28.07.2010 New! The
brand new Office2Flash SDK
is released! It comes now
with a lot of new features such as single SWF and multiple SWF converting, XML generation, Thumbnail creation, custom out filenames. The SDK is completly improved and provide now more stability and more performance.
Try it today! 22.07.2010 New! The
brand new VoIP Video EVO SDK
is released! It provide a high performance VoIP and Video conferencing solution and delivering crystal clear sound and video capturing even for both low and high-bandwidth users. Users are arranged in a tree-structure where each node is a room/ channel where users can talk, see Users, send instant messages and share files.
Completly improved! Try it today! 08.07.2010 Update! The
brand new VoIP SIP Client SDK
v1.3 is released! Now supported
are g729 and g723 codec, codec selection, play WAV
to phone lines, Failure codes support, Redirect call and much more. Many new samples for C#.NET, VB.NET, Delphi,
VB6 and fully new webdemo.
Completly improved! Try it today! 02.07.2010 New! The
brand new Flipdoo Publisher
v1.0 is released! Make you one Flipdoo Book in a few Minutes! This is the best way to create digital editions such as any type of online publications, magazines and catalogues. You want convert and share your Presentation or documents online?
The Flipdoo Publisher provides a great solution to convert PDF documents into powerfull Flash page turned flipping books. Try it today!
If you have any questions, need information, have suggestions or need help with conaito SDK, then please click here: conaito Support Team
VoIP SIP/RTP based dial and receive phone calls features in your software applications and websites?
Superior voice quality by advanced digital voice processing?
Then you really should decide on
conaito VoIP SIP Client SDK!
Our brand-new SIP Client 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 Client 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 and delivers
superior voice quality by integrating digital voice processing features including auto gain controller (AGC), acoustic echo suppression (AES) and noise suppression. It supports multiple lines, multi-party voice conference, call hold, call forwarding and transfer, DTMF, Packet Loss Concealment (PLC), adaptive jitter buffer, record and playing WAV and much more!
conaito VoIP SIP Client SDK is based on IETF standards (SIP, RTP, STUN, TURN, ICE etc.), so it should be compatible with other standard based products such as: SER, Sip EXpress, OpenSER and Asterisk.
New features of the conaito VoIP SIP Client SDK - version 1.3:
• g729 and g723 Codec´s support
• Multiple and single Codec selection support
• Failure codes support (get SIP Message Response Code, SIP Message Response Text)
• RTP/RTCP Port setting (for inbound RTP traffic)
• Reduce audio latency and audio latency settings (properties: MinPrefetchCount, MaxPrefetchCount, MaxRTPPackets)
• Media status (Events: OnLocalMediaStarted, OnLocalMediaStoped, OnRemoteMediaStarted, OnRemoteMediaStoped)
• Get used codec per line
• Custom Ringtone (play wav) support (property: RingtoneFile)
• Play wav to a selected phone line (methods: StartPlayingAtLine, StopPlayingAtLine)
• Redirect Call to other phone line
• Load and Save Configurations (methods: LoadConfiguration, StoreConfiguration)
• Complete new, re-written and updated samples with source code
• and much more!
Here is a list of the main features of the conaito VoIP SIP Client SDK:
• 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, Speex-wb, GSM6.10, iLBC, L16 and g729 & g723 Codec
• Open standards-based and interoperable with all of the major equipment vendors
• UDP and TCP support
• Multi-party voice conference support/ Conference split and join, locally mixed conferences
• Multi-line support (multiple simultaneous calls)
• SIP Instant/Chat Messaging with send/receive controlling
• Integrated STUN, TURN and ICE support • Comes with new sample SIP Proxy Server to provide in bundle with the conaito SIP Client ActiveX a ready up own SIP VoIP and Instant Messaging network solution.
• P2P support for directly connections between 2 SIP clients without SIP Server
• Outbound proxy server support
• Encrypted SIP account settings (encrypted SIP account settings in your webpage)
• Line Hold/Un-hold support
• Call forwarding and rejection
• Call transfer support
• Select media input/output devices
on-the-fly - also during a conversation/ conference)
• Mute microphone/speaker + level indicator
• Auto-answer
• DND (Do Not Disturb)
• Adaptive Jitter buffer
• PLC (Packet Lost Concealment)
• AGC (auto gain controller)
• AES (Acoustic echo cancellation or suppression)
• Noise cancellation or suppression
• DTMF tones support (generation/detection)
• Recording voice conversation into PCM WAVE (.wav) file
• Playing PCM WAVE (.wav) files to the remote end
• Audio file memory cache
• Extended SIP URL functions
• Dynamically loadable codec support (coming soon)
• Comes as ActiveX control (Web demo with ready-up signed CAB included)
• Registration on SIP Server (SIP Registrar)
• Log file on/off setting
• Microphone and Speaker Volume with Mute support
• Keep-alive packets to NAT/firewall
• Fully-customizable user interface
• Microsoft Authenticode Certificate
• Works with all kind of Internet connections
• Friendly to NAT and other firewalls
• Keep-alive packets to NAT/firewall
• Royalty free licensing
• No Yearly/Monthly fee
• Very easy to incorporate
• Fully sample applications for various programming languages such as sample source code for C#, VB.NET, JavaScript (Webdemo), VB 6.0 and Delphi
• For .NET framework as well and all development environments with ActiveX support
Easy, familiar, event-driven call control ActiveX
• Easy to use; quick development
• Support for .NET framework and all development environments with ActiveX support
• Very easy to incorporate
Rich call control feature set
• Multi-party voice conference support (Conference split/ join, locally mixed conferences)
• Multi-line support (multiple simultaneous calls)
• SIP Instant messaging
• Locally mixed conferences
• Hold/Mute
• Call transfer
• Call forwarding and rejection
Industry leading SIP support
• RFC3261 compliant SIP stack
• RFC 2833 out-of-band DTMF signaling
• Integrated STUN, TURN and ICE support
Comprehensive configuration support
• Select media input/output devices (on-the-fly as well during a conversation/conference)
• Configurable ports (RTP, SIP UDP, SIP TCP, STUN, TURN, ICE)
• SIP proxy
Advanced digital voice processing features
• AGC (auto gain controller)
• AES (Acoustic echo cancellation or suppression)
• Noise cancellation or suppression
… and much more!
Having the above features available makes it simple to develop any type of VoIP-enabled application, like e.g. a SIP softphone, IVR solution, teaching tool, live support, voip 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 Client 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 and website.
We hope you enjoy the new conaito VoIP SIP Client SDK – A powerful and highly versatile VoIP SDK to accelerate development of SIP applications and websites.
The contents of conaito VoIP SIP Client SDK and the supported development environments include:
The conaito VoIP SIP Client SDK provides the documentation, samples and related libraries you need to integrate with other applications or websites.
Conaito VoIP SIP Client SDK includes a ActiveX that can be used from any programming language like Visual Basic, .NET (VB and C#), C++, Visual Basic, Delphi, ASP, JSP, PHP, JavaScript, VBScript, etc. The conaito VoIP SIP Client SDK is designed to be used by Automation clients.
The contents of conaito VoIP SIP Client SDK and the supported development environments include all of the necessary software components for building systems based on conaito VoIP SIP Client SDK including documented operational software applications, examples (with source code), explanations as well as necessary service programs, libraries and components.
The supported development environments include:
• Visual Basic .NET
• Visual C++ .NET
• Visual C# .NET
• ASP.NET
• ASP, JSP, PHP
• JavaScript/HTML
• Visual Basic
• Visual C++
• Borland Delphi
• and all development environments with ActiveX support
System requirements:
Operating system: Windows XP, 2000, 2003, 2008, Vista, Windows 7
28.07.2010
VoIP SIP/RTP based dial and receive phone calls features in your software applications and websites?
Superior voice quality by advanced digital voice processing?
Then you really should decide on
conaito VoIP SIP Client SDK!
Our brand-new SIP Client 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.
• Comes with new sample SIP Proxy Server to provide in bundle with the conaito SIP Client ActiveX a ready up own SIP VoIP and Instant Messaging network solution.
