Speech codec in OCS 2007

One of the huge improvements we're making in Office Communications Server 2007 is in the area of speech quality and Quality of Experience in VoIP phonecalls.

We have a document available for download with an overview of our RTAudio advanced VoIP codec, which has the ability to dynamically adjust to network conditions such as latency, packet loss, jitter, etc; and because it is a variable-bitrate codec, calls will not drop out if the network is not "perfect".  This is a huge change from traditional Enterprise VoIP technologies, which require end-to-end management of the network.  Rather than adjusting the network to fit the call, we can adjust the call to fit the network.

So what does this mean?  Think better voice quality with less bandwidth, and the ability for remote/mobile users to connect to OCS and make voice/video calls across the Internet and from mobile networks, wireless hotspots, etc.

Download the document from https://www.microsoft.com/downloads/details.aspx?familyid=5d79b584-79c9-42a8-90c4-4ab3f03d19c4&displaylang=en