SIP Phone DLL
Publisher: conaito
Version: 3.0
License: Commercial
Trial Limitations: 30 Days, Trial have all features like Full Version
Price: $999.00
Operating Systems: WinXP,Windows2000,Windows2003,Windows Vista Starter,Windows Vista Home Basic,Windows Vista Home Premium,Windows Vista Business,Windows Vista Enterprise,Windows Vista Ultimate,Windows Vista Home Basic
System Requirements: Visual Basic .NET, Visual C++ .NET, Visual C# .NET, Delphi .NET, Visual Basic, Visual C++, Delphi, Visual Basic 6.0, all development environments with a ActiveX support
File Size: 5.58 MB
Total Views: 781
Total Downloads: 2
Rating: SIP Phone DLL not yet rated. [Submit Review] [Award Rating]
Release Date: 6/18/2008
Last Update Date: 10/14/2017
SIP Phone DLL Short Description:
SIP Phone DLL - A powerful and highly versatile VoIP SDK for SIP applications
SIP Phone DLL Long Description:
SIP Phone DLL for .NET and ActiveX - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.

Our brand-new SIP Phone DLL 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 SIP Phone DLL 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!
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