JP2000270024A - Method for exchanging capability of frame packet processing size in internet phone, terminal utilizing internet phone and medium recording program of internet phone - Google Patents

Abstract

PROBLEM TO BE SOLVED: To allow both terminals to normally reproduce compressed voice data and to efficiently send/receive voice data even when the processing capability of them differs by exchanging an available voice coding system and a processible frame packetting size between both terminals in the case of exchanging the capability. SOLUTION: At start of communication, a terminal 1 transmits a call packet to an opposite terminal to inform the opposite terminal of a desirable speech. A frame packetting size capability exchange section 15 transmits information denoting all processible frame packetting sizes with respect to a coding system reproducible by the own terminal 1 to the opposite terminal in a form of a capability exchange packet at that time or after that. The opposite terminal also transmits information denoting all processible frame packetting sizes with respect to a coding system reproducible by the own terminal to the terminal 1 in a form of a capability exchange packet. After the mutual capability exchange, an operating frame packetting size decision section 18 identifies the received packet and decides the coding system and the frame packetting size that are to be in use.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an Internet telephone terminal for performing two-way real-time data communication over an IP network such as the Internet. The present invention relates to a method for exchanging frame packetization size capability in an Internet telephone for exchanging with each other, an Internet telephone utilizing terminal device, and a recording medium recording an Internet telephone program.

[0002]

2. Description of the Related Art In an Internet telephone, in order to reduce the amount of data flowing on a network, voice data is generally encoded (= compressed), packetized, and transmitted to the other party.

FIG. 14 shows an example of a basic processing sequence in an Internet telephone. As shown in FIG. 14, the audio data transmitting side inputs an analog audio signal from a microphone, converts the analog audio signal into a digital signal by A / D conversion, compression-encodes it by a predetermined encoding method, and further converts it into packets. To send. The receiving side of the audio data assembles the received packets in the transmission sequence, expands the data by decoding, converts the data into an analog signal by D / A conversion, and reproduces and outputs the audio.

[0004] Internet telephone programs and their devices that realize such processing functions differ in the encoding system implemented for each product. Therefore, in an Internet telephone, at the start of communication, the coding schemes of both parties are exchanged, the other party recognizes the reproducible coding scheme, and encodes and transmits using the reproducible coding scheme. As a system for this purpose, ITU-T H.264, an international standard system for Internet telephony, is used. 323 /
H. The H.245 system is typical.

[0005] In speech coding, input speech data is coded in units of a certain amount (= frame). The base value of the frame size differs depending on the coding system. ing. For example, G.10 compresses audio data to 6.3 kbps or 5.3 kbps. 72
30ms (milliseconds) in the 3.1 system, 8k audio data
G. compress to bps In the 729 system, it is 10 ms.

However, since the number of packets to be transmitted and received increases as the frame size becomes smaller, it is necessary to perform a higher-speed data transfer process. Depending on the processing performance of a personal computer on which an Internet telephone program is installed, the time is too short. With a frame size, processing cannot catch up, processing becomes impossible, and sound interruption may occur.

In such a case, if speech is encoded and packetized for each multiple of the base frame size serving as a base, the number of transmitted packets can be reduced, so that a personal computer with low capacity can be processed. In some cases, For example, G. In the case of the 723.1 system, it is 60 ms, 90 ms, and 120 ms, which are multiples of 30 ms. A value that is a multiple of the base frame size is called a frame packetization size. If the frame packetization size is 120 ms, four frames of G.I.
The compressed audio data of the 723.1 system is transmitted to the other party as one audio data packet.

[0008] FIG. This is an example of the frame packetization size in the 723.1 system, in which encoding and packetization are performed using the base frame size. FIG. 15B shows an example in which the frame packetization size is 120 ms and the audio data for four frames is transmitted as one packet.

[0009] On the receiving side, on the other hand, H.264 added to the audio data is used. 323 / H. By calculating the difference between the timestamp values in the RTP (Realtime Transport Protocol) header specified by the 225.0 method, the number of milliseconds in which the transmitting side has encoded and packetized the frame packetization size , And the received data can be reproduced with the correct frame packetization size. FIG. 16 shows a timestamp value in the RTP header.

However, the aforementioned H. 323 / H. 225.
In the 0 method, there is no means for exchanging the capacity of the frame packetization size that both terminals can process.
A compressed data packet having a frame packetized size when the transmitting terminal has been compressed and packetized is unilaterally transmitted. As a result, conventionally, as shown in FIG. 17, for example, as shown in FIG. 17, when the processing performance on the transmission side is high, but the processing performance on the reception side is low, the reception side cannot process the encoded packet data received in the frame packet size. However, there is a problem that sound cannot be reproduced correctly.

In the Internet telephone, when packetizing compressed audio data, packetization is performed by adding header data for a communication protocol. However, regardless of the size of the compressed audio data, the header size is fixed. Therefore, as the frame packetization size is shorter, a larger number of packets having a smaller size are created, and the overhead of the header increases. Therefore, even when the same encoding is used, the amount of data flowing through the network increases.

FIG. 18 shows the difference in the amount of transmission data depending on the frame packetization size. For example, as shown in FIG. When data of 120 ms in base frame size is encoded and packetized by the 723.1 system, the amount of transmission data is 328 bytes and the bit rate of transmission data is about 22 kbps for 4 frames. In contrast, the frame packetization size is encoded at 120 ms.
In the case of packetization, as shown in FIG.
The transmission data amount is 154 bytes, and the bit rate of the transmission data is about 10 kbps.

On the other hand, in an actual Internet communication line, since the band (communication speed) differs depending on the type of line, use of an Internet telephone in an environment with a low communication speed may be considered. Further, even with the same communication line, the communication speed may temporarily decrease, for example, when the network is crowded. In such a case, the frame packetization size is increased to reduce overhead,
Although it is effective to reduce the amount of transmission data, there is a problem that such control cannot be performed because there is no means for exchanging frame packetization sizes with each other.

[0014]

SUMMARY OF THE INVENTION In order to solve the above-mentioned problems, the present invention can be used between both terminals at the time of exchanging abilities at the start of communication or at the time of exchanging abilities during communication. Exchanges frame packet sizes that can be processed together with various audio coding methods, enables normal playback of compressed audio on both terminals even with different processing capabilities, and enables efficient transmission and reception of audio data It is intended to be.

[0015]

In the present invention, the Internet telephone terminal has means for recording the coding capability of the terminal itself, and has means for recording the frame packetization size of the coding capability. Then, the terminal making the Internet call (calling terminal) carries information indicating the coding capability of the terminal itself and the frame packetization size corresponding to the coding capability in the call packet or the capability exchange packet. Means for recognizing information indicating the coding capability of the other terminal in the incoming call packet or the capability exchange packet and the size of the frame packet corresponding to the coding capability, and coding used for communication between the two terminals. Means for determining the frame packetization size for the encoding scheme and the encoding scheme.

The terminal (called terminal) that can make an Internet telephone call has a coding capability of the partner terminal in a call packet or a capability exchange packet from the partner and a frame packetization size corresponding to the coding capability. Means for recognizing the information indicating the communication terminal, means for mounting the information indicating the encoding capability of the own terminal and the frame packetization size corresponding to the encoding capability in the incoming call packet or the capability exchange packet for reply, It has means for determining an encoding system used during communication between the devices and a frame packetization size for the encoding system.

[0017]

FIG. 1 is a block diagram of an Internet telephone terminal according to an embodiment of the present invention. In the figure,
Reference numeral 1 denotes a terminal equipped with a program for Internet telephone (hereinafter referred to as Internet telephone software), 2 denotes a microphone for voice input, and 3 denotes a speaker for voice output.

The device control section 11 is a module for controlling the entire terminal, and is realized by an operating system or the like when a personal computer is used as the terminal. Network interface unit 12
Is a part for transmitting and receiving packets via a network. The communication control unit 13 converts communication data into packets according to a predetermined protocol in order to communicate with other terminals via the network interface unit 12 and the network. This is the part that performs control. The Internet telephone function / control unit 14 is a main module of Internet telephone software for realizing Internet telephone.
Since the details of this part are the same as in the prior art,
Detailed description here is omitted.

Frame packetization size capability exchange unit 15
Has frame packetized size information mounting means 16 and frame packetized size information recognizing means 17. The frame packetized size information mounting means 16 includes the encoding capability of its own terminal and the encoding capability in the packet to the destination terminal. Is loaded with information indicating the frame packetization size corresponding to. Also, the frame packetization size information recognition means 17
Recognizes information indicating the coding capability of the partner terminal and the frame packetization size corresponding to the coding capability in the packet received from the partner terminal.

Used frame packetizing size determining unit 18
Recognizes an encoding method and a frame packetization size that can be reproduced by a partner terminal, and determines an encoding method and a frame packetization size to be used at the time of transmission. The frame packetization size recording unit 19 is a means for recording an encoding method that can be used by the terminal at the time of voice communication and a frame packetization size that can be transmitted and received in the encoding method.

The voice input section 20 is a section for inputting a voice analog signal input from the microphone 2, and the A / D conversion section 21 converts the input analog signal into a digital signal. The audio encoding unit 22 compression-encodes the digitized audio data. The compression-encoded audio data is sent to the communication control unit 13, packetized with the packetization size determined by the used frame packetization size determination unit 18, and transmitted.

When the communication control unit 13 receives a packet of audio data, the audio data is decoded by an audio decoding unit 23 and converted into an analog audio signal by a D / A conversion unit 24. The analog audio signal is output to the audio output unit 2
The sound is output to the speaker 3 via the speaker 5 and the sound is reproduced.

FIG. 2 shows an example of a capability exchange sequence of the Internet telephone according to the present embodiment. FIG.
In the figure, it is shown that the terminal A originates a call. First, a call packet is transmitted from the terminal A to the terminal B, and the terminal B returns a call packet in response to the call packet. Thereafter, the terminal A and the terminal B mutually transmit and receive the capability exchange packet, and start a voice call.

FIG. 3 shows a flow of determining a frame packetization size in a terminal according to the present embodiment.

At the start of communication, when a destination number is input from the terminal, a call packet is transmitted to the destination terminal to inform the destination terminal that a call is desired. At that time or thereafter, all the frame packets that can be processed for the reproducible coding scheme of the own terminal are transmitted in a capability exchange packet, and the other terminal also can process all the frame packets that can be processed for the reproducible coding scheme. Information indicating the packetized size is transmitted as a capability exchange packet, and the capability exchange of frame packetized sizes that can be processed mutually is performed (step S1).
0, S20).

After the capability exchange, the received packet is identified, and the coding method to be used and the frame packetizing size are determined. That is, a frame packetization size that can be processed by the partner terminal is selected (step S
30). The audio data is transmitted using the frame packetization size for the determined encoding method (step S40).

FIG. 4 shows an example of the capability exchange packet used in the present embodiment. The packet identifier in the packet indicates the type of the packet, such as whether the packet is an outgoing call packet, an incoming call packet, or a capability exchange packet. Also, the message type indicates that this packet notifies the receiving capability. In this example, when the receivable encoding method is A, the receivable frame packetization sizes are 30 ms, 60 ms, and 90 ms. When the receivable encoding method is B, the receivable frame packetization size is 40 ms. 60 ms and 80 ms.

Each terminal determines an encoding method and a frame packet size to be used at the time of transmission based on the capability information received from the other party, and uses the encoding method and the frame packet size at the time of voice communication. I do. In the terminal 1 of FIG. 1, the packetizing means of the communication control unit 13 receives the size determined by the used frame packetizing size determining unit 18 under the control of the Internet telephone function / control unit 14, Make The information of the capability exchange packet may be included in an outgoing call packet or an incoming call packet.

As described above, by exchanging the capability between the reproducible coding method and the processable frame packetization size, it becomes possible to receive compressed audio data with the reproducible frame packetization size. .

As the terminal 1 for the Internet telephone described above, a personal computer in which Internet telephone software is installed, a terminal connecting device for connecting a general analog telephone, a facsimile machine, or the like to a network can be considered. An Internet telephone program for realizing each of the above means by a computer constituting a terminal can be stored in an appropriate recording medium such as a portable medium memory, a semiconductor memory, and a hard disk which can be read by the computer.

[0031]

[First Embodiment] FIG. 5 is a block diagram showing a first embodiment of the present invention. The present embodiment is an example in the case of exchanging the capacity of the frame packet size at the start of a call.

A data communication terminal A101 and a data communication terminal B111 are connected on a network (LAN) 105. A microphone 102, a speaker 103, and Internet telephone software 104 are installed in the data communication terminal A101, and a data communication terminal B111 is installed in the data communication terminal B111. Microphone 112,
A speaker 113 and Internet telephone software 114 are installed, and each constitutes an Internet telephone terminal.

The data communication terminals A101 and B111
Here, it is assumed that the Internet telephone software 104, 114 is running. Also, the IP address of the data communication terminal A101 is “129.60.10.11”, and the data communication terminal B1
The IP address of No. 11 is “129.60.10.12”.

At the data communication terminal A101, the user performs a calling operation using the Internet telephone software 104 to make a call to the data communication terminal B111. Data communication terminal A
Reference numeral 101 denotes a transmission packet including information on a receivable encoding system and its frame packetization size registered in advance in its own device as shown in FIG.

The data communication terminal B111, which is the destination,
Upon receiving a call packet from the data communication terminal A101,
Emit a signal indicating that there is an incoming call. When the data communication terminal B111 performs an incoming call operation, the information on the receivable coding method and its frame packet size registered in advance in its own device is included in the incoming call packet as shown in FIG. To the data communication terminal A101.

The data communication terminal A101 and the data communication terminal B111 compare the reception capability of the partner terminal with the transmission capability of the own terminal, and the coding method that can be used in common by both terminals is G.100. 723.1, and the usable frame packetization size is 60 ms and 90 ms.

The data communication terminals A101 and B111 are connected to the LAN 105, recognize that there is no need to be aware of the amount of communication data, and use a frame packetization size of 60 ms, which has a smaller delay compared to 90 ms. Decide to do it. As a result, both terminals determine the encoding scheme to be used as G.264. Audio data is transmitted / received to / from each other in a 723.1 system and a frame packetization size of 60 ms, and both terminals can normally reproduce compressed audio.

[Second Embodiment] FIG. 8 is a block diagram showing a second embodiment of the present invention. The present embodiment is an example of a case where the capability exchange of the frame packet size is performed after the call connection is established.

A data communication terminal A 201 and a data communication terminal B 211 are connected on a network (LAN) 205, a microphone 202, a speaker 203, and Internet telephone software 204 are installed in the data communication terminal A 201, and a data communication terminal B 211 is installed in the data communication terminal B 211. Microphone 212,
A speaker 213 and Internet telephone software 214 are installed, and each constitutes an Internet telephone terminal.

The data communication terminals A201, B211
Here, it is assumed that the Internet telephone software 204, 214 is running. The IP address of the data communication terminal A 201 is “129.60.10.11”, and the data communication terminal B 201
The IP address of No. 11 is “129.60.10.12”.

At the data communication terminal A 201, the user performs a calling operation using the Internet telephone software 204 to make a call to the data communication terminal B 211. The data communication terminal B211 that is the destination receives the call packet from the data communication terminal A201, and issues a signal indicating that there is an incoming call. When the data communication terminal B211 performs an incoming call operation, it sends an incoming packet to the data communication terminal A201 of the call originator, establishes a call connection, and then receives the reception packet registered in advance from the data communication terminal A201 in its own device. Information on the possible encoding schemes and their frame packetization sizes is transmitted in the capability exchange packet as shown in FIG.

Similarly to the data communication terminal A201, the data communication terminal B211 which is the communication partner uses the information on the receivable coding method and its frame packet size registered in advance in its own device as shown in FIG. Send in the exchange packet. The data communication terminal A201 and the data communication terminal B211 compare the reception capability of the partner terminal with the transmission capability of the own terminal, and the coding method that can be used in common by both terminals is G.264. 723.1, and recognizes that the available frame packetization size is 60 ms.
As a result, both terminals determine the encoding scheme to be used as G.264.
723.1 system, frame packetization size 60ms
Thus, audio data is transmitted / received to / from each other, and both terminals can normally reproduce compressed audio.

[Third Embodiment] FIG. 11 shows a third embodiment of the present invention.
FIG. 3 is a configuration diagram showing an example of the embodiment. This embodiment is an example of a case where the capability of the frame packet size is exchanged during a call and the frame packet size is changed again.

The microphone 30 is connected to the data communication terminal A301.
2. Speaker 303, Internet telephone software 304
Is installed, and a microphone 312, a speaker 313, and Internet telephone software 314 are installed in the data communication terminal B 311 to constitute Internet telephone terminals. The data communication terminal A301 has a modem 307, and the data communication terminal B311 has a modem 31.
7 is connected, and a dial-up connection is made to a telephone network (Internet provider) 305 via a modem. The communication speed of both modems is 28.8 kbps. Further, the data communication terminal A301 has a communication data amount measuring unit 306 for measuring the amount of data during communication.

In the data communication terminals A301 and B311, the Internet telephone software 30
4 and 314 are activated, both are connected to the Internet provider 305 via the modem, and each is acquiring an IP address. Data communication terminal A30
1 is “129.60.10.11”, and the IP address of the data communication terminal B 311 is “129.60.10.12”.

At the data communication terminal A301, the user performs a calling operation using the Internet telephone software 304 to make a call to the data communication terminal B311. The data communication terminal B311 that is the other party receives the call packet from the data communication terminal A301 and issues a signal indicating that there is an incoming call. When the data communication terminal B 311 performs an incoming call operation, it sends out an incoming call packet to the data communication terminal A 301 of the call originator, establishes a call connection, and then receives data from the data communication terminal A 301 registered in advance in its own device. Information on a possible encoding method and its frame packetization size is transmitted in a capability exchange packet as shown in FIG.

Similarly to the data communication terminal A301, the data communication terminal B311 that is the other party uses the information of the receivable coding method and its frame packet size registered in advance in its own device as shown in FIG. Send in the exchange packet. The data communication terminal A301 and the data communication terminal B311 compare the reception capability of the partner terminal with the transmission capability of the own terminal, and the coding method that can be commonly used by both terminals is G.300. 723.1, and recognizes that the usable frame packetization size is 30 ms.
As a result, both terminals determine the encoding scheme to be used as G.264.
723.1 system, frame packetization size 30 ms
Thus, audio data is transmitted / received to / from each other, and both terminals can normally reproduce compressed audio.

G. When the frame packetization size is 30 ms in the 723.1 system, the communication data amount is about 22 kbps. Therefore, even in the communication band (28.8 kbps) in the case of the dial-up connection by the modem, there is no data loss.
Transmission and reception of voice can be performed normally. However, when another data communication is performed during the communication and the effective communication band is reduced to 15 kbps, the G.264 communication is performed. Data having a size of 723.1 / frame packetization size of 30 ms cannot be transmitted completely, and sound loss due to data loss occurs. The communication bandwidth measurement unit 306 of the data communication terminal A detects this decrease in the communication band.

The communication data amount measuring unit 306 that has detected that the effective communication band has dropped to 15 kbps notifies the Internet telephone software 304. Upon receiving this notification, the Internet telephone software 304 sends the G.P. By changing the frame packetization size of the 723.1 system to only 120 ms, the data communication terminal B311
Send to The data communication terminal B311 that has received the capability exchange packet with the frame packetization size changed,
The capability exchange packet of the own terminal (FIG. 13) is transmitted again. As a result, the mutual capability exchange is performed again, and the common encoding method and framing size of both are set to G.264. 72
The 3.1 format / frame packetization size becomes 120 ms, and voice communication is resumed in this state. Since the communication band in this communication mode is about 10 kbps, even if the effective communication band is 15 kbps, there is no data loss, and voice transmission and reception can be performed normally.

In the embodiment of the present invention described above, milliseconds are used as the unit of the receivable frame packetization size, but the present invention can be similarly applied when the number of frame packets is used.

[0051]

As described above, according to the present invention, when the capability is exchanged at the start of communication or at the time of capability exchange during communication, the present invention can be used together with the speech coding method usable between both terminals. It is possible to exchange possible frame packetization sizes, and it is possible to efficiently transmit and receive audio data without occurrence of sound interruption or the like.

[Brief description of the drawings]

FIG. 1 is a block diagram of an Internet telephone terminal according to an embodiment of the present invention.

FIG. 2 is a diagram showing an example of a capability exchange sequence.

FIG. 3 is a diagram illustrating a flow of determining a frame packetization size in a terminal.

FIG. 4 is a diagram illustrating an example of a capability exchange packet.

FIG. 5 is a configuration diagram showing a first embodiment of the present invention.

FIG. 6 is a diagram illustrating an example of a call packet of a data communication terminal A;

FIG. 7 is a diagram illustrating an example of a call packet of a data communication terminal B;

FIG. 8 is a configuration diagram showing a second embodiment of the present invention.

FIG. 9 is a diagram showing an example of a capability exchange packet of the data communication terminal A.

FIG. 10 is a diagram showing an example of a capability exchange packet of the data communication terminal B.

FIG. 11 is a configuration diagram showing a third embodiment of the present invention.

FIG. 12 is a diagram showing an example of a capability exchange packet of the data communication terminal A.

FIG. 13 is a diagram showing an example of a capability exchange packet of the data communication terminal B.

FIG. 14 is a diagram showing an example of a basic processing sequence in an Internet telephone.

FIG. It is a figure which shows the example of the frame packetization size in 723.1 system.

FIG. 16 shows a time stamp (times) in an RTP header.
It is a figure which shows a tamp) value.

FIG. 17 is a diagram illustrating an example of a case where frame packetization sizes that can be processed by terminals are different, for explaining a problem of the related art.

FIG. 18 is a diagram illustrating a difference in transmission data amount depending on a frame packetization size.

[Explanation of symbols]

 DESCRIPTION OF SYMBOLS 1 Terminal 2 Microphone 3 Speaker 11 Device control unit 12 Network interface unit 13 Communication control unit 14 Internet telephone function / control unit 15 Frame packet size exchange unit 16 Frame packet size information mounting unit 17 Frame packet size information recognition unit 18 Used frame packetization size determination unit 19 Frame packetization size recording unit 20 Audio input unit 21 A / D conversion unit 22 Audio encoding unit 23 Audio decoding unit 24 D / A conversion unit 24 25 Audio output unit

 ──────────────────────────────────────────────────続 き Continuing on the front page (72) Inventor Yasuhito Hayashi 3-19-2 Nishi-Shinjuku, Shinjuku-ku, Tokyo Japan Telegraph and Telephone Corporation (72) Inventor Hiroshi Otani 3- 192-1 Nishi-Shinjuku, Shinjuku-ku, Tokyo No. Nippon Telegraph and Telephone Co., Ltd. F-term (reference) CC06 CC08 EE04 HH15 JJ25

Claims (9)

[Claims] 1. A method for exchanging frame packetization size capability in an Internet telephone in an IP network in which a plurality of terminals are accommodated each having an IP address, the method comprising: The process of exchanging the capabilities of the available coding systems and frame packetization sizes, the process of determining the coding system and frame packetization size to be used at the time of transmission based on the information, and the processes of the determined coding system and frame Compressing and packetizing voice data at a packetized size and transmitting a transmission data packet. The method for exchanging frame packetized size capability in an Internet telephone. 2. An Internet telephone using terminal device in an IP network in which a plurality of terminals are accommodated with respective IP addresses. Means for recording a proper frame packetization size, means for exchanging the receivable coding method and frame packetization size with the partner terminal at the start of communication, and a coding method and frame capable of being reproduced by the partner terminal. Means for recognizing the packetization size and determining the encoding method and frame packetization size to be used for transmission, means for transmitting voice packets using the determined encoding method and frame packetization size, and control of these means Means for performing Internet telephone use. 3. A recording medium storing a program for realizing an Internet telephone system in an IP network in which a plurality of terminals are accommodated with respective IP addresses, wherein the terminal can be used during a voice call. Recognizes the encoding method and the frame packet size that can be transmitted and received in the encoding method, and exchanges the encoding method and the frame packet size possessed by both at the start of communication so that the partner terminal can reproduce. The computer recognizes the encoding method and the frame packetizing size, determines the encoding method and the frame packetizing size to be used at the time of transmission, and executes a process of transmitting a voice packet using the encoding method and the frame packetizing size to the computer. Recording an Internet telephone program characterized by recording The recording medium on which it was recorded. 4. A method for exchanging frame packet size capability in an Internet telephone in an IP network in which a plurality of terminals are accommodated with respective IP addresses, the method comprising the steps of: The process of exchanging the capabilities of the available coding systems and frame packetization sizes, the process of determining the coding system and frame packetization size to be used at the time of transmission based on the information, and the processes of the determined coding system and frame Compressing and packetizing voice data at a packetized size and transmitting a transmission data packet. The method for exchanging frame packetized size capability in an Internet telephone. 5. An Internet telephone using terminal device in an IP network in which a plurality of terminals are accommodated with respective IP addresses. Means for recording an appropriate frame packetization size, means for exchanging the receivable coding scheme and frame packetization size with a partner terminal after a call connection is established, and a coding system capable of reproducing the partner terminal And means for recognizing the frame packetization size and determining the encoding method and frame packetization size to be used at the time of transmission, means for transmitting a voice packet using the determined encoding method and frame packetization size, and these means Controlling means for controlling an Internet telephone. 6. A recording medium for recording a program for realizing an Internet telephone system in an IP network in which a plurality of terminals are accommodated with respective IP addresses, wherein the own terminal can be used during a voice call. It recognizes the encoding method and the frame packet size that can be transmitted and received in the encoding method, exchanges the encoding method and the frame packet size of both after the call connection is established, and enables the partner terminal to reproduce. The computer recognizes a certain encoding system and frame packetization size, determines an encoding system and a frame packetization size to be used at the time of transmission, and transmits a voice packet using the encoding system and the frame packetization size to a computer. Internet telephone program characterized by recording the program to be executed The recording medium on which it was recorded. 7. A method for exchanging frame packetization size capability in an Internet telephone in an IP network in which a plurality of terminals are accommodated with respective IP addresses, the method being usable between terminals during Internet telephone communication. The process of exchanging the coding system and frame packetization size capability, the process of determining and changing the coding system and frame packetization size to be used for transmission based on the information, and the process of changing the coding system and frame Compressing and packetizing voice data at a packetized size and transmitting a transmission data packet. The method for exchanging frame packetized size capability in an Internet telephone. 8. An Internet telephone terminal device in an IP network in which a plurality of terminals are accommodated with respective IP addresses. Means for recording an appropriate frame packetization size, means for exchanging the receivable coding scheme and frame packetization size with a partner terminal during communication, and coding scheme and frame capable of being reproduced by the partner terminal. Means for recognizing the packetization size and determining and changing the coding method and frame packetization size to be used at the time of transmission; means for transmitting voice packets using the changed coding method and frame packetization size; Controlling means for controlling an Internet telephone. 9. A recording medium for recording a program for realizing an Internet telephone system in an IP network in which a plurality of terminals are accommodated with respective IP addresses, wherein the terminal can be used during a voice call. Recognizes the encoding method and the frame packet size that can be transmitted and received in the encoding method, exchanges the encoding method and the frame packet size possessed by both during communication, and enables the partner terminal to reproduce. Recognize the encoding method and frame packet size, change the encoding method and frame packet size used for transmission,
A recording medium storing a program for an Internet telephone, wherein a program for causing a computer to execute a process of transmitting a voice packet with the changed encoding method and frame packetization size is recorded.