Voip Codecs

Licenced & Open Source Voice Codecs in IP Telephony (VOIP) – A brief Introduction
In this article, we will see why its important to select the right voice codec for IP Telephony (VOIP) implementations and a short introduction to some of the popular voice codec’s – G.711, G.729, G,723.1, iLBC, Speex & GSM codec. These are narrow band codec’s.
What is a Voice Codec?
A codec digitallyencodes and compresses analog audio signals using complex mathematical models. A codec’s primary role is to seek a balance between the transmission efficiency (bandwidth) and the quality of voice signals. That is to say, transmitting the best quality of digital voice signals at the lowest bandwidth possible.
A codec refers to both coder/ decoder & compression/ decompression.
Why is a VoiceCodec required?
A voice codec is required mainly for compressing the digital voice signals so that they can be transmitted across IP networks (including lossy networks like the Internet) with least possible bandwidth and affordable quality. When codec’s are used in IP Telephony, they generally introduce processing delays as complex mathematical formulas are used for encoding/compression are CPUintensive.
Codec’s are especially required in large IP Telephony/ multi-location VOIP installations as each call on the IP network consumes a certain amount of bandwidth and the total bandwidth consumed for uncompressed voice packets could be enormous.
If you are going with a default IP Telephony installation, there is a good chance that you might be dealing with G.711 Voice Codec for all your calls.But its important to note that, on larger/ multi-site VOIP installations it might be better to go with one of the licensed codec’s or even the open-source codec’s that are mentioned below (even if only partially). These codec’s make a considerable difference to the overall voice quality and bandwidth consumption for IP Telephony implementations.
Some techniques used to compress the digital voicedata:
* VAD – Voice Activity Detection: In IP Telephony, both the conversations as well as the silence in between the conversations are digitized. So, we have both – packets containing voice as well as packets containing silence. Using VAD, packets of silence can be discarded after their duration is appropriately marked. So, the total number of packets transmitted after compression is lesser(generally around 30% lesser).
* CNG – Comfort Noise Generation: This is not a compression technique, but when voice is compressed using VAD, the awkward silence between the speech might be interpreted as lost connection and hence white noise is generated locally at both ends using CNG. This makes the call appear connected to both the parties during silence, as some background noise is audibleduring that duration.
* CELP – Code Excited Linear Prediction: In this method, various human sounds are mathematically modeled and a code book of all possible sounds is produced. So, instead of sending the actual sound packets across, only their codes are sent across. This is a very simplistic explanation, and a lot more techniques are involved.
* In some compression techniques, theheaders can be compressed separately like the payload compression which provides additional bandwidth efficiency while transmission.
These are some methods by which voice is compressed (to be decompressed at the other end), and they are by no means comprehensive. But this is given here to get a hang of how voice compression is done in digital networks.
G.711:
G.711 is considered to be the base codecin IP Telephony. It gives the highest quality of voice but takes up largest possible bandwidth. A small compression is provided by a technique called companding, and for this codec CPU utilization is least. This technique is recommended for smaller IP Telephony implementations and in places with large network backbone bandwidths where high quality voice is required. This could also be used in...