Voip Bandwidth Calculation

VoIP Bandwidth
Calculation
VoIP Bandwidth Calculation

Executive Summary
alculating 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.
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
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.
Latency and Packet Overhead

The choice of the number of frames per packet is a trade-off
between two characteristics: latency and packet overhead.
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.
Long sample periods produce high latency, which can affect

the perceived quality of the call. Long delays make interactive
conversations awkward, with the two parties often talking over
each other. Based on this fact alone, the shorter the sample
period, the better the perceived quality of the call. However, there
is a price to pay. The shorter the sample period, the smaller the
frames and the more significant the packet headers become. For
the smallest packets, well over half of the bandwidth used is taken
up by the packet headers; clearly an undesirable case. See figure
1 below:
Figure 1 - IP header forms a significant part of small Voice over IP packets
The Codec
IP
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
UDP
RTP
Data
IP Header
(Overhead)
Voice Payload
(Useful Data)
AI0106A
The IP Header
The term IP header is used to refer to the combined IP, UDP and
RTP information placed in the packet. The payload generated by
the codec is wrapped in successive layers of information in order
to deliver it to its destination. These layers are:
IP Internet Protocol
UDP User Datagram Protocol
RTP Real-time Transport Protocol
VoIP Bandwidth Calculation Page 3

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2005 Newport Networks Ltd
[email protected]
RTP is the first, or innermost, layer added. This is 12 octets. RTP

allows the samples to be reconstructed in the correct order and
provides a mechanism for measuring delay and jitter.
UDP adds 8 octets, and routes the data to the correct destination
port. It is a connectionless protocol and does not provide any
sequence information or guarantee of delivery.
Transmission of IP over other mediums will result in different

overhead calculations.
Voice over IP over Ethernet, Example 1:

G.711
Example 1: G.711 codec, 1 x 20 ms sample per packet - 95,200 kbps
Ethernet Inter-Frame Gap (12)
Ethernet Preamble (8)
Ethernet CRC (4)
Ethernet Header (14)
IP adds 20 octets, and is responsible for delivering the data to

the destination host. It is connectionless and does not guarantee
delivery or that packets will arrive in the same order they were
sent.
IP Header (40)
20 ms G.711 Payload (160)

AI0105A
In total, the IP/UDP/RTP headers add a fixed 40 octets to the

payload. With a sample period of 20 ms, the IP headers will
generate an additional fixed 16 kbps to whatever codec is being
used.
The payload for the G.711 codec and 20 ms sample period
calculated above is 160 octets, the IP header adds 40 octets.
This means 200 octets, or 1,600 bits sent 50 times a second
result 80,000 bits per second. This is the bandwidth needed to
transport the Voice over IP only, it does not take into account the
physical transmission medium.
There are other factors, which can reduce the overhead incurred
by the IP headers, such as compressed RTP (cRTP). This can be
implemented on point-to-point links and reduces the IP header from
40 to just 2 or 4 octets. Though this is not that common today, its
use will become more widespread with it being implemented with
3G mobile networks.
Codec G.711 64 kbps, 20 ms sample period
1 frames per packet (20 ms)
Standard IP headers
Ethernet transmission medium
One packet is sent every 20 ms, 50 packets per second. Payload

is 64,000 50 = 1,280 bits (160 octets). Fixed IP overhead 40
octets, fixed Ethernet overhead 38 octets. Total size 238 octets.
Bandwidth required is (160 + 40 + 38) x 50 x 8 = 95,200
kbps.
Voice over IP over Ethernet, Example 2:

G729a
Example 2: G.729a codec, 2 x 10 ms samples per packet - 39,200 kbps
Ethernet Preamble (8)
Ethernet Header (14)
The Transmission Medium
IP Header (40)
In order to travel through the IP network, the IP packet is wrapped

in another layer by the physical transmission medium. Most Voice
over IP transmissions will probably start their journey over Ethernet,
and parts of the core transmission network are also likely to be
Ethernet.
Ethernet has a minimum payload size of 46 octets. Carrying IP
packets with a fixed IP header of 40 means that the codec data must
be at least 6 octets typically not a problem. The Ethernet packet
starts with an 8 octet preamble followed by a header made up of
14 octets defining the source and destination MAC addresses and
the length. The payload is followed by a 4 octet CRC. Finally, the
packets must be separated by a minimum 12 octet gap. The result
is an additional Ethernet overhead of 38 octets.
Ethernet adds a further 38 octets to our 200 octets of G.711 codec
frame and IP header. Sent 50 times a second result 95,200 bits
per second, see example 1 below. This is the bandwidth needed
to transmit Voice over IP over Ethernet.
Ethernet CRC (4)

Ethernet Inter-Frame Gap (12)
20 ms G.729a Payload (20)

AI0107a
Codec G.729a 8 kbps, 10 ms sample period
2 frames per packet (20 ms)
Standard IP headers
Ethernet transmission medium
One packet is sent every 20 ms, 50 packets per second. Payload

is 8,000 50 = 160 bits (20 octets). Fixed IP overhead 40
octets, fixed Ethernet overhead 38 octets. Total size 98 octets.
Bandwidth required is (20 + 40 + 38) x 50 x 8 = 39,200 kbps.
Silence Suppression
Certain codecs support silence suppression. Voice Activity
Detection (VAD) suppresses the transmission of data during silence
periods. As only one person normally speaks at a time, this can
reduce the demand for bandwidth by as much as 50 percent. The
receiving codec will normally generate comfort noise during the
silence periods.

[email protected]
Additional Codec Data
Glossary
ACELP
Algebraic CELP (codec)
ADPCM
Adaptive Differential Pulse Code Modulation

(codec)
AMR
Adaptive Multi-Rate (GSM2 codec)
AMR-WB
Adaptive Multi-Rate Wideband (3GPP codec

G.722.2)
CELP
Code Excited Linear Predictive (codec)
Codec
Coder/Decoder
Summary
CRC
Cyclic Redundancy Code
Although there are many factors that influence the amount of

bandwidth required to transmit a voice call over an IP network,
by approaching the problem one element at a time the final
calculation becomes relatively simple. Other factors may influence
the actual bandwidth used, such as RTP header compression,
silence suppression and other techniques still under development.
cRTP
Compressed RTP (RFC 2508)
CS-CELP
Conjugate-Structure CELP
IP
Internet Protocol
LD-CELP
Low Delay CELP (codec)
MP-MLQ
Multi-Pulse Maximum Likelihood Quantization
Octet
A sequence of 8 bits
PCM
Pulse Code Modulation (codec)
RFC
Request For Comment
RTP
Real-time Transport Protocol
SB-ADPCM
Sub-Band ADPCM
UDP
User Datagram Protocol
VAD
Voice Activity Detection
Codec
G.711 (PCM)
G.723.1A (ACELP)
G.723.1A (MP-MLQ)
G.726 (ADPCM)
G.728 (LD-CELP)
G.729A (CS-CELP)
AMR (ACELP)
AMR (ACELP)
AMR (ACELP)
AMR-WB/G.722.2
(ACELP)
Bandwidth
64 kbps
5.3 kbps
6.4 kbps
32 kbps
16 kbps
8 kbps
4.75 kbps
7.4 kbps
12.2 kbps
6.6 kbps
Sample
period
20 ms
30 ms
30 ms
20 ms
2.5 ms
10 ms
20 ms
20 ms
20 ms
20 ms
Frame
size
160
20
24
80
5
10
12
19
31
17
Frames/
packet
1
1
1
1
4
2
1
1
1
1
Ethernet
Bandwidth
95.2 kbps
26.1 kbps
27.2 kbps
63.2 kbps
78.4 kbps
39.2 kbps
36.0 kbps
38.8 kbps
43.6 kbps
38.0 kbps

[email protected]
[email protected]
The Newport Networks logo is a registered trademark of Newport Networks Ltd. MediaProxy and SignallingProxy are trademarks of Newport Networks Ltd. 2005 Newport Networks Limited. All rights reserved. Whilst every effort has been made to
ensure that the information included in this publication is accurate at the time of going to press, Newport Networks Ltd assumes no responsibility for the accuracy of the information. Newport Networks Ltd reserves the right to change their specifications at any
time without prior notice. Some features described in this document may be planned for future releases and may not be available in the current product. Newport Networks Ltd reserves the right to modify its product development schedule without notice and
does not guarantee that any feature or product mentioned in this document will be produced or produced in the form described.
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VoIP Bandwidth

VoIP Bandwidth Calculation

alculating how much bandwidth a Voice over IP

Codec (coder/decoder) and sample period

Frames and Packets

Latency and Packet Overhead

Long sample periods produce high latency, which can affect

The conversion of the analogue waveform to a digital form is carried

The number of bits produced per second

The sample period - this defines how often the samples

UDP User Datagram Protocol

RTP Real-time Transport Protocol

VoIP Bandwidth Calculation Page 3

2005 Newport Networks Ltd

RTP is the first, or innermost, layer added. This is 12 octets. RTP

Transmission of IP over other mediums will result in different

Voice over IP over Ethernet, Example 1:

Ethernet Preamble (8)

Ethernet CRC (4)

Ethernet Header (14)

IP adds 20 octets, and is responsible for delivering the data to

20 ms G.711 Payload (160)

In total, the IP/UDP/RTP headers add a fixed 40 octets to the

Codec G.711 64 kbps, 20 ms sample period

1 frames per packet (20 ms)

Ethernet transmission medium

One packet is sent every 20 ms, 50 packets per second. Payload

Voice over IP over Ethernet, Example 2:

The Transmission Medium

In order to travel through the IP network, the IP packet is wrapped

Ethernet CRC (4)

20 ms G.729a Payload (20)

Codec G.729a 8 kbps, 10 ms sample period

2 frames per packet (20 ms)

Ethernet transmission medium

One packet is sent every 20 ms, 50 packets per second. Payload

VoIP Bandwidth Calculation Page 4

2005 Newport Networks Ltd

Additional Codec Data

Algebraic CELP (codec)

Adaptive Differential Pulse Code Modulation

Adaptive Multi-Rate (GSM2 codec)

Adaptive Multi-Rate Wideband (3GPP codec

Code Excited Linear Predictive (codec)

Cyclic Redundancy Code

Although there are many factors that influence the amount of

Compressed RTP (RFC 2508)

Low Delay CELP (codec)

Multi-Pulse Maximum Likelihood Quantization

Pulse Code Modulation (codec)

Request For Comment

Real-time Transport Protocol

User Datagram Protocol

Voice Activity Detection

VoIP Bandwidth Calculation Page 5

2005 Newport Networks Ltd

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