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Cisco Voice over IP - Part 1 - ciscoman2010@gmail.com

Hi Guys,


This doc is the continuation of "Cisco Voice over IP - ciscoman2010@gmail.com"


How to turn your voice in to bits ?


Do you ever wondered how your voice you speak in telephone ( IP world ) converted in to bits ?


It's a 4 step process which involves,


Take many samples of analog signals, as you say oooooooohhhhhhhhhhhh, see if u turn this in to a graph, it's a smooth analog waveform,

starting from zero to a high value then decrement back to zero and go to negative value tn come back to zero as well, :-)


Then calculate the no representing each samples, just may the analog wave to y axis value ( Familiar with Communication guys )

which is normally called as Quantization.


Convert that no to binary.


Compress the signal to make the bandwidth more efficient.


Have a look at few fact ?


The frequency chart is given below


Human Ear can hear a frequency - 20 to 20000 Hz

Human Speech - 200 to 9000 Hz

Telephone channel - 300 to 3400 Hz

Nyquist theorem - 300 to 4000 Hz


As per the Nyquist theorem, If you sample a signal at regular intervals of atleast twice the channel frequency, the sample will

contain enough information to accurately reconstruct that signal.


So we can have a math calculation - Nyquist highest freq is 4000 so 4000*2 = 8000

ie. we need to sample 8000 times a sec, each is 1 byte of info

so 8000*8 = 64000Kbps :-) which is equiv to 1 voice call (uncompressed audio)


Codec is the main term we speak in the industry for audio compression


G.711 uses PCM - Pulse Code Modulation for 64kbps


G.726 - ADPCM - Adaptive Differential Pulse Code Modulation take up to 32/24/16Kbps depending up of the algorithm it use.

which means, think u say AAAAhhhhh, instead of fully sending full samples i can take few of it then predict the other

It's a bit notation which u use, if i take last 4  bits it end up 32.

last 3 bits end up 24.

last 2 bits end up 16.


How to conclude which codec is good ?


Based on MoS Score we can conclude the quality of sound based on codec's

Higher the bit rate lower the delay and vice versa

G.711 has a MoS score of 4.1 with delay 0.75ms and bit rate of 64Kbps - Milliseconds

G.729 has a MoS score of 3.9 with delay of 10ms and bit rate of 8Kbps - Milliseconds


What's the function of DSP or Digital signal Processing ?


Normally this DSP cards can be installed on Router to do the following functions

Analog to Digital conversions, conferencing features,echo cancellation, Voice Activity Detection etc...


Voice Protocols that take over the IPNGN?


RTP - Real time transport protocol mainly used for reordering of voice packets & time stamping function


UDP - Multiplexing multiple streams in to one based on the port numbers.


Best Regards,

Pramod

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