When considering building a VoIP network, no matter the scope or size, one of the first questions to ask is "How much bandwidth do I need?" Thorough capacity planning is crucial to ensure good voice quality during peak and off-peak hours. Part of capacity planning is determining just how much bandwidth is required for a voice call.

Generally, a VoIP call will use between 10 kbps and 110 kbps in one direction. The exact amount depends on a number of factors, namely:

• Codec type used and sample length (how often the samples are transmitted)
• IP header
• Transmission medium (i.e. DSL, ATM)
• Silence suppression/echo cancellation

Codec

The codec type determines the amount of bandwidth used for the actual voice information, and also defines the sample period (how often the samples are transmitted). For example, the uncompressed G.711 codec has a rate of 64,000 bps. It has a sampling rate of 20 ms, meaning it generates 50 frames per second. Each frame then contains 64,000/50= 1,280 bits or 160 octets.

IP header

The IP header is relatively constant among voice calls. It refers to the additional bandwidth needed for the various headers: IP, UDP, RTP. RTP adds 12 octets, UDP 8, and IP 20, for a total of 40 octets. Sticking with a sample period of 20 ms, the IP header will require an additional 16 kbps to whatever codec is used above. The main variation here is when the cRTP (compressed RTP) standard is used. Compressed RTP reduces the IP header to only 2 or 4 octets. It is geared toward mobile networks where bandwidth is at a premium.

Transmission medium

The transmission medium, like Ethernet and/or ATM, adds its own headers and checksums to the packet. For example, Ethernet adds another 38 octets. ATM will add even more.

Silence Suppression

When silence suppression is enabled, it can reduce the necessary bandwidth by as much as 50%. Silence suppression takes advantage of the inevitable gaps in speech during a call — e.g. when one party is talking, the other usually is silent.

Putting it all together

As a starting point, the G.711 codec itself requires 64 kbps. Adding in IP header overhead brings the total to 71.6 kbps. And if ATM over DSL is used as the transmission medium, the bandwidth jumps to 106 kbps for each direction of the call.

If you don’t want to go through the tedious calculation steps, you can find several bandwidth tools online. Simply input your data and the tool will calculate the number for you.

Check out: http://www.packetizer.com/voip/diagnostics/bandcalc.html

See also:

• Codecs
• Low-Bandwidth Solutions
• Asterisk: Open-Source PBX

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