VoIP Video SIP SDK 

Crystal clear and crisp video collaboration at the glance 

The Conaito VoIP Video SIP SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive video phone calls features in your software applications. It accelerates the development of SIP/RTP compliant soft phone with a fully-customizable user interface and brand name. 

It contains a high performance voip video conferencing client capable of delivering crystal clear sound and crisp video 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 and video 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 or MP3 and much more! 

VoIP Video SIP SDK has all the same functionality as Standard VoIP SIP Client SDK and additionally provides video stream support on the basis of H.263, H.263+, H.261, H.264, VP8 WebM and Theora codecs. It is possible to control and disable video when you do not need it. 

The VoIP Video SIP 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 version 1.3: 

• Recording voice conversation into mp3 (.mp3) file (without or ID3v2.1 tag supported)
• Playing mp3 (.mp3) files to the remote end (without or ID3v2.1 tag supported)
• VP8 WebM video codec support
• Codec selection and sorting
• Change User Agent (UA) with Identity property
• Webdemo sample with video support
• Video start/stop function including AutoStartSendingVideo
• Fixed VB6 example bug
• Fixed STUN event bug
• Fixed no Webcam bug
• Fixed Outbound Proxy bug
• Additional minor improvements as well file size optimization and performance of the SDK

New features of the version 1.1/1.2: 

• Supported selectable CIF (Common Intermediate Format)
• Multiple sip accounts registration support (unlimited)
• H264 video codec support
• Improved video conference
• Improved NAT Traversal for video conferences
• Max Bitrate for video stream (VideoMaxBitrate) - free configurable
• Many minor improvements

Key features of the VoIP Video SIP SDK: 

• Easily make and receive SIP (Session Initiation Protocol) based video phone calls through any SIP gateway or SIP compliant IP-Telephony service provider
• VoIP video conferencing with crystal clear sound and crisp video even for both low and high-bandwidth users
• Supported audio codec´s: G729, G723, G711 A-Law, G711 U-Law, Speex, Speex-wb, GSM6.10 and iLBC
• Supported video codec´s: H263, H263+, H261, H264, VP8 WebM and Theora 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)
• Multiple sip accounts registration support (unlimited)
• SIP Instant/Chat Messaging with send/receive controlling
• Integrated STUN, TURN and ICE NAT Traversal
• Comes with sample SIP Proxy Server to provide in bundle with the Conaito SIP Client 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
• Line Hold/Un-hold support
• Call forwarding and rejection
• Call transfer support
• Select media input/output devices - on-the-fly (as well during a conversation/conference)
• Microphone and Speaker volume with Mute (including 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 and mp3 (.wav and .mp3) file
• Playing PCM WAVE and mp3 (.wav and .mp3) files to the remote end
• Audio file memory cache
• Extended SIP URL functions
• Comes as ActiveX control (Webdemo with ready-up signed CAB included) and native DLL with .NET Interface (no ActiveX registration necessary) for easy usage in .NET development
• Registration on SIP Server (SIP Registrar)
• Log file on/off setting
• Keep-alive packets to NAT/firewall
• Fully-customizable user interface
• Production-ready Microsoft Authenticode Certificate certified
• Works with all kind of Internet connections
• Friendly to NAT and other firewalls
• Royalty free licensing
• No Yearly/Monthly fee
• Very easy to incorporate
• SDK comes in bundle with ActiveX control (Webdemo with ready-up signed CAB included), native DLL with .NET Interface for easy usage in .NET development (no ActiveX registration necessary)
• Fully sample applications for various programming languages such as sample source code for C#, VB.NET, C++, VB 6.0, Delphi and HTML/JavaScript (Webdemo) for ActiveX, DLL and .NET version
• For .NET framework as well and all development environments with native DLL, .NET or ActiveX support

Easy, familiar, event-driven call control ActiveX, DLL and .NET

• Easy to use; quick development
• Powerful .NET Interface for easy .NET development (no ActiveX registration necessary)
• Support for .NET framework as well and all development environments with native DLL, .NET or 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)
• SIP Instant messaging
• Locally mixed conferences
• Hold/Mute
• Call transfer and redirect
• Call forwarding and rejection

Industry leading SIP support

• RFC3261 compliant SIP stack
• RFC2833 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-video-enabled application, like e.g. a SIP video softphone, IVR solution, teaching tool, live support, voip video chat, meeting tool or any other type of application which requires users being able to talk, see each other and type messages to each other. 

For Conaito VoIP Video 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 Video 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-video-enable your application. 

We hope you enjoy the Conaito VoIP Video SIP SDK – A powerful and highly versatile VoIP Video SDK to accelerate development of SIP applications. 

Conaito Technologies 

http://www.conaito.com