WO2016183774A1 - 一种通话录音的方法、装置及系统 - Google Patents

一种通话录音的方法、装置及系统 Download PDF

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Publication number
WO2016183774A1
WO2016183774A1 PCT/CN2015/079195 CN2015079195W WO2016183774A1 WO 2016183774 A1 WO2016183774 A1 WO 2016183774A1 CN 2015079195 W CN2015079195 W CN 2015079195W WO 2016183774 A1 WO2016183774 A1 WO 2016183774A1
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WIPO (PCT)
Prior art keywords
recording
media stream
telephone terminal
call
pbx
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PCT/CN2015/079195
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English (en)
French (fr)
Inventor
杨一金
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华为技术有限公司
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Application filed by 华为技术有限公司 filed Critical 华为技术有限公司
Priority to PCT/CN2015/079195 priority Critical patent/WO2016183774A1/zh
Publication of WO2016183774A1 publication Critical patent/WO2016183774A1/zh

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L9/00—Cryptographic mechanisms or cryptographic arrangements for secret or secure communications; Network security protocols
    • H04L9/40—Network security protocols
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M7/00—Arrangements for interconnection between switching centres

Definitions

  • the embodiment of the invention relates to the field of communication services, and in particular, to a method, device and system for recording a call.
  • IP call recording services have become an important service in VoIP voice services, in the financial industry, insurance. Industry, government agencies and other fields are widely used, among which the point-to-point recording service is more commonly used, that is, point-to-point recording when user A and user B talk.
  • the user B of the prior art is a user with recording permission.
  • the user B can manually initiate a recording operation according to the need (if the automatic recording is performed, the user is not required. Manually start the recording operation), perform point-to-point recording, the specific steps are as follows:
  • Step 1 user A calls user B;
  • Step 2 user B answers
  • Step 3 User A and User B talk
  • step 4 user B manually starts the recording operation (if it is automatic recording, user B does not need to manually start the recording operation);
  • Step 5 After receiving the recording operation message, the IP-based voice switch (IP-PBX, IP Private Branch eXchange) creates a three-party conference site and uses the recording server (Recording Server) as a third-party participant.
  • the second party is the user B) pulled into the three-party venue;
  • Step 6 The IP-PBX gateway sends the media stream of the venue mixing to the recording server;
  • step 7 the recording server generates and stores the recording file and the recording data.
  • the solution is to use the IP-PBX gateway in the IP-PBX system to pull the recording server as a third-party participant into the conference site through the three-party venue (mixing of the venue), so that the media stream received by the recording server is IP-PBX.
  • the media stream after the gateway site is mixed, and finally the recording server generates and stores the recording file and the recording data.
  • DSP digital signal processor
  • the user B of the prior art 2 is a user with recording permission.
  • the user B can manually start the recording operation as needed (if it is automatic recording, the user is not required. Manually start the recording operation), perform point-to-point recording, the specific process is as follows:
  • step 1 user A calls user B.
  • Step 2 User B answers.
  • Step 3 In the automatic recording mode, the recording process is automatically activated (if it is a manual recording, the IP phone B of the user B sends a start recording message to the IP-PBX gateway).
  • Step 4 The IP-PBX gateway calls the BIB (built-in bridge) module of the recorded IP phone B to record the RTP media stream generated by the user, and the BIB module will automatically answer.
  • BIB built-in bridge
  • Step 5 The IP-PBX gateway sends a message to the recording server to start recording of the IP phone B (via the SIP protocol), and the recorded phone B starts to copy its own RTP media stream 2 and transmits it to the recording server.
  • Step 6 The IP-PBX gateway sends a recording start message of the user's phone A to the recording server (via the SIP protocol), and the recorded phone B starts to copy the RTP media stream 1 generated by the phone A and transmits it to the recording server.
  • step 7 the recording server generates and stores the recording file and the recording data.
  • the IP phone B copies the RTP media stream 2 generated by itself to the recording server during the call, and simultaneously copies the received RTP media stream 1 of the peer IP telephone terminal A.
  • One copy is sent directly to the recording server, and two recording files and recording data are generated and stored by the recording server.
  • the inventor has found that the solution of the prior art 2 has the following drawbacks: (1) The recorded IP phone B needs to copy two media streams to the recording server, so two recording files are generated on the recording server side, and the recording server needs to further The two recording files are combined into one, resulting in an increase in the processing traffic of the recording server. (2) The IPB module needs to be additionally set in the recorded IP phone B, resulting in an increase in the complexity and cost of the IP phone B.
  • the purpose of the embodiments of the present invention is to provide a call recording method, apparatus, and system capable of reducing the occupation of DSP media resources of an IP-PBX gateway and satisfying the demand for point-to-point recording of large-capacity users.
  • a call recording method including the following steps:
  • the IP telephone terminal receives the voice signal of the local user, and converts the voice signal into a first media stream;
  • the IP telephone terminal receives the second media stream sent by the peer communication device
  • the digital signal processor DSP module of the IP telephone terminal performs mixing processing on the first media stream and the second media stream to obtain a third media stream;
  • the IP telephone terminal sends the third media stream to the recording server for recording.
  • the method further includes:
  • the IP telephone terminal initiates a recording operation to obtain the media address and port number of the recording server.
  • the step of transmitting the third media stream to the recording server at the IP telephone terminal in conjunction with the first aspect or the first possible implementation of the first aspect, in a second possible implementation of the first aspect, the step of transmitting the third media stream to the recording server at the IP telephone terminal Previously, it also included:
  • the IP phone terminal receives the first Re-Invite message sent by the IP-PBX network, temporarily holds the current call through the first Re-Invite message, and simultaneously plays a voice prompt tone, prompting the user that the call will be recorded;
  • the IP telephone terminal receives the second Re-Invite message sent by the IP-PBX network, and restores the currently held call to the call state by using the second Re-Invite message.
  • the step of the IP phone terminal to start a recording operation, so as to obtain a media address and a port number of the recording server specifically includes:
  • the IP phone terminal sends an Invite message to the IP-PBX gateway, where the Invite message carries the operation code of the recording service and the recording service identifier;
  • the IP-PBX gateway initiates a call to the recording system via an Invite message
  • the IP-PBX gateway obtains the media address and port number of the recording server
  • the IP-PBX gateway sends a message to the recording system to start recording
  • the IP telephone terminal receives the message sent by the IP-PBX gateway, where the message carries the media address and port number of the recording server.
  • an IP telephone terminal for call recording including:
  • a microphone for converting the voice of the local user into an electrical signal and transmitting the signal to the DSP module of the digital signal processor;
  • a digital signal processor DSP module configured to process the electrical signal to be converted into a first volume media stream signal, and further configured to process the second media stream signal sent by the peer-to-peer device to be converted into the speaker Played voice signal;
  • the DSP module has a mixing unit, configured to perform mixing processing on the first media stream and the second media stream, generate a third media stream after mixing, and combine the third media stream. Send to the recording server.
  • a call recording system comprising:
  • the IP telephone terminal further includes: a speaker for playing the received voice signal to the user; and a microphone for converting the voice of the local user into an electrical signal and transmitting the signal to the digital signal processor DSP module; the digital signal processor DSP a module, configured to process the electrical signal into a first volume media stream signal, and further configured to process the second media stream signal sent by the peer-to-peer device to be converted into a voice signal that can be played by the speaker; And a mixing unit, configured to mix the first media stream and the second media stream, generate a third media stream after mixing, and send the third media stream to a recording server;
  • the peer-to-peer device is configured to establish a call with the IP telephone terminal, receive the first physical media stream signal by the IP telephone terminal, and send a second body media stream signal to the IP telephone terminal;
  • An IP-PBX gateway configured to instruct the recording server to start recording and stop recording, and obtain a media address and a port number of the recording server, and send the same to the IP telephone terminal;
  • a recording server configured to record the received third media stream according to an instruction of the IP-PBX gateway.
  • the method before the step of the IP telephony terminal transmitting the third media stream to the recording server, the method further includes:
  • the IP phone terminal receives the first Re-Invite message sent by the IP-PBX network, temporarily holds the current call through the first Re-Invite message, and simultaneously plays a voice prompt tone, prompting the user that the call will be recorded;
  • the IP telephone terminal receives the second Re-Invite message sent by the IP-PBX network, and restores the currently held call to the call state by using the second Re-Invite message.
  • the IP phone terminal is further configured to send an Invite message to the IP-PBX gateway, where the Invite message carries an operation code and a recording of the recording service.
  • the IP-PBX gateway is further configured to initiate a call to the recording server to obtain a media address and a port number of the recording server;
  • the IP telephone terminal receives a message sent by the IP-PBX gateway, where the message carries a media address and a port number of the recording server.
  • the utility model has the beneficial effects that the mixing processing unit of the IP telephone terminal performs mixing processing on the first media stream of the local user and the second media stream sent by the peer-talking device to obtain the third media stream after the mixing. Then, the IP telephone terminal sends the third media stream to the recording server for recording. Therefore, the occupation of the DSP media resources of the IP-PBX gateway can be reduced by the mixing process, and the point-to-point recording requirement of the large-capacity user can be simultaneously satisfied.
  • FIG. 1 is a schematic diagram of a call recording system of the prior art
  • FIG. 2 is a schematic diagram of a call recording system of the prior art 2;
  • FIG. 3 is a schematic diagram of a call recording system according to an embodiment of the present invention.
  • FIG. 4 is a schematic diagram of an IP telephone terminal according to an embodiment of the present invention.
  • Figure 5 is a flow chart of the method of the first embodiment of the present invention.
  • Figure 6 is a flow chart of a method of the second embodiment of the present invention.
  • Figure 7 is a flow chart of a method in accordance with a third embodiment of the present invention.
  • a call recording system is composed of two subsystems: an IP-PBX subsystem and a recording subsystem.
  • the IP-PBX subsystem further includes an IP-PBX gateway, an IP phone terminal A of User A, and an IP phone terminal B of User B.
  • the IP-PBX gateway is used to provide number registration and call connection. It is also responsible for SIP signaling interaction with the Session Initiation Protocol Server (SIP Server), and sends the start recording and stop recording request, call message, etc. to the SIP Server. And sending a real-time transport protocol (RTP) media stream or a Secure Real-time Transport Protocol (SRTP) media stream to the recording server.
  • RTP real-time transport protocol
  • SRTP Secure Real-time Transport Protocol
  • the recording subsystem further includes: a Recording Server, a SIP Server, a database, and an Integrated Management Platform (IMP).
  • the recording server is configured to receive the RTP/SRTP media stream sent by the IP-PBX subsystem, start recording and stop recording according to the instruction, and generate and store the recording file and the recording data.
  • the SIP Server is used to interact with the IP-PBX gateway for SIP messages, and sends a message to the recording server to start and stop recording.
  • the database is used to save the configuration parameters and recording data of the recording server.
  • the integrated management platform is used to manage the centralized configuration and maintenance of the recording server, and manages and configures various parameters of the recording server.
  • the IP telephone terminal of the present invention comprises: a DSP module, a speaker, a microphone, and a Graphical User Interface (GUI) module.
  • the speaker is used to play the received voice signal to the user
  • the microphone is used to convert the voice of the local user into an electrical signal and send it to the digital signal processor DSP module
  • the GUI module provides the user with a visual interface, such as a call.
  • the DSP module is configured to process the electrical signal and convert the signal into a first volume media stream signal, and also process the second media stream signal sent by the peer-to-peer device to be converted into a voice signal that can be played by the speaker;
  • the DSP module further has a mixing unit (DSP mixer) for mixing the first media stream and the second media stream, and mixing to generate a third media stream, and The third media stream is sent to the recording server.
  • the mixing unit can also perform mixing processing independently of the DSP module.
  • the processing flow of the mixing unit of the DSP module is as follows:
  • the DSP module receives the voice signal sent by the microphone of the local phone, and converts the voice signal into a first RTP media stream and sends the voice signal to the peer communication device.
  • the DSP module receives the second RTP media stream sent by the peer-to-peer device and converts it into voice Signal and send the voice signal to the speaker for playback.
  • the mixing unit of the DSP module mixes the sent first RTP media stream and the received second RTP media stream as needed, and generates a third RTP media stream after mixing, and the third RTP is generated.
  • the media stream is sent to a specific device on the network side (such as a recording server).
  • the method for manual recording of an IP telephone terminal according to the first embodiment of the present invention after a call includes the following steps:
  • the IP telephone terminal A of the user A and the IP telephone terminal B of the user B establish a call.
  • the IP telephone terminal B receives the media stream from the IP telephone terminal A, and on the other hand, the IP telephone terminal B transmits the media stream to the IP telephone terminal A.
  • the user B IP phone terminal B having the manual recording service permission initiates a recording operation, so that the IP-PBX gateway obtains the media address and port number of the recording server.
  • the manual recording service authority can be notified to the IP telephone terminal B by the IP-PBX gateway by means of service synchronization.
  • the step S102 specifically includes the following steps:
  • the IP telephone terminal B sends an Invite message to the IP-PBX gateway.
  • the Invite message carries an operation code (Access code) and a recording service identifier (recording: on) of the recording service.
  • the IP-PBX gateway initiates a call to the recording system through the Invite message, where the connection address in the SDP (Session Description Protocol) message body of the Invite message is the IP address and port number of the IP phone terminal B.
  • SDP Session Description Protocol
  • the recording system returns 180 ringing response code and 200 OK response code to the IP-PBX gateway.
  • the SDP message body carried in the 200 OK response code includes the media address and port number of the recording server.
  • the IP-PBX gateway After receiving the 200 OK response code, the IP-PBX gateway sends an ACK acknowledgement message to the recording system, and the IP-PBX gateway completes signaling negotiation with the recording system to obtain the media address and port number of the recording server.
  • the IP-PBX gateway starts recording, and the step S104 specifically includes the following steps:
  • the IP-PBX gateway sends an INFO message (including the recording recording start: start) to the recording system, and notifies the recording server to start recording.
  • the recording system feeds back a 200 OK response code to the IP-PBX gateway.
  • the IP-PBX gateway after receiving the 200 OK response code of the recording system, the IP-PBX gateway replies with a 200 OK response to the IP phone terminal B, wherein the SDP message body carried in the 200 OK response code includes the media address and port number of the recording server.
  • the IP telephone terminal B sends an ACK acknowledgement message to the IP-PBX gateway, and starts the DSP module to mix the sent first media stream and the received second media stream.
  • the IP-PBX gateway sends the first Re-Invite message to the IP telephone terminal A and the IP telephone terminal B, respectively.
  • the current call is temporarily held by the first Re-Invite message, and the recording prompt is played: "Your call will be recorded.".
  • the IP-PBX gateway After the recording prompt tone is played, the IP-PBX gateway sends a second Re-Invite message to the IP telephone terminal A and the IP telephone terminal B, respectively, and the currently held call is restored to the call state by using the second Re-Invite message. .
  • the IP telephone terminal B sends the third media stream generated after the mixing to the recording server corresponding to the media address and the port number.
  • the recording server generates and stores the recording file and the recording data according to the received third media stream.
  • the IP telephone terminal B As shown in FIG. 6, the IP telephone terminal B according to the second embodiment of the present invention, as a method for the called party to automatically record during a call, includes the following steps:
  • the IP phone terminal A of the user A calls the IP phone terminal B of the user B to establish a call.
  • the IP telephone terminal B receives the media stream from the IP telephone terminal A, and on the other hand, the IP telephone terminal B transmits the media stream to the IP telephone terminal A.
  • the IP telephone terminal B having the automatic recording service authority determines that automatic recording is required, and starts the recording operation, so that the IP-PBX gateway obtains the media address and port number of the recording server.
  • the automatic recording service authority can notify the IP telephone terminal B by the IP-PBX gateway by means of service synchronization.
  • the step S202 specifically includes the following steps:
  • the IP telephone terminal B determines that an automatic recording is required, and then sends an Invite message to the IP-PBX gateway.
  • the Invite message carries an operation code (Access code) and a recording service identifier (recording: on) of the recording service.
  • the IP-PBX gateway initiates a call to the recording system through the Invite message, where the connection address in the SDP message body of the Invite message is the IP address and port number of the IP phone B.
  • the recording system returns 180 ringing response code and 200 OK response code to the IP-PBX gateway.
  • the SDP message body carried in the 200 OK response code includes the media address and port number of the recording server.
  • the IP-PBX gateway after receiving the 200 OK response code, the IP-PBX gateway sends an ACK confirmation message to the recording system, and the IP-PBX gateway completes the signaling negotiation with the recording system to obtain the media location of the recording server. Address and port number.
  • the IP-PBX gateway starts recording, and the step S204 specifically includes the following steps:
  • the IP-PBX gateway sends an INFO message (including the recording recording start: start) to the recording system, and notifies the recording server to start recording.
  • the recording system feeds back a 200 OK response code to the IP-PBX gateway.
  • the IP-PBX gateway after receiving the 200 OK response code of the recording system, replies with a 200 OK response to the IP phone terminal B, wherein the SDP message body carried in the 200 OK response code includes the media address and port number of the recording server.
  • the IP telephone terminal B sends an ACK acknowledgement message to the IP-PBX gateway, and starts the DSP module to mix the sent first media stream and the received second media stream.
  • the IP-PBX gateway sends the first Re-Invite message to the IP telephone terminal A and the IP telephone terminal B, respectively.
  • the current call is temporarily held by the first Re-Invite message, and the recording prompt is played: "Your call will be recorded.".
  • the IP-PBX gateway sends a second Re-Invite message to the IP telephone terminal A and the IP telephone terminal B, respectively, and the currently held call is restored to the call state by using the second Re-Invite message. .
  • the IP telephone terminal B sends the third media stream generated after the mixing to the recording server corresponding to the media address and the port number.
  • the recording server generates and stores the recording file and the recording data according to the received third media stream.
  • the IP telephone terminal B of the third embodiment of the present invention as a method for the caller to automatically record during a call, includes the following steps:
  • user B's IP telephone terminal B calls user A's IP telephone terminal A to establish a call.
  • the IP telephone terminal B receives the media stream from the IP telephone terminal A, and on the other hand, the IP telephone terminal B transmits the media stream to the IP telephone terminal A.
  • the IP telephone terminal B with the automatic recording service authority determines that automatic recording is required, and starts the recording operation, so that the IP-PBX gateway obtains the media address and port number of the recording server.
  • the automatic recording service authority can notify the IP telephone terminal B by the IP-PBX gateway by means of service synchronization.
  • the step S302 specifically includes the following steps:
  • the IP telephone terminal B determines that an automatic recording is required, and then sends an Invite message to the IP-PBX gateway.
  • the Invite message carries the operation code (Access code) and recording service of the recording service.
  • Identification on recording: on).
  • the IP-PBX gateway initiates a call to the recording system through the Invite message, where the connection address in the SDP message body of the Invite message is the IP address and port number of the IP Phone B.
  • the recording system returns 180 ringing response code and 200 OK response code to the IP-PBX gateway.
  • the SDP message body carried in the 200 OK response code includes the media address and port number of the recording server.
  • the IP-PBX gateway After receiving the 200 OK response code, the IP-PBX gateway sends an ACK acknowledgement message to the recording system, and the IP-PBX gateway completes signaling negotiation with the recording system to obtain the media address and port number of the recording server.
  • the IP-PBX gateway starts recording, and the step S304 specifically includes the following steps:
  • the IP-PBX gateway sends an INFO message (including the recording starting: start) to the recording system, and notifies the recording server to start recording.
  • the recording system feeds back a 200 OK response code to the IP-PBX gateway.
  • the IP-PBX gateway after receiving the 200 OK response code of the recording system, the IP-PBX gateway replies with a 200 OK response to the IP telephone terminal B, wherein the SDP message body carried in the 200 OK response code includes the media address and port number of the recording server.
  • S304d The IP telephone terminal B sends an ACK acknowledgement message to the IP-PBX gateway, and starts the DSP module to mix the sent first media stream and the received second media stream.
  • the IP-PBX gateway sends the first Re-Invite message to the IP telephone terminal A and the IP telephone terminal B, respectively.
  • the current call is temporarily held by the first Re-Invite message, and the recording prompt is played: "Your call will be recorded.".
  • the IP-PBX gateway sends a second Re-Invite message to the IP telephone terminal A and the IP telephone terminal B, respectively, and the currently held call is restored to the call state by using the second Re-Invite message. .
  • the IP telephone terminal B sends the third media stream generated after the mixing to the recording server corresponding to the media address and the port number.
  • the recording server generates and stores the recording file and the recording data according to the received third media stream.
  • the present invention can transmit the first RTP media stream to the peer-to-peer device by using the DSP module for mixing on the first IP telephone terminal (by the first IP telephone terminal The media stream generated by the microphone) and the second RTP media from the peer device The stream (the media stream arriving at the speaker of the first IP telephone terminal) and the mixing are performed to generate a third media stream.
  • the IP telephone terminal sends the third media stream after the mixing to the recording server, and the recording server completes the recording.
  • the prior art mixing processing required by the DSP module of the IP-PBX gateway is transferred to the DSP module of the IP telephone terminal for processing, and the media resources of the IP-PBX gateway are not required to be occupied, so that the user's point-to-point recording is not affected by the IP-PBX.
  • the limitation of gateway DSP resources enables large-capacity users to perform point-to-point recording at the same time.
  • the recording call signaling of the embodiment of the present invention is controlled by the IP-PBX gateway, and the IP-PBX gateway notifies the IP terminal of the media address and port number negotiated with the recording system, and plays the message to the terminal of the call before recording. The call will be recorded." The tone of the call avoids the problem of the user's call being suspected of being intercepted.
  • the disclosed system, apparatus, and method may be implemented in other manners.
  • the device implementations described above are merely illustrative.
  • the division of the modules or units is only a logical function division.
  • there may be another division manner for example, multiple units or components may be used. Combinations can be integrated into another system, or some features can be ignored or not executed.
  • the mutual coupling or direct coupling or communication connection shown or discussed may be an indirect coupling or communication connection through some interface, device or unit, and may be in an electrical, mechanical or other form.
  • the units described as separate components may or may not be physically separated, and the components displayed as units may or may not be physical units, that is, may be located in one place, or may be distributed to multiple network units. Some or all of the units may be selected according to actual needs to achieve the purpose of the solution of the present embodiment.
  • each functional unit in each embodiment of the present application may be integrated into one processing unit, or each unit may exist physically separately, or two or more units may be integrated into one unit.
  • the above integrated unit can be implemented in the form of hardware or in the form of a software functional unit.
  • the integrated unit if implemented in the form of a software functional unit and sold or used as a standalone product, may be stored in a computer readable storage medium.
  • the technical solution of the present application in essence or the contribution to the prior art, or all or part of the technical solution may be embodied in the form of a software product stored in a storage medium.
  • a computer device which may be a personal computer, server, or network device, etc.
  • the foregoing storage medium includes: a U disk, a mobile hard disk, a read-only memory (ROM), a random access memory (RAM), a magnetic disk, or an optical disk, and the like. .

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Abstract

本发明公开了一种通话录音方法，包括：IP电话终端接收本端用户的语音信号，并将所述语音信号转换为第一媒体流；IP电话终端接收对端通话设备发送的第二媒体流；IP电话终端的数字信号处理器DSP模块将所述第一媒体流和第二媒体流收进行混音处理，得到第三媒体流；IP电话终端将所述第三媒体流发送给录音服务器进行录音。本发明的实施例可以减少混音处理对IP-PBX网关的DSP媒体资源的占用，满足大容量用户同时进行的点对点录音需求。

Description

一种通话录音的方法、装置及系统 技术领域
本发明实施例涉及通信业务领域，具体涉及一种通话录音的方法、装置及系统。
背景技术
随着IP(Internet Protocol)网络承载的语音以及其他增值业务语音业务应用(Voice over IP，简称VoIP)的发展，IP通话录音业务也成为VoIP语音业务中的一项重要业务，在金融行业、保险行业、政府机构等多领域被广泛应用，其中比较常用的是点对点录音业务，即用户A与用户B通话时的点对点录音。
如图1所示，现有技术一的用户B是具有录音权限的用户，当用户A与用户B在通话过程中，用户B可以根据需要手工进行启动录音操作(如果是自动录音，则无需用户手工启动录音操作)，进行点对点录音，具体步骤如下：
步骤①，用户A呼叫用户B；
步骤②，用户B应答；
步骤③，用户A和用户B通话；
步骤④，用户B手工启动录音操作(如果是自动录音，则无需用户B手工启动录音操作)；
步骤⑤，基于IP网络的语音交换机(IP-PBX，IP Private Branch eXchange)收到启动录音操作消息后，创建三方会场，把录音服务器(Recording Server)作为第三方与会者(第一方为用户A，第二方为用户B)拉入到三方会场中；
步骤⑥，IP-PBX网关把会场混音的媒体流发送给录音服务器；
步骤⑦，录音服务器生成并存储录音文件和录音数据。
该方案是由IP-PBX系统中的IP-PBX网关通过三方会场(会场混音)的方式把录音服务器作为第三方与会者拉入到会场中，这样录音服务器收到的媒体流就是IP-PBX网关会场混音后的媒体流，最后由录音服务器生成并存储录音文件和录音数据。
发明人发现，现有技术一的方案存在如下缺陷：(1)由IP-PBX网关会 场混音实现点对点录音时，需要占用IP-PBX网关的三路数字信号处理器(Digital Signal Processor，简称DSP)的媒体资源，而IP-PBX网关的DSP媒体资源是有限的，因此很难满足大容量用户点对点录音的需求。例如IP-PBX网关的总用户数为10000个，而DSP媒体资源为960路，那么能够同时进行的点对点录音的业务数量仅能达到960/3＝320个，因此无法满足大容量用户同时进行的点对点录音需求。
如图2所示，现有技术二的用户B是具有录音权限的用户，当用户A与用户B在通话过程中，用户B可以按需手工进行启动录音操作(如果是自动录音，则无需用户手工启动录音操作)，进行点对点录音，具体过程如下：
步骤①，用户A呼叫用户B。
步骤②，用户B应答。
步骤③，在自动录音模式下，录音进程被自动激活(如果是手工录音，则由用户B的IP话机B向IP-PBX网关发送启动录音消息)。
步骤④，IP-PBX网关呼叫被录音的IP话机B的BIB(built-in bridge)模块，以便记录用户产生的RTP媒体流，BIB模块将会自动应答。
步骤⑤，IP-PBX网关向录音服务器发送IP话机B录音开始的消息(通过SIP协议)，同时被录音的话机B开始复制自身的RTP媒体流2并把它传送给录音服务器。
步骤⑥，IP-PBX网关向录音服务器发送用户话机A的录音开始消息(通过SIP协议)，同时被录音的话机B开始复制话机A产生的RTP媒体流1并把它传送给录音服务器。
步骤⑦，录音服务器生成并存储录音文件和录音数据。
现有技术二的方案，在通话过程中IP话机B把自身产生的RTP媒体流2复制一份后直接发送给录音服务器，同时把接收到的对端IP电话终端A的RTP媒体流1也复制一份后直接发送给录音服务器，由录音服务器生成并存储两份录音文件和录音数据。
发明人发现，现有技术二的方案存在如下缺陷：(1)被录音的IP话机B需要复制两份媒体流给录音服务器，所以在录音服务器侧会产生两份录音文件，录音服务器需要进一步将两份录音文件合并为一份，导致录音服务器的处理业务量增加。(2)录音的IP话机B中需要额外设置BIB模块，导致IP话机B的复杂度和成本上升。
发明内容
本发明实施例的目的是提供一种能够减少对IP-PBX网关的DSP媒体资源的占用、满足大容量用户的点对点录音需求的通话录音方法、装置及系统。
第一方面，提供一种通话录音方法，包括以下步骤：
IP电话终端接收本端用户的语音信号，并将所述语音信号转换为第一媒体流；
IP电话终端接收对端通话设备发送的第二媒体流；
IP电话终端的数字信号处理器DSP模块将所述第一媒体流和第二媒体流收进行混音处理，得到第三媒体流；
IP电话终端将所述第三媒体流发送给录音服务器进行录音。
结合第一方面，在第一方面的第一种可能的实现方式中，在所述IP电话终端接收对端通话设备发送的第二媒体流的步骤之后，还包括：
IP电话终端启动录音操作，以便获取到录音服务器的媒体地址和端口号。
结合第一方面或第一方面的第一种可能的实现方式，在第一方面的第二种可能的实现方式中，在所述IP电话终端将所述第三媒体流发送给录音服务器的步骤之前，还包括：
IP电话终端接收IP-PBX网发送的第一Re-Invite消息，通过该第一Re-Invite消息把当前通话暂时保持，同时播放语音提示音，提示用户通话将被录音；
IP电话终端接收IP-PBX网发送的第二Re-Invite消息，通过该第二Re-Invite消息把当前被保持的通话恢复为通话状态。
结合第一方面的第一种可能，在第一方面的第三种可能的实现方式中，所述IP电话终端启动录音操作，以便获取到录音服务器的媒体地址和端口号的步骤具体包括：
IP电话终端向IP-PBX网关发送Invite消息，所述Invite消息中携带有录音业务的操作码和录音业务标识；
IP-PBX网关向录音系统通过Invite消息发起呼叫；
IP-PBX网关获取到录音服务器的媒体地址和端口号；
IP-PBX网关向录音系统发送启动录音的消息；
IP电话终端接收IP-PBX网关发送的消息，所述消息中携带录音服务器的媒体地址和端口号。
第二方面，提供一种用于通话录音的IP电话终端，包括：
扬声器，用于把收到的语音信号播放给用户；
麦克风，用于把本端用户的语音转换为电信号，并发送给数字信号处理器DSP模块；
数字信号处理器DSP模块，用于对所述电信号进行处理后转换为第一体媒体流信号，还用于将对端通话设备发送的第二媒体流信号进行处理，转换为所述扬声器可以播放的语音信号；
其特征在于，所述DSP模块具有混音单元，用于把所述第一媒体流和所述第二媒体流进行混音处理，混音后生成第三媒体流，并将该第三媒体流发送到录音服务器。
第三方面，提供一种通话录音系统，其特征在于，包括：
IP电话终端，进一步包括：扬声器，用于把收到语音信号播放给用户；麦克风，用于把本端用户的语音转换为电信号，并发送给数字信号处理器DSP模块；数字信号处理器DSP模块，用于对所述电信号进行处理后转换为第一体媒体流信号，还用于将对端通话设备发送的第二媒体流信号进行处理，转换为所述扬声器可以播放的语音信号；以及，混音单元，用于把所述第一媒体流和所述第二媒体流进行混音处理，混音后生成第三媒体流，并将该第三媒体流发送到录音服务器；
对端通话设备，用以与所述IP电话终端建立通话，接收所述IP电话终端发送第一体媒体流信号，并向所述IP电话终端发送第二体媒体流信号；
IP-PBX网关，用于指令录音服务器启动录音和停止录音，以及获取录音服务器的媒体地址和端口号，并发送给所述IP电话终端；
录音服务器，用于根据所述IP-PBX网关的指令，将接收的所述第三媒体流进行录音。
结合第三方面，在第三方面的第一种可能的实现方式中，在所述IP电话终端将所述第三媒体流发送到录音服务器的步骤之前，还包括：
IP电话终端接收IP-PBX网发送的第一Re-Invite消息，通过该第一Re-Invite消息把当前通话暂时保持，同时播放语音提示音，提示用户通话将被录音；
IP电话终端接收IP-PBX网发送的第二Re-Invite消息，通过该第二Re-Invite消息把当前被保持的通话恢复为通话状态。
结合第三方面，在第三方面的第二种可能的实现方式中，所述IP电话终端还用于向IP-PBX网关发送Invite消息，所述Invite消息中携带有录音业务的操作码和录音业务标识；
所述IP-PBX网关还用于向所述录音服务器发起呼叫，获取录音服务器的媒体地址和端口号；
所述IP电话终端接收所述IP-PBX网关发送的消息，所述消息中携带录音服务器的媒体地址和端口号。
本发明的有益效果是：IP电话终端的混音处理单元将本端用户的第一媒体流和对端通话设备发送的第二媒体流收进行混音处理，得到混音后的第三媒体流，然后IP电话终端将该第三媒体流发送给录音服务器进行录音。从而可以减少混音处理对IP-PBX网关的DSP媒体资源的占用，满足大容量用户同时进行的点对点录音需求。
附图说明
为了更清楚地说明本发明实施例的技术方案，下面将对本发明实施例中所需要使用的附图作简单地介绍，显而易见地，下面所描述的附图仅仅是本发明的一些实施例，对于本领域普通技术人员来讲，在不付出创造性劳动的前提下，还可以根据这些附图获得其它的附图。
图1是现有技术一的通话录音系统的示意图；
图2是现有技术二的通话录音系统的示意图；
图3是本发明实施例的通话录音系统的示意图；
图4是本发明实施例的IP电话终端的示意图；
图5是本发明第一实施例的方法流程图；
图6是本发明第二实施例的方法流程图；
图7是本发明第三实施例的方法流程图。
具体实施方式
参阅图3，本发明实施例的一种通话录音系统由两个子系统组成：IP-PBX子系统和录音子系统。
其中，IP-PBX子系统进一步包括IP-PBX网关、用户A的IP电话终端A以及用户B的IP电话终端B。IP-PBX网关用于提供号码注册和呼叫接续，同时负责与会话初始协议服务器(Session Initiation Protocol Server，简称SIP Server)进行SIP信令交互，向SIP Server发送启动录音和停止录音请求、通话消息等，并向录音服务器发送实时传输协议(Real-time Transport Protocol，简称RTP)媒体流或安全实时传输协议(Secure Real-time Transport Protocol，简称SRTP)媒体流。用户A和用户B可以分别通过IP电话终端A和IP电话终端B发起呼叫，并且可以在呼叫过程中启动录音/停止录音。
录音子系统进一步包括：录音服务器(Recording Server)、SIP Server、数据库(Database)以及集成管理平台(Integrated Management Platform, 简称IMP)。其中，录音服务器用于接收IP-PBX子系统发送的RTP/SRTP媒体流，根据指令启动录音和停止录音，生成并存储录音文件和录音数据。SIP Server用以与IP-PBX网关进行SIP消息交互，并向录音服务器发送启动和停止录音的消息。数据库用于保存录音服务器的配置参数和录音记录数据。集成管理平台用于对录音服务器进行集中配置维护的管理平台，对录音服务器各参数进行管理配置。
如图4所示，本发明的IP电话终端包括：DSP模块、扬声器、麦克风以及可视化界面(Graphical User Interface，简称GUI)模块。其中，扬声器用于把收到的语音信号播放给用户，麦克风用于把本端用户的语音转换为电信号，并发送给数字信号处理器DSP模块，GUI模块提供给用户可视化的界面，如呼叫信息、操作按钮等通过界面呈现，DSP模块用于对所述电信号进行处理后转换为第一体媒体流信号，还用于将对端通话设备发送的第二媒体流信号进行处理，转换为扬声器可以播放的语音信号；该DSP模块还具有混音单元(DSP mixer)，用于把第一媒体流和所述第二媒体流进行混音处理，混音后生成第三媒体流，并将该第三媒体流发送到录音服务器。可选的，该混音单元也可以独立于DSP模块，专门进行混音处理。
DSP模块的混音单元的处理流程如下：
(1)扬声器、麦克风已准备就绪，DSP模块的通道打开。
(2)DSP模块接收本端话机的麦克风发送的语音信号，将该语音信号转换为第一RTP媒体流发送到对端通话设备。
(3)DSP模块接收对端通话设备发送的第二RTP媒体流后转换为语音 信号，并把语音信号发送给扬声器进行播放。
(4)DSP模块的混音单元根据需要把发送出去的第一RTP媒体流和收到的第二RTP媒体流进行混音处理，混音后生成第三RTP媒体流，并将该第三RTP媒体流发送到网络侧的特定设备(如录音服务器)。
如图5所示，本发明第一实施例的IP电话终端在通话后手工录音的方法，包括以下步骤：
S100，用户A的IP电话终端A和用户B的IP电话终端B建立通话。一方面，IP电话终端B接收来自IP电话终端A的媒体流，另一方面，IP电话终端B向IP电话终端A发送媒体流。
S102，具有手工录音业务权限的用户B(IP电话终端B)启动录音操作，以便IP-PBX网关获取到录音服务器的媒体地址和端口号。手工录音业务权限可通过业务同步的方式由IP-PBX网关通知IP电话终端B。
该步骤S102具体包括以下步骤：
S102a，IP电话终端B向IP-PBX网关发送Invite消息。其中，Invite消息中携带有录音业务的操作码(Access code)和录音业务标识(recording:on)。
S102b，IP-PBX网关向录音系统通过Invite消息发起呼叫，其中Invite消息的SDP(Session Description Protocol，会话描述协议)消息体中的连接地址为IP电话终端B的IP地址和端口号。
S102c，录音系统向IP-PBX网关回复180振铃(ringing)响应码、200OK响应码。其中，200 OK响应码中携带的SDP消息体包含有录音服务器的媒体地址和端口号。
S102d，IP-PBX网关收到200 OK响应码后，向录音系统发送ACK确认消息，IP-PBX网关与录音系统完成信令协商，获取到录音服务器的媒体地址和端口号。
S104，IP-PBX网关启动录音，该步骤S104具体包括以下步骤：
S104a，IP-PBX网关向录音系统发送INFO消息(包含开始录音的标识recording:start)，通知录音服务器开始录音。
S104b，录音系统反馈200 OK响应码给IP-PBX网关。
S104c，IP-PBX网关在收到录音系统的200 OK响应码后，向IP电话终端B回复200 OK响应，其中200 OK响应码中携带的SDP消息体包含有录音服务器的媒体地址和端口号。
S104d，IP电话终端B向IP-PBX网关发送ACK确认消息，同时启动DSP模块对发送的第一媒体流和收到的第二媒体流进行混音。
S106，IP-PBX网关分别向IP电话终端A和IP电话终端B发送第一Re-Invite消息。通过该第一Re-Invite消息把当前通话暂时保持，同时播放录音提示音：“您的通话将被录音……”。
S108，录音提示音播放完毕后，IP-PBX网关分别向IP电话终端A和IP电话终端B发送第二Re-Invite消息，通过该第二Re-Invite消息把当前被保持的通话恢复为通话状态。
S110，IP电话终端B把混音后生成的第三媒体流发送给媒体地址和端口号对应的录音服务器。
S112，录音服务器根据接收到的第三媒体流生成并存储录音文件和录音数据。
如图6所示，本发明第二实施例的IP电话终端B作为被叫在通话时自动录音的方法，包括以下步骤：
S200，用户A的IP电话终端A呼叫用户B的IP电话终端B，建立通话。一方面，IP电话终端B接收来自IP电话终端A的媒体流，另一方面，IP电话终端B向IP电话终端A发送媒体流。
S202，具有自动录音业务权限的IP电话终端B判断需要自动录音，启动录音操作，以便IP-PBX网关获取到录音服务器的媒体地址和端口号。自动录音业务权限可通过业务同步的方式由IP-PBX网关通知IP电话终端B。该步骤S202具体包括以下步骤：
S202a，IP电话终端B判断需要自动录音后，向IP-PBX网关发送Invite消息。其中，Invite消息中携带有录音业务的操作码(Access code)和录音业务标识(recording:on)。
S202b，IP-PBX网关向录音系统通过Invite消息发起呼叫，其中Invite消息的SDP消息体中的连接地址为IP Phone B的IP地址和端口号。
S202c，录音系统向IP-PBX网关回复180振铃(ringing)响应码、200OK响应码。其中，200 OK响应码中携带的SDP消息体包含有录音服务器的媒体地址和端口号。
S202d，IP-PBX网关收到200 OK响应码后，向录音系统发送ACK确认消息，IP-PBX网关与录音系统完成信令协商，获取到录音服务器的媒体地 址和端口号。
S204，IP-PBX网关启动录音，该步骤S204具体包括以下步骤：
S204a，IP-PBX网关向录音系统发送INFO消息(包含开始录音的标识recording:start)，通知录音服务器开始录音。
S204b，录音系统反馈200 OK响应码给IP-PBX网关。
S204c，IP-PBX网关在收到录音系统的200 OK响应码后，向IP电话终端B回复200 OK响应，其中200 OK响应码中携带的SDP消息体包含有录音服务器的媒体地址和端口号。
S204d，IP电话终端B向IP-PBX网关发送ACK确认消息，同时启动DSP模块对发送的第一媒体流和收到的第二媒体流进行混音。
S206，IP-PBX网关分别向IP电话终端A和IP电话终端B发送第一Re-Invite消息。通过该第一Re-Invite消息把当前通话暂时保持，同时播放录音提示音：“您的通话将被录音……”。
S208，录音提示音播放完毕后，IP-PBX网关分别向IP电话终端A和IP电话终端B发送第二Re-Invite消息，通过该第二Re-Invite消息把当前被保持的通话恢复为通话状态。
S210，IP电话终端B把混音后生成的第三媒体流发送给媒体地址和端口号对应的录音服务器。
S212，录音服务器根据接收到的第三媒体流生成并存储录音文件和录音数据。
如图7所示，本发明第三实施例的IP电话终端B作为主叫在通话时自动录音的方法，包括以下步骤：
S300，用户B的IP电话终端B呼叫用户A的IP电话终端A，建立通话。一方面，IP电话终端B接收来自IP电话终端A的媒体流，另一方面，IP电话终端B向IP电话终端A发送媒体流。
S302，具有自动录音业务权限的IP电话终端B判断需要自动录音，启动录音操作，以便IP-PBX网关获取到录音服务器的媒体地址和端口号。自动录音业务权限可通过业务同步的方式由IP-PBX网关通知IP电话终端B。该步骤S302具体包括以下步骤：
S302a，IP电话终端B判断需要自动录音后，向IP-PBX网关发送Invite消息。其中，Invite消息中携带有录音业务的操作码(Access code)和录音业务 标识(recording:on)。
S302b，IP-PBX网关向录音系统通过Invite消息发起呼叫，其中Invite消息的SDP消息体中的连接地址为IP Phone B的IP地址和端口号。
S302c，录音系统向IP-PBX网关回复180振铃(ringing)响应码、200OK响应码。其中，200 OK响应码中携带的SDP消息体包含有录音服务器的媒体地址和端口号。
S302d，IP-PBX网关收到200 OK响应码后，向录音系统发送ACK确认消息，IP-PBX网关与录音系统完成信令协商，获取到录音服务器的媒体地址和端口号。
S304，IP-PBX网关启动录音，该步骤S304具体包括以下步骤：
S304a，IP-PBX网关向录音系统发送INFO消息(包含开始录音的标识recording:start)，通知录音服务器开始录音。
S304b，录音系统反馈200 OK响应码给IP-PBX网关。
S304c，IP-PBX网关在收到录音系统的200 OK响应码后，向IP电话终端B回复200 OK响应，其中200 OK响应码中携带的SDP消息体包含有录音服务器的媒体地址和端口号。
S304d，IP电话终端B向IP-PBX网关发送ACK确认消息，同时启动DSP模块对发送的第一媒体流和收到的第二媒体流进行混音。
S306，IP-PBX网关分别向IP电话终端A和IP电话终端B发送第一Re-Invite消息。通过该第一Re-Invite消息把当前通话暂时保持，同时播放录音提示音：“您的通话将被录音……”。
S308，录音提示音播放完毕后，IP-PBX网关分别向IP电话终端A和IP电话终端B发送第二Re-Invite消息，通过该第二Re-Invite消息把当前被保持的通话恢复为通话状态。
S310，IP电话终端B把混音后生成的第三媒体流发送给媒体地址和端口号对应的录音服务器。
S312，录音服务器根据接收到的第三媒体流生成并存储录音文件和录音数据。
从本发明的以上实施例可以看出，本发明通过在第一IP电话终端上利用DSP模块进行混音，能够把向对端通话设备发送的第一RTP媒体流(由第一IP电话终端的麦克风产生的媒体流)和来自对端通话设备的第二RTP媒体 流(到达第一IP电话终端的扬声器的媒体流)和进行混音，产生第三媒体流。IP电话终端把混音后的第三媒体流发送给录音服务器，由录音服务器完成录音。将现有技术的需要由IP-PBX网关的DSP模块来完成的混音处理转移到IP电话终端的DSP模块处理，无需占用IP-PBX网关的媒体资源，从而使得用户点对点录音不受IP-PBX网关DSP资源的限制，实现大容量用户同时进行点对点录音功能。
另外，本发明实施例的录音呼叫信令由IP-PBX网关进行控制，IP-PBX网关把与录音系统协商得到的媒体地址和端口号告知IP终端，并且在录音前向通话的终端播放“您的通话将被录音……”的提示音，避免涉及到用户的通话疑似被监听的问题。
在本申请所提供的几个实施方式中，应该理解到，所揭露的系统，装置和方法，可以通过其它的方式实现。例如，以上所描述的装置实施方式仅仅是示意性的，例如，所述模块或单元的划分，仅仅为一种逻辑功能划分，实际实现时可以有另外的划分方式，例如多个单元或组件可以结合或者可以集成到另一个系统，或一些特征可以忽略，或不执行。另一点，所显示或讨论的相互之间的耦合或直接耦合或通信连接可以是通过一些接口，装置或单元的间接耦合或通信连接，可以是电性，机械或其它的形式。
所述作为分离部件说明的单元可以是或者也可以不是物理上分开的，作为单元显示的部件可以是或者也可以不是物理单元，即可以位于一个地方，或者也可以分布到多个网络单元上。可以根据实际的需要选择其中的部分或者全部单元来实现本实施方式方案的目的。
另外，在本申请各个实施方式中的各功能单元可以集成在一个处理单元中，也可以是各个单元单独物理存在，也可以两个或两个以上单元集成在一个单元中。上述集成的单元既可以采用硬件的形式实现，也可以采用软件功能单元的形式实现。
所述集成的单元如果以软件功能单元的形式实现并作为独立的产品销售或使用时，可以存储在一个计算机可读取存储介质中。基于这样的理解，本申请的技术方案本质上或者说对现有技术做出贡献的部分或者该技术方案的全部或部分可以以软件产品的形式体现出来，该计算机软件产品存储在一个存储介质中，包括若干指令用以使得一台计算机设备(可以是个人计算机，服务器，或者网络设备等)或处理器(processor)执行本申请各个实施 方式所述方法的全部或部分步骤。而前述的存储介质包括：U盘、移动硬盘、只读存储器(ROM，Read-Only Memory)、随机存取存储器(RAM，Random Access Memory)、磁碟或者光盘等各种可以存储程序代码的介质。
以上所述仅为本申请的实施方式，并非因此限制本申请的专利范围，凡是利用本申请说明书及附图内容所作的等效结构或等效流程变换，或直接或间接运用在其他相关的技术领域，均同理包括在本申请的专利保护范围内。

Claims (8)

  1. 一种通话录音的方法，包括以下步骤：
    IP电话终端接收本端用户的语音信号，并将所述语音信号转换为第一媒体流；
    IP电话终端接收对端通话设备发送的第二媒体流；
    IP电话终端的数字信号处理器DSP模块将所述第一媒体流和第二媒体流收进行混音处理，得到第三媒体流；
    IP电话终端将所述第三媒体流发送给录音服务器进行录音。
  2. 根据权利要求1所述的方法，其特征在于，在所述IP电话终端接收对端通话设备发送的第二媒体流的步骤之后，还包括：
    IP电话终端启动录音操作，以便获取到录音服务器的媒体地址和端口号。
  3. 根据权利要求1或2所述的方法，其特征在于，在所述IP电话终端将所述第三媒体流发送给录音服务器的步骤之前，还包括：
    IP电话终端接收IP-PBX网发送的第一Re-Invite消息，通过该第一Re-Invite消息把当前通话暂时保持，同时播放语音提示音，提示用户通话将被录音；
    IP电话终端接收IP-PBX网发送的第二Re-Invite消息，通过该第二Re-Invite消息把当前被保持的通话恢复为通话状态。
  4. 根据权利要求2所述的方法，其特征在于，所述IP电话终端启动录音操作，以便获取到录音服务器的媒体地址和端口号的步骤具体包括：
    IP电话终端向IP-PBX网关发送Invite消息，所述Invite消息中携带有录音业务的操作码和录音业务标识；
    IP-PBX网关向录音系统通过Invite消息发起呼叫；
    IP-PBX网关获取到录音服务器的媒体地址和端口号；
    IP-PBX网关向录音系统发送启动录音的消息；
    IP电话终端接收IP-PBX网关发送的消息，所述消息中携带录音服务器的媒体地址和端口号。
  5. 一种用于通话录音的IP电话终端，包括：
    扬声器，用于把收到的语音信号播放给用户；
    麦克风，用于把本端用户的语音转换为电信号，并发送给数字信号处理器DSP模块；
    数字信号处理器DSP模块，用于对所述电信号进行处理后转换为第一体媒体流信号，还用于将对端通话设备发送的第二媒体流信号进行处理，转换为所述扬声器可以播放的语音信号；
    其特征在于，所述DSP模块具有混音单元，用于把所述第一媒体流和所述第二媒体流进行混音处理，混音后生成第三媒体流，并将该第三媒体流发送到录音服务器。
  6. 一种通话录音系统，其特征在于，包括：
    IP电话终端，进一步包括：扬声器，用于把收到语音信号播放给用户；麦克风，用于把本端用户的语音转换为电信号，并发送给数字信号处理器DSP模块；数字信号处理器DSP模块，用于对所述电信号进行处理后转换为第一体媒体流信号，还用于将对端通话设备发送的第二媒体流信号进行处理，转换为所述扬声器可以播放的语音信号；以及，混音单元，用于把所述第一媒体流和所述第二媒体流进行混音处理，混音后生成第三媒体流，并将该第三媒体流发送到录音服务器；
    对端通话设备，用以与所述IP电话终端建立通话，接收所述IP电话终端发送第一体媒体流信号，并向所述IP电话终端发送第二体媒体流信号；
    IP-PBX网关，用于指令录音服务器启动录音和停止录音，以及获取录音服务器的媒体地址和端口号，并发送给所述IP电话终端；
    录音服务器，用于根据所述IP-PBX网关的指令，将接收的所述第三媒体流进行录音。
  7. 根据权利要求6所述的系统，其特征在于，在所述IP电话终端将所述第三媒体流发送到录音服务器的步骤之前，还包括：
    IP电话终端接收IP-PBX网发送的第一Re-Invite消息，通过该第一Re-Invite消息把当前通话暂时保持，同时播放语音提示音，提示用户通话将被录音；
    IP电话终端接收IP-PBX网发送的第二Re-Invite消息，通过该第二Re-Invite消息把当前被保持的通话恢复为通话状态。
  8. 根据权利要求6所述的系统，其特征在于，所述IP电话终端还用于向IP-PBX网关发送Invite消息，所述Invite消息中携带有录音业务的操作 码和录音业务标识；
    所述IP-PBX网关还用于向所述录音服务器发起呼叫，获取录音服务器的媒体地址和端口号；
    所述IP电话终端接收所述IP-PBX网关发送的消息，所述消息中携带录音服务器的媒体地址和端口号。
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