JPH1198560A - Speech system among plural slave sets in digital cordless telephone system - Google Patents

Abstract

PROBLEM TO BE SOLVED: To surely attain the service of direct speech among plural slave sets without increasing traffic of a radio speech channel and to allow a slave set to call other slave sets in the radio zone simultaneously with respect to the speech system among the plural slave sets. SOLUTION: Each of plural slave sets is provided with a control means 1 that selects itself to be in the main slave set mode or in the sub slave set mode and executes communication in the selected mode, an A/D converter means 2 that convert a transmission voice signal into voice data with plural coding speeds respectively, and a D/A converter means 3 that converts the received voice data with a corresponding to coding speed into a voice signal. The main slave set uses a call signal of a control channel to inform the operating condition of an incoming voice transmission slot and a coding speed of transmission data depending on number (n) of sub-slave sets to each sub-slave set, and each sub-slave set sends incoming voice data with the informed coding speed by an informed incoming voice transmission slot.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a communication system between a plurality of slave units in a digital cordless telephone system. 2. Description of the Related Art In recent years, digital cordless telephone systems that provide convenience of portable telephones that can be easily handled in homes, offices, and outdoors have been put into practical use. In particular, PHS (P
For the Personal Handy phone System), the standard for the communication procedure is “RCR-ST” by the Association of Radio Industries and Businesses.
D28 ". This PHS is realized by combining digital mobile communication technologies such as adoption of a 32k ADPCM as a standard audio coding system and adoption of a TDMA system. It should be noted that a half rate (16 kADPCM) and a quarter rate (8 kADPCM) for a full rate of 32 kADPCM are not specified as an audio coding method. This PHS
Is characterized by good call quality, low call charges, intensive use in office environments, and long continuous operation time with battery driving power. For this purpose, PHS services are provided by a plurality of telecommunications carriers. FIG. 32 is a diagram showing a basic configuration of the PHS. In FIG. 32, a plurality of radio base stations 41, 41,... Are connected to a public network 43 via a mobile exchange 42, respectively. In the wireless zone formed by each wireless base station 41, the mobile units PS1 to PS1 are provided.
PSn is arranged. In the PHS having such a configuration, each of the slave units PS1 to PSn is provided with a business mode in which each of the mobile devices PS1 to PSn is operated as a digital cordless telephone or a business cordless terminal installed in a business.
A public mode that operates as a simple mobile phone terminal that can access the public network in the outdoors, underground shopping malls, and other public spaces, and a direct communication mode between handset units that enables calls with other mobile stations without going through a wireless base station Having.

[0002]

2. Description of the Related Art FIG. 33 is a block diagram showing a configuration of a conventional slave unit. In FIG. 33, a conventional slave unit includes a transmission / reception antenna 9, a transmission / reception switch (SW) 10, a transmission unit 11, a reception unit 12, a modulation unit 13, a demodulation unit 14, a TDMA control unit 15, Encoding unit 16, decoding unit 17, A / D
A converter 18, a D / A converter 19, a microphone 20, a speaker 21, a control unit 22b, a RAM 23, a ROM 24 and an EEPROM 25 connected to the control unit 22b, an external terminal interface unit 26, a liquid crystal display ( LC
D), and a human interface unit 27 having a light emitting display unit (LED) and a keyboard.

[0003] The control unit 22b receives a command from the external terminal interface unit 26 and designates the ROM 24 or EEP according to the designation of the office mode, the public mode, or the direct communication mode between the slave units.
The corresponding program is taken out from the ROM 25 and the TDMA control unit 15 is controlled to realize communication in each mode. At this time, the information input from the keyboard of the human interface unit 27 is reflected on the control and the like.
The required display is performed on a CD or LED.

[0004] The encoder 16 converts a digital audio signal (64 kPCM) input from the A / D converter 18 into 32 kA
The data is converted into voice coded data of DPCM and output to the TDMA control unit 15. The decoding unit 17 includes the TDMA control unit 15
32k ADPCM voice encoded data input from
The signal is converted into a digital audio signal of kPCM and output to the D / A converter 19.

Hereinafter, referring to FIGS. 34 and 35, RCR-
A direct communication system between slave units specified in STD 28 will be described. FIG. 34 is a diagram illustrating a TDMA frame configuration of the PHS. FIG. 35 shows a sequence of a direct communication method between slave units defined by RCR-STD28. PHS TDM
As shown in FIG. 34, the A frame is 5 m in length.
s, and each frame has a downlink transmission / reception area “T” in the first half in order to enable full-duplex communication.
(r) "and the latter half of the uplink transmission / reception area" R (t) ". That is, when two opposing slave units 1 and 2 communicate, the opposite slave unit 1 transmits to the opposite slave unit 2 at "T (r)", and the opposite slave unit 2 opposes at "R (t)". Send to slave 1.

Specifically, each frame is time-divided into eight time slots (communication channels). That is, the downlink transmission / reception area “T” and the uplink transmission / reception area “R” are each configured with four time slots. In FIG. 34,
For convenience of explanation, these time slots recognized by the originating slave unit are “T1” to “T4”, respectively.
“R1” to “R4” are given and shown, and conversely, those time slots that are recognized in the same manner by the receiving handset are “r1” to “r4” and “t1” to “t”, respectively.
4 ".

The time width of one time slot is 62
5 μs. That is, the transmission information for 5 ms is 625
It is compressed to μs and transmitted in bursts. The elapsed time of the four time slots is 2.5 ms. That is, 2.5 ms from the slot where the downlink transmission data was received.
Uplink transmission data is transmitted in a later slot. As described above, one call uses one slot therein at full duplex, but in the direct communication mode between handset units, the "carrier for direct call between handset units" Several waves are reserved in advance as carriers to be assigned.

FIG. 35 shows a sequence of a communication procedure in the case where, for example, the portable unit PS1 among the portable units PS1 to PSn makes a call to the portable unit PS2 and makes a direct call. Slave units PS1, P
In S2, the mode shifts to the direct communication mode between the slave units based on the operator's instruction input from the external terminal interface unit 26. Asynchronously with this mode transition control, the slave unit PS2 supplies drive power to the receiving unit 12 via the TDMA control unit 15, and performs continuous reception for the above-mentioned several waves of "carriers for direct communication between slave units". (FIG. 35 (1)).

[0009] On the other hand, when a call is made in response to an instruction given by the operator, handset PS1 transmits eight time slots T1.
To T4, R1 to R4, a time slot not captured by any other call is obtained based on the electric field strength of the “carrier for direct communication between slave units” and the like (hereinafter, this time slot is referred to as “time slot”). Rk "), and transmits a calling signal (including the identification information of the destination handset PS2) during the time slot Tk corresponding to the time slot Rk under the above-described frame configuration (FIG. 35).
(2)).

This transmission trial is performed in order to secure a time period in which the opposite handset called handset PS2 can determine whether or not a paging signal has been received for all “direct call carriers between handset”. Up to 1 until a synchronization signal described later is received
Repeated for 0 seconds. In the slave unit PS1, the receiving unit 12 is driven at a timing capable of performing a receiving operation in a time slot Rk separated by 2.5 ms corresponding to a half of a frame with respect to each of the above-described time slots Tk.

[0011] Then, when the slave unit PS2 receives the above-mentioned calling signal (FIG. 35 (3)), the slave unit PS1 which starts from that point in time transmits the signal transmitted to the slave unit PS2 next. The time slot Rk period to be received is determined,
The synchronization signal is transmitted during the reception time slot Rk (FIG. 35 (4)). That is, since such a synchronization signal is transmitted under the synchronization that the slave unit PS2 autonomously takes with respect to the time when the slave unit PS1 transmits, the slave unit PS1
Can be received smoothly without further synchronization. Then, when the slave unit PS1 can receive and recognize the synchronization signal,
A synchronization signal indicating this is transmitted in the time slot Tk of the subsequent frame in the same manner as the above-mentioned calling signal (FIG. 35 (5)).

When the slave unit PS2 can receive and recognize the synchronization signal, it drives hardware (not shown) (ringer drive) to output a ringtone (FIG. 35 (6)).
Prompt for off-hook. When the operator answers, the slave unit PS2 stops outputting the ringtone (FIG. 35 (7)) and sends a response signal in the time slot Rk in the subsequent frame (FIG. 35 (8)). On the other hand, slave unit PS1
Receives the response signal described above, and transmits and receives an idle burst with the slave unit PS2 (FIG. 35 (9)).

Thus, PS2 can capture the time slot tk in the subsequent frame and the above-mentioned time slot rk (Tk) as a full-duplex communication channel. Tk and time slot Rk can be captured as a full-duplex communication channel, and each transitions to a communication state (FIG. 3).
5 (10)).

In the handsets PS1 and PS2 during a call, the reception signal received by the transmission / reception antenna 9 is input to the reception unit 12 via the SW 10, and is amplified and input to the demodulation unit 14. The demodulation unit 14 demodulates the reception data of the time slot Rk detected by the reception unit 12 into 32 k ADPCM data. This demodulated 32k ADPCM data is
64 kPC in the decoding unit 17 via the TDMA control unit 15
The data is converted into M data, further converted into an analog audio signal by the D / A converter 19, and output from the speaker 21.

Further, the analog audio signal taken in from the microphone 20 is converted into 64 kPCM data by the A / D converter 18, then converted into 32 kADPCM data by the encoding unit 16, and converted into the time slot Tk by the TDMA control unit 15. The signal is inserted, modulated by the modulation unit 13, and transmitted via the transmission unit 11, the SW 10, and the transmission / reception antenna 9.

As described above, in the PHS, two opposing slave units can directly communicate with each other without going through a radio base station. On the other hand, assuming a case where three or more persons have a meeting, the possibility of direct communication between a plurality of handset units in a PHS has been examined and proposed (for example, Japanese Patent Application Laid-Open No. H8-182045). . In this publication, four examples are given as examples, but the following is an example.

First, the child device 1 and the child device 2 make one time slot (for example, T1, R1 in FIG. 33 viewed from the child device 1).
Then, enter the call. As a result, a direct call between the slave units 1 and 2 between the slave units using the time slots T1 and R1 is realized. Next, the handset 1 uses another empty slot (for example, T3 and R3 in FIG. 34 viewed from the handset 1) to enter a call with the handset 3. As a result, direct communication between the slaves 1 and 3 using the time slots T3 and R3 is realized.

The handset 1 mixes the input sound, the upstream sound of the handset 2 and the upstream sound of the handset 3, and
By transmitting a downstream voice signal to the slave unit 3 using the time slot T3 and transmitting a downstream voice signal to the slave unit 3 using the time slot T3, a direct call between the slave units is realized.

[0019]

As described above, in the direct communication method between a plurality of slave units introduced in the above publication, a communication channel is assigned to each slave unit. If the traffic of the communication line is increased and a plurality of communication channels cannot be captured, a service of direct communication between a plurality of slave units cannot be provided.

Further, in the direct communication method between handset units specified by the RCR-STD 28, since a call between one handset unit is premised, one handset unit simultaneously calls another handset unit in the wireless zone. Can not do.

It is an object of the present invention to provide a direct call service between a plurality of slave units without increasing the traffic of a wireless communication line, and a certain slave unit to simultaneously call another slave unit in the wireless zone. It is an object of the present invention to provide a communication system between a plurality of cordless handsets in a digital cordless telephone system capable of performing the above.

[0022]

FIG. 1 is a block diagram showing the principle of the first to fifth aspects of the present invention. FIG. 2 shows an operation sequence according to the first and fifth aspects of the present invention. FIG.
Is an operation sequence diagram according to the second and fifth aspects of the present invention. FIG. 4 is an operation sequence diagram according to the third and fifth aspects of the present invention. FIG. 5 is an operation sequence diagram according to the fourth and fifth aspects of the present invention. 2 to 4 show a call connection sequence between the main slave (PSm) and the two sub slaves (PS1, PS2).

According to the first aspect of the present invention, in a digital cordless telephone system,
A control means for switching and setting the own handset between a main handset mode and a sub handset mode, and executing communication in the set mode; A / D conversion means 2 for converting each of the audio data having the encoding speed of the above, and D / A for converting the received audio data having the corresponding encoding speed into the audio signal
The control means 1 of the slave unit in the main slave unit mode uses the paging signal of the control channel to determine the use condition of the uplink voice transmission slot and the number n of the slave units in the sub slave unit mode.
(N = 1, 2, 3,...) To notify the coding rate of the transmission data to the predetermined number n of the slave units in the sub-slave mode, and transmit the received uplink voice data in the corresponding downlink transmission slot. Then, the control units 1 of the sub-units in the sub-unit mode of the predetermined number n control the corresponding A / D conversion unit 2 in accordance with the notified encoding speed, and transmit the uplink audio data to the predetermined number. It is formed at the coding rate and is transmitted in the notified uplink voice transmission slot.

That is, according to the first aspect of the present invention, as shown in FIG. 2, the control means 1 of PSm transmits a paging signal of the control channel to PS1 and PS2 (11). This paging signal includes the use condition of the uplink voice transmission slot and the designation of the encoding speed (1/2 encoding speed) of the transmission data determined by the number n = 2 of the slave units in the sub-slave mode. The use condition of the uplink voice transmission slot is a condition that the two sub-child units use the use time slot in each unit frame of the TDMA frame in what order and at what timing. In the illustrated example, the used time slot is the first slot, PS1 is specified to be transmitted in the first cycle (first frame), third cycle (third frame), etc., and PS2 is specified in the second cycle. Eyes (second frame), fourth
This shows the case where transmission is specified in the lap (fourth frame).

Then, PS1 and PS2 transmit a response signal of the control channel to PSm to make PSm confirm that the transmission is in the time slot specified in (12) and (13).
Next, PS1 transmits uplink audio data at a 1/2 coding rate in the first cycle of the first slot (14). In response, the PSm transmits downlink audio data at the standard coding rate in the first slot (15).

Similarly, PS2 transmits uplink audio data at 1/2 coding rate in the second cycle of the first slot (1).
6). In response to this, PSm transmits downlink audio data at the standard coding rate in the first slot (17). That is, since the two sub-child units can transmit the uplink audio data once every two frames, the audio signals are compressed to half of the standard encoded data and transmitted. Then, the downlink audio data is simultaneously received by both PS1 and PS2.

As described above, a connection in which a plurality of slave units can share a single time slot and perform simultaneous communication can be realized by a simple communication procedure. According to a second aspect of the present invention, in the digital cordless telephone system, each of the plurality of slave units is set by switching its own slave unit between the main slave unit mode and the sub slave unit mode, and executes communication in the set mode. A / D converter 2 for converting a transmission audio signal into audio data of a plurality of encoding speeds up to a standard encoding speed as a system, and a corresponding encoding speed received And D / A conversion means 3 for converting the audio data into an audio signal, and mixing control means 4 for mixing the audio input of the slave unit with the output of the D / A conversion means 3, Control means 1 uses the paging signal of the control channel to determine the use condition of the uplink voice transmission slot and the number n (n = 1, 2,
The encoding speed of the transmission data determined by 3,.
To the slave unit in the sub-slave unit mode, and controls the D / A conversion means 2 and the mixing control means 4 to mix the audio signal of the own slave unit into a signal obtained by converting the received upstream audio data into an audio signal. Controlling the A / D conversion means 2 to form transmission audio data having a standard coding rate as a system among a plurality of coding rates and transmitting the mixed audio signal in a corresponding downlink transmission slot, Each of the control units 1 of the slave units in the sub-slave mode forms the uplink audio data to be transmitted by controlling the A / D conversion unit 2 according to the notified coding speed at a predetermined coding speed, and Transmission in the specified uplink audio transmission slot, and reception of a mixed audio signal transmitted by the child device in the main child device mode.

That is, according to the second aspect of the present invention, as shown in FIG. 3, the control means 1 for PSm transmits a paging signal of the control channel to PS1 and PS2 (11). This paging signal includes the use condition of the uplink voice transmission slot and the designation of the encoding speed (1/2 encoding speed) of the transmission data determined by the number n = 2 of the slave units in the sub-slave mode. The use condition of the uplink voice transmission slot is a condition that the two sub-child units use the use time slot in each unit frame of the TDMA frame in what order and at what timing. In the illustrated example, the used time slot is the first slot, PS1 is specified to be transmitted in the first cycle (first frame), third cycle (third frame), etc., and PS2 is specified in the second cycle. Eyes (second frame), fourth
This shows the case where transmission is specified in the lap (fourth frame).

Then, PS1 and PS2 transmit the response signal of the control channel to PSm, and (12) and (13) cause PSm to confirm that the transmission is in the specified time slot.
Next, PS1 transmits uplink audio data at a 1/2 coding rate in the first cycle of the first slot (14). In response to this, the PSm converts the upstream audio data received from the PS1 into an audio signal, mixes the audio signal of the own device with the audio signal, and supplies the mixed audio signal to the A / D conversion means 2 to perform standard encoding. The transmission voice data of the speed is formed and transmitted in the corresponding downlink transmission slot (first slot) (25).
This is simultaneously received by PS1 and PS2.

Similarly, PS2 transmits uplink audio data having a 1/2 coding rate in the second cycle of the first slot (1).
6). In response to this, the PSm converts the upstream audio data received from the PS2 into an audio signal, mixes the audio signal of the own unit with the audio signal, and supplies the mixed audio signal to the A / D conversion means 2 to perform standard encoding. The transmission voice data of the speed is formed and transmitted in the corresponding downlink transmission slot (first slot) (27). This is simultaneously received by PS1 and PS2.

Also, PSm is, depending on the timing,
The audio signal of the slave unit is mixed with the audio data converted from the uplink audio data received from the PS2 and the audio signal converted from the uplink audio data received from the PS2, and the mixed audio signal is converted into an A / D signal. It may be provided to the conversion means 2 to form transmission voice data at a standard coding rate and transmit it in the corresponding downlink transmission slot (first slot). This is likewise received simultaneously by PS1 and PS2.

As described above, since a plurality of slave units can share a single time slot and perform a simultaneous call, it is possible to reduce the traffic on the radio line. According to a third aspect of the present invention, in the digital cordless telephone system according to the first or second aspect, in the communication method between a plurality of slave units, the control unit 1 of the slave unit in the main slave mode determines that n is already in communication. Notifying one slave unit of a change in the use condition of the uplink voice transmission slot and the encoding speed of the transmission data on the out channel of the downlink voice channel, and for one new slave unit by the out channel of the downlink voice channel. A paging signal including a notification of the use condition of the uplink voice transmission slot and the encoding speed of the transmission data is transmitted, and n
The control means 1 of one slave unit performs control to shift to a call at the coding rate as instructed in accordance with the change notification, and the control means 1 of one new slave unit responds according to the notified coding rate. A / D conversion means that controls the A / D conversion means to perform transmission, forms uplink audio data to be transmitted at a predetermined coding rate, and transmits the data in the notified uplink audio transmission slot.

That is, the invention described in claim 3 is based on claim 1
Alternatively, the present invention relates to a communication procedure for newly subscribing a slave unit after a call between a plurality of slave units is started by the invention according to claim 2. For example, as shown in FIG. 4, when PSm and PS1 are initially communicating at a standard coding rate (31), PSm first transmits upstream speech to outgoing outgoing voice channel of PS1 during the speech. The use condition of the transmission slot and the change of the encoding speed of the transmission data are notified at the standard encoding speed (32).

On the other hand, PS1 transmits the uplink audio data at the 1/2 encoding speed in the first cycle of the first slot as instructed (33). That is, PS1 shifts to a call at the encoding speed as instructed in accordance with the change notification. And
When the PSm can confirm the transition control of PS1, the PSm transmits a paging signal including the notification of the use condition of the upstream voice transmission slot and the coding speed of the transmission data to the newly added PS2 on the out channel of the downstream voice channel ( 34). On the other hand, PS2 transmits the uplink audio data at the 1/2 encoding speed in the second cycle of the first slot as instructed (3.
Five).

As a result, the call is changed from a two-party call to a three-party call. In response to this, the PSm transmits downlink voice data at a standard coding rate in the first slot (36).
This is received simultaneously by PS1 and PS2. In this way, a change from a two-party call to a three-party call can be made by a simple connection procedure. According to a fourth aspect of the present invention, in the digital cordless telephone system according to the first or second aspect, in the communication method between a plurality of slave units, the control unit 1 of the slave unit in the main slave mode determines that n is already in communication. Executing the call termination sequence for one of the child devices that has finished the call, and transmitting the upstream voice on the out channel of the downstream voice channel to the (n-1) mobile devices excluding the one that has finished the call. It notifies the use condition of the slot and the change of the encoding speed of the transmission data, and controls the control means 1 of the n-1 slave units which are already talking.
Is characterized by performing control to shift to a call at the encoding speed as instructed in accordance with the change notification.

That is, the invention described in claim 4 is the same as in claim 1
Alternatively, the present invention relates to a communication procedure in a case where a connection between one slave unit during a call is disconnected after a call between a plurality of slave units is started by the invention according to claim 2. The disconnection procedure includes a case in which the call is disconnected in response to a call termination request from the child device, and a case in which PSm terminates / disconnects the child device. FIG. 5 shows a case where a call is disconnected in accordance with a call termination request from a slave unit.

For example, as shown in FIG. 5, when a call is initially made between PSm and PS1 and PS2 (41), (42) and (43), PSm is transmitted from PS2 to the radio channel. When a disconnection request is received (44), a wireless channel disconnection completion is sent to PS2 (45) to execute a call termination sequence. And
The PSm notifies the PS1 of the use condition of the uplink audio transmission slot and the change of the encoding rate of the transmission data on the out channel of the downlink audio channel at the 1/2 encoding rate (4).
6). In the illustrated example, since the three-party communication is changed to the two-party communication, the encoding speed is changed from the 1/2 encoding speed to the standard encoding speed.

As a result, PS1 transmits uplink audio data at the standard encoding rate (47), and PSm transmits downlink audio data at the standard encoding rate (48). In this way, a change from a three-party call to a two-party call can be made by a simple connection procedure. According to a fifth aspect of the present invention, in the digital cordless telephone system according to any one of the first to fourth aspects, the A / D conversion means and the D / A conversion means include: m _i (m ₀ <m ₁ <m ₂ <m ₃ <m ₄
When (<...) is specified, (standard coding speed as system / m ₁ ), standard coding speed as system / m ₂ , standard coding speed as system / m ₃ ,
(Standard encoding speed as a system / m ₄ )... Which handles speech encoded data at a rate of
When the number of slave units in the sub-slave unit mode is n (n = 1, 2, 3,
4,5 When a ...), the coding rate of the transmission voice data of the slave unit of the sub-child device mode, m _{i -1} a <n ≦ m _i m _i
And select (standard coding rate for system / m _i )
Is determined.

That is, according to the fifth aspect of the invention, the number of slave units in the sub-slave unit mode is n (n = 1, 2, 3, 4, 5,.
..), the encoding speed of the transmission data of the slave unit in each sub-slave unit mode is 1 / m _i (m = 1, 2, 4, 6,...) Of the standard encoding speed for the system. It is characterized by being. That is, in the invention according to claim 5, the TDMA
An appropriate coding rate required for a call between a plurality of slave units connected via a frame can be formed from the standard coding rate of the system.

Incidentally, the application in the current PHS is as follows. In the current PHS, the standard coding rate for the system is 32 kbps. This is called full rate. Since m _i = m _n = 2 ⁿ⁻¹ , m ₀ =
0, m ₁ = 2 ⁰ = 1, m ₂ = 2 ¹ = 2, m ₃ = 2 ² = 4, m ₄
= 2 ³ = 8, m ₅ = 2 ⁴ = 16... Then, n
= 1 (between two parties), n = 2 (between three parties), n = 3 (between four parties), n = 4 (between five parties), and n = 5 (between six parties). Let's find the rate of conversion.

[0041] In n = 1 (two-party), since the _{m 0 <n ≦ m 1,} C = 32 / m 1 = 32/1 = 32kbps ··
At the standard coding rate (full rate), n = 2 (three-way), m ₁ <n ≦ m ₂ , so C = 32 / m ₂ = 32 /
2 = 16 kbps... 1/2 coding rate (half rate), n = 3 (four), m ₂ <n ≦ m ₃ , so C = 32 / m ₃ = 32/4 = 8 kbps.・ 1/4
At the coding rate (quarter rate), n = 4 (five), since m ₂ <n ≦ m ₃ , C = 32 / m ₃ = 32 /
4 = 8 kbps... １ ／ coding rate (quarter rate), n = 5 (six parties), m ₃ <n ≦ m ₄ , so C = 32 / m ₄ = 32/8 = 4 kbps.・ 1/8
It is the encoding speed. Therefore, with the current PHS, up to the quota rate is possible, so that it is possible to make a direct telephone call between slave units up to five parties.

According to a sixth aspect of the present invention, in a digital cordless telephone system comprising a wireless base station and a plurality of slaves connected to the wireless base station via a wireless line, It has the same configuration as that of the slave unit in the main slave unit mode according to any one of claims 1 to 4, and controls a call between a plurality of slave units via a wireless base station. That is, according to the invention described in claim 6, the wireless base station originally provided as a system can have the same function as that of the slave unit in the main slave unit mode described in claims 1 to 4.

[0043]

Embodiments of the present invention will be described below with reference to the drawings. FIG. 6 is a configuration diagram of an embodiment corresponding to claims 1 to 5. The slave according to this embodiment is different from the conventional slave shown in FIG.
A mixing control unit 30 is provided between the control unit 15 and the microphone 20 and the speaker 21, and a control unit 22a is obtained by slightly adding a function to the control unit 22b.

In this embodiment, the control section 22a is provided with a function for controlling communication in a direct communication mode between a plurality of slave units so as to be switchable between a main slave unit control mode and a sub slave unit control mode. These modes are designated by a setting input from the external terminal interface unit 26,
Execute each communication control. That is, each slave in the cordless telephone system of this embodiment is arbitrarily a main slave and a sub slave. The main handset controls a direct call between the plurality of handset units via the own main handset.
At this time, the encoding speed of the uplink audio transmission data is determined according to the number of sub-child units, and the sub-child units are notified.

As shown in FIG. 7, the mixing control section 30 includes a switch (SW) 31, a first D / A conversion section 32, a second D / A conversion section 33, a third D / A conversion section 34, a first A /
D converter 35, second A / D converter 36, third A / D converter 37, audio mixer 38, switches (SW) 39, 40
Etc. are provided. As described above, the current PHS employs 32 kADPCM as a full-rate audio coding system, and does not particularly define a half-rate (16 kADPCM) or quarter-rate (8 kADPCM) audio coding system, but can use it. Therefore, in this embodiment, the sub-slave unit transmits the upstream voice data to be transmitted to the main slave unit also in the half-rate (16 kADPCM) and quarter-rate (8 kADPCM) voice coding schemes, so that simultaneous communication between a plurality of slave units is performed. Is realized.

That is, the first D / A converter 32 and the first A / D
The conversion unit 35 handles the full-rate encoded audio data,
The second D / A converter 33 and the second A / D converter 36 handle half-rate audio encoded data, and the third D / A converter 34 and the third A / D converter 37 execute quarter-rate audio encoding. Handle data. These operate under the control of the control unit 22a, and the control unit 22a operates the SWs 31, 39, 40, 4
By controlling the switching between 1 and 42, required signals are input and output.

The SW 31 is a switch of one input and three outputs, and the input terminal receives the received voice data from the TDMA control unit 15. In the output terminal of the SW 31, a first output terminal is connected to an input terminal of the first D / A converter 32, a second output terminal is connected to an input terminal of the second D / A converter 33, and a third output terminal is connected. The third D / A converter 34 is connected to an input terminal. The first D / A conversion unit 32 is a conversion unit that decodes and converts 32 k ADPCM (full rate) audio encoded data input from the first output terminal of the SW 31 into an analog audio signal. The output terminal of the first D / A converter 32 is a two-input one-output SW3
9 is connected to one input terminal. In the SW 39, the output terminal of the audio mixer 38 is connected to the other input terminal, and the output terminal is connected to the speaker 21.

[0048] The second D / A converter 33 is provided with the second
16k ADPCM input from output terminal (half rate)
Is a conversion unit that decodes and converts the encoded audio data into an analog audio signal. Two output terminals of the second D / A converter 33 are connected to two input terminals of a 6-input, 4-output SW41, respectively. The third D / A conversion unit 34 is the first D / A converter
8k ADPCM (quarter rate) input from output terminal
Is a conversion unit that decodes and converts the encoded audio data into an analog audio signal. Four output terminals of the third D / A conversion unit 34 are connected to four input terminals of a 6-input 4-output SW 41, respectively. The four output terminals of the SW 41 are the audio mixer 3
8 input terminals.

The one-input two-output SW 40 has an input terminal to which an audio signal is applied from the microphone 20, one output terminal connected to the input terminal of the audio mixer 38, and the other output terminal together with the output terminal of the audio mixer 38. It is connected to the input terminal of one-input three-output SW42. The output terminal of the SW 42 has the first output terminal
The A / D converter 35 is connected to an input terminal, the second output terminal is connected to an input terminal to a second A / D converter 36, and the third output terminal is connected to an input terminal to a third A / D converter 37. You.

The first A / D converter 35 converts an audio signal input from the audio mixer 38 or the microphone 20 via the SW 42 into audio encoded data of 32 kADPCM (full rate). The second A / D converter 36 converts an audio signal input from the audio mixer 38 or the microphone 20 via the SW 42 into audio encoded data of 16 kADPCM (half rate). The third A / D conversion unit 37 is a switch 42
The audio signal input from the audio mixer 38 or the microphone 20 via the microphone is converted into 8 kADPCM (quarter rate) audio encoded data. The output (transmission audio data) of each of the conversion units 35, 36, and 37 is transmitted to the TDMA control unit 15 respectively.
To enter.

The correspondence between the above configuration and the claims is as follows. The control means 1 mainly includes a control unit 22
a corresponds. The first A / D converter 32, the second A / D converter 33, and the third A / D converter 34 correspond to the A / D converter 2. The first D / A converter 35, the second D / A converter 36, and the third D / A converter 37 correspond to the D / A converter 3. The audio mixer 38 mainly corresponds to the mixing control means 4.

The operation of this embodiment will now be described with reference to FIGS.
This will be described with reference to FIG. Since the communication control procedure in this embodiment complies with RCR-STD28,
Detailed description of each information element, signal name, etc. is omitted. A) Assumptions of Embodiment First, assumptions of this embodiment will be described with reference to FIGS. FIG. 7 shows a control channel (SCCH) and a communication channel (TCH, FACC) used in this embodiment.
It is a block diagram of H).

As shown in FIG. 7A, the SCCH has a ramp time R for transient response (4 bits), a start symbol SS (2 bits), a preamble PR (62 bits),
Synchronization word UW (32 bits), channel type CI (4
Bit), destination identification code (42 bits), calling identification code (2
8 bits), I (SCCH, 34 bits), CRC (16 bits), and guard bits (16 bits).

As shown in FIG. 7 (2), the communication channel (TCH) includes a transient response ramp time R (4 bits), a start symbol SS (2 bits), and a preamble PR.
(6 bits), synchronization word UW (16 bits), channel type CI (4 bits), SA (SACCH, 16 bits), I (TCH, 160 bits), CRC (16 bits), and guard bits (16 bits) Be composed. This TCH specifies each unit frame of the TDMA frame shown in FIG. In the channel type CI,
The identification number of the TCH (that is, the sequence number of the frame) is shown (see FIG. 9).

The communication channel (FACCH) is shown in FIG.
As shown in FIG. 5, a ramp time R for transient response (4 bits), a start symbol SS (2 bits), a preamble PR (6 bits), a synchronization word UW (16 bits),
Channel type CI (4 bits), SA (SACCH, 1
6 bits), I (FACCH, 160 bits), CRC (1
6 bits) and guard bits (16 bits).

FIG. 9 is a CI coding diagram of the TCH according to the embodiment. In this embodiment, simultaneous calls between a plurality of slave units and a plurality of slave units among three to five persons are realized using the current PHS. The simultaneous three-way call is a call between the main handset and two sub-handsets. In this case, two TCH channel types are used. The four-party simultaneous call is a call between the main handset and three sub-handsets. In this case, three TCH channel types are used. The five-party simultaneous call is a call between the main handset and four sub-handsets. In this case, four TCH channel types are used.

That is, in simultaneous communication between a plurality of slave units among three to five parties, a maximum of four TCH channel types are used. Therefore, as shown in FIG. 9, the identification numbers 1 to 4 of the four TCHs for simultaneous communication between a plurality of slave units are set by a 4-bit channel type (CI). This TCH number indicates which sub child device is the TCH used. This is related to an in-slot sequence number described later.

In a simultaneous communication between a plurality of slave units among three to five parties, communication is performed using a maximum of four frames. Therefore, in the simultaneous call method between a plurality of slave units of this embodiment,
The TDMA frame has the three configurations shown in FIGS. FIG. 10 shows a frame configuration during a simultaneous three-way call. The TDMA frame in a simultaneous three-way call is 2
Two TCHs (TCH1 and TC
H2) is alternately repeated.

FIG. 11 shows a frame configuration at the time of simultaneous communication between four parties. In this case, the number of sub-child units is three, but T
The DMA frame has four TCHs (TCH1, TCH1,
2, TCH3, TCH4) are repeated. The three sub handsets use three of the four TCHs. This is a constraint based on the relationship with the adopted encoding speed.

FIG. 12 shows a frame configuration at the time of simultaneous communication between five parties. In this case, the TDMA frame includes four TCHs (TCH1, TCH1,
2, TCH3, TCH4) are repeated. Then, as shown in FIG. 34, in the current PHS, one frame includes a downlink transmission / reception area “T” and an uplink transmission / reception area “R”.
Each area is composed of four time slots. In this embodiment, one of the four time slots is used by a plurality of slave units as a common communication channel, thereby realizing simultaneous communication between the plurality of slave units.

Therefore, as shown in FIGS. 10 to 12, the T used in each of the four time slots (communication channels) of the downlink transmission / reception area “T” and the uplink transmission / reception area “R”.
An identification code is attached to CH. TCH used in each of the four time slots is, for example, T11 to TCH1 in TCH1.
For T41, R11-R41 and TCH2, T12-T42, R12-
For R42 and TCH3, T13 to T43, R13 to R43, TCH
In No. 4, they are described as T14 to T44 and R14 to R44.

For example, the first subscript "1" of T11 is the sequence number of the TCH indicating which time slot in the frame it is. The second subscript "1" of T11 is an in-slot sequence number indicating which slave unit uses the time slot. The in-slot sequence number is a number indicating which frame the time slot is a time slot of, and has a one-to-one correspondence with the TCH number shown in FIG.

After all, in the simultaneous three-way communication, the two sub-child units transmit the upstream voice data to the main child unit in two frames.
Since transmission can be performed twice, transmission is performed at a half rate (16 kADPCM) obtained by compressing the full rate (32 kADPCM) to half. In a simultaneous four-party call, three sub-child units use three of the four TCHs so as to transmit upstream audio data to the main child unit once every four frames. In other words, in a simultaneous four-party call, the three sub handsets transmit at a quarter rate (8 kADPCM) obtained by compressing the full rate (32 kADPCM) to 1/4.

In a simultaneous five-party call, the four sub handsets transmit upstream audio data to the main handset in four frames.
Times, so the quota rate (8 kADP
CM). On the other hand, in any case, the main slave transmits the downlink audio data to each sub-slave to the current PH.
Transmit at full rate (32 k ADPCM) as per S.

It is to be noted that which of the main sub units and which of the sub sub units, that is, the operator of the sub unit
Whether to set the main slave unit mode or the sub slave unit mode in “a” is a known matter at the time of starting a telephone call, for example, because a discussion is made between the parties. In addition, the number of slave units participating in the call is also known. B) Simultaneous call between plural slave units Connection of simultaneous call between plural slave units is realized by transmission and reception of a call signal and a response signal shown in FIGS.

FIG. 13 shows a call message format (SCCH), and FIG. 14 shows a response message format (SCCH). 13 and 14, the information elements related to this embodiment are a call between a plurality of slave units (message type), a response between a plurality of slave units (message type), a codec type, and an in-slot sequence number. The other information elements are based on RCR-STD28, and the description is omitted.

In the case of calling between a plurality of slave units (message type),
The condition of the uplink transmission in the call signal transmitted for each child device is transmitted to the sub child device. In the response (message type) between a plurality of slave units, a response message corresponding to the calling signal is added. The codec type specifies three types of transmission rates of uplink audio data by using upper three bits (8, 7, 6). For example, the full rate is “000” and the half rate is “00”.
1 ", and the quota rate is designated by" 010 ". In the call signal, the transmission rate to the slave unit is specified,
The response signal confirms the designation.

The sequence number in the slot is, as described above,
This is a sequence number indicating in which frame the time slot to be used by the sub-child device is present, and a number (FIG. 9) specifying the TCH to be used by the sub-child device. The in-slot sequence number is specified using the lower two bits (2, 1). For example, the slot sequence number 1 is “00”.
The sequence number 2 in the slot is “01”, the sequence number 3 in the slot is “10”, and the sequence number 4 in the slot is “1”.
1 ". The calling signal specifies the TCH to be used for the slave, and the response signal specifies the specified TCH.

(First Embodiment: Simultaneous Three-way Call) FIG.
Is a connection sequence for a three-way simultaneous call. The same applies to the various sequences described below, except that the main handset is PSm and the sub handset is PS1, PS2, PS3, PS4.
It is written. In FIG. 15, T1m, R1m (m
= 1 to 4) is the time slot T shown in FIG.
1, indicating that SCCH and TCH with m as the in-slot sequence number are transmitted and received when R1 is a used slot. This is the matter described with reference to FIGS. The notations of T1m and R1m (m = 1 to 4) are the same in the following various sequences.

FIG. 15 (1): When the main handset PSM and the sub-handsets PS1 and PS2 perform a three-way call by simultaneous calling, the main handset PSm is not used in the time slots T1 and R1 shown in FIG. The "carrier for direct communication between slave units" not captured by any of the above-mentioned calls is obtained based on the electric field strength and the like, and if it is determined to be "vacant", the sub slave units PS1 and PS2 have the format shown in FIG. Paging signals 1 and 2 are alternately and continuously transmitted in time slot T1 (T11, T12).

At this time, main handset PSm calls call signal 1 using SCCH to call sub-handset PS1.
(FIG. 13) is transmitted including the following two instructions. The first instruction is “a downlink SCC having an in-slot sequence number“ 1 ”.
The response signal (FIG. 14) must be transmitted using the SCCH only in the uplink time slot 2.5 ms after the time slot of the TDMA frame that can receive H.
Transmission is prohibited in other time slots. " The second instruction is “During a call, a downlink TC having a channel type (CI) (FIG. 9) with an in-slot sequence number“ 1 ”.
The uplink TCH must be transmitted at the half rate only in the uplink time slot 2.5 ms after the time slot of the TDMA frame that can receive H. Transmission is prohibited in other time slots. "

The main slave unit PSm is connected to the sub slave unit PS.
Paging signal 2 using SCCH to page
3) is transmitted including the following two instructions. The first instruction is “2.5 timeslot of a time slot of a TDMA frame that can receive a downlink SCCH having an in-slot sequence number“ 2 ”.
The response signal (FIG. 14) must be transmitted using the SCCH only in the uplink time slot after ms. Transmission is prohibited in other time slots. " The second instruction is that "during a call, only in an uplink time slot 2.5 ms after a time slot of a TDMA frame capable of receiving a downlink TCH having a channel type (CI) (FIG. 9) of sequence number" 2 "in a slot. The uplink TCH must be transmitted at a half rate, and transmission is prohibited in other time slots. "

FIG. 15 (2): When the main handset PSm receives the response signals 1 and 2 for the call signals 1 and 2 from the sub handset PS1 and PS2, the downstream TCH (T11, T12)
Then, transmission is started by alternately attaching channel types (CI) of “1” and “2” as slot sequence numbers. FIG. 15 (3): After that, the main slave unit PSm and the sub slave unit PS
1, PS2 is in a three-way communication state according to the above-mentioned calling signals 1, 2. At this time, during the three-way call between the sub-child device PS1 and the child device PS2, the main child device PSm
The upstream half-rate audio of S1 and PS2 is mixed with the input audio of the slave unit, and the downstream TCH
In the time slot T1 (T11, T12). This mixing control will be described later (FIG. 30).

(Second Embodiment: Simultaneous Calls Between Five Persons) FIG.
Is a connection sequence of a simultaneous call between five parties. PSm
For the sub slave units PS1, PS2, PS3, PS4,
The connection of the five-party simultaneous communication is performed by the same procedure as the procedures (1), (2), and (3) shown in FIG. This mixing control will be described later (FIG. 31).

As described above, a plurality of sub-child units cyclically use one common time slot T1 (T11, T12, T13, T14) to realize simultaneous communication. C) Change in the number of slave units for simultaneous calls In the actual application of simultaneous calls, calls are not always continued between the same number of slave units until the end. Ends a handset during a call,
Or, a child device may end the call by itself.

When subscribing to another slave unit during simultaneous communication of a plurality of slave units, control signals shown in FIGS. 17 to 20 are used. In addition, in the case of terminating and disconnecting one slave unit during simultaneous communication of a plurality of slave units, the control signals in FIGS. 17 and 18 are used. FIG. 17 shows a codec change instruction message format (SACCH). "Codec change instruction (message type)" and "codec type" are included. The “codec change instruction” is an instruction to change the uplink transmission rate. The main handset sends. FIG. 18 shows a codec change completion message format (SACCH). "Codec change completed (message type)" and "codec type" are included. This is a response indicating that the uplink transmission rate has been changed as instructed. The sub handset transmits.

FIG. 19 shows a call message format during communication (SACCH). It includes "call during communication (message type)", "codec type", and "sequence number in slot". When calling and subscribing to another slave during simultaneous communication between a plurality of slaves, a transmission rate or the like is designated. The main handset sends. FIG. 20 shows a call response message format during communication (FACCH). It includes "call response during communication (message type)", "codec type", and "sequence number in slot". This is a response to the effect that a subscription has been made at a transmission rate or the like as instructed. The sub handset transmits.

(First Embodiment: Change from Two-Party Call to Three-Party Call) FIGS. 21 and 22 show a connection sequence for changing to a simultaneous three-party call during a two-party call. FIG. 21 (1): The main handset PSm makes a call to the sub handset PS1, and direct communication between the handset at a full rate is performed between the main handset PSm and the sub handset PS1.

FIG. 21B: In this state, the main slave unit PSm
However, when a call is made to the sub slave unit PS2 to perform a three-way call,
The main handset PSm transmits a codec change instruction message (FIG. 17) including the following instruction to the sub handset PS1 using the SACCH. In other words, this codec change instruction message includes “uplink time slot 2.5 ms after the time slot of the TDMA frame that can receive the downlink TCH having the channel type (CI) (FIG. 9) with the in-slot sequence number“ 1 ”. Only the uplink TCH must be transmitted at a half rate, and transmission is prohibited in other time slots. "

After transmitting such a codec change instruction message, the main handset PSm sends to the sub handset PS1 a channel type (CI) of “1” and “2” as in-slot sequence numbers on the downlink TCH (T11). ) And start sending. After receiving the codec change instruction message and receiving the downlink TCH having the channel type (CI) (FIG. 9) of the in-slot sequence number “1”, the sub-child device PS1 receives the time slot ( T11) 2.5 ms after the upstream time slot (R
At 11), a codec change completion message (FIG. 18) is transmitted.

FIG. 22 (3): Upon receiving the codec completion message, the main handset PSm receives the codec completion message.
In response to this, a paging signal during communication (FIG. 19) including the next instruction is continuously transmitted using the SACCH of the downlink TCH (T11, T12). In other words, the paging signal during communication includes "uplink 2.5 ms after the time slot (T12) of the TDMA frame capable of receiving the downlink TCH having the channel type (CI) of the in-slot sequence number" 2 "(FIG. 9). Time slot (R1
Only in 2), the uplink TCH must be transmitted at half rate. In other time slots, transmission is prohibited. "

The sub-child device PS2 receives the above-described calling signal during communication, and sets the channel type (C
I) After receiving the downlink TCH having (FIG. 9), 2. of the time slot (T12) of the received TDMA frame.
A response signal during communication (FIG. 20) is transmitted to the main handset PSm by FACCH in the upstream time slot (R12) 5 ms later.

During this time, the main slave unit PSm and the sub slave unit PS
A call is being made with 1. FIG. 22 (4): Upon receiving the response signal during communication, the main handset PSm performs an uplink half-rate of the sub handset PS1 and PS2 for a three-way call with the sub handset PS1 and PS2. The voice and the input voice of the slave unit are mixed, and the downstream TCH is supplied to T1 (T11, T12) as a full rate voice.

(Second Embodiment: Change from Three-Party Call to Four-Party Call) FIGS. 23 and 24 show a connection sequence for changing to a simultaneous three-party call during a three-party call. The above two people →
The change is made in the same procedure as in the three cases. That is, FIG.
3, in FIG. 24, the main PSm and the sub slave unit PS1,
Simultaneous call with PS2 (FIG. 23)
(1)), the main slave units PSm are the sub slave units PS1 and PS2.
, A codec change instruction to change the half rate to the quarter rate is issued (FIG. 23 (2)), and then the sub slave unit P
A calling signal is transmitted to S3 to make it join (FIG. 23).
(3), FIG. 24 (3)), and shift to simultaneous communication between the four parties (FIG. 24 (4)).

(Third Embodiment: Change from Four-Party Call to Five-Party Call) FIGS. 25 and 26 show a connection sequence for changing to a simultaneous call of five parties during a four-party call. In this case, basically, the change is performed in the same procedure as in the above-described two-to-three case. However, there is a slight difference between the change from the four parties to the five parties because there is no change in the transmission rate.

FIG. 25 (1): The direct communication between the slave units is performed between the main slave unit PSm and the sub slave units PS1, PS2 and PS3. FIG. 25 (2): When the main handset PSm calls the sub-handset PS4 to perform a five-way call during a four-way call, the main handset PSm descends to the handset PS4 and receives the TCH (T11, T12, T
The communication paging signal (FIG. 19) including the next instruction is continuously transmitted using the SACCH of (13, T14).

That is, this paging signal during communication includes “2.5 ms of the time slot (T14) of the TDMA frame that can receive the downlink TCH having the channel type (CI) (FIG. 9) of the sequence number“ 4 ”in the slot. The uplink TCH must be transmitted at the quarter rate only in the subsequent uplink time slot (R14), and transmission is prohibited in other time slots. "

The sub-child unit PS4 receives the above-mentioned calling signal during communication, and sets the channel type (C
I) After receiving the downlink TCH having (FIG. 9), 2. of the time slot (T14) of the received TDMA frame.
A communication response signal (FIG. 20) is transmitted to the main handset PSm on the FACCH in the up time slot (R14) after 5 ms.

FIG. 26 (3): When the main handset PSm receives the communication response signal, the sub handset PS1, PS2,
For a five-party call between PS3 and PS4, the uplink quarter-rate audio of sub-child units PS1, PS2, PS3, and PS4 and the input audio of its own child device are mixed, and the time slot of downlink TCH as full-rate audio. T1 (T11, T
12, T13, T14).

(Fourth Embodiment: Change from Three-Party Call to Two-Party Call) FIG. 27 shows a connection sequence for shifting from a simultaneous three-party call to a two-party call. In FIG. 27, during a three-way call between main slave unit PSm and sub-slave units PS1 and PS2 (FIG. 27 (1)), sub-slave unit PS during a call is in progress.
2 terminates the call itself (illustrated example), or when the main sub unit PSm disconnects the sub sub unit PS2 during a call, the main sub unit PSm performs a call termination sequence (FIG. 2).
7 (2)), the SACCH is used to transmit a codec change instruction message (FIG. 17) including the following instruction to the sub-child device PS1, and thereafter, the downlink TCH of only the channel type of “1” as the in-slot sequence number is transmitted. Start transmitting continuously (Fig. 2
7 (3)).

That is, the codec change instruction message includes “channel type (C
I) The uplink TCH must be transmitted at a rate only in the uplink time slot (R11) 2.5 ms after the time slot (T11) of the TDMA frame that can receive the downlink TCH having (FIG. 9). " included. Upon receiving the codec change instruction message, the sub-child device PS1 receives the channel type (C
I) After receiving the downlink TCH having (FIG. 9), 2. of the time slot (T11) of the received TDMA frame.
The codec change completion message (FIG. 18) is transmitted in the uplink time slot (R11) after 5 ms (FIG. 27 (4)), and the process shifts to a full-rate direct communication between handset units (FIG. 27 (5)). )).

Fifth Embodiment Change from Four-Party Call to Three-Party Call FIG. 28 shows a connection sequence for shifting from a simultaneous four-party call to a three-party call. In the same procedures (1), (2), (3), (4), and (5) as described with reference to FIG. 27, the simultaneous four-way communication (quarter rate) is shifted to a three-way communication (half rate). (Sixth embodiment: change from five-party call to four-party call)
FIG. 29 shows a connection sequence for shifting from simultaneous five-party communication to four-party communication. In this case also, basically, the process proceeds according to the procedure described with reference to FIG. 27. However, during a five-way call, the sub-slave PS4 during the call is disconnected or the main slave PSm is switched to the sub-slave PS4. When the call is cut off, the main handset PSm continues the four-way call in which the upstream voice transmission is performed at the quarter rate after the call end sequence, and the sub-handset PS.
1, PS2 and PS3. PS1,
This is because it is necessary for the three sub-child units, PS2 and PS3, to perform uplink voice transmission once every four frames.

D) Description of Mixing Control (Mixing Control During Three-Party Communication) FIG. 30 is a conceptual diagram of half-rate mixing control. This is during a three-way call between the main slave unit PSm and the sub slave units PS1 and PS2 (for example, FIG. 15).

In FIG. 30A, the main slave unit PSm
Is the upstream half-rate audio data (16 kADP) in time slot R11 of sub-child device PS1 in FIG.
CM) and the upstream half-rate audio data (16 kADPCM) in the time slot R12 of the slave unit PS2 are input to the second D / A converter 33 and converted into analog audio signals, respectively, output to the speaker 21 and output from the microphone 20. The analog audio signal and the analog audio signal are mixed in the audio mixer 38 and supplied to the first A / D converter 35 to be converted into a downstream full-rate audio signal (32 kADPCM).
Are transmitted to slave units PS1 and PS2 in time slots T11 and T12 in.

In FIG. 30 (b), the sub slave units PS1 and PS2 are connected to the downlink full rate audio signal (32
When receiving the KADPCM), it is input to the first D / A converter 32, converted into an analog audio signal, and output to the speaker 21. The audio signal input from the microphone 20 is the second A /
The data is converted into half-rate (16 k ADPCM) uplink audio data by the D conversion unit 36.

As described above, the conversation in the three-way call is realized. (Mixing control during a four- or five-party call) FIG.
FIG. 3 is a conceptual diagram of quarter rate mixing control. This is because the main child unit PSm and the sub child units PS1, PS2,
During a four-way call with PS3, or sub-child units PS1, P
During a five-party call with S2, PS3 and PS4 (for example, FIG.
6).

In FIG. 31 (a), main slave unit PSm
Is the upstream quarter-rate audio data (8 kAD) in the time slot R11 of the sub-child device PS1 in FIG.
PCM), upstream quarter-rate audio data (8 kADPCM) in time slot R12 of sub-subunit PS2, uplink quarter-rate audio data (8 kADPCM) in time slot R13 of sub-subunit PS3, and time slot R14 of sub-subunit PS4. Is input to the third D / A converter 37 to be converted into an analog audio signal, output to the speaker 21, and mixed with the analog audio signal from the microphone 20 by the audio mixer 38. Then, the first A / D converter 35
And input to the downstream full rate audio data (32k ADPC
M) and converted to T1 (T11, T12, T1) in FIG.
In (3, T14), the data is transmitted to the sub child device PS1, the sub child device PS2, the sub child device PS3, and the sub child device PS4.

In FIG. 31 (b), the sub slave units PS1,
The sub slave unit PS2, the sub slave unit PS3, and the sub slave unit PS4 are
When receiving the downlink full-rate audio data (32 kADPCM), the data is input to the first D / A converter 32, converted into an analog audio signal, and output to the speaker 21. The audio signal input from the microphone 20 is transmitted to the third A / D converter 3.
6 is the upstream audio data of the quarter rate (8 k ADPCM).

As described above, a conversation in a four-party or five-party call is realized. In the embodiment described above,
An example is shown in which a call is made between the slave units by defining the main slave unit and the sub-slave unit. However, as shown in FIG.
S is connected to the wireless base station 41 via a wireless line, so that the wireless base station 41 has the function of the main slave unit,
The wireless base station 41 may control a call between a plurality of slave units. That is, this is an embodiment corresponding to claim 6.

In the current PHS, up to the quarter rate (8 k ADPCM) can be adopted, but further 4 k
If ADPCM can also be adopted, it goes without saying that direct communication between six or more slave units becomes possible.

[0101]

As described above, according to the first aspect of the present invention, the main handset gives a plurality of sub-handsets the use condition of the upstream voice transmission slot and the coding speed of the transmission data. With multiple communication procedures, multiple slave units
A connection in which one time slot is shared and simultaneous calls can be made can be realized.

According to the second aspect of the present invention, the main handset gives the use condition of the upstream voice transmission slot and the encoding speed of the transmission data to the plurality of sub handset, so that the plurality of handset can be set to one. Simultaneous communication can be performed by sharing one time slot, so that traffic on the wireless line can be reduced. According to the third aspect of the present invention, when a new sub-unit joins a call after a call between a plurality of sub-units is started according to the first or second aspect of the present invention, one of the sub-units during the call is
A simple procedure in which the main handset shares a change of time with the sub-handset during a call by sharing one time slot and calls the sub-handset that newly joins the call. It is possible to change to n + 1 simultaneous calls.

According to the fourth aspect of the present invention, when a call between a plurality of slave units is started according to the first or second aspect of the present invention, one of the slave units during the call is disconnected. , Perform a call termination sequence with the sub-child device that ends the call,
By a simple procedure of notifying the remaining sub-child devices of the change, it is possible to change from simultaneous communication between n parties to simultaneous communication between n-1 parties.

According to the fifth aspect of the present invention, it is possible to form an appropriate coding rate necessary for a call between a plurality of slave units connected via a TDMA frame from a standard coding rate as a system. According to the sixth aspect of the present invention, it is possible to cause the wireless base station originally provided as the system to have the same function as that of the slave unit in the main slave unit mode according to any one of the first to fourth aspects. Thus, for example, when there is no free space in the communication channel, which is the time slot of the TDMA frame, it is possible to make the slave unit waiting for the free space use the busy communication channel.

In short, according to the present invention, there is an advantage that simultaneous communication between a plurality of slave units and simultaneous calling of a plurality of slave units by a certain slave unit can be easily and reliably realized without increasing traffic.

[Brief description of the drawings]

FIG. 1 is a block diagram showing the principle of the invention described in claims 1 to 5;

FIG. 2 is an operation sequence according to the first and fifth aspects of the present invention (when n = 2);

FIG. 3 shows an operation sequence according to the second and fifth aspects of the present invention (when n = 2).

FIG. 4 shows an operation sequence according to the third and fifth aspects of the present invention (when n = 1 → n = 2).

FIG. 5 is an operation sequence according to the fourth and fifth aspects of the present invention (when n = 2 → n = 1);

FIG. 6 is a configuration diagram of an embodiment corresponding to claims 1 to 5;

FIG. 7 is a configuration diagram of a mixing control unit.

FIG. 8 is a configuration diagram of a channel.

FIG. 9 is a diagram of TCH CI coding according to the embodiment;

FIG. 10 is a diagram illustrating a frame configuration during a three-way call.

FIG. 11 is a diagram illustrating a frame configuration during a four-party call.

FIG. 12 is a diagram illustrating a frame configuration during a five-party call.

FIG. 13 is a call message format.

FIG. 14 is a response message format.

FIG. 15 is a connection sequence of a simultaneous three-way call.

FIG. 16 is a connection sequence of a simultaneous call between five persons.

FIG. 17 shows a codec change instruction message format (SACCH).

FIG. 18 shows a codec change completion message format (SACCH).

FIG. 19 is a communication call message format (SAC).
CH).

FIG. 20 shows a call response message format during communication (F
ACCH).

FIG. 21 is a connection sequence for changing to a simultaneous call of three parties during a two-party call (1/2).

FIG. 22 is a connection sequence for changing to a simultaneous call of three parties during a two-party call (2/2).

FIG. 23 is a connection sequence for changing to a simultaneous call of four parties during a three-party call (1/2).

FIG. 24 is a connection sequence for changing to a simultaneous call of four parties during a three-party call (2/2).

FIG. 25 is a connection sequence for changing to a simultaneous call of five parties during a four-party call (1/2).

FIG. 26 is a connection sequence for changing to a simultaneous call of five parties during a four-party call (2/2).

FIG. 27 is a connection sequence for changing from simultaneous three-way communication to two-way communication.

FIG. 28 is a connection sequence for changing from simultaneous four-party communication to three-party communication.

FIG. 29 is a connection sequence for changing a simultaneous call between five parties to a call between four parties.

FIG. 30 is a conceptual diagram of half-rate mixing control. (A) shows the case of the main sub unit, and (b) shows the case of the sub sub unit.

FIG. 31 is a conceptual diagram of quarter rate mixing control. (A) shows the case of the main sub unit, and (b) shows the case of the sub sub unit.

FIG. 32 is a basic configuration diagram of a PHS.

FIG. 33 is a configuration diagram of a conventional slave unit.

FIG. 34 is a diagram illustrating a TDMA frame configuration of a PHS.

FIG. 35 shows a sequence of a direct communication system between slave units specified by RCR-STD28.

[Explanation of symbols]

 REFERENCE SIGNS LIST 1 control means 2 A / D conversion means 3 D / A conversion means 4 mixing control means 9 transmission / reception antenna 10 transmission / reception switch (SW) 11 transmission section 12 reception section 13 modulation section 14 demodulation section 15 TDMA control section 20 microphone 21 speaker 22a Control unit 23 RAM 24 ROM 25 EEPROM 27 Human interface unit 30 Mixing control unit 31, 39, 40, 41, 42 Switch (SW) 32 First D / A conversion unit 33 Second D / A conversion unit 34 Third D / A conversion Unit 35 1st A / D conversion unit 36 2nd A / D conversion unit 37 3rd A / D conversion unit 38 Audio mixer

Claims (6)

[Claims] In a digital cordless telephone system, control means for setting a slave unit in a plurality of slave units by switching between a main slave unit mode and a sub slave unit mode, and executing communication in the set mode; A / D conversion means for converting a transmission audio signal into audio data of a plurality of encoding speeds up to a standard encoding speed as a system, and converting the received audio data of the corresponding encoding speed into an audio signal. And D / A conversion means for converting the data into a digital signal. The control means for the child device in the main child device mode uses the paging signal of the control channel to use the upstream voice transmission slot and the number of child devices in the sub child device mode. n (n =
1, 2, 3,...) To notify the predetermined number n of the sub-units of the sub-units in the sub-unit mode, and to transmit the received uplink voice data in the corresponding downlink transmission slot. The control means for the predetermined number n of the slave units in the sub-slave mode respectively control the corresponding A / D converter in accordance with the notified encoding speed and transmit the upstream audio data to the predetermined encoding speed. And transmitting in the notified uplink voice transmission slot in the digital cordless telephone system. 2. In a digital cordless telephone system, control means for setting a slave unit to a main slave unit mode and a sub slave unit mode for each of a plurality of slave units, and executing communication in the set mode. A / D conversion means for converting a transmission audio signal into audio data of a plurality of encoding speeds up to a standard encoding speed as a system, and converting the received audio data of the corresponding encoding speed into an audio signal. D / A conversion means for converting the audio signal of the slave unit to the output of the D / A conversion means, and the control means of the slave unit in the main slave mode comprises: The use condition of the uplink voice transmission slot using the calling signal and the number n (n =
(1), (2), (3), and (3), notifying a predetermined number n of the slave units in the sub-slave mode of the transmission data, and controlling the D / A conversion unit and the mixing control unit to receive the data. The audio signal of the slave unit is mixed with the audio data obtained by converting the uplink audio data into an audio signal. Forming transmission audio data at the coding rate and transmitting the data in the corresponding downlink transmission slot, and the control means of the predetermined number of slave units in the sub-slave mode respectively control the A / A according to the notified encoding speed. Forming upstream voice data to be transmitted by controlling the D-conversion means at a predetermined coding rate and transmitting the data in the notified upstream voice transmission slot; Fukusuko machine between call system in a digital cordless telephone system which is characterized in that to receive the audio signal. 3. The communication method between a plurality of slave units in the digital cordless telephone system according to claim 1 or 2, wherein the control unit of the slave unit in the main slave mode includes n slave units that are already in a call. To notify the use condition of the upstream voice transmission slot and the change of the coding rate of the transmission data on the out channel of the downstream voice channel, and transmit the upstream voice transmission slot on the out channel of the downstream voice channel to one new slave unit. The calling means including the notification of the use condition and the transmission speed of the transmission data is transmitted, and the control means of the n slave units which are already in the telephone call communicates at the coding speed as instructed in accordance with the change notification. The control means of one new slave unit controls the A / D conversion means in accordance with the notified encoding speed and transmits the upstream audio data. Predetermined form by the coding rate, notified Fukusuko machine between call system in a digital cordless telephone system and transmits an uplink voice transmission slot. 4. The communication method between a plurality of slave units in the digital cordless telephone system according to claim 1 or 2, wherein the control unit of the slave unit in the main slave mode includes n slave units that are already in a call. Executing the end-call sequence for one of the sub-units that ended the call, and using the up-speech transmission slot on the out-channel of the down-channel for the n-1 sub-units excluding the end-unit. And a notification of a change in the encoding speed of the transmission data, and the control means of the n-1 slave units that are already talking on the telephone perform control to shift to a call at the encoding speed as instructed in accordance with the change notification. A communication method between a plurality of cordless handsets in a digital cordless telephone system. 5. The Fukusuko machine between call system in a digital cordless telephone system according to any one of claims 1 to 4, wherein the A / D converter and D / A conversion means, m _i ( When m ₀ <m ₁ <m ₂ <m ₃ <m ₄ <...], (standard coding rate for system / m ₁ ), (standard coding rate for system / m ₂ ), (Standard coding speed as system / m ₃ ), voice coding data of speed of (standard coding speed as system / m ₄ )... The number is n (n = 1, 2,
When 3,4,5 ...) is the coding rate of the transmission voice data of the slave unit of the sub-child device mode, select the m _i as a _{m i-1 <n ≦ m} i, ( A communication method between a plurality of cordless handsets in a digital cordless telephone system, wherein the coding rate is determined to be a standard coding rate / m _i ). 6. A digital cordless telephone system comprising a wireless base station and a plurality of slaves connected to the wireless base station via a wireless line, wherein the wireless base station comprises: A communication method between a plurality of slave units in a digital cordless telephone system, which has a configuration similar to that of the slave unit in the main slave mode described in 1 above, and controls a call between the plurality of slave units via a wireless base station.