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Newsletter 03. September 2009
VoIP SIP Client SDK v1.0 released

www.conaito.com
conaito Technologies Introduces
VoIP SIP Client SDK - Release v1.0
A powerful and highly versatile VoIP SDK to accelerate development of SIP applications and websites

conaito VoIP SIP Client SDK A powerful and highly versatile VoIP SDK to accelerate development of SIP applications and websites 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 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 etc.), so it should be compatible with other standard based products such as: SER, Sip EXpress, OpenSER and Asterisk.

Lost no time and start a massive jump in development and a couple of new customers. Enjoy!

Learn more about our VoIP SIP solution here!

conaito VoIP SIP Client SDK
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VoIP SIP Client SDK v1.0 released

A powerful and highly versatile VoIP SDK to accelerate development
of SIP applications and websites


Here is a list of the main features of the conaito VoIP SIP Client SDK:

• 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, GSM6.10, iLBC, L16 and g729 Codec)
• Multi-party voice conference support
• Multi-line (simultaneous calls) support (Multiple concurrent calls)
• 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
• Select media input/output devices
• 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 commented sample applications for various programming languages
• Sample source code for C#, VB.NET, JavaScript (Webdemo), C++, VB 6.0 and Delphi 7.0
• For .NET framework as well and all development environments with ActiveX support

Advanced digital voice processing features:

• AGC (auto gain controller)
• Acoustic echo cancellation or suppression (AES)
• 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.

Download a free Trial version

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conaito online documenationNeed help? conaito Support

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