Bandwidth VS. Codec 101

Discussion in '3CX Phone System - General' started by cjammer, May 23, 2007.

  1. cjammer

    cjammer New Member

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    What bandwidth does each codec take? For a low speed circuit what would be the best Codec to use and why? Any info on codecs will be a BIG help.

    Thanks,

    Cjammer
     
  2. Costas3CX

    Costas3CX New Member

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    well from my understanding out of the codecs 3cx uses, GSM is the least bandwidth demanding at 16kbps or is it 13? Not too sure right now. The other two we are using are at 64kbps.
     
  3. cjammer

    cjammer New Member

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    Does one have better quality over the other?

    Cjammer
     
  4. Anonymous

    Anonymous Guest

    G711
    Sampling Rate: 8 Khz
    Bandwidth: 64 kbps
    Nominal Bandwidt: 87.2 kbps
    Payload size: 20 ms

    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

    The good:
    Designed to deliver precise transmission of speech
    Very low processing overheads

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


    GSM

    Sampling Rate: 8 Khz
    Bandwidth: 13 kbps
    Nominal Bandwidt: ? kbps
    Payload size: ? ms

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

    The good:
    Relatively high compression ratio.

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


    Often mentioned but not used by 3cx is the G729

    Sampling Rate: 8 Khz
    Bandwidth: 8 kbps
    Nominal Bandwidt: 31.2 kbps
    Payload size: 20 ms

    The good:
    Excellent bandwidth utilisation for toll quality speech
    Performs well under random bit errors

    The bad:
    License required.

    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.


    Well there you have it :).
     
  5. cjammer

    cjammer New Member

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    Thanks!!!!

    That is REALLY good information. That fills in a lot of the gaps I needed.

    This should be high on everyone's reading list that is new to VoIP

    Thanks Again!

    Cjammer
     

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