VoIP: Coder Decoder

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If you want to know how a voice produced by a codec sounds like, you can listen to a 8 Kbps mono WAV, which can be downloaded from http://www.signalogic.com/index.pl?page=codec_samples. Question often asked when using this service is how much bandwidth is cunsumed. When idle, the system does not need large amount of bandwidth, but when you communicate, the bandwidth needed could be as big as 32 kbps up and 32 kbps down per channel (1 communication session), depending which codec is being used.

There is a number of codecs used:

Codec Bandwidth consumption
GIPS 13.3 Kbps or higher
GSM 13 Kbps (full rate), 20 ms frame size
iLBC 15 Kbps, 20 ms frame size, 13.3 Kbps, 30 ms frame size
ITU G.711 64 Kbps, sample-based is also known as alaw/ulaw
ITU G.722 48/56/64 Kbps
ITU G.723.1 5.3/6.3 Kbps, 30ms frame size
ITU G.726 16/24/32/40 Kbps
ITU G.728 16 Kbps
ITU G.729 8 Kbps, 10 ms frame size
Speex 2.15 to 44.2 Kbps
LPC10 2.5 Kbps
DoD CELF 4.8 Kbps

The next question would be which codec is the most suitable for a provider? The answer depends on the amount of bandwidth you have. If you have a maximum bandwidth of 32 Kbps both up and down for a VoIP traffic, it is recommended that you use GSM or iLBC as your codec. On the other hand, if the amount of bandwidth is higher, say, higher than 128 Kbps, you can use G711u (PCMU), which will increase the voice quality in a communication session, with clearer voice and lower delay. Other codec that could produce optimal result is the G.729 Codec. Unfortunately, it is a proprietary codec which is not favourable for those who use open source platform.

The most commonly used codecs are the G.729, GSM, and G.711. Of these, the G.711 is favorable as it delivers good quality in LAN network, GSM is preferred by open source users as GSM is not copyrighted, while many VoIP devices use G.729 for their codec, the one which is copyrighted.

[edit] Pranala Menarik

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