conaito Technologies Introduces the VoIP SIP SDK with DLL, ActiveX and NET - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications - Release v24
Released on: February 21, 2008, 1:27 am
Press Release Author: conaito Technologies
Press Release Summary: VoIP SIP SDK with DLL and ActiveX - A highly versatile SDK for SIP applications
Press Release Body: Software Product: VoIP SIP SDK for .NET and ActiveX - Version 2.4 Author: conaito Technologies
VoIP SIP SDK with DLL, ActiveX and .NET - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.
VoIP SIP SDK with DLL, ActiveX and .NET for VoIP conferencing 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 VoIP SIP SDK with DLL and ActiveX 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 microphone/speaker for each line.
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) * NEW in v2.4! Encrypt SIP account setting * NEW in v2.3! Secure Weblicensing (protect your license in websites) * NEW in v2.2! Multi-User conference support * NEW in v2.0! Multi-line (simultaneous calls) support (Multiple Concurrent calls) * NEW in v2.0! Call Hold support * NEW in v2.0! Call Transfer support * NEW in v2.0! Instant text messaging (MIME) support and typing indication. * Mute microphone/speaker for each line * DNS SRV resolution for SIP servers (RFC 3263) * Stereo codec (L16) * RTCP * Auto-answer * Do Not Disturb (DND) * Adaptive jitter buffer * Adaptive silence detection * PLC (Packet Lost Concealment) * DTMF tones support (with INFO - RFC 2976) * Recording voice conversation into PCM WAVE (.wav) file * Advanced configurable digital voice processing features
Having the above features available makes it simple to develop any type of VoIP-enabled application, like e.g. a SIP softphone, teaching tool, live support, chat, meeting tool or any other type of application which requires users being able to talk and type messages to each other.
For conaito VoIP SIP clients to be able to interact with each other they must connect to a SIP gateway or SIP based IP-Telephony service provider.
Just relax! Please, don\'t hesitate trying our VoIP SIP SDK at once and get yourself, as well as your customers, the exciting experience of easy, fast and high quality standard applications which VoIP-enable your application.
Web Site: http://www.conaito.com/voip_sip_sdk_ueberblick.asp
Contact Details: Address: T. Korodi, Chemnitz Germany