KR101483270B1 - apparatus and method of management service flow in portable internet system - Google Patents

Abstract

The present invention relates to a method for setting up at least one service flow (SF) to be used by at least one MS (mobile station) connected to a portable Internet network, a first signaling A first DSC (Dynamic Service Change) process for each MS to change the set SF to an admittable state; and a step of, when the called MS answers a call request, A third DSC process of changing the SF to a provisioning state when a call session between the calling MS and the called MS ends; The SF management method of the system is started to dynamically allocate and set the service flow to be used by the MS without implementing the expensive PCRF, .

Description

[0001] The present invention relates to a method and apparatus for managing a service flow of a portable Internet system,

The present invention relates to a service flow management method and apparatus for a portable Internet system, and more particularly, to a method and apparatus for managing a service flow in a portable Internet system, It is possible to dynamically allocate the service flow to the MS and to change the service flow to the prevision state even after the end of the call session to improve the management efficiency for the service flow or to change the QoS parameter value of the service flow to be used for the MS The present invention relates to a service flow management method and apparatus for a portable Internet system for accurately predicting VoIP service quality of a MS and granting QoS by granting a QoS profile based on the setting.

In 4G (4th Generation) communication system, which is a next generation communication system, researches for providing users with high-speed services with high quality of service (QoS) have been actively conducted.

Particularly, in the 4G communication system, a mobility is provided to a broadband wireless access (BWA) communication system such as a wireless local area network (LAN) system and a wireless metropolitan area network (MAN) (IEEE 802.16a / d) communication system and IEEE (Institute of Electrical and Electronics Engineers) IEEE (Institute of Electrical and Electronics Engineers) 802.16e communication system.

The IEEE 802.16a / d communication system and the IEEE 802.16e communication system use an Orthogonal Frequency Division Multiplexing (OFDM) / Orthogonal Frequency Division Multiplexing (OFDM) system to support a broadband transmission network on a physical channel of a wireless MAN system. (Orthogonal Frequency Division Multiple Access) system.

The IEEE 802.16a / d communication system is a system in which a subscriber station (SS) is currently fixed, that is, a system considering only mobility and a single cell structure, or an IEEE 802.16e communication system, It is a system considering mobility of SS in communication system. An SS having mobility is called a mobile station (MS).

In the IEEE 802.16e communication system, in order for a MS to access a base station (BS) and provide a service, the MS successfully performs a network entry process to enter a service area of the BS, (service flow) is set.

The service flow refers to a unidirectional flow of service data units (SDUs) transmitted over a connection guaranteeing a specific QoS. The service flow includes a provision QoS parameter set, There can be three types of QoS parameter sets: a parameter set (an admit QoS Paramemer Set), and an active QoS parameter set.

Here, the provision QoS parameter set is a 32-bit identifier assigned to a service flow by the BS, with a service flow identifier (SFID) being set by setting a service flow property.

The MS has logically several connections and service flows with the BS, and there are different connections corresponding to the respective service flows.

Therefore, the SFID and the connection identifier (CID) are used to distinguish different service flows and connections between the MS and the BS.

The admit QoS parameter set occupies a resource but is not actually active. The 16-bit CID is allocated from the BS. The active QoS parameter set is a real resource, and the active QoS parameter A set of QoS parameters that real data can be transmitted and received.

Meanwhile, when registration is completed through initialization, the MS performs a DSA process of transmitting and receiving a dynamic service addition (DSA) message between the MS and the BS. Since the DSA message includes a provisioning QoS parameter set, Lt; RTI ID = 0.0 > service flow.

In addition, when the MS changes the service parameter of the currently set service flow while it is transmitting / receiving traffic, the MS transmits a Dynamic Service Change (DSC) message to the BS to change the required service parameter Perform the DSC process. Meanwhile, when there is no traffic exchanged through the service flow and a transport CID (TCID) is returned, the DSC process is performed.

When terminating the connected service flow, the MS transmits a Dynamic Service Deletion (DSD) message to the BS to delete the corresponding service flow through the DSD process.

Meanwhile, a voice over IP (VoIP) service, which is a telephone service through an IP network, is a service that exchanges voice in packets in real time through a call session established through an IP network. To provide a voice call service similar to that of the telephone service of the present invention.

In the VoIP service, the call connection between the sender and the receiver is based on a call setup signaling protocol, for example, Session Initiation Protocol (SIP) and Session Description Protocol (SDP).

SIP defines a communication path between a sender and a receiver of SS7 (Signaling System Number 7) in the existing telephone network, that is, a call session establishment function, and SDP defines a characteristic of a packet exchanged through a set communication session do.

When a call session is established by SIP / SDP, the voice of the sender / receiver is sampled and included in an RTP (Real Time Protocol) packet and exchanged over UDP / IP through a call session.

In the case of providing a VoIP service through a portable Internet system such as the IEEE 802.16e communication system described above, QoS is supported in the wireless network between the MS and the BS using SIP / SDP in order to guarantee the QoS of the VoIP service .

For example, a network group (NWG Network Working Group) for defining IEEE 802.16 defines a Policy Charging Rules Function (PCRF) for managing service flows so that MSs can dynamically allocate QoS service flows supporting QoS whenever needed, Is recommended to be used.

However, since PCRF is an expensive device, implementation of PCRF in a portable Internet system is a real problem in terms of implementation cost.

On the other hand, if the mobile Internet system that does not implement the defined PCRF supports static QoS, it is necessary to activate all the service flows required by the MS at the beginning of the MS entry, so resource management for the service flows It is inefficient.

Hereinafter, a method of supporting static QoS in a portable Internet system will be briefly described.

The static QoS can set a preliminary QoS parameter set by a network management server (NMS) that manages the resources of the portable Internet system.

The set of preliminary QoS parameters may be forwarded to the MS via an authentication server (RADIUS server or an Authorization, Authentication and Accounting (AAA) server), or may be predefined by the management server to the ASN-GW (Access Service Network Gateway) or BS.

The authentication server transmits the service flow to be used by the MS as a service profile ID (service profile ID) or a service class name.

Meanwhile, the operator of the portable Internet system can define the initial state of the corresponding QoS parameter set, and the static QoS has the state of provision / admit / active (provision / admit / active) , And a combination of TLV ((Type, Length, Value) and the corresponding state can be transmitted as a QoS parameter set value.

There are two types of service flows. First, initial service flow (ISF) is the initial service flow for signaling before dynamic host configuration protocol (DHCP). If the initial service flow is not set, the MS can not perform a signaling procedure for establishing a call session through the portable Internet network, and thus leaves the portable Internet network.

The second is the Pre-Provision Service Flow (PPSF) used in the application. The PPSF is processed differently according to the CRR (Combined Resource Required) flag if the MS is not created at the time of initial connection or handoff. If the CRR flag is ON, the mobile terminal moves out of the portable Internet network. If the CRR flag is OFF, the service flow is released (Silently Release).

1 is a flowchart illustrating a flow of providing a VoIP service through a general portable Internet system.

1, the originating MS 10-1 and the terminating MS 10-2 are connected to base station systems 20-1 and 20-2, each of which is comprised of a base station (BS) and an access control router (ACR) -2), the initial service flow is set through the DSA process (S 10, S 11). At this time, the type of the initial service flow may be a type having a provisioning and an active QoS parameter set.

The source MS 10-1 and the destination MS 10-2 send a resource for a service flow for exchanging RTP packets for providing a VoIP service with a call admission control (CAC) and an Unsolicited Granted Service (UGS) .

When the initial service flow is established, the source MS 10-1 transmits SIP INVITE, which is call request signaling, to the destination MS 10-2 via the call server 30 (S12, S13).

When the SIP INVITE is received, the terminating MS 10-2 sends SIP 183, which is a response signaling indicating that it is a call request, to the calling MS 10-1 (S14, S15).

When the user hooks off the call request, the terminating MS 10-2 sends the SIP 200, which is acknowledgment signaling for the SIP INVITE, to the calling MS 10-1 (S16, S17 ).

The originating MS 10-1 transmits the SIP ACK, which is confirmation signaling, to the terminating MS 10-2 (S18, S19) and establishes a call session (S20) when the SIP 200 becomes available.

Then, the source MS 10-1 and the destination MS 10-2 exchange VoIP traffic, that is, RTP packets including voice information through a call session.

On the other hand, if the user of the source MS 10-1 selects the termination of the call (Hook on), the SIP BYE, which is the call termination signaling, is transmitted to the terminating MS 10-2 (S21, S22) When the SIP BTE is received, the terminating MS 10-2 sends the SIP ACK, which is the response signaling, to the originating MS 10-1 (S23, S24) and terminates the call session.

If there is no traffic exchanged through the service flow and the transport CID (TCID: Transport CID) is to be returned, the destination MS 10-2 changes the state of the service flow through the DSC process (S25 ).

Since the source MS 10-1 must terminate the service flow, the service flow is deleted through the DSD process (S26).

There is a problem in processing the PPSF to provide the VoIP service with the static QoS that does not implement the PCRF in the general portable Internet system. In other words, the PPSF is not active initially, but the only way to implement the PCRF is to implement the DSC process in the provisioned state.

Also, if some of the PPSFs are released in the state that the CRR flag is ODD, the static QoS must turn on the CRR flag to make the ISF and PPSF always active in order to recover the released service flow again. It is very inefficient.

On the other hand, even if the CCR flag is ON, if the resources of all the PPSFs can not be reserved when the MS enters the initial network, it can not enter the portable Internet network, and the call setup success rate is lowered.

The present invention proposes a service flow management method of a portable Internet system capable of dynamically managing a service flow for providing a VoIP service without implementing an expensive PCRF in a portable Internet system And an object thereof is to provide such a device.

According to an aspect of the present invention, there is provided a method for managing SF (Service Flow) of a portable Internet system, the method comprising: setting at least one SF to be used by at least one MS (mobile station) A first DSC for establishing a call session between the called MS and a called MS; a first DSC (Dynamic Service Change) process for each MS to change the set SF to an admittable state; A second DSC process of changing the SF changed to the admittable state to the active state in the first DSC process when the response is answered by the first DSC process; And a third DSC process for changing the state of the second DSC.
The step of setting the SF sets the SF to one of a provision state and an active state or more.
The first signaling process includes transmitting the SIP INVITE to the called MS by the calling MS, and transmitting the SIP 183 to the calling MS by the called MS.
Wherein the first DSC process comprises: generating service change information from a SIP message in which each MS is received; selecting an SF based on the QoS parameter of the preset SF and the service change information; And sending a DSC request message to the base station system to change the SF to an admittable state.
The process of generating the service change information may include a method of analyzing and generating information included in the SDP field of the SIP message in each application of the MS or a method of analyzing and generating the SIP message by a Convergence Sublayer It is generated in any one way.
The service change information includes at least one of a counterpart IP address, a port number, or CS rule generation information for transmitting a packet in the CS layer.
The service change information further includes QoS parameter information of at least one of Maximum Sustained Traffic Rate, Maximum Latency, Tolerated Jitter, and Request / Transmission Policy.
The setting of the SF is performed by setting at least one service profile ID or service class name for at least one SF to be used by the at least one MS.
The step of selecting the SF is to check whether the SF is designated by the service class name of each MS, and if a service class name is not designated, a service whose QoS parameter value among the service flows set in the service profile ID is subsetted The flow or the CS rule selects the SF based on the service flow having the same relative IP address or port number or the service change information and the QoS parameter value, and b) if the service class name is designated, The SF is selected based on the SF QoS parameter value of the provisioned state and the service change information.
The first DSC procedure is performed before the destination MS transmits the SIP 200 to the calling MS.
According to another aspect of the SF management method of the portable Internet system according to the present invention, the base station system sets a QoS profile for determining the suitability of the SF, A DSA (Dynamic Service Addition) process in which each MS adds a SF for establishing a call session; a DSC process for changing an SF added to each MS through the DSA process to an active state; Setting a call session through the SF changed to the active state; and performing a DSD (Dynamic Service Delete) process in which each MS deletes the SF when the call session ends.
Wherein the DSA process comprises: generating service grant information from an SIP message in which each MS is received; selecting an SF based on the service grant information; and transmitting DSA Transmitting a request message;
And the base station system granting the SF based on the service grant information and the QoS profile.
The step of generating the service grant information includes a method of analyzing and generating information included in the SDP field of the SIP message in each application of the MS or a method of analyzing and generating the SIP message by a Convergence Sublayer It is generated in any one way.
The service grant information includes at least one of an IP address of a partner, a port number, or CS rule generation information for transmitting a packet in the CS layer.
The service grant information further includes QoS parameter information of at least one of Maximum Sustained Traffic Rate, Maximum Latency, Tolerated Jitter, and Request / Transmission Policy.
The DSA procedure is performed before the called MS sends the SIP 200 to the calling MS.
According to an aspect of the present invention, there is provided a portable Internet system in which at least one SF to be used for a VoIP service is set in advance, and after a first signaling process with an opposite MS, , And after the call session is terminated after the call session is established by changing to the active state through the second DSC process, the SF is changed to the provision state through the third DSC process And includes at least one MS.
The first signaling process exchanges SIP INVITE and SIP 183 between the calling MS and the called MS.
The originating and receiving MSs generate service change information from the SDP field of the SIP message and perform the first DSC process by selecting an SF based on the QoS parameters of the preset SF and the service change information.
The service change information includes at least one of an IP address, a port number, and a CS rule generation information for transmitting a packet in the CS layer, and information of at least one of Maximum Sustained Traffic Rate, Maximum Latency, Tolerated Jitter, or Request / Transmission Policy And at least one QoS parameter information.
The preset SF is designated as a provision or active state, and is set to a service profile ID or a service class name for at least one SF.
The originating and receiving MSs confirm whether the SF is designated as a service class name, and if a) the service class name is not specified, the service flow or CS rule in which the QoS parameter value among the service flows set in the service profile ID is subsetted Selects the SF based on the service flow or the service change information and the QoS parameter value equal to the relative IP address and the port number, and b) if the service class name is designated, the service class name is set to the provisioning state And selects the SF based on the SF QoS parameter value and the service change information.
According to another aspect of the present invention, there is provided a portable Internet system including at least one base station system for setting a QoS profile for determining suitability of an SF added through a DSA process, A DSC process for changing an added SF added after performing the DSA process for adding a SF for setting a call session after the process is set up, And performs a DSD (Dynamic Service Delete) process for deleting the SF.
The at least one MS generates service grant information from a SIP message, selects an SF, and transmits the DSA request message to the base station system to perform the DSA procedure.
The base station system grants the SF based on the service grant information and the QoS profile received from the at least one MS.
Wherein the at least one MS analyzes the information included in the SDP field of the SIP message in an application and generates the SIP message in one of a method in which a Convergence Sublayer layer analyzes and generates the SIP message, Information.
Meanwhile, according to the operation method of the MS connected to the portable Internet system according to the present invention, the method of acquiring the request information of the SF from the SIP message received through the portable Internet network, and checking whether the SF is designated as the service class name Determining whether the corresponding SF is in a provision state if it is designated as the service class name; adding a SF through a DSA process if the SF is not in a provisioned state; Information and an QS parameter for the SF in the provisioned state, and if the QoS is appropriate, changing the state to an admittable state through a DSC process and rejecting the SF setting if it is not proper; , And selecting the SF of the provisioned state and changing it to the admittable state through the DSC process.
The request information of the SF includes at least one of IP address, port number, or CS rule generation information for transmitting a packet in the CS layer, and information of at least one of Maximum Sustained Traffic Rate, Maximum Latency, Tolerated Jitter, or Request / Transmission Policy.

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According to the present invention, when a VoIP service is provided through a portable Internet system, a service flow can be dynamically allocated to an MS by presetting a service flow to be used by the MS without implementing an expensive PCRF, The service flow is changed to the prevision state even after the end of the call session so that the management efficiency for the service flow can be improved.

In addition, if the base station system makes a DSA request from the MS, the service flow can be dynamically allocated based on the QoS profile based on the conformance to the addition of the corresponding service flow.

In addition, QoS parameters of the service flows to be used by the MS can be set in advance, or the QoS parameters can be provided on the basis of the QoS profile, thereby accurately estimating the VoIP service quality of the MS and ensuring QoS.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS Hereinafter, a method and apparatus for managing a service flow of a portable Internet system according to a preferred embodiment of the present invention will be described in detail with reference to the accompanying drawings, wherein the detailed description of known technology is omitted do.

The detailed description of the technical contents described in 'IEEE 802.16' is omitted from the detailed description of the terms and components according to the present invention.

2 is a flowchart illustrating a flow of providing a VoIP service in a portable Internet system according to a first embodiment of the present invention.

2, an operator of the portable Internet system transmits a call request message to an originating MS 100-1 (for example, an originating MS 100-1) in an authentication server (Radius or AAA server) (400 in FIG. 5) or a base station system (RAS / ACR) And provisioning of a plurality of service flows to be used by the destination MS 100-2 (S 101, S 102). At this time, the profile of the service flow set in advance can be divided into the service profile ID or the service class name.

In addition, the QoS parameter set value of the pre-set service flow can be set to the provisioning state when the originating MS 100-1 and the terminating MS 100-2 make initial connection.

On the other hand, the QoS parameter set value of the service flow can be set to the provisioning & active state, and even if the provisioning & active state is designated, the service flow is PPSF and the service flow can be allowed in the state where the CRR flag is OFF However, the service flow is released without being rejected.

At this time, the state of the corresponding service flow can be deleted, but if the CRR flag is OFF in the first embodiment of the present invention, it is maintained in the prevision state.

If there is a VoIP service request from the user, the source MS 100-1 transmits a SIP INVITE to the destination MS 100-2 through a previously used TCID (Transport Connection ID) (S 103, S 104).

That is, the SIP INVITE message is transmitted from the source MS 100-1 to the destination MS 100-2 through the base station system 200 and the call server 300, and the SDP field, i.e., the SDP Offer The destination IP address, port and QoS information, and the like.

When the SIP INVITE is received, the destination MS 100-2 transmits a SIP 183 to the calling MS 100-1 (S 105, S 106) informing that it is a call request (during call session establishment).

Then, the terminating MS 100-2 and the originating MS 100-1 perform a first DSC process for changing the state of the preset service flow to the admittable state (S 107, S 108).

At this time, the terminating MS 100-2 and the originating MS 100-1 perform the DSC process before the user answers a hook up call request, In order to pre-occupy service flow resources for providing a VoIP service before a call session is established, an IP terminal supporting the VoIP service can provide a call service in the same manner as a general phone terminal.

In addition, the destination MS 100-2 and the originating MS 100-1 occupy service flow resources in advance and provide various services (for example, multimedia services) in a standby state during the call through previously occupied resources .

Meanwhile, in the network group (NWG Network Working Group) defining EEE 802.16, for example, since the admission status change of the service flow is optional, the admission status is implemented according to the MS or policy of the portable Internet system Otherwise, the process of changing to the admittied state is skipped, and the state is changed to the active state.

In this way, when the service flow is converted into the active state, the service flow is converted through the following process.

The DSC process of the SDP field of the SIP signaling message, i.e., the information included in the SDP Offer message included in the SIP message, is performed.

The method by which the originating MS 100-1 and the terminating MS 100-2 generate service change information (service flow request information necessary for providing a VoIP service) for the DSC process is largely divided into a SIP message The information included in the SDP field is analyzed and transmitted to the modem, and the CS (Convergence Sublayer) layer parses the SIP message.

1) An application may use an application programming interface (API) or a shared file method to analyze the SDP Offer message of a SIP message and deliver it to the modem. The QoS API does not allow the modem to parse the SIP message, And transmits the service change information to the modem.

The service change information includes information for generating a CS rule for transmitting packets in the CS layer such as the IP address and the port number of the peer and information for generating a QoS rule such as Maximum Sustained Traffic Rate, Maximum Latency, Tolerated Jitter, or Request / Information corresponding to the parameter.

2) The CS layer parses the SIP message and searches the internal SDP field. That is, information for creating a CS rule such as the IP address and the port number of a partner included in the SDP field of the SIP message and QoS parameter information such as Maximum Sustained Traffic Rate, Maximum Latency, Tolerated Jitter, or Request / Transmission Policy As service change information.

Then, the source MS 100-1 and the destination MS 100-2 extract the request information for the requested service flow based on the identified service change information.

At this time, if the service class name is designated, the application can use the service claim name as request information for the service flow.

1) If the service class name is not specified

1-1) Select a required service flow among preset service flows, and select a service flow having a QoS parameter value as a subset among service flows in the prevision state.

On the other hand, if there is a service flow in which the CS rule is the same as the relative IP address and port number delivered from the application, the corresponding service flow is selected.

1-2) If the Maximum Sustained Traffic Rate, Maximum Latency, and Tolerated Jitter are defined in the service flow in the provisioning state, the comparison is made based on the following conditions.

(Required Maximum Sustained Traffic Rate) && (Required Maximum Latency <= Providence Maximum Latency) && (Required Tolerated Jitter <= Provided Tolerated Jitter)

If there is a service flow satisfying the above condition, the MS 100 selects the corresponding service flow.

On the other hand, when a plurality of service flows are selected, the service change information required for the service flow and the service having the closest value are firstly selected.

In addition, the QoS parameter values which are not recognized in the SDP field of the SIP message, but are necessarily set, are set based on the QoS parameter values of the service flows in the prevision state.

The originating MS 100-1 and the terminating MS 100-2 perform a DSC process for converting the service flow of the selected provisioning state into the admission state. That is, the DSC-REQ message is transmitted to the base station system 200 based on the service change information identified from the SIP message, the QoS parameter value of the service flow in the provisioning state, and the QoS parameter value set in the portable Internet system, .

Therefore, the source MS 100-1 and the destination MS 100-2 can select a service flow satisfying the required one of preset service flows and occupy resources for providing VoIP service.

2) When the service class name is specified

For example, if an operator wishes to designate a Samsung VOIP Application using G.711 as a service claim, the service claim name including a plurality of service flows can be designated as 'G711_Samsung_1', and the maximum Sustained Rate, Maximum Latency, or CS rule.

On the other hand, when the service class name is specified, the required QoS parameter value for the service flow in the provisioning state can be designated.

In addition, in the case of a public network instead of the private network, a required service flow can be designated by using a global service claim name. Since the global service class name is based on rule naming, a required QoS parameter value is set.

The source MS 100-1 and the destination MS 100-2 check if the service class corresponding to the service class name is in the provision state if the service class name is specified, The service change information is compared with the QoS parameter value of the service flow in the provisioning state.

For example, the condition is compared with the above condition, and if satisfied, the corresponding service flow is selected.

On the other hand, if several service flows are selected, the service change information required for the service flow and the service having the closest value are selected in advance.

In addition, the QoS parameter values which are not recognized in the SDP field of the SIP message, but are necessarily set, are set based on the QoS parameter values of the service flows in the prevision state.

The originating MS 100-1 and the terminating MS 100-2 perform a DSC process for converting the service flow of the selected provisioning state into the admission state. That is, the DSC-REQ message is transmitted to the base station system 200 based on the service change information identified from the SIP message, the QoS parameter value of the service flow in the provisioning state, and the QoS parameter value set in the portable Internet system, .

On the other hand, when the service flow designated by the service class name is in a NULL state other than the provisioned state, the source MS 100-1 and the destination MS 100-2 perform the DSA process and the active state, If this is in use, convert it to an admittable state according to the usage policy, or use error handling.

Thus, the originating MS 100-1 and the terminating MS 100-2 can perform the DSC procedure for changing the selected service flow to the admittable state, i.e., the originating MS 100-1 and the terminating MS 100 2 transmits a DSC-REQ message to the base station system 200 and the base station system 200 transmits a DSC-RSP message to the calling MS 100-1 and the called MS 100-2 The TCID to which the traffic is exchanged is allocated to the calling MS 100-1 and the terminating MS 100-2.

Meanwhile, the called MS 100-2 transmits a SIP 200 (ok) message to the calling MS 100-1 when the user hangs off after performing the first DSC process (S 109 , S110).

At this time, the SIP 200 includes an SDP Answer corresponding to the SDP Offer included in the SIP INVITE.

The source MS 100-1 and the destination MS 100-2 perform a second DSC process of changing the service flow converted from the provisioned state to the admittable state into the active state in the first DSC procedure S 111, S 112).

That is, the calling MS 100-1 and the terminating MS 100-2 change the service flow, which is occupied in advance in the first DSC process and changed to the admittable state, to the active state.

On the other hand, when the admission status is not supported in the portable Internet system, the provisional status is changed to the active status in the second DSC process, not the first DSC process, and the TCID is allocated.

When the admission status of the service flow is changed to the active status, if the CS rules included in the SIP 200 are the same or match the service class name, the admission status of the corresponding service flow is changed to the active status and the QoS parameter set is omitted And a DSC process can be performed based on an existing admit QoS parameter set.

When the service flow is changed to the active state in the second DSC process, the source MS 100-1 transmits a SIP ACK to the destination MS 100-2 (S113, S114).

Then, the source MS 100-1 and the destination MS 100-2 exchange traffic including voice information through the TCID assigned in the first DSC or the second DSC process.

In this case, the traffic is exchanged by being included in the RTP packet, QoS for the RTP packet is guaranteed by scheduling of the portable Internet system, and some of the QoS parameters are modified in the second DSC process according to the characteristics of the traffic including voice information. It is possible.

When the VoIP service is terminated, the calling MS 100-1 and the terminating MS 100-2 exchange the SIP BYE and the SIP 200 OK (S 115 to S 118).

The originating MS 100-1 and the terminating MS 100-2 are configured so that the CS rule included in the SIP message is the same or the third rule for changing the service flow having the same service class name into the provisioning state DSC process is performed (S 119, S 120).

That is, when the call is terminated, the calling MS 100-1 and the terminating MS 100-2 change the ACTIVE state to the provisioning state through the third DSC process without deleting the service flow.

Thus, dynamic management of the service flow is possible by the originating MS 100-1 and the terminating MS 100-2 changing to the provisioning state without releasing the preset service flow.

At this time, if the state of the service flow managed by the source MS 100-1 and the destination MS 100-2 is (active & provision) or (active & admit & provision) exists , And changes to the provisioning state through the third DSC process. If the provisioning state does not exist, the service flow is deleted through the DSD process and the TCID for exchanging traffic is released.

3 is a flowchart illustrating a service flow management method of a portable Internet system according to a first embodiment of the present invention.

Referring to FIG. 3, the operator of the portable Internet system sets a plurality of service flows to be used by the MS in advance (S 200).

At this time, the operator can preset the service flow to be used by the MS in the authentication server 400 (Radius or AAA server), the base station system 200 (RAS / ACR), or the ASN-GW in the service profile ID or service class name .

Then, the operator can designate the QoS parameter set value of the service flow preset for the MS to the provisioning state when the source MS 100-1 and the destination MS 100-2 make initial connection.

If there is a VoIP service request from the user, the source MS 100-1 transmits a SIP INVITE to the destination MS 100-2 through a TCID (Transport Connection ID) that has been used (S 210).

When the SIP INVITE is received, the terminating MS 100-2 transmits a SIP 183 indicating that it is a call request (during a call session establishment) to the calling MS 100-1 (S220).

Then, the receiving MS 100-2 and the originating MS 100-1 select a service flow in the prevision state among the preset service flows, change the state from the prevision state to the admittable state through the first DSC process , And receives a TCID (S 230).

At this time, the first DSC process performs the DSC process before the user answers the call request (Hook-off).

4 is a flowchart illustrating a first DSC process according to the first embodiment of the present invention.

Referring to FIG. 4, the MS grants the information included in the SDP Offer message included in the SDP field of the SIP signaling message (SIP INVITE and SIP 183), i.e., the SIP message, to generate the service change information (S 231).

As described above, the method of generating the service change information includes a method of analyzing the information included in the SDP field of the SIP message in the application of the MS and delivering the information to the modem, and a method in which the Convergence Sublayer layer parses the SIP message Method may be applicable.

Then, the MS confirms whether a preset service flow is designated as a service class name (S 232).

If the service class name is not specified, the MS determines whether the service flow or the CS rule having the QoS parameter value of the subset of the service flow set in the service profile ID has the same service flow or service change information as the relative IP address and port number delivered from the application The QoS parameter value is compared with the above conditions and a satisfactory service flow is selected (S 233).

On the other hand, when the service class name is designated, the MS compares the QoS parameter value of the service flow, which is a provisioned state among a plurality of service flows assigned with the service class name, with the service change information, (S 234).

Meanwhile, the called MS 100-2 transmits a SIP 200 (ok) to the calling MS 100-1 when the user hangs off after performing the first DSC process (S 240) .

At this time, the SIP 200 includes an SDP Answer corresponding to the SDP Offer included in the SIP INVITE.

The source MS 100-1 and the destination MS 100-2 perform a second DSC process of changing the service flow converted from the provisioned state to the admittable state into the active state in the first DSC procedure S 250).

That is, the calling MS 100-1 and the terminating MS 100-2 change the service flow, which is occupied in advance in the first DSC process and changed to the admittable state, to the active state.

When the service flow is changed to the active state in the second DSC process, the source MS 100-1 transmits a SIP ACK to the destination MS 100-2 (S260).

The originating MS 100-1 and the terminating MS 100-2 then transmit a call session for exchanging traffic including voice information through the TCID allocated in the first DSC or the second DSC (S 270).

The calling MS 100-1 and the terminating MS 100-2 exchange SIP BYE and SIP 200 OK when the VoIP service is terminated (S280).

The originating MS 100-1 and the terminating MS 100-2 can not change the provision rule state without changing the CS rule contained in the SIP message or deleting the service flow whose service class name agrees A third DSC process is performed (S 290).

5 is a flowchart illustrating a flow of providing a VoIP service in a portable Internet system according to a second embodiment of the present invention.

In the flow of the second embodiment of the present invention described in FIG. 5, the detailed description of the same message flow as described in FIG. 2 is omitted.

The operator of the portable Internet system according to the second embodiment of the present invention sets a QoS profile for a service class to be used by the MS, that is, a QoS parameter value for a service flow to be used by the MS, The base station system 200 downloads the QoS profile for judging the conformity of the mobile terminal 200 from the authentication server 400 (S 301, S 302).

Since the PCRF is responsible for determining the suitability of the DSA, the portable Internet system according to the present invention does not implement the expensive PCRF, so the ASN-GW takes charge of policy making and the base station system 200 downloads the QoS profile do.

At this time, the option of R3 Action Trigger is set to 0x10, and the R3 Activation Trigger of the QoS profile is as follows.

= without PCC (0x10) & QoSParamSetValue

0x11: QoS Profile for Provisioning

0x12: QoS Profile for Admits

0x14: QoS Profile for Activate

0x16: QoS profile for all state

The QoS profile starting with 0x1X is not downloaded by the MS and is stored by the ASN.

The originating MS 100-1 transmits the SIP INVITE to the terminating MS 100-2 (S 303, S 304) through the TCID that has been used in accordance with the request of the subscriber, Sends the SIP 183 to the originating MS 100-1 (S305, S306).

The SIP message includes an SDP Offer message. The SDP Offer message includes a relative IP address and port number of the VoIP service, QoS information of the VoIP service, and the like.

The originating MS 100-1 and the terminating MS 100-2 perform a DSA process to add the service flow to the ADMIT state after transmitting the SIP INVTE or SIP 183 (S307, S308) .

That is, the source MS 100-1 and the destination MS 100-2 perform a DSA process for adding a service flow for exchanging voice information before the call session is established, and occupy resources in advance .

However, since the admission status change of the service flow is optional in the network group of the WiMAX Forum, for example, defining the EEE 802.16, the policy of the MS or the portable Internet system is implemented as an admittable state If not, the DSA process is skipped and the DSC process is performed to convert the DSC process to the active state.

Thus, the DSA process for adding a service flow is as follows.

The MS grasps the SDP field of the SIP signaling message, i.e., the information included in the SDP Offer message included in the SIP message, and generates the service grant information.

The method by which the source MS 100-1 and the destination MS 100-2 generate the service grant information for the DSA procedure (request information of the service flow necessary for providing VoIP service) is the same as the first DSC procedure Lt; RTI ID = 0.0 &gt; service change information. &Lt; / RTI &gt;

That is, a method of generating service grant information can be roughly divided into a method of analyzing information included in an SDP field of a SIP message in an application and delivering it to a modem, and a method of parsing a SIP message by a CS (Convergence Sublayer) layer.

1) An application may use an application programming interface (API) or a shared file method to analyze the SDP Offer message of a SIP message and deliver it to the modem. The QoS API does not allow the modem to parse the SIP message, Send service grant information to the modem.

The service change information includes information for generating a CS rule for transmitting packets in the CS layer such as the IP address and the port number of the peer and information for generating a QoS rule such as Maximum Sustained Traffic Rate, Maximum Latency, Tolerated Jitter, or Request / Information corresponding to the parameter.

2) The CS layer parses the SIP message and searches the internal SDP field. That is, information for creating a CS rule such as the IP address and the port number of a partner included in the SDP field of the SIP message and QoS parameter information such as Maximum Sustained Traffic Rate, Maximum Latency, Tolerated Jitter, or Request / Transmission Policy As service grant information.

Then, the source MS 100-1 and the destination MS 100-2 extract request information for the requested service flow based on the grasped service grant information.

At this time, if the service class name is designated, the application can use the service claim name as request information for the service flow.

Then, the MS transmits the DSA-REQ message including the service grant information to the base station system 200 to perform the DSA procedure.

At this time, the MS confirms whether the service class name is designated.

1) If the service class name is not specified, the MS transmits a DSA-REQ message including the service grant information (QoS parametric value) to the base station system 200 after selecting the required service flow.

The base station system 200 verifies that it is valid based on the QoS profile stored in the stored ASN-GW if the PCC of the R3 activation trigger in the DSA-REQ message is a QoS parameter set value set to 0x1X.

The base station system 200 accepts a DSA-REQ message if the QoS parameter set value is a subset of the QoS profile in the ASN. Then, the base station system 200 confirms whether the VoIP service can be provided through the corresponding service flow through the CAC process.

2) If the service class name is specified

The MS transmits the service class name and the service grant information to the base station system 200 through the DSA-REQ message, and the base station system 200 transmits the VoIP service through the corresponding service flow through the CAC process. Make sure you can provide it.

Since the system has a service class name assigned to the DSA-REQ message, the base station compares the required service class with the required capacity.

In the case of a global service class, the global capacity of the service class is included, so the MS may directly compare the capacity.

The base station system 200 compares the service class to be used by the MS according to the following conditions when Maximum Sustained Traffic Rate, Maximum Latency, or Tolerated Jitter is defined.

(Required Maximum Sustained Traffic Rate) && (Required Maximum Latency <= Service Class Maximum Latency) && (Required Tolerated Jitter <= Use Service Class Tolerated Jitter) Check

When the above conditions are satisfied, the base station system 200 permits the corresponding service flow.

On the other hand, the QoS parameter value not extracted from the SDP field of the SIP message is based on the value set in the service class.

If the service grant information included in the DSA-REQ message is allowed, the base station system 200 transmits a DSA-RSP message to the corresponding MS, generates a TCID for the resource, allocates the TCID to the MS, (S309, S310).

Therefore, the portable Internet system according to the second embodiment of the present invention can not determine the service class to be used by the MS in the same manner as the first embodiment of the present invention, The service flow can be dynamically generated.

The called MS 100-2 sends a SIP 200 (ok) to the calling MS 100-1 (S 311, S 312) when the user answers (Hook off).

When the destination MS 100-2 transmits the SIP 200 and the originating MS 100-1 receives the SIP 200, the service flow of the ADMITTED state added through the DSA process is made active through the DSC process (S 313, S 314).

At this time, if the MS matches the CS rule included in the SIP 200 in the DSC process or if the service class name matches, the MS changes the service flow to the active state.

In addition, the QoS parameter set can be omitted in the DSC process, and it is changed to the active state based on the existing admit QoS parameter set.

The originating MS 100-1 sends a SIP ACK to the terminating MS 100-2 and each base station system 200 compares the QoS profile for the changed service flow to the active state (S 315, S 316 ).

Then, the source MS 100-1 and the destination MS 100-2 exchange traffic including voice information through the TCID assigned in the first DSC or the second DSC process.

At this time, the traffic is exchanged by being included in the RTP packet, and the QoS for the RTP packet is guaranteed by the scheduling of the portable Internet system.

The calling MS 100-1 and the terminating MS 100-2 exchange the SIP BYE and the SIP 200 OK when the VoIP service is terminated (S 317 to S 320).

When the VoIP service is terminated, the source MS 100-1 and the destination MS 100-2 perform a DSD process of deleting the same service flow having the same CS rule or the same service class name (S321 , S 322).

6 is a flowchart illustrating a service flow management method of a portable Internet system according to a second embodiment of the present invention.

Referring to FIG. 6, the operator of the portable Internet system sets a QoS profile for a service class to be used by the MS, that is, a QoS parameter value for a service flow to be used by the MS (S 400).

The base station system 200 downloads the QoS profile for determining the suitability of the service flow set by the MS through the DSA process from the authentication server 400 by the base station system 200 in operation S 410.

The originating MS 100-1 transmits a SIP INVITE to the terminating MS 100-2 through the TCID that has been used according to the subscriber's request (S 420), and the terminating MS 100-2 transmits the SIP INVITE to the SIP 183 to the calling MS 100-1 (S 430).

The originating MS 100-1 and the terminating MS 100-2 perform a DSA procedure to add the service flow to the ADMIT state after transmitting the SIP INVTE or SIP 183 (S440).

7 is a flowchart illustrating a DSA process according to a second embodiment of the present invention.

Referring to FIG. 7, the MS grants the information included in the SDP Offer message included in the SDP field of the SIP signaling message, that is, the SIP message, to generate service grant information (S441).

Then, the MS transmits a DSA-REQ message including the service grant information to the base station system 200 (S442).

The base station system 200 permits the service flow by comparing the QoS grant value included in the DSA-REQ message, that is, the QoS parameter value with the QoS profile (S443).

If the service grant information included in the DSA-REQ message is allowed, the base station system 200 transmits a DSA-RSP message to the corresponding MS, generates a TCID for the resource, allocates the TCID to the MS, (S444).

Meanwhile, the called MS 100-2 transmits a SIP 200 (ok) to the calling MS 100-1 when the user responds (Hook off) (S 450).

When the destination MS 100-2 transmits the SIP 200 and the originating MS 100-1 receives the SIP 200, the service flow of the ADMITTED state added through the DSA process is made active through the DSC process (S 460).

The originating MS 100-1 sends a SIP ACK to the terminating MS 100-2, and each base station system 200 compares the QoS profile for the changed service flow to the active state (S 470).

Then, the source MS 100-1 and the destination MS 100-2 exchange traffic including voice information through the TCID assigned in the first DSC or the second DSC process (S 480).

When the VoIP service is terminated, the source MS 100-1 and the destination MS 100-2 perform a DSD process of deleting the service flow having the same CS rule or matching the service class name (S 490 ).

8 to 10 are flowcharts illustrating an operation method of an MS according to a preferred embodiment of the present invention.

8 is a diagram illustrating an operation method of a terminal supporting both the first embodiment (DSC process) and the second embodiment (DSA process) according to the present invention, in which a SIP message, for example, SIP 183 or SIP INVITE If it is received (S 510), the SDP field is analyzed to analyze required request information (service change information or service grant information) (S 511).

The MS confirms whether the service flow is designated as a service class name (S 512). If the service flow name is designated as a service class name, the MS checks whether the corresponding service flow is in a provision state (S 513).

If the MS is not in the provisioning state, it processes the DSA procedure for adding the corresponding service flow (S 514).

If the MS is in the provisioning state, the MS compares the requested service flow with the corresponding service flow and confirms that it is reasonable (S 515). If it is, the MS processes the DSC process (S 516) (S 517).

If the service class is not set, the MS determines whether there is a service flow that satisfies the request information among the service flows in the provisioned state among the designated service flows (S 518).

If there is a service flow satisfying the request information, the MS processes the DSC process for changing the service flow to the admittable state (S 519). If not, the MS processes or rejects the DSA process (S 520).

If the service class name is not specified, the MS compares the provisioned service flow with the requested service flow (S 518). If the MS satisfies the service flow, the MS performs the DSC process (S 519) , The setting of the corresponding service flow is rejected (S 520).

9 is an operation flow of the MS in which the service flow in the admittable state can be changed to the active state,

MS For example, the originating MS 100-1 analyzes the request information of the requested service flow by analyzing the SDP field of the SIP 200 (S530), that is, the SIP 200 (S531).

The MS checks whether the corresponding service flow is in the admission state based on the CS rule or the service class name (step 532). If not, the MS processes the DSA process or the DSC process (S 533).

The MS confirms whether the request information of the requested service flow is included in the service flow in the admittable state (S 534), and if so, processes the DSC process to change the active state to the active state (S 536) If not, the setting of the service flow is rejected (S 535).

10 is an operation flow of an MS that processes a DSC process or a DSD process in a service flow in an active state. When the MS terminates the call, that is, when SIP BYE and SIP 200 OK are exchanged (S 540), the MS analyzes the SDP field The requirements of the required service flow are analyzed (S 541).

The MS determines whether the corresponding service flow is in the active state based on the CS rule or the service class name (S 542). If not, the MS determines that the resource has already been released in a certain state (S 543).

The MS checks whether the corresponding service flow is in the provision state (S 544). If not, the MS processes the DSC process for changing to the provision state (S 545). If the provision state is present, (S 546).

While the present invention has been particularly shown and described with reference to exemplary embodiments thereof, it is evident that many alternatives, modifications and variations will be apparent to those skilled in the art.

FIG. 1 is a flowchart illustrating a flow of providing a VoIP service through a general portable Internet system.

FIG. 2 is a flowchart illustrating a flow of providing a VoIP service in a portable Internet system according to a first embodiment of the present invention; FIG.

3 is a flowchart illustrating a service flow management method of a portable Internet system according to a first embodiment of the present invention.

FIG. 4 is a flowchart illustrating a first DSC process according to the first embodiment of the present invention; FIG.

FIG. 5 is a flowchart illustrating a flow of providing a VoIP service in a portable Internet system according to a second embodiment of the present invention; FIG.

6 is a flowchart illustrating a service flow management method of a portable Internet system according to a second embodiment of the present invention.

FIG. 7 is a flowchart illustrating a DSA process according to a second embodiment of the present invention; FIG.

8 to 10 are flowcharts illustrating an operation method of an MS according to a preferred embodiment of the present invention.

Claims (28)

A method for managing Service Flow (SF) of a portable Internet system, Setting at least one SF to be used by at least one mobile station (MS) connecting to a portable Internet network; A first signaling process for establishing a call session between a calling MS and a called MS, A first DSC (Dynamic Service Change) process for each MS to change the set SF to an admittable state; A second DSC process of changing the SF changed to the admittable state in the first DSC process to an active state when the called MS responds to the talk request, And a third DSC process of changing the SF to a provisioning state when a call session between the calling MS and the called MS ends. The method of claim 1, wherein the setting of the SF comprises: Wherein the SF is set to one of a provision state and an active state or more. 2. The method of claim 1, wherein the first signaling process comprises: Sending the SIP INVITE to the called MS; And the destination MS sending SIP 183 to the originating MS. The method of claim 1, wherein the first DSC process comprises: Generating service change information from the SIP message in which each MS is received; Selecting an SF based on the QoS parameter of the preset SF and the service change information; And transmitting a DSC request message to the base station system to change the selected SF to an admittable state. 5. The method of claim 4, wherein the step of generating the service change information comprises: A method of analyzing and generating information included in the SDP field of the SIP message in an application of each MS or a method of analyzing and generating the SIP message by a CS (Convergence Sublayer) layer SF management method of portable internet system. The method as claimed in claim 4, And information on at least one of a destination IP address, a port number, or CS rule generation information for transmitting a packet in the CS layer. 7. The method according to claim 6, Wherein the QoS parameter information includes at least one of a Maximum Sustained Traffic Rate, a Maximum Latency, a Tolerated Jitter, and a Request / Transmission Policy. The method of claim 1, wherein the setting of the SF comprises: Wherein the at least one MS sets a service profile ID or a service class name for at least one SF to be used by the at least one MS. 5. The method of claim 4, wherein the step of selecting SF comprises: Each MS determines whether the SF is designated as a service class name, a) If the service class name is not specified, the service flow or CS rule in which the QoS parameter value of the service flow set in the service profile ID is subsetted is the same as the relative IP address and port number, or the service change information and the QoS parameter value The SF is selected based on the SF, and b) if the service class name is designated, the service class name is selected based on the SF QoS parameter value in the provisioned state of the specified service flow and the service change information. . The method of claim 1, wherein the first DSC process comprises: Wherein the destination MS is performed before the destination MS transmits the SIP 200 to the originating MS. A method of managing SF in a portable Internet system, Setting a QoS profile for the base station system to determine the suitability of the SF, A first signaling process between the calling MS and the called MS, A DSA (Dynamic Service Addition) process in which each MS adds an SF for establishing a call session; A DSC process for changing the SF added to each MS through the DSA process to an active state, Establishing a call session through the SF changed to the active state by each MS; And a DSD (Dynamic Service Delete) process for deleting the SF when each MS ends the call session. 12. The method according to claim 11, Generating service grant information from each SIP message received by the MS; Selecting an SF based on the service grant information, Transmitting a DSA request message to the base station system to add the selected SF; And allowing the base station system to grant the SF based on the service grant information and the QoS profile. 13. The method of claim 12, wherein the generating the service- A method of analyzing and generating information included in the SDP field of the SIP message in an application of each MS or a method of analyzing and generating the SIP message by a CS (Convergence Sublayer) layer SF management method of portable internet system. 14. The method of claim 13, wherein the service- And information on at least one of a destination IP address, a port number, or CS rule generation information for transmitting a packet in the CS layer. 15. The method of claim 14, wherein the service- Wherein the QoS parameter information includes at least one of a Maximum Sustained Traffic Rate, a Maximum Latency, a Tolerated Jitter, and a Request / Transmission Policy. 12. The method according to claim 11, Wherein the destination MS is performed before the destination MS transmits the SIP 200 to the originating MS. In a portable Internet system, A first DSC process for setting at least one or more SFs to be used for the VoIP service and changing the selected SF among the set SFs to the admittable state after the first signaling process with the other MS is performed, And changing the SF to a provisioning state through a third DSC process when the call session is terminated after changing a call session to an active state. 18. The method of claim 17, wherein the first signaling process comprises: Wherein the calling MS and the called MS exchange SIP INVITE and SIP 183. 19. The method as claimed in claim 18, Generates the service change information from the SDP field of the SIP message, and performs the first DSC process by selecting the SF based on the QoS parameter of the preset SF and the service change information. The method as claimed in claim 19, The QoS parameter information of at least one of the IP address, the port number of the peer or the CS rule generation information for transmitting packets in the CS layer and the maximum sustained traffic rate, the maximum latency, the tolerated jitter, or the request / Includes a portable Internet system. 18. The method as claimed in claim 17, The service profile ID or the service class name for at least one of the SFs. 18. The method of claim 17, The originating and receiving MSs confirm whether the SF is designated as a service class name, a) If the service class name is not specified, the service flow or CS rule in which the QoS parameter value among the service flows set in the service profile ID is subsetted is the same as the relative IP address and the port number, or the service change information and the QoS parameter Selecting the SF based on the value, b) if the service class name is designated, selecting the SF based on the SF QoS parameter value in the provisioned state of the specified service flow with the service class name and the service change information. In a portable Internet system, At least one base station system for setting a QoS profile for judging suitability of the SF added through the DSA process, A DSC process for changing the added SF added to the active state after performing the DSA process for adding a SF for establishing a call session with the other MS for the VoIP service after the first signaling process, And a DSD (Dynamic Service Delete) process for deleting the SF when the call session is terminated. 24. The method of claim 23, wherein the at least one MS comprises: Generates the service grant information from the SIP message, selects the SF, and transmits the DSA request message to the base station system to perform the DSA process. 25. The base station of claim 24, And grants the SF based on the service grant information and the QoS profile received from the at least one MS. 25. The method of claim 24, wherein the at least one MS comprises: The service providing information is generated by analyzing the information included in the SDP field of the SIP message in an application and generating the service grant information by a method of analyzing and generating the SIP message by a CS (Convergence Sublayer) layer Portable internet system. A method of operating an MS connected to a portable Internet system, The method comprising the steps of: receiving request information of an SF from a SIP message received through a portable Internet network; Confirming whether the SF is designated as a service class name; Checking whether the corresponding SF is provisioned if the service class name is designated; If not in the provisioning state, the process of adding SF through the DSA process, If it is in the provision state, the request information of the SF is compared with the QS parameter of the SF in the provisioned state. If the request information is valid, the state is changed to the admittable state through the DSC process. and, And selecting an SF in the provisioned state and changing to an admittable state through a DSC process if the service class is not designated. 28. The method as claimed in claim 27, Information on at least one of the IP address, the port number, and the CS rule generation information for transmitting the packet in the CS layer and the maximum sustained traffic rate, the maximum latency, the tolerated jitter, or the request / transmission policy A method of operation of an MS accessing a portable Internet system comprising at least one QoS parameter information.

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