In the short term at least, companies will be migrating their existing private voice networks onto their IP data networks, so the trick will be able to work out in advance whether their IP network has the capacity to take the voice traffic, or if it does not, then to calculate how much additional bandwidth is required. I guess that the nearest thing I can think of to the Erlang in the data world is the TRIB (transfer rate of information bits), and is a measure of the effective quantity of information over a circuit vs. time!
However, I am not sure that this will help, because data traffic is not so time critical as voice. Some data traffic is more so than others, eg: real-time data such as that for air traffic control would need to be delivered ASAP whereas email could be left for hours without too much inconvenience. This is where traffic shaping comes in. You can hold back on the email and give the ATC high priority. Most data traffic models use some form of queuing technique to smooth out the traffic flows. I am not sure that you need to get into this as there are a lot of organisations that specialise in it already. What I would like is to be able to input the info. from my data traffic tools into your voice traffic tool. This could be to input the maximum data bandwidth required during the data busy hour and the maximum data bandwidth required during the voice busy hour. Then should be able to compute the bandwidth required for voice and data combined. BTW – admission control works OK when the network is only being used for voice. The problem is that to get maximum benefit most companies want a converged network. On some of the products I have worked on, they have been capable of routing voice traffic onto the PSTN if the IP network is unavailable, however if there is huge packet loss and/ or latency then they continue to connect calls to the IP network!