WO2014017455A1 - 通信システムと方法とプログラム - Google Patents
通信システムと方法とプログラム Download PDFInfo
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- WO2014017455A1 WO2014017455A1 PCT/JP2013/069843 JP2013069843W WO2014017455A1 WO 2014017455 A1 WO2014017455 A1 WO 2014017455A1 JP 2013069843 W JP2013069843 W JP 2013069843W WO 2014017455 A1 WO2014017455 A1 WO 2014017455A1
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
- H04M1/724—User interfaces specially adapted for cordless or mobile telephones
- H04M1/72403—User interfaces specially adapted for cordless or mobile telephones with means for local support of applications that increase the functionality
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/80—Responding to QoS
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04B—TRANSMISSION
- H04B7/00—Radio transmission systems, i.e. using radiation field
- H04B7/14—Relay systems
- H04B7/15—Active relay systems
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/60—Network streaming of media packets
- H04L65/75—Media network packet handling
- H04L65/765—Media network packet handling intermediate
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M1/00—Substation equipment, e.g. for use by subscribers
- H04M1/253—Telephone sets using digital voice transmission
- H04M1/2535—Telephone sets using digital voice transmission adapted for voice communication over an Internet Protocol [IP] network
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M11/00—Telephonic communication systems specially adapted for combination with other electrical systems
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M7/00—Arrangements for interconnection between switching centres
- H04M7/006—Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer
Definitions
- the present invention is based on a Japanese patent application: Japanese Patent Application No. 2012-163959 (filed on July 24, 2012), and the entire description of the application is incorporated herein by reference.
- the present invention relates to a communication system, method, and program.
- a system using a virtual machine is known.
- a virtual machine also referred to as “virtual terminal” or “virtual client”
- a virtual OS Operating System
- client OS an application on the virtual machine. Execute.
- the terminal 170A is communicatively connected to the server apparatus 110A via the network, operates the virtual terminal on the server apparatus 110A as if operating the real terminal, operates the application, and generates screen information, for example.
- the application on the virtual terminal transmits the screen information to the terminal 170A and displays the screen information on the display device of the terminal 170A.
- the terminal 170A of such a system is also referred to as a thin client terminal.
- a connection is made to a virtual terminal on the server device 110A (a terminal that virtualizes the terminal 170A), and the server device is accessed from the terminal 170A at home or away from home, You can safely connect to your own (terminal 170A) virtual terminal and perform your work. Since no data is left in the terminal 170A, for example, even if the user loses the terminal 170A, secret information, company information, and the like are not leaked to the outside.
- Patent Document 1 a response signal including a voice response signal for an email message received by a thin client terminal for communication is sent via the communication network to the sender of the email message.
- An information communication device for transmission to a communication point of a thin client terminal for converting an audio response signal into an audio data file to be transmitted to a communication network, an e-mail and an audio data file
- An e-mail server connected to the communication network, an identification unit for identifying the audio data file from the response signal, and an audio mail for receiving the audio data file from the identification unit and converting the audio data file into an audio stream Interface and the above voice stream to the recipient's voice mailbox
- a system including a voice mail system includes the recipient's voice mailbox is disclosed. This system enables a voice response signal to be sent as a response to an e-mail message from a thin client terminal for communication via a voice data file to a recipient's voice mailbox via a voice data file and transmitted to the recipient.
- Patent Document 2 discloses a call destination telephone number and a VoIP gateway included in a connection request transmitted from a telephone terminal. Based on the stored information, the connection request is transmitted to either an IP network or a PSTN (Public Switched Network), and if it is determined that communication via the IP network is impossible, the IP network A configuration for transmitting the connection request determined to be transmitted to the PSTN is disclosed.
- an accessory for an electronic device includes an interface for exchanging data between the accessory and the electronic device, and a VoIP circuit, and the VoIP circuit is included in the electronic device or the accessory.
- a configuration is disclosed that is configured to implement at least a portion of VoIP.
- Patent Documents 1-3 do not disclose a configuration for eliminating a voice call processing delay on a server of a virtual machine type thin client system.
- VoIP Voice Over IP: IP (Internet Protocol) network
- VoIP Voice Over IP: IP (Internet Protocol) network
- VoIP Voice Over IP
- IP Internet Protocol
- the delay of each process on the virtual terminal (virtual client unit) on the server device is added, so that the total delay from end to end is, for example, 1 second. It may be significantly longer than the above. For this reason, troubles may occur in voice calls.
- the present invention has been made in view of the above problems, and its purpose is to allow a terminal to access a server device via a network and perform a voice call with a connection destination terminal via the server device. It is an object to provide a communication system, a server device, a method, and a program that can reduce the processing delay in a virtual terminal that occurs when performing a call and improve the comfort of voice calls.
- the server device operates on the virtual client functioning as a virtual terminal of the terminal by an operation on the terminal.
- the terminal is a communication system that displays the information based on the information from the server device.
- the server device receives an operation signal or a packet from the terminal.
- a voice signal received from the terminal based on an instruction from the control unit when the control unit determines that the voice call is based on the operation signal or the packet;
- a communication system including a relay unit that relays a packet storing data to a connection partner terminal without going through the virtual client. That.
- a server device that connects to a terminal via a network and transfers information obtained by operating an application on a virtual client functioning as a virtual terminal of the terminal to the terminal by an operation on the terminal.
- a control unit that receives an operation signal or packet from the terminal and determines from the operation signal or packet that the call is a voice call, and the control unit when the control unit determines that the call is a voice call
- a relay unit that relays a packet storing voice data received from the terminal to a connection partner terminal without passing through the virtual client based on an instruction from the terminal.
- a server device that connects to a terminal via a network and transfers information obtained by operating an application on a virtual client functioning as a virtual terminal of the terminal to the terminal by an operation on the terminal.
- On the computers that make up A process of receiving an operation signal or packet from the terminal and determining from the operation signal or packet that it is a voice call; A process for relaying a packet storing voice data received from the terminal to a connection partner terminal without passing through the virtual client based on an instruction from the control unit when it is determined as a voice call.
- a program for executing is provided.
- a computer-readable recording medium memory device, optical recording medium (device), magnetic storage medium (device)
- the server device connects to a terminal via a network and transfers information obtained by operating an application on a virtual client functioning as a virtual terminal of the terminal to the terminal by an operation on the terminal.
- a communication method The server device receives an operation signal or packet from the terminal and determines from the operation signal or packet that the call is a voice call, Provided is a communication method for relaying a packet storing voice data received from the terminal to a connection partner terminal without going through the virtual client when it is determined that the voice call is made.
- the present invention it is possible to reduce processing delay in a virtual terminal that occurs when a terminal accesses a server device via a network and makes a voice call with a counterpart terminal via the server device. Comfort can be improved.
- connection structure of illustrative 1st Embodiment of this invention It is a figure which shows the structure of the server apparatus of illustrative 1st Embodiment of this invention. It is a figure which shows the connection structure of the 2nd Embodiment of this invention. It is a figure explaining a virtual machine type thin client system.
- FIG. 1 is a diagram illustrating the configuration of the first exemplary embodiment of the present invention.
- a mobile 3G packet network is used as a network
- an SGSN Serving GPRS (General Packet Radio Service) Support Node)
- GGSN Gateway GPRS Support Node
- the configuration of is shown.
- an SGSN / GGSN device 190 represents a device in which an SGSN device and a GGSN device are integrated.
- a terminal (mobile terminal) 170_1 connects to the server apparatus 110 installed in the corporate network 130 and performs data transfer by a thin client
- the mobile terminal 170_1 uses the server apparatus 110 to perform the mobile terminal.
- the case where a voice call is performed with respect to 170_2 is illustrated.
- the telephone book 111 necessary for making a call is held in the server apparatus 110 of the thin client system.
- user names and telephone numbers are registered in the telephone directory 111 of the server apparatus 110.
- the portable terminals (170_1, 170_2) do not need to have a phone book. For this reason, when a portable terminal is lost, the security with respect to a telephone number, a user name, etc. can be ensured.
- a voice call VoIP application is executed on the virtual terminal of the server apparatus 110 (not shown in FIG. 1, corresponding to the virtual machine in FIG. 4).
- the screen data generated by the voice call VoIP application is transferred from the server device 110 to the mobile terminal 170_1, and the mobile terminal 170_1 decodes the screen data and displays it on a display device (not shown) of the mobile terminal 170_1.
- the user makes a voice call from the mobile terminal 170_1 to the mobile terminal 170_2 while watching a screen displayed on a display device (not shown) of the mobile terminal 170_1.
- the voice call VoIP application When the voice call VoIP application is activated on the virtual terminal (not shown) of the server apparatus 110 in order for the portable terminal 170_1 to start a voice call, the voice call VoIP is sent from the portable terminal 170_1 to the server apparatus 110.
- a packet storing an operation signal for starting the application is transmitted.
- the operation signal means a signal transmitted from the mobile terminal 170_1 by an operation such as a key operation, a screen touch operation, or a character input on the mobile terminal 170_1.
- the control unit (192 in FIG. 2) of the server device 110 determines that the voice call is made, and the voice call VoIP application is executed on the virtual terminal. (Corresponding to the application of FIG. 4) is activated, the voice call VoIP application generates screen information, encodes the screen information, and transfers it from the server device 110 to the portable terminal 170_1.
- the screen information is decoded by the mobile terminal 170_1, the screen information is displayed on the display device (not shown) of the mobile terminal 170_1, and operations such as selection of a voice call partner user name and a telephone number are performed.
- the audio signal accompanying the screen information is processed by a path different from the path for voice call. That is, in the server apparatus 110, after being captured by the screen capture unit (180 in FIG. 2), the audio signal is compression-encoded by the audio encoder unit (189 in FIG. 2) to become a compression-encoded stream. The packet is sent to the portable terminal 170_1 as a packet different from the voice call according to the protocol.
- the mobile terminal 170_1 transmits a packet storing a session control message according to a well-known session control protocol, and a packet storing a bit stream obtained by compressing and encoding an audio signal in the mobile terminal 170_1.
- SIP Session Initiation Protocol
- SIP Session Initiation Protocol
- the SGSN / GGSN device 190 is integrally configured, but it is needless to say that the SGSN device and the GGSN device may be separately arranged.
- FIG. 2 is a diagram illustrating a configuration of the server device 110.
- a virtual client unit 211 operates on a guest OS (virtual OS) in a virtual machine on a virtualization layer above the host OS.
- a well-known OS can be used as the host OS and guest OS.
- Linux registered trademark
- Android registered trademark
- Windows registered trademark
- the virtualization mechanism is not limited to a host OS type in which virtualization software is installed on a specific host OS and a virtual machine is operated on the virtualization software. You may comprise by a visor type.
- the virtual client unit 211 includes a control unit 192 and a screen generation unit 193.
- the mobile terminal 170_1 transmits an operation signal packet for starting the voice call VoIP application software on the virtual terminal in order to start the voice call.
- the packet receiving unit 186 of the server apparatus 110 receives the operation signal packet, extracts the operation signal, and outputs it to the control unit 192 of the virtual client unit 211.
- control unit 192 When the control unit 192 receives the operation signal and determines that it is a voice call, the control unit 192 executes the voice call VoIP application software. By executing the voice call VoIP application software, screen information for screen display to the end user (portable terminal 170_1) is generated, and the screen generation unit 193 generates a screen using the screen information and performs screen capture. Output to the unit 180.
- the screen capture unit 180 captures the screen generated by the screen generation unit 193 at a predetermined screen resolution and a predetermined frame rate (the number of frames per second), and outputs the captured image to the image encoder unit 188. .
- the image encoder unit 188 compresses and encodes the input screen using a predetermined encoder at a predetermined screen resolution, bit rate, and frame rate to obtain a compressed encoded stream.
- a compression encoding method H.264 is used.
- H.264 MPEG (Moving Picture Experts Group) -4, JPEG (Joint Photographic Experts Group) 2000, and other known compression coding systems can be used.
- the second packet transmission unit 176 stores the compressed encoded stream input from the image encoder unit 188 in a predetermined packet and outputs the packet to the SGSN / GGSN device 190 (see FIG. 1).
- the packet protocol may be RTP (Real-time Transport Protocol) / UDP (User Data Protocol) / IP (Internet Protocol), UDP / IP, or TCP (Transmission Control Protocol) / IP.
- UDP / IP is used as an example.
- the mobile terminal 170_1 receives the compression-encoded stream for the screen output from the server device 110 to the SGSN / GGSN device 190, and decodes the compressed stream with a predetermined screen resolution and frame rate. Then, it is displayed on the display device (not shown in FIG. 1) of the portable terminal 170_1.
- the virtual client unit (virtual terminal) 211 reads the destination telephone number (in this case, the telephone number of the mobile terminal 170_2 (see FIG. 1)) and the user name from the telephone directory 111, and uses this as screen information to carry the mobile phone.
- Terminal 170_1 send to FIG. 1).
- the mobile terminal 170_1 When the user of the mobile terminal 170_1 selects a user name and a telephone number on the screen while watching this screen on the display device (not shown in FIG. 1) of the mobile terminal 170_1 and starts a call, the mobile terminal 170_1 A packet storing the SIP message is transmitted to the server apparatus 110. Subsequently, the mobile terminal 170_1 transmits a packet storing a stream obtained by compressing and encoding the audio signal to the server apparatus 110.
- the processing delay is shortened by processing the voice call packet from the portable terminal 170_1 using a path different from the audio accompanying the screen.
- the packet reception unit 186 of the server device 110 includes, among the packets received from the mobile terminal 170_1 (see FIG. 1). Output the packet storing the SIP message to the control unit 192 of the virtual client unit 211; A packet (voice call packet) in which a compressed and encoded stream for voice is stored is output to the voice call packet relay unit 185.
- a compression encoding method for speech G.I. 711, G.G. 711.1, G.G. 729, AMR_NB (Adaptive Multi-Rate Narrow Band), AMR-WB (Adaptive Multi-Rate Wide Band), and the like can be used.
- control unit 192 of the virtual client unit 211 When receiving the operation signal, the control unit 192 of the virtual client unit 211 performs the following operation.
- the control unit 192 of the virtual client unit 211 analyzes the operation signal from the mobile terminal 170_1 (FIG. 1), and uses the voice call VoIP application software as an application that operates on the virtual machine (virtual client unit 211). to start.
- the control unit 192 of the virtual client unit 211 receives the SIP message from the packet reception unit 186.
- the control unit 192 of the virtual client unit 211 obtains the destination telephone number selected by the end user (the user of the portable terminal 170_1) from the VoIP application software, and obtains the destination IP address from the telephone number. Then, after rewriting the destination IP address of the received SIP message with the obtained IP address, the first packet transmission unit 187 is notified of the rewritten SIP message and the destination IP address.
- the control unit 192 of the virtual client unit 211 outputs an instruction to the voice call packet relay unit 185 so as to relay the packet in which the compressed encoded stream for the voice call is stored as it is.
- the voice call packet relay unit 185 receives a packet storing a compressed encoded stream for the voice call received from the portable terminal 170_1 (FIG. 1) by the packet receiving unit 186, and relays it from the control unit 192 of the virtual client unit 211.
- the input packet is relayed and output to the first packet transmitting unit 187 without processing delay in the virtual client unit 211 (without passing through the processing in the virtual client unit 211).
- the first packet transmission unit 187 receives the destination IP address and the SIP message from the control unit 192 of the virtual client unit 211, and the voice call packet relay unit 185 stores a packet in which a compressed encoded stream for the voice call is stored. Is sent to the mobile terminal 170_2 (see FIG. 1) via the SGSN / GGSN device 190 (see FIG. 1) or the like. In addition, the voice call packet relay unit 185 changes the protocol of the packet, if necessary, for the packet storing the compressed encoded stream for the voice call, and then the SGSN / GGSN device 190 (see FIG. 1). Etc., and sent to the portable terminal 170_2 (see FIG. 1).
- the protocol may be changed to the RTP / UDP / IP protocol, the UDP / IP protocol, or, in some cases, the TCP / IP protocol.
- the SGSN / GGSN device 190 tunnels the packet received from the server device 110 by, for example, GTP (GPRS Tunneling Protocol) -U protocol, and forwards the packet to the RNC device 195_2.
- GTP GPRS Tunneling Protocol
- the RNC device 195_2 It is transmitted to the portable terminal 170_2 wirelessly through 194_2.
- the mobile 3G network has been described as an example of the mobile network 150, but a mobile LTE (Long Term Evolution) network may be used. Also, a fixed network, a wireless local area network (WiFi) network such as WiFi (Wireless Fidelity), an NGN (Next Generation Network) network, a W-LAN network, an Internet network, or the like may be used.
- WiFi wireless local area network
- WiFi Wireless Fidelity
- NGN Next Generation Network
- W-LAN network Wireless Local Area Network
- Internet network or the like.
- the server apparatus 110 includes a virtual PC obtained by virtualizing a PC as a virtual terminal as a virtual machine.
- a screen generated by starting an application (voice call VoIP application software) in the virtual client unit 211 of the server apparatus 110 is captured by the screen capture unit 180, and is compressed and encoded by the image encoder unit 188.
- the configuration has been described in which the packet transmission unit 176 packetizes and sends the packet to the mobile terminal, and the mobile terminal decodes and displays the screen.
- the server apparatus 110 does not compress the screen and sends it to the mobile terminal.
- the server device 110 may send a drawing command for drawing the screen to the mobile terminal, receive the drawing command at the mobile terminal, and draw and display the screen using the drawing command.
- the server apparatus 110 receives a relay instruction from the control unit 192 (FIG. 2) in the server apparatus 110 for the packet in which the compressed encoded stream for the voice call from the partner mobile terminal 170_2 (FIG. 1) is stored.
- the unit 185 relays the packet without delay, and transmits the packet from the first packet transmission unit 187 to the mobile terminal 170_1 (see FIG. 1).
- the virtual client unit 211 of the server device 110 and the processing of each unit may be realized by a program executed by the server device 110.
- a storage medium such as a semiconductor memory device (programmable non-volatile memory) and a magnetic / optical disk in which the program is recorded is provided.
- the embodiment is realized by installing and executing the server apparatus 110.
- FIG. 3 is a diagram illustrating the configuration of the second embodiment.
- the server apparatus 110 is arranged in the cloud 230 and connected to the server apparatus 110 via a mobile network 150 from a mobile terminal (for example, 170_1).
- the cloud (cloud network) 230 may be a network infrastructure for cloud computing, and may be the Internet or an IP-VPN (Virtual Private Network) constructed by connecting IP networks.
- IP-VPN Virtual Private Network
- a mobile LTE network may be used instead of the mobile network 150.
- a fixed network an NGN network, a W-LAN network, an Internet network, or the like can also be used.
- a fixed terminal can be used instead of the portable terminal.
- a screen generated by starting an application in the virtual client unit of the server apparatus 110 is captured, compressed and encoded by the image encoder, packetized by the second packet transmission unit, and sent to the mobile terminal.
- the drawing command for drawing the screen is sent to the portable terminal, and the portable terminal is displayed.
- the drawing command may be received and used to draw and display the screen.
- the problem that the delay becomes significantly longer than that of normal VoIP not using the thin client method can be solved. it can.
- an operation signal or a packet is received from the terminal to determine whether the call is a voice call.
- the packet storing the voice data transmitted from the terminal is relayed. This is because there is almost no delay and the data can be sent to the other party.
- 110, 110A Server device 111 Telephone book 130 Corporate network 150 Mobile network 170_1, 170_2, 170A Terminal (portable terminal) 176 Second packet transmitter 177 Third packet transmitter 180 Screen capture unit 185 Voice call packet relay unit 186 Packet receiver 187 First packet transmitter 188 Image encoder unit 189 Audio encoder unit 190 SGSN / GGSN device 192 Control Unit 193 screen generation unit 194_1, 194_2 base station 195_1, 195_2 RNC device 211 virtual client unit 230 cloud network
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Abstract
Description
本発明は、日本国特許出願:特願2012-163959号(2012年7月24日出願)に基づくものであり、同出願の全記載内容は引用をもって本書に組み込み記載されているものとする。
本発明は、通信システムと方法とプログラムに関する。
・音声データの復号、
・音声データの再符号化、
・音声データの再パケット化
等の処理が入るためである。
前記端末から操作信号又はパケットを受信し前記操作信号又は前記パケットから、音声通話であることを判別する処理と、
音声通話と判別された場合に、前記制御部からの指示に基づき、前記端末から受信した、音声データが格納されたパケットを、前記仮想クライアントを経由せずに、接続相手端末宛に中継する処理と、
を実行させるプログラムが提供される。本発明によれば、該プログラムを記録したコンピュータ読み出し可能な記録媒体(メモリ装置、光記録媒体(装置)、磁気記憶媒体(装置))が提供される。
サーバ装置は、前記端末から操作信号又はパケットを受信し前記操作信号又は前記パケットから、音声通話であることを判別し、
音声通話と判別された場合に、前記端末から受信した、音声データが格納されたパケットを、前記仮想クライアントを経由せずに、接続相手端末宛に中継する、通信方法が提供される。
図1は、本発明の第1の実施形態の構成を例示した図である。特に制限されないが、図1では、ネットワークとしては、モバイル3Gパケットネットワークを用い、パケット転送装置として、SGSN(Serving GPRS(General Packet Radio Service) Support Node)/GGSN(Gateway GPRS Support Node)装置を用いる場合の構成が示されている。図1において、SGSN/GGSN装置190は、SGSN装置とGGSN装置を一体化した装置を表している。
・SIPメッセージが格納されたパケットを、仮想クライアント部211の制御部192に出力し、
・音声に対する圧縮符号化ストリームが格納されたパケット(音声通話のパケット)を、音声通話パケット中継部185に出力する。ここで、音声に対する圧縮符号化の方式としては、G.711、G.711.1、G.729、AMR_NB(Adaptive Multi-Rate Narrow Band)、AMR-WB(Adaptive Multi-Rate Wide Band)等、周知の圧縮符号化方式を用いることができる。
図3は、第二の実施の形態の構成を示す図である。図3を参照すると、サーバ装置110をクラウド230に配置し、携帯端末(例えば170_1)から、モバイルネットワーク150を経由して、サーバ装置110に接続する構成とされている。クラウド(クラウド網)230は、クラウドコンピューティングのネットワーク基盤を提供し、インターネットや、IP網同志を接続して構築したIP-VPN(Virtual Private Network:仮想施設網)等であってもよい。仮想化技術を利用することで、サーバリソース、ストレージリソース等はユーザ間でリソースの利用効率の向上、リソース分離によるセキュリティ確保等が行われる。図3において、図1の要素と同一又は同等の要素には、同一の参照符号が付されており、図1の要素と同じ働きをするので、説明は省略する。
111 電話帳
130 企業網
150 モバイルネットワーク
170_1、170_2、170A 端末(携帯端末)
176 第2のパケット送信部
177 第3のパケット送信部
180 画面キャプチャ部
185 音声通話パケット中継部
186 パケット受信部
187 第1のパケット送信部
188 画像エンコーダ部
189 オーディオエンコーダ部
190 SGSN/GGSN装置
192 制御部
193 画面生成部
194_1、194_2 基地局
195_1、195_2 RNC装置
211 仮想クライアント部
230 クラウド網
Claims (10)
- 端末と、ネットワークを介して前記端末に接続するサーバ装置と、を備え、前記サーバ装置は、前記端末での操作により、前記端末の仮想端末として機能する仮想クライアント上でアプリケーションを動作させて得た情報を前記端末に転送する通信システムであって、
前記サーバ装置が、
前記端末から操作信号又はパケットを受信し、前記操作信号又は前記パケットから音声通話であることを判別する制御部と、
前記制御部で音声通話と判別された場合に、前記制御部からの指示に基づき、前記端末から受信した、音声データが格納されたパケットを、前記仮想クライアントを経由せずに、接続相手の端末宛に中継する中継部と、
を備えた、ことを特徴とする通信システム。 - ネットワークを介して端末に接続し、前記端末での操作により、前記端末の仮想端末として機能する仮想クライアント上でアプリケーションを動作させて得た情報を前記端末に転送するサーバ装置であって、
前記端末から操作信号又はパケットを受信し前記操作信号又は前記パケットから、音声通話であることを判別する制御部と、
前記制御部で音声通話と判別された場合に、前記制御部からの指示に基づき、前記端末から受信した、音声データが格納されたパケットを、前記仮想クライアントを経由せずに、接続相手の端末宛に中継する中継部と、
を備えた、ことを特徴とするサーバ装置。 - 前記中継部で中継された音声データが格納されたパケットを前記接続相手の端末宛に送信する第1のパケット送信部と、
前記制御部で音声通話と判別された場合に、前記仮想クライアント上で音声通話アプリケーションソフトウェアを実行して得られる情報から画面を生成する画面生成部と、
前記生成された画面を第1のエンコーダで圧縮符号化して得たデータを格納したパケットを送信する第2のパケット送信部と、
を備えた、ことを特徴とする請求項2記載のサーバ装置。 - 前記生成された画面に含まれるオーディオ信号を第2のエンコーダで圧縮符号化して得たデータを格納したパケットを送信する第3のパケット送信部を備えた、ことを特徴とする請求項3記載のサーバ装置。
- 前記画面生成部で生成された画面を、予め定められた解像度、及び、フレームレートでキャプチャし、前記第1及び第2のエンコーダにそれぞれ出力する画面キャプチャ部を備えたことを特徴とする請求項4記載のサーバ装置。
- 前記仮想クライアントに、前記制御部と前記画面生成部を備えた、ことを特徴とする請求項3記載のサーバ装置。
- ネットワークを介して端末に接続し、前記端末での操作により、前記端末の仮想端末として機能する仮想クライアント上でアプリケーションを動作させて得た情報を前記端末に転送するサーバ装置を構成するコンピュータに、
前記端末から操作信号又はパケットを受信し前記操作信号又は前記パケットから、音声通話であることを判別する処理と、
音声通話と判別された場合に、前記制御部からの指示に基づき、前記端末から受信した、音声データが格納されたパケットを、前記仮想クライアントを経由せずに、接続相手の端末宛に中継する処理と、
を実行させるプログラム。 - ネットワークを介して端末に接続し、前記端末での操作により、前記端末の仮想端末として機能する仮想クライアント上でアプリケーションを動作させて得た情報を前記端末に転送するサーバ装置による通信方法であって、
サーバ装置では、
前記端末から操作信号又はパケットを受信し前記操作信号又は前記パケットから、音声通話であることを判別し、
音声通話と判別された場合に、前記端末から受信した、音声データが格納されたパケットを、前記仮想クライアントを経由せずに、相手先の端末宛に中継する、ことを特徴とする通信方法。 - 前記中継された音声データが格納されたパケットを第1のパケット送信部から送信し、
音声通話と判別された場合に、音声通話アプリケーションソフトウェアを実行して得られる情報から画面を生成し、
前記生成された画面を、第1のエンコーダで圧縮符号化して得たデータを格納したパケットを第2のパケット送信部から送信する、ことを特徴とする請求項8記載の通信方法。 - 前記生成された画面にオーディオ信号が含まれる場合、前記オーディオ信号を第2のエンコーダで圧縮符号化して得たデータを格納したパケットを第3のパケット送信部から送信する、ことを特徴とする請求項9記載の通信方法。
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JP2014526917A JP5920466B2 (ja) | 2012-07-24 | 2013-07-23 | 通信システムと方法とプログラム |
EP13822803.6A EP2879362A4 (en) | 2012-07-24 | 2013-07-23 | COMMUNICATION SYSTEM, METHOD AND PROGRAM |
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CN104509083A (zh) | 2015-04-08 |
JP5920466B2 (ja) | 2016-05-18 |
KR101709244B1 (ko) | 2017-02-23 |
EP2879362A4 (en) | 2016-03-30 |
JPWO2014017455A1 (ja) | 2016-07-11 |
KR20150029712A (ko) | 2015-03-18 |
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