CN100452784C - Phonetical gate communication device and method - Google Patents

Phonetical gate communication device and method Download PDF

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CN100452784C
CN100452784C CN 03123797 CN03123797A CN100452784C CN 100452784 C CN100452784 C CN 100452784C CN 03123797 CN03123797 CN 03123797 CN 03123797 A CN03123797 A CN 03123797A CN 100452784 C CN100452784 C CN 100452784C
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voice
digital
interface
gateway
division
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CN1553666A (en )
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孙建明
孙文举
张自习
徐永德
滕学金
董有荣
赵培吉
赵海防
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北京共业科技有限公司
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Abstract

本发明公开了一种语音网关通讯设备,该设备不仅包括一般语音网关通讯设备包括的部件,还包括一个时分数字交换网络芯片,所有的电话接口、中继接口及DSP都通过编解码模块与时分数字交换网络芯片相连接。 The present invention discloses a voice gateway communications equipment, the apparatus comprising means generally comprises a voice gateway not only communication equipment, further comprising a digital time-division switching network chip, all the telephone interface, trunk interface and codec DSP through a time division module digital switching network chip is connected. 本发明还公开了一种语音网关通讯方法,其特征是,该方法既包括通过IP(互联网协议)网络传送语音信号的步骤,同时还包括通过程控交换方式传送语音信号的步骤。 The present invention also discloses a method of voice communications gateway, wherein the method comprises the step of both IP (Internet Protocol) network to transmit voice signals, and further comprising the step of transmitting a speech signal through the remote exchange. 本发明方法和设备结合了传统程控交换设备及方法和语音网关设备及方法的优点,不仅实现了通讯设备功能的完善,有效地降低成本,并且明显提高了内部通话的语音质量。 The method and apparatus of the present invention combines a conventional telephone switching equipment and methods and advantages of the voice gateway apparatus and methods, not only the improvement of telecommunications equipment, reduces costs, and significantly improve the quality of voice intercom.

Description

一种语音网关通讯设备和方法 A voice gateway communications equipment and methods

技术领域 FIELD

本发明涉及一种语音网关通讯设备和语音网关通讯方法,尤其涉及一种结合传统的电话交换机技术与基于互联网协议的语音传输技术(VOIP技术)的语音 The present invention relates to a communication device and a voice gateway voice gateway communication method, and particularly to a combination of traditional voice telephone exchange technology and voice transmission technology based on Internet Protocol (VOIP technology)

网关通讯i殳备和语音网关通讯方法。 Communication Gateway and Voice Gateway i Shu apparatus communication method.

背景技术 Background technique

传统上,为了通讯的目的,几乎每个企业用户或集团用户一般使用专用程控 Traditionally, for the purposes of communications, almost every business user or group of users generally use a dedicated programmable

交换机(PBX)来实现固定办公场所内部的互相通话,并通过接入外部公网的方式实现与外部用户的通话。 Exchange (PBX) call each other to achieve a fixed inner office and a conversation with an external user via an external public network access. 传统的数字程控交换机的工作原理是,首先将呼叫端电话的模拟信号按照统一的标准,在编解码模块内将其转换为数字信号;再进行编码,形成每秒64K位的数据流;该数据流进入到时分数字交换网络芯片,时分数字交换网络可以将此数据流交换到被呼叫端电话的接口上;然后在编解码模块内将此数字信号进行解码;再转换成模拟信号;最后传输到受话端电话机上,这样可以听到另一方的声音。 The traditional digital PBX operating principle is, first end of the telephone call to the analog signal in accordance with a uniform standard, in the encoding and decoding module converts it into a digital signal; re-encoded to form a 64K bit per second data stream; data flow proceeds to a time division digital switching network chip, time division digital switching network, this data stream to the exchange on the called telephone terminal interface; then this digital signal is decoded in the codec module; converted to an analog signal; finally transmitted to then by the end of the phone, so you can hear the other voice. 其中,时分数字交换网络芯片是程控交换机的核心, 它受中央处理器CPU的控制。 Wherein the time-division digital switching network, the core chip program-controlled switches, which controlled by a central processing unit CPU. 中央处理器CPU中包括了一套进行程控交换控制的较复杂的专用软件系统。 A central processing unit CPU includes a more complex set of dedicated software program-controlled switching system control. 使用程控交换机的好处主要是,对于企业内部用户或集团内部用户与外部用户的通话而言,它可以使较多的用户充分共享较少的中继线资源来实现同时通话;对于企业内部用户或集团内部用户之间的通话而言,它可以使这些内部用户之间的通话不受中继线资源的限制,而可以自由地免费通话, 从而显著节省费用;此外,还可以拓展出许多补充业务功能。 The benefits of using PBX is mainly for internal users and external users within an enterprise user or group call, the more it allows users to fully share fewer resources to achieve the trunk of simultaneous calls; or for internal users within the group For calls between users, which can make internal calls between these unrestricted trunk resources, and can freely free calls to significantly save costs; in addition, can also develop a lot of supplementary business functions.

实现电话通信的另一种设备为利用IP网络来传输语音的设备一一V0IP语音网关设备。 Another device is implemented using an IP telephone communication network to transmit voice device eleven V0IP voice gateway device. 随着计算机的日益普及和IP技术的广泛应用,各行各业的许多大、 中、小企业及许多政府部门都通过租用或自建的方式拥有了自身的IP网,VOIP 语音网关设备也应运而生。 With the widespread use of computers and the growing popularity of IP technology, many businesses large, medium and small businesses and many government departments have their own IP network or leased by way of self-built, VOIP voice gateway equipment also came into being students. 其工作原理是:接收到呼叫语音信号后,利用压缩算法对语音数据进行压缩处理,然后将语音数据按TCP/IP标准进行打包,经过IP网络把数据包送至被呼叫地,再把这些语音数据包串起来,经过解压处理后, Its working principle is: after receiving the call voice signals using a compression algorithm to compress the voice data processing, and then packetizes the voice data according to TCP / IP standard, IP network via the packet is sent to the call, then the voice the packet string together, after the decompression process,

恢复成原来的语音信号,送到被呼叫端,达到由互联网实时传送语音的目的。 Restored to the original speech signal to the called terminal, to achieve real time voice transmission by the Internet. 使用VOIP语音网关系统的好处是,它能直接使用因特网,通用性好,极大地扩大 The advantage of using VOIP voice gateway system is that it can directly use the Internet, common good, which greatly expanded

了传递语音信息的范围;节省拨打长途电话的开支;还能拓展出许多其它功能, The transmission of both voice information; saving on long distance calls expenses; also develop many other functions,

如同时进行图象传输,等等。 The simultaneous image transmission, and the like.

但是,前述两种电话通信设备各自都有不足之处,即: 一方面,传统PBX 的缺点在于,不同厂家提供的PBX产品含有各厂家的专有技术,产品之间兼容性差、扩展性差;而且,传统PBX不能简单地通过IP (互联网协议)网络传输语音,即不具有V01P设备所具有的功能。 However, at each of the two kinds of telephone communication device has shortcomings, namely: on the one hand, traditional PBX disadvantage is that different manufacturers to provide products containing various manufacturers of PBX's proprietary technology, poor compatibility between products, poor scalability; and , not simply by traditional PBX IP (Internet protocol) network to transmit voice, i.e. having no V01P apparatus has the functions. 而V01P技术是实现从传统封闭的电信系统向开放的、以IP为基础的电信平台转变的革命性技术,它开放了以往传统的各个交换控制的平台系统,使得新产品与新应用层出不穷,前景十分广阔。 The V01P technology is realized from the traditional closed system to an open telecommunications and revolutionary technology to IP-based telecommunications platform change, it opened up the platform of traditional system of exchange control, making the new products and new applications emerging, prospects very broad. 传统的PBX如果不与IP技术相结合,其应用将会受到极大的限制。 The traditional PBX if not combined with IP technology, its application will be greatly hampered. 此外,PBX设备的费用较高,使用PBX拨打长途电话费用亦高。 In addition, higher costs of PBX equipment, the cost of long distance calls using a PBX is also high. 另一方面,语音网关设备不具有数字程控交换机的许多功能,尤其是专用电话交换的许多功能,这便导致在许多应用中,语音网关所提供的功能不能满足用户的需求。 On the other hand, voice gateway device does not have many of the features of a digital PBX, especially many of the features of a dedicated telephone exchange, which would lead to many applications, voice gateway functionality provided can not meet the needs of users. 如,在有关电话通讯的国家标准中规定了许多的补充业务,语音网关不能实现所有的补充业务功能,这同样会限制其应用范围。 For example, provisions in national standards relevant telephone communications in a number of supplementary services, voice gateway can not achieve all the supplementary business functions, which will also limit its scope of application. 同时,当语音网关的一个电话用户呼叫语音网关内部的另一个电话用户时,通话质量下降。 Meanwhile, when the interior of the telephone user to another user calls a telephone voice gateway voice gateway, call quality. 因为不论是网关内部还是外部的通话,语音信号均要经过压缩过程后被传输,这样不仅不可避免地要产生失真,导致通话质量下降,并且,语音网关对语音信号的压缩和解压缩过程都要占用系统大量的处理能力,尤其是占用系统的DSP处理能力,这将会导致系统的处理能力降低,或者说,同样多的用户通话,将会占用系统更多的资源,导致对系统的处理能力的要求提高。 Because both internal or external gateway calls, voice signals are to be transmitted after the process is compressed, so that not only inevitably produce distortion, leading to decline in call quality and voice gateway process to be occupied for the compression and decompression of speech signal compression system lot of processing power, especially occupy system of DSP processing power, which will result in reduced processing capacity of the system, or, as many users call, it will take up more system resources, resulting in the processing capability of the system requirements increase. 此外,为了同时使用两者提供的功能,用户不得不同时购置PBX设备和VOIP设备各一套,增加了固定资产成本和维护费用。 In addition, in order to simultaneously provide both functions, the user has to purchase simultaneously VOIP PBX equipment and each set of equipment, increasing the cost of fixed assets and maintenance costs.

发明内容 SUMMARY

本发明的目的是,提供一种将传统的PBX技术与VOIP技术相结合的新型语音网关通讯设备和新型语音网关通讯方法。 Object of the present invention is to provide a novel voice gateway communications equipment and a traditional PBX VOIP technology and novel technology combined voice gateway communication method.

按本发明的一个目的,提供了一种将传统的PBX技术与VOIP技术相结合的新型语音网关通讯设备,包括若干个电话接口及中继接口、数字信号处理器(DSP)、中央处理器(CPU)以及以太网接口;以太网一端接到中央处理器(CPU) 上,另一端接到外部数据网络上;其特征在于,还包括一个时分数字交换网络芯 According to a purpose of the present invention, there is provided a novel apparatus for voice communications gateway traditional PBX technology and VOIP technology combination, comprising a plurality of telephony interface and trunk interface, a digital signal processor (DSP), a central processing unit ( CPU), and an Ethernet interface; end of the Ethernet to the central processor (CPU) on the other end to the external data network; characterized by further comprising a time division digital switching network, the core

片,所有的电话接口和中继接口都通过编解码模块与时分数字交换网络芯片相连接;DSP也与此时分数字交换网络芯片相连接。 Sheet, all the telephone interface and trunk interface are connected through a switching network chip codec module and a digital time-division; the DSP is also connected to the digital switching network chip sub-case. 这样,所有电话接口、中继接口、 DSP接口都能通过时分数字交换网络芯片中的时分数字交换网络进行信号的互相交换。 Thus, all the telephone interface, trunk interface, the interface can be switched network the DSP exchange signals time-division digital switching network via time-division digital chip. 当被呼叫对象位于网关外部时,在中央处理器CPU的控制下,电话语音通过编解码模块转换成数字信号,然后通过时分数字交换网络传输到DSP,由DSP 对数字信号进行压缩处理,形成语音包,之后,中央处理器CPU将此语音包发送到外部数据网络,通过外部数据网络到达被呼叫对象。 When the call destination is located outside the gateway, under the control of the central processing unit CPU, telephone voice codec converted by the module into a digital signal, and then transmitted to the switching network via time-division digital DSP, digital signal processing by the DSP for compression, forms a voice package, then, the CPU transmits this voice packet to an external data network and arriving at the call destination via an external data network. 因此,本发明具有了通过互联网传输语音(VOIP)的功能。 Accordingly, the present invention has the function of Voice over Internet (VOIP) a. 而当被呼叫对象位于网关内部或者是中继接口时,在中央处理器CPU的控制下,将语音信号转换成数字信号,编码后直接通过时分数字交换网络交换到被呼叫端对应的编解码器,解码还原成模拟语音信号, 最终达到被呼叫对象。 And when the call destination is located within the gateway or trunk interface, under the control of the central processing unit CPU, a voice signal into a digital signal, encoded by the corresponding call to the exchange end of codecs directly through a time division digital switching network decoded into an analog voice signal reduction, and ultimately achieve the called object. 因而,本发明具有了数字程控交换机的所有功能。 Accordingly, the present invention has all the features of a digital PBX.

按本发明的另一个目的,提供了一种将传统的FBX技术与VOIP技术相结合的新型语音网关通讯方法,如图5所示,包括以下步骤: According to another object of the present invention, there is provided a novel method for the voice gateway communication technology and conventional FBX VOIP technology combining 5, comprising the steps of:

a. 从电话接口或中继接口端接受呼叫信号,通过编解码模块将模拟语音呼叫模拟信号转换成数字信号并进行编码; . A receiving a call signal from the telephone interface, or trunk interface terminal, via the codec module converts the analog voice call analog signals to digital signals and encoded;

其特征在于,还包括以下步骤: Characterized by further comprising the steps of:

b. 判断被呼叫对象位于网关内部还是外部;如果被呼叫对象位于网关外部, 则执行以下步骤c;如果被呼叫对象位于网关内部或者是中继接口,则执4亍以下步骤d; . B determines the call destination is located inside or outside the gateway; if the call destination is located outside the gateway, the step C is executed; if the call destination is located within the relay gateway, or interface, perform the steps of the right foot 4 D;

c. 通过时分数字交换网络传输到数字信号处理器DSP,由DSP对数字信号进行压缩处理形成语音包,然后由中央处理器CPU将此语音包发送到外部数据网络,通过外部数据网络到达被呼叫对象; c. By time division digital switching network to the digital signal processor DSP, DSP voice packets to form the digital signal processed by compression, the central processor CPU then sends this voice data packet to the external network, the external data is a call through the network to objects;

d. 在中央处理器CPU的控制下,将编码了的数字信号直接通过时分数字交换网络交换到达被呼叫对象对应的编解码模块,还原成模拟语音信号, 最终到达被呼叫对象。 d. Under control of the central processing unit CPU, a digital signal will be encoded directly reach the exchange switching network corresponding to the call destination codec module through a digital time-division, reduction into an analog voice signal, and finally to the called object.

本发明在通常的语音网关中加入了时分数字交换网络,同时通过中央处理器的软件控制处理流程,实现通信功能。 The present invention is generally added in a voice gateway in a time division digital switching network, while controlling the flow of processing by the central processor software, communication functions. 与传统的PBX相比,本发明有着明显的优势,如:将电话网与计算机网络统一成一个整体, 一机两用或多用,除了能为传统的电话用户提供服务外,还能方便地为因特网用户提供服务;同时,除了提供电话程控交换功能外,还能提供语音网关通讯功能;为用户显著节约了设备投资和运行成本;全新的硬件平台和高度集成的系统功能可更加方便和容易地实现增值服务,比如建立呼叫中心、实施V0IP等;将专用的通讯平台搬到大众普遍较 Compared with the conventional the PBX, the present invention has obvious advantages, such as: the telephone network and a computer network into a unified whole, one machine or use, in addition to conventional telephone service for the user, but also can conveniently be internet users with services; at the same time, in addition to providing program-controlled telephone switching function, but also provides a voice gateway communication function; the user significant savings in equipment investment and operating costs; new hardware platforms and highly integrated system functions can be more convenient and easily value-added services, such as call centers, such as the implementation of V0IP; dedicated communications platform to move to a more general public

熟悉的计算机平台上,有开放的标准,互通性、通用性和实用性更强;使用、配置和维护更加简单;功能更加强大且集成度高,单一系统就可以完成使用传统PBX需外配许多设备才能完成的功能,比如自动话务台、语音信箱等;由于采用了计算机平台,使系统的扩容和升级更加简单和节省投资;应用开发方便简单。 On familiar computer platform, open standards, interoperability, versatility and more practical; use, configuration and maintenance easier; more powerful and high integration, a single system can be completed using a traditional PBX to be outside with many equipment to complete functionality, such as automated attendant, voice mail and so on; As a result of computer platforms, expansion and upgrade of the system easier and saving investment; application development easy and convenient. 而与一般的VOIP语音网关产品相比较,本发明能够实现语音交换功能;内部通话时由于免去了对数据的压缩和解压缩处理,因此减少了语音的失真,提高了通话质量,同时降低了DSP通道的占用,提高了DSP系统的处理能力,有效地降低了系统的成本;此外,能够实现一般VOIP产品所不具备的电话增值业务,如呼叫保持、呼叫等待、三方协商等补充业务;在同样用户数的情况下,本发明的性价比高。 Compared with the general VOIP voice gateway products, the present invention enables voice switching function; internal call due eliminates compression and decompression processing of compressed data, thus reducing speech distortion and improve call quality, while reducing the DSP occupied channel, increasing the processing power of the DSP system, effectively reducing the cost of the system; moreover, enables the general VOIP products do not have a phone value-added services, such as call hold, call waiting, three-way consultations and other supplementary services; at the same when the number of users, the present invention is cost-effective. 因此,本发明不仅能分别克服PBX设备和VOIP设备各自的不足之处, 还结合了PBX和V0IP两者的优点,即在实现VOIP功能的同时,又提供了所有的程控交换机的功能,能够满足用户不同场合的各种需求。 Accordingly, the present invention not only to overcome the PBX equipment and VOIP device respectively at their shortcomings, but also combines the advantages of both PBX and V0IP, i.e. at the same time implementing a VOIP functionality, and provides all of the functionality of the PBX to meet user needs of different occasions. 因此本发明不仅实现了通讯设备功能的完善,有效地降低成本,并且明显提高了内部通话的语音质量。 Thus, the present invention not only achieves improved communications equipment functions, effectively reduce costs, and significantly improve the quality of voice calls inside.

附图说明 BRIEF DESCRIPTION

图1为现有技术的专用程控交换机的线路连接方框图。 Figure 1 is a block diagram showing a connection line dedicated PBX prior art. 图2为现有技术的VOIP语音网关设备的线路连接方框图。 Line VOIP voice gateway device FIG 2 is a block diagram showing a prior art connector. 图3为本发明语音网关通讯设备的线路连接方框图。 FIG 3 line voice gateway device connected to the communication block diagram of the present invention.

图4为本发明设备的时分数字交换网络芯片和中央处理器(CPU)的功能模块图示。 FIG 4 division digital switching network chip devices and a central processor (CPU) of the present invention illustrating a functional module.

图5为本发明方法的流程图。 5 is a flowchart of a method of the present invention FIG.

图6为本发明语音网关通讯方法一个方案的流程图。 A flowchart of a method of voice communications gateway embodiment of the present invention of FIG. 具体实施方式 Detailed ways

下面结合附图对本发明设备作进一步的详细说明。 BRIEF apparatus of the present invention will be further described in detail below in conjunction.

图3为本发明设备的线路连接方框图,本发明的语音网关通讯设备包括若干个电话接口及中继接口、数字信号处理器(DSP)、中央处理器(CPU)以及以太网接口;以太网一端接到中央处理器(CPU)上,另一端接到外部数据网络上; 还包括一个时分数字交换网络芯片,所有的电话接口和中继接口都通过编解码模块与时分数字交换网络芯片相连接;DSP也与此时分数字交换网络芯片相连接。 FIG 3 a block diagram of the line connecting device of the present invention, the voice gateway communication apparatus according to the invention comprises a plurality of telephony interface and trunk interface, a digital signal processor (DSP), a central processing unit (CPU), and an Ethernet interface; Ethernet end to the central processing unit (CPU), the other end to an external data network; further comprising a digital time-division switching network chip, all the telephone interface and trunk interface chip are connected by a switching network codec module and a digital time-division; DSP is also connected to the digital switching network chip sub-case. 这样,所有电话接口、中继接口、 DSP接口都能通过时分数字交换网络芯片中的时分数字交换网络进行信号的互相交换。 Thus, all the telephone interface, trunk interface, the interface can be switched network the DSP exchange signals time-division digital switching network via time-division digital chip. 其中,电话接口、中继接口、数字信号处理器(DSP )、以太网接口和编解码模块都是PBX技术与VOIP技术领域中的常用部件,可以市购得到。 Wherein the telephone interface, trunk interface, a digital signal processor (DSP), an Ethernet interface and a codec module are PBX and VOIP technology used in the field of component technology, can be obtained commercially. 时分数字交换网络芯片和中央处理器(CPU)则按本发 Time-division digital switching network and a central processor chip (CPU) press the invention

明设备的功能需要,安装有为此设计制造的逻辑电路和控制软件系统,它们的模 Light - the device needs to install the software and the control logic circuit manufacturing systems designed for this purpose, their mode

块示于图4中。 Block shown in Figure 4. 在了解了本发明的内容的基础上,本领域普通技术人员容易得知各个模块的具体内容,并且知道如何实现它们。 In the understanding of the present invention on the basis of those of ordinary skill in the art readily know the specific contents of each module, and to know how to implement them.

图6为本发明语音网关通讯方法一个方案的流程图,包括步骤为: 第l步:电话摘机,CPU控制时分数字交换网络给电话接口送拨号音; 第2步:用户拨号,CPU控制时分数字交换网络切断拨号音,数据库软件进 Dial-up users, the CPU controls division:; voice gateway 6 is a flowchart of the communication method of one aspect of the present invention, comprising the steps of:: Step 2 Step l phone off hook, the CPU controls the time division digital switching network, to send a dial tone to the telephone interface digital switching network cut off the dial tone, database software into

行号码分析,如果是V0IP呼叫,则执行以下第3步、第4步,否则执行第5步、 Line number analysis, if it is V0IP call, perform the following step 3, step 4, otherwise go to step 5,

第6步; Step 6;

第3步:VOIP任务将呼叫通过VOIP协议发送到网络,到达被叫网关,电话接口通过时分数字交换网络连接到DSP; Step 3: VOIP call transmission task to the network through a VOIP protocol, reaches the called gateways, telephone interface via a digital time division switching network is connected to the DSP;

第4步:被叫应答,主被叫通话,DSP压缩语音,CPU通过网络接口将语音发送到被叫,然后执行以下第7步 Step 4: the called party answers, the main incoming call, the DSP voice compression, voice is sent to the CPU through the network interface called, then perform the following step 7

第5步:如果是内部电话,则被叫振铃;如果是中继,则发送呼叫信息到对方。 Step 5: If the internal telephone, ringing the called; if it is a relay, sending call information to the counterpart. 通过时分数字交换网络将主叫电话连接到回铃音; Connect the phone to the caller ringback tone by time-division digital switching network;

第6步:被叫应答,通过时分数字交换网络连接主叫、被叫,主叫、被叫通话。 Step 6: the called party answers, a network connecting the caller, the called party, calling party, an incoming call by a time-division digital switching.

第7步:主叫挂机,切断交换网络,给被叫发送呼叫释放信号,释放系统资源。 Step 7: the caller hangs up, cut off the switching network, to the called party transmits a call release signal to release the system resources.

为实现以上步骤,本发明方法使用的控制系统采用实时操作系统并以多任务实现,系统中创建的任务模块包括网络任务、接口通信任务、呼叫处理任务、数据库任务、V0IP协议任务、DSP控制通信任务,此外,还可以包括计费任务、维 To achieve the above steps, the method of the present invention, the control system uses real-time operating system, and a multitasking implemented task module created in the system includes a network task, the interface communication task, the call processing tasks, database tasks, V0IP protocol task, the DSP controls the communication task, in addition, may include billing tasks, Victoria

护配置任务等。 Protection configuration tasks. 其中,各任务的具体内容为: Among them, the specific content of each task as follows:

1. 网络任务:负责从网络上接受数据和发送数据,实现网络通信。 1. Network Task: responsible for receiving and sending data from the network, network communication.

2. 接口通信任务:完成接口板和系统软件之间的通信,收集硬件的事件, 并且将系统软件的控制传达到接口板。 2. Interface communication tasks: the communication between the interface board and system software, hardware collection of events, and will be communicated to the control system software of the interface board.

3. 呼叫处理任务:完成电话、中继的控制,实现数字程控交换机的通信协议、补充业务等。 3. The call processing tasks: completion of calls, control relay, digital PBX implementing communication protocols, and other supplementary service. 采用有限消息状态级机制,完成对电话接口、中继接口的控制。 Finite status message mechanism stage, complete control of the telephone interface, a relay interface. 电话接口、中继接口、交换网络、DSP通道在呼叫处理任务的控制下互相传递语音信号流,完成数字程控交换机的功能。 Telephone interface, a relay interfaces, switching network, the DSP voice channel transmission signal streams with each other under the control of call processing task, completion of a digital PBX. 还能与VOIP 协议密切配合,实现V0IP功能。 Also work closely with VOIP protocol to achieve V0IP function.

4. 数据库任务:为呼叫处理任务和VOIP协议任务服务,提供号码分析、路由选择、资源管理、资源分配等功能。 4. Database task: to call VOIP protocol processing tasks and task services, providing number analysis, routing, resource management, resource allocation and other functions.

5. VOIP协议任务实现VOIP的功能,实现与其他语音网关的通信,与呼叫处理任务密切协作,在提供语音通信的同时,还能提供传真业务。 5. VOIP protocol task of implementing a VOIP functionality, implementation, communication in close collaboration with the other voice gateways and call processing tasks, while providing voice communications, but also provides fax services.

6. 计费任务从呼叫处理任务:和VOIP协议任务收集所有的计费原始信息, 并发送到计费服务器,使所有的呼叫能够在计费服务器上进行计费,适合运营商或虚拟运营商的运营要求。 6. billing tasks from call-handling tasks: collecting and VOIP protocol task all the original billing information, and sent to the accounting server, so that all calls can bill on the billing server for virtual operator or operators operational requirements.

7. 维护配置任务:完成系统的数据维护和管理,提供多种维护方式,包括HTTP方式、Telnet方式、SNMP方式等多种維护办法。 7. Maintenance configuration tasks: complete system maintenance and data management, offers a variety of maintenance, including maintenance of a variety of ways through HTTP, Telnet mode, SNMP methods. 维护配置任务要与数据库任务密切协作,共同完成对系统的配置和维护。 Configuration Tasks to maintain close collaboration with the database tasks together to complete the configuration and maintenance of the system.

8. DSP控制通信任务:完成对数字信号处理器(DSP)的控制,并且实现系统软件和DSP的通信,能将DSP输出的经过压缩的语音包交给网络任务, 也能将网络任务收到的语音数据包发送给DSP,由DSP解压后输出到数字交换网络,由数字交换网络输出到各接口(电话接口和中继接口),实现VOIP功能。 8. The DSP controls the communication tasks: complete control of a digital signal processor (DSP), and implement a communication system software and DSP, DSP can output through the compressed voice packets to the network task, the network task can also receive voice data packet to the DSP, to the digital switching network output after decompression of the DSP, the digital switching network output to the respective connector (telephone interface and trunk interface), for implementing a VOIP functionality.

Claims (3)

1. 一种语音网关通讯设备,包括若干个电话接口及中继接口、数字信号处理器、 中央处理器以及以太网"l姿口;以太网一端接到中央处理器上,另一端接到外部数据网络上;其特征在于,还包括一个时分数字交换网络芯片,所有的电话接口和中继接口都通过编解码模块与时分数字交换网络芯片相连接,数字信号处理器也与此时分数字交换网络芯片相连接,在中央处理器的控制下, 当被呼叫对象位于网关外部时,电话语音由编解码模块转换成数字信号后, 通过时分数字交换网络传输到数字信号处理器进行压缩处理,然后由中央处理器通过外部数据网络送达被呼叫对象;当被呼叫对象位于网关内部或者是中继接口时,语音信号转换成数字信号后直接通过时分数字交换网络交换到被呼叫端对应的编解码模块,解码还原后到达被呼叫对象。 A voice gateway communication apparatus comprising a plurality of telephony interface and trunk interface, a digital signal processor, the central processor and the Ethernet "l posture port; end of the Ethernet to the central processor, and the other end connected to an external a data network; characterized in that, further comprising a digital time-division switching network chip, all the telephone interface and trunk interface chip are connected by a switching network codec module and a digital time-division, digital signal processors at that time division digital switching network, after the chip is connected, under the control of the central processor when the call destination is located external to the gateway, the telephone voice codec converted by the module into a digital signal by time-division digital switching network, the digital signal processor to transmit the compression process, then the the central processor is served by an external data network call destination; when the call destination is located within the gateway or trunk interface, switching the call to the corresponding end of the encoding and decoding module for converting speech signals into digital signals by time-division digital switching network directly after reduction of decoded reach the call destination.
2. 根据权利要求1所述的语音网关通讯设备,其特征在于:在中央处理器的控制下,所有电话接口、中继接口、数字信号处理器接口都能通过时分数字交换网络芯片中的时分数字交换网络进行信号的互相交换。 The voice gateway communication apparatus according to claim 1, wherein: under the control of the central processor, all the telephone interface, trunk interface, the interface can be a digital signal processor chip division switching network by time-division digital digital switching exchange network signals. ,
3. —种语音网关通讯方法,包括:a. 从电话接口或中继接口端接受呼叫信号,通过编解码模块将模拟语音呼叫模拟信号转换成数字信号并进行编码;其特征在于,还包括以下步骤:b. 判断被呼叫对象位于网关内部还是外部;如果被呼叫对象位于网关外部, 则执行以下步骤c;如果被呼叫对象位于网关内部或者是中继接口,则执行以下步骤d;c. 通过时分数字交换网络传输到数字信号处理器,由数字信号处理器对数字信号进行压缩处理形成语音包,然后由中央处理器将此语音包发送到外部数据网络,通过外部数据网络到达被呼叫对象;d. 在中央处理器的控制下,将编码了的数字信号直接通过时分数字交换网络交换到达被呼叫对象对应的编解码模块,还原成模拟语音信号,最终到达被呼叫对象。 3 - Species voice gateway communication method, comprising: a receiving interface or from the telephone trunk interface end of the call signal by the codec module converts the analog voice call converts an analog signal into a digital signal and encoded; characterized by further comprising. step:. b determines the call destination is located inside or outside the gateway; if the incoming call destination is located outside the gateway, the step C is performed; if it is located within the gateway or the call destination trunk interface, perform the following steps d; c through. time-division digital switching network to the digital signal processor, digital signal processing forms a voice packet is compressed by a digital signal processor, the central processor then sends this voice data packet to the external network, the call reaches the object via an external data network; d. under control of the central processor, the encoded digital signal directly reaches the exchange switching network corresponding to the call destination codec module through a digital time-division, reduction into an analog voice signal, and finally to the called object.
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