study a Voice-over-IP (VoIP) application | My Assignment Tutor

Hello. I am in need of help with this exercise; that is I need answers to the questions. I messed up and didn’t see exercise before 20 minutes ago, and the deadline is in 1 hour 30 minutes. I need answers to the questions, but I dont need an illustration on the “make an illustration” questions. I can make those myself, but it would be nice with some notes on how it should be constructed, or a quick draft (but not a must!). I have uploaded two pages from the textbook about skype (referenced in exercise). — In this exercise, you will study a Voice-over-IP (VoIP) application. The application is similar to the example in chapter 9.3 in 7th edition (7.3 in 6th edition). Even though many VoIP applications like Skype no longer provide host-to-host services, this exercise helps generate a general understanding of different concepts in multimedia networking. The goal is to learn about packet loss, loss recovery schemes, delay, jitter and RTP. To pass this exercise, you are expected to show an understanding of the concepts and theory of VoIP by answering the questions. Simple, numerical answers will not be accepted on their own. In this application, the sender uses normal RTP and transmits G.722-encoded voice at 48 Kbps. The application collects encoded data in 16 millisecond chunks. Question 1. What is the rate at which data is generated at the sender (in bytes)? Question 2. What is the size of the IP datagrams sent? You must clearly show the steps in how you calculated your answer and what elements the datagram consists of. Question 3. Explain how an arbitrary RTP packet in the application will look like. Use actual values from the application when possible. Include the size of the fields and elements of the packet. Loss recovery schemes can be used to preserve acceptable audio quality in the presence of packet loss. We will focus on three types of loss anticipation schemes: forward error correction (FEC) with redundant encoded chunks, FEC with redundant lower-resolution audio stream, and interleaving. Question 4. For each of the three schemes listed in the previous paragraph, show how much additional bandwidth each of them require of our application. For the first type of FEC, suppose a redundant chunk is generated for every five original chunks. For the other type, suppose GSM is used for the low-bit rate encoded stream. For each scheme, include both the new transmission rate and the percentage increase Question 5. For each of the three schemes, explain what happens if the first packet is lost in every group of five packets. Which scheme will have the better audio quality? Question 7. For each of the three schemes, explain what happens if the first packet is lost in every group of two packets. Which scheme will have the better audio quality? Question 8. Make an illustration of the scenario in question 7 for FEC with redundant lowerresolution audio stream. The figure should be made such that it’s clear that the student understands how the scheme works Question 9. For each of the FEC schemes, how much playback delay does each scheme add? What can be said about the delay of the interleaving scheme?

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