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White Paper - VoIP Bandwidth Calculation
Executive Summary
Calculating how much bandwidth a Voice over IP call occupies can feel a bit like trying to answer the question; "How elastic is a piece of string?" However, armed with a basic understanding of the parts that make up the whole, the question becomes easier to understand. This White Paper examines the process that turns voice into Voice over IP. Also see the on-line VoIP Bandwidth Calculator.
Overview
The amount of bandwidth required to carry voice over an IP network is dependent upon a number of factors. Among the most important are:
- Codec (coder/decoder) and sample period
- IP header
- Transmission medium
- Silence suppression
The codec determines the actual amount of bandwidth that the voice data will occupy. It also determines the rate at which the voice is sampled. The IP/UDP/RTP header can generally be thought of as a fixed overhead of 40 octets per packet, though on point-to-point links RTP header compression can reduce this to 2 to 4 octets (RFC 2508). The transmission medium, such as Ethernet, will add its own headers, checksums and spacers to the packet. Finally, some codecs employ silence suppression, which can reduce the required bandwidth by as much as 50 percent.
The Codec
The conversion of the analogue waveform to a digital form is carried out by a codec. The codec samples the waveform at regular intervals and generates a value for each sample. These samples are typically taken 8,000 times a second. These individual values are accumulated for a fixed period to create a frame of data. A sample period of 20 ms is common. Some codecs use longer sample periods, such as 30 ms employed by G.723.1. Others use shorter periods, such as 10 ms employed by G.729a.
The important characteristics of the codec are:
- The number of bits produced per second
- The sample period - this defines how often the samples are transmitted
Together, these give us the size of the frame. For example, take a G.711 codec sampling at 20 ms. This generates 50 frames of data per second. G.711 transmits 64,000 bits per second so each frame will contain 64,000 ÷ 50 = 1,280 bits or 160 octets.
Frames and Packets
Many IP phones simply place one frame of data in each packet. However, some place more than one frame in each packet. For example, the G.729a codec works with a 10 ms sample period and produces a very small frame (10 bytes). It is more efficient to place two frames in each packet. This decreases the packet transmission overhead without increasing the latency excessively.
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IPsec and VoIP
