An overview graphic of Microsoft's RTAudio codec.

You’ve probably heard a bit in the past few days about how Microsoft is increasing their presence in the VoIP biz. (By the way, since VoIP is clearly and acronym, why don’t we get to capitalize the ‘o’?) (And lets stop this tendency to pronounce it as a word “voip”, it’s V-O-I-P folks.) So maybe it’s not related, but coincidental with this news is the publication of a white paper on their RT Audio Codec.

This codec is clearly designed to operate in a network (as opposed to circuit switched) environment, with lots of provisions for latency, variable bandwidth, FEC, and so on. I have a feeling that if you are making calls across Microsoft’s new phone system you’re going to be using this guy to transport your voice. Of course, when you go out to the PSTN, it’s going to require dropping down to G.711 or some other standard - which always galls Microsoft, because in some cases it’s going to mean paying royalties.

This suband coder is not just Microsoft’s, you’re free to fork out $US 50K to get a development kit and a use license for your product. Which probably means you need it if you’re building a headset or handset that connects to the new, non-existent Microsoft PBX.