With the Conaito VoIP SIP Client SDK you may easily establish calls to regular phones (PSTN) from your desktop and Web applications. You can utilize it to develop IVR or more complex SIP Server/Media center solutions. The SDK comes with new sample SIP Proxy Server to provide in bundle with the Conaito SIP Client a ready up SIP VoIP and Instant Messaging network solution. 

Typical scenario - How to make a call:

1. Build SIPActiveX object (specify required properties such as LoginID, Password, RegistrationProxy, LicenseKey etc.)
2. Initialize SIPActiveX.
3. Register with SIP Proxy server. Register method takes no parameters because all SIP user account settings must be set before calling Initialize method.
4. Handle calls.
5. Unregister on SIP Proxy Server (see UnRegister method for more information).
5. Shutdown SIPActiveX.

Usage GUI applications:

1. Register Conaito SIP Client ActiveX control (in Web applications the control is registered automatically - CAB is used).
2. Register the control with your development environment.
3. Put the control on main form. Set it's "Visible" property to "false".

Transports: The core SIP stack provides following transports:

• UDP (configurable by UDPPort)
• TCP (configurable by TCPPort)

Networking: The Conaito VoIP SIP Client SDK is fully compatible with most typical SIP servers (Asterisk, Sip EXpress e.t.c.). It supports both NAT traversing (STUN, ICE) and reliable TCP for the signaling protocol. Also it's usable for sending text messages, play and record conferences (WAV) and DTMF tones. 

Media: You may use 8000/16000 Hz sampling codecs (G711 A-Law, G711 U-Law, Speex, Speex-wb, GSM6.10, iLBC and g729 & g723 Codec support). The SDK also provides sending and receiving of DTMF tones, playing ring-tone is also supported. You can play WAV files and record conversations in WAV format.