Codec Processing
We already mention that analog audio is turned into packets for VoIP transmission. This transmission of packet is called codec. Codec stands for `compressor-decompressor'.
Reducing bandwidth consumption is a major task handled in VoIP communication. Your digitized voice may require huge space (bits or bytes) for transmission over internet. The solution is compressing the digitized data into a compact format or smaller size.
Think about the song you use to listen on your mp3 player. MP3 is a compressed format of original audio or song file. But, this mp3 is very good to listen to, compared to huge pure audio files. That’s why to store more song in your device memory, you listen mp3 songs.
Codec task of VoIP do the same thing, it compress your digitized voice by using proper compression algorithm. This compressed data is sent over the network and once it reaches Media Gateway Controller server; the softswitch convert the voice from one codec to the other if it’s needed.
Suppose your call has buzzed on a softswitch server and it needs to be routed to a UK based media gateway which does not support or understand your voice codec. The softswitch will translate the voice codec to a supported codec of UK media gateway before routing. Like different audio codec ex. mp3, wma; there are several codec used by VoIP:
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Codec Name
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Bandwidth (Kbps)
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Description
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G.711
|
64
|
Delivers High quality speech transmission
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|
G. 722
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48/56/64
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Adaptable with network congestion
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|
G.723.1
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5.3 0r 6.3
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High compression with high quality
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|
G.726
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16/24/32/40
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Improved version before G.723.1
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|
G.729
|
8
|
For optimized bandwidth utilization
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|
iLBC
|
15
|
Deliver less Packet loss
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