Our VoIP SIP SDK provides a powerful and highly customizable solution to quickly add SIP based dial and receive phone calls features in your software applications. It accelerates the development of SIP compliant soft phone with a fully-customizable user interface and brand name. The Abto LLC 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 delivers superior voice quality. It supports DTMF, adaptive silence detection, adaptive jitter buffer! Abto LLC VoIP SIP SDK is based on IETF standards (SIP, STUN, etc.), so it should be compatible with other standard based products such as Asterisk, OpenSER other. Dynamically loadable codecs Registrar support STUN support Samples on Delphi, C#, VB, VB.NET, C#, HTML echo cancelation support of playing audio files Hold/Retrieve with sample Record conversation into WAV Delphi sample updated VB sample updated HTML JavaScript sample updated Extended OnIncomingCall event arguments Call transfer support Added status code on call reject Added Unauthorized and Trying events for user registration Fixed bug with default sound device Fixed to be recognized as valid ActiveX component Samples updated Mute Sound VPN support Fixed VB6 compilation issue Support of Authorization Id FAX recognition Noise reduction Auto gain Local number parameter Help updated Jitter buffer parameters Windowless samples on C++ and .NET The new Version 3.0.4 comes with: * Explicit multilines handling added * Conferences quality issues fixed * Configuation handling enhanced * Stability has been greatly improved * New samples added * Help updated * Samples updated