Give Your App a Voice in 30 Minutes
Powerful and highly versatile VoIP SIP SDK
The Conaito VoIP 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 softphone 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.
- Easy, familiar, event-driven call control ActiveX, DLL and .NET
It´s very easy to incorporate and provide you quick development using the .NET framework and all development environments with native DLL, .NET or ActiveX support.
- Industry leading SIP support
Conaito VoIP SIP Client SDK is based on IETF standards, RFC3261 compliant SIP stack, RFC 2833 out-of-band DTMF signaling, Integrated STUN, TURN and ICE support. more
- Advanced voice processing features
Comes out-of-the-box with AGC (automatic gain control), AES (acoustic echo cancellation) and Noise cancellation. more
- Fair royalty free licensing
We offer fair royalty free licensing, no hidden costs and no yearly/monthly fees like other competitors. more
- Rich call control feature set
Features like Multi-party conference, Multi-line support, SIP Instant Messaging, locally mixed conferences, Hold/Mute, Call Transfer, Call Forwarding and Rejection are standard features by Conaito! more
- Comprehensive configuration support
Highly configurable, such as: select media input/output devices on-the-fly, configurable Ports (RTP, SIP UDP, SIP TCP, STUN, TURN, ICE) and SIP Proxy etc.. more
- Crystal clear VoIP conferencing
even for both low and high-bandwidth users (G729, G723, G711 A-Law, G711 U-Law, Speex, Speex-wb, GSM6.10 and iLBC codec). 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
are able to interact with each other they must connect to
a SIP gateway or SIP based IP-Telephony service provider.
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 Conaito VoIP SIP Client SDK - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications and websites.