1. Tutorial 12
Solutions

2. Q.1
In the Internet phone example in the lecture notes, let h be the total number of header bytes added to each chunk, including UDP and IP header.
a. Assuming an IP datagram is emitted every 20 msecs, find the transmission rate in bits per second for the datagrams generated by one side of this application.
b. What is a typical value of h when RTP is used?

3. a) PCM-encoded voice: 64kbps=8000 bytes/s
8000 bytes/s * 20 msec/chunk = 160 bytes/chunk
(160 + h) bytes / datagram
1s / (20 msec) = 50 datagrams/s
Transmission rate = 50 * (160+h) bytes/s
= 0.4 * (160 + h) kbps
= ( h) kbps

4. b) IP header: 20 bytes
UDP header: 8 bytes
RTP header: 12 bytes
h = 40 bytes

5. Q.2
Consider the adaptive playout strategy described in the lecture notes
a) How can two successive packets received at the destination have time-stamps that differ by more than 20 msecs when the two packets belong to the same talk spurt?
b) How can the receiver use sequence numbers to determine whether a packet is the first packet in a talk spurt?

6. a). If there is packet lost, then other packets may have been transmitted in between the two received packets.

7. b). Let Si denote the sequence number of the ith received packet
b). Let Si denote the sequence number of the ith received packet. If ti – ti-1 > 20 msec and Si = Si-1 + 1then packet i begins a new talkspurt

8. Q.3
Recall the two FEC schemes for Internet phone described in the lecture notes. Suppose the first scheme generates a redundant chunk for every four original chunks. Suppose the second scheme uses a low-bit rate encoding whose transmission rate is 25 percent of the transmission rate of the nominal stream.

9. Question 3 a) How much additional bandwidth does each scheme require?
Both schemes require 25% more bandwidth.

10. b) How do the two schemes perform if the first packet is lost in every group of five packets? Which scheme will have better audio quality?
The first scheme will be able to reconstruct the original high-quality audio encoding. The second scheme will use the low quality audio encoding for the lost packets and will therefore have lower overall quality.

11. c) How do the two schemes perform if the first packet is lost in every group of two packets? Which scheme will have better audio quality?
For the first scheme, many of the original packets will be lost and audio quality will be very poor. For the second scheme, every audio chunk will be available at the receiver, although only the low quality version will be available for every other chunk. Audio quality will be acceptable.

12. Q.4
Consider an RTP session consisting of four users, all of which are sending and receiving RTP packets into the same multicast address. Each user sends video at 100 kbps.

13. a) RTCP will limit its traffic rate to what rate?
The session bandwidth is 4* 100 kbps = 400 kbps. Five percent of the session bandwidth is 20 kbps.

14. b) A particular receiver will be allocated how much RTCP bandwidth?
c) A particular sender will be allocated how much RTCP bandwidth?
Each user is both a sender and receiver, each user gets 5 kbps for RTCP packets (receiver reports, sender reports, and source description packets).