KR100636279B1 - Call admission contorl system and method using resource in voice over internet protocal system - Google Patents

Abstract

According to the present invention, a VIP is used to efficiently perform call control by reflecting the bandwidth of an internet trunk during call signaling in a VoIP system using a session initialization protocol (SIP). The present invention relates to a call control system and method using resource information of a system. According to the present invention, when a call connection request message including SDP protocol information is received from an arbitrary terminal, resources of a network system are transmitted to a router of a source and a destination. After transmitting the option message for requesting the information, modifying the SDP protocol information according to the resource information of the network system received from the router of the source and the destination to transmit the modified SDP protocol information to the destination terminal to control the call connection It characterized in that it comprises a relay server.

SIP protocol, call signaling, internet trunk, bandwidth, codec information

Description

Call control system and its method using resource information of VIO system {CALL ADMISSION CONTORL SYSTEM AND METHOD USING RESOURCE IN VOICE OVER INTERNET PROTOCAL SYSTEM}

1 is a diagram illustrating a location information registration process of a user agent in a conventional VoIP system.

2 is a diagram illustrating a call connection process using a SIP protocol in a conventional VoIP system.

3 is a diagram showing a schematic configuration of a call control system of VoIP according to the present invention.

4 is a diagram illustrating an example of an option message according to the present invention;

5 is a diagram illustrating an example of a message including resource information according to the present invention.

FIG. 6 is a diagram illustrating parameters of a P-CAC-Codec header in the resource information message header of FIG. 5; FIG.

FIG. 7 is a diagram illustrating an example of a media format among parameters of a P-CAC-Codec header of FIG. 6. FIG.

8 illustrates a configuration of a proxy server according to the present invention.

9 is a diagram illustrating a process of a call control method using resource information of a BIO system according to the present invention;

FIG. 10 is a diagram illustrating a process when a call is rejected among the call control processes of FIG. 9; FIG.

Explanation of symbols on the main parts of the drawings

10, 50: VoIP terminal 20, 40: router

30: proxy server 31: IP interface unit

32: SIP protocol analysis unit 33: call control unit

34: resource information management unit 35: subscriber management unit

The present invention relates to a call control system using a resource information of a Voice over Internet Protocol (VoIP) system and a method thereof, and more particularly to a session initialization protocol (SIP). The present invention relates to a call control system and a method using resource information of a VoIP system that efficiently performs call control by reflecting a bandwidth of an Internet trunk during call signaling in a used VoIP system.

Session Initial Protocol (SIP) is a signaling protocol for creating, modifying and terminating Internet-based multimedia service sessions. The role of such a signaling protocol is to exchange information for controlling a message session between subjects who want to exchange messages.

In order to make a call in a general telephone network, the process of pressing a dial number, listening for a waiting sound until the other party picks up the telephone receiver, and returning a line resource after the call is performed through control message transmission through signaling.

VoIP network using SIP is composed of user agent and SIP server. In particular, the SIP server includes a proxy / redirect server for call signaling, a location server for managing location information of a user agent, and a user agent. It consists of a register for managing the registration information of the user agent).

More specifically, a user agent is a part that operates with an actual user, and may be divided into a requesting UAC (client) and a response UAS (server), but in reality, both parts are one. It is included in the program.

The proxy server forwards the request message to the place where the actual message should be delivered. In other words, the proxy mode continues to receive and forward subsequent messages, but in the redirect mode, the SIP mode delivers a SIP-URI that can be requested immediately to the client.

In addition, a register registers a user agent and, when a request is received from another user agent, provides a registered information so that a message can be delivered to the user agent. The authentication process also distinguishes between legitimate users.

The location server is a DB in which a user's location is registered, not a SIP UA, and information is exchanged between a location server and another SIP member by a method other than SIP.

FIG. 1 is a diagram illustrating a location information registration process of a user agent in a VoIP system according to the related art. FIG. 2 is a diagram illustrating a call connection process using a SIP protocol in a conventional VoIP system. SIP Call Signaling is used as follows.

First, as shown in FIG. 1, the first user 1 and the second user 2 corresponding to the user agent register their location information with the SIP registrar server 3. After S1), a 200 OK message is received (S2).

Next, as shown in FIG. 2, when the first user 1 calls the second user 2, the first user 1 first sends an INVITE message to the proxy server 4 (S3). This INVITE message includes media information along with call signaling information. In SIP, Session Description Protocol (SDP) is used.

Subsequently, the proxy server 4 analyzes the INVITE message and sends an INVITE message to the second user 2 corresponding to the destination (S4). The 100 Trying message indicates that the corresponding call is normally signaled. Through the notification (S5) to the first user (1) through.

Subsequently, when the second user 2 receives the call, the second user 2 transmits a 180 ringing message to the first user 1 (S6), and then the first user indicates that call signaling has been normally completed through the 200 OK message. When (1) is notified (S7), the first user 1 transfers an ACK message to the second user 2 (S8).

As a result, a real-time transport protocol (RTP) for substantially media transmission between the first user 1 and the second user 2 is set. At this time, RTP is set based on the contents of SDP included in the initial INVITE message.

In this conventional technique, the SDP for setting the actual media is generated and referenced by each user agent. In this case, the user agent simply includes codec information that can be supported by the user agent and selects an appropriate codec from the user agent of the transmitting and receiving end, but actually the user agent Does not know whether the call is set up in an environment with sufficient bandwidth, such as a Local Area Network (LAN), or in a limited environment, such as a Wide Area Network (WAN). do.

In addition, if you look at the general network configuration, the bandwidth information is actually managed by a router that performs a traffic transfer by configuring a LAN or WAN. There was a problem that the agent (User Agent) as well as the proxy server (Proxy Server) can not know.

An object of the present invention for solving the above problems, in the VoIP system using the Session Initial Protocol (SIP) to check the resource (Resource) information during call signaling (Call Signaling) to efficiently accommodate the call By selecting the appropriate codec to reject or rejecting the call, the system can efficiently use the limited Internet trunk bandwidth and reduce the system's unnecessary call signaling to reduce the system's efficiency and performance. To provide a call control system and a method using the resource information of the VIP system to increase the.

According to an aspect of the call control system using the resource information of the VIP system according to the present invention for achieving the above object, when a call connection request message including SDP protocol information is received from any terminal, the source and After transmitting an option message for requesting the resource information of the network system to the router of the destination, modifying the SDP protocol information according to the resource information of the network system received from the router of the source and the destination to modify the SDP protocol information to the destination terminal. It includes a relay server to transmit to control the call connection.

The option message includes IP address information for selecting a router trunk interface of the destination.

The resource information of the network system includes at least one of codec information and bandwidth information.

The relay server, when a call connection request is generated from any terminal, the call control unit for controlling the call connection according to the resource information of the network system received from the router of the source and destination for the option message, and the call control unit of The apparatus further includes a resource information management unit for checking the resource information of the network system and modifying the SDP protocol information according to a request.

The call controller connects the requested call when the resource information of the network system is secured and rejects the requested call when the resource information of the network system is not secured.

On the other hand, according to one aspect of the call control method using the resource information of the VIP system according to the present invention for achieving the above object, when a call connection request message containing SDP protocol information from any terminal, Transmitting an option message for requesting resource information of a network system to a router of a source and a destination, modifying the SDP protocol information according to resource information of a network system received from the router of the source and the destination; And transmitting the modified SDP protocol information to the destination terminal to control the call connection.

DESCRIPTION OF THE PREFERRED EMBODIMENTS Hereinafter, detailed descriptions of preferred embodiments of the present invention will be described with reference to the accompanying drawings. It should be noted that reference numerals and like elements among the drawings are denoted by the same reference numerals and symbols as much as possible even though they are shown in different drawings. In the following description of the present invention, if it is determined that a detailed description of a related known function or configuration may unnecessarily obscure the subject matter of the present invention, the detailed description thereof will be omitted.

3 is a diagram illustrating a schematic configuration of a call control system of VoIP according to the present invention, FIG. 4 is a diagram illustrating an example of an option message according to the present invention, and FIG. 5 is a message including resource information according to the present invention. FIG. 6 is a diagram illustrating a parameter of a P-CAC-Codec header in the resource information message header of FIG. 5, and FIG. 7 is a diagram illustrating a media format among parameters of the P-CAC-Codec header of FIG. 6. It is a figure which shows an example.

As shown in FIG. 3, the present invention provides a plurality of first VoIP terminals 10a, 10b, 10c, and 10d, a first router 20, a proxy server 30, a second router 40, And a plurality of second VoIP terminals 50a, 50b, 50c, and 50d.

When a call connection request is generated from any one of the plurality of first VoIP terminals 10a, 10b, 10c, and 10d, the first router 20 supports the SDP (Session Description) supported by the terminal requesting the call connection. Protocol) The call connection request message including the protocol information is received and delivered to the proxy server 30.

When the proxy server 30 receives the call connection request message including the Session Description Protocol (SDP) protocol information, the media is set by referring to the <from> and <to> tags of the message. The source and destination IP addresses to be recognized are recognized.

Accordingly, the proxy server 30 may request an OPTIONS message for requesting resource information of the system from the router corresponding to the corresponding IP address, that is, the first router 20 and the second router 40. Send it.

That is, such an OPTIONS message is a message used to obtain resource information of a system, and a P-CAC-Destination-IP header is added to an existing message as shown in FIG. 4.

The added P-CAC-Destination-IP message header is IP address information for selecting a trunk interface to be used by the SIP aware router. For example, as shown in FIG. Destination-IP: 192.0.2.201 ".

In addition, the first router 20 and the second router 40 receiving the option message from the proxy server 30 are SIP-aware routers capable of interpreting the option message, and the IP address in the option message is occupied. By referring to the state of the trunk interface (LAN or WAN) to be transmitted, the response message to which the resource information available on the trunk interface is added is transmitted to the proxy server 30. At this time, the resource information includes information such as codec information and bandwidth.

That is, the response message to the option message is a message for providing resource information of the system, and the P-CAC-Codec header and the P-CAC-Bandwidth header are added to the existing message as shown in FIG. 5. It is done.

As such, the P-CAC-Codec message header added to the response message has codec information to be used in the corresponding trunk interface. Detailed parameters for this will have the format of <media>, <transport>, and <fmt list> as shown in FIG. In particular, the media format has a format as shown in FIG. 7. For example, the P-CAC-Codec message header is added as "P-CAC-Codec: audio RTP / AVP 0 4 18" as shown in FIG.

In addition, the P-CAC-Bandwidth message header has resource information available on the corresponding trunk interface. In this case, the unit uses bps. For example, the P-CAC-Bandwidth message header is also added as shown in FIG. 5 as "P-CAC-Bandwidth: 384000".

However, if the SIP-aware router cannot secure adequate resources, the requested call is rejected.

The proxy server 30 checks the resource information of the system transmitted from the first router 20 and the second router 40 and modifies the first received Session Description Protocol (SDP) protocol information.

In addition, the proxy server 30 transmits the modified Session Description Protocol (SDP) protocol information to the receiving terminal receiving a call of the calling terminal, and the receiving terminal receiving the call receives the modified SDP. (Session Description Protocol) Protocol information is transmitted to the calling terminal requesting a call connection.

Accordingly, the calling terminal requesting a call connection compares the original Session Description Protocol (SDP) protocol information with the modified Session Description Protocol (SDP) protocol information to determine an appropriate codec.

Accordingly, the transmitting side and the receiving side terminal set the real time transmission protocol (RTP) for the actual media transmission with the SDP information reflecting the network information to perform the transmission / reception.

8 is a diagram illustrating a configuration of a proxy server according to the present invention.

As shown in Fig. 8, the proxy server of the present invention includes an IP interface unit 31, a SIP protocol analysis unit 32, a call control unit 33, a resource information management unit 34, and a subscriber management unit 35. It is composed of

The IP interface unit 31 performs interfacing with the VoIP terminals.

The SIP protocol analyzer 32 analyzes the SIP protocol information transmitted between the VoIP terminals and transmits the SIP protocol information to the call controller 33.

When a call (CALL) is requested from any terminal, the call controller 33 connects or rejects the requested call according to whether system resources transmitted from the transmitting and receiving SIP-aware router are secured.

That is, if the resource information is secured from the SIP-aware router and the resource information of the system is transmitted through the corresponding message, the call controller 33 connects the requested call to the receiving terminal of the destination. -aware If the router receives a message that it cannot obtain adequate resources, it rejects the requested call.

The resource information manager 34 receives and manages resource information transmitted from the SIP-aware router. That is, at the request of the call controller 33, the resource information of the system transmitted from the SIP-aware router is checked, and the SDP information is corrected based on the checked resource information of the system and transmitted to the call controller 33. Done.

Accordingly, the call controller 33 controls the message including the modified SDP information to be transmitted to the receiving terminal through the IP interface 31.

The subscriber manager 35 manages the information of the system subscriber.

FIG. 9 is a diagram illustrating a process of a call control method using resource information of a BIO system according to the present invention, and FIG. 10 is a diagram illustrating a process when a call is rejected during the call control process of FIG. 9.

As shown in FIG. 9, when the proxy server 30 receives an INVITE message including Session Description Protocol (SDP) protocol information from the first user 10 (S10), the proxy server 30 may respond to the 100 Trying in response thereto. The first user 10 transmits the data (S20). That is, the INVITE message includes Original SDP information that can be supported by the first user 10.

Accordingly, after receiving the INVITE message, the proxy server 30 can refer to the <from> and <to> tags of the message to know the source and destination IP address where the media is set. Will be.

Subsequently, the proxy server 30 transmits an OPTIONS message for requesting resource information of the system to the first router 20 which is the sending side router and the second router 40 which is the receiving side router ( S30).

Accordingly, the first router 20 and the second router 40 that receive the option message from the proxy server 30 interpret the option message and the trunk interface (LAN or WAN) to which the IP address in the option message is occupied. In step S40, a "200 OK" response message in which resource information available in the corresponding trunk interface is added to the proxy server 30 is transmitted. At this time, the resource information includes information such as codec information and bandwidth.

Subsequently, the proxy server 30 receives a "200 OK" message and then checks resource information of the system (S50), and then first receives a Session Description Protocol based on the resource information (Resource). ) Modify the protocol information (Modification, S60).

Subsequently, the proxy server 30 transmits an INVITE message (INVITE +) to a receiving terminal that receives a call of the calling terminal (S70). In this case, the revised Session Description Protocol (SDP) protocol information (Resource aware SDP) is included in the INVITE message.

Subsequently, the second user 50 receives the requested call and transmits a "180 Ringing" message to the first user 10 (S80), and then transmits "200 OK" (S90). Here, the "200 OK" message includes modified Session Description Protocol (SDP) protocol information (SDP negotiated).

Accordingly, the first user 10 transmits an "ACK" message to the second user 50 in response (S100).

On the other hand, when the proxy server 30 sends the option message to request the resource information of the system, when the receiving SIP-aware router cannot secure the appropriate resources, as shown in FIG. In response to the call rejection (480 Temporary Not Available) message is transmitted to the proxy server 30 (S110), the proxy server 30 sends a "480 Temporary Not Available" message to the first user 10. Transmission (S120).

In the above, specific preferred embodiments of the present invention have been illustrated and described. However, the present invention is not limited to the above-described embodiment, and various modifications can be made by any person having ordinary skill in the art without departing from the gist of the present invention attached to the claims. will be.

According to the present invention, by selecting the appropriate codec reflecting the resource information of the trunk interface of the router constituting the network, the bandwidth of the entire network can be efficiently used, and the network By taking into account the resources or accepting or rejecting them properly, it is possible to improve the resources and performance of the system.

Claims (7)

In the call control system of the VoIP network, When a call connection request message including SDP protocol information is received from an arbitrary terminal, the network received from the router of the source and the destination after transmitting an option message for requesting resource information of the network system to the router of the source and the destination. The call control system using the resource information of the BIP network, comprising: a relay server controlling the call connection by modifying the SDP protocol information according to the resource information of the system and transmitting the modified SDP protocol information to a destination terminal. . The method of claim 1, The option message is a call control system using resource information of a BIO IP system, characterized in that it comprises IP address information for selecting a router trunk interface (Trunk Interface) of the destination. The method of claim 1, The resource information of the network system includes at least one of codec information and bandwidth information. The method of claim 1, The relay server, A call controller for controlling a call connection according to resource information of a network system received from a router of a source and a destination for the option message when a call connection request is generated from an arbitrary terminal; And a resource information management unit for checking the resource information of the network system and modifying the SDP protocol information in response to a request of the call control unit. The method of claim 4, wherein The call control unit, When the resource information of the network system is secured, the call is requested, and if the resource information of the network system is not secured, the call using the resource information of the VIP system is characterized in that the rejected request. Control system. In the call control method of the VoIP network, When a call connection request message including SDP protocol information is received from an arbitrary terminal, transmitting an option message for requesting resource information of a network system to a router of a source and a destination; Modifying the SDP protocol information according to resource information of a network system received from routers of the source and destination; And controlling the call connection by transmitting the modified SDP protocol information to a destination terminal. The method of claim 6, After the transmission of the option message, if the resource information of the network system is not secured, further comprising the step of rejecting the requested call.

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