Q3 2013 Release v1.3

Highlights 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

Q2 2013 Release v1.2

Highlights of the version 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

Q1 2013 Release v1.0

Highlights of the version 1.0

• Easily make and receive SIP (Session Initiation Protocol) based 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 (coming soon) 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)
• 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 a ready up own SIP VoIP and Instant Messaging network solution.
• P2P support for directly connectionsbetween 2 SIP clients without SIP Server
• Registration on SIP Server (SIP Registrar)
• Outbound Proxy Server support
• 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)
• Microphone and speaker volume access
• Mute microphone/speaker + level indicator
• Auto-answer
• DND (do not disturb)
• Adaptive Jitter buffer
• PLC (packet lost concealment)
• AGC (automatic 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
• Comes as ActiveX control and native DLL with .NET Interface for easy usage in .NET development
• Log file on/off setting
• Keep-alive packets for NAT/Firewall
• Fully-customizable user interface
• Microsoft Authenticode Certificate
• Fair 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, C++, VB 6.0 and Delphi
• For .NET framework as well and all development environments with native DLL, .NET or ActiveX support