Consider a real-time audio stream sent from host A to host B. Assume that at time 0, A starts sending audio packets at a rate of one packet every 10 milliseconds with the audio encoded using 64 Kb/s PCM.

a) How many audio samples are contained in each packet?

b) Suppose that B uses adaptive playout with parameter K=4 and that before the start of the talk spurt, the average delay measured at the receiver is 100 ms, and the average delay variation is 10 ms. Suppose that at time 170 ms, B receives a packet that was sent by A at time 50 ms. At what time will B start to play out the audio data contained in this packet? At what time will it finish playing the audio data from this packet?

c) Suppose that the audio packets are sent using UDP and RTP across an Ethernet network and that the RTP header of 12 bytes. Approximately what fraction of each packet is actual audio data (as opposed to protocol fields). Account for all protocol fields, but you may ignore the Ethernet preamble and flag fields.

d) Suppose that the audio data is sent using RTP and the timestamp in the first RTP packet of the talk spurt is 30. What is the value of the timestamp in the second RTP packet?

please provide steps.