VoIP SDK provides a powerful and highly customizable solution (SDK
includes such features: SIP activeX control, Dynamically loadable
codecs, DTMF, STUN support, IM interface, Adaptive silence detection
and many more) to quickly add SIP based dial and receive phone calls
(to make a long story short - voip client) features in your software
applications. It accelerates the development of SIP compliant
softphone with a fully-customizable user interface and brand name.
The VoIP soft phone 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
by voip soft phone. It supports DTMF, adaptive silence detection,
adaptive jitter buffer! Also using our voice conversation sdk you can
work through firewall or NAT.
Voip sdk is based on IETF standards (SIP, STUN, etc.), so it should be
compatible with other standard based products such as Asterisk,
OpenSER other.
Please check Features page for more details about our voice internet
phone sdk. Also you can download latest evaluation version on our
Download page (evaluation version includes VoIP C++, VoIP Delphi, Sip
ActiveX, C# and other languages examples, so you can see our sip dll
in work).