The SIP SDK is a potent and flexible VoIP SDK that enables you to integrate SIP-based dialing and phone call features into your software applications quickly. With our SDK, you can provide your customers with a reliable and high-quality audio experience.
Looking for a powerful and highly versatile solution to accelerate the development of SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications and websites? Look no further than the SIP SDK for .NET and ActiveX.
Our brand-new SIP SDK is packed with features to make adding SIP functionality to your applications a breeze. It comes complete with a fully-customizable user interface and branding options, making it easy to create soft phone applications that meet your exact specifications.
One of the standout features of the conaito SIP SDK is its high performance VoIP conferencing client, which delivers crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software). What's more, it enables worldwide communication over the internet or intern networks through both voice and text messages.
With advanced configurable digital voice processing features such as auto gain controller, acoustic echo suppression, noise suppression, reverb suppression and mute microphone, the conaito SIP SDK delivers superior voice quality.
Key features of the SIP SDK for .NET and ActiveX include easily making and receiving SIP based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider, VoIP conferencing with crystal clear sound, encrypted SIP account settings, secure Weblicensing, multi-user conference support, multi-line (simultaneous calls) support, 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, RTCP, auto-answer, Do Not Disturb (DND), adaptive jitter buffer, adaptive silence, PLC (Packet Lost Concealment), AGC, VAD, AEC, and Noise Concealment.
The SIP SDK for .NET and ActiveX offers much more than just these features, making it a must-have for any developer looking to add SIP functionality to their software. Try it today!
Our brand-new SIP SDK is packed with features to make adding SIP functionality to your applications a breeze. It comes complete with a fully-customizable user interface and branding options, making it easy to create soft phone applications that meet your exact specifications.
One of the standout features of the conaito SIP SDK is its high performance VoIP conferencing client, which delivers crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software). What's more, it enables worldwide communication over the internet or intern networks through both voice and text messages.
With advanced configurable digital voice processing features such as auto gain controller, acoustic echo suppression, noise suppression, reverb suppression and mute microphone, the conaito SIP SDK delivers superior voice quality.
Key features of the SIP SDK for .NET and ActiveX include easily making and receiving SIP based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider, VoIP conferencing with crystal clear sound, encrypted SIP account settings, secure Weblicensing, multi-user conference support, multi-line (simultaneous calls) support, 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, RTCP, auto-answer, Do Not Disturb (DND), adaptive jitter buffer, adaptive silence, PLC (Packet Lost Concealment), AGC, VAD, AEC, and Noise Concealment.
The SIP SDK for .NET and ActiveX offers much more than just these features, making it a must-have for any developer looking to add SIP functionality to their software. Try it today!
What's New
Version 3.0:
* 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