Step #3: Accurately estimate network resource utilization
Accurately estimate the network resource utilization for the proposed VOIP system. Unlike traditional land lines where users have the equivalent of dedicated switch ports, a VOIP system is usually placed on a shared medium. Converged networks are what everyone is trying to implement to save money. Be sure to consider the number of simultaneous calls that will occur. With that information in hand, you can easily calculate the aggregate bandwidth that will be needed to support all calls during peak calling periods.
By benchmarking the requirements for a single VOIP call, computing the aggregate call traffic and comparing these values with baselines of existing network activity, it will be simple to determine if VOIP will "fit" in your current network. If the numbers do not add up, you may need to consider architectural changes to your IT infrastructure.
Step #4: Select a codec with higher speech sampling rates
Select a codec with higher speech sampling rates. In general, the higher the speech sampling rate, the better the potential call quality (but at the expense of more bandwidth being consumed). Make tradeoffs carefully. For example, G.711 provides excellent quality. Data is delivered at 64K bps, and the codec imposes no compression delay.
Technically, G.711 delivers 8,000 bytes per second without compression so that full Nyquist-dictated samples are provided. However, if you are attempting to pack multiple phone calls over a narrow bandwidth circuit, you may need to accept slightly lower quality to use a codec that is less hungry for bandwidth.