How to calculate necessary bandwidth for a VoIP network

When I teach SIP, H.248, Cisco, or Avaya training courses, I’m often asked how to calculate necessary bandwidth for a VoIP network. I’m actually headed out the door to teach a SIP Essentials class this week, and seeing as I am anticipating this question, I thought I might post some helpful resources on the subject.

Essentially, it’s a question of the type of payload (codec you’ve selected), the sample period (how many samples per packet), and then taking into consideration the IP/UDP/RTP header information (generally, 40 octets), and then transmission overhead (generally, Ethernet 18 octets). After calculating bandwidth required for one call, you can then figure out required bandwidth for the total number of simultaneous calls you’d like to support on your network. However this number may be further muddled by employing silence suppression, which may potentially reduce necessary bandwidth by up to 50%.

Cisco has a helpful document on the subject of Voice Over IP – Per Call Bandwidth (Document ID: 7934).

At the end of the day, its important to understand the theory of what is going on, but the easiest way to calculate necessary bandwidth is to use a nice VoIP calculator, like the Packetizer VoIP Bandwidth Calculator.