JPH09312666A - Packet communication adaptor - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide the packet communication adaptor by which a buffering delay is reduced without decreasing the line utility efficiency in the case of voice communication by using a packet communication network. SOLUTION: A packet communication adaptor 3 contains a plurality of telephone terminals 1 and multiplexes voice signals from the telephone terminals 1 to assembly them as a single packet and sends the packet to a packet communication network 4. Furthermore, the adaptor 3 disassembles the multiplexed packet received from the packet communication network 4 and distributes the disassembled signals to each destination telephone terminal 1.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a packet communication adapter that enables a packet communication network to be applied to a communication mode such as telephone communication that requires bidirectional real-time communication between terminals. It relates to the device.
In particular, it is a technique aimed at eliminating the delay in packet assembly and minimizing the influence of the overhead caused by the packet header part.

[0002]

2. Description of the Related Art Telecommunications technology has been developed and developed mainly for the purpose of providing telephone services. On the other hand, in recent years, along with the development of computers, the spread of computer networks, which connect computers to each other via a telecommunication network, has been significantly expanding. In the computer network, a communication network represented by the Internet, which has a minimum unit of packets, should be used because of its high efficiency against traffic with burstiness peculiar to computer-to-computer communication and its high fault tolerance. There are many. A communication network in which packets are the smallest unit is called a packet switching network, as opposed to a circuit switching network represented by a telephone network.

In packet switching, a group of information called a packet is used as a minimum information transmission unit, and at least the header information in which the destination of the packet is written is attached to the head of each packet. A node device called a packet switch or router that configures the inside of a packet communication network refers to the header information of the packet transmitted from the user on the transmission side, and the packet received toward the adjacent node closest to the destination written in it. To transfer. As a result of repeating this operation, the packet reaches the final destination written in the destination, and communication between terminals is realized.

The information transmitted by the user (generally called the payload) is written in the part following the header information, and the receiving user discards the header part and interprets the payload part as the information from the transmitting user. To use. Generally, the header information is useless for the receiving user. Since the size of the header is constant regardless of the size of the payload, the larger the packet size, the larger the size of the payload in a single packet, and as a result, the bandwidth required by the packet communication network. The width (bit rate) is reduced and the transmission efficiency is improved. However, considering that retransmission processing occurs when a transmission error occurs, an extremely large packet size deteriorates transmission efficiency.

Further, in order to transmit a large packet, the terminal or node device must prepare a buffer corresponding to the packet size. Therefore, it is desirable that the size of each packet in the packet communication network be set as large as possible, but it is set to an optimum value in consideration of the influence of transmission error.

Communication performed between computers is typified by a file transfer or an electronic mail transfer, and is represented by a terminal-
The requirements for real-time communication between terminals are not strict. On the other hand, in telephone calls, since the voice signal is continuous, there is a severe demand for real-time performance between terminals.
For this reason, it is common to use a circuit switching system that guarantees a continuous bit stream for telephone calls.

The circuit switching service occupies a band even during a free time when data from a user is not actually transmitted, so that a data transfer request is generated in bursts, especially in computer communication. Inefficient when used for applications.

This is also the case when the application to telephone voice communication is considered, and the circuit switching occupies the band even when there is no voice. From this point of view, it is more efficient to use the network by transmitting only the voiced section using packet switching.

Considering the simultaneous transmission of n pairs of telephone conversations, circuit switching requires n times the line capacity of 64 Kbps required for a single telephone conversation. When packet switching is used, it is stochastically that n people talk at the same time, so the line capacity is less than n times 64 Kbps. ATM
(Asynchronous Transmission Mode) is a technology aimed at this statistical multiple effect.

The Internet, which has been explosively developing in recent years, is made up of a combination of a leased line, an ISDN network, etc., and its communication protocol is I.
It is based on P packets and can be classified as packet communication. On the Internet, the cost depending on time and distance does not burden the user at all, and similarly, when performing long distance communication, the cost is lower than the circuit switching service.

For the above reasons, there is an increasing demand from the user side to realize not only computer communication but also telephone communication on the packet communication network.

[0012]

When the factors causing the delay in voice packet communication are sorted out, there are a delay required for packet assembly (hereinafter referred to as a buffering delay) and a delay generated in a node device inside the packet network. .

The delay generated in the node device inside the packet network is improved by improving the performance of the node device and increasing the capacity of the line connecting the nodes. On the other hand, the buffering delay depends on the payload size of the packet and the amount of data generated by the transmitting user. The larger the amount of generated data and the smaller the payload size, the smaller the delay. For example, assuming transmission of telephone voice at 64 Kbps and the payload size of one packet is 1024 bytes, the buffering delay is 128 msec (10
24 * 8/64), but if the payload size is 128 bytes, this delay time is 16 msec.
It is shortened to (128 * 8/64). On the other hand, even if the payload size is 1024 bytes, assuming that 128 Kbps voice is transmitted, the buffering delay is 64 m.
It becomes sec (1024 * 8/128).

In telephone voice communication, if the delay becomes large,
Echo problem occurs. This is mainly because the sound reproduced from the handset acoustically wraps around the transmitter and reaches the original speaker. In the area where the transmission delay is small, the user cannot recognize the echo, but the effect becomes more significant as the delay increases. Furthermore, if the transmission delay increases, it takes an abnormally long time to receive a reply from the other party when one of the parties has finished speaking, making it very difficult to speak. This phenomenon occurs in telephone calls via satellite communication.

In order to improve the quality of telephone calls in a packet communication network, it is necessary to make the delay as small as possible. As described above, the buffering delay cannot be avoided once the payload size of the packet and the data generation rate are determined. Therefore, in order to shorten the buffering delay, the payload size will be reduced. In fact, an AT designed from the beginning for voice communication
In M, the payload size is set as small as 48 bytes.

However, as described above, when the payload size is reduced, the influence of the header overhead increases and the transmission efficiency decreases. For example, in the case of transmitting a voice compressed to 8 Kbps using a voice compression algorithm so that the buffering delay amount is within 5 msec, the payload size needs to be set to 5 bytes.

The header size of a general IP packet is 2
Since it is 0 bytes, the packet size is 25 bytes (2
0 + 5), which must be sent every 5 msec, so the required bandwidth is 40 Kbps (25
[Bytes] × 8 [bits / bytes] / 5 [msec]), which means that the effect of audio compression cannot be fully utilized. In particular, assuming that n users use this service at the same time, the required bandwidth when the statistical multiplexing effect is ignored is n × 40 Kbps.

[0018]

SUMMARY OF THE INVENTION In order to solve the above-mentioned problems, the present invention provides a means for accommodating a plurality of terminal devices (telephone terminals) and a signal input from these terminal devices. It has a means of assembling as one packet and transmitting it to the packet communication network. Further, it has means for disassembling the packet received from the packet communication network and distributing the disassembled signal to each terminal device.

As a result, it is possible to substantially increase the input data generation rate at the time of packet assembly, and it is possible to reduce the buffering delay without reducing the packet payload size.

For example, 10 telephones using voice compressed to 8 Kbps have a buffering delay of 5 msec.
It is assumed that they will communicate at the same time. For this purpose, 10 slots of 5 bytes, which is the amount of information generated by one terminal device in 5 msec, are prepared in the payload of the packet. That is, the payload size is 50 bytes (5
[Bytes / phone] x 10 [phone]). In the packet communication adapter device, a voice signal of 5 msec from each telephone is stored in each slot, a header is added, and then the packet is transmitted to the packet communication network.

As described above, the IP packet header is 20
Since it is bytes, the size of the entire IP packet at this time is 7
It becomes 0 bytes (20 + 50). As a result, the required bandwidth is 112 Kbps (70 [bytes] × 8 [bits / bytes] / 5 [msec]), and the bandwidth occupied by one telephone set is 11.2 Kbps. This is because the bandwidth required for each telephone is 4 when transmitting 8 Kbps voice with a buffering delay of 5 msec without using the present invention.
It is about 1/4 compared to 0 Kbps.

Although the quality of voice transmitted over the telephone network is almost constant at present, the required telephone voice quality may change in the recent telecommunications where user needs are diversified. is there. Specifically, there are cases where low-quality voice of about 8 Kbps is acceptable, and cases where high-quality voice communication of 128 Kbps or higher is required.

In order to meet such needs, it is necessary to adapt the size of the slot prepared when signals from a plurality of terminal devices are multiplexed in a single packet.
That is, when transmitting with a buffering delay of 5 msec, it is necessary to prepare a slot of 5 bytes for a terminal of 8 Kbps and a slot of 80 bytes for a terminal of 128 Kbps. Therefore, it has a function of flexibly changing slot allocation in response to a request from a terminal device and transmitting multiplexing information, that is, information about the configuration of a slot, to a packet communication adapter device of a communication partner. I will.

[0024]

BEST MODE FOR CARRYING OUT THE INVENTION An embodiment of the present invention will be described below with reference to the drawings. FIG. 1 is a block diagram showing the configuration of an embodiment of the present invention.

In FIG. 1A, reference numerals 1 and 2 denote telephone terminals,
3, 3'denotes a packet communication adapter device, and 4 a packet communication network. As is clear from FIG. 1 (a), a plurality of telephone terminals 1 and 2 are connected to one packet communication adapter device 3 or packet communication adapter device 3 ', and telephone terminal identification numbers are assigned to them. Has been done. Here, for convenience, each telephone terminal 1 (1-1, 1-2, ..., 1-
The telephone terminal identification number of p) is [1-1], [1-2],
, [1-p], and each telephone terminal 2 (2-1, 2-
2, ..., 2-q) telephone terminal identification number is [2-1],
It is expressed as [2-2], ..., [2-q].

The telephone terminals 1 and 2 and the packet communication adapter devices 3 and 3'may be connected directly or may be connected via a public telephone network. In addition, at least two packet communication adapter devices 3,
3'is connected via the packet communication network 4, and each packet communication adapter device 3, 3'is given an address in the packet communication network 4.

FIG. 1B is a diagram showing an internal configuration example of the packet communication adapter device 3. 31 in FIG. 1 (b)
Is a telephone terminal connection unit, 32 is a codec, 33 is a packet assembly / disassembly unit, 34 is a packet network connection unit, and 35 is a control unit. The internal configuration of the packet communication adapter device 3'is also the same.

The telephone terminal connection units 31 are provided as many as the telephone terminals that can be simultaneously connected to the packet communication adapter device 3, and have a function of recognizing a dial signal sent from the telephone terminal, an incoming signal to the telephone terminal, and It has the function of sending ringback tones and the function of sending and receiving voice signals to and from telephone terminals. When the telephone terminals 1 and 2 and the packet communication adapter devices 3 and 3'are connected via a public telephone network, the telephone terminal connecting unit 31 recognizes an incoming signal received from the public telephone network and recognizes it. It must have both the function of responding to the call and the function of sending a dial signal to the public telephone network.

The codecs 32 are provided in the same number as the telephone terminal connection units 31, and have a function of converting an analog voice signal received from the telephone terminal into a digital signal in real time, and conversely, a digital signal into an analog voice signal in real time. And has a function of transmitting the data to the telephone terminal via the telephone terminal connection unit 31.

It is assumed that the codec 32 can select a method of digitizing an analog signal in response to a request from the telephone terminal. Specifically, each bit rate of 16K, 32K, 64Kbps can be selected in response to a request from the telephone terminal.

The packet assembling / disassembling unit 33 is connected to the plurality of codecs 32, collects the digitized voice signals from the respective codecs 32 and stores them in the payload portion, and at the same time, the header information that the packet communication network 4 can recognize. Packet is added to assemble the packet, and the packet network connection unit 34
And the packet network connection unit 34 decomposes the packet received from the packet communication network 4 to extract the payload portion and transfers the signal therein to the codec 32.

The packet network connection unit 34 transfers the packet delivered from the packet assembling / disassembling unit 33 to the packet communication network 4
And a function of receiving a packet addressed to itself from the packet communication network 4 and delivering it to the packet assembling / disassembling unit 33.

The operation of this embodiment will be described below with reference to FIGS. In any one of the telephone terminals 1, the user performs a calling operation. Here, it is assumed that the source is the telephone terminal 1-1 and the destination is the telephone terminal 2-1 of the telephone terminals 2 having the telephone terminal identification number [2-1].

The telephone terminal 1-1 sends the telephone terminal identification number [2-1] of the telephone terminal 2-1 which is the transmission destination to the packet communication adapter device 3 by a dial signal, and at the same time requests a voice quality of 16K, Specify from 32K and 64Kbps. When the telephone terminal 1-1 and the packet communication adapter device 3 are connected via the public telephone network, the telephone terminal 1-1 first connects the packet communication adapter device 3 to the public telephone network. The established telephone number is specified to establish a connection with the packet communication adapter device 3, and then the telephone terminal identification number [2-1] of the telephone terminal 2-1 which is the transmission destination is packet-transmitted by a dial signal. While sending to the adapter device 3, at the same time request voice quality 1
Specify from 6K, 32K, 64Kbps.

In the packet communication adapter device 3, the telephone terminal connection unit 31 receives this dial signal and the control unit 35
The packet communication adapter device 3'in which the telephone terminal 2-1 whose destination is designated is accommodated is discriminated, and the packet communication adapter device 3'from the packet network connection unit 34 is connected to the corresponding packet communication adapter device 3 '. A connection for packet communication is established between them.

After the connection is established in the packet communication, the packet communication adapter device 3 is connected to the packet communication network 4
The signal is used to notify the packet communication adapter device 3 ′ of the telephone terminal identification number [2-1] of the telephone terminal 2-1 designated as the destination by the user, and the telephone making the call. The telephone terminal connection unit 31 connected to the terminal 1-1 sends a ringback tone to the telephone terminal 1-1.

FIG. 2 (a) shows the contents of the packet sent from the packet communication adapter device 3 to the packet communication adapter device 3'at this time. Upon receiving this packet, the packet communication adapter device 3'transmits an incoming signal to the telephone terminal 2-1 designated as the destination.
The telephone terminal 2-1 and the packet communication adapter device 3 '
If is connected to the public telephone network via the public telephone network, the packet communication adapter device 3'designates the telephone number of the telephone terminal 2-1 designated as the destination to the public telephone network. An incoming signal is transmitted from the network to the telephone terminal 2-1 designated as the destination.

When the user of the telephone terminal 2-1 responds to this incoming call, the incoming signal which has been sent to the telephone terminal 2-1 until then is stopped, and the response packet shown in FIG. , Is transmitted from the packet communication adapter device 3 ′ to the packet communication adapter device 3. Upon receiving this packet, the packet communication adapter device 3 stops the ringback tone being sent to the telephone terminal 1-1, and starts a call between the telephone terminals 1-1 and 2-1.

As shown in FIG. 2B, in this response packet, the slot number information used by the packet communication adapter device 3'to store the transmission voice signal and the size information of the slot. Is included. Further, the packet communication adapter device 3 also sends a response packet to the packet communication adapter device 3 ', and specifies the slot number and its slot size used for storing the transmitted voice signal therein.

In the calling operation of the telephone terminal 1-1,
Assuming that a voice quality of 16 Kbps is requested, if the buffering delay needs to be 5 msec, the size of one slot is 10 bytes (16 [Kbps] × 5 [m
sec] / 8 [bit / byte]). Telephone terminal 1-
When 1 specifies the voice quality of 32K and 64Kbps in the transmission operation, the slot sizes are 20 and 40, respectively.
Become a part-time job.

FIG. 3 (a) shows the contents of packets exchanged between the packet communication adapter devices 3 and 3'during a call between the telephone terminals 1 and 2. Here, a slot can be virtually specified for the payload.

The voice of the talker is converted into an electric analog voice signal by an acoustoelectric converter normally included in the telephone terminal 1-1 and sent to the packet communication adapter device 3. In the packet communication adapter device 3, the codec 32 digitizes the analog voice signal, and the packet assembling / disassembling unit 33 stores the continuous digital signal in the slot (slot number 0).

When only one of the telephone terminals 1 connected to the packet communication adapter device 3 is busy, when a voice signal is stored in one slot of 10 bytes, this packet is connected to the packet network. It is sent to the packet communication network 4 via the unit 34. That is, one packet is sent to the packet communication network 4 every 5 msec.

The packet assembling / disassembling unit 33 in the packet communication adapter 3'which receives this packet sends the digital voice signal of one slot in the payload to the codec 32, and the codec 32 converts this into an analog signal. This analog signal is sent to the telephone terminal 2-1 through the telephone terminal connection unit 31, and is converted into an acoustic signal by the electroacoustic transducer normally included in the telephone terminal 2-1.
It is delivered to the user's hearing.

During this call, a call start request is made between a pair of the telephone terminals 1 and 2 other than the telephone terminals 1-1 and 2-1 such as the telephone terminal 1-3 and the telephone terminal 2-4. There was. Also at this time, in the same procedure as described above, the transmission of the incoming signal to the telephone terminal 2-4 on the receiving side and the transmission of the ring back tone to the telephone terminal 1-3 on the transmitting side are performed by the respective packet communication adapter devices 3, 3. 3 '.

When the user responds to the incoming call, the packet communication adapter devices 3 and 3'indicate the slot number and slot size used for storing the transmission signal together with the response message. At this time, the slot number specified is different from the slot used in the call because there is an ongoing call. Here, the slot number is assumed to be 1.

FIG. 3B shows the contents of packets exchanged between the packet communication adapter devices 3 and 3'when two calls are present at the same time. The voice signal between the telephone terminals 1-1 and 2-1 is stored in slot 0, and the voice signal between the telephone terminals 1-3 and 2-4 is stored in slot 1. In this example, the telephone terminal 1
-3 and the telephone terminal 2-4, a size of slot 1 is set to 40 bytes on the assumption that a voice call of 64 Kbps is made.

FIG. 4 shows the contents of the slot number table managed by the packet assembling / disassembling unit 33 of the packet communication adapter device 3 at this point. The packet assembling / disassembling unit 33
The voice signals received from the two codecs 32 connected to the telephone terminal 1-1 and the telephone terminal 1-2 via the telephone terminal connection unit 31 are collected for 5 msec and are stored in the slot number table shown in FIG. The packets are stored in the determined slots, and they are transmitted from the packet network connection unit 34 to the packet communication network 4 every 5 msec. As for the voice on the receiving side, the slots of the packet to be received are disassembled every 5 msec and distributed to each codec 32 as determined by the slot number table shown in FIG.

In FIG. 3 (c), when a call request is further generated between the telephone terminals 1 and 2, and there are r pairs of calls, the packet communication adapter device 3 and the packet communication adapter device 3'are shown. It shows the contents of the packets exchanged between them.

A disconnection request from the telephone terminals 1 and 2 during a call is realized by exchanging a special disconnection request packet between the packet communication adapter devices 3 and 3 ', as in the case of starting a call. When this disconnection request is generated, the slot number table managed by the packet assembling / disassembling unit 33 is rewritten accordingly.

When a human usually has a conversation, a silent portion is included in a considerable proportion. Since it is originally not necessary to transmit information regarding the silent portion, in such a case, it is possible to temporarily set the size of the slot allocated for transmission of the telephone terminal in the silent state to zero. . As a result, further improvement of transmission efficiency is realized.
In order to reduce the transmission amount by not transmitting the silent portion, the codec 32 has a function of detecting the silence, and when the silence is detected, it is transmitted to the packet assembling / disassembling unit 33, and the packet assembling / disassembling unit 33 In the unit 33, it is possible to adopt a method of designating a slot number for temporarily setting the size to zero in the payload portion of the packet.

As described above, using the packet communication network,
In particular, when a plurality of users simultaneously perform voice communication, it is possible to reduce the buffering delay without degrading the line utilization efficiency.

FIG. 5 is a graph showing how much the required bandwidth (bit rate) can be reduced by the present invention. The horizontal axis of the graph is the number of terminals communicating simultaneously, and the vertical axis is the required bandwidth (bit rate). As a premise of this graph, the voice signal generated by each telephone terminal is 8 Kbps, the size of the packet header is 20 bytes, and the buffering delay is 5 msec.

In FIG. 5, the bandwidth required when the present invention is used, and the packet size is determined so as to simply realize a low buffering delay without using the present invention, the communication is performed. The required bandwidth is plotted. As is clear from FIG. 5, as the number of terminals communicating at the same time increases, the required bandwidths differ, and the advantage of the present invention becomes clear.

Further, since each packet simultaneously transmits the configuration information of the slots virtually provided inside the payload, it becomes possible to accommodate terminal devices that generate various different bit rates.

[0056]

As described above, by using the present invention, a plurality of packets can be stored under a single header information, especially when a plurality of users simultaneously perform voice communication by using a packet communication network. Since the signals from the terminal devices can be stored and transmitted collectively, the influence of the overhead caused by the header information is suppressed, and the line utilization efficiency is not reduced,
It is possible to reduce the buffering delay. In addition, the signal generation rate from each terminal can be set for each terminal.

[Brief description of drawings]

FIG. 1 is a block diagram showing a configuration according to an embodiment of the present invention.

FIG. 2 is a diagram showing the contents of a packet transmitted from a packet communication adapter device when making a call or making a response between telephone terminals.

FIG. 3 is a diagram showing contents of a packet transmitted between packet communication adapter devices during a telephone call of a telephone terminal.

FIG. 4 is a diagram showing an example of a slot number table managed by a packet assembling / disassembling unit.

FIG. 5 shows how much bandwidth (bit rate) is required when a plurality of telephone terminals communicate via a packet communication network, with and without the present invention. It is a figure of a graph showing whether a difference occurs.

[Explanation of symbols]

 1, 2 Telephone Terminal 3, 3'Packet Communication Adapter Device 4 Packet Communication Network 31 Telephone Terminal Connection Section 32 Codec 33 Packet Assembly / Disassembly Section 34 Packet Network Connection Section 35 Control Section

 ─────────────────────────────────────────────────── ─── Continuation of the front page (72) Inventor Shoji Takano 3-19-2 Nishishinjuku, Shinjuku-ku, Tokyo Inside Nippon Telegraph and Telephone Corporation (72) Takeshi Ikeda 3-19-3 Nishishinjuku, Shinjuku-ku, Tokyo No. 2 within Nippon Telegraph and Telephone Corporation

Claims (3)

[Claims] 1. A packet communication network for performing communication using a packet as a minimum communication unit, and a terminal device operated by a user of the packet communication network, and packetizing a signal generated by the terminal device into the packet. It is possible to accommodate a plurality of terminal devices in a packet communication adapter device that sends out to a communication network, decomposes a packet received from the packet communication network, restores a signal, and outputs it to a terminal device that accommodates it. Means, a means for assembling the signals input from the respective terminal devices and then assembling them as a single packet, and a means for separating the signals of the single received packet and distributing them to a plurality of terminal devices. And multiplexes the signals from the plurality of terminal devices into a single packet, and outputs the packet to the packet communication network. A packet communication adapter device having a function of allocating a packet received from a communication network to a plurality of accommodated terminal devices after the packet is disassembled. 2. The packet communication adapter device according to claim 1, wherein the means for assembling the signals input from the respective terminal devices into a single packet is a signal input from each terminal device. A packet communication adapter device having means for deciding a band to be allocated to each terminal device in response to a request from each terminal device when multiplexing the packets. 3. The packet communication adapter device according to claim 1, further comprising means for adding information regarding a signal multiplexing method to a packet to be transmitted to the packet communication network. Adapter device.