The GIPS Developer Community was launched today by Global IP Sound to enable developers to build Voice over IP applications. According to GIPS, "The goal of the Community is to easily allow developers to utilize Global IP Sound’s ground-breaking technology to develop exciting applications and innovative ways of communicating. By making its software available to anyone, GIPS is helping to break down boundaries and encourage innovation and expression."
GIPS is making available as a free download, GIPS VoiceEngine Lite, which provides an easy to use, flexible API that includes voice processing capabilities such as GIPS patented jitter buffer/error concealment module NetEQ, GIPS codecs, and the tools to create VoIP applications. GIPS VoiceEngine Lite is a light version of GIPS VoiceEngine Standard.
The GIPS developer community website consists of the needed software libraries and components to develop VoIP applications. From this website you can access and use to following components:
- GIPS VoiceEngine Lite libraries (.dll and .h)
- GIPS VoiceEngine Lite demo client - with simple GUI example code
- SIP demo - An example implementation of a VoIP application utilizing a SIP source code from reSIProcate’s
- GIPS SoundWare demo (.exe) - Based on GIPS VoiceEngine Advanced with echo cancellation
In addition, you can also use Libjingle, a set of C++ components, provided by Google, to interoperate with Google Talk's peer-to-peer and voice calling capabilities. In addition to Libjingle, Google is providing the code to simplify the integration between Libjingle and GIPS VoiceEngine Lite. The GIPS VoiceEngine Lite library is limited with a time parameter setting (computer clock) that enables usage for approximately 3 months. This time limitation can be extended through additional free download from their community site.
Features for VoiceEngine Lite include:
- High level API
- Low latency and high packet loss robustness with GIPS NetEQ
- GIPS narrowband and wideband codecs, including iLBC, GIPS Enhanced G.711, iSAC, iPCM-wb
- Send DTMF
- Sound device selection
- Speech level indicator
- Microphone and speaker volume controls
- Compatible with any call setup protocol (e.g.SIP, H.323)
- RTP Handling or support for customer
- Specific media transport protocol
- RTCP and statistics
- SRTP
- Video synchronization
- Support for Windows 98, 2000, NT, ME, XP
