VOIP, Codecs, Quality - still confused

[Guest guest]

There was a mention of Codecs when using VOIP telephone

systems – and how the codec affects the VOIP sound quality.

Since I am thinking of moving to a virtual pbx service (like

vocalocity) and I hadn’t heard of this issue before, I thought I would

check it out.

Having read over the information – I’m still not

sure I understand which Codec is ideal and which works better for a medical

office.

Anyway…Here is what Dr. Google says…

Codecs

Codecs

are used to convert an analog voice signal to digitally encoded version. Codecs

vary in the sound quality, the bandwidth required, the computational

requirements, etc.

Each service, program, phone, gateway, etc typically supports several different

codecs, and when talking to each other, negotiate which codec they will use.

=============================

http://voip.about.com/od/voipbasics/a/voipcodecs.htm

There are many codecs

for audio, video, fax and text. Below is a list of the most common codecs for

VoIP. As a user, you may think that you have little to do with what these are,

but it is always good to know a minimum about these, since you might have to

make decisions one day relating codecs concerning VoIP in your business; or at

least might one day understand some words in the Greek VoIP people speak! I

won’t drag you into all the technicalities of codecs, but will just

mention them.

If you are a techie and want to know more about each one of these codecs in

detail, have a

look there.

Common VoIP Codecs

Codec

Bandwidth/kbps

Comments

G.711

64

Delivers precise speech transmission. Very low processor

requirements. Needs at least 128 kbps for two-way.

G.722

48/56/64

Adapts to varying compressions and bandwidth is conserved

with network congestion.

G.723.1

5.3/6.3

High compression with high quality audio. Can use with

dial-up. Lot of processor power.

G.726

16/24/32/40

An improved version of G.721 and G.723 (different from

G.723.1)

G.729

8

Excellent bandwidth utilization. Error tolerant. License

required.

GSM

13

High compression ratio. Free and available in many

hardware and software platforms. Same encoding is used in GSM cellphones

(improved versions are often used nowadays).

iLBC

15

Robust to packet loss. Free

Speex

2.15 / 44

Minimizes bandwidth usage by using variable bit rate.

===========================

http://communication.howstuffworks.com/ip-telephony6.htm

VoIP: Codecs

VoIP software processes and routes the

calls.

A codec, which stands for coder-decoder, converts an audio

signal into compressed digital form for transmission and then back into an

uncompressed audio signal for replay. It's the essence of VoIP.

Codecs accomplish the conversion by sampling the audio

signal several thousand times per second. For instance, a G.711 codec samples the audio at 64,000 times a second. It

converts each tiny sample into digitized data and compresses it for

transmission. When the 64,000 samples are reassembled, the pieces of audio

missing between each sample are so small that to the human ear, it sounds like

one continuous second of audio signal. There are different sampling rates in

VoIP depending on the codec being used:

64,000 times per second

32,000 times per second

8,000 times per second

A G.729A codec has a sampling rate of 8,000 times per second

and is the most commonly used codec in VoIP.

Codecs use advanced algorithms to help sample, sort, compress and packetize

audio data. The CS-ACELP algorithm (CS-ACELP =

conjugate-structure algebraic-code-excited linear prediction) is one of the

most prevalent algorithms in VoIP. CS-ACELP organizes and streamlines the

available bandwidth. Annex B is an aspect of CS-ACELP that

creates the transmission rule, which basically states " if no one is

talking, don't send any data. " The efficiency created by this rule is one

of the greatest ways in which packet switching is superior to circuit

switching. It's Annex B in the CS-ACELP algorithm that's responsible for that

aspect of the VoIP call.

The codec works with the algorithm to convert and sort everything out, but

it's not any good without knowing where to send the data. In VoIP, that task is

handled by soft switches.

==================================

http://www.ozvoip.com/codecs.php

Introduction to Codecs

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.

These pages attempt to demistify codecs and give a brief overview of the

different codecs and when they are used. 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 others.

If you would like to read up more about codecs with respect to VoIP, the

following links may be of interest:

Codecs page the VoIP Wiki

Find out which hardware supports

which codecs.

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 each codec. If you would like to

voice your opinion about a particular codec, or discuss the merits of one over

another, feel free to do so in our voice over IP forums.

Codec

Sampling

Rate (kHz)

Bandwidth

(kbps)

Nominal Bandwidth

(kbps)

Payload

Size

(ms)

License

Comments

Pros

Cons

?

DVI4

unknown

unknown

unknown

Not a very common codec.

G.711

8

64

87.2

20

Open Source

G.711u/a often refered to as

u-law/a-law: where a-law is the European version and u-law the US/Japanese

version

Designed to deliver precise transmission of

speech

Very low processing overheads

Including overheads, uses

>64kbps, thus at least 128kbps bandwidth in each direction is required

G.722

16

48

unknown

Open Source

An ITU standard codec.

16

56

unknown

30

16

64

unknown

G.723.1

8

5.3

20.8

30

Proprietry

Often used by dialup VoIP

users for optimal quality.

Very high compression whilst

maintaining high quality audio.

Requires a lot of processor

power.

8

6.3

21.9

30

G.726

8

16

unknown

Open Source

An improved version of G.721

and G.723 (totally different from G.723.1)

CPU overhead is relatively

low for level of compression obtained.

8

24

47.2

20

8

32

55.2

20

8

40

unknown

G.728

unknown

16

31.5

Open Source

An ITU standard codec.

G.729

8

8

31.2

20

Patented

An ITU standard codec.

Excellent bandwidth utilisation for toll

quality speech

Performs well under random bit errors

License

required for use

GSM

8

13

unknown

Proprietry

Same encoding as used in GSM

mobile phones (though improved version are often used nowadays).

Relatively high compression ratio.

Royalty free means it is available in many

hardware and software platforms.

iLBC

unknown

13.33

unknown

30

Free to use

High robustness to packet

loss

unknown

15

unknown

20

Siren

unknown

unknown

unknown

Not much known about this

codec, and does not appear to be commonly supported.

Speex

8

unknown

unknown

Open Source

Uses variable bit rate to

minimise bandwidth usage

16

unknown

unknown

32

unknown

unknown

Notes

The information provided here is for information purposes only, if you find

errors or ommissions, please report them in the relevant discussion forum.

Bandwidth

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.

Sampling Rate

The sampling rate is the rate at which the analogue audio signal is sampled.

Nyquist's Theorem states that in order to record a certain frequency, sampling

must occur at at least twice that frequency. Thus, the higher the sampling

rate, the greater the frequency range in the encoded audio stream. The human

ear is capable of hearing from about 20Hz to about 20,000Hz. Typically, speech

is around 100-4,000Hz. Thus, a sampling rate of at least 8kHz is required to

accurately encode the human voice. Greater sampling rates will capture higher

frequencies (this is useful, for example, if you are playing music down the

phone), but will also increase bandwidth as there are more samples to encode

and transmit.

Payload Size

The size of the payload of each encoded voice packet influences two things:

lag and bandwidth. Every encoded packet that is sent incurs fixed bandwidth

overheads (due to IP and other headers added to the data in the network). Thus,

larger payloads incur a proportionately smaller overhead, thus reducing the

nominal bandwidth utilisation. However, by using larger payloads, more audio

(ie., a longer period of time) is required to construct a single packet, which

in turn increases the amount of time it takes for even the beginning of the

packet to reach the other end and be decoded, thus increasing the lag in the

conversation. This is a typical trade-off in VoIP. Most codecs use payload

sizes of 10-40ms.

[Guest guest]

I learned about these when trying to figure out why my Windows Mobile would not play the voicemails attached to incoming emails.  The answer, I was led to beleive,  was that the codec that the phone company uses is NOT  one that WM  recognizes (they do play OK  on Iphone).   So far, I have not figured  way to listen to them on my WinMobile device.  (there are conversion pgms,  but cant save,convert,then listen to each voicemail).     FYI

 

There was a mention of Codecs when using VOIP telephone systems – and how the codec affects the VOIP sound quality.

 

Since I am thinking of moving to a virtual pbx service (like vocalocity) and I hadn’t heard of this issue before, I thought I would check it out.

 

Having read over the information – I’m still not sure I understand which Codec is ideal and which works better for a medical office.

 

Anyway…Here is what Dr. Google says…

 

Codecs

Codecs are used to convert an analog voice signal to digitally encoded version. Codecs vary in the sound quality, the bandwidth required, the computational requirements, etc.

Each service, program, phone, gateway, etc typically supports several different codecs, and when talking to each other, negotiate which codec they will use.

=============================

http://voip.about.com/od/voipbasics/a/voipcodecs.htm

 

There are many codecs for audio, video, fax and text. Below is a list of the most common codecs for VoIP. As a user, you may think that you have little to do with what these are, but it is always good to know a minimum about these, since you might have to make decisions one day relating codecs concerning VoIP in your business; or at least might one day understand some words in the Greek VoIP people speak! I won’t drag you into all the technicalities of codecs, but will just mention them.

If you are a techie and want to know more about each one of these codecs in detail, have a look there.

Common VoIP Codecs

Codec

Bandwidth/kbps

Comments

G.711

64

Delivers precise speech transmission. Very low processor requirements. Needs at least 128 kbps for two-way.

G.722

48/56/64

Adapts to varying compressions and bandwidth is conserved with network congestion.

G.723.1

5.3/6.3

High compression with high quality audio. Can use with dial-up. Lot of processor power.

G.726

16/24/32/40

An improved version of G.721 and G.723 (different from G.723.1)

G.729

8

Excellent bandwidth utilization. Error tolerant. License required.

GSM

13

High compression ratio. Free and available in many hardware and software platforms. Same encoding is used in GSM cellphones (improved versions are often used nowadays).

iLBC

15

Robust to packet loss. Free

Speex

2.15 / 44

Minimizes bandwidth usage by using variable bit rate.

 

===========================

http://communication.howstuffworks.com/ip-telephony6.htm

VoIP: Codecs

VoIP software processes and routes the calls.

A codec, which stands for coder-decoder, converts an audio signal into compressed digital form for transmission and then back into an uncompressed audio signal for replay. It's the essence of VoIP.

Codecs accomplish the conversion by sampling the audio signal several thousand times per second. For instance, a G.711 codec samples the audio at 64,000 times a second. It converts each tiny sample into digitized data and compresses it for transmission. When the 64,000 samples are reassembled, the pieces of audio missing between each sample are so small that to the human ear, it sounds like one continuous second of audio signal. There are different sampling rates in VoIP depending on the codec being used:

64,000 times per second

32,000 times per second

8,000 times per second

A G.729A codec has a sampling rate of 8,000 times per second and is the most commonly used codec in VoIP.

Codecs use advanced algorithms to help sample, sort, compress and packetize audio data. The CS-ACELP algorithm (CS-ACELP = conjugate-structure algebraic-code-excited linear prediction) is one of the most prevalent algorithms in VoIP. CS-ACELP organizes and streamlines the available bandwidth. Annex B is an aspect of CS-ACELP that creates the transmission rule, which basically states " if no one is talking, don't send any data. " The efficiency created by this rule is one of the greatest ways in which packet switching is superior to circuit switching. It's Annex B in the CS-ACELP algorithm that's responsible for that aspect of the VoIP call.

The codec works with the algorithm to convert and sort everything out, but it's not any good without knowing where to send the data. In VoIP, that task is handled by soft switches.

==================================

http://www.ozvoip.com/codecs.php

Introduction to Codecs

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.

These pages attempt to demistify codecs and give a brief overview of the different codecs and when they are used. 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 others.

If you would like to read up more about codecs with respect to VoIP, the following links may be of interest:

Codecs page the VoIP Wiki

Find out which hardware supports which codecs.

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 each codec. If you would like to voice your opinion about a particular codec, or discuss the merits of one over another, feel free to do so in our voice over IP forums.

Codec

SamplingRate (kHz)

Bandwidth(kbps)

Nominal Bandwidth(kbps)

Payload Size(ms)

License

Comments

Pros

Cons

?

DVI4

unknown

unknown

unknown

Not a very common codec.

G.711

8

64

87.2

20

Open Source

G.711u/a often refered to as u-law/a-law: where a-law is the European version and u-law the US/Japanese version

Designed to deliver precise transmission of speech

Very low processing overheads

Including overheads, uses >64kbps, thus at least 128kbps bandwidth in each direction is required

G.722

16

48

unknown

Open Source

An ITU standard codec.

16

56

unknown

30

16

64

unknown

G.723.1

8

5.3

20.8

30

Proprietry

Often used by dialup VoIP users for optimal quality.

Very high compression whilst maintaining high quality audio.

Requires a lot of processor power.

8

6.3

21.9

30

G.726

8

16

unknown

Open Source

An improved version of G.721 and G.723 (totally different from G.723.1)

CPU overhead is relatively low for level of compression obtained.

8

24

47.2

20

8

32

55.2

20

8

40

unknown

G.728

unknown

16

31.5

Open Source

An ITU standard codec.

G.729

8

8

31.2

20

Patented

An ITU standard codec.

Excellent bandwidth utilisation for toll quality speech

Performs well under random bit errors

License required for use

GSM

8

13

unknown

Proprietry

Same encoding as used in GSM mobile phones (though improved version are often used nowadays).

Relatively high compression ratio.

Royalty free means it is available in many hardware and software platforms.

iLBC

unknown

13.33

unknown

30

Free to use

High robustness to packet loss

unknown

15

unknown

20

Siren

unknown

unknown

unknown

Not much known about this codec, and does not appear to be commonly supported.

Speex

8

unknown

unknown

Open Source

Uses variable bit rate to minimise bandwidth usage

16

unknown

unknown

32

unknown

unknown

Notes

The information provided here is for information purposes only, if you find errors or ommissions, please report them in the relevant discussion forum.

Bandwidth

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.

Sampling Rate

The sampling rate is the rate at which the analogue audio signal is sampled. Nyquist's Theorem states that in order to record a certain frequency, sampling must occur at at least twice that frequency. Thus, the higher the sampling rate, the greater the frequency range in the encoded audio stream. The human ear is capable of hearing from about 20Hz to about 20,000Hz. Typically, speech is around 100-4,000Hz. Thus, a sampling rate of at least 8kHz is required to accurately encode the human voice. Greater sampling rates will capture higher frequencies (this is useful, for example, if you are playing music down the phone), but will also increase bandwidth as there are more samples to encode and transmit.

Payload Size

The size of the payload of each encoded voice packet influences two things: lag and bandwidth. Every encoded packet that is sent incurs fixed bandwidth overheads (due to IP and other headers added to the data in the network). Thus, larger payloads incur a proportionately smaller overhead, thus reducing the nominal bandwidth utilisation. However, by using larger payloads, more audio (ie., a longer period of time) is required to construct a single packet, which in turn increases the amount of time it takes for even the beginning of the packet to reach the other end and be decoded, thus increasing the lag in the conversation. This is a typical trade-off in VoIP. Most codecs use payload sizes of 10-40ms.