Now, uniform quantization is actually the normal thing to do. So, we will also look into the fact that what are the various compression mechanisms that are used for the transport of voice packets over the IP. So, really speaking, the step size you can adopt depending upon what are the voltage ranges between which the signal amplitude is likely to vary. Table of Contents Executive Overview Audio accuracy or quality Echo: Apart from transport mechanism, we would also need signaling mechanism to establish the sessions between the sender and the receivers.

So, what is done at the transmitter? This packet which was transmitted here, it arrived here. The exam is closed book. It may be possible to have the end to end delays smaller than milliseconds that may be possible to have smaller than milliseconds. During this 3-day course delegates will examine SIP technology and architecture and learn how a functioning VoIP service can be established. Now, as I have already pointed out the principle of vocoder is speech synthesis through a model of the vocal tract.

But in this kxrandikar, let me just briefly review some of the issues which are there in carrying the voice over IP in terms of analog to digital conversions. So, those are called waveform coders.

Digitizing Analog Signals 1.

NPTEL :: Electronics & Communication Engineering – Broadband Networks: Concepts and Technology

So, this packet which was transmitted at this time, it arrived here but it was played here. If the variance v i hat is estimated as 1 minus alpha v i minus 1 plus alpha into r i minus t i minus d i hat. To carry out this More information. And then, this bit is used by the decoder to increment or decrement the output by one step size. So, however as we have seen in practice, it may not be possible to do that and therefore moreover this value of d max may not be known at the receiver.

If d hat is kept equal to d max for a very large value, then it increases the latency. While this may not be a serious problem in the streaming networjs applications, actually this may become a significant issue in the two way interactive phone conversations. If it is kept close to 1, then this d i hat is more a reflection of the current delay.


Data continuous audio vs. The advantages are many, but because of the technology. So therefore, a trade-off needs to be achieved between the delay broadbqnd would occur in accumulating the packets and the overheads that would be associated with the headers.

Basic Components of a Telephony Network 8 A number of components. Now, as you see that the packet loss can occur because the network in a non QoS network. So, this is like what we are doing is that the delay d i hat is estimated using this equation. The question really is that what should be the value of this d hat that needs to be used. Indian Institute of Technology, Bombay.

Now, several issues come up here, we will we will take up many karandikra the issues. Automatic Call Distribution is a system used to determine how incoming calls are routed. So, let me just sketch brief block diagram of the voice over IP phone that may be present. Now, what is non uniform quantization?

Broadband Networks: Concepts and Technology

To make this website work, we log user data and share it with processors. If you really see that let us say networkd t i is the time of generation of prog packet, let us say that t i is time of generation of i’th packet and r i is the time of reception of i’th packet, then as you can see here that in the fixed playout buffer algorithm, in the fixed playout buffer algorithm, as you can see the packet p i will be played at the time t i plus d max where d max happens to be the maximum delay in the network.

Requirements of MN 4. Queuing discipline and scheduling Traffic. On other hand, if this alpha is close to 0, then this actually filters out any networls fluctuations in the delays. So, this is the quantization. So, let us look at first the problem of the delay jitter. Now, this will be typically the transport of the packet voice over the internet. Summary By Maheshwar Jayaraman 1 1. So, if these speech segments are given to the receiver for play xeries, then the speech quality will be definitely degraded.


Voiceis analog in character and moves in the form of waves.

So, when we do this, when we achieve the when we try to achieve the compressions; there are two issues which come up in the context of audio coding. Introduction to PCM encoding and decoding. A key reason for this is More information. So, as you can see here that sampling actually discretizes the signal in time domain and as from the Nyquist sampling rate that analog signals should be lecturw at a rate which is twice the highest frequency in the signal. So, if the i’th packet arrives such that r i happens to be larger than p i, then i th packet will be dropped.

Now obviously, you achieve a very great degree of compressions by using these vocoders.

So, this is like discretization this is discretizing the signal amplitude to fixed levels. So, that is why the packets are sent using UDP as a transport mechanisms.

So, at the receiver it may broadbaand some irritating feeling and therefore typically at the receiver, you will generate what is lectuge as the comfort noise. Basic principles of Voice over IP Dr. Just to give you an example of fixed playout buffer algorithms. Many data More information.

How is Voice Quality measured?