• 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 (coming soon) Codec)
• Open standards-based and interoperable with all of the major equipment vendors
• Multi-party voice conference support (Conference split and join, locally mixed conferences)
• Multi-line support (multiple simultaneous calls)
• Integrated STUN, TURN and ICE support

• 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 as well during a conference)
• Mute microphone/speaker
• 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
• Dynamically loadable codec support
• Comes as ActiveX control (Web demo with ready-up signed CAB included)
• Registration on SIP Server (SIP Registrar)
• 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), C++, 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 as well and all development environments with ActiveX support
• Very easy to incorporate
Rich call control feature set
• Multi-party voice conference support (Conference split and join, locally mixed conferences)
• Multi-line support (multiple simultaneous calls)
• 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.