• 通過任何的SIP網(wǎng)管或SIP兼容的IP電話服務提供商都能夠輕松地撥打和接聽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編解碼器
• 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/聊天消息
• 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
• 呼叫轉接和拒絕
• 支持呼叫轉移
• 選擇媒體輸入/輸出設備
• 聯(lián)機——會談/會議中也同樣支持
• 靜音麥克風/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（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
• 音頻文件內存緩存
• 可擴展的SIP URL功能
• 支持動態(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靜音支持
• 為NAT/防火墻保持活動數(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

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