The award-winning VoIP SDK is a powerful and highly versatile set of tools designed to dramatically accelerate development of SIP applications. VoIP SDK enable flexible, rapid application development options for creating desktop softphone clients, Rich Internet Applications (RIA) and telephony extensions into existing enterprise and web applications, It's complies with IETF and 3GPP standards, delivers high performance, and provides advanced API layer for full user control and flexibility.
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 3.6.1 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
| 