Conaito VoIP Video SIP SDK为快速添加基于拨打和接收视频电话的SIP，提供了一个功能强大并且高度灵活的解决方案。它加速了SIP/RTP兼容协议的软工具的发展，另外它还能够定制用户界面和品牌名称。
It contains a high performance voip video conferencing client capable of delivering crystal clear sound and crisp video 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 and video quality by integrating digital voice processing features including auto gain controller (AGC), acoustic echo suppression (AES) and noise suppression. It supports multiple lines, multi-party voice conference, call hold, call forwarding and transfer, DTMF, Packet Loss Concealment (PLC), adaptive jitter buffer, record and playing WAV or MP3 and much more!
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产品主要特征：
通过任何一个网关或协议的网络电话服务提供商，轻松地制作和接收一个会话启动协议（会话启动协议）的电话。
声音清晰的网络视频会议和低带宽或高带宽用户都能拥有的清晰的视频播放功能。
支持的音频编解码器有：G729，G723，G.711 A-Law，G711 u-Law，Speex，Speex WB，gsm6.10和iLBC
支持的视频编解码器有：H263，H263 +，H261，H264，VP8 WebM和Theora
以开放标准为基础，同所有主要设备供应商的共同操作功能
UDP和TCP的支持
多党派语音会议支持会议的拆分与人员加入以及局部的混合型会议
多行支持（多个同时调用）
多个账户注册支持（无限制）
SIP即时聊天消息的发送与接收控制的功能
集成对STUN, TURN 和ICE的支持
SIP服务器演示与conaito SIP客户端的捆绑。
在没有SIP服务器的情况下，P2P支持直接连接2个SIP客户端。
注册服务器（注册商）
出站代理服务器支持
加密SIP帐户设置（加密您的网页帐户设置）
查看产品特征详情请点击这里>
Key features of the VoIP Video SIP SDK
- Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider
- VoIP video conferencing with crystal clear sound and crisp video even for both low and high-bandwidth users
- Supported audio codec´s: G729, G723, G711 A-Law, G711 U-Law, Speex, Speex-wb, GSM6.10 and iLBC
- Supported video codec´s: H263, H263+, H261, H264, VP8 WebM and Theora 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)
- Multiple sip accounts registration support (unlimited)
- SIP Instant/Chat Messaging with send/receive controlling
- Integrated STUN, TURN and ICE support
- Comes with SIP Server demo to provide in bundle with the Conaito SIP Client a ready up own SIP VoIP and Instant Messaging network solution.
- P2P support for directly connectionsbetween 2 SIP clients without SIP Server
- Registration on SIP Server (SIP Registrar)
- 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)
- Microphone and speaker volume access
- Mute microphone/speaker + level indicator
- Auto-answer
- DND (do not disturb)
- Adaptive Jitter buffer
- PLC (packet lost concealment)
- AGC (automatic gain controller)
- AES (acoustic echo cancellation or suppression)
- Noise cancellation or suppression
- DTMF tones support (generation/detection)
- Recording voice conversation into PCM WAVE and mp3 (.wav and .mp3) file
- Playing PCM WAVE and mp3 (.wav and .mp3) files to the remote end
- Audio file memory cache
- Extended SIP URL functions
- Comes as ActiveX control (Webdemo with ready-up signed CAB included) and native DLL with .NET Interface (no ActiveX registration necessary) for easy usage in .NET development
- Log file on/off setting
- Keep-alive packets for NAT/Firewall
- Fully-customizable user interface
- Production-ready Microsoft Authenticode Certificate certified
- Fair royalty free licensing
- No Yearly/Monthly fee
- Very easy to incorporate
- SDK comes in bundle with ActiveX control (Webdemo with ready-up signed CAB included), native DLL with .NET Interface for easy usage in .NET development (no ActiveX registration necessary)
- Fully sample applications for various programming languages such as sample source code for C#, VB.NET, C++, VB 6.0, Delphi and HTML/JavaScript (Webdemo) for ActiveX, DLL and .NET version
- For .NET framework as well and all development environments with native DLL, .NET or ActiveX support
Easy, familiar, event-driven call control ActiveX, DLL and .NET
- Easy to use; quick development
- Powerful .NET Interface for easy .NET development (no ActiveX registration necessary)
- Support for .NET framework as well and all development environments with native DLL, .NET or ActiveX support
- Very easy to incorporate
Rich call control feature set
- Multi-party voice conference support (conference split/ join, locally mixed conferences)
- Multi-line support (multiple simultaneous calls)
- Multiple sip accounts registration support (unlimited)
- SIP Instant Messaging
- Locally mixed conferences
- Hold/Mute
- Call Transfer
- Call Forwarding and Rejection
Industry leading SIP support
- RFC3261 compliant SIP stack
- RFC2833 out-of-band DTMF signaling
- Integrated STUN, TURN and ICE support
Comprehensive configuration support
- Select media input/output devices (on-the-fly as well during a conversation/conference)
- Configurable Ports (RTP, SIP UDP, SIP TCP, STUN, TURN, ICE)
- SIP Proxy
Advanced digital voice processing features
- AGC (automatic gain control)
- AES (acoustic echo cancellation or suppression)
- Noise cancellation or suppression
正式授权
京公网安备 11010102002019号
