VoIP SIP SDK for .NET and ActiveX 3.0
VoIP SIP SDK - A powerful and highly versatile VoIP SDK. Our 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.
Product Specifications
User Rating: | item has less then 1 vote |
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FileCart Rating | ||
Submitted by: | conaito | |
Homepage | Visit Homepage | |
License | Commercial | |
Expires | 30 Days | |
Price | $999 | |
Downloads | 543 | |
OS Support | Windows Install and Uninstall | |
File Name | conaitoVoIP_SIP_SDK.zip | |
File Size | 5.58 MB ( 5,718 KB ) | |
Published | Sep 03, 2007 (17 years ago) | |
Updated | Dec 17, 2013 (11 years ago) |
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Product Details
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!
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!
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 |
Release Info: | Major Update on Jun 17, 2008 |
Release Notes: | * 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 |
Related Tags: | sip sdk SIP/ RTP RTP SIP activeX SIP Lib |
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