我們?nèi)碌腟IP Client SDK提供了一個功能強大和高度靈活的解決方案，用于添加快的基于和接聽電話功能的SIP（會話初始協(xié)議）到您的軟件應用程序和網(wǎng)站。它以一個*可定制的用戶界面和品牌名稱加快了SIP/RTP兼容的軟電話的開發(fā)。該conaito VoIP SIP Client SDK包含了一個高性能的VoIP會議客戶端，他具有為低、高帶寬用戶和SIP兼容的設備（硬件和軟件）提供高清晰的語音能力。它通過互聯(lián)網(wǎng)或內(nèi)部網(wǎng)使得世界范圍內(nèi)的通信成為可能，或者通過集成數(shù)字設備處理功能，包括自動增益控制器（AGC），回聲抑制（AES）和噪音抑制來提供高品質(zhì)的語音質(zhì)量。
Our brand-new SIP Client SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications and websites. It accelerates the development of SIP/RTP compliant soft phone with a fully-customizable user interface and brａnd name. The conaito VoIP SIP Client SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software). It enables a worldwide communication over the internet or intern networks either and delivers superior voice quality by integrating digital voice processing features including auto gain controller (AGC), acoustic echo suppression (AES) and noise suppression.
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主要特點
- 通過任何的SIP網(wǎng)管或SIP兼容的服務提供商都能夠輕松地和接聽SIP（會話初始協(xié)議）電話
- 能夠為低、高帶寬用戶提供高清晰的語音的VoIP會議。
- G711 A-Law, G711 U-Law, Speex, Speex-wb, GSM6.10, iLBC, L16和g729 & g723編
- 基于開放標準并且與所有的主要設備供應商具有互操作性
- 支持UDP和TCP
- 多方語音會議支持/會議分割和加入，本地混合會議
- 多線路支持（多個同步呼叫）
- 具有控制發(fā)送/接受的SIP即時/消息
- 支持集成STUN,TURN和ICE
- 配備新樣例SIP代理服務器提供于一個擁有SIP VoIP和即時消息網(wǎng)絡解決方案的conaito SIP Client ActiveX的捆綁中。
- P2P支持用于直接連接兩個SIP客戶端而不需要SIP服務器
- 支持外界代理服務器加密
- SIP賬戶設置（在您的網(wǎng)頁中加密SIP賬戶設置）
- 支持線路暫停/取消暫停
- 呼叫轉(zhuǎn)接和拒絕
- 支持呼叫轉(zhuǎn)移
- 選擇媒體輸入/輸出設備
- 聯(lián)機——會談/會議中也同樣支持
- *麥克風/揚聲器+音量指示器
- 自動應答
- 免打擾（請勿打擾）
- 自適應抖動緩沖器
- PLC（包丟失隱藏）
- AGC（自動增益控制器）
- AES（聲學回聲消除或抑制）
- 噪音消除或抑制
- 支持DTMF音調(diào)（產(chǎn)生/檢測）
- 記錄語音會話到PCM WAVE（.wav）文件中
- 在遠程終端播放PCM WAVE（.wav）文件
- 音頻文件內(nèi)存緩存
- 可擴展的SIP URL功能
- 支持動態(tài)的可加載（即將推出）
- 作為ActiveX控件推出（包含具有已注冊簽名的CAB的Web demo）
- 在SIP服務器上注冊（SIP注冊器）
- 日志文件開/關設置
- 麥克風和揚聲器音量的*支持
- 為NAT/防火墻保持活動數(shù)據(jù)包
- *可定制的用戶界面
- 微軟的Authenticode證書
- 與所有的Internet連接兼容
- 對NAT和其他防火墻友好
- 為NAT/防火墻保持活動數(shù)據(jù)包
- 免版稅許可
- 無需每年/每月收費
- 非常易于整合
- 為各種編程語言提供全功能的示例應用程序，如C#，VB.NET，JavaScript（Web 演示），VB 6.0和Delphi的示例源代碼。
- 支持.NET框架以及所有的ActiveX環(huán)境
獎項
Main Features
- Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider
- VoIP conferencing with crystal clear sound even for both low and high-bandwidth users
- G711 A-Law, G711 U-Law, Speex, Speex-wb, GSM6.10, iLBC, L16 and g729 & g723 Codec
- Open standards-based and interoperable with all of the major equipment vendors
- UDP and TCP support
- Multi-party voice conference support/ Conference split and join, locally mixed conferences
- Multi-line support (multiple simultaneous calls)
- SIP Instant/Chat Messaging with send/receive controlling
- Integrated STUN, TURN and ICE support
- Comes with new sample SIP Proxy Server to provide in bundle with the conaito SIP Client ActiveX a ready up own SIP VoIP and Instant Messaging network solution.
- P2P support for directly connections between 2 SIP clients without SIP Server
- Outbound proxy server support
- Encrypted SIP account settings (encrypted SIP account settings in your webpage)
- Line Hold/Un-hold support
- Call forwarding and rejection
- Call transfer support
- Select media input/output devices
- on-the-fly - also during a conversation/ conference)
- Mute microphone/speaker + level indicator
- Auto-answer
- DND (Do Not Disturb)
- Adaptive Jitter buffer
- PLC (Packet Lost Concealment)
- AGC (auto gain controller)
- AES (Acoustic echo cancellation or suppression)
- Noise cancellation or suppression
- DTMF tones support (generation/detection)
- Recording voice conversation into PCM WAVE (.wav) file
- Playing PCM WAVE (.wav) files to the remote end
- Audio file memory cache
- Extended SIP URL functions
- Dynamically loadable codec support (coming soon)
- Comes as ActiveX control (Web demo with ready-up signed CAB included)
- Registration on SIP Server (SIP Registrar)
- Log file on/off setting
- Microphone and Speaker Volume with Mute support
- Keep-alive packets to NAT/firewall
- Fully-customizable user interface
- Microsoft Authenticode Certificate
- Works with all kind of Internet connections
- Friendly to NAT and other firewalls
- Keep-alive packets to NAT/firewall
- Royalty free licensing
- No Yearly/Monthly fee
- Very easy to incorporate
- Fully sample applications for various programming languages such as sample source code for C#, VB.NET, JavaScript (Webdemo), VB 6.0 and Delphi
- For .NET framework as well and all development environments with ActiveX support
Awards
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