Conaito's brand-new SIP 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 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 by speaking and/or by text messages and delivers superior voice quality by integrating advanced configurable digital voice processing features including auto gain controller, acoustic echo cancellation, noise cancellation, reverb cancellation and Voice activity detection.
Here is a list of the main features of the conaito VoIP SIP client:
* 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, GSM6.10, g723 and iLBC Codec).
* NEW in v1.3! Multi-line support.
* NEW in v1.3! DTMF tones support (SIP INFO and RTP)
* NEW in v1.3! Recording voice conversation into PCM WAVE (.wav) file
* NEW in v1.3! Playing PCM WAVE (.wav) files to a voice conversation
* NEW in v1.3! Dynamically loadable codec´s support (GSM6.10, g723 codec's plug-in samples included)
* NEW in v1.3! Comes as ActiveX control (Webdemo included)
* Registration on SIP Server (SIP Registrar).
* Support UDP, TCP and TLS (currently experimental) as transport type.
* Plain text messaging.
* Microphone and Speaker Visualization support.
* Microphone and Speaker Volume with Mute support.
* Audio device selection.
* Fully-customizable user interface.
* Packetloss resistant (by using iLBC codec).
* Supports OLE Automation (scripting) by providing IDispatch interface and custom interfaces for C/C++ developers.
* Works with all kind of Internet connections.
* Royalty free licensing
* No Yearly/Monthly fee
* Very easy to incorporate
Advanced configurable digital voice processing features:
* AGC (auto gain controller).
* Acoustic echo cancellation or suppression.
* Noise cancellation or suppression.
* Reverb cancellation or suppression.
* VAD (Voice activity detection).
Having the above features available makes it simple to develop any type of VoIP-enabled application, like e.g. a SIP softphone, teaching tool, live support, chat, meeting tool or any other type of application which requires users being able to talk and type messages to each other.
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