JP2009213084A - Telephone apparatus - Google Patents

Telephone apparatus Download PDF

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JP2009213084A
JP2009213084A JP2008056754A JP2008056754A JP2009213084A JP 2009213084 A JP2009213084 A JP 2009213084A JP 2008056754 A JP2008056754 A JP 2008056754A JP 2008056754 A JP2008056754 A JP 2008056754A JP 2009213084 A JP2009213084 A JP 2009213084A
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voice signal
voice
call
communication
received
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Yasushi Kudo
康 工藤
Shinji Nishimura
眞次 西村
Koichi Nagoya
光一 名児耶
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Nakayo Telecommunications Inc
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Nakayo Telecommunications Inc
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Abstract

<P>PROBLEM TO BE SOLVED: To provide a telephone apparatus in which a voice talk with another conference participant or an outside line partner is enabled during a voice conference and while continuously hearing both reception voices of another partner during and not during the conference, a transmission voice during the speech is transmitted only to the speaking partner. <P>SOLUTION: In the case that there is an incoming call from a second communication apparatus or there is an outgoing call to the second communication apparatus during a speech with a first communication apparatus, a reception voice output means converts a reception voice signal synthesized by a reception voice signal synthesizing means into a voice and outputs it, and a transmission voice signal switching and transmitting means performs switching to transmit to the second communication apparatus a transmission voice signal inputted by a transmission voice input means. <P>COPYRIGHT: (C)2009,JPO&INPIT

Description

本発明は、非会議通話と会議通話を同時に行う電話装置に関する。   The present invention relates to a telephone device that performs a non-conference call and a conference call simultaneously.

特許文献1には、音声会議中に特定の会議参加者との個別通話を行う電話会議装置が開示されている。   Patent Document 1 discloses a telephone conference apparatus that performs an individual call with a specific conference participant during an audio conference.

特開平8−018676号公報Japanese Patent Laid-Open No. 8-018676

特許文献1に記載の技術では、音声会議中に特定の会議参加者との個別通話を行うと、その間は音声会議の内容を聞く事が出来ない。また、取引相手等との重要な電話協議中に、電話会議装置から会議参加を促す呼び出しがあった場合に、当該通話中の呼を終了する必要がある。   With the technology described in Patent Document 1, if an individual call is made with a specific conference participant during an audio conference, the content of the audio conference cannot be heard during that time. In addition, when a call for inviting a conference is received from the conference call device during an important phone call with a business partner or the like, it is necessary to terminate the call during the call.

本発明の目的は、上記事情に鑑みてなされたものであり、音声会議中に他の会議参加者または非参加者との音声通話を可能とし、会議中と通話中の両方の受話音声を聞き続けながら、通話中の送話音声は通話相手のみに送信する電話装置を提供することにある。   The object of the present invention has been made in view of the above circumstances, and enables voice calls with other conference participants or non-participants during an audio conference and listens to received voices during and during the conference. It is another object of the present invention to provide a telephone device that transmits transmitted voice during a call only to the other party.

上述した課題は、複数の通信装置との通話機能を有し、第1の通信装置からの受話音声信号を受信する第1受話音声信号受信手段と、第2の通信装置からの受話音声信号を受信する第2受話音声信号受信手段と、第1受話音声信号受信手段が受信した第1の音声信号と第2受話音声信号受信手段が受信した第2の音声信号とを合成する受話音声信号合成手段と、受話音声信号を音声に変換して出力する受話音声出力手段と、送話音声を音声信号に変換して入力する送話音声入力手段と、送話音声入力手段が入力した送話音声信号を第1の通信装置または第2の通信装置のいずれかへ切り替えて送信する送話音声信号切替送信手段と、を有し、第1の通信装置との通話中に、かつ第2の通信装置からの着信があった場合または第2の通信装置への発信があった場合、受話音声出力手段は、受話音声信号合成手段が合成した受話音声信号を音声に変換して出力し、送話音声信号切替送信手段は、送話音声入力手段が入力した送話音声信号を第2の通信装置へ送信するように切り替える電話装置により、達成できる。   The above-described problem is that the first received voice signal receiving means for receiving a received voice signal from the first communication apparatus and the received voice signal from the second communication apparatus have a call function with a plurality of communication apparatuses. Second received voice signal receiving means for receiving, received voice signal synthesis for synthesizing the first voice signal received by the first received voice signal receiving means and the second voice signal received by the second received voice signal receiving means. Means, a received voice output means for converting a received voice signal into a voice and outputting it, a transmitted voice input means for converting a transmitted voice into a voice signal, and a transmitted voice input by the transmitted voice input means Transmission voice signal switching transmission means for switching and transmitting the signal to either the first communication device or the second communication device, and during the call with the first communication device, the second communication When there is an incoming call from the device or to the second communication device When there is a call, the reception voice output means converts the reception voice signal synthesized by the reception voice signal synthesis means into a voice and outputs it, and the transmission voice signal switching transmission means sends the transmission voice input by the transmission voice input means. This can be achieved by a telephone device that switches to transmit a speech signal to the second communication device.

本発明によれば、音声会議中に他の通話相手と音声通話を行っているときでも音声会議の音声が聞こえるため、音声会議の様子がわかり容易に音声会議へ戻ることができる。   According to the present invention, since the voice of the audio conference can be heard even when a voice call is performed with another call partner during the audio conference, the state of the audio conference can be easily understood and the user can easily return to the audio conference.

以下、本発明の実施の形態について、実施例を用い図面を参照しながら説明する。なお、実質同一部位には、同じ参照番号を用い、説明は繰り返さない。
まず、図1を参照して、電話システムを説明する。ここで、図1は、複数の相手との通話機能を有する電話装置を含む電話システムの接続構成図である。図1において、複数の相手との通話機能を有する電話装置200は、IP網100に接続されたルータ(またはゲートウェイ)600が接続されたLAN回線700に接続される。IP網100には、電話会議を実現する多地点接続装置(MCU:Multipoint Control Unit)400とSIPサーバ500と電話会議を行うIP機器300−1と300−2が接続され、さらに非会議通話を行うIP機器300−3も接続される。
Hereinafter, embodiments of the present invention will be described with reference to the drawings using examples. Note that the same reference numerals are used for substantially the same parts, and description thereof will not be repeated.
First, a telephone system will be described with reference to FIG. Here, FIG. 1 is a connection configuration diagram of a telephone system including a telephone device having a function of calling with a plurality of parties. In FIG. 1, a telephone device 200 having a call function with a plurality of parties is connected to a LAN line 700 to which a router (or gateway) 600 connected to an IP network 100 is connected. The IP network 100 is connected to a multipoint connection unit (MCU: Multipoint Control Unit) 400 that realizes a conference call and an IP device 300-1 and 300-2 that performs a conference call with a SIP server 500, and further performs a non-conference call. The IP device 300-3 to be performed is also connected.

電話装置200が、複数のIP機器300−1と300−2と電話会議を行うには、次の手順による。まず、電話装置200は、MCU400と接続するためにSIPサーバ500へ発信を行う。SIPサーバ500は、MCU400へ接続を行い、音声の通話路を電話装置200とMCU400との間で確立する。   The telephone device 200 performs a telephone conference with the plurality of IP devices 300-1 and 300-2 by the following procedure. First, the telephone device 200 makes a call to the SIP server 500 in order to connect to the MCU 400. The SIP server 500 connects to the MCU 400 and establishes a voice communication path between the telephone device 200 and the MCU 400.

さらに、IP機器300−1とIP機器300−2も同様に、MCU400と接続するためにSIPサーバ500へ発信を行って、MCU400と接続を行い、音声の通話路をMCU400とIP機器300−1との間、さらにMCU400とIP機器300−2との間で確立する。   Similarly, the IP device 300-1 and the IP device 300-2 make a call to the SIP server 500 to connect to the MCU 400, connect to the MCU 400, and connect the voice communication path between the MCU 400 and the IP device 300-1. Between the MCU 400 and the IP device 300-2.

MCU400は、接続された電話装置200、IP機器300−1、IP機器300−2の音声をミキシング(合成)して、複数の相手による電話会議を実現する。電話会議中に電話装置200へIP機器300−3から着信があると、電話装置200は、SIPサーバ500より着信を受ける。電話装置200は、IP機器300−1とIP機器300−2との電話会議を継続したままでIP機器300−3からの着信を受け付け通話する。   The MCU 400 mixes (synthesizes) voices of the connected telephone device 200, the IP device 300-1, and the IP device 300-2 to realize a conference call by a plurality of parties. If there is an incoming call from the IP device 300-3 to the telephone device 200 during the conference call, the telephone device 200 receives the incoming call from the SIP server 500. The telephone device 200 accepts an incoming call from the IP device 300-3 and makes a call while continuing the conference call between the IP device 300-1 and the IP device 300-2.

このとき、電話装置200は、MCU400からの受話の音声は受信するが、MCU400に対して電話装置200の送話の音声は送信しない。これより、電話装置200の送話は、着信のあったIP機器300−3に対して行い、MCU400へ接続しているIP機器300−1とIP機器300−2へは送信しない。このため電話装置200の音声は、IP機器300−1とIP機器300−2では、聞こえない。しかし、電話装置200は、MCU400からの受話の音声は受信しているため、IP機器300−3と通話を行っても、IP機器300−1とIP機器300−2の音声は聞くことができ、容易に電話会議に復帰することができる。   At this time, the telephone device 200 receives the voice of reception from the MCU 400 but does not transmit the voice of transmission of the telephone device 200 to the MCU 400. As a result, the telephone device 200 transmits to the IP device 300-3 that has received the incoming call, and does not transmit to the IP device 300-1 and the IP device 300-2 connected to the MCU 400. For this reason, the voice of the telephone device 200 cannot be heard by the IP device 300-1 and the IP device 300-2. However, since the telephone device 200 receives the voice of the incoming call from the MCU 400, the voice of the IP device 300-1 and the IP device 300-2 can be heard even when a call is made with the IP device 300-3. You can easily return to the conference call.

図2を参照して、複数の相手との通話機能を有する電話装置を説明する。ここで、図2は電話装置の機能ブロック図である。図2において、電話装置200は、キー入力部401と、電話会議登録処理部402と、参加者テーブル403と、主制御部404と、音声通話切替通知部405と、音声方向通知部406と、表示部407と、2つの受話音声信号受信処理部408と、受話音声合成部410と、受話レベル調整部411と、受話音声出力部412と、送話音声入力部413と、送話音声切替部414と、RTP処理部415と、SIP処理部416と、TCP/UDP/IP処理部417と、イーサネット(登録商標)処理部418を有する。   With reference to FIG. 2, the telephone apparatus which has a telephone call function with a some other party is demonstrated. Here, FIG. 2 is a functional block diagram of the telephone device. 2, the telephone device 200 includes a key input unit 401, a conference call registration processing unit 402, a participant table 403, a main control unit 404, a voice call switching notification unit 405, a voice direction notification unit 406, Display unit 407, two received voice signal reception processing unit 408, received voice synthesis unit 410, received level adjustment unit 411, received voice output unit 412, transmitted voice input unit 413, and transmitted voice switching unit 414, an RTP processing unit 415, an SIP processing unit 416, a TCP / UDP / IP processing unit 417, and an Ethernet (registered trademark) processing unit 418.

キー入力部401は、IP機器への発信や着信応答の操作による機能と電話会議中の送話の音声の切替の機能を有し、キー操作の情報を主制御部404へ通知する。キー入力部401は、さらに、キー操作情報を音声通話切替通知部405へ通知する。   The key input unit 401 has a function for switching to an IP device or an incoming call response and a function for switching the voice of a transmission during a conference call, and notifies the main control unit 404 of key operation information. The key input unit 401 further notifies the key operation information to the voice call switching notification unit 405.

電話会議登録処理部402は、主制御部404からの指示により通話中の相手のIP機器を電話会議中端末として登録する。登録したIP機器は、参加者テーブル403として保存する。   The telephone conference registration processing unit 402 registers the other party's IP device as a telephone conference terminal in response to an instruction from the main control unit 404. The registered IP device is stored as a participant table 403.

主制御部404は、キー入力部401から通知された情報と発信状態または着信状態を判定し、状態を遷移させる。また、音声の切替、音声レベルの調節などの指示を行い、音声送信方向の表示の指示を行う。音声通話切替通知部405は、複数の電話機との通話中に他の電話機との通話切替指示を主制御部404へ通知する。   The main control unit 404 determines the information notified from the key input unit 401 and the transmission state or the incoming state, and changes the state. Also, instructions such as voice switching and voice level adjustment are given, and instructions for displaying the voice transmission direction are given. The voice call switching notification unit 405 notifies the main control unit 404 of a call switching instruction with another telephone during a call with a plurality of telephones.

音声方向通知部406は、主制御部404からの指示により、電話装置200が、MCU400の会議通話へ音声を送信しているか、非会議通話を行うIP機器300−3へ音声を送信しているかの表示データを作成し、表示部407へ送信する。表示部407は、複数の相手との電話会議の通話中に他の非会議の相手との通話中であることを表示させるためにLCDなどへ表示を行う。   In response to an instruction from the main control unit 404, the voice direction notifying unit 406 transmits the voice to the conference call of the MCU 400 or transmits the voice to the IP device 300-3 that performs the non-conference call. Display data is generated and transmitted to the display unit 407. The display unit 407 displays on the LCD or the like in order to display that a call is being made with another non-conference partner during a conference call with a plurality of other parties.

受話音声信号受信処理部408−1は、電話会議の相手または非電話会議の第一の相手からの受話音声信号を受信し、受話音声合成部410へ送信する。受話音声信号受信処理部408−2は、電話会議の相手または非電話会議の第二の相手からの受話音声信号を受信し、受話音声合成部410へ送信する。   The received voice signal reception processing unit 408-1 receives the received voice signal from the other party of the telephone conference or the first party of the non-telephone conference, and transmits it to the received voice synthesis unit 410. The received voice signal reception processing unit 408-2 receives the received voice signal from the other party of the telephone conference or the second party of the non-telephone conference, and transmits it to the received voice synthesis unit 410.

ここで、第一の相手とは、IP機器300−1やIP機器300−2の会議通話を行う相手である。また、第二の相手とは、非会議通話を行うIP機器300−3である。受話音声合成部410は、第一の相手からの音声と第二の相手からの音声をミキシングし、音声合成を行う機能を有する。合成した音声は、受話音声出力部412へ通知する。   Here, the first partner is a partner who performs a conference call between the IP device 300-1 and the IP device 300-2. The second partner is the IP device 300-3 that performs a non-conference call. The received voice synthesizing unit 410 has a function of mixing voices from the first partner and voices from the second partner and performing voice synthesis. The synthesized voice is notified to the received voice output unit 412.

受話レベル調整部411は、受話音声合成部410に対して、主制御部404からの指示により、送話音声を送信していない相手からの受話音声の信号レベルを低下させる。   In response to an instruction from the main control unit 404, the reception level adjustment unit 411 lowers the signal level of the reception voice from the other party not transmitting the transmission voice to the reception voice synthesis unit 410.

受話音声出力部412は、受話音声信号を音声にD/A変換して出力する。送話音声入力部413は、送話音声を音声信号にA/D変換して入力する。送話音声切替部414は、送話音声入力部413から入力した音声を主制御部404の制御により、送話音声信号を第一の相手または第二の相手のいずれかへ切り替えて送信する。   The received voice output unit 412 D / A converts the received voice signal into voice and outputs the voice. The transmitted voice input unit 413 inputs A / D converted voice signals into voice signals. The transmission voice switching unit 414 transmits the voice input from the transmission voice input unit 413 while switching the transmission voice signal to either the first partner or the second partner under the control of the main control unit 404.

RTP処理部415は、音声データからIPパケットを形成する。RTP処理部415は、また、IPパケットから音声データを形成する。SIP処理部416は、通話するための発信や着信などのシーケンスを制御する。TCP/UDP/IP処理部417は、RTP処理部415またはSIP処理部416で形成したIPパケットをプロトコルとして送受信する。イーサネット処理部418は、IPパケットについてLANネットワーク700との間で送受信する。   The RTP processing unit 415 forms an IP packet from the voice data. The RTP processing unit 415 also forms voice data from the IP packet. The SIP processing unit 416 controls sequences such as outgoing and incoming calls for making a call. The TCP / UDP / IP processing unit 417 transmits and receives the IP packet formed by the RTP processing unit 415 or the SIP processing unit 416 as a protocol. The Ethernet processing unit 418 transmits and receives IP packets to and from the LAN network 700.

図3を参照して、複数の相手との通話機能を有する電話装置で会議通話中に非会議通話を行う動作を説明するためのシーケンスを説明する。ここで、図3はMCUとSIPサーバと電話装置と複数のIP機器との間のシーケンス図である。図3において、電話装置200は、会議通話を行うためにSIPサーバ500へ発信を行う(S501)。SIPサーバ500は、MCU400へ接続を行う(S502)。MCU400は、電話装置200へ応答を返送する(S503)。この結果、通話路ができて電話装置200とMCU400で通話状態となる。   With reference to FIG. 3, a sequence for explaining an operation of performing a non-conference call during a conference call by a telephone device having a call function with a plurality of opponents will be described. FIG. 3 is a sequence diagram among the MCU, the SIP server, the telephone device, and a plurality of IP devices. In FIG. 3, the telephone device 200 makes a call to the SIP server 500 to make a conference call (S501). The SIP server 500 connects to the MCU 400 (S502). The MCU 400 returns a response to the telephone device 200 (S503). As a result, a call path is established and the telephone device 200 and the MCU 400 enter a call state.

次に、IP機器300−1が会議通話を行うために同じく、SIPサーバ500へ発信を行う(S505)。SIPサーバ500は、MCU400へ接続を行う(S506)。MCU400は、IP機器300−1へ応答を返送する(S507)。この結果、通話路ができてIP機器300−1とMCU400で通話状態となる。   Next, the IP device 300-1 makes a call to the SIP server 500 in order to make a conference call (S505). The SIP server 500 connects to the MCU 400 (S506). The MCU 400 returns a response to the IP device 300-1 (S507). As a result, a call path is established and the IP device 300-1 and the MCU 400 are in a call state.

さらに、IP機器300−2が会議通話を行うために同じく、SIPサーバ500へ発信を行う(S509)。SIPサーバ500は、MCU400へ接続を行う(S510)。MCU400は、IP機器300−2へ応答を返送する(S511)。この結果、通話路ができてIP機器300−2とMCU400で通話状態となる。   Further, the IP device 300-2 makes a call to the SIP server 500 in order to make a conference call (S509). The SIP server 500 connects to the MCU 400 (S510). The MCU 400 returns a response to the IP device 300-2 (S511). As a result, a call path is established and the IP device 300-2 and the MCU 400 are in a call state.

MCU400は、電話装置200、IP機器300−1、IP機器300−2の音声を合成して、電話装置200、IP機器300−1、IP機器300−2へ送信することにより、3者による会議通話状態となる。   The MCU 400 synthesizes voices of the telephone device 200, the IP device 300-1, and the IP device 300-2, and transmits the synthesized speech to the telephone device 200, the IP device 300-1, and the IP device 300-2, thereby allowing a conference by three parties. It becomes a call state.

会議状態であるときに、第二の通話として電話装置200に対してIP機器300−3からの着信を説明する。IP機器300−3は、電話装置200へ発信するためにSIPサーバ500へ発信動作を行う(S513)。SIPサーバ500は、電話装置200へ着信する(S514)。電話装置200は、IP機器300−3に対して応答を送信する(S515)。電話装置200は、送話音声切替部414により、MCU400へ送出していた送話の音声を切り替えて(S516)、IP機器300−3へ送信し、電話装置200とIP機器300−3で通話状態となる。   An incoming call from the IP device 300-3 to the telephone device 200 as a second call when in a conference state will be described. The IP device 300-3 performs a call operation to the SIP server 500 in order to make a call to the telephone device 200 (S513). The SIP server 500 receives an incoming call to the telephone device 200 (S514). The telephone device 200 transmits a response to the IP device 300-3 (S515). The telephone device 200 switches the transmission voice transmitted to the MCU 400 by the transmission voice switching unit 414 (S516), and transmits the voice to the IP device 300-3. The telephone device 200 and the IP device 300-3 make a call. It becomes a state.

IP機器300−1はMUC400と通話状態を継続しており、IP機器300−2もMUC400と通話状態を継続している。そのため、会議通話は継続している。電話装置200は、MCU400から送られるIP機器300−1とIP機器300−2の第一の音声を受話レベル調整部411への制御により、IP機器300−3の第二の音声よりレベルを下げる制御を行う(S520)。この結果、MCU400からのIP機器300−1とIP機器300−2の受話の音声が小さくなり、IP機器300−3からの受話の音声が聞きやすくなる。ここで、MCU400から電話装置200への、片方向矢印は、片方向送話であることを示している。   The IP device 300-1 continues to talk with the MUC 400, and the IP device 300-2 also keeps talking to the MUC 400. Therefore, the conference call continues. The telephone device 200 lowers the level of the first voice of the IP device 300-1 and the IP device 300-2 sent from the MCU 400 from the second voice of the IP device 300-3 by controlling the reception level adjustment unit 411. Control is performed (S520). As a result, the voices received by the IP device 300-1 and the IP device 300-2 from the MCU 400 are reduced, and the voices received from the IP device 300-3 are easy to hear. Here, a one-way arrow from the MCU 400 to the telephone device 200 indicates a one-way transmission.

会議通話へ戻るときは、電話装置200の切替操作を受けて、送話音声切替部414にてIP機器300−3へ送信していた送話の音声をMCU400へ切り替える(S522)。MCU400を経由してIP機器300−1とIP機器300−2の会議通話を行うことが出来る。IP機器300−1とMCU400は通話が継続しており、またIP機器300−2とMCU400の通話も継続している。このため、会議通話を終了することなく、会議通話である第一の通話と非会議通話である第二の通話を自由に切り替えることができる。第一の通話に切り替えたことにより、IP機器300−3からの受話の音声レベルを下げ(S526)、IP機器300−3からの音声を受話しながら会議通話を行うことができる。ここで、IP機器300−3から電話装置200への、片方向矢印は、片方向送話であることを示している。   When returning to the conference call, in response to the switching operation of the telephone device 200, the transmission voice transmitted to the IP device 300-3 by the transmission voice switching unit 414 is switched to the MCU 400 (S522). A conference call between the IP device 300-1 and the IP device 300-2 can be performed via the MCU 400. IP device 300-1 and MCU 400 continue to talk, and IP device 300-2 and MCU 400 also continue talking. Therefore, the first call that is a conference call and the second call that is a non-conference call can be freely switched without terminating the conference call. By switching to the first call, the voice level of the incoming call from the IP device 300-3 can be lowered (S526), and the conference call can be performed while receiving the voice from the IP device 300-3. Here, a one-way arrow from the IP device 300-3 to the telephone device 200 indicates a one-way transmission.

図4を参照して、複数の相手との通話機能を有する電話装置の動作を説明する。ここで、図4は電話装置の動作フローチャートである。図4において、電話装置200は、キー操作(S601)により、キー入力部401で発信操作と、着信操作と、音声出力切替を判断する(S602、S613、S624)。また、主制御部404にて通話中であるかの状態を監視している(S623)
発信の場合(S602でYes)、主制御部404は、会議発信かを判定する(S603)。会議発信であれば(S603でYes)、主制御部404は、MCU400へSIPサーバ500を経由して発信する(S604)。MCU400からの応答を暫時待って、主制御部404は、応答を判定する(S605)。応答が無ければ(S605でNo)、主制御部404は、待機状態となる。
With reference to FIG. 4, the operation of the telephone device having a call function with a plurality of opponents will be described. Here, FIG. 4 is an operation flowchart of the telephone device. In FIG. 4, the telephone device 200 determines a call operation, an incoming call operation, and a voice output switch by the key input unit 401 by a key operation (S601) (S602, S613, S624). In addition, the main control unit 404 monitors whether or not a call is in progress (S623).
In the case of outgoing (Yes in S602), the main control unit 404 determines whether the outgoing call is a conference (S603). If it is a conference call (Yes in S603), the main control unit 404 sends the call to the MCU 400 via the SIP server 500 (S604). After waiting for a response from the MCU 400 for a while, the main control unit 404 determines the response (S605). If there is no response (No in S605), the main control unit 404 enters a standby state.

応答があると(S605でYes)、電話装置200は、MCU400との音声通話中となる(S606)。電話装置200の電話会議登録処理部402は、MCU400からの会議参加者情報を受信し(S607)、参加者テーブル403へ保存する(S608)。参加者テーブル403は、通話しているIP機器が会議通話であるか非会議通話であるかを認識し、第一と第二の通話相手の同時通話であることや送話方向の表示データに使用する。電話装置200は、音声方向通知部406よりMCU400へ通話の方向であることを表示して(S609)、待機状態(キー操作待ち)となる。   If there is a response (Yes in S605), the telephone device 200 is in a voice call with the MCU 400 (S606). The telephone conference registration processing unit 402 of the telephone device 200 receives the conference participant information from the MCU 400 (S607) and stores it in the participant table 403 (S608). The participant table 403 recognizes whether the IP device that is making a call is a conference call or a non-conference call. use. The telephone device 200 displays from the voice direction notification unit 406 that the call direction is to the MCU 400 (S609), and enters a standby state (waiting for key operation).

電話装置200の発信が非会議通話への発信のときは(S603でNo)、主制御部404は、非会議通話としてIP機器へ発信する(S610)。相手のIP機器からの応答を暫時待って、主制御部404は、応答を判定する(S611)。応答があれば(S611でYes)、主制御部404は、会議通話中かを判定する(S616)。会議通話中であれば(S616でYes)、送話音声切替部414より、MCU400との音声通話を着信があったIP機器300−3へ送話を切り替える(S617)。続けて、MCU400からの受話音声とIP機器300−3からの受話音声を受話音声合成部410で合成する(S618)。そして、主制御部404は、受話レベル調整部411を制御してMCU400からの受話音声のレベルを下げる(S619)。電話装置200の送話方向は、着信応答したIP機器300−3であるため、送話方向を表示部407を制御して、IP機器300−3への送話であることの表示を行う(S620)。そして、会議通話の第一の通話と非会議通話の第二の同時通話であることを表示する(S621)。電話装置200とIP機器300−3とで音声通話状態となって(S622)、主制御部404は待機状態となる。主制御部404は、複数の通話中でなければ(S616でNo)、そのままステップ622に遷移する。
主制御部404は、応答が無い場合(S611でNo)、電話装置200は、そのまま待機状態となる。
When the call from the telephone device 200 is a call to a non-conference call (No in S603), the main control unit 404 makes a call to the IP device as a non-conference call (S610). After waiting for a response from the counterpart IP device for a while, the main control unit 404 determines the response (S611). If there is a response (Yes in S611), the main control unit 404 determines whether a conference call is in progress (S616). If the conference call is in progress (Yes in S616), the transmission voice switching unit 414 switches the transmission to the IP device 300-3 that has received the voice call with the MCU 400 (S617). Subsequently, the received voice from the MCU 400 and the received voice from the IP device 300-3 are synthesized by the received voice synthesis unit 410 (S618). Then, the main control unit 404 controls the reception level adjustment unit 411 to lower the level of the reception voice from the MCU 400 (S619). Since the transmission direction of the telephone device 200 is the IP device 300-3 that responds to the incoming call, the transmission direction is controlled by the display unit 407 to display that the transmission is to the IP device 300-3 ( S620). Then, it is displayed that the first call of the conference call and the second simultaneous call of the non-conference call (S621). The telephone device 200 and the IP device 300-3 enter a voice call state (S622), and the main control unit 404 enters a standby state. If the plurality of telephone conversations are not being performed (No in S616), the main control unit 404 proceeds to step 622 as it is.
When there is no response (No in S611), the main control unit 404 is in a standby state as it is.

電話装置200のキー操作(S601)により、キー入力部401で着信の操作をした場合(S602でNoかつS613でYes)、主制御部404は、表示部407へ着信表示を行い(S614)、着信応答した後(S615)、ステップ616に遷移する。   When an incoming operation is performed at the key input unit 401 by key operation (S601) of the telephone device 200 (No at S602 and Yes at S613), the main control unit 404 displays an incoming call on the display unit 407 (S614). After answering the incoming call (S615), the process proceeds to step 616.

キー操作(S601)で、発信操作、着信操作以外の操作を行った場合(S602でNoかつS613でNo)、主制御部404は、通話中か判定する(S623)。通話中のとき(S623でYes)、主制御部404は、ステップ601のキー操作が切替か判定する(S624)。Yesのとき、ステップ616に遷移する。主制御部404はステップ624でNoのとき、待機状態となる。   When the key operation (S601) performs an operation other than the outgoing call operation and the incoming call operation (No in S602 and No in S613), the main control unit 404 determines whether a call is in progress (S623). When the call is in progress (Yes in S623), the main control unit 404 determines whether the key operation in Step 601 is switched (S624). If yes, the process proceeds to step 616. When the main controller 404 is No in step 624, the main controller 404 enters a standby state.

ステップ623でNoのとき、主制御部404は、複数通話終了か判定する(S625)。Yesのとき、主制御部404は同時通話を解除して(S626)、待機状態に遷移する。一方、ステップ625でNoのとき、そのまま待機状態に遷移する。   When No in step 623, the main control unit 404 determines whether a plurality of calls are finished (S625). When Yes, the main control unit 404 cancels the simultaneous call (S626) and transitions to a standby state. On the other hand, when the result is No in step 625, the state transits to the standby state as it is.

送話音声の切り替えは、第二の通話相手と通話中に特定のキー(例えば#キー)を押下すると、送話が第二の通話相手から第一の通話相手へ切り替わる。また、特定のキーを押下し続けると、一時的に第二の通話相手との通話から第一の通話相手へ送話が切り替わり、第一の通話相手へ話すことができる。このとき、特定のキー押下を止めると、通話していた第二の通話相手へ送話の方向が戻ることも出来る。   To switch the transmission voice, when a specific key (for example, # key) is pressed during a call with the second call partner, the transmission is switched from the second call partner to the first call partner. Further, when a specific key is continuously pressed, the transmission is temporarily switched from the call with the second call partner to the first call partner, and the first call partner can be talked to. At this time, if the pressing of a specific key is stopped, the direction of the transmission can be returned to the second calling party who was talking.

また、第一の通話の相手がMCU400ではなく、一般通話でも良く、一般通話中にMCU400への発信またはMCU400からの呼び出しによる通話を第二の通話として複数での通話でもよい。   Further, the other party of the first call may be a general call instead of the MCU 400, and a plurality of calls may be made with a call made to the MCU 400 or a call from the MCU 400 during the general call as a second call.

以上、本発明の複数の相手との通話機能を有する電話装置の実施例を説明したが、電話装置200は、MCU付きの電話装置でもよく、会議通話を行うIP機器300−1やIP機器300−2も本実施例の電話装置でも良い。   The embodiment of the telephone device having a call function with a plurality of opponents according to the present invention has been described above. However, the telephone device 200 may be a telephone device with an MCU, and the IP device 300-1 and the IP device 300 that perform a conference call. -2 may also be the telephone device of this embodiment.

電話システムの接続構成図である。It is a connection block diagram of a telephone system. 電話装置の機能ブロック図である。It is a functional block diagram of a telephone apparatus. MCUとSIPサーバと電話装置と複数のIP機器間のシーケンス図である。It is a sequence diagram among MCU, a SIP server, a telephone apparatus, and several IP apparatus. 電話装置の動作フローチャートである。It is an operation | movement flowchart of a telephone apparatus.

符号の説明Explanation of symbols

100…IP網、200…複数の相手との通話機能を有する電話装置、300…IP機器、400…多地点接続装置、500…SIPサーバ、600…ルータ、700…ローカルエリアネットワーク。   DESCRIPTION OF SYMBOLS 100 ... IP network, 200 ... Telephone apparatus which has a call function with a several party, 300 ... IP apparatus, 400 ... Multipoint connection apparatus, 500 ... SIP server, 600 ... Router, 700 ... Local area network.

Claims (4)

複数の通信装置との通話機能を有する電話装置において、
第1の通信装置からの受話音声信号を受信する第1受話音声信号受信手段と、第2の通信装置からの受話音声信号を受信する第2受話音声信号受信手段と、前記第1受話音声信号受信手段が受信した第1の音声信号と前記第2受話音声信号受信手段が受信した第2の音声信号とを合成する受話音声信号合成手段と、受話音声信号を音声に変換して出力する受話音声出力手段と、送話音声を音声信号に変換して入力する送話音声入力手段と、前記送話音声入力手段が入力した送話音声信号を前記第1の通信装置または前記第2の通信装置のいずれかへ切り替えて送信する送話音声信号切替送信手段と、を有し、
前記第1の通信装置との通話中に、前記第2の通信装置からの着信があった場合または前記第2の通信装置への発信があった場合、前記受話音声出力手段は、前記受話音声信号合成手段が合成した受話音声信号を音声に変換して出力し、前記送話音声信号切替送信手段は、前記送話音声入力手段が入力した送話音声信号を前記第2の通信装置へ送信するように切り替えることを特徴とする電話装置。
In a telephone device having a call function with a plurality of communication devices,
A first received voice signal receiving means for receiving a received voice signal from the first communication apparatus; a second received voice signal receiving means for receiving a received voice signal from the second communication apparatus; and the first received voice signal. Received voice signal synthesizing means for synthesizing the first voice signal received by the receiving means and the second voice signal received by the second received voice signal receiving means; and the received voice signal that converts the received voice signal into voice and outputs the voice. A voice output means; a voice input means for converting a voice to be converted into a voice signal; and a voice signal input by the voice input means for the first communication device or the second communication. Transmission voice signal switching transmission means for switching and transmitting to any of the devices,
When there is an incoming call from the second communication device or a call to the second communication device during a call with the first communication device, the received voice output means The received voice signal synthesized by the signal synthesizing means is converted into a voice and outputted, and the transmission voice signal switching transmission means sends the transmission voice signal inputted by the transmission voice input means to the second communication device. The telephone device is characterized in that switching is performed.
請求項1に記載の電話装置であって、
前記第1の通信装置との通信と前記第2の通信装置との通信とを同時に行っていることを表示する手段をさらに有することを特徴とする電話装置。
The telephone device according to claim 1,
The telephone apparatus further comprising means for displaying that communication with the first communication apparatus and communication with the second communication apparatus are simultaneously performed.
請求項1または請求項2に記載の電話装置であって、
着信時または発信時に当該通信装置が通信会議装置であるか否かを判定する通信会議判定手段をさらに有し、
前記第1の通信装置または前記第2の通信装置のいずれかが通信会議装置であると判定された場合、前記送話音声信号切替送信手段は、予め定められた特定のキーが押下されている間のみ、前記送話音声入力手段が入力した送話音声信号を前記通信会議装置と判定された側の通信装置へ送信することを特徴とする電話装置。
The telephone device according to claim 1 or 2, wherein
A communication conference determination means for determining whether the communication device is a communication conference device at the time of incoming call or outgoing call;
When it is determined that either the first communication device or the second communication device is a communication conference device, the transmission voice signal switching transmission unit has a predetermined specific key pressed. A telephone apparatus characterized by transmitting a transmission voice signal input by the transmission voice input means to a communication apparatus on the side determined to be the communication conference apparatus only during a period.
請求項1ないし請求項3のいずれか一つに記載の電話装置であって、
前記受話音声信号合成手段は、前記送話音声信号切替送信手段が送話音声信号を送信していない側の通信装置から受話音声信号の音声レベルを低くして合成することを特徴とする電話装置。
The telephone device according to any one of claims 1 to 3,
The received voice signal synthesizing unit synthesizes the received voice signal by lowering the voice level of the received voice signal from the communication device on the side where the transmitted voice signal switching transmission unit does not transmit the transmitted voice signal. .
JP2008056754A 2008-03-06 2008-03-06 Telephone apparatus Pending JP2009213084A (en)

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Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2015525044A (en) * 2012-08-03 2015-08-27 クゥアルコム・インコーポレイテッドQualcomm Incorporated Voice call combination in multi-SIM devices

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2015525044A (en) * 2012-08-03 2015-08-27 クゥアルコム・インコーポレイテッドQualcomm Incorporated Voice call combination in multi-SIM devices
JP2017153097A (en) * 2012-08-03 2017-08-31 クゥアルコム・インコーポレイテッドQualcomm Incorporated Combining voice calls in multi-sim device

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