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VoIP SIP SDK for .NET and ActiveX 3.0

VoIP SIP SDK - A powerful and highly versatile VoIP SDK for SIP applications
Publisher: conaito Technologies
Category: Games
Version: 3.0
License: commercial
Cost: 999$
Size: 5.58 MB
Updated: 11 Nov 2008
Screenshot: view screenshot
Bookmark: Click here
Review
VoIP SIP SDK for .NET and ActiveX - 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)
* RTCP
* Auto-answer
* Do Not Disturb (DND)
* Adaptive jitter buffer
* Adaptive silence
* PLC (Packet Lost Concealment)
* AGC, VAD, AEC, Noise Concealment

...and much more. Try it today!

Changelog:
Version: 3.0(17 Jun 2008)
* Multi-Line, 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/U, Speex, GSM6.10, iLBC, L16, g723, g729)
* UDP, TCP, STUN, TURN, ICE, VAD, DTMF support
* Secure SIP account settings
Version: 2.5.0.4(15 May 2008)
* 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
Version: 2.4(25 Feb 2008)
* 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
* NEW in v2.4! Secure Weblicensing, Encrypt SIP account settings
Version: 1.0
Tags: Sip Sdk | Sip Rtp | Rtp | Sip Activex | Sip Lib
download (conaitoVoIP_SIP_SDK.zip - 5.58 MB)