SIP Phone DLL is a potent and adaptable VoIP SDK. It allows a fast inclusion of dial and receive SIP-based phone calls features in software applications.
The SIP Phone DLL for .NET and ActiveX is an amazing VoIP SDK that allows developers to build SIP applications quickly. This new software provides a highly versatile solution to quickly add Session Initiation Protocol (SIP) based dial and receive phone call features in software applications and websites. The SIP Phone DLL accelerates the development of soft phones that are SIP/RTP compliant, with a fully-customizable user interface and brand name.
The Conaito SIP Phone DLL cordially 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 worldwide communication over the internet or intern networks, both by speaking and/or by text messages, delivering an outstanding voice quality feature by integrating advanced configurable digital processing features consisting of an auto gain controller, acoustic echo suppression, noise suppression, reverb suppression, and mute microphone.
Key features include making and receiving SIP based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider, VoIP conferencing with crystal clear sound, encryption for SIP account settings and secure web licensing, multi-user conference support, multi-line support, call hold, 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, Noise Concealment, and much more.
In conclusion, the SIP Phone DLL for .NET and ActiveX is a powerful and highly versatile VoIP SDK that makes developing SIP applications relatively easy. By taking advantage of its advanced features and customizable interface, developers can build powerful SIP and RTP compliant softphones that provide crystal clear voice quality and ease of use. We highly recommend trying out the SIP Phone DLL today!
The Conaito SIP Phone DLL cordially 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 worldwide communication over the internet or intern networks, both by speaking and/or by text messages, delivering an outstanding voice quality feature by integrating advanced configurable digital processing features consisting of an auto gain controller, acoustic echo suppression, noise suppression, reverb suppression, and mute microphone.
Key features include making and receiving SIP based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider, VoIP conferencing with crystal clear sound, encryption for SIP account settings and secure web licensing, multi-user conference support, multi-line support, call hold, 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, Noise Concealment, and much more.
In conclusion, the SIP Phone DLL for .NET and ActiveX is a powerful and highly versatile VoIP SDK that makes developing SIP applications relatively easy. By taking advantage of its advanced features and customizable interface, developers can build powerful SIP and RTP compliant softphones that provide crystal clear voice quality and ease of use. We highly recommend trying out the SIP Phone DLL 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