the network destined for the terminal 100 may be performed.

On the other hand, when the R4 handover is made, the RR-Req message according to the R4 handover may be received from the Serving SFA of the target ASN-GW to the Anchor SFA 320 (Service Flow Administrator) of the ASN-GW 300. When the Anchor SFA 320 of the ASN-GW 300 receives the RR-Req message according to the R4 handover, the Anchor SFA 320 may dynamically adjust for the UGS based on the subscriber's class information from the AAA 410. It is possible to determine whether to allow the QoS request.

For example, the initial dynamic QoS setting of the terminal, which may be provided through the ASN-GW 300, may include a dynamic service addition request (DSA-REQ) message as defined in the IEEE 802.16e / mobile WiMAX standard. ) To the base station 200. Then, the base station 200 transmits a Path-Reg-Req message to the serving ASN-GW / Anchor ASN-GW through the R6 interface. The serving SFA of the Anchor ASN-GW sends a Path-Reg-Req message to the Anchor ASN-GW via the R4 interface. Here, when the serving ASN-GW and the Anchor ASN-GW are different because the R4 handover is performed, the serving ASN-GW transmits a Path-Reg-Req message to the Anchor ASN-GW through the R4 interface.

In another embodiment of the present invention, an initial dynamic QoS setting of a network that can be provided through the ASN-GW 300 may be configured such that the Anchor SFA of the ASN-GW 300 is RR from the serving SFA of the Anchor ASN-GW via the R4 interface. This can be done by receiving a Req message. Here, the setting of the dynamic QoS may be made based on the classification information and the QoS information from the SIP signaling message as described above, and based on the subscriber class information from the AAA 410 regardless of whether the dynamic QoS request is allowed or not. It may be done.

The VoIP CSN 500 transmits data including VoIP to a terminal in another network, for example, Peer 600, and transmits VoIP data from the Peer 600 terminal to the terminal 100. 600).

The soft switch 510 of the VoIP CSN 500 supports call session establishment by controlling equipment such as a gateway in a broadband communication network such as a portable Internet network, and supports the SGW 540 and the MGW 530 and the ASN-GW 300. ) To interface with each other. The soft switch 510 performs call processing according to a call request from the terminal 100 connected to the portable Internet network and a VoIP call request from the peer terminal using the SIP signaling message.

The soft switch 510 uses the SIP signaling message to operate the MGW 530, SGW 540, MRCF 550 of the VoIP CSN 500, and the SIP Snooper 310 of the ASN-GW 300. You can control and check the call session. At this time, the soft switch 510 queries the call session state set to the MGW 530, the SGW 540, the MRCF 550, and the SIP snooper 310, and uses the response to the call session state to determine the call session. Perform audit function for.

The SGW 540 is a network element that converts and transmits call signaling between the PST / PLMN 600 and the portable Internet network. The SGW 540 converts a call request from a Peer terminal into a protocol message and transmits it to the soft switch 510. In addition, the call request received from the terminal 100 via the soft switch 510 is converted into a protocol message and transmitted to the peer terminal.

The MGW 530 is a device for converting the format of the transmitted traffic. For example, the MGW 530 converts time division multiplexing (TDM) traffic of a circuit network into an asynchronous transmission (ATM) or IP traffic of a packet network in a packet voice transmission network. To perform. At this time, for call control, the soft switch and the control protocol (eg, MGCP, Megaco) are interworked to determine the path of data traffic. Information transfer between gateways may use a real-time transport protocol (RTP) in the case of IP.

The MGW 530 performs conversion of VoIP data and general voice data. That is, the MGW 530 supports the performance and transmission of the mutual conversion between the data received from the Peer 600 terminal and the data received in the portable Internet network. The MRCF (550, Media Resource Control Function) identifies and controls media resources according to data transmission in the network.

As shown in FIG. 1, the portable Internet system including the above-described configuration performs call processing according to a call request from the terminal 100 and a call request from a peer terminal. QoS triggering of the requested service and setup and release of service classes (BE, UGS) may be performed. Through this, the VoIP service can be supported in the portable Internet system including the soft switch.

3 to 12, specific embodiments of the VoIP service supporting method of the present invention will be described.

The VoIP service supporting method of the present invention can support both a VoIP service request from a terminal and a VoIP service request from a peer terminal. Hereinafter, as an example, a method of processing call signaling and a QoS allocation performed in the terminal and the network when the terminal requests the VoIP service will be described. In the following description, description of general matters among the procedures performed in the portable Internet system will be omitted.

The VoIP call request initiated by the terminal (specifically the VoIP client) is made using a SIP signaling message, and the SIP signaling message can be transmitted and received through a BE connection. Therefore, after the terminal is entered into the network, it creates a BE connection, and performs the SIP registration procedure (registration procedure) between the terminal and the soft switch through the generated BE connection to start the VoIP service.

3 is a diagram illustrating a SIP session establishment method and QoS allocation method for VoIP service.

Referring to FIG. 3, since the terminal needs to enter the network in order to receive the VoIP service from the network, the terminal accesses the network by performing a network entry procedure with the base station and the ASN-GW ( S100).

After the terminal accesses the network, the terminal may request a VoIP service from the network. When the VoIP call request is made from the terminal, the ASN-GW may receive subscriber information from the AAA of the CSN to obtain a QoS policy.

In addition, the network may request the VoIP service according to the call request of the peer terminal. Here, the request of the VoIP service according to the call request of the peer terminal may be made regardless of whether the terminal to which the peer terminal wants to access is connected to the network, and from the peer terminal before the terminal performs the network entry or at the time of performing the network entry. When a VoIP call request is made to the network, a QoS policy may be obtained by receiving subscriber information from the AAA of the CSN in the network entry procedure of S100.

After the terminal accesses the network, an initial service flow (ISF), that is, BE service flow, for establishing a BE connection so that data can be transmitted and received between the terminal and the ASN-GW through an Internet Protocol (IP). The assignment procedure is performed (S102).

Subsequently, an IP acquisition procedure is performed between the UE and the ASN-GW through the BE connection (S104).

Subsequently, the charging initiation procedure according to the BE service flow is performed between the ASN-GW and the AAA (S106).

Subsequently, a soft switch discovery procedure is performed between the UE and the ASN-GW (S108).

Subsequently, a SIP session is established by transmitting and receiving a SIP signaling message through a BE connection between the UE and the soft switch of the VoIP CSN via the ASN-GW to establish a SIP session (S110).

Subsequently, the VoIP service flow, i.e., the creation of the UGS connection, includes a dynamic QoS allocation procedure in the terminal and a dynamic QoS allocation procedure in the network, where the API of the terminal converts a SIP signaling message between the VoIP client and the modem. And relaying.

The terminal (specifically, the API) performs a SIP session establishment and QoS allocation procedure for the VoIP service using the SIP signaling message transmitted to the network (S112). At this time, the conversion and relaying of the SIP signaling message performed in the AIP may be equally applied to the SIP signaling message in the uplink (UL) and the downlink (DL).

In addition, the ASN-GW performs a SIP session establishment and QoS allocation procedure for VoIP service by snooping and analyzing a SIP signaling message received from the terminal (S114).

In detail, the SIP snooper snoops and analyzes a SIP signaling message transmitted and received in a network, and provides information obtained through the analysis to the Anchor SFA. The Anchor SFA generates a service flow based on SIP signaling information provided by the SIP snoop, subscriber information received from the AAA, and QoS information.

In this case, the analysis and relaying procedure of the SIP signaling message performed in the SIP snoop and the generation procedure of QoS allocation and service flow performed in the Anchor SFA may be similarly applied in the downlink (DL) and the uplink (UL). .

Through the S112 and 114, the SIP session is established and the UGS connection for the VoIP service is generated by assigning QoS for the VoIP service (S116).

Subsequently, when the UGS connection for the VoIP service is generated, charging and receiving according to the use of the UGS connection is started by transmitting and receiving the subscriber information and the QoS information according to the service class between the ASN-GW and the AAA (S118).

Subsequently, the VoIP service is provided through the transmission and reception of VoIP data between the terminal and the peer terminal using the allocated UGS connection (S120).

Meanwhile, in the portable Internet system, R4 handover may be performed due to the movement of the terminal. In addition, the terminal may be in an idle mode to reduce radio resources and power consumption (S128).

When a VoIP call request is made from the network, the VoIP call request may be made regardless of the state of the terminal (network entry, handover, idle mode). Accordingly, in the state of S128, a SIP signaling message (specifically, a SIP INVITE message) may be received from the soft switch to the ASN-GW. In this case, the ASN-GW buffers a SIP INVITE message received from the soft switch (S130).

Here, the SIP request message is buffered until the terminal reenters the network, that is, performs a network reentry procedure, until the handover is completed or the terminal changes from the idle mode to the active mode.

Subsequently, when the call using the VoIP service is terminated between the terminal of the portable Internet network and the peer terminal, a procedure of releasing a SIP session generated between the terminal and the network is performed (S122). Together with S122, the QoS assignment of the UGS connection that has been created for the VoIP service is released (S124).

Subsequently, when the UGS connection for the VoIP service is released, the subscriber information and the QoS information according to the service class are transmitted / received between the ASN-GW and the AAA to terminate the charging according to the use of the UGS connection (S126). Through the above-described S100 to S126 it is possible to support the VoIP service to the terminal in the portable Internet system.

Hereinafter, an embodiment of the SIP session establishment method and the QoS allocation method in S110 to S116 of FIG. 3 will be described with reference to FIGS. 4 to 9, and with reference to FIGS. 10 to 12, FIG. 3. A specific embodiment of the SIP session release method and the QoS allocation release method in S122 and S124 will be described. 6 to 9 include a case where the serving ASN-GW and the Anchor ASN-GW are different from each other by the UE performing handover.

4 and 5, when the VoIP service request is made from the subscriber, the terminal transmits a SIP INVITE message to the soft switch for the VoIP call request (S200). Here, the SIP INVITE message is transmitted from the VoIP client to the API in the terminal, and the AIP converts the VoIP call request from the VoIP client to be recognized by the terminal and transmits it to the modem. The modem transmits the SIP INVITE message to the base station of the network. In the network, the SIP INVITE message is transmitted from the base station to the soft switch of the VoIP CSN via the ASN-GW. In this case, the SIP INVITE message includes URL, phone number, IP address, port number, MSTR, MRTR, and Codec information to define information of a receiver to be connected and information of data to be transmitted.

Here, the VoIP client transmits the SIP INVITE message through the URL, phone number, IP address, port number, MSTR, MRTR, and Codec included in the SIP INVITE message. Obtaining information (classification information), and stores the obtained information (S202).

In addition, the SIP snoop of the ASN-GW snoops the SIP INVITE message transmitted in the network as shown in FIG. 5, and then analyzes the SIP INVITE message to identify the terminal's identification information, QoS information (QoS information), and the like. After classification information is detected, it is stored (S360).

Subsequently, the soft switch transmits the SIP INVITE message received from the terminal to the peer (S204).

Thereafter, the peer receiving the SIP INVITE message transmits a 183 session progress message to the soft switch (S206). Subsequently, the 183 session progress message is transmitted from the soft switch to the terminal via the ASN-GW and the base station (S208).

Subsequently, the terminal receiving the 183 session progress message transmits a Provisional Response ACKnowledgement (PRACK) message to the soft switch (S210). The soft switch transmits the received PRACK message to the peer (S212). Here, the PRACK message is used to give a confidence in a session not yet established before the transmission and reception of the SIP 200 OK message.

Subsequently, the peer transmits the SIP OK message including the IP address and the port number to the soft switch (S214). The soft switch receiving the SIP OK message transmits the SIP OK message to the terminal (S216).

Here, the SIP snoop of the ASN-GW snoops a SIP OK message transmitted in the network, as shown in FIG. 5, and then detects local information having an identifier of the matched terminal. Thereafter, the QoS information and the classification information of the destination are added to the local information and then stored together with the information in the S360 (S370).

In addition, the terminal receives the SIP OK message received from the peer via the soft switch, and then detects the SIP OK message received from the VoIP client. Thereafter, the destination QoS information and classification information are obtained from the SIP OK message, and then stored together with the information in S202 (S218).

Subsequently, after the terminal finishes updating information necessary for establishing a SIP session through S202 and S218, the terminal transmits the updated information to the soft switch by including the updated information in the UPDATE message (S220). The soft switch transmits the UPDATA message received from the terminal to the peer (S222).

Subsequently, the peer transmits the UPDATA OK message to the soft switch in response to the received UPDATA message (S224). The soft switch transmits the SIP 200 OK message to the terminal (S226). When the terminal receives the SIP 200 OK message of S226, a SIP session for VoIP service is established.

Subsequently, when the SIP session is established through S200 to S226, ringing is performed in the peer, and the peer transmits a 180 ringing message to the soft switch, thereby informing that ringing (call connection) is performed (S228). The soft switch transmits a 180 ringing message to the terminal and informs that ringing is performed in the peer (S230). When the terminal receives the 180 ringing message, the subscriber of the terminal can receive a ring back tone.

Subsequently, when the peer requests the call to start, a SIP 200 OK message is transmitted from the soft switch to the terminal to inform the terminal that VoIP service (call) with the Peer terminal is possible (S232).

Subsequently, the terminal receiving the SIP 200 OK message in S232 transmits the SIP ACK message to the soft switch in response (S234). The soft switch transmits the received SIP 200 OK message to the peer terminal to inform that the connection for the VoIP service is possible between the terminal and the peer terminal (S236).

After the SIP session is established between the terminal and the peer terminal with the soft switch in between, QoS assignment must be made to send and receive VoIP data. Here, QoS allocation may be performed by a request from the terminal, as shown in Figure 4 (S238). In addition, as shown in Figure 5, it may be performed by a request from the ASN-GW of the network (S380).

In the QoS assignment by the request of the terminal of S238, the VoIP client of the terminal requests the QoS allocation for the VoIP service through the API.

Here, the API converts the QoS allocation request from the VoIP client so that the terminal can receive it, and relays the QoS allocation request to the modem. The modem performs a QoS assignment procedure for generating a UGS service class with the base station based on the QoS assignment received from the VoIP client via the API (S240).

Here, the QoS allocation request is made by including a dynamic service addition request, and the QoS allocation request message includes a URL, phone number, source IP address, source port number, MSTR, MRTR, updated between the terminal and the peer terminal in S202 and S220. Codec, destination IP address, destination port number information is included.

Subsequently, when QoS is allocated according to the request of the terminal, a UGS connection for the VoIP service is generated, and the VoIP service is provided through the transmission and reception of VoIP data between the terminal and the terminal connected to the peer through the generated UGS connection (S120). ).

With reference to FIGS. 6 and 7, a method for allocating QoS by a request of a terminal will be described in detail. In FIG. 6, it is assumed that available resources that can be allocated to a service flow requested by a terminal through a connection admission control (CAC) are identified, and QoS allocation is approved in an SFA. On the other hand, in FIG. 7, it is assumed that there are not enough resources available or the QoS assignment is rejected in the SFA. 6 and 7 include an R4 handover.

First, referring to FIG. 6, after a SIP session is established, dynamic service addition should be performed in order for a terminal to receive a VoIP service. To this end, the terminal transmits a dynamic service addition request (DSA-REQ) message to the base station (S140).

Subsequently, the base station receiving the DSA-REQ message from the terminal applies a connection admission control (CAC) to check whether the VoIP service can be provided to the terminal through the UGS (S142). Here, the CAC is for checking whether there is sufficient resource to allocate the UGS according to the dynamic service addition request requested from the terminal at the base station.

At this time, applying the CAC is requested from the terminal on the basis of the information included in the QoS assignment request message received from the terminal in the base station (information stored in the S218), for example, QoS information parameters (QoS information parameters). This is done by checking if there are enough radio resources (available resources) to provide dynamic services. The base station may determine whether to add the dynamic service and the QoS allocation request from the terminal by checking available resources through the application of CAC.

Subsequently, the base station transmits an R6 Path-Reg-Req message including the QoS value and service class information to be allocated to the terminal to the serving ASN-GW and requests the establishment of a data path between the base station and the serving ASN-GW (S144). .

Thereafter, the Anchor SFA determines whether to allocate dynamic QoS based on the subscriber information received from the AAA and the service flow information (SF information) received from the serving SFA of the Anchor ASN-GW. Serving SFA and Anchor SFA are the same.

If an ASN-GW serving the current service is different from the Anchor ASN-GW that performed the initial network entry by performing an R4 handover, the serving ASN-GW includes an R4 Path-Reg including QoS values and service class (UGS) information. Send a Req message to the Anchor ASN-GW. A request for data path establishment between the serving ASN-GW and the Anchor ASN-GW is performed through the R4 Path-Reg-Req message (S146).

Subsequently, the Anchor ASN-GW transmits the service flow information including the service flow identifier (SFID) in the RR-Req message to the Anchor SFA in order to generate a route of the VoIP data (S148).

Subsequently, the Anchor SFA determines whether to allocate dynamic QoS based on the subscriber information received from the AAA and the service flow information (SF information) received from the serving SFA of the Anchor ASN-GW. Here, if the dynamic service requested by the terminal corresponds to the subscriber information received from the AAA, the Anchor SFA allows the dynamic QoS allocation (S150).

Subsequently, in response to the RR-Req message received at S148, the Anchor SFA transmits an RR-Rsp message including service flow information (SF information) to the Anchor ASN-GW (S152). Subsequently, the Anchor ASN-GW transmits an R4 Path-Reg-Rsp message to the serving ASN-GW in response to the R4 Path-Reg-Req message received in S146 (S154).

Subsequently, the serving ASN-GW transmits an R6 Path-Reg-Rsp message to the base station in response to the R6 Path-Reg-Req message received in S144 (S156). At this time, the R6 Path-Reg-Rsp message includes PHS and service flow (eg, UGS) generation information.

Subsequently, the base station applies the CAC to check whether there are enough resources to generate UGS. As a result of checking through the CAC, if there is sufficient resources to generate the UGS, in step S140, the dynamic service requested from the terminal is approved (S158). Subsequently, the base station transmits a DSA-RSP / ACK message to the terminal in response to the DSA-REQ message (S160).

Subsequently, the base station includes the UGS generation information in the R6 Path-Reg-Ack message and transmits the information to the serving ASN-GW (S162). Subsequently, the serving ASN-GW receiving the R6 Paht-Reg-Ack message includes the UGS generation information in the R4 Paht-Reg-Ack message and transmits it to the Anchor ASN-GW (S164). Subsequently, the Anchor ASN-GW transmits an RR-Ack message to the Anchor SFA in response to the RR-Rsp message received in S152 (S166).

Referring to FIG. 7, after S148, if the dynamic service requested by the terminal does not correspond to subscriber information received from the AAA, the Anchor SFA rejects the dynamic QoS allocation request (S170).

Subsequently, the Anchor SFA includes the reject code of the dynamic QoS allocation determined in S170 in the RR-Rsp message and transmits the same to the Anchor ASN-GW (S172).

Subsequently, the Anchor ASN-GW receiving the RR-Rsp message including the reject code transmits an R4 Path-Reg-Rsp message including the reject code to the serving ASN-GW. (S174). Subsequently, the serving ASN-GW transmits an R6 Path-Reg-Rsp message including the reject code to the base station (S176).

As described above, when the Anchor SFA rejects the dynamic QoS allocation request from the terminal, the base station may reject the addition of the dynamic service requested from the terminal in S130 without checking whether there is sufficient resource for UGS allocation.

Even when Anchor SFA approves the dynamic QoS allocation request, the base station verifies that there are available resources available for UGS allocation through CAC application, and if the available resources do not exist, the rejection of the dynamic QoS allocation in the DSA-RSP message It transmits to the terminal including the (reject code) (S178). Through the above-described S130 to S178 may be the approval / rejection according to the dynamic service addition request from the terminal.

On the other hand, the QoS assignment request may be made in the network as well as the terminal of the portable Internet network.

Referring to Figures 8 and 9, it will be described in detail with respect to the QoS allocation method by the request of the network. In FIG. 8, it is assumed that available resources that can be allocated to a service flow requested by a terminal through a connection admission control (CAC) are identified, and QoS allocation is approved in an SFA. Meanwhile, in FIG. 9, it is assumed that there are not enough resources available, or the QoS assignment is rejected in the SFA. 8 and 9 include an R4 handover.

First, referring to FIG. 8, after a SIP session is established, dynamic service addition should be performed to provide a VoIP service to a terminal.

To this end, the serving SFA includes service flow information including a service flow identifier (SFID) in an RR-Req message and transmits it to the Anchor SFA in which the terminal performs an initial network entry (S300).

Subsequently, the Anchor SFA determines whether to allocate dynamic QoS based on the subscriber information received from the AAA and the service flow information (SF information) received from the serving SFA of the Anchor ASN-GW. Here, if the dynamic service requested from the terminal corresponds to subscriber information received from the AAA, the Anchor SFA allows dynamic QoS allocation (S302).

Subsequently, in response to the RR-Req message received at S300, the Anchor SFA transmits an RR-Rsp message including service flow information (SF information) to the Anchor ASN-GW (S304).

Subsequently, if the Anchor ASN-GW performing the initial network entry and the ASN-GW serving the current service are different from each other, the Anchor ASN-GW includes an R4 Path-Reg including QoS values and service class (UGS) information. Send a Req message to the serving ASN-GW. A request for data path establishment between the serving ASN-GW and the Anchor ASN-GW is performed through the R4 Path-Reg-Req message (S306).

Subsequently, the serving ASN-GW requests an establishment of a data path between the base station and the serving ASN-GW by transmitting an R6 Path-Reg-Req message including a QoS value and service class information to be allocated to the terminal (S308). .

Subsequently, the base station applies the CAC to check whether the VoIP service can be provided to the terminal through the UGS, and checks whether there are enough resources to allocate the UGS (S310).

Subsequently, as a result of the CAC checking at the base station, if there are enough resources to generate the UGS, the mobile station approves the addition of the dynamic service and transmits a dynamic service addition request (DSA-REQ) message to the terminal (S312).

Subsequently, the terminal transmits a DSA-RSP / ACK message to the base station in response to the DSA-REQ message of S312 (S314).

Subsequently, the base station transmits the R6 Path-Reg-Rsp message to the serving ASN-GW in response to the R6 Path-Reg-Req message received in S308 (S316). At this time, the R6 Path-Reg-Rsp message includes PHS and service flow (eg, UGS) generation information.

Subsequently, the serving ASN-GW transmits an R4 Path-Reg-Rsp message to the Anchor ASN-GW in response to the R4 Path-Reg-Req message received in S306 (S318). Subsequently, the Anchor ASN-GW transmits an R4 Path-Reg-Ack message to the serving ASN-GW in response to the R4 Path-Reg-Rsp message (S320).

Subsequently, the serving ASN-GW includes the UGS generation information in the R6 Paht-Reg-Ack message and transmits the information to the base station (S322). In addition, the Anchor ASN-GW transmits an RR-Ack message to the Anchor SFA in response to the RR-Rsp message received in S304 (S324).

Referring to FIG. 9, even when dynamic QoS allocation is approved in the Anchor SFA, as a result of confirming that there are available resources that can be allocated to UGS through application of CAC, if the available resources do not exist, the dynamic QoS allocation may be rejected. It may be (S336).

In this case, the base station includes a reject code of the dynamic QoS allocation determined in S336 in the R6 Path-Reg-Rsp message and transmits it to the serving ASN-GW (S338).

Subsequently, the serving ASN-GW receiving the R6 Path-Reg-Rsp message containing the reject code of the QoS assignment includes the rejection of the QoS assignment in the R4 Path-Reg-Rsp message to the Anchor ASN-GW. It transmits to (S340).

Subsequently, when receiving the R4 Path-Reg-Rsp message including the rejection of the QoS assignment, the Anchor ASN-GW includes the rejection of the QoS assignment in the RR-Ack message and transmits it to the Anchor SFA (S346).

As described above, even if the dynamic QoS allocation request is approved in the Anchor SFA, as a result of confirming that there are available resources that can be allocated to UGS through the application of CAC, if the available resources do not exist, the dynamic service addition may be rejected.

Through the above-described S300 to S346 may be approved / rejected the dynamic service addition request according to the request of the network.

After the VoIP service is provided between the terminal and the peer using the SIP session and the UGS service class through the above-described procedures of FIGS. 3 to 9, when the conversation is completed, the SIP session and the UGS service class generated between the terminal and the soft switch are completed. Will be released.

Hereinafter, a SIP session release method and a QoS release method will be described with reference to FIGS. 10 to 12.

The release of the SIP session may be made by a request from the terminal and a request from the network. In FIG. 10, an SIP session release request is made by the terminal.

When the conversation is completed between the terminal and the peer terminal using VoIP, the terminal transmits a SIP BYE message to the soft switch to release the SIP session (S400). In this case, the SIP BYE message includes information of URL, phone number, IP address, port number, MSTR, MRTR, and Codec that were applied when the SIP session was created.

Here, in the process of transmitting the SIP BYE message from the terminal to the soft switch, the VoIP client, the identification information, QoS information of the terminal through the URL, phone number, IP address, Port number, MSTR, MRTR, Codec included in the SIP BYE message (QoS information) and classification information (classification information) are obtained and the obtained information is stored (S402). In addition, the SIP snoop of the ASN-GW snoops the SIP BYE message transmitted in the network, and analyzes the SIP BYE message to detect identification information, QoS information, and classification information of the terminal. After that, it stores (S404).

Subsequently, the soft switch transmits the SIP BYE message received from the terminal to the peer (S406). Thereafter, the peer terminal receiving the SIP BYE message transmits a SIP OK message to the terminal via a soft switch in response to the release of the SIP session requested by the terminal (S408).

When the terminal receives the SIP OK message from the peer in response to the SIP BYE message, the SIP session created between the terminal and the peer terminal is released, and the terminal sends a SIP Ack message to the peer terminal to release the SIP session successfully. This informs (S410).

Subsequently, when the SIP session is released between the terminal and the peer terminal, the QoS allocated for the transmission and reception of VoIP data is released (S416). Here, the release of QoS may be performed by a request from the terminal (S412), and may be performed by a request from the ASN-GW of the network (S414).

Referring to FIG. 11, after the SIP session is released, the dynamic service generated for providing the VoIP service is deleted. To this end, the terminal transmits a dynamic service deletion request (DSD-REQ) message to the base station (S420).

Subsequently, the base station receiving the DSD-REQ message from the terminal generates an R6 Path-DeReg-Req message including the QoS value and service class information assigned to the terminal to the serving ASN-GW to generate the base station and the serving ASN-GW. A request is made to release the data path that has been used (S422).

Thereafter, the Anchor SFA releases the dynamic QoS allocated for the VoIP service based on the service flow information (SF information). When the R4 handover is not performed, the serving SFA and the Anchor SFA are the same. If an ASN-GW serving the current service is different from the Anchor ASN-GW that performed the initial network entry due to an R4 handover, the serving ASN-GW includes an R4 Path-DeReg- containing QoS value and service class (UGS) information. Send a Req message to the Anchor ASN-GW. The release of the data path generated between the serving ASN-GW and the Anchor ASN-GW is requested through the R4 Path-DeReg-Req message (S424).

Subsequently, the Anchor ASN-GW transmits the service flow information including the service flow identifier (SFID) in the RR-Req message to the Anchor SFA in order to release the path for the transmission of the VoIP data (S426). ).

Subsequently, the Anchor SFA releases the dynamic QoS allocated for the VoIP service based on the service flow information (SF information) (S428).

Subsequently, in response to the RR-Req message received at S426, the Anchor SFA transmits an RR-Rsp message including service flow information (SF information) to the Anchor ASN-GW (S430).

Subsequently, the Anchor ASN-GW transmits an R4 Path-DeReg-Rsp message to the serving ASN-GW in response to the R4 Path-DeReg-Req message received in S424 (S432).

Subsequently, the serving ASN-GW transmits an R6 Path-DeReg-Rsp message to the base station in response to the R6 Path-DeReg-Req message received in S422 (S434).

Subsequently, the base station receiving the R6 Path-DeReg-Rsp message from the serving ASN-GW releases QoS according to the VoIP service (S436). This releases the UGS that was created for the VoIP service.

Subsequently, the base station transmits a DSD-RSP / ACK message to the terminal in response to the DSD-REQ message to inform that the UGS generated for the VoIP service is released (S438).

Subsequently, the base station includes the UGS release information in the R6 Path-DeReg-Ack message and transmits the information to the serving ASN-GW (S440). Subsequently, the serving ASN-GW receiving the R6 Path-DeReg-Ack message includes the UGS generation information in the R4 Path-DeReg-Ack message and transmits it to the Anchor ASN-GW (S442).

Subsequently, the Anchor ASN-GW transmits an RR-Ack message to the Anchor SFA in response to the RR-Rsp message received at S430 to complete a release procedure of QoS allocated for VoIP (S444).

Meanwhile, the request for releasing QoS may be made in the network as well as the terminal.

Referring to FIG. 12, after the SIP session is released, the dynamic service generated for providing the VoIP service is deleted. To this end, the Anchor ASN-GW transmits an RR-Req message including the service flow information to the Anchor SFA (S470).

The Anchor SFA releases the QoS allocated for the VoIP service based on the service flow information included in the RR-Req message (S472). Thereafter, the Anchor SFA transmits an RR-Rsp message including QoS release information to the Anchor ASN-GW in response to the RR-Req message of S470 (S474).

Subsequently, the Anchor ASN-GW transmits an R4 Path-DeReg-Req message including a QoS value and service class (UGS) information to the serving ASN-GW (S476). The R4 Path-DeReg-Req message is used to request the release of the data path that was created between the serving ASN-GW and the Anchor ASN-GW.

Subsequently, the serving ASN-GW transmits an R6 Path-DeReg-Req message including the QoS value and service class information allocated to the terminal, to request the release of the data path established between the base station and the serving ASN-GW (S478). ).

Subsequently, the base station transmits a dynamic service deletion request (DSD-REQ) message to the terminal (S480). QoS is released according to the VoIP service (S482). This releases the UGS that was created for the VoIP service.

Subsequently, the terminal transmits a DSD-RSP / ACK message to the base station in response to the DSD-REQ message to inform that the release of the UGS generated for the VoIP service was successfully performed (S484).

Subsequently, the base station includes the UGS release information in the R6 Path-DeReg-Rsp message and transmits the information to the serving ASN-GW (S486). Subsequently, the serving ASN-GW receiving the R6 Path-DeReg-Rsp message includes the UGS generation information in the R4 Path-DeReg-Rsp message and transmits it to the Anchor ASN-GW (S488). In addition, the serving ASN-GW transmits an R6 Path-DeReg-Ack message to the base station in response to the received R6 Path-DeReg-Rsp message (S490).

Subsequently, the Anchor ASN-GW receiving the R4 Path-DeReg-Rsp message sends the RR-Ack message to the Anchor SFA in response to the RR-Rsp message received at S474 to complete the release procedure of the QoS allocated for VoIP. (S492).

The VoIP service supporting method according to the above-described embodiment of the present invention performs call processing based on a call request from a terminal and a VoIP service request from a peer terminal, and the QoS triggering and service class (BE) according to the requested VoIP service. UGS) can be turned on / off. This enables VoIP services in soft switch-based portable Internet networks.

The VoIP service supporting method according to the above-described embodiments of the present invention may be implemented in the form of program instructions that can be executed by various computer means and recorded in a computer-readable recording medium. In this case, the computer-readable recording medium may include program instructions, data files, data structures, and the like, alone or in combination. On the other hand, the program instructions recorded on the recording medium may be those specially designed and configured for the present invention or may be known and available to those skilled in the computer software.

The computer-readable recording medium includes a magnetic recording medium such as a magnetic medium such as a hard disk, a floppy disk and a magnetic tape, an optical medium such as a CD-ROM and a DVD, a magnetic disk such as a floppy disk, A magneto-optical media, and a hardware device specifically configured to store and execute program instructions such as ROM, RAM, flash memory, and the like. In addition, program instructions include not only machine code generated by a compiler, but also high-level language code that can be executed by a computer using an interpreter or the like. The hardware device described above may be configured to operate as one or more software modules to perform the operations of the present invention, and vice versa.

On the other hand, those skilled in the art will understand that the present invention described above may be implemented in other specific forms without changing the technical spirit or essential features.

1 is a diagram illustrating a portable Internet system supporting a VoIP service according to an embodiment of the present invention.

2 is a diagram illustrating a packet classification method in ASN-GW and a traffic transmission method using QoS allocation of a service class.

3 is a diagram illustrating a SIP session establishment method and QoS allocation method for VoIP service.

4 is a diagram illustrating a SIP session establishment and QoS allocation method in a terminal.

5 is a diagram illustrating a SIP session establishment and QoS allocation method in a network.

6 is a diagram illustrating a dynamic service adding method in response to a request from a terminal.

FIG. 7 is a diagram illustrating a method for rejecting a dynamic service addition request when a subscriber who requests a service in FIG. 6 is not a service object.

8 illustrates a method for adding a dynamic service in response to a request from a network.

FIG. 9 is a diagram illustrating a method for rejecting a dynamic service addition request when a subscriber who requests a service in FIG. 8 is not a service object.

10 is a diagram illustrating a SIP session release method.

11 is a diagram illustrating a method for releasing QoS at a terminal.

12 illustrates a method of releasing QoS in a network.

DESCRIPTION OF THE REFERENCE SYMBOLS

100: terminal (MS) 110: VoIP client

120: application programming interface

130: modem 200: base station (BS)

210: PHS (Personal Handy System) 300: ASN-GW

310: SIP Snooper 320: Anchor SFA

400: CSN 410: AAA

500: VoIP CSN 510: Soft Switch

520: Application Server (AS) 530: Media Gateway (MGW)

540: Signaling GateWay (SGW)

550: Media Resource Control Function (MRCF)

600: Public Switching Telephone Network (PSTN)

Claims (26)

In the soft switch-based wireless communication network, Delivering a SIP signaling message to establish a session for VoIP service between the terminal and the soft switch after a network entry of the terminal and a Session Initiation Protocol (SIP) registration procedure are performed; Transmitting VoIP data over the session; Snooping said SIP signaling message during said forwarding, and QoS triggering to allocate QoS resources for said session based on information extracted from said SIP signaling message. The method of claim 1, And if the terminal is in the idle mode, buffering a SIP signaling message transmitted to the terminal. The method of claim 1, The extracted information includes a terminal identification information, QoS information and classification information. The method of claim 3, wherein The QoS information includes a guaranteed data rate (Guaranteed data rate) and the maximum data rate (Maximum data rate). The method of claim 3, wherein The classification information includes a source IP, a source port number, a protocol, a destination IP, and a destination port number. The method of claim 3, wherein When the terminal performs the handover, the allocation of the QoS resources, VoIP service support method, characterized in that performed in the Anchor SFA (Service Flow Administrator) of ASN-GW (Access Service Network-GateWay). The method of claim 6, After the QoS resource allocation, further comprising initiating charging according to the use of the VoIP service. The method of claim 7, wherein when the transmission of the VoIP data through the session is completed, Releasing the session; And And releasing the allocated QoS resources. The method of claim 8, And after the release of the QoS resource, stopping the charging due to the use of the VoIP service. The method of claim 1, And the SIP signaling message is a SIP INVITE message or a SIP OK message. In the soft switch-based wireless communication network, Transmitting a SIP signaling message to establish a session for VoIP service between the terminal and the soft switch after the network entry of the terminal and the SIP registration procedure are performed; Transmitting VoIP data over the session after establishing the session; Snooping the SIP signaling message in the terminal and QoS triggering so that QoS resources for the session are allocated based on information extracted from the SIP signaling message. The method of claim 11, The extracted information includes a terminal identification information, QoS information and classification information. 13. The method of claim 12, The QoS information includes a guaranteed data rate (Guaranteed data rate) and the maximum data rate (Maximum data rate). 13. The method of claim 12, The classification information includes a source IP, a source port number, a protocol, a destination IP, and a destination port number. The method of claim 12, wherein the terminal And requesting QoS resource allocation based on identification information, QoS information, and classification information of the terminal. The method of claim 12, wherein when the transmission of the VoIP data through the session is completed, Releasing the session; And And releasing the allocated QoS resources. The method of claim 11, And the SIP signaling message is a SIP INVITE message or a SIP OK message. In the soft switch-based wireless communication network, An ASN-GW for transmitting a SIP signaling message and transmitting VoIP data over the session to establish a session for VoIP service between the terminal and the soft switch after the network entry of the terminal and the SIP registration procedure are performed; The ASN-GW includes a snooper that snoops the SIP signaling message during the delivery and QoS triggers to allocate QoS resources for the session based on information extracted from the SIP signaling message. VoIP service support device. The method of claim 18, The information extracted from the SIP signaling message includes identification information, QoS information, and classification information of the terminal. The ASN-GW further includes a service flow administrator (SFA) for allocating QoS resources based on identification information, QoS information, and classification information of the terminal. The method of claim 19, VoIP service supporting apparatus further comprising AAA (Authentication-Authorization-Accounting) for providing the ASN-GW with information for billing according to VoIP service usage. The method of claim 20, The AAA provides the VoIP service according to the service class of the terminal, the QoS information according to the BE service flow or QoS information according to the UGS service flow to the SFA, characterized in that the device. The method of claim 18, The snooper buffers a SIP signaling message transmitted to the terminal when the terminal is in the idle mode. The method of claim 22, And the snooper transmits the SIP signaling message to the terminal after the terminal changes from active mode to idle mode. A modem for transmitting and receiving a SIP signaling message in response to a VoIP service request with a soft switch-based wireless communication network; A VoIP client establishing a session for the VoIP service with the soft switch based on identification information, QoS information, and classification information of the terminal extracted from the SIP signaling message, and performing QoS triggering so that QoS resources for the session are allocated; And an application programming interface (API) for converting and relaying the SIP signaling message between the modem and the VoIP client. 25. The method of claim 24, The QoS information includes a guaranteed data rate and a maximum data rate, The classification information includes a source IP, a source port number, a protocol, a destination IP, and a destination port number. And the VoIP client transmits and receives VoIP data to and from the wireless communication network through the session based on the QoS information and classification information via the API and the modem. The method of claim 25, wherein the API In the uplink, converts the SIP signaling message from the VoIP client and sends it to the modem, In the downlink, the VoIP service support apparatus, characterized in that for converting the SIP signaling message received from the wireless communication network via the modem to transmit to the VoIP client.

Images