conaito Technologies Introduces the SIP Phone DLL - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications - Release v24

Released on: February 28, 2008, 1:39 am

Press Release Author: conaito Technologies

Industry: Software

Press Release Summary: SIP Phone DLL - A powerful and highly versatile VoIP SDK for
SIP applications

Press Release Body: Software Product: SIP Phone DLL for .NET and ActiveX - Version 2.4
Author: conaito Technologies

SIP Phone DLL, ActiveX and .NET - A powerful and highly versatile VoIP SDK to
accelerate development of SIP applications.

SIP Phone 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

The SIP Phone 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)
* 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 SIP Phone DLL 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.

conaito Technologies

Web Site:

Contact Details: Address: T. Korodi, Chemnitz

Phone Number: 00493713179019

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