• 通過任何的SIP網(wǎng)管或SIP兼容的IP電話服務(wù)提供商都能夠輕松地?fù)艽蚝徒勇燬IP（會(huì)話初始協(xié)議）電話
• 能夠?yàn)榈汀⒏邘捰脩籼峁└咔逦恼Z音的VoIP會(huì)議。
• G711 A-Law, G711 U-Law, Speex, Speex-wb, GSM6.10, iLBC, L16和g729 & g723編解碼器
• 基于開放標(biāo)準(zhǔn)并且與所有的主要設(shè)備供應(yīng)商具有互操作性
• 支持UDP和TCP
• 多方語音會(huì)議支持/會(huì)議分割和加入，本地混合會(huì)議
• 多線路支持（多個(gè)同步呼叫）
• 具有控制發(fā)送/接受的SIP即時(shí)/聊天消息
• 支持集成STUN,TURN和ICE
• 配備新樣例SIP代理服務(wù)器提供于一個(gè)擁有SIP VoIP和即時(shí)消息網(wǎng)絡(luò)解決方案的conaito SIP Client ActiveX的捆綁中。
• P2P支持用于直接連接兩個(gè)SIP客戶端而不需要SIP服務(wù)器
• 支持外界代理服務(wù)器加密
• SIP賬戶設(shè)置（在您的網(wǎng)頁中加密SIP賬戶設(shè)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(shè)備
• 聯(lián)機(jī)——會(huì)談/會(huì)議中也同樣支持
• 靜音麥克風(fēng)/揚(yáng)聲器+音量指示器
• 自動(dòng)應(yīng)答
• 免打擾（請(qǐng)勿打擾）
• 自適應(yīng)抖動(dòng)緩沖器
• PLC（包丟失隱藏）
• AGC（自動(dòng)增益控制器）
• AES（聲學(xué)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(cè)）
• 記錄語音會(huì)話到PCM WAVE（.wav）文件中
• 在遠(yuǎn)程終端播放PCM WAVE（.wav）文件
• 音頻文件內(nèi)存緩存
• 可擴(kuò)展的SIP URL功能
• 支持動(dòng)態(tài)的可加載解碼器（即將推出）
• 作為ActiveX控件推出（包含具有已注冊(cè)簽名的CAB的Web demo）
• 在SIP服務(wù)器上注冊(cè)（SIP注冊(cè)器）
• 日志文件開/關(guān)設(shè)置
• 麥克風(fēng)和揚(yáng)聲器音量的靜音支持
• 為NAT/防火墻保持活動(dòng)數(shù)據(jù)包
• 完全可定制的用戶界面
• 微軟的Authenticode證書
• 與所有的Internet連接兼容
• 對(duì)NAT和其他防火墻友好
• 為NAT/防火墻保持活動(dòng)數(shù)據(jù)包
• 免版稅許可
• 無需每年/每月收費(fèi)
• 非常易于整合
• 為各種編程語言提供全功能的示例應(yīng)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

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