Calculating
the bandwidth used for VoIP may seem like a daunting task, however it is rather
simple, especially after understanding simple some principles.
VoIP calls
consist of two main parts:
·
The call management section is the part which
gets the call going. It signals call initiation, ringing, disconnect, and other
communications performed between the two endpoints in order to get the call
going.
·
The second part is the audio, which is
transmitted using RTP. Since the bandwidth consumed by SIP is insignificant, we
shall focus on calculating the bandwidth consumed by audio in the rest of this
article.
Since raw
audio can be rather large, it needs to be encoded before it is sent on the
network. This is done using a codec. Different codecs produce different audio
quality, consume different levels of bandwidth, and some are more CPU intensive
than others. Thus it is important that you select the right codec for your
application.
Before we
delve into the differences of the common codecs, let’s introduce another
principle which would allow us to accurately calculate the bandwidth utilised.
When you need to send data over the network, the data needs to be packaged. The
“packaging” contains information which allows the data to be sent to the
destination and to be rebuilt correctly. As you might imagine, the packaging
does not come free – it adds to the bandwidth consumption.
There are
different network packaging layers (required by the 7 layer OSI model). The
encoded audio needs to be packaged into RTP packets. In turn, the RTP packets
need to be packaged into UDP packets, which then need to be packaged into IP
packets. Ethernet is the most common type of network, and this requires another
wrapper.
We shall
refer to these packages collectively as overhead. Irrelevant of the codec used,
the overhead introduced in the packet is fixed. Below is the overhead
introduced by each overhead item:
·
RTP – 4.8 kbps
·
UDP – 3.2 kbps
·
IP – 8 kbps
·
Ethernet (not using QOS) – 15.2 kbps
The total
overhead is 31.2 kbps.
Codecs
Now that
we understand the basics, let’s proceed with explaining the differences between
the common codecs which can be used to encapsulate the Audio in a VoIP call.
The following table shows the audio quality expected, the CPU resources required
to encode and decode the audio, the base size of the audio packets, and total
bandwidth consumption after taking into consideration the overhead.
Codec
|
Audio Quality
|
CPU Resources
|
Base Size
|
Total Size (Base + Overhead)
|
G711
|
Good
|
Very Low
|
64 kbps
|
95.2 kbps
|
G722
|
Very Good
|
Low
|
64 kbps
|
95.2 kbps
|
GSM
|
Acceptable
|
Average
|
13 kbps
|
44.2 kbps
|
G729
|
Average
|
High
|
8 kbps
|
39.2 kbps
|
Note that
the above bandwidth consumption is in kilobits per second. You will need to
divide by 8 in order to get the equivalent in kilobytes per second. Using the
above data, we can come up with the following stats:
Codec
|
Kilobits per second
|
Kilobytes per second
|
Kilobytes per minute
|
Megabytes per hour
|
G711
|
95.2
|
11.9
|
714
|
41.8
|
G722
|
95.2
|
11.9
|
714
|
41.8
|
GSM
|
44.2
|
5.525
|
331.5
|
19.4
|
G729
|
39.2
|
4.9
|
294
|
17.2
|
Here are
some notes and suggestions for the application of specific codecs:
·
The above are stats for one audio stream. A VoIP
call will use one audio stream for each leg (or endpoint). Thus a call between
two parties will use double the bandwidth shown above.
·
G729 is the codec which consumes the least
bandwidth and has a relatively good audio quality. However that comes with two
drawbacks:
1. Its efficiency comes
at a cost, CPU usage. It is more CPU intensive to encode audio in such a low
size while maintaining the quality.
2. G729 is a proprietary
codec. Because of this, the number of simultaneous G729 channels cannot exceed
half your VoIP phone depending on the number of participants and type of calls.
·
Because of this, G729 should only be used when
really required, such as for external calls to VoIP Providers, calls across a
bridge, or for remote extensions (basically all calls being done over the
internet). You can configure the PBX to fall back to GSM if G729 calls cannot
be made.
·
Although G711 and G722 consume over twice as much
bandwidth as the other codecs, most Local Area Networks are able to handle this
bandwidth. Using the above table, a 1 hour call using G711 is equivalent to
transferring a 41.8 MB file. If that causes an issue, you should consider
upgrading your network.
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