I'm working with a radio company partner Icom, and we have purchased this SDK/API for integrating into there communication architecture. Using the SDK we can control messaging, radio id, and audio in/out.
What i'm interesting in is developing a custom audio gateway plugin that would allow audio to flow in and out to SIP enabled topologies (like asterisk/Etherstack and many others) . We have the API for this.
Normally I ask them to work on a transcoder, but the audio codec ICOM uses is proprietary and they only licenced us to use a hardware decoder/encoder (API's provided). The AUDIO IN/OUT and Push-To-Talk and Incoming Voice Call is exposed in the API layer, which may allow use to code to an open SIP stack.
I have contacted Etherstack and they love the IDEA, although there server application runs on LINUX, which is no problem running under a VM, they can licence us to support SIP (Direct RTP streams, G711 codecs).
Note the current development will be in windows, but possibly moved into windows embedded later.