Download VoIP SIP SDK with DLL, ActiveX and .NET 2.5.2

VoIP SIP SDK with DLL and ActiveX - A highly versatile SDK for SIP applications

Download (5.13 MB)

User Rating:: ( Unrated )
License:: Commercial ( $999 to buy )
Platform:: Windows ( 2000/XP/Vista )
Updated:: May 13, 2008
Downloads:: 17

About VoIP SIP SDK with DLL, ActiveX and .NET

VoIP SIP SDK - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.

Our brand-new SIP SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications and websites. It accelerates the development of SIP/ RTP compliant soft phone with a fully-customizable user interface and brand name.

The conaito 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/or by text messages and delivers superior voice quality by integrating advanced configurable digital voice processing features including auto gain controller, acoustic echo suppression, noise suppression, reverb suppression and mute microphon

Key Features

Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider
VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A-Law, G711 U-Law, Speex, GSM6.10, iLBC, L16, g723 and g729 Codec)
Encrypt SIP account settings (protect your SIP account settings in websites)
Secure Weblicensing (protect your license in websites)
Multi-User conference support
Multi-line (simultaneous calls) support (Multiple Concurrent calls)
Call Hold support
Call Transfer support
Instant text messaging (MIME) support and typing indication
Mute microphone/speaker for each line
DNS SRV resolution for SIP servers (RFC 3263)
Stereo codec (L16)
RTCP
Auto-answer
Do Not Disturb (DND)
Adaptive jitter buffer
Adaptive silence
Advanced configurable digital voice processing features

And much more. Try it today!.

Changes in this version:

* Multi-User conference support* Make and receive SIP based phone calls* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A-Law, G711 U-Law, Speex, GSM6.10, iLBC, L16, g723 and g729* Secure Weblicensing, Encrypt SIP account settings