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 v211

Released on: January 21, 2008, 3:31 am

Press Release Author: conaito Technologies

Industry: Software

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 2.1.1
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 v2.0! Multi-line (max 512 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.
* NEW in v2.0! Mute microphone/speaker for each line
* NEW in v2.0! DNS SRV resolution for SIP servers (RFC 3263)
* NEW in v2.0! Stereo codec (L16)
* NEW in v2.0! RTCP
* NEW in v2.0! Auto-answer
* NEW in v2.0! Do Not Disturb (DND)
* NEW in v2.0! Adaptive jitter buffer
* NEW in v2.0! Adaptive silence detection
* NEW in v2.0! PLC (Packet Lost Concealment)
* NEW in v2.0! DTMF tones support (with INFO - RFC 2976)
* NEW in v2.0! 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.

conaito Technologies

Web Site: http://www.conaito.com/voip_sip_sdk_ueberblick.asp

Contact Details: Address: T. Korodi, Chemnitz
Germany

Phone Number: 00493713179019
Email: agent@conaito.com

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