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VoIP SIP SDK for .NET and ActiveX 3.0
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!
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This file archive is 100% SAFE to download and install.
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* 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
* 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
* 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
* 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
* 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 g7
* NEW in v1.3: DTMF tones support (SIP INFO and RTP)
* NEW in v1.3: Recording voice conversation into PCM WAVE (.wav) file
* NEW in v1.3: Playing PCM WAVE (.wav) files to a voice conversation
* NEW in v1.3: Dynamically loadable codec´s support GSM6.10, g723 codec
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