SIP VoIP SDK

The SIP VoIP SDK provides a means to add SIP based dial and receive phone call features in software applications. You can use it in the development of a SIP compliant soft phone with a fully-customizable user interface and brand name.

The ABTO LLC SIP VoIP SDK contains a VoIP conferencing client capable of delivering sound for both low and high-bandwidth users and SIP compatible devices (hardware and software). It supports DTMF, adaptive silence detection, and an adaptive jitter buffer.

The vendor states that the product is based on IETF standards (SIP, STUN, etc.); it should be compatible with other standard based products such as Asterisk, OpenSER, and others.

Version features:

- v1.1: Dynamically loadable codecs; registrar support; STUN support; samples on Delphi, C#, VB, VB.NET, C#, HTML; echo cancellation; support of playing audio files

- v1.2: Hold/retrieve with sample; record conversation into WAV; Delphi, VB, and HTML/JavaScript sample updated; extended OnIncomingCall event arguments

- v1.3: 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