There is mainly two most inportant quanization techniques
a) Wave Codecs
b) Source Codecs
Wave Codecs
G.711 falls in wave codec category and takes bandwidth of 64Kbps approx by using Pulse Code Modulation technique(PCM). It is the same as described by Nyquist. G.726 uses 32,24 or 16 Kbps and uses Adapative Differential Pulse Code Modulation (ADPCM). When G.726 uses 32 Kbps during that time it takes 4 bits for error control and slips the reoccurence bits, for 24 Kbps it uses 3 bit and for 16 Kbps it uses 2 bits. Definately during the less error control bits increases the error rate because you can skips those bits also which need to be used. This is aka quantization error.
Source Codecs
Sorce codecs are designed to work with human voice. Like cisco G.729 takes 8Kbps of bandwidth to deliver voice. Now the question comes in mind why it takes so less? Actually it is having in built codebook in the database and on the basics of frequeny it matches the code and place the code in a packet. Lets assume if you want to say CISCO, it will go like "ccceessccoo" and it is having built in binary for the words. Thats why it eats less bandwidth rather than other protocol.
As move in the less bandwidth protocols the delay need to be considered because it always increases. So one can say bandwidth is inversely proportional to delay.
regards
shivlu jain
SDN and NFV is the next phase of technology change which will help service provider to launch the services in single click. This is all about the programmability of the networks by using open source software defined network controller.
Thursday, May 21, 2009
Quantization Techniques In Voip Network
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2 comments:
Nice post!
Just few notes:
G.729 is not a Cisco standard. It's an ITU standard for digital voice compression.
But G.729 is fairly old, there are new emerging codecs like the open source speex or iLBC (ilBC is now supported as wideband {HQ} codec on some newer Cisco IP Phones).
btw: There is one bad thing about all these low bandwidth stuff, even tho the codec can generate raw 8kbps digital compressed voice stream, this stream must be encapsulated before the actual transmission over an IP network. And usually the overhead from the encapsusation is even higher that the payload itself :-/ RTP header and IP header compression can in some situations help with this issue.
Hi Jozef
Thanks for adding more to it. As far my knowledge is concerned, G.729 is cisco proprietry. Any how I am new to voice and need to dig more.
From you comments it seems like that you are having much experience in voip network.
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