Now, several issues come up here, we will we will take up many of the issues. Audio accuracy or quality Echo: That is what will be the part of our discussions that we will do in the in the next few minutes. This was the time at which i’th packet arrived and this was the time at which the i’th packet was transmitted. So, what are the, what are really the attributes of a low bit rate speech coders? So, while the packet loss can be contained by using forward error correcting codes, the delays even if there are packet delays of the order of to milliseconds depending upon the applications; if it is a streaming audio kind of applications, then larger delays like milliseconds may be acceptable.

Basic principles of Voice over IP Dr. Is it possible to have that acceptable voice communications? Network administrators must design a network to operate within an acceptable delay budget. But if it so happens that the probability that the signal amplitudes hovers between the low amplitude values if that is higher than the probability that it lies it in the high amplitude zones; then what maybe done is that we may resort to non uniform quantizations. So therefore, a trade-off needs to be achieved between the delay that would occur in accumulating the packets and the overheads that would be associated with the headers. So, this may actually introduce latency. So, this is a low pass filtered versions and as you can see here that alpha is a constant which lies between 0 and 1.

Fundamentals Series Analog vs.

So, as I just pointed So, d i is equal to r i minus t i was the delay of the i’th packet. A complete and systematic treatment of signal processing for VoIP voice. It will not affect or degrade the speech quality as much and therefore really speaking, somehow if the load on the internet is light, if it is not heavily loaded and if the packet losses can be contained to not a very significant fraction of the total packets transmitted; then it should be possible for us to have a very good intelligible speech without having appreciable degradations.

So, as a result the number of parameters or the number of bits that are required to be sent to the receiver decreases considerably and thereby we achieve a low bit rate speech coding.

Then the question is that we need some more information other than Introduction to VoIP Voice. So now, we will actually see that since the retransmission mechanism is not possible in the packet loss; how to address the problem of packet loss in the voice over IP conversation, how to address the problem of the delay jitter which occurs due to random queuing delay and different packets experiencing the different delays?


So, this way each packet can wait some amount of time, appropriate amount of time in the playout buffer and then the packets can be played out. Now, uniform quantization is actually the normal thing to do. Steven Holmes 3 years ago Views: The playout buffer is actually a buffer which is used at the receiver where the arriving packets are stored and then these packets are played out at an appropriate time. Introduction Before More information. Let us say that we are not using any compressions.

Recognizing Voice Over IP: Now, as you can see here that the periodicity with which the packets were generated at the transmitter, that periodicity is completely lost in the receiver here. Now, as you know that the problem of the delay jitter is that it happens because at the transmitter so and this can be of course through the internet and then we have the receiver. So, this is how d i hat is estimated.

Broadband Networks: Concepts and Technology

So, if these speech segments are given to the receiver for play out, then the speech quality will be definitely degraded. But even if there is some packet loss which is above this, then I have already pointed out that since retransmission mechanism is not suitable, the mechanism that will be used for voice over IP applications will be forward error correcting codes or FECS.

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.

Why we would like to use the the UDP as a protocol? Now, as I have already pointed out the principle of vocoder is speech synthesis through a model of the vocal tract. So now, suppose if some packets are lost, so how will you determine that it is the first packet if the packet has been lost? Jean-Yves Le Boudec Prof. Now, as you know that in any analog to digital speech conversions there are three steps that are required and these three steps are: This must be constrained to a defined bandwidth.

But apart from the bit rate, as we have already seen that the delay which will be there as the speech codec, the quality the intelligibity of the speech and the computational complexity are also important issues.


Either it is positive, then you can represent it by 1, if it is negative, you can represent it by 0. 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.

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

How is Voice Quality measured? Vu Thi Anh Nguyet 1 Outlines 1. Digitizing Analog Signals 1. Chapter 3 ATM and Multimedia Traffic In the middle of thethe telecommunications world started the design of a network technology that could act as a great unifier to support all digital services, including low-speed telephony and very More information.

In sampling, we sample the signal which is like discretization of the signal in the time domain and this sampling is done at Nyquist sampling rate and elcture the samples are obtained, we round them to the nearest level of quantization.

So, what are the forms of the audio codecs? Now, instead of then quantizing the samples themselves, what you do is that you encode the difference between the input samples metworks the predicted value and then the prediction coefficients are selected to minimize the prediction errors. However, amplitude the signal 4.

So, the solution for this is to use the playout buffer. Audio accuracy or quality Echo: In the middle of theprrof telecommunications world started the design of a network technology that could act as a great unifier to support all digital services, including low-speed telephony and very. To use this website, you must agree to our Privacy Policyincluding cookie policy. So, as a result you get a bit rate of into 8 bits per second which gives you 64 kilo bits per second.

So, thereby also you can result a reduction in the bit rate. So you can actually quantize not the incoming signals but the log of it and as a result you can get logarithmic quantizations. Summary By Maheshwar Jayaraman 1 1. The exam is closed book. So, what is alternative? What network mechanisms provide which kinds of quality assurances?