NORTH SOUTH UNIVERSITY
Compare Study of CODEC. Which is
suitable for Bangladesh (Low bandwidth)?
Dr. Mashiur Rahman
Date of Submission: 16.04.08
M. Raisul Islam # 063441556
With respect to voice over IP, a codec is an algorithm used to encode and decode the
voice conversation. Since voice and sound as we hear it is analogue, it needs to be
converted (or encoded) to a digital format suitable for transmission over the Internet.
Once at the other end, it needs to be decoded again so the other person can hear what you
are saying. There are a variety of different ways this encoding and decoding can be done
- many of which utilise compression in order to reduce the required bandwidth of the
conversation. A key thing to remember with VoIP, is that encoding, particularly when
heavy compression is used, takes time, which adds a delay to the conversation. Thus, the
holy grail is a codec which not only maintains good quality with compression, but is able
to do the encoding and decoding in a minimal amount of time.
It is important to keep in mind that different VoIP clients support different codecs, and
each VoIP provider will only support a subset of the codecs too. Generally, when a VoIP
call is established, you will need to use a codec that both parties and the provider support.
No need to worry though, this sort of negotiation is handled automatically, but knowing
the details will enable you to force or encourage certain codecs to be used. Understanding
codecs will also help you understand why some VoIP clients sound better than others,
and why voice quality with some providers, or through certain ISPs, are better than
1.2 What is CODEC?
"Codec" is a technical name for "compression/decompression". It also stands for
"compressor/decompressor" and "code/decode". Codecs are standard methods of
Compressing and decompressing data. All of these variations mean the same thing: a
codec is a computer program that both shrinks large movie files,data and makes them
playable on your computer. Codec programs are required for your media player to play
your downloaded music and movies.
1.3 Why do we need codecs?
Because video and music files are large, they become difficult to transfer across the
Internet quickly. To help speed up downloads, mathematical "codecs" were built to
encode ("shrink") a signal for transmission and then decode it for viewing or editing.
Without codecs, downloads would take three to five times longer than they do now.
1.4 Is there only One CODEC Need?
Sadly, there are hundreds of codecs being used on the Internet, and will need
combinations that specifically play files. There are codecs for audio and video
compression, for streaming media over the Internet, videoconferencing, playing mp3's,
speech, or screen capture. To make matters more confusing, some people who share their
files on the Net choose to use very obscure codecs to shrink their files. This makes it very
frustrating for users who download these files, but do not know which codecs to get to
play these files. If you are a regular downloader, you will probably need ten to twelve
codecs to play your music and movies.
Some common codec examples are MP3, WMA, RealVideo, RealAudio, DivX and
XviD. There are many other more obscure codecs.
There is no single best answer to this question. There are so many codec choices. The
easiest option is to download "codec packs". Codec packs are collections of codecs
gathered in single large files. There is much debate over whether it is necessary to get a
large group of codec files, but it certainly is the easiest and least-frustrating option for
new downloaders. Here are the codec packs we recommend at About.com:
2.1 Codec Comparison
The following table lists the various codecs used in voice over IP, and in particular SIP.
Many codecs come in a few varieties, and we have attempted to list all such version of
Sampling Nominal Payload
Codec Rate Bandwidth Size License Comments Pros Cons ?
(kHz) (kbps) (ms)
Not a very
DVI4 unknown unknown unknown common
to as u-law/a- Including
law: where a-
Open law is the >64kbps, thus
G.711 8 64 87.2 20 of speech at least
version and u- bandwidth in
Very low each direction
processing is required
16 48 unknown An ITU
G.722 16 56 unknown 30 standard
16 64 unknown
8 5.3 20.8 30 Very high
Often used by
whilst Requires a lot
G.723.1 Proprietry users for of processor
8 6.3 21.9 30 maintaining power.
8 16 unknown An improved
version of CPU
8 24 47.2 20
Open G.721 and
G.726 8 32 55.2 20 relatively low
Source G.723 (totally for level of
different from obtained.
8 40 unknown
G.728 unknown 16 31.5 standard
G.729 8 8 31.2 20 Patented standard required for
encoding as compression
used in GSM ratio.
phones Royalty free
GSM 8 13 unknown Proprietry
(though means it is
improved available in
version are many
often used hardware and
unknown 13.33 unknown 30 High
iLBC robustness to
unknown 15 unknown 20 use
Siren unknown unknown unknown and does not
appear to be
Speex 8 unknown unknown Open Uses variable
16 unknown unknown
Source bit rate to
32 unknown unknown
• Bandwidth values represent the amount of data in the payload of the IP packets.
• Bandwidth values indicate the bandwidth in each direction - not the sum of
upstream and downstream bandwidths.
• Bandwidth values assume continuous transmission of voice in both direction with
no silence suppression.
• The 'nominal bandwidth' column indicates the typical Ethernet bandwidth one can
expect the codec to use.
Codec Information Bandwidth Calculations
Codec Codec Codec Mean Voice Voice Packets Bandwidth
& Bit Sample Sample Opinion Payload Payload Per MP or
MP or Ethernet
Rate Size Interval Score Size Size Second FRF.12
(Kbps) (Bytes) (ms) (MOS) (Bytes) (ms) (PPS) (Kbps)
(64 10 ms 4.1 20 ms 50 82.8 Kbps 67.6 Kbps 87.2 Kbps
(8 10 ms 3.92 20 ms 50 26.8 Kbps 11.6 Kbps 31.2 Kbps
(6.3 30 ms 3.9 30 ms 34 18.9 Kbps 8.8 Kbps 21.9 Kbps
(5.3 30 ms 3.8 30 ms 34 17.9 Kbps 7.7 Kbps 20.8 Kbps
G.726 20 80
5 ms 3.85 20 ms 50 50.8 Kbps 35.6 Kbps 55.2 Kbps
(32 Bytes Bytes
(24 5 ms 20 ms 50 42.8 Kbps 27.6 Kbps 47.2 Kbps
(16 5 ms 3.61 30 ms 34 28.5 Kbps 18.4 Kbps 3
2.3 Explanation of Terms
Based on the codec, this is the number of bits per second that need to be
Codec Bit Rate transmitted to deliver a voice call. (codec bit rate = codec sample size / codec
Based on the codec, this is the number of bytes captured by the Digital Signal
Processor (DSP) at each codec sample interval. For example, the G.729 coder
Codec Sample operates on sample intervals of 10 ms, corresponding to 10 bytes (80 bits) per
sample at a bit rate of 8 Kbps. (codec bit rate = codec sample size / codec sample
This is the sample interval at which the codec operates. For example, the G.729
coder operates on sample intervals of 10 ms, corresponding to 10 bytes (80 bits)
Interval (ms) per sample at a bit rate of 8 Kbps. (codec bit rate = codec sample size / codec
MOS is a system of grading the voice quality of telephone connections. With
MOS, a wide range of listeners judge the quality of a voice sample on a scale of
one (bad) to five (excellent). The scores are averaged to provide the MOS for the
The voice payload size represents the number of bytes (or bits) that are filled into
a packet. The voice payload size must be a multiple of the codec sample size. For
Size (Bytes) example, G.729 packets can use 10, 20, 30, 40, 50, or 60 bytes of voice payload
The voice payload size can also be represented in terms of the codec samples. For
Voice Payload example, a G.729 voice payload size of 20 ms (two 10 ms codec samples)
represents a voice payload of 20 bytes [ (20 bytes * 8) / (20 ms) = 8 Kbps ]
PPS represents the number of packets that need to be transmitted every second in
order to deliver the codec bit rate. For example, for a G.729 call with voice
payload size per packet of 20 bytes (160 bits), 50 packets need to be transmitted
every second [50 pps = (8 Kbps) / (160 bits per packet) ]
2.4 Bandwidth Calculation Formulas
The following calculations are used:
• Total packet size = (L2 header: MP or FRF.12 or Ethernet) + (IP/UDP/RTP
header) + (voice payload size)
• PPS = (codec bit rate) / (voice payload size)
• Bandwidth = total packet size * PPS
2.5 Sample Calculation
For example, the required bandwidth for a G.729 call (8 Kbps codec bit rate) with cRTP,
MP and the default 20 bytes of voice payload is:
• Total packet size (bytes) = (MP header of 6 bytes) + ( compressed IP/UDP/RTP
header of 2 bytes) + (voice payload of 20 bytes) = 28 bytes
• Total packet size (bits) = (28 bytes) * 8 bits per byte = 224 bits
• PPS = (8 Kbps codec bit rate) / (160 bits) = 50 pps
Note: 160 bits = 20 bytes (default voice payload) * 8 bits per byte
• Bandwidth per call = voice packet size (224 bits) * 50 pps = 11.2 Kbps
2.6 Which Codec for which network?
A lot of work about the characterization of Codecs in terms of bandwidth utilisation, but
has there any work been done on the suitability of different codecs based on network
SLA parameters. For example, bandwidth on a broadband Internet connection is
essentially free up to ~512K per second, so bandwidth efficiency of one call in a truly
distributed switched environment is probably not at all interesting for an end user.
3.1 Which CODEC is suitable in BD (Low Bandwidth)?
At first there are some example are given below for low BW network then we will get a
clear idea of codecs in BD.
Although voice quality may suffer, VOIP can be used over a dial-up Internet or other low
Some fancy mathematics comes into play as VoIP streams are compressed below 16
kbps, but they come at the expense of call quality, with the following codecs offering
some insight into the tricks needed to deliver voice in low-bandwidth environments.
3.2 ITU G.726 & ITU G.727: 16, 24, 32 or 40 kbps
Uses an 8 kHz sampling frequency and employs Adaptive Differential Pulse Code
Modulation to encode PCM values as differences between the current and the previous
value. While the more bandwidth intensive codecs use 8-bit PCM sampling, G.726
reduces this to 2, 3, 4 or 5-bit. It is the standard codec used in DECT wireless phone
systems. G.726 replaced G.721 (32 kbps) and G.723 (24 and 40 kbps). G.727 offers the
same bit rates as G.726 but is optimised for Packet Circuit Multiplex Equipment.
3.3 ITU G.728: 16 kbps
Uses an 8 kHz sampling frequency and employs a Low-Delay version of Code Excited
Linear Prediction - an algorithm designed specifically for low bit rate speech
compression. It is favoured for some video, cellular and satellite applications.
3.4 Internet Low Bitrate Codec (iLBC): 13.33 or 15.2 kbps
Uses an 8 kHz sampling frequency and employs Block-Independent Linear-Predictive
Coding. iLBC is free to use but not open source
Some users have reported being able to use the Vonage service over dial-up, although
Vonage doesn't offically support this.
3.5 ITU G.729: 8 kbps
Uses an 8 kHz sampling frequency and employs Conjugate Structure Algebraic-Code
Excited Linear Prediction to squeeze a VoIP call into 8 kbps. G.729 is the codec of
choice for consumer VoIP providers running over the open internet, as Australia's engin
and MyNetFone. G.729A is compatible with G.729 but requires less computation, while
G.729B uses Discontinuous Transmission (DTX), Voice Activity Detection (VAD) and
Comfort Noise Generation (CNG) to reduce bandwidth usage during silence in a call.
3.6 ITU G.723.1: 5.3/6.3 kbps
Uses an 8 kHz sampling frequency and employs Algebraic Code Excited Linear
Prediction to achieve 5.3 kbps or Multipulse LPC with Maximum Likelihood
Quantization to achieve 6.3 kbps.
Hints for Low Bandwith VOIP
• OrbisTelecom offer G.711 and other low codecs with thier software.
• Choose your codec to minimize bandwidth — experiment those available in your
Some companies seem cater to the low bandwidth market - for example:
• PCPhoneline.com — H.323 and SIP products support G.723.1/G.711 codecs and
work with dialup connections as low as 19.2 kbps. Prices under $50
• IPmental — says requires minimum 10Kbps
• CuPhone — says requires minimum dialup
• Azatel — SIP ATA for dial-up Internet connections
3.7 Services officially supporting low bandwidth VOIP
• NetzeroVoice — works over dialup and connects through most firewalls
3.8 Features of Voice Pulse
* Inter-Asterisk Exchange 2 (IAX2)
* Session Initiation Protocol (SIP)
Supported User Agents
• Asterisk The Open Source PBX, AsteriskNOW, AA50, Swith
* Fonality PBX
* trixbox CE, SE, EE, CCE
* Cisco/Linksys SIP devices
* Aastra, Grandstream, Snom SIP devices
So we can say on depending above discussion that G.711ulaw, G.711alaw, GSM
ADPCM, ILBC codec are suitable for Bangladesh