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conaito Technologies Introduces the Softphone SDK, SIP SDK, VOIP SDK, Softphone, IVR SDK, IMS SDK

Software Product: Softphone SDK, SIP SDK, VOIP SDK, Softphone, IVR SDK, IMS SDK - Version 1.3
Author: conaito Technologies

Softphone SDK, SIP SDK, VOIP SDK, Softphone, IVR SDK, IMS SDK - ActiveX and .NET - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.

Softphone SDK, SIP SDK, VOIP SDK, Softphone, IVR SDK, IMS SDK for VoIP conferencing

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 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. It supports multiple lines, multi-party voice conference, call hold, call transfer, DTMF, Packet Loss Concealment (PLC), adaptive jitter buffer, record and playing

* Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users
(G711 A-Law, G711 U-Law, Speex, GSM6.10, iLBC, L16 g723 and g729 Codec)
* Multi-party voice conference support
* Multi-line (simultaneous calls) support
* Line Hold/Retrieve support
* Call Transfer support
* Mute microphone/speaker
* Auto-answer
* Do Not Disturb (DND)
* 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
* Dynamically loadable codec support
* Comes as ActiveX control (Web demo with ready-up signed CAB included)
* Registration on SIP Server (SIP Registrar)
* Microphone and Speaker Volume with Mute support

... and much more!

Having the above features available makes it simple to develop any type of VoIP-enabled application, like e.g. a SIP softphone, IVR solution, teaching tool, live support, voip chat, meeting tool or any other type of application which requires users being able to talk and type messages to each other.

Just relax!
Please, don't hesitate trying our VoIP SIP Client SDK at once and get yourself, as well as your customers, the exciting experience of easy, fast and high quality standard applications which VoIP-enable your application and website.