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 conaito 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 forwarding and transfer, DTMF, Packet Loss Concealment (PLC), adaptive jitter buffer, record and playing WAV and much more!

conaito VoIP SIP Client SDK is based on IETF standards (SIP, RTP, STUN, TURN, ICE etc.), so it should be compatible with other standard based products such as: SER, Sip EXpress, OpenSER and Asterisk.

• 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 (coming   soon) Codec)
• Open standards-based and interoperable with all of the major equipment vendors
• Multi-party voice conference support (Conference split and join, locally mixed conferences)
• Multi-line support (multiple simultaneous calls)
• Integrated STUN, TURN and ICE support
• 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 as well during a conference)
• Mute microphone/speaker
• 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
• 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
• 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), C++, VB 6.0 and Delphi
• For .NET framework as well and all development environments with ActiveX support

Easy, familiar, event-driven call control ActiveX
• Easy to use; quick development
• Support for .NET framework as well and all development environments with ActiveX support
• Very easy to incorporate

Rich call control feature set
• Multi-party voice conference support (Conference split and join, locally mixed conferences)
• Multi-line support (multiple simultaneous calls)
• Locally mixed conferences
• Hold/Mute
• Call transfer
• Call forwarding and rejection

Industry leading SIP support
• RFC3261 compliant SIP stack
• RFC 2833 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 (auto gain controller)
• AES (Acoustic echo cancellation or suppression)
• Noise cancellation or suppression

… 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.

For conaito VoIP SIP clients to be able to interact with each other they must connect to a SIP gateway or SIP based IP-Telephony service provider.

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.

We hope you enjoy the new conaito VoIP SIP Client SDK – A powerful and highly versatile VoIP SDK to accelerate development of SIP applications and websites.

The conaito VoIP SIP Client SDK provides the documentation, samples and related libraries you need to integrate with other applications or websites.

Conaito VoIP SIP Client SDK includes a ActiveX that can be used from any programming language like Visual Basic, .NET (VB and C#), C++, Visual Basic, Delphi, ASP, JSP, PHP, JavaScript, VBScript, etc. The conaito VoIP SIP Client SDK is designed to be used by Automation clients.

The contents of conaito VoIP SIP Client SDK and the supported development environments include all of the necessary software components for building systems based on conaito VoIP SIP Client SDK including documented operational software applications, examples (with source code), explanations as well as necessary service programs, libraries and components.

The supported development environments include:

• Visual Basic .NET
• Visual C++ .NET
• Visual C# .NET
• ASP.NET
• ASP, JSP, PHP
• JavaScript/HTML
• Visual Basic
• Visual C++
• Borland Delphi
• and all development environments with ActiveX support

System requirements:
Operating system: Windows XP, 2000, 2003, Vista, Windows 7