語音網關 的英文怎麼說

中文拼音 [yǔyīnwǎngguān]
語音網關 英文
ag audio gateway
  • 語 : 語動詞[書面語] （告訴） tell; inform
  • 音 : 名詞1. （聲音） sound 2. （消息） news; tidings 3. [物理學] （音質） tone 4. （姓氏） a surname
  • 網 : Ⅰ名詞1 （捕魚捉鳥的器具） net 2 （像網的東西） thing which looks like a net 3 （像網一樣的組織或...
  • 關 : Ⅰ動詞1 （使開著的物體合攏） close; shut 2 （圈起來） shut in; lock up 3 （倒閉; 歇業） close down...
  • 語音 : speech sounds; pronunciation; voice
  1. Each channelized voice t1 / e1 line terminated into a voice gateway supports approximately 24 simultaneous phone calls

    每個終止於語音網關的通道化語音t1 / e1線可支持約24個同時進行的電話呼叫。
  2. The basic characteristics of the current data network are point - to - point, connectless, doing one ' s endeavor, no quality of service, etc. these characteristics do not meet the requirement of real - time services, therefore, the realization of voip need support of the some key technology. these technologies includes : speech sound coding and data compression, real - time transmission and control, mute compression and multicast, acoustic - echo cancellation and comfort noise generator, dynamic monitor and guarantee of quality of network service, as well as, the compatible of different network and different protocol with each other

    但現有的數據網路的基本特性：點對點的、無連接的、盡力而為的、沒有服務質量保證等特性並不適合與實時的業務要求，因此voip的實現需要一些關鍵技術的支持，這些技術包括：語音編碼和壓縮技術、實時傳輸和控制技術、組播技術、靜音壓縮和舒適噪聲生成技術、回聲消除技術、網路服務質量的動態監測和保證技術、以及不同的網路、不同的協議之間的互連互通等等。
  3. Ip voice gateway is a node between the communication control unit and the server of trunking system in trunking system network. its duty is digital compress coding and packetization of the voice in trunking system. it located on the edge of the network and accessed the trunking system into ip network from the kernel network angle

    語音網關作為集群通信系統ip聯網的關鍵設備，它處于集群系統話路控制單元和集群系統服務器之間，起著語音壓縮、分組傳輸的重要作用;從網路的角度來看，它處于ip網路和集群無線通信系統之間，把集群通信系統接入ip網路。
  4. It can provide many functions : the conversion between transfer formats, the conversion between communication protocols, the conversion between analog signals and digital signals, the conversion of control commands, real time audio code / decode, ip call control, control of rtp / rtcp, call detail logs, accounting and some of gatekeeper ' s functions such as address tansfer / search.

    它提供以下功能：傳輸格式間的轉換，通信規程間的轉換，音頻、數據格式轉換，呼叫控制信令的轉換，對pstn的呼叫控制功能， pstn電路交換功能，實時語音處理語音壓縮功能， ip網路呼叫控制功能，對實時對話協議rtp rtcp的控制功能，呼叫日誌，計費功能，還包括關守的地址翻譯搜索等功能。
  5. This dissertation has discussed the r & d process of a broadcasting system based on network for fire fighting. this broadcasting system is based on the lan, and it has used the technology of voip. it has implemented the network speech communication by packet switching, and it has removed the shortcoming of the traditional broadcasting system which is based on circuit

    本文系統的論述了網路消防廣播系統的研製過程，該系統基於tcp / ip局域網路，利用了voip的相關技術，採用分組交換的原理來實現語音的網路傳輸，易於擴展，不受距離的限制，消除了傳統基於音頻電路消防廣播的弊端。
  6. Design and implementation of sociel insurance voice gateway system

    社保語音網關系統的設計與實現
  7. The rest of this paper is organized as follows. in section 1, the relevant knowledge about voip, mobile ipv6 and qos are presented

    論文第一章討論了無線移動網路實時語音業務qos研究的相關背景知識，對基於ipv6的voip 、移動ipv6及qos模型進行了闡述。
  8. The characteristic and key technologies of the system are as follows : ( 1 ) in realizing the live broadcast of audio and video, the problem of immense multimedia data and low networks bandwidth utilization ratio is solved by using mpeg - 4 as format of audio and video data. audio and video data are collected by video card cv500 which developed by beijing sum tone company ; meanwhile, the contradictory between the delay of networks transmitting and the quality of the image is well solved by setting a " bi - buffer area "

    系統實現中解決的關鍵問題和特色主要有以下幾個方面： （ 1 ）在視音頻直播功能的實現中，通過使用北京算通公司的cv500視頻採集卡和cv500sdk進行視音頻數據採集，並採用當今最新的圖像和語音編碼壓縮標準mpeg - 4作為視音頻數據的採集格式，既保證了圖像的質量，又大大縮減了視音頻所佔的帶寬，從而解決了多媒體數據量大、網路帶寬利用率低的問題；同時，通過設置環形緩沖區的辦法來調和網路傳輸延時與圖像質量之間的矛盾，取得了較好的效果。
  9. New initiatives include setting up of more regional cyber centres, exploring the feasibility of text - to - speech translation services for government web sites and collaborating with it professional bodies to encourage the private sector to adopt to barrier - free web design methods

    有關的新措施包括設立更多地區性的數碼中心，研究可否為政府網站提供文字轉換語音的翻譯服務，以及與資訊科技專業組織合力鼓勵私營機構採用無障礙網頁的設計方法。
  10. We build a phone gateway to connect the pstn and intranet and add telephone function to every multimedia computers. in this thesis there is detail description for the technologies and ways to implement the system

    我們通過建立一個配備語音卡的電話網關來實現電話網和數據網路的互聯，並且為每臺計算機增加一個普通電話的功能。
  11. As conforming with the h. 323v2 and relevant national standards, the terminal has high interoperability with other h. 323 device, and has some main function units defined by standards such as system control unit, h. 225 layer unit, lan interface unit, and audio codec unit

    323國際標準第二版和相關國內標準，能夠和其它h 323設備互通，具有系統控制單元、 h 225layer單元、網路介面單元和語音編解碼單元等協議中所規定功能單元。
  12. At the end the author discusses the real - time feature through analyzing the device driver. in the third chapter, the author provides the specification and general design structure for the control unit of mdsl. he discusses different communication types between the record unit and the control unit and provides methods of how to resave data from the record unit and replay local voice data rapidly

    控制單元要對記錄單元進行監測和控制，涉及到大量的網路通信，因此作者對網路通信的並發服務方式進行了歸納總結，深入探討了記錄單元和控制單元多種的通信實現方式，對控制單元系統初始化，控制單元與記錄單元交互的時序，控制單元的關鍵任務，數據轉存及語音數據的本機回放進行了細致深入的研究，並在文中給出了自己的演算法。
  13. From the view of our practice, the problem of network plan and build, integrated oam of services based on ngn / softswitch is study in this paper. first, look insight what is ngn and softswitch, main features and advance of these technique, analysis the develop policy ; with the requirement of network build and service management, existed network and services, schemas of man integrated service network and softswitch test network are analyzed mainly focused on the topic of relationship between these network and the existed, the impact on the network and services development in next stage. some suggest is also given

    首先在闡述ngn和軟交換基本技術、主要特點、優勢和需要解決的主要問題基礎上，分析了ngn和軟交換的發展方式和策略;根據市場對網路建設和營運管理的需求，結合以ngn和軟交換為核心的新一代電信技術、現有的網路與業務條件，詳細分析了城域綜合業務網和語音、數據、視頻綜合業務的軟交換試驗網這兩種當前主要的建設方案:在與現有網路和業務關系、對今後網路和業務開發的影響等方面，比較了實施方案的優缺點，提出了修正的網路建設方案。
  14. This article mainly discusses the basic theories and related protocol and technology of speech sound communication based on ip. the key discuss is the support of real - time services that is provided by the current ip network, codec and the acoustic - echo cancellation, as well as the dynamic monitor of network communication quality of service. it gains some conclusion by compare the merit and shortcoming between the h. 323 protocol and sip protocol and analyzing their bases : rtp / rtcp protocol

    本文主要論述基於ip的語音通信所涉及的基本理論和相關的協議與技術，重點論述當前ip數據網路對實時業務提供的支持、語音編碼、回聲消除和網路通信服務質量的動態監測，在協議的分析上比較h . 323和sip協議之間的優缺點，並具體分析這兩個協議的基礎： rtp rtcp協議。
  15. This paper researched the network structure of the gsm - r system according to the operation requirement. according to the gsm supplementary service protocols established by etsi, this paper researched the pivotal technologies of the gsm - r system : vgcs and vbs, and at the same time designed the realization project of the vgcs vbs module. this paper also designed the service flow of the vgcs vbs call in the network side and at last validated the function of the module with the specific testing tool

    本文針對gsm - r系統的業務需求對gsm - r系統的網路結構進行了分析和研究;參照etsi制定的gsm補充業務相關協議，深入研究分析了gsm - r系統中的關鍵技術:語音廣播呼叫（ vbs ）和語音組呼（ vgcs ） ，同時參考現有的zxg10 - msc系統（中興公司基於g10交換機的移動交換系統）進行了vgcs / vbs模塊的方案設計，並詳細規劃了vgcs和vbs功能在網路側的業務流程，最後使用測試工具對模塊的功能進行了驗證。
  16. In this paper, we give a detail discussion on the key technology, including software and hardware designing of g. 729a multi - channel speech codec realtime implemention on a simple dsp processor - tms320c6202. in combination with the requirements of a military communication network, the atm adaptation solution of g. 729 coder bit stream is analyzed. a kind of new atm adaptation technology - aal2 is introduced. the analyse and research of aal2 are provided

    本文詳細討論了多路g . 729a語音編解碼器在一片dsp處理器tms320c6202上實時實現的軟硬體設計和關鍵技術。結合某軍事通信網設備的需要，進而對g . 729語音編碼的碼流的atm適配方案進行了分析。提出了用一種新的atm適配技術- - aal2進行適配的方案。
  17. In the paper, the embeded process platform and voice compression moudle were introduced firstly, then host port interface circuit was designed by which the host processor mpc852t communicated with ac48304. and then em interface and ethernet interface of the gateway were designed, which communicatewith trunking system and server respectively

    本文的主要工作是在了解嵌入式處理平臺和語音壓縮模塊的基礎上，設計了powerpc處理器擴展ac48304的hpi介面電路;並設計了語音網關與集群系統的em介面以及以太網介面。
  18. In this simple, non - personalized example, a caller dials the phone number of the voice gateway

    在這一簡單的、非個性化的例子示例中，主叫用戶方撥打語音網關的電話號碼。
  19. On the other hand the voice packets from the server were transmitted to the trunking system in the converse process

    另一方面，從集群系統服務器來的語音包在語音網關中經過相反的過程到達集群系統。
  20. The analoge voice from the trunking system was pcm coded, compressed coded, packtized and then transmitted to the server by udp / ip link in the voice gateway

    語音網關把來自集群系統的模擬語音，經過pcm編碼，語音壓縮編碼，分組， rtp包封裝，經udp / ip鏈路傳輸到集群系統服務器。
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