You can determine how much voip bandwidth to set aside for voice traffic using simple math. However, in a converged voice and data network, you have to make decisions on how much voip bandwidth to give each service. These decisions are based on careful consideration of your priorities and the available voip bandwidth you can afford. If you allocate too little voip bandwidth for voice service, there might be unacceptable quality issues.
Another consideration is that voice services are less tolerant to voip bandwidth depletion than that of Internet traffic. Therefore, voip bandwidth for voice services and associated signaling must take a priority over that of best-effort Internet traffic.
If a network were to use the same prevailing encoding (CODEC) scheme as the current PSTN system, voip bandwidth requirements for Voip networks would tend to be larger than that of a circuit-switched voice network of similar capacity. The reason is the overhead in the protocols used to deliver the voice service.
Typically, you would need speeds of OC-12c/STM-4 and higher to support thousands of call sessions. However, Voip networks that employ compression and silence suppression could actually use less voip bandwidth than a similar circuit-switched network. The reason is because of the greater granularity in voip bandwidth usage that a packet-based network has in comparison to a fixed, channel size TDM network.
Allocations of network voip bandwidth are based on projected numbers of calls at peak hours. Any over-subscription of voice voip bandwidth can cause a reduction in voice quality. Also, you must set aside adequate voip bandwidth for signaling to ensure that calls are complete and to reduce service interruptions. The formula for calculating total voip bandwidth needed for voice traffic is relatively straightforward. The formula to calculate RTP bearer voice voip bandwidth usage for a given number of phone calls is as follows:
Bits per sec = packet creation rates per sec x packet size x number of calls x 8 bits per sec
Where samples per sec = 1,000 ms / packet creation rate
Example: 2,000 full-duplex G.711 encoded voice channels that have a packet creation
Rate of 20 ms, with a packet size of 200 bytes (40 byte IP header + 160 byte payload)
50 samples per second = 1,000 ms / 20 ms
160 Mbps = 50 x 200 x 2,000 x 8
Note that this number is a raw measure of Voice over IP traffic and does not take in account the overhead used by the transporting media (links between the routers) and data-link layer protocols. Add this raw IP value to that of the overhead to determine the link speeds needed to support this number of calls. Note this value represents only the bearer (voice) content. Signaling voip bandwidth requirements vary depending on the rate at which the calls are generated and signaling protocol used. If a large number of calls are initiated in a relatively short period, the peak voip bandwidth needs for the signaling could be quite high. A general guideline for the maximum voip bandwidth requirement that an IP signaling protocol needs is roughly three percent of all bearer traffic. Using the previous example, signaling voip bandwidth requirements if all 2,000 calls were initiated in one second would be approximately 4.8 Mbps (3 percent of 160-megabits).
With the calculation of bearer and signaling, the total voip bandwidth needed to support two thousand G.711 encoded calls would be an approximate maximum of 164.8 MB. This voip bandwidth requirement is a theoretical maximum for this specific case. If the parameters change, such as call initiation rate, voice encoding method, packet creation rate, employment of compression, and silence suppression, the voip bandwidth requirements would change as well. With large Voice over IP implementations requiring sizable voip bandwidth, it becomes imperative that the IP network delivers the needed service at predictably high performance.