One of the most common terms that come to mind when discussing migration to VoIP systems is network bandwidth. This term - often interchangeably used with network speed - refers to the data transfer rates supported by the network, expressed in either Kilobits per second (Kbps), Megabit per second (Mbps) or Gigabit per second (Gbps). Bandwidth plays an important part in networking, as it defines not only the speed in which data moves from one location to another, but also the overall capacity of the network connection.
Bandwidth can be likened to a pipeline supplying a building with water. The pipe has a maximum flow capacity defined by the water source and by the size of the pipe installed. Thus, there is no way that water flow can be increased beyond the line's maximum capacity. The same thing happens in a network: data transfer rate is limited by both the ISP (i.e., subscription), and the capacity of the cable/circuit.
Bandwidth in VoIP terms
Using the same water pipe analogy, let's say we are installing an indoor pool inside the building. Naturally, the existing water pipeline will not be adequate to supply water to the pool, since the capacity of the pipe was intended only for the original needs of the building. It is thus necessary to either redesign the existing pipeline or install a separate line to bring water for the indoor pool. This is exactly what happens when VoIP is added to an existing network - sometimes, it requires either increasing the bandwidth or perhaps even allotting a separate line for it.
VoIP, like the pool, can be an extremely voracious application. It requires a lot of bandwidth in the same way that the pool requires a lot of water. In both cases, we can make do with the existing setups in a worst case scenario, but that might entail sacrificing other users: water supply in lavatory faucets may be reduced to a trickle while we fill-up the pool, and downloading files or browsing webpages may become very slow when a VoIP call is underway.
Using more bandwidth
A regular PSTN call normally consumes just over 64 Kbps of bandwidth, so that a 1.5 Mbps dedicated circuit (called T1 or DS-1 circuit) could be used for 24 consecutive calls. VoIP, on the other hand, requires more bandwidth overhead due to the packetizing it must do, so the same call would consume more than 120 Kbps in uncompressed VoIP, and the same DS-1 circuit could only be used for 13 calls.
Fortunately, VoIP allows compression of the media (the voice part converted to digital form using VoIP codecs). The most basic compression available allows twice the number of uncompressed consecutive calls to be sent using the same DS-1 line.
Managing bandwidth
While it is not possible to determine accurately the bandwidth needed and actually used by each hardware or software in a VoIP network, a number of tools are available for measuring network throughput. For local networks, benchmarking tools like netperf and tccp are useful in measuring TCP and UDP performance between two workstations. There are also network speed/bandwidth tests available online, that are useful in assessing network throughput.
But even with these tools, precise measurement of bandwidth utilization is still not possible. Factors such as hardware configurations, software specifications and latency are dynamically changing all the time, making it impossible to gauge throughput accurately, particularly in an environment with a lot of VoIP activity.
Saiju is an IP communications and business software expert. He actively promotes great hosted business VoIP and hosted IP PBX solutions.
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