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Voice bandwidth traffic is calculated in Erlangs. Erlangs represent number of calls per hour*duration of calls / 60 minutes. To transfer that to VoIP depends on the codec and compression scheme used. http://www.erlang.com/calculator/lipb/ has a useful calculator that can assist with the calculations. I will caution, however, that if you do not utilize one of the newer routers that work with the QoS bit or do not have a traffic shaper in use, you will want to add more than the standard 1/3 growth factor to the equation.
FEEDBACK TO THIS Q&A
- Where you say, "To transfer that to VoIP depends on the codec and compression scheme used" - let's assume 120 circuit switched channels are needed, and let's say that the codec is a G.711, can you give an example of how to apply the circuit switched configuration to the specifications of the codec to determine the corresponding VoIP bandwidth?
—Jamie
Response from Carrie: A G.711 transfers 64k uncompressed per channel. Depending on your sampling, which for this example I assumed to be 20 milliseconds (which is pretty standard), you would need 9600 kbps to transmit the traffic.
If you have a phone switch, you should look up your BHT number (busy hour traffic). This figure will let you know what is used when your phone system is at its busiest and allow you to figure your erlangs. The figure may change based on that number. There are several calculators available at www.erlang.com.
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