KR20110023053A - Message data structure in system for transmitting the voice using internet network - Google Patents

Abstract

PURPOSE: A message data structure in a system for transmitting a voice using an internet network is provided to reduce the size of a packet, thereby analyzing the packet of the reduced size in a command analyzer. CONSTITUTION: A flag field includes information about message formats classified into a request and a response. A message field includes the information about the message formats. A from proxy field comprises a from IP field with ID information of a transmitting person of the message, a from IP field with IP information of the transmitting person, and proxy server IP information. A to proxy field comprises a to ID field with ID information of a receiving person of the message, a to IP field with IP information of the receiving person, and proxy server IP information.

Description

Message data structure in system for transmitting the voice using internet network

The present invention relates to a data structure, and more particularly, to a message data structure in a system for transmitting voice using an internet network.

As the Internet spreads widely today, various services using the Internet have emerged. Among these services, a voice communication service using a packet-based network such as the Internet, instead of a Switched Circuit Network (SCN) including a conventional Public Switched Telephone Network (PSTN), is also available. have. Voice over Internet Protocol (VoIP) service, which is one of voice communication services, is a system for transmitting voice using an Internet Protocol (IP) network. In addition to a simple telephone function, a multimedia function and various additional services are supported. VoIP requires a signaling protocol to find and establish a call, and ITU-T's H.323 and IETF's Session Initiation Protocol (SIP).

SIP is a protocol designed to make it easy to invite a party to a conference or telephone call rather than to control a user when a video conference or a telephone call is made. SIP is not extended to any protocol stack and is defined as text based such as HTTP (HyperText Transfer Protocol), so it is easy to extend and easy to use. SIP uses the existing e-mail address and provides mobility through location-directed messages, making it a simple protocol for receiving calls at designated locations.

However, a SIP data packet transmitted and received in such a system has a problem that a large field is unnecessary and thus a load is required to interpret the packet in the SIP command interpreter.

SUMMARY OF THE INVENTION The present invention has been made in an effort to provide a message data structure in which a size of a packet transmitted and received in a system for transmitting voice using an internet network is reduced.

According to an aspect of the present invention, in a message data structure transmitted and received in a system for transmitting voice using an internet network, a flag having information classified as a request and a reply is classified into a format of the message. Field, a Message field with information about the format of the message, a From ID field with ID information of the sender of the message, a From IP field with IP information of the sender, and proxy server IP information used by the sender. Provide a message data structure including a From Proxy field having a To ID field having an ID information of the recipient of the message, a To IP field having the recipient's IP information, and a To Proxy having proxy server IP information used by the recipient. do.

The message data structure may further include a field indicating a Voice over Internet Protocol (VoIP) used in a system for transmitting voice using the Internet network and a version thereof.

The message data structure may further include an Option field having additional information according to the message.

If the message data structure is a request in the flag field, the message format is INVITE, which is a message transmitted when starting a session, and a response message for the INVITE message is received. ACK replying to, BYE indicating that the call has ended in the server, CANCLE used to cancel a request that has been requested previously and not completed, OPTION used to request information about the server regardless of call setup And a register used to register its own address, IP address information, and the like.

In the message data structure, the format of the message in the existing message field is Login at Login, INVITE at User Invitation, OK at Accept Request, ACK at Login Success, NACK at Login Failure, BYE at End of Voice Transmission, At least one of 180 ringing when waiting for reception approval, 302 MovedTemp when changing the IP, and busy when the user transmits or receives a status may be at least one.

Reduce the size of packets in a system that transmits voice using internet network. The command interpreter interprets small packets and therefore takes less load.

As the invention allows for various changes and numerous embodiments, particular embodiments will be illustrated and described in the drawings. However, this is not intended to limit the present invention to specific embodiments, it should be understood to include all modifications, equivalents, and substitutes included in the spirit and scope of the present invention.

Terms including ordinal numbers, such as second and first, may be used to describe various components, but the components are not limited by the terms. The terms are used only for the purpose of distinguishing one component from another. For example, without departing from the scope of the present invention, the second component may be referred to as the first component, and similarly, the first component may also be referred to as the second component. And / or < / RTI > includes any combination of a plurality of related listed items or any of a plurality of related listed items.

When a component is referred to as being "connected" or "connected" to another component, it may be directly connected to or connected to that other component, but it may be understood that other components may be present in between. Should be. On the other hand, when a component is said to be "directly connected" or "directly connected" to another component, it should be understood that there is no other component in between.

The terminology used herein is for the purpose of describing particular example embodiments only and is not intended to be limiting of the present invention. Singular expressions include plural expressions unless the context clearly indicates otherwise. In this application, the terms "comprise" or "have" are intended to indicate that there is a feature, number, step, operation, component, part, or combination thereof described in the specification, and one or more other features. It is to be understood that the present invention does not exclude the possibility of the presence or the addition of numbers, steps, operations, components, components, or a combination thereof.

Unless defined otherwise, all terms used herein, including technical or scientific terms, have the same meaning as commonly understood by one of ordinary skill in the art. Terms such as those defined in the commonly used dictionaries should be construed as having meanings consistent with the meanings in the context of the related art and shall not be construed in ideal or excessively formal meanings unless expressly defined in this application. Do not.

DETAILED DESCRIPTION Hereinafter, exemplary embodiments will be described in detail with reference to the accompanying drawings, and the same or corresponding components will be given the same reference numerals regardless of the reference numerals, and redundant description thereof will be omitted.

1 is a configuration diagram of a SIP network providing a VoIP service.

Referring to FIG. 1, a SIP network includes SIP user agents 111 through 115 and a SIP server 110. Here, the user agent (UA) is a telephone apparatus or a terminal, which is a logical entity for initiating and terminating a SIP transaction by transmitting or receiving a message for establishing a call.

When the SIP UA (111 to 115) in the Internet network requests registration to the SIP server 110, the registration request message ('Resister' message) transmitted to the SIP server 110, the connection header ('Contact' header) It is included. The 'Contact' header has information on the address for the call connection of the SIP UA requested to register, and specifically includes the IP address, port, priority (Q value), and message transmission time information (Expire Time).

Accordingly, when the SIP server 110 performs a call connection by receiving a call connection request message ('INVITE' message) from the SIP UAs 111 to 115 in the Internet network, the IP address and port of the receiver from 'INVITE' are performed. Find and connect the arcs.

2 is a flowchart illustrating a message transmission method between SIP UAs.

Here, the user agent (UA) is a telephone apparatus or a terminal, which is a logical entity that initiates and terminates a SIP transaction by sending or receiving a message for call setup. 2, SIP UA1 is a message sender and SIP UA2 is a message receiver. The SIP UA1 transmits an INVITE message, which is a call connection request message, to the SIP UA2 (S210). Then, if the SIP UA2 receiving the INVITE message is busy, transmits a 180 ringing message (S220-1), and accepts the media session information and accepts the call connection, and transmits a 200 OK message (S220-2).

When the SIP UA1 receives the 200 OK message, it transmits an ACK message indicating that the 200 OK message has been received (S230). Then, the SIP UA1 and the SIP UA2 exchange the data by relaying the RTP message to perform voice transmission and reception (S240). The SIP UA2 transmits a BYE message at the end of voice transmission and reception (S250), and terminates when the SIP UA1 transmits a 200 OK message (S260).

3 is a diagram schematically illustrating a SIP message data structure transmitted and received in a SIP network. The SIP protocol generally uses UDP sockets and 5060 ports and follows the Ethernet frame format for the physical transmission of SIP message packets.

Referring to FIG. 3, when a packet of a SIP message is a UDP packet, the message data structure in the system for transmitting voice using the Internet network is an IP header 310, a UDP header 330, and a database unit 350. -1).

The IP header 310 is composed of a source IP part 310a having source IP information and a destination IP part 310b having destination IP information, and the UDP header 330. ) Is composed of a source port portion 330a having source port information and a receiving port portion 330b having destination port information. Substantial SIP messages are stored in the database 350-1.

The format of the SIP message stored in the database unit 350-1 has a format as shown in the figure, and includes a message type, a SIP version, Via, FROM, TO, Call-ID, CSeq, and Contact. The SIP message 350-1 is a case where the type of message is INVITE.

The Via field records the route taken, the FROM field contains the user (sender) ID information, the TO field contains the recipient ID information, and the Call-ID field as a unique identifier for grouping a series of messages. For each call, it must be the same for all requests and responses sent by the UA in the call and must be the same for each registration from the UA. The SIP UA has a means to ensure that the Call-ID field is not duplicated by another UA.

The CSeq field contains information identifying and sorting the transaction, the Contact field contains the UA's own address information that sends the SIP request message, and once the call is established, the destination UA needs the information needed to connect to the requesting UA. Have

The database unit 350-1 may contain other data 350-2 having a message type of 200 OK.

The present embodiment provides a message data structure that reduces the size of packets transmitted and received in a system for transmitting voice using an internet network. The data structure according to the present embodiment may be variously used in a system for transmitting voice using an internet network such as a UDP socket and a 5060 port or a TCP socket and a 10080 port.

4 is a diagram schematically illustrating a message data structure transmitted and received in a system for transmitting voice using the Internet network according to the present embodiment.

Referring to FIG. 4, the message data structure 400 is composed of Flag, SIP version, Message, From ID, From IP, From Proxy, To ID, To IP, To Proxy, Option.

The Flag field has information that classifies a message format into a request and a reply. If the message format is a request, it may contain a 'Q', and if the response is a 'R'. For example, the message format Request is INVITE, which is the message sent at the beginning of the session, ACK that receives the last response message for the INVITE message and responds to the response. BYE notifying that it has been terminated, CANCLE used to cancel a request that has not been completed yet, OPTION used to request information about the server regardless of call setup, and own address and IP address information are registered. There is a REGISTER to use.

Reply message types include Information message, 100th message 1XX, 182 and 200 OK messages indicating that the call has been queued.

The SIP version field indicates the VoIP and its version used in the system for transmitting voice over the Internet. The Message field is Login at login, INVITE at user invitation, OK at accept request, ACK at login success, NACK at login failure, BYE at end of voice transmission and reception, 180 Ringing at reception approval waiting, 302 MovedTemp when IP is changed, and user has Busy information when sending / receiving.

The From ID, From IP, and From Proxy fields contain the sender ID, IP, and proxy server IP information used by the sender. To ID, To IP, and To Proxy have the receiver's ID, IP, and proxy server IP information used by the receiver. The Option field has additional information according to the message.

In this case, the message data size is a limit not exceeding the value of the MTU for a network that knows the maximum transmission unit (MTU), and may be 1K bytes or less for a network whose MTU is unknown.

Compared with the SIP message data structure, in this embodiment, there is no Via field, which is a field for problem occurrence, and the FROM and TO fields are divided into fields having an ID field, an IP field, and proxy server IP information of the sender and receiver, respectively. . Also, the CSeq field is excluded because it can contain the same contents as the message field.

The specific numerical values used in the above-described embodiments are only for describing one embodiment of the present invention, and thus the content of the present invention is not limited to these specific numerical values.

Although the embodiments of the present invention have been described in detail above, the scope of the present invention is not limited thereto, and various modifications and improvements of those skilled in the art using the basic concepts of the present invention defined in the following claims are also provided. It belongs to the scope of rights.

All of the above functions may be performed by a processor such as a microprocessor, a controller, a microcontroller, an application specific integrated circuit (ASIC), or the like according to software or program code coded to perform the function. The design, development and implementation of the code will be apparent to those skilled in the art based on the description of the present invention.

While the present invention has been particularly shown and described with reference to exemplary embodiments thereof, it will be understood by those skilled in the art that various changes and modifications may be made therein without departing from the spirit and scope of the invention. You will understand. Therefore, the present invention is not limited to the above-described embodiment, and the present invention will include all embodiments within the scope of the following claims.

1 is a configuration diagram of a SIP network providing a VoIP service.

2 is a flowchart illustrating a message transmission method between SIP UAs.

3 is a diagram schematically illustrating a SIP message data structure transmitted and received in a SIP network.

4 is a diagram schematically illustrating a message data structure transmitted and received in a system for transmitting voice using the Internet network according to the present embodiment.

Claims (5)

In the message data structure transmitted and received in the system for transmitting voice using the Internet network, A flag field having information classifying the format of the message into a request and a reply; A Message field having information about the format of the message; A From ID field with ID information of the sender of the message, a From IP field with IP information of the sender and a From Proxy field with proxy server IP information used by the sender; and A To ID field having ID information of a receiver of the message, a To IP field having IP information of the receiver, and a To Proxy having proxy server IP information used by the receiver. The method of claim 1, And a field for indicating a Voice over Internet Protocol (VoIP) and a version thereof used in a system for transmitting voice using the Internet network. The method of claim 1, And a Option field having additional information according to the message. The method of claim 1, In the flag field If the format of the message is a request, the message format is INVITE, which is a message transmitted at the start of a session, an ACK that receives a response message for the INVITE message and responds to the response, and the server responds. BYE notifying that the call has ended, CANCLE used to cancel a previously requested request that has not yet been completed, OPTION used to request information about the server regardless of the call setup, and own address and IP address information. Message data structure, characterized in that at least one of the REGISTER used to register. The method of claim 1, In the message field The message format is Login at login, INVITE at user invitation, OK at accept request, ACK at login success, NACK at login failure, BYE at end of voice transmission, 180 Ringing, IP At least one of 302 MovedTemp at the time of change and Busy at the time of transmission and reception by the user.

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