VoIP SIP SDK is a sip soft phone solution to quickly build voip softphone to dial and receive phone calls or add voip chat features in your software application.
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 sip api 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.
The new Version 3.0.11 comes with:
* Multiple lines explicit selection!
* Conferencing!
* Dynamically loadable codecs
* SIP Proxy authentication
* Multi-party voice conference
* Registrar support
* Play wav files into conversation
* Record conversation into file
* Hold/Retrieve call
* Forward Call (Blind Call Transfer)
* Transfer Call (Attended Transfer)
* Mute Sound
* VPN support
* Authorization Id
* Noise reduction
* Auto gain
* Jitter buffer parameters
* Samples on Delphi, C#, VB, VB.NET, C++ 2005, C++ 6.0, HTML (SIP ActiveX)
* Windowless samples on C++ and .NET
* DTMF
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